freeswitch/src/mod/endpoints/mod_sofia/mod_sofia.c

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Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
/*
* FreeSWITCH Modular Media Switching Software Library / Soft-Switch Application
* Copyright (C) 2005/2006, Anthony Minessale II <anthmct@yahoo.com>
*
* Version: MPL 1.1
*
* The contents of this file are subject to the Mozilla Public License Version
* 1.1 (the "License"); you may not use this file except in compliance with
* the License. You may obtain a copy of the License at
* http://www.mozilla.org/MPL/
*
* Software distributed under the License is distributed on an "AS IS" basis,
* WITHOUT WARRANTY OF ANY KIND, either express or implied. See the License
* for the specific language governing rights and limitations under the
* License.
*
* The Original Code is FreeSWITCH Modular Media Switching Software Library / Soft-Switch Application
*
* The Initial Developer of the Original Code is
* Anthony Minessale II <anthmct@yahoo.com>
* Portions created by the Initial Developer are Copyright (C)
* the Initial Developer. All Rights Reserved.
*
* Contributor(s):
*
* Anthony Minessale II <anthmct@yahoo.com>
* Ken Rice, Asteria Solutions Group, Inc <ken@asteriasgi.com>
* Paul D. Tinsley <pdt at jackhammer.org>
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
*
*
* mod_sofia.c -- SOFIA SIP Endpoint
*
*/
/* Best viewed in a 160 x 60 VT100 Terminal or so the line below at least fits across your screen*/
/*************************************************************************************************************************************************************/
/*Defines etc..*/
/*************************************************************************************************************************************************************/
#define HAVE_APR
#include <switch.h>
struct outbound_reg;
typedef struct outbound_reg outbound_reg_t;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
struct sofia_profile;
typedef struct sofia_profile sofia_profile_t;
#define NUA_MAGIC_T sofia_profile_t
struct sofia_private {
switch_core_session_t *session;
outbound_reg_t *oreg;
};
typedef struct sofia_private sofia_private_t;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
struct private_object;
typedef struct private_object private_object_t;
#define NUA_HMAGIC_T sofia_private_t
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
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#define MY_EVENT_REGISTER "sofia::register"
#define MY_EVENT_EXPIRE "sofia::expire"
#define MULTICAST_EVENT "multicast::event"
#define SOFIA_REPLACES_HEADER "_sofia_replaces_"
#define SOFIA_USER_AGENT "FreeSWITCH(mod_sofia)"
#define SOFIA_CHAT_PROTO "sip"
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
#include <sofia-sip/nua.h>
#include <sofia-sip/sip_status.h>
#include <sofia-sip/sdp.h>
#include <sofia-sip/sip_protos.h>
#include <sofia-sip/auth_module.h>
#include <sofia-sip/su_md5.h>
#include <sofia-sip/su_log.h>
#include <sofia-sip/nea.h>
extern su_log_t tport_log[];
static switch_frame_t silence_frame = { 0 };
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
static char silence_data[13] = "";
static char reg_sql[] =
"CREATE TABLE sip_registrations (\n"
" user VARCHAR(255),\n"
" host VARCHAR(255),\n"
" contact VARCHAR(1024),\n"
" status VARCHAR(255),\n"
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
" rpid VARCHAR(255),\n"
" expires INTEGER(8)"
");\n";
static char sub_sql[] =
"CREATE TABLE sip_subscriptions (\n"
" proto VARCHAR(255),\n"
" user VARCHAR(255),\n"
" host VARCHAR(255),\n"
" sub_to_user VARCHAR(255),\n"
" sub_to_host VARCHAR(255),\n"
" event VARCHAR(255),\n"
" contact VARCHAR(1024),\n"
" call_id VARCHAR(255),\n"
" full_from VARCHAR(255),\n"
" full_via VARCHAR(255),\n"
" expires INTEGER(8)"
");\n";
static char auth_sql[] =
"CREATE TABLE sip_authentication (\n"
" user VARCHAR(255),\n"
" host VARCHAR(255),\n"
" passwd VARCHAR(255),\n"
" nonce VARCHAR(255),\n"
" expires INTEGER(8)"
");\n";
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
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static const char modname[] = "mod_sofia";
#define STRLEN 15
static switch_memory_pool_t *module_pool = NULL;
#define set_param(ptr,val) if (ptr) {free(ptr) ; ptr = NULL;} if (val) {ptr = strdup(val);}
#define set_anchor(t,m) if (t->Anchor) {delete t->Anchor;} t->Anchor = new SipMessage(m);
/* Local Structures */
/*************************************************************************************************************************************************************/
struct sip_alias_node {
char *url;
nua_t *nua;
struct sip_alias_node *next;
};
typedef struct sip_alias_node sip_alias_node_t;
typedef enum {
PFLAG_AUTH_CALLS = (1 << 0),
PFLAG_BLIND_REG = (1 << 1),
PFLAG_AUTH_ALL = (1 << 2),
PFLAG_FULL_ID = (1 << 3),
PFLAG_PRESENCE = (1 << 4)
} PFLAGS;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
typedef enum {
TFLAG_IO = (1 << 0),
TFLAG_CHANGE_MEDIA = (1 << 1),
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
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TFLAG_OUTBOUND = (1 << 2),
TFLAG_READING = (1 << 3),
TFLAG_WRITING = (1 << 4),
TFLAG_HUP = (1 << 5),
TFLAG_RTP = (1 << 6),
TFLAG_BYE = (1 << 7),
TFLAG_ANS = (1 << 8),
TFLAG_EARLY_MEDIA = (1 << 9),
TFLAG_SECURE = (1 << 10),
TFLAG_VAD_IN = ( 1 << 11),
TFLAG_VAD_OUT = ( 1 << 12),
TFLAG_VAD = ( 1 << 13),
TFLAG_TIMER = (1 << 14),
TFLAG_READY = (1 << 15),
TFLAG_REINVITE = (1 << 16),
TFLAG_REFER = (1 << 17),
TFLAG_NOHUP = (1 << 18),
TFLAG_XFER = (1 << 19),
TFLAG_NOMEDIA = (1 << 20),
TFLAG_BUGGY_2833 = (1 << 21),
TFLAG_SIP_HOLD = (1 << 22),
TFLAG_RWLOCK = (1 << 23)
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
} TFLAGS;
static struct {
switch_hash_t *profile_hash;
switch_mutex_t *hash_mutex;
uint32_t callid;
int32_t running;
switch_mutex_t *mutex;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
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} globals;
typedef enum {
REG_FLAG_AUTHED = (1 << 0),
} reg_flags_t;
typedef enum {
REG_STATE_UNREGED,
REG_STATE_TRYING,
REG_STATE_REGISTER,
REG_STATE_REGED,
REG_STATE_FAILED,
REG_STATE_EXPIRED
} reg_state_t;
struct outbound_reg {
sofia_private_t sofia_private;
nua_handle_t *nh;
sofia_profile_t *profile;
char *name;
char *register_scheme;
char *register_realm;
char *register_username;
char *register_password;
char *register_from;
char *register_to;
char *register_proxy;
char *expires_str;
uint32_t freq;
time_t expires;
time_t retry;
uint32_t flags;
reg_state_t state;
switch_memory_pool_t *pool;
struct outbound_reg *next;
};
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
struct sofia_profile {
int debug;
char *name;
char *dbname;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
char *dialplan;
char *context;
char *extrtpip;
char *rtpip;
char *sipip;
char *extsipip;
char *username;
char *url;
char *sipdomain;
char *timer_name;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
int sip_port;
char *codec_string;
char *codec_order[SWITCH_MAX_CODECS];
int codec_order_last;
int running;
int codec_ms;
int dtmf_duration;
unsigned int flags;
unsigned int pflags;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
uint32_t max_calls;
nua_t *nua;
switch_memory_pool_t *pool;
su_root_t *s_root;
sip_alias_node_t *aliases;
switch_payload_t te;
uint32_t codec_flags;
switch_mutex_t *ireg_mutex;
switch_mutex_t *oreg_mutex;
outbound_reg_t *registrations;
su_home_t *home;
switch_hash_t *profile_hash;
switch_hash_t *chat_hash;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
};
struct private_object {
sofia_private_t sofia_private;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
uint32_t flags;
switch_payload_t agreed_pt;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
switch_core_session_t *session;
switch_frame_t read_frame;
const switch_codec_implementation_t *codecs[SWITCH_MAX_CODECS];
int num_codecs;
switch_codec_t read_codec;
switch_codec_t write_codec;
uint32_t codec_index;
uint32_t codec_ms;
switch_caller_profile_t *caller_profile;
int32_t timestamp_send;
int32_t timestamp_recv;
switch_rtp_t *rtp_session;
int ssrc;
switch_time_t last_read;
sofia_profile_t *profile;
char *local_sdp_audio_ip;
switch_port_t local_sdp_audio_port;
char *remote_sdp_audio_ip;
switch_port_t remote_sdp_audio_port;
char *adv_sdp_audio_ip;
switch_port_t adv_sdp_audio_port;
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
char *proxy_sdp_audio_ip;
switch_port_t proxy_sdp_audio_port;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
char *from_uri;
char *to_uri;
char *from_address;
char *to_address;
char *callid;
char *far_end_contact;
char *contact_url;
char *from_str;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
char *rm_encoding;
char *rm_fmtp;
char *fmtp_out;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
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char *remote_sdp_str;
char *local_sdp_str;
char *dest;
char *key;
char *xferto;
char *kick;
char *origin;
char *hash_key;
char *chat_from;
char *chat_to;
char *e_dest;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
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unsigned long rm_rate;
switch_payload_t pt;
switch_mutex_t *flag_mutex;
switch_payload_t te;
nua_handle_t *nh;
nua_handle_t *nh2;
su_home_t *home;
sip_contact_t *contact;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
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};
/* Function Prototypes */
/*************************************************************************************************************************************************************/
static switch_status_t sofia_on_init(switch_core_session_t *session);
static switch_status_t sofia_on_hangup(switch_core_session_t *session);
static switch_status_t sofia_on_loopback(switch_core_session_t *session);
static switch_status_t sofia_on_transmit(switch_core_session_t *session);
static switch_status_t sofia_outgoing_channel(switch_core_session_t *session, switch_caller_profile_t *outbound_profile,
switch_core_session_t **new_session, switch_memory_pool_t *pool);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
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static switch_status_t sofia_read_frame(switch_core_session_t *session, switch_frame_t **frame, int timeout,
switch_io_flag_t flags, int stream_id);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
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static switch_status_t sofia_write_frame(switch_core_session_t *session, switch_frame_t *frame, int timeout,
switch_io_flag_t flags, int stream_id);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
static switch_status_t config_sofia(int reload);
static switch_status_t sofia_kill_channel(switch_core_session_t *session, int sig);
static switch_status_t activate_rtp(private_object_t *tech_pvt);
static void deactivate_rtp(private_object_t *tech_pvt);
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
static void set_local_sdp(private_object_t *tech_pvt, char *ip, uint32_t port, char *sr, int force);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
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static void tech_set_codecs(private_object_t *tech_pvt);
static void attach_private(switch_core_session_t *session,
sofia_profile_t *profile,
private_object_t *tech_pvt,
char *channame);
static void terminate_session(switch_core_session_t **session, switch_call_cause_t cause, int line);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
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static switch_status_t tech_choose_port(private_object_t *tech_pvt);
static void do_invite(switch_core_session_t *session);
static uint8_t negotiate_sdp(switch_core_session_t *session, sdp_session_t *sdp);
static char *get_auth_data(char *dbname, char *nonce, char *npassword, size_t len, switch_mutex_t *mutex);
static void establish_presence(sofia_profile_t *profile);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
static void sip_i_state(int status,
char const *phrase,
nua_t *nua,
sofia_profile_t *profile,
nua_handle_t *nh,
sofia_private_t *sofia_private,
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
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sip_t const *sip,
tagi_t tags[]);
static void sip_i_refer(nua_t *nua,
sofia_profile_t *profile,
nua_handle_t *nh,
sofia_private_t *sofia_private,
sip_t const *sip,
tagi_t tags[]);
static void sip_i_info(nua_t *nua,
sofia_profile_t *profile,
nua_handle_t *nh,
switch_core_session_t *session,
sip_t const *sip,
tagi_t tags[]);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
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static void sip_i_invite(nua_t *nua,
sofia_profile_t *profile,
nua_handle_t *nh,
sofia_private_t *sofia_private,
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
sip_t const *sip,
tagi_t tags[]);
static void sip_i_register(nua_t *nua,
sofia_profile_t *profile,
nua_handle_t *nh,
sofia_private_t *sofia_private,
sip_t const *sip,
tagi_t tags[]);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
static void event_callback(nua_event_t event,
int status,
char const *phrase,
nua_t *nua,
sofia_profile_t *profile,
nua_handle_t *nh,
sofia_private_t *sofia_private,
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
sip_t const *sip,
tagi_t tags[]);
static void *SWITCH_THREAD_FUNC profile_thread_run(switch_thread_t *thread, void *obj);
static void launch_profile_thread(sofia_profile_t *profile);
static switch_status_t config_sofia(int reload);
static switch_status_t chat_send(char *proto, char *from, char *to, char *subject, char *body, char *hint);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
/* BODY OF THE MODULE */
/*************************************************************************************************************************************************************/
typedef enum {
AUTH_OK,
AUTH_FORBIDDEN,
AUTH_STALE,
} auth_res_t;
static char *get_url_from_contact(char *buf, uint8_t dup)
{
char *url = NULL, *e;
if ((url = strchr(buf, '<')) && (e = strchr(url, '>'))) {
url++;
if (dup) {
url = strdup(url);
e = strchr(url, '>');
}
*e = '\0';
}
return url;
}
static auth_res_t parse_auth(sofia_profile_t *profile, sip_authorization_t const *authorization, char *regstr, char *np, size_t nplen)
{
int index;
char *cur;
su_md5_t ctx;
char uridigest[2 * SU_MD5_DIGEST_SIZE + 1];
char bigdigest[2 * SU_MD5_DIGEST_SIZE + 1];
char *nonce, *uri, *qop, *cnonce, *nc, *response, *input = NULL, *input2 = NULL;
auth_res_t ret = AUTH_OK;
char *npassword = NULL;
int cnt = 0;
nonce = uri = qop = cnonce = nc = response = NULL;
if (authorization->au_params) {
for(index = 0; (cur=(char*)authorization->au_params[index]); index++) {
char *var, *val, *p, *work;
var = val = work = NULL;
if ((work = strdup(cur))) {
var = work;
if ((val = strchr(var, '='))) {
*val++ = '\0';
while(*val == '"') {
*val++ = '\0';
}
if ((p = strchr(val, '"'))) {
*p = '\0';
}
if (!strcasecmp(var, "nonce")) {
nonce = strdup(val);
cnt++;
} else if (!strcasecmp(var, "uri")) {
uri = strdup(val);
cnt++;
} else if (!strcasecmp(var, "qop")) {
qop = strdup(val);
cnt++;
} else if (!strcasecmp(var, "cnonce")) {
cnonce = strdup(val);
cnt++;
} else if (!strcasecmp(var, "response")) {
response = strdup(val);
cnt++;
} else if (!strcasecmp(var, "nc")) {
nc = strdup(val);
cnt++;
}
}
free(work);
}
}
}
if (cnt != 6) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Invalid Authorization header!\n");
goto end;
}
if (switch_strlen_zero(np)) {
if (!get_auth_data(profile->dbname, nonce, np, nplen, profile->ireg_mutex)) {
ret = AUTH_STALE;
goto end;
}
}
npassword = np;
if ((input = switch_mprintf("%s:%q", regstr, uri))) {
su_md5_init(&ctx);
su_md5_strupdate(&ctx, input);
su_md5_hexdigest(&ctx, uridigest);
su_md5_deinit(&ctx);
}
if ((input2 = switch_mprintf("%q:%q:%q:%q:%q:%q", npassword, nonce, nc, cnonce, qop, uridigest))) {
memset(&ctx, 0, sizeof(ctx));
su_md5_init(&ctx);
su_md5_strupdate(&ctx, input2);
su_md5_hexdigest(&ctx, bigdigest);
su_md5_deinit(&ctx);
if (!strcasecmp(bigdigest, response)) {
ret = AUTH_OK;
} else {
ret = AUTH_FORBIDDEN;
}
}
end:
if (input) {
switch_safe_free(input);
}
if (input2) {
switch_safe_free(input2);
}
switch_safe_free(nonce);
switch_safe_free(uri);
switch_safe_free(qop);
switch_safe_free(cnonce);
switch_safe_free(nc);
switch_safe_free(response);
return ret;
}
static void execute_sql(char *dbname, char *sql, switch_mutex_t *mutex)
{
switch_core_db_t *db;
if (mutex) {
switch_mutex_lock(mutex);
}
if (!(db = switch_core_db_open_file(dbname))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error Opening DB %s\n", dbname);
goto end;
}
switch_core_db_persistant_execute(db, sql, 25);
switch_core_db_close(db);
end:
if (mutex) {
switch_mutex_unlock(mutex);
}
}
struct callback_t {
char *val;
switch_size_t len;
int matches;
};
static int find_callback(void *pArg, int argc, char **argv, char **columnNames){
struct callback_t *cbt = (struct callback_t *) pArg;
switch_copy_string(cbt->val, argv[0], cbt->len);
cbt->matches++;
return 0;
}
static int del_callback(void *pArg, int argc, char **argv, char **columnNames){
switch_event_t *s_event;
if (argc >=3 ) {
if (switch_event_create_subclass(&s_event, SWITCH_EVENT_CUSTOM, MY_EVENT_EXPIRE) == SWITCH_STATUS_SUCCESS) {
switch_event_add_header(s_event, SWITCH_STACK_BOTTOM, "profile-name", "%s", argv[0]);
switch_event_add_header(s_event, SWITCH_STACK_BOTTOM, "user", "%s", argv[1]);
switch_event_add_header(s_event, SWITCH_STACK_BOTTOM, "host", "%s", argv[2]);
switch_event_add_header(s_event, SWITCH_STACK_BOTTOM, "contact", "%s", argv[3]);
switch_event_add_header(s_event, SWITCH_STACK_BOTTOM, "expires", "%d", argv[4]);
switch_event_fire(&s_event);
}
}
return 0;
}
static void check_expire(sofia_profile_t *profile, time_t now)
{
char sql[1024];
char *errmsg;
switch_core_db_t *db;
if (!(db = switch_core_db_open_file(profile->dbname))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error Opening DB %s\n", profile->dbname);
return;
}
switch_mutex_lock(profile->ireg_mutex);
snprintf(sql, sizeof(sql), "select '%s',* from sip_registrations where expires > 0 and expires < %ld", profile->name, (long) now);
switch_core_db_exec(db, sql, del_callback, NULL, &errmsg);
if (errmsg) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "SQL ERR [%s][%s]\n", sql, errmsg);
switch_safe_free(errmsg);
errmsg = NULL;
}
snprintf(sql, sizeof(sql), "delete from sip_registrations where expires > 0 and expires < %ld", (long) now);
switch_core_db_persistant_execute(db, sql, 1000);
snprintf(sql, sizeof(sql), "delete from sip_authentication where expires > 0 and expires < %ld", (long) now);
switch_core_db_persistant_execute(db, sql, 1000);
snprintf(sql, sizeof(sql), "delete from sip_subscriptions where expires > 0 and expires < %ld", (long) now);
switch_core_db_persistant_execute(db, sql, 1000);
switch_mutex_unlock(profile->ireg_mutex);
switch_core_db_close(db);
}
static char *find_reg_url(sofia_profile_t *profile, char *user, char *host, char *val, switch_size_t len)
{
char *errmsg;
struct callback_t cbt = {0};
switch_core_db_t *db;
if (!(db = switch_core_db_open_file(profile->dbname))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error Opening DB %s\n", profile->dbname);
return NULL;
}
cbt.val = val;
cbt.len = len;
switch_mutex_lock(profile->ireg_mutex);
if (host) {
snprintf(val, len, "select contact from sip_registrations where user='%s' and host='%s'", user, host);
} else {
snprintf(val, len, "select contact from sip_registrations where user='%s'", user);
}
switch_core_db_exec(db, val, find_callback, &cbt, &errmsg);
if (errmsg) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "SQL ERR [%s][%s]\n", val, errmsg);
switch_safe_free(errmsg);
errmsg = NULL;
}
switch_mutex_unlock(profile->ireg_mutex);
switch_core_db_close(db);
return cbt.matches ? val : NULL;
}
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
static void set_local_sdp(private_object_t *tech_pvt, char *ip, uint32_t port, char *sr, int force)
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
{
char buf[1024];
switch_time_t now = switch_time_now();
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
if (!force && !ip && !sr && switch_test_flag(tech_pvt, TFLAG_NOMEDIA)) {
return;
}
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
if (!ip) {
if (!(ip = tech_pvt->adv_sdp_audio_ip)) {
ip = tech_pvt->proxy_sdp_audio_ip;
}
}
if (!port) {
if (!(port = tech_pvt->adv_sdp_audio_port)) {
port = tech_pvt->proxy_sdp_audio_port;
}
}
if (!sr) {
