freeswitch/src/switch_channel.c

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/*
* 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>
* Michael Jerris <mike@jerris.com>
*
*
* switch_channel.c -- Media Channel Interface
*
*/
#include <switch.h>
#include <switch_channel.h>
struct switch_cause_table {
const char *name;
switch_call_cause_t cause;
};
static struct switch_cause_table CAUSE_CHART[] = {
{ "UNALLOCATED", SWITCH_CAUSE_UNALLOCATED },
{ "SUCCESS", SWITCH_CAUSE_SUCCESS },
{ "NO_ROUTE_TRANSIT_NET", SWITCH_CAUSE_NO_ROUTE_TRANSIT_NET },
{ "NO_ROUTE_DESTINATION", SWITCH_CAUSE_NO_ROUTE_DESTINATION },
{ "CHANNEL_UNACCEPTABLE", SWITCH_CAUSE_CHANNEL_UNACCEPTABLE },
{ "CALL_AWARDED_DELIVERED", SWITCH_CAUSE_CALL_AWARDED_DELIVERED },
{ "NORMAL_CLEARING", SWITCH_CAUSE_NORMAL_CLEARING },
{ "USER_BUSY", SWITCH_CAUSE_USER_BUSY },
{ "NO_USER_RESPONSE", SWITCH_CAUSE_NO_USER_RESPONSE },
{ "NO_ANSWER", SWITCH_CAUSE_NO_ANSWER },
{ "SUBSCRIBER_ABSENT", SWITCH_CAUSE_SUBSCRIBER_ABSENT },
{ "CALL_REJECTED", SWITCH_CAUSE_CALL_REJECTED },
{ "NUMBER_CHANGED", SWITCH_CAUSE_NUMBER_CHANGED },
{ "REDIRECTION_TO_NEW_DESTINATION", SWITCH_CAUSE_REDIRECTION_TO_NEW_DESTINATION },
{ "EXCHANGE_ROUTING_ERROR", SWITCH_CAUSE_EXCHANGE_ROUTING_ERROR },
{ "DESTINATION_OUT_OF_ORDER", SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER },
{ "INVALID_NUMBER_FORMAT", SWITCH_CAUSE_INVALID_NUMBER_FORMAT },
{ "FACILITY_REJECTED", SWITCH_CAUSE_FACILITY_REJECTED },
{ "RESPONSE_TO_STATUS_ENQUIRY", SWITCH_CAUSE_RESPONSE_TO_STATUS_ENQUIRY },
{ "NORMAL_UNSPECIFIED", SWITCH_CAUSE_NORMAL_UNSPECIFIED },
{ "NORMAL_CIRCUIT_CONGESTION", SWITCH_CAUSE_NORMAL_CIRCUIT_CONGESTION },
{ "NETWORK_OUT_OF_ORDER", SWITCH_CAUSE_NETWORK_OUT_OF_ORDER },
{ "NORMAL_TEMPORARY_FAILURE", SWITCH_CAUSE_NORMAL_TEMPORARY_FAILURE },
{ "SWITCH_CONGESTION", SWITCH_CAUSE_SWITCH_CONGESTION },
{ "ACCESS_INFO_DISCARDED", SWITCH_CAUSE_ACCESS_INFO_DISCARDED },
{ "REQUESTED_CHAN_UNAVAIL", SWITCH_CAUSE_REQUESTED_CHAN_UNAVAIL },
{ "PRE_EMPTED", SWITCH_CAUSE_PRE_EMPTED },
{ "FACILITY_NOT_SUBSCRIBED", SWITCH_CAUSE_FACILITY_NOT_SUBSCRIBED },
{ "OUTGOING_CALL_BARRED", SWITCH_CAUSE_OUTGOING_CALL_BARRED },
{ "INCOMING_CALL_BARRED", SWITCH_CAUSE_INCOMING_CALL_BARRED },
{ "BEARERCAPABILITY_NOTAUTH", SWITCH_CAUSE_BEARERCAPABILITY_NOTAUTH },
{ "BEARERCAPABILITY_NOTAVAIL", SWITCH_CAUSE_BEARERCAPABILITY_NOTAVAIL },
{ "SERVICE_UNAVAILABLE", SWITCH_CAUSE_SERVICE_UNAVAILABLE },
{ "CHAN_NOT_IMPLEMENTED", SWITCH_CAUSE_CHAN_NOT_IMPLEMENTED },
{ "FACILITY_NOT_IMPLEMENTED", SWITCH_CAUSE_FACILITY_NOT_IMPLEMENTED },
{ "SERVICE_NOT_IMPLEMENTED", SWITCH_CAUSE_SERVICE_NOT_IMPLEMENTED },
{ "INVALID_CALL_REFERENCE", SWITCH_CAUSE_INVALID_CALL_REFERENCE },
{ "INCOMPATIBLE_DESTINATION", SWITCH_CAUSE_INCOMPATIBLE_DESTINATION },
{ "INVALID_MSG_UNSPECIFIED", SWITCH_CAUSE_INVALID_MSG_UNSPECIFIED },
{ "MANDATORY_IE_MISSING", SWITCH_CAUSE_MANDATORY_IE_MISSING },
{ "MESSAGE_TYPE_NONEXIST", SWITCH_CAUSE_MESSAGE_TYPE_NONEXIST },
{ "WRONG_MESSAGE", SWITCH_CAUSE_WRONG_MESSAGE },
{ "IE_NONEXIST", SWITCH_CAUSE_IE_NONEXIST },
{ "INVALID_IE_CONTENTS", SWITCH_CAUSE_INVALID_IE_CONTENTS },
{ "WRONG_CALL_STATE", SWITCH_CAUSE_WRONG_CALL_STATE },
{ "RECOVERY_ON_TIMER_EXPIRE", SWITCH_CAUSE_RECOVERY_ON_TIMER_EXPIRE },
{ "MANDATORY_IE_LENGTH_ERROR", SWITCH_CAUSE_MANDATORY_IE_LENGTH_ERROR },
{ "PROTOCOL_ERROR", SWITCH_CAUSE_PROTOCOL_ERROR },
{ "INTERWORKING", SWITCH_CAUSE_INTERWORKING },
{ "ORIGINATOR_CANCEL", SWITCH_CAUSE_ORIGINATOR_CANCEL },
{ "CRASH", SWITCH_CAUSE_CRASH },
{ "SYSTEM_SHUTDOWN", SWITCH_CAUSE_SYSTEM_SHUTDOWN },
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
{ "LOSE_RACE", SWITCH_CAUSE_LOSE_RACE },
{ "MANAGER_REQUEST", SWITCH_CAUSE_MANAGER_REQUEST },
{ "BLIND_TRANSFER", SWITCH_CAUSE_BLIND_TRANSFER },
{ "ATTENDED_TRANSFER", SWITCH_CAUSE_ATTENDED_TRANSFER },
{ NULL, 0 }
};
struct switch_channel {
char *name;
switch_buffer_t *dtmf_buffer;
switch_mutex_t *dtmf_mutex;
switch_mutex_t *flag_mutex;
switch_mutex_t *profile_mutex;
switch_core_session_t *session;
switch_channel_state_t state;
uint32_t flags;
uint32_t state_flags;
switch_caller_profile_t *caller_profile;
const switch_state_handler_table_t *state_handlers[SWITCH_MAX_STATE_HANDLERS];
int state_handler_index;
switch_hash_t *variables;
switch_hash_t *private_hash;
switch_call_cause_t hangup_cause;
};
SWITCH_DECLARE(char *) switch_channel_cause2str(switch_call_cause_t cause)
{
uint8_t x;
char *str = "UNKNOWN";
for(x = 0; CAUSE_CHART[x].name; x++) {
if (CAUSE_CHART[x].cause == cause) {
str = (char *) CAUSE_CHART[x].name;
}
}
return str;
}
SWITCH_DECLARE(switch_call_cause_t) switch_channel_str2cause(char *str)
{
uint8_t x;
switch_call_cause_t cause = SWITCH_CAUSE_UNALLOCATED;
for(x = 0; CAUSE_CHART[x].name; x++) {
if (!strcasecmp(CAUSE_CHART[x].name, str)) {
cause = CAUSE_CHART[x].cause;
}
}
return cause;
}
SWITCH_DECLARE(switch_call_cause_t) switch_channel_get_cause(switch_channel_t *channel)