sr = "sendrecv";
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
snprintf(buf, sizeof(buf),
"v=0\n"
"o=FreeSWITCH %d%"APR_TIME_T_FMT" %d%"APR_TIME_T_FMT" IN IP4 %s\n"
"s=FreeSWITCH\n"
"c=IN IP4 %s\n"
"t=0 0\n"
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
"a=%s\n"
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
"m=audio %d RTP/AVP",
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
port,
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
now,
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
port,
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
now,
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
ip,
ip,
sr,
port
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
);
if (tech_pvt->rm_encoding) {
snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), " %d", tech_pvt->pt);
} else if (tech_pvt->num_codecs) {
int i;
for (i = 0; i < tech_pvt->num_codecs; i++) {
const switch_codec_implementation_t *imp = tech_pvt->codecs[i];
snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), " %d", imp->ianacode);
}
}
if (tech_pvt->te > 95) {
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), " %d", tech_pvt->te);
}
snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "\n");
if (tech_pvt->rm_encoding) {
snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "a=rtpmap:%d %s/%ld\n", tech_pvt->pt, tech_pvt->rm_encoding, tech_pvt->rm_rate);
if (tech_pvt->fmtp_out) {
snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "a=fmtp:%d %s\n", tech_pvt->pt, tech_pvt->fmtp_out);
}
if (tech_pvt->read_codec.implementation) {
snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "a=ptime:%d\n", tech_pvt->read_codec.implementation->microseconds_per_frame / 1000);
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
} else if (tech_pvt->num_codecs) {
int i;
for (i = 0; i < tech_pvt->num_codecs; i++) {
const switch_codec_implementation_t *imp = tech_pvt->codecs[i];
snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "a=rtpmap:%d %s/%d\n", imp->ianacode, imp->iananame, imp->samples_per_second);
if (imp->fmtp) {
snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "a=fmtp:%d %s\n", imp->ianacode, imp->fmtp);
}
snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "a=ptime:%d\n", imp->microseconds_per_frame / 1000);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
}
}
if (tech_pvt->te > 95) {
snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "a=rtpmap:%d telephone-event/8000\na=fmtp:%d 0-16\n", tech_pvt->te, tech_pvt->te);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
}
tech_pvt->local_sdp_str = switch_core_session_strdup(tech_pvt->session, buf);
}
static void tech_set_codecs(private_object_t *tech_pvt)
{
if (switch_test_flag(tech_pvt, TFLAG_NOMEDIA)) {
return;
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (tech_pvt->num_codecs) {
return;
}
if (tech_pvt->profile->codec_string) {
tech_pvt->num_codecs = switch_loadable_module_get_codecs_sorted(tech_pvt->codecs,
SWITCH_MAX_CODECS,
tech_pvt->profile->codec_order,
tech_pvt->profile->codec_order_last);
} else {
tech_pvt->num_codecs = switch_loadable_module_get_codecs(switch_core_session_get_pool(tech_pvt->session), tech_pvt->codecs,
sizeof(tech_pvt->codecs) / sizeof(tech_pvt->codecs[0]));
}
}
static void attach_private(switch_core_session_t *session,
sofia_profile_t *profile,
private_object_t *tech_pvt,
char *channame)
{
switch_channel_t *channel;
char name[256];
assert(session != NULL);
assert(profile != NULL);
assert(tech_pvt != NULL);
switch_core_session_add_stream(session, NULL);
channel = switch_core_session_get_channel(session);
switch_mutex_init(&tech_pvt->flag_mutex, SWITCH_MUTEX_NESTED, switch_core_session_get_pool(session));
switch_mutex_lock(tech_pvt->flag_mutex);
tech_pvt->flags = profile->flags;
switch_mutex_unlock(tech_pvt->flag_mutex);
tech_pvt->profile = profile;
tech_pvt->te = profile->te;
tech_pvt->session = session;
tech_pvt->home = su_home_new(sizeof(*tech_pvt->home));
su_home_init(tech_pvt->home);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
switch_core_session_set_private(session, tech_pvt);
tech_set_codecs(tech_pvt);
snprintf(name, sizeof(name), "sofia/%s/%s", profile->name, channame);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
switch_channel_set_name(channel, name);
}
static void terminate_session(switch_core_session_t **session, switch_call_cause_t cause, int line)
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
{
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Term called from line: %d\n", line);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (*session) {
switch_channel_t *channel = switch_core_session_get_channel(*session);
switch_channel_state_t state = switch_channel_get_state(channel);
struct private_object *tech_pvt = NULL;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
tech_pvt = switch_core_session_get_private(*session);
if (tech_pvt) {
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (state > CS_INIT && state < CS_HANGUP) {
switch_channel_hangup(channel, cause);
}
if (!switch_test_flag(tech_pvt, TFLAG_READY)) {
if (state > CS_INIT && state < CS_HANGUP) {
sofia_on_hangup(*session);
}
switch_core_session_destroy(session);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
}
} else {
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
switch_core_session_destroy(session);
}
}
}
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
static switch_status_t tech_choose_port(private_object_t *tech_pvt)
{
char *ip = tech_pvt->profile->rtpip;
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
switch_channel_t *channel;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
switch_port_t sdp_port;
char *err;
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
char tmp[50];
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
channel = switch_core_session_get_channel(tech_pvt->session);
if (switch_test_flag(tech_pvt, TFLAG_NOMEDIA) || tech_pvt->adv_sdp_audio_port) {
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
return SWITCH_STATUS_SUCCESS;
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
tech_pvt->local_sdp_audio_ip = ip;
tech_pvt->local_sdp_audio_port = switch_rtp_request_port();
sdp_port = tech_pvt->local_sdp_audio_port;
if (tech_pvt->profile->extrtpip) {
if (!strncasecmp(tech_pvt->profile->extrtpip, "stun:", 5)) {
char *stun_ip = tech_pvt->profile->extrtpip + 5;
if (!stun_ip) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Stun Failed! NO STUN SERVER\n");
terminate_session(&tech_pvt->session, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER, __LINE__);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
return SWITCH_STATUS_FALSE;
}
if (switch_stun_lookup(&ip,
&sdp_port,
stun_ip,
SWITCH_STUN_DEFAULT_PORT,
&err,
switch_core_session_get_pool(tech_pvt->session)) != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Stun Failed! %s:%d [%s]\n", stun_ip, SWITCH_STUN_DEFAULT_PORT, err);
terminate_session(&tech_pvt->session, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER, __LINE__);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
return SWITCH_STATUS_FALSE;
}
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "Stun Success [%s]:[%d]\n", ip, sdp_port);
} else {
ip = tech_pvt->profile->extrtpip;
}
}
tech_pvt->adv_sdp_audio_ip = switch_core_session_strdup(tech_pvt->session, ip);
tech_pvt->adv_sdp_audio_port = sdp_port;
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
snprintf(tmp, sizeof(tmp), "%d", sdp_port);
switch_channel_set_variable(channel, SWITCH_LOCAL_MEDIA_IP_VARIABLE, tech_pvt->adv_sdp_audio_ip);
switch_channel_set_variable(channel, SWITCH_LOCAL_MEDIA_PORT_VARIABLE, tmp);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
return SWITCH_STATUS_SUCCESS;
}
static void do_invite(switch_core_session_t *session)
{
char rpid[1024] = { 0 };
char alert_info[1024] = { 0 };
char *alertbuf;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
private_object_t *tech_pvt;
switch_channel_t *channel = NULL;
switch_caller_profile_t *caller_profile;
char *cid_name, *cid_num;
char *e_dest = NULL;
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
char *holdstr = "";
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
tech_pvt = (private_object_t *) switch_core_session_get_private(session);
assert(tech_pvt != NULL);
caller_profile = switch_channel_get_caller_profile(channel);
cid_name = (char *) caller_profile->caller_id_name;
cid_num = (char *) caller_profile->caller_id_number;
if ((tech_pvt->from_str = switch_mprintf("\"%s\" <sip:%s@%s>",
cid_name,
cid_num,
tech_pvt->profile->sipip
))) {
char *rep = switch_channel_get_variable(channel, SOFIA_REPLACES_HEADER);
if ((alertbuf = switch_channel_get_variable(channel, "alert_info"))) {
snprintf(alert_info, sizeof(alert_info) - 1, "Alert-Info: %s", alertbuf);
}
tech_choose_port(tech_pvt);
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
set_local_sdp(tech_pvt, NULL, 0, NULL, 0);
switch_set_flag_locked(tech_pvt, TFLAG_READY);
// forge a RPID for now KHR -- Should wrap this in an if statement so it can be turned on and off
if (switch_test_flag(caller_profile, SWITCH_CPF_SCREEN)) {
char *priv = "no";
if (switch_test_flag(caller_profile, SWITCH_CPF_HIDE_NAME)) {
priv = "name";
if (switch_test_flag(caller_profile, SWITCH_CPF_HIDE_NUMBER)) {
priv = "yes";
}
} else if (switch_test_flag(caller_profile, SWITCH_CPF_HIDE_NUMBER)) {
priv = "yes";
}
snprintf(rpid, sizeof(rpid) - 1, "Remote-Party-ID: %s;party=calling;screen=yes;privacy=%s", tech_pvt->from_str, priv);
}
if (!tech_pvt->nh) {
tech_pvt->nh = nua_handle(tech_pvt->profile->nua, NULL,
SIPTAG_TO_STR(tech_pvt->dest),
SIPTAG_FROM_STR(tech_pvt->from_str),
SIPTAG_CONTACT_STR(tech_pvt->profile->url),
TAG_END());
tech_pvt->sofia_private.session = session;
nua_handle_bind(tech_pvt->nh, &tech_pvt->sofia_private);
}
if (tech_pvt->e_dest && (e_dest = strdup(tech_pvt->e_dest))) {
char *user = e_dest, *host = NULL;
char hash_key[256] = "";
if ((host = strchr(user, '@'))) {
*host++ = '\0';
}
snprintf(hash_key, sizeof(hash_key), "%s%s%s", user, host, cid_num);
tech_pvt->chat_from = tech_pvt->from_str;
tech_pvt->chat_to = tech_pvt->dest;
tech_pvt->hash_key = switch_core_session_strdup(tech_pvt->session, hash_key);
switch_core_hash_insert(tech_pvt->profile->chat_hash, tech_pvt->hash_key, tech_pvt);
free(e_dest);
}
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
holdstr = switch_test_flag(tech_pvt, TFLAG_SIP_HOLD) ? "*" : "";
nua_invite(tech_pvt->nh,
TAG_IF(rpid, SIPTAG_HEADER_STR(rpid)),
TAG_IF(alert_info, SIPTAG_HEADER_STR(alert_info)),
//SIPTAG_CONTACT_STR(tech_pvt->profile->url),
SOATAG_USER_SDP_STR(tech_pvt->local_sdp_str),
SOATAG_RTP_SORT(SOA_RTP_SORT_REMOTE),
SOATAG_RTP_SELECT(SOA_RTP_SELECT_ALL),
TAG_IF(rep, SIPTAG_REPLACES_STR(rep)),
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
SOATAG_HOLD(holdstr),
TAG_END());
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Memory Error!\n");
}
}
static void do_xfer_invite(switch_core_session_t *session)
{
char rpid[1024];
private_object_t *tech_pvt;
switch_channel_t *channel = NULL;
switch_caller_profile_t *caller_profile;
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
tech_pvt = (private_object_t *) switch_core_session_get_private(session);
assert(tech_pvt != NULL);
caller_profile = switch_channel_get_caller_profile(channel);
if ((tech_pvt->from_str = switch_mprintf("\"%s\" <sip:%s@%s>",
(char *) caller_profile->caller_id_name,
(char *) caller_profile->caller_id_number,
tech_pvt->profile->sipip
))) {
char *rep = switch_channel_get_variable(channel, SOFIA_REPLACES_HEADER);
tech_pvt->nh2 = nua_handle(tech_pvt->profile->nua, NULL,
SIPTAG_TO_STR(tech_pvt->dest),
SIPTAG_FROM_STR(tech_pvt->from_str),
SIPTAG_CONTACT_STR(tech_pvt->profile->url),
TAG_END());
nua_handle_bind(tech_pvt->nh2, &tech_pvt->sofia_private);
nua_invite(tech_pvt->nh2,
TAG_IF(rpid, SIPTAG_HEADER_STR(rpid)),
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
SIPTAG_CONTACT_STR(tech_pvt->profile->url),
SOATAG_USER_SDP_STR(tech_pvt->local_sdp_str),
SOATAG_RTP_SORT(SOA_RTP_SORT_REMOTE),
SOATAG_RTP_SELECT(SOA_RTP_SELECT_ALL),
TAG_IF(rep, SIPTAG_REPLACES_STR(rep)),
TAG_END());
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Memory Error!\n");
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
}
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
static void tech_absorb_sdp(private_object_t *tech_pvt)
{
switch_channel_t *channel;
char *sdp_str;
channel = switch_core_session_get_channel(tech_pvt->session);
assert(channel != NULL);
if ((sdp_str = switch_channel_get_variable(channel, SWITCH_B_SDP_VARIABLE))) {
sdp_parser_t *parser;
sdp_session_t *sdp;
sdp_media_t *m;
if ((parser = sdp_parse(tech_pvt->home, sdp_str, (int)strlen(sdp_str), 0))) {
if ((sdp = sdp_session(parser))) {
for (m = sdp->sdp_media; m ; m = m->m_next) {
tech_pvt->proxy_sdp_audio_ip = switch_core_session_strdup(tech_pvt->session, (char *)sdp->sdp_connection->c_address);
tech_pvt->proxy_sdp_audio_port = (switch_port_t)m->m_port;
if (tech_pvt->proxy_sdp_audio_ip && tech_pvt->proxy_sdp_audio_port) {
break;
}
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
}
}
sdp_parser_free(parser);
}
tech_pvt->local_sdp_str = switch_core_session_strdup(tech_pvt->session, sdp_str);
}
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
/*
State methods they get called when the state changes to the specific state
returning SWITCH_STATUS_SUCCESS tells the core to execute the standard state method next
so if you fully implement the state you can return SWITCH_STATUS_FALSE to skip it.
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
*/
static switch_status_t sofia_on_init(switch_core_session_t *session)
{
private_object_t *tech_pvt;
switch_channel_t *channel = NULL;
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
tech_pvt = (private_object_t *) switch_core_session_get_private(session);
assert(tech_pvt != NULL);
tech_pvt->read_frame.buflen = SWITCH_RTP_MAX_BUF_LEN;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
switch_channel_set_variable(channel, "endpoint_disposition", "INIT");
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "SOFIA INIT\n");
if (switch_channel_test_flag(channel, CF_NOMEDIA)) {
switch_set_flag_locked(tech_pvt, TFLAG_NOMEDIA);
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
tech_absorb_sdp(tech_pvt);
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (switch_test_flag(tech_pvt, TFLAG_OUTBOUND)) {
do_invite(session);
}
/* Move Channel's State Machine to RING */
switch_channel_set_state(channel, CS_RING);
return SWITCH_STATUS_SUCCESS;
}
static switch_status_t sofia_on_ring(switch_core_session_t *session)
{
switch_channel_t *channel = NULL;
private_object_t *tech_pvt = NULL;
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
tech_pvt = (private_object_t *) switch_core_session_get_private(session);
assert(tech_pvt != NULL);
switch_channel_set_variable(channel, "endpoint_disposition", "RING");
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "SOFIA RING\n");
return SWITCH_STATUS_SUCCESS;
}
static switch_status_t sofia_on_execute(switch_core_session_t *session)
{
switch_channel_t *channel = NULL;
private_object_t *tech_pvt = NULL;
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
tech_pvt = (private_object_t *) switch_core_session_get_private(session);
assert(tech_pvt != NULL);
switch_channel_set_variable(channel, "endpoint_disposition", "EXECUTE");
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "SOFIA EXECUTE\n");
return SWITCH_STATUS_SUCCESS;
}
// map QSIG cause codes to SIP from RFC4497 section 8.4.1
static int hangup_cause_to_sip(switch_call_cause_t cause) {
switch (cause) {
case SWITCH_CAUSE_UNALLOCATED:
case SWITCH_CAUSE_NO_ROUTE_TRANSIT_NET:
case SWITCH_CAUSE_NO_ROUTE_DESTINATION:
return 404;
case SWITCH_CAUSE_USER_BUSY:
return 486;
case SWITCH_CAUSE_NO_USER_RESPONSE:
return 408;
case SWITCH_CAUSE_NO_ANSWER:
return 480;
case SWITCH_CAUSE_SUBSCRIBER_ABSENT:
return 480;
case SWITCH_CAUSE_CALL_REJECTED:
return 603;
case SWITCH_CAUSE_NUMBER_CHANGED:
case SWITCH_CAUSE_REDIRECTION_TO_NEW_DESTINATION:
return 410;
case SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER:
return 502;
case SWITCH_CAUSE_INVALID_NUMBER_FORMAT:
return 484;
case SWITCH_CAUSE_FACILITY_REJECTED:
return 501;
case SWITCH_CAUSE_NORMAL_UNSPECIFIED:
return 480;
case SWITCH_CAUSE_NORMAL_CIRCUIT_CONGESTION:
case SWITCH_CAUSE_NETWORK_OUT_OF_ORDER:
case SWITCH_CAUSE_NORMAL_TEMPORARY_FAILURE:
case SWITCH_CAUSE_SWITCH_CONGESTION:
return 503;
case SWITCH_CAUSE_OUTGOING_CALL_BARRED:
case SWITCH_CAUSE_INCOMING_CALL_BARRED:
case SWITCH_CAUSE_BEARERCAPABILITY_NOTAUTH:
return 403;
case SWITCH_CAUSE_BEARERCAPABILITY_NOTAVAIL:
return 503;
case SWITCH_CAUSE_BEARERCAPABILITY_NOTIMPL:
return 488;
case SWITCH_CAUSE_FACILITY_NOT_IMPLEMENTED:
case SWITCH_CAUSE_SERVICE_NOT_IMPLEMENTED:
return 501;
case SWITCH_CAUSE_INCOMPATIBLE_DESTINATION:
return 503;
case SWITCH_CAUSE_RECOVERY_ON_TIMER_EXPIRE:
return 504;
case SWITCH_CAUSE_ORIGINATOR_CANCEL:
return 487;
default:
return 500;
}
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
static switch_status_t sofia_on_hangup(switch_core_session_t *session)
{
switch_core_session_t *asession;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
private_object_t *tech_pvt;
switch_channel_t *channel = NULL;
switch_call_cause_t cause;
int sip_cause;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
tech_pvt = (private_object_t *) switch_core_session_get_private(session);
assert(tech_pvt != NULL);
cause = switch_channel_get_cause(channel);
sip_cause = hangup_cause_to_sip(cause);
deactivate_rtp(tech_pvt);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Channel %s hanging up, cause: %s\n",
switch_channel_get_name(channel), switch_channel_cause2str(cause), sip_cause);
if (tech_pvt->hash_key) {
switch_core_hash_delete(tech_pvt->profile->chat_hash, tech_pvt->hash_key);
}
if (tech_pvt->kick && (asession = switch_core_session_locate(tech_pvt->kick))) {
switch_channel_t *a_channel = switch_core_session_get_channel(asession);
switch_channel_hangup(a_channel, switch_channel_get_cause(channel));
switch_core_session_rwunlock(asession);
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (tech_pvt->nh) {
if (!switch_test_flag(tech_pvt, TFLAG_BYE)) {
if (switch_test_flag(tech_pvt, TFLAG_ANS)) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Sending BYE\n");
nua_bye(tech_pvt->nh, TAG_END());
} else {
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
if (!switch_test_flag(tech_pvt, TFLAG_OUTBOUND)) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Responding to INVITE with: %d\n", sip_cause);
nua_respond(tech_pvt->nh, sip_cause, NULL, TAG_END());
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Sending CANCEL\n");
nua_cancel(tech_pvt->nh, TAG_END());
}
}
switch_set_flag_locked(tech_pvt, TFLAG_BYE);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
}
}
if (tech_pvt->from_str) {
switch_safe_free(tech_pvt->from_str);
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
switch_clear_flag_locked(tech_pvt, TFLAG_IO);
if (tech_pvt->home) {
su_home_deinit(tech_pvt->home);
tech_pvt->home = NULL;
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
return SWITCH_STATUS_SUCCESS;
}
static switch_status_t sofia_on_loopback(switch_core_session_t *session)
{
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "SOFIA LOOPBACK\n");
return SWITCH_STATUS_SUCCESS;
}
static switch_status_t sofia_on_transmit(switch_core_session_t *session)
{
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "SOFIA TRANSMIT\n");
return SWITCH_STATUS_SUCCESS;
}
static void deactivate_rtp(private_object_t *tech_pvt)
{
int loops = 0;//, sock = -1;
if (switch_rtp_ready(tech_pvt->rtp_session)) {
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
while (loops < 10 && (switch_test_flag(tech_pvt, TFLAG_READING) || switch_test_flag(tech_pvt, TFLAG_WRITING))) {
switch_yield(10000);
loops++;
}
switch_rtp_destroy(&tech_pvt->rtp_session);
}
}
static switch_status_t tech_set_codec(private_object_t *tech_pvt, int force)
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
{
switch_channel_t *channel;
assert(tech_pvt->codecs[tech_pvt->codec_index] != NULL);
if (tech_pvt->read_codec.implementation) {
if (!force) {
return SWITCH_STATUS_SUCCESS;
}
if (strcasecmp(tech_pvt->read_codec.implementation->iananame, tech_pvt->rm_encoding) ||
tech_pvt->read_codec.implementation->samples_per_second != tech_pvt->rm_rate) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Changing Codec from %s to %s\n",
tech_pvt->read_codec.implementation->iananame, tech_pvt->rm_encoding);