{
assert(channel != NULL);
return channel->hangup_cause;
}
SWITCH_DECLARE(switch_channel_timetable_t *) switch_channel_get_timetable(switch_channel_t *channel)
{
switch_channel_timetable_t *times = NULL;
assert(channel != NULL);
if (channel->caller_profile) {
switch_mutex_lock(channel->profile_mutex);
times = channel->caller_profile->times;
switch_mutex_unlock(channel->profile_mutex);
}
return times;
}
SWITCH_DECLARE(switch_status_t) switch_channel_alloc(switch_channel_t **channel, switch_memory_pool_t *pool)
{
assert(pool != NULL);
if (((*channel) = switch_core_alloc(pool, sizeof(switch_channel_t))) == 0) {
return SWITCH_STATUS_MEMERR;
}
switch_core_hash_init(&(*channel)->variables, pool);
switch_core_hash_init(&(*channel)->private_hash, pool);
switch_buffer_create_dynamic(&(*channel)->dtmf_buffer, 128, 128, 0);
switch_mutex_init(&(*channel)->dtmf_mutex, SWITCH_MUTEX_NESTED, pool);
switch_mutex_init(&(*channel)->flag_mutex, SWITCH_MUTEX_NESTED, pool);
switch_mutex_init(&(*channel)->profile_mutex, SWITCH_MUTEX_NESTED, pool);
(*channel)->hangup_cause = SWITCH_CAUSE_UNALLOCATED;
return SWITCH_STATUS_SUCCESS;
}
SWITCH_DECLARE(switch_size_t) switch_channel_has_dtmf(switch_channel_t *channel)
{
switch_size_t has;
assert(channel != NULL);
switch_mutex_lock(channel->dtmf_mutex);
has = switch_buffer_inuse(channel->dtmf_buffer);
switch_mutex_unlock(channel->dtmf_mutex);
return has;
}
SWITCH_DECLARE(switch_status_t) switch_channel_queue_dtmf(switch_channel_t *channel, char *dtmf)
{
switch_status_t status;
register switch_size_t len, inuse;
switch_size_t wr = 0;
char *p;
assert(channel != NULL);
switch_mutex_lock(channel->dtmf_mutex);
inuse = switch_buffer_inuse(channel->dtmf_buffer);
len = strlen(dtmf);
if (len + inuse > switch_buffer_len(channel->dtmf_buffer)) {
switch_buffer_toss(channel->dtmf_buffer, strlen(dtmf));
}
p = dtmf;
while(wr < len && p) {
if (is_dtmf(*p)) {
wr++;
} else {
break;
}
p++;
}
status = switch_buffer_write(channel->dtmf_buffer, dtmf, wr) ? SWITCH_STATUS_SUCCESS : SWITCH_STATUS_MEMERR;
switch_mutex_unlock(channel->dtmf_mutex);
return status;
}
SWITCH_DECLARE(switch_size_t) switch_channel_dequeue_dtmf(switch_channel_t *channel, char *dtmf, switch_size_t len)
{
switch_size_t bytes;
switch_event_t *event;
assert(channel != NULL);
switch_mutex_lock(channel->dtmf_mutex);
if ((bytes = switch_buffer_read(channel->dtmf_buffer, dtmf, len)) > 0) {
*(dtmf + bytes) = '\0';
}
switch_mutex_unlock(channel->dtmf_mutex);
if (bytes && switch_event_create(&event, SWITCH_EVENT_DTMF) == SWITCH_STATUS_SUCCESS) {
switch_channel_event_set_data(channel, event);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "DTMF-String", dtmf);
switch_event_fire(&event);
}
return bytes;
}
SWITCH_DECLARE(void) switch_channel_uninit(switch_channel_t *channel)
{
switch_buffer_destroy(&channel->dtmf_buffer);
}
SWITCH_DECLARE(switch_status_t) switch_channel_init(switch_channel_t *channel,
switch_core_session_t *session,
switch_channel_state_t state, uint32_t flags)
{
assert(channel != NULL);
channel->state = state;
channel->flags = flags;
channel->session = session;
return SWITCH_STATUS_SUCCESS;
}
SWITCH_DECLARE(void) switch_channel_presence(switch_channel_t *channel, char *rpid, char *status)
{
char *id = switch_channel_get_variable(channel, "presence_id");
switch_event_t *event;
switch_event_types_t type = SWITCH_EVENT_PRESENCE_IN;
if (!status) {
type = SWITCH_EVENT_PRESENCE_OUT;
}
if (!id) {
return;
}
if (switch_event_create(&event, type) == SWITCH_STATUS_SUCCESS) {
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "proto", __FILE__);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "login", "%s", __FILE__);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "from", "%s", id);
if (type == SWITCH_EVENT_PRESENCE_IN) {
if (!rpid) {
rpid = "unknown";
}
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "rpid", "%s", rpid);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "status", "%s", status);
}
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "event_type", "presence");
switch_event_fire(&event);
}
}
SWITCH_DECLARE(char *) switch_channel_get_variable(switch_channel_t *channel, char *varname)
{