switch_core_codec_destroy(&tech_pvt->read_codec);
switch_core_codec_destroy(&tech_pvt->write_codec);
switch_core_session_reset(tech_pvt->session);
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Already using %s\n",
tech_pvt->read_codec.implementation->iananame);
return SWITCH_STATUS_SUCCESS;
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
}
channel = switch_core_session_get_channel(tech_pvt->session);
assert(channel != NULL);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (switch_core_codec_init(&tech_pvt->read_codec,
tech_pvt->rm_encoding,
tech_pvt->rm_fmtp,
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
tech_pvt->rm_rate,
tech_pvt->codec_ms,
1,
SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE | tech_pvt->profile->codec_flags,
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
NULL,
switch_core_session_get_pool(tech_pvt->session)) != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Can't load codec?\n");
terminate_session(&tech_pvt->session, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER, __LINE__);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
return SWITCH_STATUS_FALSE;
} else {
if (switch_core_codec_init(&tech_pvt->write_codec,
tech_pvt->rm_encoding,
tech_pvt->rm_fmtp,
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
tech_pvt->rm_rate,
tech_pvt->codec_ms,
1,
SWITCH_CODEC_FLAG_ENCODE | SWITCH_CODEC_FLAG_DECODE | tech_pvt->profile->codec_flags,
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
NULL,
switch_core_session_get_pool(tech_pvt->session)) != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Can't load codec?\n");
terminate_session(&tech_pvt->session, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER, __LINE__);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
return SWITCH_STATUS_FALSE;
} else {
int ms;
tech_pvt->read_frame.rate = tech_pvt->rm_rate;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
ms = tech_pvt->write_codec.implementation->microseconds_per_frame / 1000;
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "Set Codec %s %s/%d %d ms\n",
switch_channel_get_name(channel),
tech_pvt->codecs[tech_pvt->codec_index]->iananame, tech_pvt->rm_rate, tech_pvt->codec_ms);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
tech_pvt->read_frame.codec = &tech_pvt->read_codec;
switch_core_session_set_read_codec(tech_pvt->session, &tech_pvt->read_codec);
switch_core_session_set_write_codec(tech_pvt->session, &tech_pvt->write_codec);
tech_pvt->fmtp_out = switch_core_session_strdup(tech_pvt->session, tech_pvt->write_codec.fmtp_out);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
}
}
return SWITCH_STATUS_SUCCESS;
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
static switch_status_t activate_rtp(private_object_t *tech_pvt)
{
int bw, ms;
switch_channel_t *channel;
const char *err = NULL;
switch_rtp_flag_t flags;
switch_status_t status;
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
char tmp[50];
assert(tech_pvt != NULL);
channel = switch_core_session_get_channel(tech_pvt->session);
assert(channel != NULL);
if (switch_test_flag(tech_pvt, TFLAG_NOMEDIA)) {
return SWITCH_STATUS_SUCCESS;
}
if (switch_rtp_ready(tech_pvt->rtp_session) && !switch_test_flag(tech_pvt, TFLAG_REINVITE)) {
return SWITCH_STATUS_SUCCESS;
}
if ((status = tech_set_codec(tech_pvt, 0)) != SWITCH_STATUS_SUCCESS) {
return status;
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
bw = tech_pvt->read_codec.implementation->bits_per_second;
ms = tech_pvt->read_codec.implementation->microseconds_per_frame;
flags = (switch_rtp_flag_t) (SWITCH_RTP_FLAG_RAW_WRITE | SWITCH_RTP_FLAG_AUTOADJ | SWITCH_RTP_FLAG_DATAWAIT);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (switch_test_flag(tech_pvt, TFLAG_BUGGY_2833)) {
flags |= SWITCH_RTP_FLAG_BUGGY_2833;
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "RTP [%s] %s:%d->%s:%d codec: %u ms: %d\n",
switch_channel_get_name(channel),
tech_pvt->local_sdp_audio_ip,
tech_pvt->local_sdp_audio_port,
tech_pvt->remote_sdp_audio_ip,
tech_pvt->remote_sdp_audio_port,
tech_pvt->agreed_pt,
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
tech_pvt->read_codec.implementation->microseconds_per_frame / 1000);
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
snprintf(tmp, sizeof(tmp), "%d", tech_pvt->remote_sdp_audio_port);
switch_channel_set_variable(channel, SWITCH_LOCAL_MEDIA_IP_VARIABLE, tech_pvt->adv_sdp_audio_ip);
switch_channel_set_variable(channel, SWITCH_LOCAL_MEDIA_PORT_VARIABLE, tmp);
if (tech_pvt->rtp_session && switch_test_flag(tech_pvt, TFLAG_REINVITE)) {
switch_clear_flag_locked(tech_pvt, TFLAG_REINVITE);
if (switch_rtp_set_remote_address(tech_pvt->rtp_session,
tech_pvt->remote_sdp_audio_ip,
tech_pvt->remote_sdp_audio_port,
&err) != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "RTP REPORTS ERROR: [%s]\n", err);
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "RTP CHANGING DEST TO: [%s:%d]\n",
tech_pvt->remote_sdp_audio_ip, tech_pvt->remote_sdp_audio_port);
}
return SWITCH_STATUS_SUCCESS;
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
tech_pvt->rtp_session = switch_rtp_new(tech_pvt->local_sdp_audio_ip,
tech_pvt->local_sdp_audio_port,
tech_pvt->remote_sdp_audio_ip,
tech_pvt->remote_sdp_audio_port,
tech_pvt->agreed_pt,
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
tech_pvt->read_codec.implementation->encoded_bytes_per_frame,
tech_pvt->codec_ms * 1000,
(switch_rtp_flag_t) flags,
NULL,
tech_pvt->profile->timer_name,
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
&err,
switch_core_session_get_pool(tech_pvt->session));
if (switch_rtp_ready(tech_pvt->rtp_session)) {
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
uint8_t vad_in = switch_test_flag(tech_pvt, TFLAG_VAD_IN) ? 1 : 0;
uint8_t vad_out = switch_test_flag(tech_pvt, TFLAG_VAD_OUT) ? 1 : 0;
uint8_t inb = switch_test_flag(tech_pvt, TFLAG_OUTBOUND) ? 0 : 1;
tech_pvt->ssrc = switch_rtp_get_ssrc(tech_pvt->rtp_session);
switch_set_flag_locked(tech_pvt, TFLAG_RTP);
switch_set_flag_locked(tech_pvt, TFLAG_IO);
if ((vad_in && inb) || (vad_out && !inb)) {
switch_rtp_enable_vad(tech_pvt->rtp_session, tech_pvt->session, &tech_pvt->read_codec, SWITCH_VAD_FLAG_TALKING);
switch_set_flag_locked(tech_pvt, TFLAG_VAD);
}
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "RTP REPORTS ERROR: [%s]\n", err);
terminate_session(&tech_pvt->session, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER, __LINE__);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
switch_clear_flag_locked(tech_pvt, TFLAG_IO);
return SWITCH_STATUS_FALSE;
}
switch_set_flag_locked(tech_pvt, TFLAG_IO);
return SWITCH_STATUS_SUCCESS;
}
static switch_status_t sofia_answer_channel(switch_core_session_t *session)
{
private_object_t *tech_pvt;
switch_channel_t *channel = NULL;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
assert(session != NULL);
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
tech_pvt = (private_object_t *) switch_core_session_get_private(session);
assert(tech_pvt != NULL);
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
if (switch_channel_test_flag(channel, CF_NOMEDIA)) {
switch_set_flag_locked(tech_pvt, TFLAG_NOMEDIA);
tech_absorb_sdp(tech_pvt);
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (!switch_test_flag(tech_pvt, TFLAG_ANS) && !switch_channel_test_flag(channel, CF_OUTBOUND)) {
switch_set_flag_locked(tech_pvt, TFLAG_ANS);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
tech_choose_port(tech_pvt);
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
set_local_sdp(tech_pvt, NULL, 0, NULL, 0);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
activate_rtp(tech_pvt);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (tech_pvt->nh) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Local SDP %s:\n%s\n",
switch_channel_get_name(channel),
tech_pvt->local_sdp_str);
nua_respond(tech_pvt->nh, SIP_200_OK,
SIPTAG_CONTACT_STR(tech_pvt->profile->url),
SOATAG_USER_SDP_STR(tech_pvt->local_sdp_str),
SOATAG_AUDIO_AUX("cn telephone-event"),
NUTAG_INCLUDE_EXTRA_SDP(1),
TAG_END());
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
}
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
return SWITCH_STATUS_SUCCESS;
}
static switch_status_t sofia_read_frame(switch_core_session_t *session, switch_frame_t **frame, int timeout,
switch_io_flag_t flags, int stream_id)
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
{
private_object_t *tech_pvt = NULL;
size_t bytes = 0, samples = 0, frames = 0, ms = 0;
switch_channel_t *channel = NULL;
int payload = 0;
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
tech_pvt = (private_object_t *) switch_core_session_get_private(session);
assert(tech_pvt != NULL);
if (switch_test_flag(tech_pvt, TFLAG_HUP)) {
return SWITCH_STATUS_FALSE;
}
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
while (!(tech_pvt->read_codec.implementation && switch_rtp_ready(tech_pvt->rtp_session))) {
if (switch_channel_ready(channel)) {
switch_yield(10000);
} else {
return SWITCH_STATUS_GENERR;
}
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
tech_pvt->read_frame.datalen = 0;
switch_set_flag_locked(tech_pvt, TFLAG_READING);
bytes = tech_pvt->read_codec.implementation->encoded_bytes_per_frame;
samples = tech_pvt->read_codec.implementation->samples_per_frame;
ms = tech_pvt->read_codec.implementation->microseconds_per_frame;
if (tech_pvt->last_read) {
#if 0
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
elapsed = (unsigned int)((switch_time_now() - tech_pvt->last_read) / 1000);
if (elapsed > 60000) {
return SWITCH_STATUS_TIMEOUT;
}
#endif
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
}
if (switch_test_flag(tech_pvt, TFLAG_IO)) {
switch_status_t status;
if (!switch_test_flag(tech_pvt, TFLAG_RTP)) {
return SWITCH_STATUS_GENERR;
}
assert(tech_pvt->rtp_session != NULL);
tech_pvt->read_frame.datalen = 0;
while (!switch_test_flag(tech_pvt, TFLAG_BYE) && switch_test_flag(tech_pvt, TFLAG_IO) && tech_pvt->read_frame.datalen == 0) {
tech_pvt->read_frame.flags = SFF_NONE;
status = switch_rtp_zerocopy_read_frame(tech_pvt->rtp_session, &tech_pvt->read_frame);
if (status != SWITCH_STATUS_SUCCESS && status != SWITCH_STATUS_BREAK) {
return SWITCH_STATUS_FALSE;
}
payload = tech_pvt->read_frame.payload;
#if 0
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
elapsed = (unsigned int)((switch_time_now() - started) / 1000);
if (timeout > -1) {
if (elapsed >= (unsigned int)timeout) {
return SWITCH_STATUS_BREAK;
}
}
elapsed = (unsigned int)((switch_time_now() - last_act) / 1000);
if (elapsed >= hard_timeout) {
return SWITCH_STATUS_BREAK;
}
#endif
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (switch_rtp_has_dtmf(tech_pvt->rtp_session)) {
char dtmf[128];
switch_rtp_dequeue_dtmf(tech_pvt->rtp_session, dtmf, sizeof(dtmf));
switch_channel_queue_dtmf(channel, dtmf);
}
if (tech_pvt->read_frame.datalen > 0) {
tech_pvt->last_read = switch_time_now();
if (tech_pvt->read_codec.implementation->encoded_bytes_per_frame) {
bytes = tech_pvt->read_codec.implementation->encoded_bytes_per_frame;
frames = (tech_pvt->read_frame.datalen / bytes);
} else
frames = 1;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
samples = frames * tech_pvt->read_codec.implementation->samples_per_frame;
ms = frames * tech_pvt->read_codec.implementation->microseconds_per_frame;
tech_pvt->timestamp_recv += (int32_t) samples;
tech_pvt->read_frame.samples = (int) samples;
break;
}
}
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
switch_clear_flag_locked(tech_pvt, TFLAG_READING);
if (tech_pvt->read_frame.datalen == 0) {
*frame = NULL;
return SWITCH_STATUS_GENERR;
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
*frame = &tech_pvt->read_frame;
return SWITCH_STATUS_SUCCESS;
}
static switch_status_t sofia_write_frame(switch_core_session_t *session, switch_frame_t *frame, int timeout,
switch_io_flag_t flags, int stream_id)
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
{
private_object_t *tech_pvt;
switch_channel_t *channel = NULL;
switch_status_t status = SWITCH_STATUS_SUCCESS;
int bytes = 0, samples = 0, frames = 0;
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
tech_pvt = (private_object_t *) switch_core_session_get_private(session);
assert(tech_pvt != NULL);
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
while (!(tech_pvt->read_codec.implementation && switch_rtp_ready(tech_pvt->rtp_session))) {
if (switch_channel_ready(channel)) {
switch_yield(10000);
} else {
return SWITCH_STATUS_GENERR;
}
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (switch_test_flag(tech_pvt, TFLAG_HUP)) {
return SWITCH_STATUS_FALSE;
}
if (!switch_test_flag(tech_pvt, TFLAG_RTP)) {
return SWITCH_STATUS_GENERR;
}
if (!switch_test_flag(tech_pvt, TFLAG_IO)) {
return SWITCH_STATUS_SUCCESS;
}
switch_set_flag_locked(tech_pvt, TFLAG_WRITING);
if (tech_pvt->read_codec.implementation->encoded_bytes_per_frame) {
bytes = tech_pvt->read_codec.implementation->encoded_bytes_per_frame;
frames = ((int) frame->datalen / bytes);
} else
frames = 1;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
samples = frames * tech_pvt->read_codec.implementation->samples_per_frame;
#if 0
printf("%s %s->%s send %d bytes %d samples in %d frames ts=%d\n",
switch_channel_get_name(channel),
tech_pvt->local_sdp_audio_ip,
tech_pvt->remote_sdp_audio_ip,
frame->datalen,
samples,
frames,
tech_pvt->timestamp_send);
#endif
switch_rtp_write_frame(tech_pvt->rtp_session, frame, samples);
tech_pvt->timestamp_send += (int) samples;
switch_clear_flag_locked(tech_pvt, TFLAG_WRITING);
return status;
}
static switch_status_t sofia_kill_channel(switch_core_session_t *session, int sig)
{
private_object_t *tech_pvt;
switch_channel_t *channel = NULL;
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
tech_pvt = (private_object_t *) switch_core_session_get_private(session);
assert(tech_pvt != NULL);
switch_clear_flag_locked(tech_pvt, TFLAG_IO);
switch_set_flag_locked(tech_pvt, TFLAG_HUP);
if (switch_rtp_ready(tech_pvt->rtp_session)) {
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
switch_rtp_kill_socket(tech_pvt->rtp_session);
}
return SWITCH_STATUS_SUCCESS;
}
static switch_status_t sofia_waitfor_read(switch_core_session_t *session, int ms, int stream_id)
{
private_object_t *tech_pvt;
switch_channel_t *channel = NULL;
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
tech_pvt = (private_object_t *) switch_core_session_get_private(session);
assert(tech_pvt != NULL);
return SWITCH_STATUS_SUCCESS;
}
static switch_status_t sofia_waitfor_write(switch_core_session_t *session, int ms, int stream_id)
{
private_object_t *tech_pvt;
switch_channel_t *channel = NULL;
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
tech_pvt = (private_object_t *) switch_core_session_get_private(session);
assert(tech_pvt != NULL);
return SWITCH_STATUS_SUCCESS;
}
static switch_status_t sofia_send_dtmf(switch_core_session_t *session, char *digits)
{
private_object_t *tech_pvt;
tech_pvt = (private_object_t *) switch_core_session_get_private(session);
assert(tech_pvt != NULL);
return switch_rtp_queue_rfc2833(tech_pvt->rtp_session,
digits,
tech_pvt->profile->dtmf_duration * (tech_pvt->read_codec.implementation->samples_per_second / 1000));
}
static switch_status_t sofia_receive_message(switch_core_session_t *session, switch_core_session_message_t *msg)
{
switch_channel_t *channel;
private_object_t *tech_pvt;
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
tech_pvt = (private_object_t *) switch_core_session_get_private(session);
assert(tech_pvt != NULL);
switch (msg->message_id) {
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
case SWITCH_MESSAGE_INDICATE_NOMEDIA: {
char *uuid;
switch_core_session_t *other_session;
switch_channel_t *other_channel;
char *ip = NULL, *port = NULL;
switch_set_flag_locked(tech_pvt, TFLAG_NOMEDIA);
tech_pvt->local_sdp_str = NULL;
if ((uuid = switch_channel_get_variable(channel, SWITCH_BRIDGE_VARIABLE)) && (other_session = switch_core_session_locate(uuid))) {
other_channel = switch_core_session_get_channel(other_session);
ip = switch_channel_get_variable(other_channel, SWITCH_REMOTE_MEDIA_IP_VARIABLE);
port = switch_channel_get_variable(other_channel, SWITCH_REMOTE_MEDIA_PORT_VARIABLE);
switch_core_session_rwunlock(other_session);
if (ip && port) {
set_local_sdp(tech_pvt, ip, atoi(port), NULL, 1);
}
}
if (!tech_pvt->local_sdp_str) {
tech_absorb_sdp(tech_pvt);
}
do_invite(session);
}
break;
case SWITCH_MESSAGE_INDICATE_MEDIA: {
switch_clear_flag_locked(tech_pvt, TFLAG_NOMEDIA);
tech_pvt->local_sdp_str = NULL;
if (!switch_rtp_ready(tech_pvt->rtp_session)) {
tech_set_codecs(tech_pvt);
tech_choose_port(tech_pvt);
}
set_local_sdp(tech_pvt, NULL, 0, NULL, 1);
do_invite(session);
while (!switch_rtp_ready(tech_pvt->rtp_session) && switch_channel_get_state(channel) < CS_HANGUP) {
switch_yield(1000);
}
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
}
break;
case SWITCH_MESSAGE_INDICATE_HOLD: {
switch_set_flag_locked(tech_pvt, TFLAG_SIP_HOLD);
do_invite(session);
}
break;
case SWITCH_MESSAGE_INDICATE_UNHOLD: {
switch_clear_flag_locked(tech_pvt, TFLAG_SIP_HOLD);
do_invite(session);
}
break;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
case SWITCH_MESSAGE_INDICATE_BRIDGE:
if (switch_test_flag(tech_pvt, TFLAG_XFER)) {
switch_clear_flag_locked(tech_pvt, TFLAG_XFER);
if (msg->pointer_arg) {
switch_core_session_t *asession, *bsession = msg->pointer_arg;
if ((asession = switch_core_session_locate(tech_pvt->xferto))) {
private_object_t *a_tech_pvt = switch_core_session_get_private(asession);
private_object_t *b_tech_pvt = switch_core_session_get_private(bsession);
switch_set_flag_locked(a_tech_pvt, TFLAG_REINVITE);
a_tech_pvt->remote_sdp_audio_ip = switch_core_session_strdup(asession, b_tech_pvt->remote_sdp_audio_ip);
a_tech_pvt->remote_sdp_audio_port = b_tech_pvt->remote_sdp_audio_port;
a_tech_pvt->local_sdp_audio_ip = switch_core_session_strdup(asession, b_tech_pvt->local_sdp_audio_ip);
a_tech_pvt->local_sdp_audio_port = b_tech_pvt->local_sdp_audio_port;
activate_rtp(a_tech_pvt);
b_tech_pvt->kick = switch_core_session_strdup(bsession, tech_pvt->xferto);
switch_core_session_rwunlock(asession);
}
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
msg->pointer_arg = NULL;
return SWITCH_STATUS_FALSE;
}
}
if (tech_pvt->rtp_session && switch_test_flag(tech_pvt, TFLAG_TIMER)) {
switch_rtp_clear_flag(tech_pvt->rtp_session, SWITCH_RTP_FLAG_USE_TIMER);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "De-activate timed RTP!\n");
}
break;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
case SWITCH_MESSAGE_INDICATE_UNBRIDGE:
if (tech_pvt->rtp_session && switch_test_flag(tech_pvt, TFLAG_TIMER)) {
switch_rtp_set_flag(tech_pvt->rtp_session, SWITCH_RTP_FLAG_USE_TIMER);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Re-activate timed RTP!\n");
}
break;
case SWITCH_MESSAGE_INDICATE_RINGING:
nua_respond(tech_pvt->nh, SIP_180_RINGING, SIPTAG_CONTACT_STR(tech_pvt->profile->url), TAG_END());
break;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
case SWITCH_MESSAGE_INDICATE_PROGRESS: {
struct private_object *tech_pvt;
switch_channel_t *channel = NULL;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
tech_pvt = switch_core_session_get_private(session);
assert(tech_pvt != NULL);
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
if (!switch_test_flag(tech_pvt, TFLAG_EARLY_MEDIA) && !switch_test_flag(tech_pvt, TFLAG_ANS)) {
switch_set_flag_locked(tech_pvt, TFLAG_EARLY_MEDIA);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "Asked to send early media by %s\n", msg->from);
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
if (switch_test_flag(tech_pvt, TFLAG_NOMEDIA)) {
tech_absorb_sdp(tech_pvt);
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
/* Transmit 183 Progress with SDP */
tech_choose_port(tech_pvt);
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
set_local_sdp(tech_pvt, NULL, 0, NULL, 0);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
activate_rtp(tech_pvt);
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_INFO, "Ring SDP:\n%s\n", tech_pvt->local_sdp_str);
nua_respond(tech_pvt->nh,
SIP_183_SESSION_PROGRESS,
SIPTAG_CONTACT_STR(tech_pvt->profile->url),
SOATAG_USER_SDP_STR(tech_pvt->local_sdp_str),
SOATAG_AUDIO_AUX("cn telephone-event"),
TAG_END());
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
}
}
break;
default:
break;
}
return SWITCH_STATUS_SUCCESS;
}
static switch_status_t sofia_receive_event(switch_core_session_t *session, switch_event_t *event)
{
switch_channel_t *channel;
struct private_object *tech_pvt;
char *body;
nua_handle_t *msg_nh;
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
tech_pvt = switch_core_session_get_private(session);
assert(tech_pvt != NULL);
if (!(body = switch_event_get_body(event))) {
body = "";
}
if (tech_pvt->hash_key) {
msg_nh = nua_handle(tech_pvt->profile->nua, NULL,
SIPTAG_FROM_STR(tech_pvt->chat_from),
NUTAG_URL(tech_pvt->chat_to),
SIPTAG_TO_STR(tech_pvt->chat_to),
SIPTAG_CONTACT_STR(tech_pvt->profile->url),
TAG_END());
nua_message(msg_nh,
SIPTAG_CONTENT_TYPE_STR("text/html"),
SIPTAG_PAYLOAD_STR(body),
TAG_END());
}
return SWITCH_STATUS_SUCCESS;
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
static const switch_io_routines_t sofia_io_routines = {
/*.outgoing_channel */ sofia_outgoing_channel,
/*.answer_channel */ sofia_answer_channel,
/*.read_frame */ sofia_read_frame,
/*.write_frame */ sofia_write_frame,
/*.kill_channel */ sofia_kill_channel,
/*.waitfor_read */ sofia_waitfor_read,
/*.waitfor_read */ sofia_waitfor_write,
/*.send_dtmf*/ sofia_send_dtmf,
/*.receive_message*/ sofia_receive_message,
/*.receive_event*/ sofia_receive_event
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
};
static const switch_state_handler_table_t sofia_event_handlers = {
/*.on_init */ sofia_on_init,
/*.on_ring */ sofia_on_ring,
/*.on_execute */ sofia_on_execute,
/*.on_hangup */ sofia_on_hangup,
/*.on_loopback */ sofia_on_loopback,
/*.on_transmit */ sofia_on_transmit
};
static const switch_endpoint_interface_t sofia_endpoint_interface = {
/*.interface_name */ "sofia",
/*.io_routines */ &sofia_io_routines,
/*.event_handlers */ &sofia_event_handlers,
/*.private */ NULL,
/*.next */ NULL
};
static const switch_chat_interface_t sofia_chat_interface = {
/*.name */ SOFIA_CHAT_PROTO,
/*.chat_send */ chat_send,
};
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
static const switch_loadable_module_interface_t sofia_module_interface = {
/*.module_name */ modname,
/*.endpoint_interface */ &sofia_endpoint_interface,
/*.timer_interface */ NULL,
/*.dialplan_interface */ NULL,
/*.codec_interface */ NULL,
/*.application_interface */ NULL,
/*.api_interface */ NULL,
/*.file_interface */ NULL,
/*.speech_interface */ NULL,
/*.directory_interface */ NULL,
/*.chat_interface */ &sofia_chat_interface
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
};
static void logger(void *logarg, char const *fmt, va_list ap)
{
char *data = NULL;
if (fmt) {
#ifdef HAVE_VASPRINTF
int ret;
ret = vasprintf(&data, fmt, ap);
if ((ret == -1) || !data) {
return;
}
#else
data = (char *) malloc(2048);
if (data) {
vsnprintf(data, 2048, fmt, ap);
} else {
return;
}
#endif
}
switch_log_printf(SWITCH_CHANNEL_LOG_CLEAN, SWITCH_LOG_CONSOLE, (char*) "%s", data);
free(data);
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
static switch_status_t sofia_outgoing_channel(switch_core_session_t *session, switch_caller_profile_t *outbound_profile,
switch_core_session_t **new_session, switch_memory_pool_t *pool)
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
{
switch_status_t status = SWITCH_STATUS_FALSE;
switch_core_session_t *nsession;
char *data, *profile_name, *dest;
sofia_profile_t *profile;
switch_caller_profile_t *caller_profile = NULL;
private_object_t *tech_pvt = NULL;
switch_channel_t *nchannel;
char *host;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
*new_session = NULL;
if (!(nsession = switch_core_session_request(&sofia_endpoint_interface, pool))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Error Creating Session\n");
goto done;
}
if (!(tech_pvt = (struct private_object *) switch_core_session_alloc(nsession, sizeof(*tech_pvt)))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Error Creating Session\n");
terminate_session(&nsession, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER, __LINE__);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
goto done;
}
data = switch_core_session_strdup(nsession, outbound_profile->destination_number);
profile_name = data;
if (!(dest = strchr(profile_name, '/'))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Invalid URL\n");
terminate_session(&nsession, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER, __LINE__);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
goto done;
}
*dest++ = '\0';
if (!(profile = (sofia_profile_t *) switch_core_hash_find(globals.profile_hash, profile_name))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Invalid Profile\n");
terminate_session(&nsession, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER, __LINE__);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
goto done;
}
if ((host = strchr(dest, '%'))) {
char buf[128];
*host = '@';
tech_pvt->e_dest = switch_core_session_strdup(nsession, dest);
*host++ = '\0';
if (find_reg_url(profile, dest, host, buf, sizeof(buf))) {
tech_pvt->dest = switch_core_session_strdup(nsession, buf);
} else {
terminate_session(&nsession, SWITCH_CAUSE_NO_ROUTE_DESTINATION, __LINE__);
goto done;
}
} else if (!strchr(dest, '@')) {
char buf[128];
tech_pvt->e_dest = switch_core_session_strdup(nsession, dest);
if (find_reg_url(profile, dest, profile_name, buf, sizeof(buf))) {
tech_pvt->dest = switch_core_session_strdup(nsession, buf);
} else {
terminate_session(&nsession, SWITCH_CAUSE_NO_ROUTE_DESTINATION, __LINE__);
goto done;
}
} else {
tech_pvt->dest = switch_core_session_alloc(nsession, strlen(dest) + 5);
snprintf(tech_pvt->dest, strlen(dest) + 5, "sip:%s", dest);
}
attach_private(nsession, profile, tech_pvt, dest);
nchannel = switch_core_session_get_channel(nsession);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
caller_profile = switch_caller_profile_clone(nsession, outbound_profile);
switch_channel_set_caller_profile(nchannel, caller_profile);
switch_channel_set_flag(nchannel, CF_OUTBOUND);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
switch_set_flag_locked(tech_pvt, TFLAG_OUTBOUND);
switch_channel_set_state(nchannel, CS_INIT);
switch_channel_set_variable(nchannel, "endpoint_disposition", "OUTBOUND");
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
*new_session = nsession;
status = SWITCH_STATUS_SUCCESS;
if (session) {
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
//char *val;
//switch_channel_t *channel = switch_core_session_get_channel(session);
switch_ivr_transfer_variable(session, nsession, SOFIA_REPLACES_HEADER);
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
done:
return status;
}
static uint8_t negotiate_sdp(switch_core_session_t *session, sdp_session_t *sdp)
{
uint8_t match = 0;
private_object_t *tech_pvt;
sdp_media_t *m;
sdp_attribute_t *a;
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
switch_channel_t *channel;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
tech_pvt = switch_core_session_get_private(session);
assert(tech_pvt != NULL);
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
channel = switch_core_session_get_channel(session);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if ((tech_pvt->origin = switch_core_session_strdup(session, (char *) sdp->sdp_origin->o_username))) {
if (strstr(tech_pvt->origin, "CiscoSystemsSIP-GW-UserAgent")) {
switch_set_flag_locked(tech_pvt, TFLAG_BUGGY_2833);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Activate Buggy RFC2833 Mode!\n");
}
}
for (a = sdp->sdp_attributes; a; a = a->a_next) {
if (!strcasecmp(a->a_name, "sendonly")) {
switch_set_flag_locked(tech_pvt, TFLAG_SIP_HOLD);
} else if (!strcasecmp(a->a_name, "sendrecv")) {
switch_clear_flag_locked(tech_pvt, TFLAG_SIP_HOLD);
}
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
for (m = sdp->sdp_media; m ; m = m->m_next) {
if (m->m_type == sdp_media_audio) {
sdp_rtpmap_t *map;
for (map = m->m_rtpmaps; map; map = map->rm_next) {
int32_t i;
if (!strcasecmp(map->rm_encoding, "telephone-event")) {
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
tech_pvt->te = (switch_payload_t)map->rm_pt;
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
for (i = 0; i < tech_pvt->num_codecs; i++) {
const switch_codec_implementation_t *imp = tech_pvt->codecs[i];
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Codec Compare [%s:%d]/[%s:%d]\n",
map->rm_encoding, map->rm_pt, imp->iananame, imp->ianacode);
if (map->rm_pt < 96) {
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
match = (map->rm_pt == imp->ianacode) ? 1 : 0;
} else {
match = strcasecmp(map->rm_encoding, imp->iananame) ? 0 : 1;
}
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (match && (map->rm_rate == imp->samples_per_second)) {
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
char tmp[50];
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
tech_pvt->rm_encoding = switch_core_session_strdup(session, (char *)map->rm_encoding);
tech_pvt->pt = (switch_payload_t)map->rm_pt;
tech_pvt->rm_rate = map->rm_rate;
tech_pvt->codec_ms = imp->microseconds_per_frame / 1000;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
tech_pvt->remote_sdp_audio_ip = switch_core_session_strdup(session, (char *)sdp->sdp_connection->c_address);
tech_pvt->rm_fmtp = switch_core_session_strdup(session, (char *)map->rm_fmtp);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
tech_pvt->remote_sdp_audio_port = (switch_port_t)m->m_port;
tech_pvt->agreed_pt = (switch_payload_t)map->rm_pt;
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
snprintf(tmp, sizeof(tmp), "%d", tech_pvt->remote_sdp_audio_port);
switch_channel_set_variable(channel, SWITCH_REMOTE_MEDIA_IP_VARIABLE, tech_pvt->remote_sdp_audio_ip);
switch_channel_set_variable(channel, SWITCH_REMOTE_MEDIA_PORT_VARIABLE, tmp);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
} else {
match = 0;
}
}
if (match) {
if (tech_set_codec(tech_pvt, 1) != SWITCH_STATUS_SUCCESS) {
match = 0;
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
}
}
}
}
return match;
}
// map sip responses to QSIG cause codes ala RFC4497 section 8.4.4
static switch_call_cause_t sip_cause_to_freeswitch(int status) {
switch (status) {
case 200:
return SWITCH_CAUSE_NORMAL_CLEARING;
case 401:
case 402:
case 403:
case 407:
case 603:
return SWITCH_CAUSE_CALL_REJECTED;
case 404:
case 485:
case 604:
return SWITCH_CAUSE_UNALLOCATED;
case 408:
case 504:
return SWITCH_CAUSE_RECOVERY_ON_TIMER_EXPIRE;
case 410:
return SWITCH_CAUSE_NUMBER_CHANGED;
case 413:
case 414:
case 416:
case 420:
case 421:
case 423:
case 505:
case 513:
return SWITCH_CAUSE_INTERWORKING;
case 480:
return SWITCH_CAUSE_NO_USER_RESPONSE;
case 400:
case 481:
case 500:
case 503:
return SWITCH_CAUSE_NORMAL_TEMPORARY_FAILURE;
case 486:
case 600:
return SWITCH_CAUSE_USER_BUSY;
case 484:
return SWITCH_CAUSE_INVALID_NUMBER_FORMAT;
case 488:
case 606:
return SWITCH_CAUSE_BEARERCAPABILITY_NOTIMPL;
case 502:
return SWITCH_CAUSE_NETWORK_OUT_OF_ORDER;
case 405:
return SWITCH_CAUSE_SERVICE_UNAVAILABLE;
case 406:
case 415:
case 501:
return SWITCH_CAUSE_SERVICE_NOT_IMPLEMENTED;
case 482:
case 483:
return SWITCH_CAUSE_EXCHANGE_ROUTING_ERROR;
case 487:
return SWITCH_CAUSE_ORIGINATOR_CANCEL;
default:
return SWITCH_CAUSE_NORMAL_UNSPECIFIED;
}
}
static void set_hash_key(char *hash_key, int32_t len, sip_t const *sip)
{
snprintf(hash_key, len, "%s%s%s",
(char *) sip->sip_from->a_url->url_user,
(char *) sip->sip_from->a_url->url_host,
(char *) sip->sip_to->a_url->url_user
);
#if 0
/* nicer one we cant use in both directions >=0 */
snprintf(hash_key, len, "%s%s%s%s%s%s",
(char *) sip->sip_to->a_url->url_user,
(char *) sip->sip_to->a_url->url_host,
(char *) sip->sip_to->a_url->url_params,
(char *) sip->sip_from->a_url->url_user,
(char *) sip->sip_from->a_url->url_host,
(char *) sip->sip_from->a_url->url_params
);
#endif
}
static void set_chat_hash(private_object_t *tech_pvt, sip_t const *sip)
{
char hash_key[256] = "";
char buf[512];
if (!sip || tech_pvt->hash_key) {
return;
}
if (find_reg_url(tech_pvt->profile, (char *) sip->sip_from->a_url->url_user, (char *) sip->sip_from->a_url->url_host, buf, sizeof(buf))) {
tech_pvt->chat_from = sip_header_as_string(tech_pvt->home, (void *)sip->sip_to);
tech_pvt->chat_to = switch_core_session_strdup(tech_pvt->session, buf);
set_hash_key(hash_key, sizeof(hash_key), sip);
} else {
return;
}
tech_pvt->hash_key = switch_core_session_strdup(tech_pvt->session, hash_key);
switch_core_hash_insert(tech_pvt->profile->chat_hash, tech_pvt->hash_key, tech_pvt);
}
static void sip_i_message(int status,
char const *phrase,
nua_t *nua,
sofia_profile_t *profile,
nua_handle_t *nh,
sofia_private_t *sofia_private,
sip_t const *sip,
tagi_t tags[])
{
if (sip) {
sip_from_t const *from = sip->sip_from;
char *from_user = NULL;
char *from_host = NULL;
sip_to_t const *to = sip->sip_to;
char *to_user = NULL;
char *to_host = NULL;
sip_subject_t const *sip_subject = sip->sip_subject;
sip_payload_t *payload = sip->sip_payload;
const char *subject = "n/a";
char *msg = NULL;
if (sip->sip_content_type) {
if (strstr((char*)sip->sip_content_type->c_subtype, "composing")) {
return;
}
}
if (from) {
from_user = (char *) from->a_url->url_user;
from_host = (char *) from->a_url->url_host;
}
if (to) {
to_user = (char *) to->a_url->url_user;
to_host = (char *) to->a_url->url_host;
}
if (!to_user) {
return;
}
if (payload) {
msg = payload->pl_data;
}
if (sip_subject) {
subject = sip_subject->g_value;
}
if (nh) {
char hash_key[512];
private_object_t *tech_pvt;
switch_channel_t *channel;
switch_event_t *event;
char *to_addr;
char *from_addr;
char *p;
char *full_from;
char proto[512] = SOFIA_CHAT_PROTO;
full_from = sip_header_as_string(profile->home, (void *)sip->sip_from);
if ((p=strchr(to_user, '+'))) {
switch_copy_string(proto, to_user, sizeof(proto));
p = strchr(proto, '+');
*p++ = '\0';
if ((to_addr = strdup(p))) {
if((p = strchr(to_addr, '+'))) {
*p = '@';
}
}
} else {
to_addr = switch_mprintf("%s@%s", to_user, to_host);
}
from_addr = switch_mprintf("%s@%s", from_user, from_host);
set_hash_key(hash_key, sizeof(hash_key), sip);
if ((tech_pvt = (private_object_t *) switch_core_hash_find(profile->chat_hash, hash_key))) {
channel = switch_core_session_get_channel(tech_pvt->session);
if (switch_event_create(&event, SWITCH_EVENT_MESSAGE) == SWITCH_STATUS_SUCCESS) {
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "proto", SOFIA_CHAT_PROTO);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "login", "%s", profile->url);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "from", "%s", tech_pvt->hash_key);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "to", "%s", to_addr);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "subject", "SIMPLE MESSAGE");
if (msg) {
switch_event_add_body(event, msg);
}
if (switch_core_session_queue_event(tech_pvt->session, &event) != SWITCH_STATUS_SUCCESS) {
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "delivery-failure", "true");
switch_event_fire(&event);
}
}
} else {
switch_chat_interface_t *ci;
if ((ci = switch_loadable_module_get_chat_interface(proto))) {
ci->chat_send(SOFIA_CHAT_PROTO, from_addr, to_addr, "", msg, full_from);
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Invaid Chat Interface [%s]!\n", proto);
}
}
switch_safe_free(to_addr);
switch_safe_free(from_addr);
if (full_from) {
su_free(profile->home, full_from);
}
}
}
}
static void pass_sdp(private_object_t *tech_pvt, char *sdp)
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
{
char *val;
switch_channel_t *channel;
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
switch_core_session_t *other_session;
switch_channel_t *other_channel;
channel = switch_core_session_get_channel(tech_pvt->session);
assert(channel != NULL);
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
if ((val = switch_channel_get_variable(channel, SWITCH_ORIGINATOR_VARIABLE)) && (other_session = switch_core_session_locate(val))) {
other_channel = switch_core_session_get_channel(other_session);
assert(other_channel != NULL);
if (!switch_channel_get_variable(other_channel, SWITCH_B_SDP_VARIABLE)) {
switch_channel_set_variable(other_channel, SWITCH_B_SDP_VARIABLE, sdp);
}
if (!switch_test_flag(tech_pvt, TFLAG_CHANGE_MEDIA) && (
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
switch_channel_test_flag(other_channel, CF_OUTBOUND) &&
switch_channel_test_flag(other_channel, CF_NOMEDIA) &&
switch_channel_test_flag(channel, CF_OUTBOUND) &&
switch_channel_test_flag(channel, CF_NOMEDIA))) {
switch_ivr_nomedia(val, SMF_FORCE);
switch_set_flag_locked(tech_pvt, TFLAG_CHANGE_MEDIA);
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
}
switch_core_session_rwunlock(other_session);
}
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
static void sip_i_state(int status,
char const *phrase,
nua_t *nua,
sofia_profile_t *profile,
nua_handle_t *nh,
sofia_private_t *sofia_private,
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
sip_t const *sip,
tagi_t tags[])
{
char const *l_sdp = NULL, *r_sdp = NULL;
int offer_recv = 0, answer_recv = 0, offer_sent = 0, answer_sent = 0;
int ss_state = nua_callstate_init;
switch_channel_t *channel = NULL;
private_object_t *tech_pvt = NULL;
switch_core_session_t *session = sofia_private ? sofia_private->session : NULL;
const char *replaces_str = NULL;
char *uuid;
switch_core_session_t *other_session = NULL;
switch_channel_t *other_channel = NULL;
char st[80] = "";
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
tl_gets(tags,
NUTAG_CALLSTATE_REF(ss_state),
NUTAG_OFFER_RECV_REF(offer_recv),
NUTAG_ANSWER_RECV_REF(answer_recv),
NUTAG_OFFER_SENT_REF(offer_sent),
NUTAG_ANSWER_SENT_REF(answer_sent),
SIPTAG_REPLACES_STR_REF(replaces_str),
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
SOATAG_LOCAL_SDP_STR_REF(l_sdp),
SOATAG_REMOTE_SDP_STR_REF(r_sdp),
TAG_END());
if (session) {
channel = switch_core_session_get_channel(session);
assert(channel != NULL);
tech_pvt = switch_core_session_get_private(session);
assert(tech_pvt != NULL);
tech_pvt->nh = nh;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Channel %s entering state [%s]\n",
switch_channel_get_name(channel),
nua_callstate_name(ss_state));
if (r_sdp) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Remote SDP:\n%s\n", r_sdp);
tech_pvt->remote_sdp_str = switch_core_session_strdup(session, (char *)r_sdp);
switch_channel_set_variable(channel, SWITCH_R_SDP_VARIABLE, (char *) r_sdp);
pass_sdp(tech_pvt, (char *) r_sdp);
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
}
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
if (status == 988) {
return;
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
switch ((enum nua_callstate)ss_state) {
case nua_callstate_init:
break;
case nua_callstate_authenticating:
break;
case nua_callstate_calling:
break;
case nua_callstate_proceeding:
if (channel) {
if (status == 180) {
if (switch_test_flag(tech_pvt, TFLAG_NOMEDIA)) {
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
if ((uuid = switch_channel_get_variable(channel, SWITCH_BRIDGE_VARIABLE)) && (other_session = switch_core_session_locate(uuid))) {
switch_core_session_message_t msg;
msg.message_id = SWITCH_MESSAGE_INDICATE_RINGING;
msg.from = __FILE__;
switch_core_session_receive_message(other_session, &msg);
switch_core_session_rwunlock(other_session);
}
} else {
switch_core_session_message_t *msg;
if ((msg = malloc(sizeof(*msg)))) {
memset(msg, 0, sizeof(*msg));
msg->message_id = SWITCH_MESSAGE_INDICATE_RINGING;
msg->from = __FILE__;
switch_core_session_queue_message(session, msg);
switch_set_flag(msg, SCSMF_DYNAMIC);
}
}
}
if (r_sdp) {
if (switch_test_flag(tech_pvt, TFLAG_NOMEDIA)) {
switch_set_flag_locked(tech_pvt, TFLAG_EARLY_MEDIA);
switch_channel_set_flag(channel, CF_EARLY_MEDIA);
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
if ((uuid = switch_channel_get_variable(channel, SWITCH_BRIDGE_VARIABLE)) && (other_session = switch_core_session_locate(uuid))) {
other_channel = switch_core_session_get_channel(other_session);
switch_channel_pre_answer(other_channel);
switch_core_session_rwunlock(other_session);
}
return;
} else {
sdp_parser_t *parser = sdp_parse(tech_pvt->home, r_sdp, (int)strlen(r_sdp), 0);
sdp_session_t *sdp;
uint8_t match = 0;
if (tech_pvt->num_codecs) {
if ((sdp = sdp_session(parser))) {
match = negotiate_sdp(session, sdp);
}
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (parser) {
sdp_parser_free(parser);
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (match) {
tech_choose_port(tech_pvt);
activate_rtp(tech_pvt);
switch_channel_set_variable(channel, "endpoint_disposition", "EARLY MEDIA");
switch_set_flag_locked(tech_pvt, TFLAG_EARLY_MEDIA);
switch_channel_set_flag(channel, CF_EARLY_MEDIA);
return;
}
switch_channel_set_variable(channel, "endpoint_disposition", "NO CODECS");
nua_respond(nh, SIP_488_NOT_ACCEPTABLE,