Ringback (sponsored by Front Logic) This addition lets you set artifical ringback on a channel that is waiting for an originated call to be answered. the syntax is <action application="set" data="ringback=[data]"/> where data is either the full path to an audio file or a teletone generation script.. syntax of teletone scripts LEGEND: 0-9,a-d,*,# (standard dtmf tones) variables: c,r,d,v,>,<,+,w,l,L,% c (channels) - Sets the number of channels. r (rate) - Sets the sample rate. d (duration) - Sets the default tone duration. v (volume) - Sets the default volume. > (decrease vol) - factor to decrease volume by per frame (0 for even decrease across duration). < (increase vol) - factor to increase volume by per frame (0 for even increase across duration). + (step) - factor to step by used by < and >. w (wait) - default silence after each tone. l (loops) - number of times to repeat each tone in the script. L (LOOPS) - number of times to repeat the the whole script. % (manual tone) - a generic tone specified by a duration, a wait and a list of frequencies. standard tones can have custom duration per use with the () modifier 7(1000, 500) to generate DTMF 7 for 1 second then pause .5 seconds EXAMPLES UK Ring Tone [400+450 hz on for 400ms off for 200ms then 400+450 hz on for 400ms off for 2200ms] %(400,200,400,450);%(400,2200,400,450) US Ring Tone [440+480 hz on for 2000ms off for 4000ms] %(2000,4000,440,480) ATT BONG [volume level 4000, even decay, step by 2, # key for 60ms with no wait, volume level 2000, 350+440hz {us dialtone} for 940ms v=4000;>=0;+=2;#(60,0);v=2000;%(940,0,350,440) SIT Tone 913.8 hz for 274 ms with no wait, 1370.6 hz for 274 ms with no wait, 1776.7 hz for 380ms with no wait %(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7) ATTN TONE (phone's off the hook!) 1400+2060+2450+2600 hz for 100ms with 100ms wait %(100,100,1400,2060,2450,2600) git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3408 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-11-19 01:05:06 +00:00
char *v = NULL;
assert(channel != NULL);
if (!(v=switch_core_hash_find(channel->variables, varname))) {
if (!channel->caller_profile || !(v = switch_caller_get_field_by_name(channel->caller_profile, varname))) {
if (!strcmp(varname, "base_dir")) {
return SWITCH_GLOBAL_dirs.base_dir;
}
Ringback (sponsored by Front Logic) This addition lets you set artifical ringback on a channel that is waiting for an originated call to be answered. the syntax is <action application="set" data="ringback=[data]"/> where data is either the full path to an audio file or a teletone generation script.. syntax of teletone scripts LEGEND: 0-9,a-d,*,# (standard dtmf tones) variables: c,r,d,v,>,<,+,w,l,L,% c (channels) - Sets the number of channels. r (rate) - Sets the sample rate. d (duration) - Sets the default tone duration. v (volume) - Sets the default volume. > (decrease vol) - factor to decrease volume by per frame (0 for even decrease across duration). < (increase vol) - factor to increase volume by per frame (0 for even increase across duration). + (step) - factor to step by used by < and >. w (wait) - default silence after each tone. l (loops) - number of times to repeat each tone in the script. L (LOOPS) - number of times to repeat the the whole script. % (manual tone) - a generic tone specified by a duration, a wait and a list of frequencies. standard tones can have custom duration per use with the () modifier 7(1000, 500) to generate DTMF 7 for 1 second then pause .5 seconds EXAMPLES UK Ring Tone [400+450 hz on for 400ms off for 200ms then 400+450 hz on for 400ms off for 2200ms] %(400,200,400,450);%(400,2200,400,450) US Ring Tone [440+480 hz on for 2000ms off for 4000ms] %(2000,4000,440,480) ATT BONG [volume level 4000, even decay, step by 2, # key for 60ms with no wait, volume level 2000, 350+440hz {us dialtone} for 940ms v=4000;>=0;+=2;#(60,0);v=2000;%(940,0,350,440) SIT Tone 913.8 hz for 274 ms with no wait, 1370.6 hz for 274 ms with no wait, 1776.7 hz for 380ms with no wait %(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7) ATTN TONE (phone's off the hook!) 1400+2060+2450+2600 hz for 100ms with 100ms wait %(100,100,1400,2060,2450,2600) git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3408 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-11-19 01:05:06 +00:00
v = switch_core_get_variable(varname);
}
}
return v;
}
SWITCH_DECLARE(switch_hash_index_t *) switch_channel_variable_first(switch_channel_t *channel, switch_memory_pool_t *pool)
{
assert(channel != NULL);
return switch_hash_first(pool, channel->variables);
}
SWITCH_DECLARE(switch_status_t) switch_channel_set_private(switch_channel_t *channel, char *key, void *private_info)
{
assert(channel != NULL);
switch_core_hash_insert_dup(channel->private_hash, switch_core_session_strdup(channel->session, key), private_info);
return SWITCH_STATUS_SUCCESS;
}
SWITCH_DECLARE(void *) switch_channel_get_private(switch_channel_t *channel, char *key)
{
assert(channel != NULL);
return switch_core_hash_find(channel->private_hash, key);
}
SWITCH_DECLARE(switch_status_t) switch_channel_set_name(switch_channel_t *channel, char *name)
{
assert(channel != NULL);
channel->name = NULL;
if (name) {
char *uuid = switch_core_session_get_uuid(channel->session);
channel->name = switch_core_session_strdup(channel->session, name);
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "New Chan %s [%s]\n", name, uuid);
}
return SWITCH_STATUS_SUCCESS;
}
SWITCH_DECLARE(char *) switch_channel_get_name(switch_channel_t *channel)
{
assert(channel != NULL);
return channel->name ? channel->name : "N/A";
}
SWITCH_DECLARE(switch_status_t) switch_channel_set_variable(switch_channel_t *channel, char *varname, char *value)
{
assert(channel != NULL);
if (varname) {
switch_core_hash_delete(channel->variables, varname);
if (!switch_strlen_zero(value)) {
switch_core_hash_insert_dup(channel->variables, varname, switch_core_session_strdup(channel->session, value));
} else {
switch_core_hash_delete(channel->variables, varname);
}
return SWITCH_STATUS_SUCCESS;
}
return SWITCH_STATUS_FALSE;
}
SWITCH_DECLARE(int) switch_channel_test_flag(switch_channel_t *channel, switch_channel_flag_t flags)
{
assert(channel != NULL);
return switch_test_flag(channel, flags) ? 1 : 0;
}
SWITCH_DECLARE(void) switch_channel_set_flag(switch_channel_t *channel, switch_channel_flag_t flags)
{
assert(channel != NULL);
switch_set_flag_locked(channel, flags);
}
SWITCH_DECLARE(void) switch_channel_set_state_flag(switch_channel_t *channel, switch_channel_flag_t flags)
{
assert(channel != NULL);
switch_mutex_lock(channel->flag_mutex);
channel->state_flags |= flags;
switch_mutex_unlock(channel->flag_mutex);
}
SWITCH_DECLARE(void) switch_channel_clear_flag(switch_channel_t *channel, switch_channel_flag_t flags)
{
assert(channel != NULL);
switch_clear_flag_locked(channel, flags);
}
SWITCH_DECLARE(switch_channel_state_t) switch_channel_get_state(switch_channel_t *channel)
{
switch_channel_state_t state;
assert(channel != NULL);
switch_mutex_lock(channel->flag_mutex);
state = channel->state;
switch_mutex_unlock(channel->flag_mutex);
return state;
}
SWITCH_DECLARE(uint8_t) switch_channel_ready(switch_channel_t *channel)
{
assert(channel != NULL);
return (channel->state > CS_RING && channel->state < CS_HANGUP && !switch_test_flag(channel, CF_TRANSFER)) ? 1 : 0;
}
static const char *state_names[] = {
"CS_NEW",
"CS_INIT",
"CS_RING",
"CS_TRANSMIT",
"CS_EXECUTE",
"CS_LOOPBACK",
"CS_HOLD",
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
"CS_HIBERNATE",
"CS_HANGUP",
"CS_DONE",
NULL
};
SWITCH_DECLARE(const char *) switch_channel_state_name(switch_channel_state_t state)
{
return state_names[state];
}
SWITCH_DECLARE(switch_channel_state_t) switch_channel_name_state(char *name)
{
uint32_t x = 0;
for(x = 0; state_names[x]; x++) {
if (!strcasecmp(state_names[x], name)) {
return (switch_channel_state_t) x;
}
}
return CS_DONE;
}
SWITCH_DECLARE(switch_channel_state_t) switch_channel_perform_set_state(switch_channel_t *channel,
const char *file,
const char *func,
int line,
switch_channel_state_t state)
{
switch_channel_state_t last_state;
int ok = 0;
assert(channel != NULL);
switch_mutex_lock(channel->flag_mutex);
last_state = channel->state;
if (last_state == state) {
goto done;
}
if (last_state >= CS_HANGUP && state < last_state) {
goto done;
}
/* STUB for more dev
case CS_INIT:
switch(state) {
case CS_NEW:
case CS_INIT:
case CS_LOOPBACK:
case CS_TRANSMIT:
case CS_RING:
case CS_EXECUTE:
case CS_HANGUP:
case CS_DONE:
default:
break;
}
break;
*/
switch (last_state) {
case CS_NEW:
switch (state) {
default:
ok++;
break;
}
break;
case CS_INIT:
switch (state) {
case CS_LOOPBACK:
case CS_TRANSMIT:
case CS_RING:
case CS_EXECUTE:
case CS_HOLD:
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 CS_HIBERNATE:
ok++;
default:
break;
}
break;
case CS_LOOPBACK:
switch (state) {
case CS_TRANSMIT:
case CS_RING:
case CS_EXECUTE:
case CS_HOLD:
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 CS_HIBERNATE:
ok++;
default:
break;
}
break;
case CS_TRANSMIT:
switch (state) {
case CS_LOOPBACK:
case CS_RING:
case CS_EXECUTE:
case CS_HOLD:
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 CS_HIBERNATE:
ok++;
default:
break;
}
break;
case CS_HOLD:
switch (state) {
case CS_LOOPBACK:
case CS_RING:
case CS_EXECUTE:
case CS_TRANSMIT:
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 CS_HIBERNATE:
ok++;
default:
break;
}
break;
case CS_HIBERNATE:
switch (state) {
case CS_LOOPBACK:
case CS_RING:
case CS_EXECUTE:
case CS_TRANSMIT:
case CS_HOLD:
ok++;
default:
break;
}
break;
case CS_RING:
switch_clear_flag(channel, CF_TRANSFER);
switch (state) {
case CS_LOOPBACK:
case CS_EXECUTE:
case CS_TRANSMIT:
case CS_HOLD:
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 CS_HIBERNATE:
ok++;
default:
break;
}
break;
case CS_EXECUTE:
switch (state) {
case CS_LOOPBACK:
case CS_TRANSMIT:
case CS_RING:
case CS_HOLD:
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 CS_HIBERNATE:
ok++;
default:
break;
}
break;
case CS_HANGUP:
switch (state) {
case CS_DONE:
ok++;
default:
break;
}
break;
default:
break;
}
if (ok) {
if (state > CS_RING) {
switch_channel_presence(channel, "unknown", (char*)state_names[state]);
}
switch_log_printf(SWITCH_CHANNEL_ID_LOG, (char *) file, func, line, SWITCH_LOG_DEBUG, "%s State Change %s -> %s\n",
channel->name,
state_names[last_state],
state_names[state]);
switch_mutex_lock(channel->flag_mutex);
channel->state = state;
switch_mutex_unlock(channel->flag_mutex);
if (state == CS_HANGUP && !channel->hangup_cause) {