TAG_END());
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
}
}
break;
case nua_callstate_completing:
nua_ack(nh, TAG_END());
break;
case nua_callstate_received:
if (channel) {
if (r_sdp) {
if (switch_test_flag(tech_pvt, TFLAG_NOMEDIA)) {
switch_channel_set_variable(channel, "endpoint_disposition", "RECEIVED_NOMEDIA");
switch_channel_set_state(channel, CS_INIT);
switch_set_flag_locked(tech_pvt, TFLAG_READY);
switch_core_session_thread_launch(session);
return;
} else {
sdp_parser_t *parser = sdp_parse(tech_pvt->home, r_sdp, (int)strlen(r_sdp), 0);
sdp_session_t *sdp;
uint8_t match = 0;
if (tech_pvt->num_codecs) {
if ((sdp = sdp_session(parser))) {
match = negotiate_sdp(session, sdp);
}
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (parser) {
sdp_parser_free(parser);
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (match) {
nua_handle_t *bnh;
sip_replaces_t *replaces;
switch_channel_set_variable(channel, "endpoint_disposition", "RECEIVED");
switch_channel_set_state(channel, CS_INIT);
switch_set_flag_locked(tech_pvt, TFLAG_READY);
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
switch_core_session_thread_launch(session);
if (replaces_str && (replaces = sip_replaces_make(tech_pvt->home, replaces_str)) && (bnh = nua_handle_by_replaces(nua, replaces))) {
sofia_private_t *b_private;
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Processing Replaces Attended Transfer\n");
while (switch_channel_get_state(channel) < CS_EXECUTE) {
switch_yield(10000);
}
if ((b_private = nua_handle_magic(bnh))) {
char *br_b = switch_channel_get_variable(channel, SWITCH_BRIDGE_VARIABLE);
char *br_a = switch_core_session_get_uuid(b_private->session);
if (br_b) {
switch_ivr_uuid_bridge(br_a, br_b);
switch_channel_set_variable(channel, "endpoint_disposition", "ATTENDED_TRANSFER");
switch_channel_hangup(channel, SWITCH_CAUSE_ATTENDED_TRANSFER);
} else {
switch_channel_set_variable(channel, "endpoint_disposition", "ATTENDED_TRANSFER_ERROR");
switch_channel_hangup(channel, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER);
}
} else {
switch_channel_set_variable(channel, "endpoint_disposition", "ATTENDED_TRANSFER_ERROR");
switch_channel_hangup(channel, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER);
}
nua_handle_unref(bnh);
}
return;
}
switch_channel_set_variable(channel, "endpoint_disposition", "NO CODECS");
nua_respond(nh, SIP_488_NOT_ACCEPTABLE,
TAG_END());
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
}
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
case nua_callstate_early:
break;
case nua_callstate_completed:
if (tech_pvt && r_sdp) {
if (r_sdp) {
if (switch_test_flag(tech_pvt, TFLAG_NOMEDIA)) {
return;
} else {
sdp_parser_t *parser = sdp_parse(tech_pvt->home, r_sdp, (int)strlen(r_sdp), 0);
sdp_session_t *sdp;
uint8_t match = 0;
if (tech_pvt->num_codecs) {
if ((sdp = sdp_session(parser))) {
match = negotiate_sdp(session, sdp);
}
}
if (match) {
tech_choose_port(tech_pvt);
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
set_local_sdp(tech_pvt, NULL, 0, NULL, 0);
switch_set_flag_locked(tech_pvt, TFLAG_REINVITE);
activate_rtp(tech_pvt);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Processing Reinvite\n");
if (parser) {
sdp_parser_free(parser);
}
}
}
}
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
case nua_callstate_ready:
if (nh == tech_pvt->nh2) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Cheater Reinvite!\n");
switch_set_flag_locked(tech_pvt, TFLAG_REINVITE);
tech_pvt->nh = tech_pvt->nh2;
tech_pvt->nh2 = NULL;
tech_choose_port(tech_pvt);
activate_rtp(tech_pvt);
return;
}
if (channel) {
if (r_sdp) {
if (switch_test_flag(tech_pvt, TFLAG_NOMEDIA)) {
switch_set_flag_locked(tech_pvt, TFLAG_ANS);
switch_channel_set_flag(channel, CF_ANSWERED);
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
if ((uuid = switch_channel_get_variable(channel, SWITCH_BRIDGE_VARIABLE)) && (other_session = switch_core_session_locate(uuid))) {
other_channel = switch_core_session_get_channel(other_session);
switch_channel_answer(other_channel);
switch_core_session_rwunlock(other_session);
}
return;
} else {
sdp_parser_t *parser = sdp_parse(tech_pvt->home, r_sdp, (int)strlen(r_sdp), 0);
sdp_session_t *sdp;
uint8_t match = 0;
if (tech_pvt->num_codecs) {
if ((sdp = sdp_session(parser))) {
match = negotiate_sdp(session, sdp);
}
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (parser) {
sdp_parser_free(parser);
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (match) {
switch_set_flag_locked(tech_pvt, TFLAG_ANS);
switch_channel_set_variable(channel, "endpoint_disposition", "ANSWER");
tech_choose_port(tech_pvt);
activate_rtp(tech_pvt);
switch_channel_set_flag(channel, CF_ANSWERED);
return;
}
switch_channel_set_variable(channel, "endpoint_disposition", "NO CODECS");
nua_respond(nh, SIP_488_NOT_ACCEPTABLE, TAG_END());
}
} else if (switch_test_flag(tech_pvt, TFLAG_EARLY_MEDIA)) {
switch_set_flag_locked(tech_pvt, TFLAG_ANS);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
switch_channel_set_variable(channel, "endpoint_disposition", "ANSWER");
switch_channel_set_flag(channel, CF_ANSWERED);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
return;
} //else probably an ack
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
case nua_callstate_terminating:
break;
case nua_callstate_terminated:
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (session) {
if (switch_test_flag(tech_pvt, TFLAG_RWLOCK)) {
switch_core_session_rwunlock(session);
switch_clear_flag(tech_pvt, TFLAG_RWLOCK);
}
if (!switch_test_flag(tech_pvt, TFLAG_BYE)) {
switch_set_flag_locked(tech_pvt, TFLAG_BYE);
if (switch_test_flag(tech_pvt, TFLAG_NOHUP)) {
switch_clear_flag_locked(tech_pvt, TFLAG_NOHUP);
} else {
snprintf(st, sizeof(st), "%d", status);
switch_channel_set_variable(channel, "sip_term_status", st);
terminate_session(&session, sip_cause_to_freeswitch(status), __LINE__);
}
}
tech_pvt->nh = NULL;
}
if (nh) {
nua_handle_destroy(nh);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
}
break;
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
}
static char *get_auth_data(char *dbname, char *nonce, char *npassword, size_t len, switch_mutex_t *mutex)
{
switch_core_db_t *db;
switch_core_db_stmt_t *stmt;
char *sql = NULL, *ret = NULL;
if (mutex) {
switch_mutex_lock(mutex);
}
if (!(db = switch_core_db_open_file(dbname))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error Opening DB %s\n", dbname);
goto end;
}
sql = switch_mprintf("select passwd from sip_authentication where nonce='%q'", nonce);
if (switch_core_db_prepare(db, sql, -1, &stmt, 0)) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Statement Error!\n");
goto fail;
} else {
int running = 1;
int colcount;
while (running < 5000) {
int result = switch_core_db_step(stmt);
if (result == SQLITE_ROW) {
if ((colcount = switch_core_db_column_count(stmt))) {
switch_copy_string(npassword, (char *)switch_core_db_column_text(stmt, 0), len);
ret = npassword;
}
break;
} else if (result == SQLITE_BUSY) {
running++;
switch_yield(1000);
continue;
}
break;
}
switch_core_db_finalize(stmt);
}
fail:
switch_core_db_close(db);
end:
if (mutex) {
switch_mutex_unlock(mutex);
}
if (sql) {
switch_safe_free(sql);
}
return ret;
}
typedef enum {
REG_REGISTER,
REG_INVITE
} sofia_regtype_t;
static uint8_t handle_register(nua_t *nua,
sofia_profile_t *profile,
nua_handle_t *nh,
sip_t const *sip,
sofia_regtype_t regtype,
char *key,
uint32_t keylen)
{
sip_from_t const *from = sip->sip_from;
sip_expires_t const *expires = sip->sip_expires;
sip_authorization_t const *authorization = sip->sip_authorization;
sip_contact_t const *contact = sip->sip_contact;
switch_xml_t domain, xml, user, param, xparams;
char params[1024] = "";
char *sql;
switch_event_t *s_event;
char *from_user = (char *) from->a_url->url_user;
char *from_host = (char *) from->a_url->url_host;
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
char contact_str[1024] = "";
char buf[512];
char *passwd = NULL;
uint8_t stale = 0, ret = 0, forbidden = 0;
auth_res_t auth_res;
long exptime = 60;
switch_event_t *event;
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
char *rpid = "unknown";
const char *display = "\"user\"";
if (contact) {
char *port = (char *) contact->m_url->url_port;
display = contact->m_display;
if (switch_strlen_zero(display)) {
if (from) {
display = from->a_display;
if (switch_strlen_zero(display)) {
display = "\"user\"";
}
}
} else {
display = "\"user\"";
}
if (!port) {
port = "5060";
}
if (contact->m_url->url_params) {
snprintf(contact_str, sizeof(contact_str), "%s <sip:%s@%s:%s;%s>",
display, contact->m_url->url_user, contact->m_url->url_host, port, contact->m_url->url_params);
} else {
snprintf(contact_str, sizeof(contact_str), "%s <sip:%s@%s:%s>",
display, contact->m_url->url_user, contact->m_url->url_host, port);
}
}
if (expires) {
exptime = expires->ex_delta;
} else if (contact->m_expires) {
exptime = atol(contact->m_expires);
}
if (regtype == REG_REGISTER) {
authorization = sip->sip_authorization;
} else if (regtype == REG_INVITE) {
authorization = sip->sip_proxy_authorization;
}
if ((profile->pflags & PFLAG_BLIND_REG)) {
goto reg;
}
if (authorization) {
auth_res = parse_auth(profile, authorization, (char *)sip->sip_request->rq_method_name, key, keylen);
if (auth_res != AUTH_OK && auth_res != AUTH_STALE) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "send %s for [%s@%s]\n",
forbidden ? "forbidden" : "challange",
from_user, from_host);
if (auth_res == AUTH_FORBIDDEN) {
nua_respond(nh, SIP_403_FORBIDDEN, NUTAG_WITH_THIS(nua), TAG_END());
} else {
nua_respond(nh, SIP_401_UNAUTHORIZED, NUTAG_WITH_THIS(nua), TAG_END());
}
return 1;
}
}
if (!authorization || stale) {
snprintf(params, sizeof(params), "from_user=%s&from_host=%s&contact=%s",
from_user,
from_host,
contact_str
);
if (switch_xml_locate("directory", "domain", "name", from_host, &xml, &domain, params) != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "can't find domain for [%s@%s]\n", from_user, from_host);
nua_respond(nh, SIP_401_UNAUTHORIZED, NUTAG_WITH_THIS(nua), SIPTAG_CONTACT(contact), TAG_END());
return 1;
}
if (!(user = switch_xml_find_child(domain, "user", "id", from_user))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "can't find user [%s@%s]\n", from_user, from_host);
nua_respond(nh, SIP_401_UNAUTHORIZED, NUTAG_WITH_THIS(nua), SIPTAG_CONTACT(contact), TAG_END());
switch_xml_free(xml);
return 1;
}
if (!(xparams = switch_xml_child(user, "params"))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "can't find params for user [%s@%s]\n", from_user, from_host);
nua_respond(nh, SIP_401_UNAUTHORIZED, NUTAG_WITH_THIS(nua), SIPTAG_CONTACT(contact), TAG_END());
switch_xml_free(xml);
return 1;
}
for (param = switch_xml_child(xparams, "param"); param; param = param->next) {
char *var = (char *) switch_xml_attr_soft(param, "name");
char *val = (char *) switch_xml_attr_soft(param, "value");
//switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "param [%s]=[%s]\n", var, val);
if (!strcasecmp(var, "password")) {
passwd = val;
}
}
if (passwd) {
switch_uuid_t uuid;
char uuid_str[SWITCH_UUID_FORMATTED_LENGTH + 1];
char *sql, *auth_str;
su_md5_t ctx;
char hexdigest[2 * SU_MD5_DIGEST_SIZE + 1];
char *input;
input = switch_mprintf("%s:%s:%s", from_user, from_host, passwd);
su_md5_init(&ctx);
su_md5_strupdate(&ctx, input);
su_md5_hexdigest(&ctx, hexdigest);
su_md5_deinit(&ctx);
switch_safe_free(input);
switch_uuid_get(&uuid);
switch_uuid_format(uuid_str, &uuid);
sql = switch_mprintf("delete from sip_authentication where user='%q' and host='%q';\n"
"insert into sip_authentication values('%q','%q','%q','%q', %ld)",
from_user,
from_host,
from_user,
from_host,
hexdigest,
uuid_str,
time(NULL) + 60);
auth_str = switch_mprintf("Digest realm=\"%q\", nonce=\"%q\",%s algorithm=MD5, qop=\"auth\"", from_host, uuid_str,
stale ? " stale=\"true\"," : "");
if (regtype == REG_REGISTER) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Requesting Registration from: [%s@%s]\n", from_user, from_host);
nua_respond(nh, SIP_401_UNAUTHORIZED,
NUTAG_WITH_THIS(nua),
SIPTAG_WWW_AUTHENTICATE_STR(auth_str),
TAG_END());
} else if (regtype == REG_INVITE) {
nua_respond(nh, SIP_407_PROXY_AUTH_REQUIRED,
NUTAG_WITH_THIS(nua),
SIPTAG_PROXY_AUTHENTICATE_STR(auth_str),
TAG_END());
}
execute_sql(profile->dbname, sql, profile->ireg_mutex);
switch_safe_free(sql);
switch_safe_free(auth_str);
ret = 1;
} else {
ret = 0;
}
switch_xml_free(xml);
if (ret) {
return ret;
}
}
reg:
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
if (exptime) {
if (!find_reg_url(profile, from_user, from_host, buf, sizeof(buf))) {
sql = switch_mprintf("insert into sip_registrations values ('%q','%q','%q','Registered', '%q', %ld)",
from_user,
from_host,
contact_str,
rpid,
(long) time(NULL) + (long)exptime);
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
} else {
sql = switch_mprintf("update sip_registrations set contact='%q', expires=%ld, rpid='%q' where user='%q' and host='%q'",
contact_str,
(long) time(NULL) + (long)exptime,
rpid,
from_user,
from_host);
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
}
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
if (switch_event_create_subclass(&s_event, SWITCH_EVENT_CUSTOM, MY_EVENT_REGISTER) == SWITCH_STATUS_SUCCESS) {
switch_event_add_header(s_event, SWITCH_STACK_BOTTOM, "profile-name", "%s", profile->name);
switch_event_add_header(s_event, SWITCH_STACK_BOTTOM, "from-user", "%s", from_user);
switch_event_add_header(s_event, SWITCH_STACK_BOTTOM, "from-host", "%s", from_host);
switch_event_add_header(s_event, SWITCH_STACK_BOTTOM, "contact", "%s", contact_str);
switch_event_add_header(s_event, SWITCH_STACK_BOTTOM, "rpid", "%s", rpid);
switch_event_add_header(s_event, SWITCH_STACK_BOTTOM, "expires", "%ld", (long)exptime);
switch_event_fire(&s_event);
}
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
if (sql) {
execute_sql(profile->dbname, sql, profile->ireg_mutex);
switch_safe_free(sql);
sql = NULL;
}
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Register:\nFrom: [%s@%s]\nContact: [%s]\nExpires: [%ld]\n",
from_user,
from_host,
contact_str,
(long)exptime
);
if (switch_event_create(&event, SWITCH_EVENT_PRESENCE_IN) == SWITCH_STATUS_SUCCESS) {
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "proto", "sip");
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "login", "%s", profile->url);
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "rpid", "%s", rpid);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "from", "%s@%s", from_user, from_host);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "status", "Registered");
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "event_type", "presence");
switch_event_fire(&event);
}
} else {
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
if ((sql = switch_mprintf("delete from sip_subscriptions where user='%q' and host='%q'", from_user, from_host))) {
execute_sql(profile->dbname, sql, profile->ireg_mutex);
switch_safe_free(sql);
sql = NULL;
}
if (switch_event_create(&event, SWITCH_EVENT_PRESENCE_OUT) == SWITCH_STATUS_SUCCESS) {
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "proto", "sip");
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "login", "%s", profile->url);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "from", "%s+%s@%s", SOFIA_CHAT_PROTO, from_user, from_host);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "status", "unavailable");
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "rpid", rpid);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "event_type", "presence");
switch_event_fire(&event);
}
}
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
if (switch_event_create(&event, SWITCH_EVENT_ROSTER) == SWITCH_STATUS_SUCCESS) {
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "proto", "sip");
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "from", "%s@%s", from_user, from_host);
switch_event_fire(&event);
}
if (regtype == REG_REGISTER) {
nua_respond(nh, SIP_200_OK, SIPTAG_CONTACT(contact),
NUTAG_WITH_THIS(nua),
TAG_END());
return 1;
}
return 0;
}
static int sub_reg_callback(void *pArg, int argc, char **argv, char **columnNames)
{
sofia_profile_t *profile = (sofia_profile_t *) pArg;
//char *proto = argv[0];
char *user = argv[1];
char *host = argv[2];
switch_event_t *event;
char *status = NULL;
if (switch_strlen_zero(status)) {
status = "Available";
}
if (switch_event_create(&event, SWITCH_EVENT_PRESENCE_IN) == SWITCH_STATUS_SUCCESS) {
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "proto", SOFIA_CHAT_PROTO);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "login", "%s", profile->url);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "from", "%s@%s", user, host);
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "status", "%s", status);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "event_type", "presence");
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "event_subtype", "probe");
switch_event_fire(&event);
}
return 0;
}
static int resub_callback(void *pArg, int argc, char **argv, char **columnNames)
{
sofia_profile_t *profile = (sofia_profile_t *) pArg;
char *user = argv[0];
char *host = argv[1];
char *status = argv[2];
char *rpid = argv[3];
char *proto = argv[4];
switch_event_t *event;
if (switch_strlen_zero(proto)) {
proto = NULL;
}
if (switch_event_create(&event, SWITCH_EVENT_PRESENCE_IN) == SWITCH_STATUS_SUCCESS) {
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "proto", proto ? proto : SOFIA_CHAT_PROTO);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "login", "%s", profile->url);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "from", "%s@%s", user, host);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "status", "%s", status);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "rpid", "%s", rpid);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "event_type", "presence");
switch_event_fire(&event);
}
return 0;
}
static int sub_callback(void *pArg, int argc, char **argv, char **columnNames)
{
sofia_profile_t *profile = (sofia_profile_t *) pArg;
char *pl;
char *id, *note;
uint32_t in = atoi(argv[0]);
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
char *status = argv[1];
char *rpid = argv[2];
char *proto = argv[3];
char *user = argv[4];
char *host = argv[5];
char *sub_to_user = argv[6];
char *sub_to_host = argv[7];
char *event = argv[8];
char *contact = argv[9];
char *callid = argv[10];
char *full_from = argv[11];
char *full_via = argv[12];
nua_handle_t *nh;
char *to;
char *open;
char *tmp;
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
if (!rpid) {
rpid = "unknown";
}
if (in) {
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
note = switch_mprintf("<dm:note>%s</dm:note>", status);
open = "open";
} else {
note = NULL;
open = "closed";
}
if (strcasecmp(proto, SOFIA_CHAT_PROTO)) {
/*encapsulate*/
id = switch_mprintf("sip:%s+%s+%s@%s", proto, sub_to_user, sub_to_host, host);
} else {
id = switch_mprintf("sip:%s@%s", sub_to_user, sub_to_host);
}
to = switch_mprintf("sip:%s@%s", user, host);
pl = switch_mprintf("<?xml version='1.0' encoding='UTF-8'?>\r\n"
"<presence xmlns='urn:ietf:params:xml:ns:pidf'\r\n"
"xmlns:dm='urn:ietf:params:xml:ns:pidf:data-model'\r\n"
"xmlns:rpid='urn:ietf:params:xml:ns:pidf:rpid'\r\n"
"xmlns:c='urn:ietf:params:xml:ns:pidf:cipid'\r\n"
"entity='pres:%s'>\r\n"
"<tuple id='t6a5ed77e'>\r\n"
"<status>\r\n"
"<basic>%s</basic>\r\n"
"</status>\r\n"
"</tuple>\r\n"
"<dm:person id='p06360c4a'>\r\n"
"<rpid:activities>\r\n"
"<rpid:%s/>\r\n"
"</rpid:activities>%s</dm:person>\r\n"
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
"</presence>", id, open, rpid, note);
nh = nua_handle(profile->nua, NULL, TAG_END());
tmp = contact;
contact = get_url_from_contact(tmp, 0);
nua_notify(nh,
NUTAG_URL(contact),
SIPTAG_TO_STR(full_from),
SIPTAG_FROM_STR(to),
SIPTAG_CONTACT_STR(profile->url),
SIPTAG_CALL_ID_STR(callid),
SIPTAG_VIA_STR(full_via),
SIPTAG_SUBSCRIPTION_STATE_STR("active;expires=3600"),
SIPTAG_EVENT_STR(event),
SIPTAG_CONTENT_TYPE_STR("application/pidf+xml"),
SIPTAG_PAYLOAD_STR(pl),
TAG_END());
switch_safe_free(id);
switch_safe_free(note);
switch_safe_free(pl);
switch_safe_free(to);
return 0;
}
static void sip_i_subscribe(int status,
char const *phrase,
nua_t *nua,
sofia_profile_t *profile,
nua_handle_t *nh,
sofia_private_t *sofia_private,
sip_t const *sip,
tagi_t tags[])
{
if (sip) {
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
long exp, exp_raw;
sip_to_t const *to = sip->sip_to;
sip_from_t const *from = sip->sip_from;