channel->hangup_cause = SWITCH_CAUSE_NORMAL_CLEARING;
}
if (state < CS_HANGUP) {
switch_event_t *event;
if (switch_event_create(&event, SWITCH_EVENT_CHANNEL_STATE) == SWITCH_STATUS_SUCCESS) {
if (state == CS_RING) {
switch_channel_event_set_data(channel, event);
} else {
char state_num[25];
snprintf(state_num, sizeof(state_num), "%d", channel->state);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "Channel-State", (char *) switch_channel_state_name(channel->state));
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "Channel-State-Number", (char *) state_num);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "Channel-Name", switch_channel_get_name(channel));
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "Unique-ID", switch_core_session_get_uuid(channel->session));
}
switch_event_fire(&event);
}
}
if (state < CS_DONE) {
switch_core_session_signal_state_change(channel->session);
}
} else {
switch_log_printf(SWITCH_CHANNEL_ID_LOG, (char *) file, func, line, SWITCH_LOG_WARNING, "%s Invalid State Change %s -> %s\n",
channel->name,
state_names[last_state],
state_names[state]);
//we won't tolerate an invalid state change so we can make sure we are as robust as a nice cup of dark coffee!
if (channel->state < CS_HANGUP) {
// not cool lets crash this bad boy and figure out wtf is going on
assert(0);
}
}
done:
if (channel->state_flags) {
channel->flags |= channel->state_flags;
channel->state_flags = 0;
}
switch_mutex_unlock(channel->flag_mutex);
return channel->state;
}
SWITCH_DECLARE(void) switch_channel_event_set_data(switch_channel_t *channel, switch_event_t *event)
{
switch_caller_profile_t *caller_profile, *originator_caller_profile = NULL, *originatee_caller_profile = NULL;
switch_hash_index_t *hi;
switch_codec_t *codec;
void *val;
const void *var;
char state_num[25];
if ((caller_profile = switch_channel_get_caller_profile(channel))) {
originator_caller_profile = caller_profile->originator_caller_profile;
originatee_caller_profile = caller_profile->originatee_caller_profile;
}
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "Channel-State", (char *) switch_channel_state_name(channel->state));
snprintf(state_num, sizeof(state_num), "%d", channel->state);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "Channel-State-Number", (char *) state_num);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "Channel-Name", switch_channel_get_name(channel));
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "Unique-ID", switch_core_session_get_uuid(channel->session));
if ((codec = switch_core_session_get_read_codec(channel->session))) {
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "Channel-Read-Codec-Name", codec->implementation->iananame);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "Channel-Read-Codec-Rate", "%u", codec->implementation->samples_per_second);
}
if ((codec = switch_core_session_get_write_codec(channel->session))) {
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "Channel-Write-Codec-Name", codec->implementation->iananame);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "Channel-Write-Codec-Rate", "%u", codec->implementation->samples_per_second);
}
/* Index Caller's Profile */
if (caller_profile) {
switch_caller_profile_event_set_data(caller_profile, "Caller", event);
}
/* Index Originator's Profile */
if (originator_caller_profile) {
switch_caller_profile_event_set_data(originator_caller_profile, "Originator", event);
}
/* Index Originatee's Profile */
if (originatee_caller_profile) {
switch_caller_profile_event_set_data(originatee_caller_profile, "Originatee", event);
}
/* Index Variables */
for (hi = switch_hash_first(switch_core_session_get_pool(channel->session), channel->variables); hi;
hi = switch_hash_next(hi)) {
char buf[1024];
switch_hash_this(hi, &var, NULL, &val);
snprintf(buf, sizeof(buf), "variable_%s", (char *) var);
switch_event_add_header(event, SWITCH_STACK_BOTTOM, buf, (char *) val);
}
}
SWITCH_DECLARE(void) switch_channel_set_caller_profile(switch_channel_t *channel, switch_caller_profile_t *caller_profile)
{
assert(channel != NULL);
assert(channel->session != NULL);
switch_mutex_lock(channel->profile_mutex);
assert(caller_profile != NULL);
if (!caller_profile->uuid) {
caller_profile->uuid = switch_core_session_strdup(channel->session, switch_core_session_get_uuid(channel->session));
}
if (!caller_profile->chan_name) {
caller_profile->chan_name = switch_core_session_strdup(channel->session, channel->name);
}
if (!caller_profile->context) {
caller_profile->chan_name = switch_core_session_strdup(channel->session, "default");
}
if (!channel->caller_profile) {
switch_event_t *event;
if (switch_event_create(&event, SWITCH_EVENT_CHANNEL_CREATE) == SWITCH_STATUS_SUCCESS) {
switch_channel_event_set_data(channel, event);
switch_event_fire(&event);
}
}
caller_profile->times = (switch_channel_timetable_t *) switch_core_session_alloc(channel->session, sizeof(*caller_profile->times));