sip_contact_t const *contact = sip->sip_contact;
char *from_user = NULL;
char *from_host = NULL;
char *to_user = NULL;
char *to_host = NULL;
char *sql, *event = NULL;
char *proto = "sip";
char *d_user = NULL;
char *contact_str = "";
char *call_id = NULL;
char *to_str = NULL;
char *full_from = NULL;
char *full_via = NULL;
switch_core_db_t *db;
char *errmsg;
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
char *sstr;
const char *display = "\"user\"";
switch_event_t *sevent;
if (contact) {
char *port = (char *) contact->m_url->url_port;
display = contact->m_display;
if (switch_strlen_zero(display)) {
if (from) {
display = from->a_display;
if (switch_strlen_zero(display)) {
display = "\"user\"";
}
}
} else {
display = "\"user\"";
}
if (!port) {
port = "5060";
}
if (contact->m_url->url_params) {
contact_str = switch_mprintf("%s <sip:%s@%s:%s;%s>",
display,
contact->m_url->url_user,
contact->m_url->url_host, port, contact->m_url->url_params);
} else {
contact_str = switch_mprintf("%s <sip:%s@%s:%s>",
display,
contact->m_url->url_user,
contact->m_url->url_host, port);
}
}
if (to) {
to_str = switch_mprintf("sip:%s@%s", to->a_url->url_user, to->a_url->url_host);//, to->a_url->url_port);
}
if (to) {
to_user = (char *) to->a_url->url_user;
to_host = (char *) to->a_url->url_host;
}
if (strstr(to_user, "ext+") || strstr(to_user, "user+") || strstr(to_user, "conf+")) {
char proto[80];
char *p;
switch_copy_string(proto, to_user, sizeof(proto));
if ((p = strchr(proto, '+'))) {
*p = '\0';
}
if (switch_event_create(&sevent, SWITCH_EVENT_PRESENCE_IN) == SWITCH_STATUS_SUCCESS) {
switch_event_add_header(sevent, SWITCH_STACK_BOTTOM, "proto", SOFIA_CHAT_PROTO);
switch_event_add_header(sevent, SWITCH_STACK_BOTTOM, "login", "%s", profile->name);
switch_event_add_header(sevent, SWITCH_STACK_BOTTOM, "from", "%s@%s", to_user, to_host);
switch_event_add_header(sevent, SWITCH_STACK_BOTTOM, "rpid", "unknown");
switch_event_add_header(sevent, SWITCH_STACK_BOTTOM, "status", "Click To Call");
switch_event_fire(&sevent);
}
}
if (strchr(to_user, '+')) {
char *h;
if ((proto = (d_user = strdup(to_user)))) {
if ((to_user = strchr(d_user, '+'))) {
*to_user++ = '\0';
if ((h = strchr(to_user, '+')) || (h = strchr(to_user, '@'))) {
*h++ = '\0';
to_host = h;
}
}
}
if (!(proto && to_user && to_host)) {
nua_respond(nh, SIP_404_NOT_FOUND, NUTAG_WITH_THIS(nua), TAG_END());
goto end;
}
}
call_id = sip_header_as_string(profile->home, (void *)sip->sip_call_id);
event = sip_header_as_string(profile->home, (void *)sip->sip_event);
full_from = sip_header_as_string(profile->home, (void *)sip->sip_from);
full_via = sip_header_as_string(profile->home, (void *)sip->sip_via);
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
exp_raw = (sip->sip_expires ? sip->sip_expires->ex_delta : 3600);
exp = (long) time(NULL) + exp_raw;
if ((sql = switch_mprintf("delete from sip_subscriptions where "
"proto='%q' and user='%q' and host='%q' and sub_to_user='%q' and sub_to_host='%q' and event='%q';\n"
"insert into sip_subscriptions values ('%q','%q','%q','%q','%q','%q','%q','%q','%q','%q',%ld)",
proto,
from_user,
from_host,
to_user,
to_host,
event,
proto,
from_user,
from_host,
to_user,
to_host,
event,
contact_str,
call_id,
full_from,
full_via,
exp
))) {
execute_sql(profile->dbname, sql, profile->ireg_mutex);
switch_safe_free(sql);
}
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
sstr = switch_mprintf("active;expires=%ld", exp_raw);
nua_respond(nh, SIP_202_ACCEPTED,
NUTAG_WITH_THIS(nua),
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
SIPTAG_SUBSCRIPTION_STATE_STR(sstr),
SIPTAG_FROM(sip->sip_to),
SIPTAG_TO(sip->sip_from),
SIPTAG_CONTACT_STR(to_str),
TAG_END());
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
switch_safe_free(sstr);
if (!(db = switch_core_db_open_file(profile->dbname))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error Opening DB %s\n", profile->dbname);
goto end;
}
if ((sql = switch_mprintf("select * from sip_subscriptions where user='%q' and host='%q'",
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
to_user, to_host, to_user, to_host))) {
switch_mutex_lock(profile->ireg_mutex);
switch_core_db_exec(db, sql, sub_reg_callback, profile, &errmsg);
switch_mutex_unlock(profile->ireg_mutex);
switch_safe_free(sql);
}
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
switch_core_db_close(db);
end:
if (event) {
su_free(profile->home, event);
}
if (call_id) {
su_free(profile->home, call_id);
}
if (full_from) {
su_free(profile->home, full_from);
}
if (full_via) {
su_free(profile->home, full_via);
}
switch_safe_free(d_user);
switch_safe_free(to_str);
switch_safe_free(contact_str);
}
}
static void sip_r_subscribe(int status,
char const *phrase,
nua_t *nua,
sofia_profile_t *profile,
nua_handle_t *nh,
sofia_private_t *sofia_private,
sip_t const *sip,
tagi_t tags[])
{
}
/*---------------------------------------*/
static void sip_i_refer(nua_t *nua,
sofia_profile_t *profile,
nua_handle_t *nh,
sofia_private_t *sofia_private,
sip_t const *sip,
tagi_t tags[])
{
/* Incoming refer */
sip_from_t const *from;
sip_to_t const *to;
sip_refer_to_t const *refer_to;
switch_core_session_t *session = sofia_private ? sofia_private->session : NULL;
if (session) {
private_object_t *tech_pvt = NULL;
char *etmp = NULL, *exten = NULL;
switch_channel_t *channel_a = NULL, *channel_b = NULL;
tech_pvt = switch_core_session_get_private(session);
channel_a = switch_core_session_get_channel(session);
if (!sip->sip_cseq || !(etmp = switch_mprintf("refer;id=%u", sip->sip_cseq->cs_seq))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Memory Error!\n");
goto done;
}
if (switch_channel_test_flag(channel_a, CF_NOMEDIA)) {
nua_notify(tech_pvt->nh, SIPTAG_CONTENT_TYPE_STR("message/sipfrag"),
SIPTAG_PAYLOAD_STR("SIP/2.0 403 Forbidden"),
SIPTAG_EVENT_STR(etmp),
TAG_END());
goto done;
}
from = sip->sip_from;
to = sip->sip_to;
if ((refer_to = sip->sip_refer_to)) {
if (profile->pflags & PFLAG_FULL_ID) {
exten = switch_mprintf("%s@%s", (char *) refer_to->r_url->url_user, (char *) refer_to->r_url->url_host);
} else {
exten = (char *) refer_to->r_url->url_user;
}
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Process REFER to [%s@%s]\n", exten, (char *) refer_to->r_url->url_host);
if (refer_to->r_url->url_headers) {
sip_replaces_t *replaces;
nua_handle_t *bnh;
char *rep;
if ((rep = strchr(refer_to->r_url->url_headers, '='))) {
char *br_a = NULL, *br_b = NULL;
char *buf;
rep++;
if ((buf = switch_core_session_alloc(session, strlen(rep) + 1))) {
rep = url_unescape(buf, (const char *) rep);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Replaces: [%s]\n", rep);
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Memory Error!\n");
goto done;
}
if ((replaces = sip_replaces_make(tech_pvt->home, rep)) && (bnh = nua_handle_by_replaces(nua, replaces))) {
sofia_private_t *b_private;
switch_channel_set_variable(channel_a, SOFIA_REPLACES_HEADER, rep);
if ((b_private = nua_handle_magic(bnh))) {
channel_b = switch_core_session_get_channel(b_private->session);
br_a = switch_channel_get_variable(channel_a, SWITCH_BRIDGE_VARIABLE);
br_b = switch_channel_get_variable(channel_b, SWITCH_BRIDGE_VARIABLE);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "Attended Transfer [%s][%s]\n", br_a, br_b);
if (br_a && br_b) {
switch_ivr_uuid_bridge(br_a, br_b);
switch_channel_set_variable(channel_b, "endpoint_disposition", "ATTENDED_TRANSFER");
nua_notify(tech_pvt->nh, SIPTAG_CONTENT_TYPE_STR("message/sipfrag"),
SIPTAG_PAYLOAD_STR("SIP/2.0 200 OK"),
SIPTAG_EVENT_STR(etmp),
TAG_END());
} else {
private_object_t *tech_pvt_b = (private_object_t *) switch_core_session_get_private(b_private->session);
if (!br_a && !br_b) {
switch_set_flag_locked(tech_pvt, TFLAG_NOHUP);
switch_set_flag_locked(tech_pvt_b, TFLAG_XFER);
tech_pvt_b->xferto = switch_core_session_strdup(b_private->session, switch_core_session_get_uuid(session));
} else if (!br_a && br_b) {
switch_core_session_t *bsession;
if ((bsession = switch_core_session_locate(br_b))) {
private_object_t *b_tech_pvt = switch_core_session_get_private(bsession);
switch_channel_t *b_channel = switch_core_session_get_channel(bsession);
private_object_t *bp_tech_pvt = switch_core_session_get_private(b_private->session);
switch_core_session_get_uuid(b_private->session);
switch_set_flag_locked(tech_pvt, TFLAG_NOHUP);
switch_channel_clear_state_handler(b_channel, NULL);
switch_channel_set_state_flag(b_channel, CF_TRANSFER);
switch_channel_set_state(b_channel, CS_TRANSMIT);
switch_set_flag_locked(tech_pvt, TFLAG_REINVITE);
tech_pvt->local_sdp_audio_ip = switch_core_session_strdup(session, bp_tech_pvt->local_sdp_audio_ip);
tech_pvt->local_sdp_audio_port = bp_tech_pvt->local_sdp_audio_port;
tech_pvt->remote_sdp_audio_ip = switch_core_session_strdup(session, b_tech_pvt->remote_sdp_audio_ip);
tech_pvt->remote_sdp_audio_port = b_tech_pvt->remote_sdp_audio_port;
activate_rtp(tech_pvt);
b_tech_pvt->kick = switch_core_session_strdup(bsession, switch_core_session_get_uuid(session));
switch_core_session_rwunlock(bsession);
}
switch_channel_hangup(channel_b, SWITCH_CAUSE_ATTENDED_TRANSFER);
}
nua_notify(tech_pvt->nh, SIPTAG_CONTENT_TYPE_STR("message/sipfrag"),
SIPTAG_PAYLOAD_STR("SIP/2.0 200 OK"),
SIPTAG_EVENT_STR(etmp),
TAG_END());
}
}
nua_handle_unref(bnh);
} else { /* the other channel is on a different box, we have to go find them */
if (exten && (br_a = switch_channel_get_variable(channel_a, SWITCH_BRIDGE_VARIABLE))) {
switch_core_session_t *asession;
switch_channel_t *channel = switch_core_session_get_channel(session);
if ((asession = switch_core_session_locate(br_a))) {
switch_core_session_t *tsession;
switch_call_cause_t cause = SWITCH_CAUSE_NORMAL_CLEARING;
uint32_t timeout = 60;
char *tuuid_str;
channel = switch_core_session_get_channel(asession);
exten = switch_mprintf("sofia/%s/%s@%s:%s",
profile->name,
(char *) refer_to->r_url->url_user,
(char *) refer_to->r_url->url_host,
refer_to->r_url->url_port
);
switch_channel_set_variable(channel, SOFIA_REPLACES_HEADER, rep);
if (switch_ivr_originate(asession,
&tsession,
&cause,
exten,
timeout,
&noop_state_handler,
NULL,
NULL,
NULL) != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Cannot Create Outgoing Channel! [%s]\n", exten);
nua_notify(tech_pvt->nh, SIPTAG_CONTENT_TYPE_STR("message/sipfrag"),
SIPTAG_PAYLOAD_STR("SIP/2.0 403 Forbidden"),
SIPTAG_EVENT_STR(etmp),
TAG_END());
goto done;
}
switch_core_session_rwunlock(asession);
tuuid_str = switch_core_session_get_uuid(tsession);
switch_ivr_uuid_bridge(br_a, tuuid_str);
switch_channel_set_variable(channel_a, "endpoint_disposition", "ATTENDED_TRANSFER");
nua_notify(tech_pvt->nh, SIPTAG_CONTENT_TYPE_STR("message/sipfrag"),
SIPTAG_PAYLOAD_STR("SIP/2.0 200 OK"),
SIPTAG_EVENT_STR(etmp),
TAG_END());
} else {
goto error;
}
} else { error:
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Invalid Transfer! [%s]\n", br_a);
switch_channel_set_variable(channel_a, "endpoint_disposition", "ATTENDED_TRANSFER_ERROR");
nua_notify(tech_pvt->nh, SIPTAG_CONTENT_TYPE_STR("message/sipfrag"),
SIPTAG_PAYLOAD_STR("SIP/2.0 403 Forbidden"),
SIPTAG_EVENT_STR(etmp),
TAG_END());
}
}
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Cannot parse Replaces!\n");
}
goto done;
}
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Missing Refer-To\n");
goto done;
}
if (exten) {
switch_channel_t *channel = switch_core_session_get_channel(session);
char *br;
if ((br = switch_channel_get_variable(channel, SWITCH_BRIDGE_VARIABLE))) {
switch_core_session_t *bsession;
if ((bsession = switch_core_session_locate(br))) {
channel = switch_core_session_get_channel(bsession);
switch_channel_set_variable(channel, "TRANSFER_FALLBACK", (char *) from->a_user);
switch_ivr_session_transfer(bsession, exten, profile->dialplan, profile->context);
switch_core_session_rwunlock(bsession);
}
switch_channel_set_variable(channel, "endpoint_disposition", "BLIND_TRANSFER");
nua_notify(tech_pvt->nh, SIPTAG_CONTENT_TYPE_STR("message/sipfrag"),
SIPTAG_PAYLOAD_STR("SIP/2.0 200 OK"),
SIPTAG_EVENT_STR(etmp),
TAG_END());
} else {
exten = switch_mprintf("sip:%s@%s:%s",
(char *) refer_to->r_url->url_user,
(char *) refer_to->r_url->url_host,
refer_to->r_url->url_port);
tech_pvt->dest = switch_core_session_strdup(session, exten);
switch_set_flag_locked(tech_pvt, TFLAG_NOHUP);
nua_notify(tech_pvt->nh, SIPTAG_CONTENT_TYPE_STR("message/sipfrag"),
SIPTAG_PAYLOAD_STR("SIP/2.0 200 OK"),
SIPTAG_EVENT_STR(etmp),
TAG_END());
do_xfer_invite(session);
}
}
done:
if (exten && strchr(exten, '@')) {
switch_safe_free(exten);
}
if (etmp) {
switch_safe_free(etmp);
}
}
}
static void sip_i_publish(nua_t *nua,
sofia_profile_t *profile,
nua_handle_t *nh,
sofia_private_t *sofia_private,
sip_t const *sip,
tagi_t tags[])
{
if (sip) {
sip_from_t const *from = sip->sip_from;
char *from_user = NULL;
char *from_host = NULL;
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
char *rpid = "unknown";
sip_payload_t *payload = sip->sip_payload;
char *event_type;
if (from) {
from_user = (char *) from->a_url->url_user;
from_host = (char *) from->a_url->url_host;
}
if (payload) {
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
switch_xml_t xml, note, person, tuple, status, basic, act;
switch_event_t *event;
uint8_t in = 0;
char *sql;
if ((xml = switch_xml_parse_str(payload->pl_data, strlen(payload->pl_data)))) {
char *status_txt = "", *note_txt = "";
if ((tuple = switch_xml_child(xml, "tuple")) && (status = switch_xml_child(tuple, "status")) && (basic = switch_xml_child(status, "basic"))) {
status_txt = basic->txt;
}
if ((person = switch_xml_child(xml, "dm:person")) && (note = switch_xml_child(person, "dm:note"))) {
note_txt = note->txt;
}
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
if (person && (act = switch_xml_child(person, "rpid:activities"))) {
if ((rpid = strchr(act->child->name, ':'))) {
rpid++;
} else {
rpid = act->child->name;
}
}
if (!strcasecmp(status_txt, "open")) {
if (switch_strlen_zero(note_txt)) {
note_txt = "Available";
}
in = 1;
} else if (!strcasecmp(status_txt, "closed")) {
if (switch_strlen_zero(note_txt)) {
note_txt = "Unavailable";
}
}
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
if ((sql = switch_mprintf("update sip_registrations set status='%q',rpid='%q' where user='%q' and host='%q'",
note_txt, rpid, from_user, from_host))) {
execute_sql(profile->dbname, sql, profile->ireg_mutex);
switch_safe_free(sql);
}
event_type = sip_header_as_string(profile->home, (void *)sip->sip_event);
if (in) {
if (switch_event_create(&event, SWITCH_EVENT_PRESENCE_IN) == SWITCH_STATUS_SUCCESS) {
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "proto", SOFIA_CHAT_PROTO);
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "rpid", "%s", rpid);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "login", "%s", profile->url);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "from", "%s@%s", from_user, from_host);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "status", "%s", note_txt);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "event_type", "%s", event_type);
switch_event_fire(&event);
}
} else {
if (switch_event_create(&event, SWITCH_EVENT_PRESENCE_OUT) == SWITCH_STATUS_SUCCESS) {
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "proto", SOFIA_CHAT_PROTO);
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "rpid", "%s", rpid);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "login", "%s", profile->url);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "from", "%s@%s", from_user, from_host);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "event_type", "%s", event_type);
switch_event_fire(&event);
}
}
if (event_type) {
su_free(profile->home, event_type);
}
switch_xml_free(xml);
}
}
}
nua_respond(nh, SIP_200_OK, NUTAG_WITH_THIS(nua), TAG_END());
}
static void sip_i_info(nua_t *nua,
sofia_profile_t *profile,
nua_handle_t *nh,
switch_core_session_t *session,
sip_t const *sip,
tagi_t tags[]) {
//placeholder for string searching
char *signal_ptr;
//Try and find signal information in the payload
signal_ptr = strstr(sip->sip_payload->pl_data, "Signal=");
//See if we found a match
if(signal_ptr) {
struct private_object *tech_pvt = NULL;
switch_channel_t *channel = NULL;
char dtmf_digit[2] = {0,0};
//Get the channel
channel = switch_core_session_get_channel(session);
//Barf if we didn't get it
assert(channel != NULL);
//make sure we have our privates
tech_pvt = switch_core_session_get_private(session);
//Barf if we didn't get it
assert(tech_pvt != NULL);
//move signal_ptr where we need it (right past Signal=)
signal_ptr = signal_ptr + 7;
//put the digit somewhere we can muck with
strncpy(dtmf_digit, signal_ptr, 1);
//queue it up
switch_channel_queue_dtmf(channel, dtmf_digit);
//print debug info
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "INFO DTMF(%s)\n", dtmf_digit);
} else { //unknown info type
sip_from_t const *from;
from = sip->sip_from;
//print in the logs if something comes through we don't understand
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Unknown INFO Recieved: %s%s" URL_PRINT_FORMAT "[%s]\n",
from->a_display ? from->a_display : "", from->a_display ? " " : "",
URL_PRINT_ARGS(from->a_url), sip->sip_payload->pl_data);
}
return;
}
static void sip_i_invite(nua_t *nua,
sofia_profile_t *profile,
nua_handle_t *nh,
sofia_private_t *sofia_private,
sip_t const *sip,
tagi_t tags[])
{
switch_core_session_t *session = sofia_private ? sofia_private->session : NULL;
char key[128] = "";
sip_unknown_t *un;
if (!session) {
if ((profile->pflags & PFLAG_AUTH_CALLS)) {
if (handle_register(nua, profile, nh, sip, REG_INVITE, key, sizeof(key))) {
return;
}
}
if ((session = switch_core_session_request(&sofia_endpoint_interface, NULL))) {
private_object_t *tech_pvt = NULL;
switch_channel_t *channel = NULL;
sip_from_t const *from = sip->sip_from;
sip_to_t const *to = sip->sip_to;
char *displayname;
char *username, *to_username = NULL;
char *url_user = (char *) from->a_url->url_user;
char *to_user, *to_host;
if (!(tech_pvt = (private_object_t *) switch_core_session_alloc(session, sizeof(private_object_t)))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Hey where is my memory pool?\n");
terminate_session(&session, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER, __LINE__);
return;
}
if (!switch_strlen_zero(key)) {
tech_pvt->key = switch_core_session_strdup(session, key);
}
to_user = (char *) to->a_url->url_user;
to_host = (char *) to->a_url->url_host;
if (switch_strlen_zero(to_user)) { /* if sofia doesnt parse the To: right, we'll have to do it */
if ((to_user = sip_header_as_string(tech_pvt->home, (sip_header_t *) to))) {
char *p;
if (*to_user == '<') {
to_user++;
}
if ((p = strchr((to_user += 4), '@'))) {
*p++ = '\0';
to_host = p;
if ((p = strchr(to_host, '>'))) {
*p = '\0';
}
}
}
}
if (switch_strlen_zero(url_user)) {
url_user = "service";
}
if (!switch_strlen_zero(from->a_display)) {
displayname = switch_core_session_strdup(session, (char *) from->a_display);
if (*displayname == '"') {
char *p;
displayname++;
if ((p = strchr(displayname, '"'))) {
*p = '\0';
}
}
} else {
displayname = url_user;
}
if (!(username = switch_mprintf("%s@%s", url_user, (char *) from->a_url->url_host))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Memory Error!\n");
return;
}
if (profile->pflags & PFLAG_FULL_ID) {
if (!(to_username = switch_mprintf("%s@%s", (char *) to_user, (char *) to_host))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Memory Error!\n");
switch_safe_free(username);
return;
}
}
attach_private(session, profile, tech_pvt, username);
switch_core_session_read_lock(session);
switch_set_flag(tech_pvt, TFLAG_RWLOCK);
channel = switch_core_session_get_channel(session);
switch_channel_set_variable(channel, "endpoint_disposition", "INBOUND CALL");
set_chat_hash(tech_pvt, sip);
switch_channel_set_variable(channel, "sip_from_user", (char *) from->a_url->url_user);
if (from->a_url->url_user && *from->a_url->url_user == '+') {
switch_channel_set_variable(channel, "sip_from_user_stripped", (char *)(from->a_url->url_user+1));
} else {
switch_channel_set_variable(channel, "sip_from_user_stripped", (char *)from->a_url->url_user);
}
switch_channel_set_variable(channel, "sip_from_host", (char *) from->a_url->url_host);
if ((tech_pvt->caller_profile = switch_caller_profile_new(switch_core_session_get_pool(session),
(char *) from->a_url->url_user,
profile->dialplan,
displayname,
(char *) from->a_url->url_user,
(char *) from->a_url->url_host,
NULL,
NULL,
NULL,
(char *)modname,
(profile->context && !strcasecmp(profile->context, "_domain_")) ?