caller_profile->times->created = switch_time_now();
if (channel->caller_profile && channel->caller_profile->times) {
channel->caller_profile->times->transferred = switch_time_now();
caller_profile->times->answered = channel->caller_profile->times->answered;
}
caller_profile->next = channel->caller_profile;
channel->caller_profile = caller_profile;
switch_mutex_unlock(channel->profile_mutex);
}
SWITCH_DECLARE(switch_caller_profile_t *) switch_channel_get_caller_profile(switch_channel_t *channel)
{
switch_caller_profile_t *profile;
assert(channel != NULL);
switch_mutex_lock(channel->profile_mutex);
profile = channel->caller_profile;
switch_mutex_unlock(channel->profile_mutex);
return profile;
}
SWITCH_DECLARE(void) switch_channel_set_originator_caller_profile(switch_channel_t *channel,
switch_caller_profile_t *caller_profile)
{
assert(channel != NULL);
if (channel->caller_profile) {
switch_mutex_lock(channel->profile_mutex);
caller_profile->next = channel->caller_profile->originator_caller_profile;
channel->caller_profile->originator_caller_profile = caller_profile;
switch_mutex_unlock(channel->profile_mutex);
}
}
SWITCH_DECLARE(void) switch_channel_set_originatee_caller_profile(switch_channel_t *channel,
switch_caller_profile_t *caller_profile)
{
assert(channel != NULL);
if (channel->caller_profile) {
switch_mutex_lock(channel->profile_mutex);
caller_profile->next = channel->caller_profile->originatee_caller_profile;
channel->caller_profile->originatee_caller_profile = caller_profile;
switch_mutex_unlock(channel->profile_mutex);
}
}
SWITCH_DECLARE(switch_caller_profile_t *) switch_channel_get_originator_caller_profile(switch_channel_t *channel)
{
switch_caller_profile_t *profile = NULL;
assert(channel != NULL);
if (channel->caller_profile) {
switch_mutex_lock(channel->profile_mutex);
profile = channel->caller_profile->originator_caller_profile;
switch_mutex_unlock(channel->profile_mutex);
}
return profile;
}
SWITCH_DECLARE(switch_caller_profile_t *) switch_channel_get_originatee_caller_profile(switch_channel_t *channel)
{
switch_caller_profile_t *profile = NULL;
assert(channel != NULL);
if (channel->caller_profile) {
switch_mutex_lock(channel->profile_mutex);
profile = channel->caller_profile->originatee_caller_profile;
switch_mutex_unlock(channel->profile_mutex);
}
return profile;
}
SWITCH_DECLARE(char *) switch_channel_get_uuid(switch_channel_t *channel)
{
assert(channel != NULL);
assert(channel->session != NULL);
return switch_core_session_get_uuid(channel->session);
}
SWITCH_DECLARE(int) switch_channel_add_state_handler(switch_channel_t *channel,
const switch_state_handler_table_t *state_handler)
{
int x, index;
assert(channel != NULL);
switch_mutex_lock(channel->flag_mutex);
for (x = 0; x < SWITCH_MAX_STATE_HANDLERS; x++) {
if (channel->state_handlers[x] == state_handler) {
index = x;
goto end;
}
}
index = channel->state_handler_index++;
if (channel->state_handler_index >= SWITCH_MAX_STATE_HANDLERS) {
index = -1;
goto end;
}
channel->state_handlers[index] = state_handler;
end:
switch_mutex_unlock(channel->flag_mutex);
return index;
}
SWITCH_DECLARE(const switch_state_handler_table_t *) switch_channel_get_state_handler(switch_channel_t *channel, int index)
{
const switch_state_handler_table_t *h = NULL;
assert(channel != NULL);
if (index > SWITCH_MAX_STATE_HANDLERS || index > channel->state_handler_index) {
return NULL;
}
switch_mutex_lock(channel->flag_mutex);
h = channel->state_handlers[index];
switch_mutex_unlock(channel->flag_mutex);
return h;
}
SWITCH_DECLARE(void) switch_channel_clear_state_handler(switch_channel_t *channel, const switch_state_handler_table_t *state_handler)
{
int index, i = channel->state_handler_index;
const switch_state_handler_table_t *new_handlers[SWITCH_MAX_STATE_HANDLERS] = {0};
switch_mutex_lock(channel->flag_mutex);
assert(channel != NULL);
channel->state_handler_index = 0;
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
if (state_handler) {
for (index = 0; index < i; index++) {
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
if (channel->state_handlers[index] != state_handler) {
new_handlers[channel->state_handler_index++] = channel->state_handlers[index];
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
}
}
}
for (index = 0; index < SWITCH_MAX_STATE_HANDLERS; index++) {
channel->state_handlers[index] = NULL;
}
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
if (state_handler) {
for (index = 0; index < channel->state_handler_index; index++) {
channel->state_handlers[index] = new_handlers[index];
*deep breath* Ok, This one adds a bunch of stuff on top of the framework restructuring from yesterday. 1) originate api function: Usage: originate <call url> <exten> [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] This will call the specified url then transfer the call to the specified extension example: originate exosip/1000@somehost 1000 XML default 2) mutiple destinations in outbound calls: This means any dialstring may contain an '&' separated list of call urls When using mutiple urls in this manner it is possible to map a certian key as required indication of an accepted call. You may also supply a filename to play possibly instructing the call recipiant to press the desired key etc... The example below will call 2 locations playing prompt.wav to any who answer and completing the call to the first offhook recipiant to dial "4" <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="set" data="call_timeout=60"/> <action application="set" data="group_confirm_file=/path/to/prompt.wav"/> <action application="set" data="group_confirm_key=4"/> <action application="bridge" data="iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> The following is the equivilant but the confirm data is passed vial the bridge parameters (This is for situations where there is no originating channel to set variables to) <extension name="3002"> <condition field="destination_number" expression="^3002$"> <action application="bridge" data=/path/to/prompt.wav:4"confirm=iax/guest@somebox/1234&exosip/1000@somehost"/> </condition> </extension> Omitting the file and key stuff will simply comeplete the call to whoever answers first. (this is similar to how other less fortunate software handles the situation with thier best effort.) This logic should be permitted in anything that establishes an outgoing call with switch_ivr_originate() Yes! That means even in this new originate api command you can call mutiple targets and send whoever answers first to an extension that calls more mutiple targets. (better test it though!) Oh, and you should be able to do the same in the mod_conference dial and dynamic conference features please report any behaviour contrary to this account to me ASAP cos i would not be terribly suprised if I forgot some scenerio that causes an explosion I did all this in 1 afternoon so it probably needs tuning still. git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2311 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-08-17 00:53:09 +00:00
}
}
switch_mutex_unlock(channel->flag_mutex);
}
SWITCH_DECLARE(void) switch_channel_set_caller_extension(switch_channel_t *channel,
switch_caller_extension_t *caller_extension)
{
assert(channel != NULL);
switch_mutex_lock(channel->profile_mutex);
caller_extension->next = channel->caller_profile->caller_extension;
channel->caller_profile->caller_extension = caller_extension;
switch_mutex_unlock(channel->profile_mutex);
}
SWITCH_DECLARE(switch_caller_extension_t *) switch_channel_get_caller_extension(switch_channel_t *channel)
{
switch_caller_extension_t *extension;
assert(channel != NULL);
switch_mutex_lock(channel->profile_mutex);
extension = channel->caller_profile->caller_extension;
switch_mutex_unlock(channel->profile_mutex);
return extension;
}
SWITCH_DECLARE(switch_channel_state_t) switch_channel_perform_hangup(switch_channel_t *channel,
const char *file,
const char *func,
int line,
switch_call_cause_t hangup_cause)
{
assert(channel != NULL);
switch_mutex_lock(channel->flag_mutex);
if (channel->caller_profile && channel->caller_profile->times && !channel->caller_profile->times->hungup) {
switch_mutex_lock(channel->profile_mutex);
channel->caller_profile->times->hungup = switch_time_now();
switch_mutex_unlock(channel->profile_mutex);
}
if (channel->state < CS_HANGUP) {
switch_event_t *event;
switch_channel_state_t last_state = channel->state;
channel->state = CS_HANGUP;
channel->hangup_cause = hangup_cause;
switch_log_printf(SWITCH_CHANNEL_ID_LOG, (char *) file, func, line, SWITCH_LOG_NOTICE, "Hangup %s [%s] [%s]\n",
channel->name,
state_names[last_state], switch_channel_cause2str(channel->hangup_cause));
if (switch_event_create(&event, SWITCH_EVENT_CHANNEL_HANGUP) == SWITCH_STATUS_SUCCESS) {
switch_event_add_header(event, SWITCH_STACK_BOTTOM, "Hangup-Cause", switch_channel_cause2str(channel->hangup_cause));
switch_channel_event_set_data(channel, event);
switch_event_fire(&event);
}
switch_channel_set_variable(channel, "hangup_cause", switch_channel_cause2str(channel->hangup_cause));
switch_channel_presence(channel, "unavailable", switch_channel_cause2str(channel->hangup_cause));
switch_core_session_kill_channel(channel->session, SWITCH_SIG_KILL);
switch_core_session_signal_state_change(channel->session);
}
switch_mutex_unlock(channel->flag_mutex);
return channel->state;
}
SWITCH_DECLARE(switch_status_t) switch_channel_perform_pre_answer(switch_channel_t *channel,
const char *file,
const char *func,
int line)
{
switch_core_session_message_t msg;
char *uuid = switch_core_session_get_uuid(channel->session);
switch_status_t status;
assert(channel != NULL);
if (channel->state >= CS_HANGUP) {
return SWITCH_STATUS_FALSE;
}
if (switch_channel_test_flag(channel, CF_ANSWERED)) {
return SWITCH_STATUS_SUCCESS;