(char *) from->a_url->url_host : profile->context,
to_username ? to_username : (char *) to_user
)) != 0) {
for (un=sip->sip_unknown; un; un=un->un_next) {
if (!strncasecmp(un->un_name, "Alert-Info", 10)) {
if (!switch_strlen_zero(un->un_value)) {
switch_channel_set_variable(channel, "alert_info", (char *)un->un_value);
}
// Loop thru Known Headers Here so we can do something with them
} else if (!strncasecmp(un->un_name, "Remote-Party-ID", 15)) {
int argc, x, screen = 1;
char *mydata, *argv[10] = { 0 };
if (!switch_strlen_zero(un->un_value)) {
if ((mydata = strdup(un->un_value))) {
argc = switch_separate_string(mydata, ';', argv, (sizeof(argv) / sizeof(argv[0])));
// Do We really need this at this time
// clid_uri = argv[0];
for (x=1; x < argc && argv[x]; x++){
// we dont need to do anything with party yet we should only be seeing party=calling here anyway
// maybe thats a dangerous assumption bit oh well yell at me later
// if (!strncasecmp(argv[x], "party", 5)) {
// party = argv[x];
// } else
if (!strncasecmp(argv[x], "privacy=", 8)) {
char *arg = argv[x] + 9;
if (!strcasecmp(arg, "no")) {
switch_clear_flag(tech_pvt->caller_profile, SWITCH_CPF_HIDE_NAME);
switch_clear_flag(tech_pvt->caller_profile, SWITCH_CPF_HIDE_NUMBER);
} else if (!strcasecmp(arg, "yes")) {
switch_set_flag(tech_pvt->caller_profile, SWITCH_CPF_HIDE_NAME | SWITCH_CPF_HIDE_NUMBER);
} else if (!strcasecmp(arg, "full")) {
switch_set_flag(tech_pvt->caller_profile, SWITCH_CPF_HIDE_NAME | SWITCH_CPF_HIDE_NUMBER);
} else if (!strcasecmp(arg, "name")) {
switch_set_flag(tech_pvt->caller_profile, SWITCH_CPF_HIDE_NAME);
} else if (!strcasecmp(arg, "number")) {
switch_set_flag(tech_pvt->caller_profile, SWITCH_CPF_HIDE_NUMBER);
}
} else if (!strncasecmp(argv[x], "screen=", 7) && screen > 0) {
char *arg = argv[x] + 8;
if (!strcasecmp(arg, "no")) {
screen = 0;
switch_clear_flag(tech_pvt->caller_profile, SWITCH_CPF_SCREEN);
}
}
}
free(mydata);
}
}
break;
}
}
switch_channel_set_caller_profile(channel, tech_pvt->caller_profile);
switch_safe_free(username);
switch_safe_free(to_username);
}
tech_pvt->sofia_private.session = session;
nua_handle_bind(nh, &tech_pvt->sofia_private);
}
}
}
static void sip_i_register(nua_t *nua,
sofia_profile_t *profile,
nua_handle_t *nh,
sofia_private_t *sofia_private,
sip_t const *sip,
tagi_t tags[])
{
handle_register(nua, profile, nh, sip, REG_REGISTER, NULL, 0);
}
static void sip_i_options(int status,
char const *phrase,
nua_t *nua,
sofia_profile_t *profile,
nua_handle_t *nh,
sofia_private_t *sofia_private,
sip_t const *sip,
tagi_t tags[])
{
nua_respond(nh, SIP_200_OK,
//SOATAG_USER_SDP_STR(tech_pvt->local_sdp_str),
//SOATAG_AUDIO_AUX("cn telephone-event"),
//NUTAG_INCLUDE_EXTRA_SDP(1),
TAG_END());
}
static void sip_r_register(int status,
char const *phrase,
nua_t *nua,
sofia_profile_t *profile,
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
nua_handle_t *nh,
sofia_private_t *sofia_private,
sip_t const *sip,
tagi_t tags[])
{
outbound_reg_t *oreg = NULL;
sip_www_authenticate_t const *authenticate = NULL;
switch_core_session_t *session = sofia_private ? sofia_private->session : NULL;
char const *realm = NULL;
char *p = NULL, *duprealm = NULL, *qrealm = NULL;
char const *scheme = NULL;
int index;
char *cur;
if (session) {
private_object_t *tech_pvt;
if ((tech_pvt = switch_core_session_get_private(session)) && switch_test_flag(tech_pvt, TFLAG_REFER)) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "received reply from refer\n");
return;
}
}
if (status == 401 || status == 407) {
if (sip->sip_www_authenticate) {
authenticate = sip->sip_www_authenticate;
} else if (sip->sip_proxy_authenticate) {
authenticate = sip->sip_proxy_authenticate;
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Missing Authenticate Header!\n");
return;
}
scheme = (char const *) authenticate->au_scheme;
if (authenticate->au_params) {
for(index = 0; (cur=(char*)authenticate->au_params[index]); index++) {
if ((realm = strstr(cur, "realm="))) {
realm += 6;
break;
}
}
}
if (!(scheme && realm)) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "No scheme and realm!\n");
}
}
if (!(scheme && realm)) {
return;
}
duprealm = strdup(realm);
qrealm = duprealm;
while(*qrealm && *qrealm == '"') {
*qrealm++;
}
if ((p = strchr(qrealm, '"'))) {
*p = '\0';
}
if (sofia_private) {
if (sofia_private->oreg) {
oreg = sofia_private->oreg;
} else if (profile) {
outbound_reg_t *oregp;
for (oregp = profile->registrations; oregp; oregp = oregp->next) {
if (scheme && qrealm && !strcasecmp(oregp->register_scheme, scheme) && !strcasecmp(oregp->register_realm, qrealm)) {
oreg = oregp;
break;
}
}
if (!oreg) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "No Match for Scheme [%s] Realm [%s]\n", scheme, qrealm);
}
}
}
switch_safe_free(duprealm);
if (!oreg) {
return;
}
if (status == 200) {
oreg->state = REG_STATE_REGISTER;
} else if (authenticate) {
char authentication[256] = "";
int ss_state;
if (realm) {
snprintf(authentication, sizeof(authentication), "%s:%s:%s:%s", scheme, realm,
oreg->register_username,
oreg->register_password);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Authenticating '%s' with '%s'.\n",
profile->username, authentication);
ss_state = nua_callstate_authenticating;
tl_gets(tags,
NUTAG_CALLSTATE_REF(ss_state),
SIPTAG_WWW_AUTHENTICATE_REF(authenticate),
TAG_END());
nua_authenticate(nh, SIPTAG_EXPIRES_STR(oreg->expires_str), NUTAG_AUTH(authentication), TAG_END());
}
}
}
static void event_callback(nua_event_t event,
int status,
char const *phrase,
nua_t *nua,
sofia_profile_t *profile,
nua_handle_t *nh,
sofia_private_t *sofia_private,
sip_t const *sip,
tagi_t tags[])
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
{
struct private_object *tech_pvt = NULL;
auth_res_t auth_res = AUTH_FORBIDDEN;
switch_core_session_t *session = sofia_private ? sofia_private->session : NULL;
if (session) {
if (switch_core_session_read_lock(session) != SWITCH_STATUS_SUCCESS) {
/* too late */
return;
}
tech_pvt = switch_core_session_get_private(session);
}
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
if (status != 100 && status != 200) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "event [%s] status [%d][%s] session: %s\n",
nua_event_name (event), status, phrase,
session ? switch_channel_get_name(switch_core_session_get_channel(session)) : "n/a"
);
}
if ((profile->pflags & PFLAG_AUTH_ALL) && tech_pvt && tech_pvt->key && sip) {
sip_authorization_t const *authorization = NULL;
if (sip->sip_authorization) {
authorization = sip->sip_authorization;
} else if (sip->sip_proxy_authorization) {
authorization = sip->sip_proxy_authorization;
}
if (authorization) {
auth_res = parse_auth(profile, authorization, (char *)sip->sip_request->rq_method_name, tech_pvt->key, strlen(tech_pvt->key));
}
if (auth_res != AUTH_OK) {
switch_channel_t *channel = switch_core_session_get_channel(tech_pvt->session);
switch_channel_hangup(channel, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER);
nua_respond(nh, SIP_401_UNAUTHORIZED, TAG_END());
goto done;
}
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
switch (event) {
case nua_r_shutdown:
//sip_r_shutdown(status, phrase, nua, profile, nh, sofia_private, sip, tags);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
case nua_r_get_params:
//sip_r_get_params(status, phrase, nua, profile, nh, sofia_private, sip, tags);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
case nua_r_invite:
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
case nua_r_register:
sip_r_register(status, phrase, nua, profile, nh, sofia_private, sip, tags);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
case nua_r_unregister:
//sip_r_unregister(status, phrase, nua, profile, nh, sofia_private, sip, tags);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
case nua_r_options:
//sip_r_options(status, phrase, nua, profile, nh, sofia_private, sip, tags);
break;
case nua_i_options:
sip_i_options(status, phrase, nua, profile, nh, sofia_private, sip, tags);
break;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
case nua_i_fork:
//sip_i_fork(status, phrase, nua, profile, nh, sofia_private, sip, tags);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
case nua_i_invite:
sip_i_invite(nua, profile, nh, sofia_private, sip, tags);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
case nua_i_publish:
sip_i_publish(nua, profile, nh, sofia_private, sip, tags);
break;
case nua_i_register:
sip_i_register (nua, profile, nh, sofia_private, sip, tags);
break;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
case nua_i_prack:
//sip_i_prack(nua, profile, nh, sofia_private, sip, tags);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
case nua_i_state:
sip_i_state(status, phrase, nua, profile, nh, sofia_private, sip, tags);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
case nua_r_bye:
//sip_r_bye(status, phrase, nua, profile, nh, sofia_private, sip, tags);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
case nua_i_bye:
//sip_i_bye(nua, profile, nh, sofia_private, sip, tags);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
case nua_i_message:
sip_i_message(status, phrase, nua, profile, nh, sofia_private, sip, tags);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
case nua_r_info:
//sip_r_info(status, phrase, nua, profile, nh, sofia_private, sip, tags);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
case nua_i_info:
sip_i_info(nua, profile, nh, session, sip, tags);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
case nua_r_refer:
//sip_r_refer(status, phrase, nua, profile, nh, sofia_private, sip, tags);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
case nua_i_refer:
sip_i_refer(nua, profile, nh, sofia_private, sip, tags);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
case nua_r_subscribe:
sip_r_subscribe(status, phrase, nua, profile, nh, sofia_private, sip, tags);
break;
case nua_i_subscribe:
sip_i_subscribe(status, phrase, nua, profile, nh, sofia_private, sip, tags);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
case nua_r_unsubscribe:
//sip_r_unsubscribe(status, phrase, nua, profile, nh, sofia_private, sip, tags);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
case nua_r_publish:
//sip_r_publish(status, phrase, nua, profile, nh, sofia_private, sip, tags);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
case nua_r_message:
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
case nua_r_notify:
if (nh) {
nua_handle_destroy(nh);
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
case nua_i_notify:
//sip_i_notify(nua, profile, nh, sofia_private, sip, tags);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
case nua_i_cancel:
//sip_i_cancel(nua, profile, nh, sofia_private, sip, tags);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
case nua_i_error:
//sip_i_error(nua, profile, nh, sofia_private, status, phrase, tags);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
case nua_i_active:
case nua_i_ack:
case nua_i_terminated:
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
case nua_r_set_params:
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
break;
default:
if (status > 100)
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "%s: unknown event %d: %03d %s\n",
nua_event_name (event), event, status, phrase);
else
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "%s: unknown event %d\n", nua_event_name (event), event);
break;
}
done:
if (session) {
switch_core_session_rwunlock(session);
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
}
static void unreg(sofia_profile_t *profile)
{
outbound_reg_t *oregp;
for (oregp = profile->registrations; oregp; oregp = oregp->next) {
nua_handle_destroy(oregp->nh);
}
}
static void check_oreg(sofia_profile_t *profile, time_t now)
{
outbound_reg_t *oregp;
for (oregp = profile->registrations; oregp; oregp = oregp->next) {
int ss_state = nua_callstate_authenticating;
reg_state_t ostate = oregp->state;
switch(ostate) {
case REG_STATE_REGISTER:
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "registered %s\n", oregp->name);
oregp->expires = now + oregp->freq;
oregp->state = REG_STATE_REGED;
break;
case REG_STATE_UNREGED:
if ((oregp->nh = nua_handle(oregp->profile->nua, NULL,
NUTAG_URL(oregp->register_proxy),
SIPTAG_TO_STR(oregp->register_to),
NUTAG_CALLSTATE_REF(ss_state),
SIPTAG_FROM_STR(oregp->register_from), TAG_END()))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "registering %s\n", oregp->name);
oregp->sofia_private.oreg = oregp;
nua_handle_bind(oregp->nh, &oregp->sofia_private);
nua_register(oregp->nh,
SIPTAG_FROM_STR(oregp->register_from),
SIPTAG_CONTACT_STR(oregp->register_from),
SIPTAG_EXPIRES_STR(oregp->expires_str),
NUTAG_REGISTRAR(oregp->register_proxy),
TAG_NULL());
oregp->retry = now + 10;
oregp->state = REG_STATE_TRYING;
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error registering %s\n", oregp->name);
oregp->state = REG_STATE_FAILED;
}
break;
case REG_STATE_TRYING:
if (oregp->retry && now >= oregp->retry) {
oregp->state = REG_STATE_UNREGED;
oregp->retry = 0;
}
break;
default:
if (oregp->expires && now >= oregp->expires) {
oregp->state = REG_STATE_UNREGED;
oregp->expires = 0;
}
break;
}
}
}
#define IREG_SECONDS 30
#define OREG_SECONDS 1
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
static void *SWITCH_THREAD_FUNC profile_thread_run(switch_thread_t *thread, void *obj)
{
sofia_profile_t *profile = (sofia_profile_t *) obj;
switch_memory_pool_t *pool;
sip_alias_node_t *node;
uint32_t ireg_loops = 0;
uint32_t oreg_loops = 0;
switch_core_db_t *db;
switch_event_t *s_event;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
profile->s_root = su_root_create(NULL);
profile->home = su_home_new(sizeof(*profile->home));
su_home_init(profile->home);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
profile->nua = nua_create(profile->s_root, /* Event loop */
event_callback, /* Callback for processing events */
profile, /* Additional data to pass to callback */
NUTAG_URL(profile->url),
NTATAG_UDP_MTU(65536),
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
TAG_END()); /* Last tag should always finish the sequence */
nua_set_params(profile->nua,
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
//NUTAG_EARLY_MEDIA(1),
NUTAG_AUTOANSWER(0),
NUTAG_AUTOALERT(0),
NUTAG_ALLOW("REGISTER"),
NUTAG_ALLOW("REFER"),
NUTAG_ALLOW("INFO"),
TAG_IF((profile->pflags & PFLAG_PRESENCE), NUTAG_ALLOW("PUBLISH")),
TAG_IF((profile->pflags & PFLAG_PRESENCE), NUTAG_ALLOW("NOTIFY")),
TAG_IF((profile->pflags & PFLAG_PRESENCE), NUTAG_ALLOW("SUBSCRIBE")),
TAG_IF((profile->pflags & PFLAG_PRESENCE), NUTAG_ENABLEMESSAGE(1)),
//TAG_IF((profile->pflags & PFLAG_PRESENCE), NUTAG_ALLOW_EVENTS("presence")),
//TAG_IF((profile->pflags & PFLAG_PRESENCE), NUTAG_ALLOW_EVENTS("presence.winfo")),
SIPTAG_SUPPORTED_STR("100rel, precondition"),
SIPTAG_USER_AGENT_STR(SOFIA_USER_AGENT),
TAG_END());
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
for (node = profile->aliases; node; node = node->next) {
node->nua = nua_create(profile->s_root, /* Event loop */
event_callback, /* Callback for processing events */
profile, /* Additional data to pass to callback */
NUTAG_URL(node->url),
TAG_END()); /* Last tag should always finish the sequence */
nua_set_params(node->nua,
NUTAG_EARLY_MEDIA(1),
NUTAG_AUTOANSWER(0),
NUTAG_AUTOALERT(0),
NUTAG_ALLOW("REGISTER"),
NUTAG_ALLOW("REFER"),
NUTAG_ALLOW("INFO"),
TAG_IF((profile->pflags & PFLAG_PRESENCE), NUTAG_ALLOW("PUBLISH")),
TAG_IF((profile->pflags & PFLAG_PRESENCE), NUTAG_ENABLEMESSAGE(1)),
SIPTAG_SUPPORTED_STR("100rel, precondition"),
SIPTAG_USER_AGENT_STR(SOFIA_USER_AGENT),
TAG_END());
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
}
if ((db = switch_core_db_open_file(profile->dbname))) {
switch_core_db_test_reactive(db, "select contact from sip_registrations", reg_sql);
switch_core_db_test_reactive(db, "select contact from sip_subscriptions", sub_sql);
switch_core_db_test_reactive(db, "select * from sip_authentication", auth_sql);
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Cannot Open SQL Database!\n");
return NULL;
}
switch_core_db_close(db);
switch_mutex_init(&profile->ireg_mutex, SWITCH_MUTEX_NESTED, profile->pool);
switch_mutex_init(&profile->oreg_mutex, SWITCH_MUTEX_NESTED, profile->pool);
ireg_loops = IREG_SECONDS;
oreg_loops = OREG_SECONDS;
if (switch_event_create(&s_event, SWITCH_EVENT_PUBLISH) == SWITCH_STATUS_SUCCESS) {
switch_event_add_header(s_event, SWITCH_STACK_BOTTOM, "service", "_sip._udp");
switch_event_add_header(s_event, SWITCH_STACK_BOTTOM, "port", "%d", profile->sip_port);
switch_event_fire(&s_event);
}
if (switch_event_create(&s_event, SWITCH_EVENT_PUBLISH) == SWITCH_STATUS_SUCCESS) {
switch_event_add_header(s_event, SWITCH_STACK_BOTTOM, "service", "_sip._tcp");
switch_event_add_header(s_event, SWITCH_STACK_BOTTOM, "port", "%d", profile->sip_port);
switch_event_fire(&s_event);
}
if (switch_event_create(&s_event, SWITCH_EVENT_PUBLISH) == SWITCH_STATUS_SUCCESS) {
switch_event_add_header(s_event, SWITCH_STACK_BOTTOM, "service", "_sip._sctp");
switch_event_add_header(s_event, SWITCH_STACK_BOTTOM, "port", "%d", profile->sip_port);
switch_event_fire(&s_event);
}
switch_mutex_lock(globals.hash_mutex);
switch_core_hash_insert(globals.profile_hash, profile->name, profile);
switch_mutex_unlock(globals.hash_mutex);
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
if (profile->pflags & PFLAG_PRESENCE) {
establish_presence(profile);
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
}
while(globals.running == 1) {
if (++ireg_loops >= IREG_SECONDS) {
check_expire(profile, time(NULL));
ireg_loops = 0;
}
if (++oreg_loops >= OREG_SECONDS) {
check_oreg(profile, time(NULL));
oreg_loops = 0;
}
su_root_step(profile->s_root, 1000);
}
unreg(profile);
su_home_deinit(profile->home);
if (switch_event_create(&s_event, SWITCH_EVENT_UNPUBLISH) == SWITCH_STATUS_SUCCESS) {
switch_event_add_header(s_event, SWITCH_STACK_BOTTOM, "service", "_sip._udp");
switch_event_add_header(s_event, SWITCH_STACK_BOTTOM, "port", "%d", profile->sip_port);
switch_event_fire(&s_event);
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
su_root_destroy(profile->s_root);
pool = profile->pool;
switch_core_destroy_memory_pool(&pool);
switch_mutex_lock(globals.mutex);
globals.running = 0;
switch_mutex_unlock(globals.mutex);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
return NULL;
}
static void launch_profile_thread(sofia_profile_t *profile)
{
switch_thread_t *thread;
switch_threadattr_t *thd_attr = NULL;
switch_threadattr_create(&thd_attr, profile->pool);
switch_threadattr_detach_set(thd_attr, 1);
switch_threadattr_stacksize_set(thd_attr, SWITCH_THREAD_STACKSIZE);
switch_thread_create(&thread, thd_attr, profile_thread_run, profile, profile->pool);
}
static switch_status_t config_sofia(int reload)
{
char *cf = "sofia.conf";
switch_xml_t cfg, xml = NULL, xprofile, param, settings, profiles, registration, registrations;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
switch_status_t status = SWITCH_STATUS_SUCCESS;
sofia_profile_t *profile = NULL;
char url[512] = "";
switch_mutex_lock(globals.mutex);
globals.running = 1;
switch_mutex_unlock(globals.mutex);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (!(xml = switch_xml_open_cfg(cf, &cfg, NULL))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "open of %s failed\n", cf);
status = SWITCH_STATUS_FALSE;
goto done;
}
if ((settings = switch_xml_child(cfg, "global_settings"))) {
for (param = switch_xml_child(settings, "param"); param; param = param->next) {
char *var = (char *) switch_xml_attr_soft(param, "name");
char *val = (char *) switch_xml_attr_soft(param, "value");
if (!strcasecmp(var, "log-level")) {
su_log_set_level(NULL, atoi(val));
} else if (!strcasecmp(var, "log-level-trace")) {
su_log_set_level(tport_log, atoi(val));
}
}
}
if ((profiles = switch_xml_child(cfg, "profiles"))) {
for (xprofile = switch_xml_child(profiles, "profile"); xprofile; xprofile = xprofile->next) {
if (!(settings = switch_xml_child(xprofile, "settings"))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "No Settings, check the new config!\n", cf);
} else {
char *xprofilename = (char *) switch_xml_attr_soft(xprofile, "name");
switch_memory_pool_t *pool = NULL;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
/* Setup the pool */
if ((status = switch_core_new_memory_pool(&pool)) != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Memory Error!\n");
goto done;
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (!(profile = (sofia_profile_t *) switch_core_alloc(pool, sizeof(*profile)))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Memory Error!\n");
goto done;
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (!xprofilename) {
xprofilename = "unnamed";
}
profile->pool = pool;
profile->name = switch_core_strdup(profile->pool, xprofilename);
snprintf(url, sizeof(url), "sofia_reg_%s", xprofilename);
profile->dbname = switch_core_strdup(profile->pool, url);
switch_core_hash_init(&profile->chat_hash, profile->pool);
profile->dtmf_duration = 100;
profile->codec_ms = 20;
for (param = switch_xml_child(settings, "param"); param; param = param->next) {
char *var = (char *) switch_xml_attr_soft(param, "name");
char *val = (char *) switch_xml_attr_soft(param, "value");
if (!strcasecmp(var, "debug")) {
profile->debug = atoi(val);
} else if (!strcasecmp(var, "use-rtp-timer") && switch_true(val)) {
switch_set_flag(profile, TFLAG_TIMER);
} else if (!strcasecmp(var, "rfc2833-pt")) {
profile->te = (switch_payload_t) atoi(val);
} else if (!strcasecmp(var, "sip-port")) {
profile->sip_port = atoi(val);
} else if (!strcasecmp(var, "vad")) {
if (!strcasecmp(val, "in")) {
switch_set_flag(profile, TFLAG_VAD_IN);
} else if (!strcasecmp(val, "out")) {
switch_set_flag(profile, TFLAG_VAD_OUT);
} else if (!strcasecmp(val, "both")) {
switch_set_flag(profile, TFLAG_VAD_IN);
switch_set_flag(profile, TFLAG_VAD_OUT);
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Invald option %s for VAD\n", val);
}
} else if (!strcasecmp(var, "ext-rtp-ip")) {
profile->extrtpip = switch_core_strdup(profile->pool, val);
} else if (!strcasecmp(var, "rtp-ip")) {
profile->rtpip = switch_core_strdup(profile->pool, val);
} else if (!strcasecmp(var, "sip-ip")) {
profile->sipip = switch_core_strdup(profile->pool, val);
} else if (!strcasecmp(var, "sip-domain")) {
profile->sipdomain = switch_core_strdup(profile->pool, val);
} else if (!strcasecmp(var, "rtp-timer-name")) {
profile->timer_name = switch_core_strdup(profile->pool, val);
} else if (!strcasecmp(var, "manage-presence")) {
if (switch_true(val)) {
profile->pflags |= PFLAG_PRESENCE;
}
} else if (!strcasecmp(var, "auth-calls")) {
if (switch_true(val)) {
profile->pflags |= PFLAG_AUTH_CALLS;
}
} else if (!strcasecmp(var, "accept-blind-reg")) {
if (switch_true(val)) {
profile->pflags |= PFLAG_BLIND_REG;
}
} else if (!strcasecmp(var, "auth-all-packets")) {
if (switch_true(val)) {
profile->pflags |= PFLAG_AUTH_ALL;
}
} else if (!strcasecmp(var, "full-id-in-dialplan")) {
if (switch_true(val)) {
profile->pflags |= PFLAG_FULL_ID;
}
} else if (!strcasecmp(var, "ext-sip-ip")) {
profile->extsipip = switch_core_strdup(profile->pool, val);
} else if (!strcasecmp(var, "bitpacking")) {
if (!strcasecmp(val, "aal2")) {
profile->codec_flags = SWITCH_CODEC_FLAG_AAL2;
}
} else if (!strcasecmp(var, "username")) {
profile->username = switch_core_strdup(profile->pool, val);
} else if (!strcasecmp(var, "context")) {
profile->context = switch_core_strdup(profile->pool, val);
} else if (!strcasecmp(var, "alias")) {
sip_alias_node_t *node;
if ((node = switch_core_alloc(profile->pool, sizeof(*node)))) {
if ((node->url = switch_core_strdup(profile->pool, val))) {
node->next = profile->aliases;
profile->aliases = node;
}
}
} else if (!strcasecmp(var, "dialplan")) {
profile->dialplan = switch_core_strdup(profile->pool, val);
} else if (!strcasecmp(var, "max-calls")) {
profile->max_calls = atoi(val);
} else if (!strcasecmp(var, "codec-prefs")) {
profile->codec_string = switch_core_strdup(profile->pool, val);
profile->codec_order_last = switch_separate_string(profile->codec_string, ',', profile->codec_order, SWITCH_MAX_CODECS);
} else if (!strcasecmp(var, "codec-ms")) {
profile->codec_ms = atoi(val);
} else if (!strcasecmp(var, "dtmf-duration")) {
int dur = atoi(val);
if (dur > 10 && dur < 8000) {
profile->dtmf_duration = dur;
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Duration out of bounds!\n");
}
}
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (switch_test_flag(profile, TFLAG_TIMER) && !profile->timer_name) {
profile->timer_name = switch_core_strdup(profile->pool, "soft");
}
if (!profile->username) {
profile->username = switch_core_strdup(profile->pool, "FreeSWITCH");
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (!profile->rtpip) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "Setting ip to '127.0.0.1'\n");
profile->rtpip = switch_core_strdup(profile->pool, "127.0.0.1");
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
}
if (!profile->sip_port) {
profile->sip_port = 5060;
}
if (!profile->dialplan) {
profile->dialplan = switch_core_strdup(profile->pool, "XML");
}
if (!profile->sipdomain) {
profile->sipdomain = switch_core_strdup(profile->pool, profile->sipip);
}
snprintf(url, sizeof(url), "sip:mod_sofia@%s:%d", profile->sipip, profile->sip_port);
profile->url = switch_core_strdup(profile->pool, url);
}
if (profile) {
if ((registrations = switch_xml_child(xprofile, "registrations"))) {
for (registration = switch_xml_child(registrations, "registration"); registration; registration = registration->next) {
char *name = (char *) switch_xml_attr_soft(registration, "name");
outbound_reg_t *oreg;
if (switch_strlen_zero(name)) {
name = "anonymous";
}
if ((oreg = switch_core_alloc(profile->pool, sizeof(*oreg)))) {
oreg->pool = profile->pool;
oreg->profile = profile;
oreg->name = switch_core_strdup(oreg->pool, name);
oreg->freq = 0;
for (param = switch_xml_child(registration, "param"); param; param = param->next) {
char *var = (char *) switch_xml_attr_soft(param, "name");
char *val = (char *) switch_xml_attr_soft(param, "value");
if (!strcmp(var, "register-scheme")) {
oreg->register_scheme = switch_core_strdup(oreg->pool, val);
} else if (!strcmp(var, "register-realm")) {
oreg->register_realm = switch_core_strdup(oreg->pool, val);
} else if (!strcmp(var, "register-username")) {
oreg->register_username = switch_core_strdup(oreg->pool, val);
} else if (!strcmp(var, "register-password")) {
oreg->register_password = switch_core_strdup(oreg->pool, val);
} else if (!strcmp(var, "register-from")) {
oreg->register_from = switch_core_strdup(oreg->pool, val);
} else if (!strcmp(var, "register-to")) {
oreg->register_to = switch_core_strdup(oreg->pool, val);
} else if (!strcmp(var, "register-proxy")) {
oreg->register_proxy = switch_core_strdup(oreg->pool, val);
} else if (!strcmp(var, "register-frequency")) {
oreg->expires_str = switch_core_strdup(oreg->pool, val);
}
}
if (switch_strlen_zero(oreg->register_scheme)) {
oreg->register_scheme = switch_core_strdup(oreg->pool, "Digest");
}
if (switch_strlen_zero(oreg->register_realm)) {
oreg->register_realm = switch_core_strdup(oreg->pool, "freeswitch.org");
}
if (switch_strlen_zero(oreg->register_username)) {
oreg->register_username = switch_core_strdup(oreg->pool, "freeswitch");
}
if (switch_strlen_zero(oreg->register_password)) {
oreg->register_password = switch_core_strdup(oreg->pool, "works");
}
if (switch_strlen_zero(oreg->register_from)) {
oreg->register_from = switch_core_strdup(oreg->pool, "FreeSWITCH <sip:freeswitch@freeswitch.org>");
}
if (switch_strlen_zero(oreg->register_to)) {
oreg->register_to = switch_core_strdup(oreg->pool, "sip:freeswitch@freeswitch.org");
}
if (switch_strlen_zero(oreg->register_proxy)) {
oreg->register_proxy = switch_core_strdup(oreg->pool, "sip:freeswitch@freeswitch.org");
}
if (switch_strlen_zero(oreg->expires_str)) {
oreg->expires_str = switch_core_strdup(oreg->pool, "300");
}
oreg->freq = atoi(oreg->expires_str);
oreg->next = profile->registrations;
profile->registrations = oreg;
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
}
}
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "Started Profile %s [%s]\n", profile->name, url);
launch_profile_thread(profile);
profile = NULL;
}
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
}
done:
if (xml) {
switch_xml_free(xml);
}
return status;
}
static void event_handler(switch_event_t *event)
{
char *subclass, *sql;
if ((subclass = switch_event_get_header(event, "orig-event-subclass")) && !strcasecmp(subclass, MY_EVENT_REGISTER)) {
char *from_user = switch_event_get_header(event, "orig-from-user");
char *from_host = switch_event_get_header(event, "orig-from-host");
char *contact_str = switch_event_get_header(event, "orig-contact");
char *exp_str = switch_event_get_header(event, "orig-expires");
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
char *rpid = switch_event_get_header(event, "orig-rpid");
long expires = (long)time(NULL) + atol(exp_str);
char *profile_name = switch_event_get_header(event, "orig-profile-name");
sofia_profile_t *profile;
char buf[512];
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
if (!rpid) {
rpid = "unknown";
}
if (!profile_name || !(profile = (sofia_profile_t *) switch_core_hash_find(globals.profile_hash, profile_name))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Invalid Profile\n");
return;
}
if (!find_reg_url(profile, from_user, from_host, buf, sizeof(buf))) {
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
sql = switch_mprintf("insert into sip_registrations values ('%q','%q','%q','Regestered', '%q', %ld)",
from_user,
from_host,
contact_str,
rpid,
expires);
} else {
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
sql = switch_mprintf("update sip_registrations set contact='%q', rpid='%q', expires=%ld where user='%q' and host='%q'",
contact_str,
rpid,
expires,
from_user,
from_host);
}
if (sql) {
execute_sql(profile->dbname, sql, profile->ireg_mutex);
switch_safe_free(sql);
sql = NULL;
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Propagating registration for %s@%s->%s\n",
from_user, from_host, contact_str);
}
}
}
static switch_status_t chat_send(char *proto, char *from, char *to, char *subject, char *body, char *hint)
{
char buf[256];
char *user, *host;
sofia_profile_t *profile;
char *ffrom = NULL;
nua_handle_t *msg_nh;
char *contact;
if (to && (user = strdup(to))) {
if ((host = strchr(user, '@'))) {
*host++ = '\0';
}
if (!host || !(profile = (sofia_profile_t *) switch_core_hash_find(globals.profile_hash, host))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Invalid Profile %s\n", host ? host : "NULL");
return SWITCH_STATUS_FALSE;
}
if (!find_reg_url(profile, user, host, buf, sizeof(buf))) {
return SWITCH_STATUS_FALSE;
}
if (!strcmp(proto, SOFIA_CHAT_PROTO)) {
from = hint;
} else {
char *fp, *p, *fu = NULL;
if (!(fp = strdup(from))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Memory Error!\n");
return SWITCH_STATUS_FALSE;
}
if ((p = strchr(fp, '@'))) {
*p = '\0';
fu = strdup(fp);
*p = '+';
}
ffrom = switch_mprintf("\"%s\" <sip:%s+%s@%s>", fu, proto, fp, profile->name);
from = ffrom;
switch_safe_free(fu);
switch_safe_free(fp);
}
contact = get_url_from_contact(buf, 1);
msg_nh = nua_handle(profile->nua, NULL,
SIPTAG_FROM_STR(from),
NUTAG_URL(contact),
SIPTAG_TO_STR(buf), // if this cries, add contact here too, change the 1 to 0 and omit the safe_free
SIPTAG_CONTACT_STR(profile->url),
TAG_END());
switch_safe_free(contact);
nua_message(msg_nh,
SIPTAG_CONTENT_TYPE_STR("text/html"),
SIPTAG_PAYLOAD_STR(body),
TAG_END());
switch_safe_free(ffrom);
free(user);
}
return SWITCH_STATUS_SUCCESS;
}
static void cancel_presence(void)
{
char *sql, *errmsg = NULL;
switch_core_db_t *db;
sofia_profile_t *profile;
switch_hash_index_t *hi;
void *val;
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
if ((sql = switch_mprintf("select 0,'unavailable','unavailable',* from sip_subscriptions where event='presence'"))) {
for (hi = switch_hash_first(apr_hash_pool_get(globals.profile_hash), globals.profile_hash); hi; hi = switch_hash_next(hi)) {
switch_hash_this(hi, NULL, NULL, &val);
profile = (sofia_profile_t *) val;
if (!(profile->pflags & PFLAG_PRESENCE)) {
continue;
}
if (!(db = switch_core_db_open_file(profile->dbname))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error Opening DB %s\n", profile->dbname);
continue;
}
switch_mutex_lock(profile->ireg_mutex);
switch_core_db_exec(db, sql, sub_callback, profile, &errmsg);
switch_mutex_unlock(profile->ireg_mutex);
switch_core_db_close(db);
}
switch_safe_free(sql);
}
}
static void establish_presence(sofia_profile_t *profile)
{
char *sql, *errmsg = NULL;
switch_core_db_t *db;
if (!(db = switch_core_db_open_file(profile->dbname))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error Opening DB %s\n", profile->dbname);
return;
}
if ((sql = switch_mprintf("select user,host,'Registered','unknown','' from sip_registrations"))) {
switch_mutex_lock(profile->ireg_mutex);
switch_core_db_exec(db, sql, resub_callback, profile, &errmsg);
switch_mutex_unlock(profile->ireg_mutex);
switch_safe_free(sql);
}
if ((sql = switch_mprintf("select sub_to_user,sub_to_host,'Online','unknown',proto from sip_subscriptions "
"where proto='ext' or proto='user' or proto='conf'"))) {
switch_mutex_lock(profile->ireg_mutex);
switch_core_db_exec(db, sql, resub_callback, profile, &errmsg);
switch_mutex_unlock(profile->ireg_mutex);
switch_safe_free(sql);
}
switch_core_db_close(db);
}
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
static char *translate_rpid(char *in, char *ext)
{
char *r = NULL;
if (in && (strstr(in, "null") || strstr(in, "NULL"))) {
in = NULL;
}
if (!in) {
in = ext;
}
if (!in) {
return NULL;
}
if (!strcasecmp(in, "dnd")) {
r = "busy";
}
if (ext && !strcasecmp(ext, "away")) {
r = "idle";
}
return r;
}
static void pres_event_handler(switch_event_t *event)
{
sofia_profile_t *profile;
switch_hash_index_t *hi;
void *val;
char *from = switch_event_get_header(event, "from");
char *proto = switch_event_get_header(event, "proto");
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
char *rpid = switch_event_get_header(event, "rpid");
char *status= switch_event_get_header(event, "status");
char *event_type = switch_event_get_header(event, "event_type");
//char *event_subtype = switch_event_get_header(event, "event_subtype");
char *sql = NULL;
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
char *euser = NULL, *user = NULL, *host = NULL;
char *errmsg;
switch_core_db_t *db;
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
if (rpid && !strcasecmp(rpid, "n/a")) {
rpid = NULL;
}
if (status && !strcasecmp(status, "n/a")) {
status = NULL;
}
if (rpid) {
rpid = translate_rpid(rpid, status);
}
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
if (!status) {
status = "Available";
if (rpid) {
if (!strcasecmp(rpid, "busy")) {
status = "Busy";
} else if (!strcasecmp(rpid, "unavailable")) {
status = "Idle";
} else if (!strcasecmp(rpid, "away")) {
status = "Idle";
}
}
}
if (!rpid) {
rpid = "unknown";
}
if (event->event_id == SWITCH_EVENT_ROSTER) {
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
if (from) {
sql = switch_mprintf("select 1,'%q','%q',* from sip_subscriptions where event='presence' and full_from like '%%%q%%'", status, rpid, from);
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
} else {
sql = switch_mprintf("select 1,'%q','%q',* from sip_subscriptions where event='presence'", status, rpid);
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
}
for (hi = switch_hash_first(apr_hash_pool_get(globals.profile_hash), globals.profile_hash); hi; hi = switch_hash_next(hi)) {
switch_hash_this(hi, NULL, NULL, &val);
profile = (sofia_profile_t *) val;
if (!(profile->pflags & PFLAG_PRESENCE)) {
continue;
}
if (sql) {
if (!(db = switch_core_db_open_file(profile->dbname))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error Opening DB %s\n", profile->dbname);
continue;
}
switch_mutex_lock(profile->ireg_mutex);
switch_core_db_exec(db, sql, sub_callback, profile, &errmsg);
switch_mutex_unlock(profile->ireg_mutex);
switch_core_db_close(db);
}
}
return;
}
if (switch_strlen_zero(event_type)) {
event_type="presence";
}
if ((user = strdup(from))) {
if ((host = strchr(user, '@'))) {
char *p;
*host++ = '\0';
if ((p = strchr(host, '/'))) {
*p = '\0';
}
} else {
switch_safe_free(user);
return;
}
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
if ((euser = strchr(user, '+'))) {
euser++;
} else {
euser = user;
}
} else {
return;
}
switch(event->event_id) {
case SWITCH_EVENT_PRESENCE_IN:
sql = switch_mprintf("select 1,'%q','%q',* from sip_subscriptions where proto='%q' and event='%q' and sub_to_user='%q' and sub_to_host='%q'",
status , rpid, proto, event_type, euser, host);
break;
case SWITCH_EVENT_PRESENCE_OUT:
sql = switch_mprintf("select 0,'%q','%q',* from sip_subscriptions where proto='%q' and event='%q' and sub_to_user='%q' and sub_to_host='%q'",
status, rpid, proto, event_type, euser, host);
break;
default:
break;
}
for (hi = switch_hash_first(apr_hash_pool_get(globals.profile_hash), globals.profile_hash); hi; hi = switch_hash_next(hi)) {
switch_hash_this(hi, NULL, NULL, &val);
profile = (sofia_profile_t *) val;
if (!(profile->pflags & PFLAG_PRESENCE)) {
continue;
}
if (sql) {
if (!(db = switch_core_db_open_file(profile->dbname))) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error Opening DB %s\n", profile->dbname);
continue;
}
switch_mutex_lock(profile->ireg_mutex);
switch_core_db_exec(db, sql, sub_callback, profile, &errmsg);
switch_mutex_unlock(profile->ireg_mutex);
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
switch_core_db_close(db);
}
}
Presence and Chat Gateway Code This is some brand new stuff to gateway chat/presence/audio from one protocol to another So far it only works between google/jingle and SIP All I had to test the SIP end was X-Lite and Eyebeam and GoogleTalk on the jingle end. With this setup registered X-Lite's can chat with each other and call each other as well as X-Lite to GoogleTalk and GoogleTalk to X-Lite audio calls. Chat May also be done between X-Lite and jabber You'll also need a jabber server configured for component login so you can interface. We have only tested with jabberd2 so far. Configure DNS so srv records for jabber for your subdomain (fs.mydomain.com in the example) so the jabber records are pointed at your jabber server. RELEVANT CONFIGS <!-- Brian has no jingle support so send calls to him over to his iax url --> <extension name="bkw"> <condition field="destination_number" expression="^jingle\+brian@agents.cylynx.com$"> <action application="bridge" data="iax/guest@brianwest.homeunix.org/9184290404"/> </condition> </extension> <!-- Assumption is made here that both sip and jingle have the same profile/domain name as documented below --> <extension name="jingle2sip"> <condition field="source" expression="mod_dingaling"/> <condition field="destination_number" expression="^sip\+([^\@]+)\@(.*)$"> <action application="bridge" data="sofia/$2/$1%$2"/> </condition> </extension> <extension name="sip2jingle"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^jingle\+([^\@]+)\@(.*)$"> <action application="bridge" data="dingaling/sip+${sip_fromuser}@${sip_fromhost}/$1@$2"/> </condition> </extension> <configuration name="sofia.conf" description="sofia Endpoint"> <global_settings> <param name="log-level" value="0"/> </global_settings> <profiles> <profile name="fs.mydomain.com"> <registrations/> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="codec-ms" value="20"/> <param name="accept-blind-reg" value="true"/> <param name="manage-presence" value="true"/> <!--<param name="full-id-in-dialplan" value="true"/>--> <!--<param name="auth-calls" value="true"/>--> <!--<param name="auth-all-packets" value="true"/>--> <param name="use-rtp-timer" value="true"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="100.200.100.200"/> <param name="sip-ip" value="fs.mydomain.com"/> </settings> </profile> </profiles> </configuration> <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="0"/> <param name="codec-prefs" value="PCMU"/> </settings> <profile type="component"> <param name="name" value="fs.mydomain.com"/> <param name="password" value="secret"/> <param name="dialplan" value="XML"/> <param name="rtp-ip" value="208.64.200.42"/> <param name="server" value="jabber.freeswitch.org:5347"/> <!-- disable to trade async for more calls --> <param name="use-rtp-timer" value="true"/> <param name="exten" value="_auto_"/> <!--<param name="vad" value="both"/>--> </profile> </configuration> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3115 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-20 06:17:00 +00:00
switch_safe_free(sql);
switch_safe_free(user);
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
SWITCH_MOD_DECLARE(switch_status_t) switch_module_load(const switch_loadable_module_interface_t **module_interface, char *filename)
{
Media Management (Sponsored By Front Logic) This modification makes it possible to change the media path of session in the switch on-the-fly and from the dialplan. It adds some API interface calls usable from a remote client such as mod_event_socket or the test console. 1) media [off] <uuid> Turns on/off the media on the call described by <uuid> The media will be redirected as desiered either into the switch or point to point. 2) hold [off] <uuid> Turns on/off endpoint specific hold state on the session described by <uuid> 3) broadcast <uuid> "<path>[ <timer_name>]" or "speak:<tts_engine>|<tts_voice>|<text>[|<timer_name>]" [both] A message will be sent to the call described by uuid instructing it to play the file or speak the text indicated. If the 'both' option is specified both ends of the call will hear the message otherwise just the uuid specified will hear the message. During playback when only one side is hearing the message the other end will hear silence. If media is not flowing across the switch when the message is broadcasted, the media will be directed to the switch for the duration of the call and then returned to it's previous state. Also the no_media=true option in the dialplan before a bridge makes it possible to place a call while proxying the session description from one endpoint to the other and establishing an immidiate point-to-point media connection with no media on the switch. <action application="set" data="no_media=true"/> <action application="bridge" data="sofia/mydomain.com/myid@myhost.com"/> *NOTE* when connecting two outbound legs by using the "originate" api command with an extension that has no_media=true enabled, the media for the first leg will be engaged with the switch until the second leg has answered and the other session description is available to establish a point to point connection at which time point-to-point mode will be enabled. *NOTE* it is reccommended you rebuild FreeSWITCH with "make sure" as there have been some changes to the core. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3245 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-10-31 21:38:06 +00:00
silence_frame.data = silence_data;
silence_frame.datalen = sizeof(silence_data);
silence_frame.buflen = sizeof(silence_data);
silence_frame.flags = SFF_CNG;
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (switch_core_new_memory_pool(&module_pool) != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "OH OH no pool\n");
return SWITCH_STATUS_TERM;
}
memset(&globals, 0, sizeof(globals));
switch_mutex_init(&globals.mutex, SWITCH_MUTEX_NESTED, module_pool);
if (switch_event_bind((char *) modname, SWITCH_EVENT_CUSTOM, MULTICAST_EVENT, event_handler, NULL) != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Couldn't bind!\n");
return SWITCH_STATUS_TERM;
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
su_init();
su_log_redirect(NULL, logger, NULL);
su_log_redirect(tport_log, logger, NULL);
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
switch_core_hash_init(&globals.profile_hash, module_pool);
switch_mutex_init(&globals.hash_mutex, SWITCH_MUTEX_NESTED, module_pool);
config_sofia(0);
if (switch_event_bind((char *) modname, SWITCH_EVENT_PRESENCE_IN, SWITCH_EVENT_SUBCLASS_ANY, pres_event_handler, NULL) != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Couldn't bind!\n");
return SWITCH_STATUS_GENERR;
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
if (switch_event_bind((char *) modname, SWITCH_EVENT_PRESENCE_OUT, SWITCH_EVENT_SUBCLASS_ANY, pres_event_handler, NULL) != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Couldn't bind!\n");
return SWITCH_STATUS_GENERR;
}
if (switch_event_bind((char *) modname, SWITCH_EVENT_ROSTER, SWITCH_EVENT_SUBCLASS_ANY, pres_event_handler, NULL) != SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Couldn't bind!\n");
return SWITCH_STATUS_GENERR;
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
/* connect my internal structure to the blank pointer passed to me */
*module_interface = &sofia_module_interface;
/* indicate that the module should continue to be loaded */
return SWITCH_STATUS_SUCCESS;
}
SWITCH_MOD_DECLARE(switch_status_t) switch_module_shutdown(void)
{
cancel_presence();
switch_mutex_lock(globals.mutex);
if (globals.running == 1) {
globals.running = -1;
}
switch_mutex_unlock(globals.mutex);
while(globals.running) {
switch_yield(1000);
}
Adding mod_sofia to the tree so we can work on it easier.... I am not adding it to the examples or to the modules.conf because it's not really ready for that yet. This is only 1.5 days old from scratch at this point but the brave hearted who want to play with it can do the following: Add this to modules.conf: ----------------------------------------------------------------------------- endpoints/mod_sofia ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration/modules.conf area ----------------------------------------------------------------------------- <load module="mod_sofia"/> ----------------------------------------------------------------------------- Add this to freeswitch.xml in the configuration section ----------------------------------------------------------------------------- <configuration name="sofia.conf" description="sofia Endpoint"> <!-- You may have multiple profiles --> <profile name="test"> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="PCMU"/> <param name="use-rtp-timer" value="true"/> <param name="rtp-ip" value="127.0.0.1"/> <param name="sip-ip" value="127.0.0.1"/> <!-- optional ; --> <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>--> <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> --> <!-- VAD choose one (out is a good choice); --> <!-- <param name="vad" value="in"/> --> <!-- <param name="vad" value="out"/> --> <!-- <param name="vad" value="both"/> --> <!--<param name="alias" value="sip:208.64.200.40:5555"/>--> </profile> </configuration> ----------------------------------------------------------------------------- The call string to use profile test would be: sofia/test/1000@1.2.3.4 as in: <action application="bridge" data="sofia/test/1000@1.2.3.4"/> git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2398 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-25 23:55:59 +00:00
su_deinit();
return SWITCH_STATUS_SUCCESS;
}