}
if (switch_channel_test_flag(channel, CF_EARLY_MEDIA)) {
return SWITCH_STATUS_SUCCESS;
}
msg.message_id = SWITCH_MESSAGE_INDICATE_PROGRESS;
msg.from = channel->name;
status = switch_core_session_message_send(uuid, &msg);
if (status == SWITCH_STATUS_SUCCESS) {
switch_event_t *event;
switch_log_printf(SWITCH_CHANNEL_ID_LOG, (char *) file, func, line, SWITCH_LOG_NOTICE, "Pre-Answer %s!\n", channel->name);
switch_channel_set_flag(channel, CF_EARLY_MEDIA);
switch_channel_set_variable(channel, "endpoint_disposition", "EARLY MEDIA");
if (switch_event_create(&event, SWITCH_EVENT_CHANNEL_PROGRESS) == SWITCH_STATUS_SUCCESS) {
switch_channel_event_set_data(channel, event);
switch_event_fire(&event);
}
}
return status;
}
SWITCH_DECLARE(switch_status_t) switch_channel_perform_ringback(switch_channel_t *channel,
const char *file,
const char *func,
int line)
{
switch_core_session_message_t msg;
char *uuid = switch_core_session_get_uuid(channel->session);
switch_status_t status;
assert(channel != NULL);
if (channel->state >= CS_HANGUP) {
return SWITCH_STATUS_FALSE;
}
if (switch_channel_test_flag(channel, CF_ANSWERED)) {
return SWITCH_STATUS_SUCCESS;
}
if (switch_channel_test_flag(channel, CF_EARLY_MEDIA)) {
return SWITCH_STATUS_SUCCESS;
}
msg.message_id = SWITCH_MESSAGE_INDICATE_RINGING;
msg.from = channel->name;
status = switch_core_session_message_send(uuid, &msg);
if (status == SWITCH_STATUS_SUCCESS) {
switch_log_printf(SWITCH_CHANNEL_ID_LOG, (char *) file, func, line, SWITCH_LOG_NOTICE, "Ringback %s!\n", channel->name);
}
return status;
}
SWITCH_DECLARE(switch_status_t) switch_channel_perform_mark_answered(switch_channel_t *channel,
const char *file,
const char *func,
int line)
{
switch_event_t *event;
assert(channel != NULL);
if (channel->state >= CS_HANGUP) {
return SWITCH_STATUS_FALSE;
}
if (switch_channel_test_flag(channel, CF_ANSWERED)) {
return SWITCH_STATUS_SUCCESS;
}
if (channel->caller_profile && channel->caller_profile->times) {
switch_mutex_lock(channel->profile_mutex);
channel->caller_profile->times->answered = switch_time_now();
switch_mutex_unlock(channel->profile_mutex);
}
switch_channel_set_flag(channel, CF_ANSWERED);
if (switch_event_create(&event, SWITCH_EVENT_CHANNEL_ANSWER) == SWITCH_STATUS_SUCCESS) {
switch_channel_event_set_data(channel, event);
switch_event_fire(&event);
}
switch_channel_set_variable(channel, "endpoint_disposition", "ANSWER");
switch_log_printf(SWITCH_CHANNEL_ID_LOG, (char *) file, func, line, SWITCH_LOG_NOTICE, "Channel [%s] has been answered\n", channel->name);
return SWITCH_STATUS_SUCCESS;
}
SWITCH_DECLARE(switch_status_t) switch_channel_perform_answer(switch_channel_t *channel,
const char *file,
const char *func,
int line)
{
assert(channel != NULL);
if (channel->state >= CS_HANGUP) {
return SWITCH_STATUS_FALSE;
}
if (switch_channel_test_flag(channel, CF_ANSWERED)) {
return SWITCH_STATUS_SUCCESS;
}
if (switch_core_session_answer_channel(channel->session) == SWITCH_STATUS_SUCCESS) {
return switch_channel_perform_mark_answered(channel, file, func, line);
}
return SWITCH_STATUS_FALSE;
}
#define resize(l) {\
char *dp;\
olen += (len + l + block);\
cpos = c - data;\
if ((dp = realloc(data, olen))) {\
data = dp;\
c = data + cpos;\
memset(c, 0, olen - cpos);\
}} \
SWITCH_DECLARE(char *) switch_channel_expand_variables(switch_channel_t *channel, char *in)
{
char *p, *c;
char *data, *indup;
Ringback (sponsored by Front Logic) This addition lets you set artifical ringback on a channel that is waiting for an originated call to be answered. the syntax is <action application="set" data="ringback=[data]"/> where data is either the full path to an audio file or a teletone generation script.. syntax of teletone scripts LEGEND: 0-9,a-d,*,# (standard dtmf tones) variables: c,r,d,v,>,<,+,w,l,L,% c (channels) - Sets the number of channels. r (rate) - Sets the sample rate. d (duration) - Sets the default tone duration. v (volume) - Sets the default volume. > (decrease vol) - factor to decrease volume by per frame (0 for even decrease across duration). < (increase vol) - factor to increase volume by per frame (0 for even increase across duration). + (step) - factor to step by used by < and >. w (wait) - default silence after each tone. l (loops) - number of times to repeat each tone in the script. L (LOOPS) - number of times to repeat the the whole script. % (manual tone) - a generic tone specified by a duration, a wait and a list of frequencies. standard tones can have custom duration per use with the () modifier 7(1000, 500) to generate DTMF 7 for 1 second then pause .5 seconds EXAMPLES UK Ring Tone [400+450 hz on for 400ms off for 200ms then 400+450 hz on for 400ms off for 2200ms] %(400,200,400,450);%(400,2200,400,450) US Ring Tone [440+480 hz on for 2000ms off for 4000ms] %(2000,4000,440,480) ATT BONG [volume level 4000, even decay, step by 2, # key for 60ms with no wait, volume level 2000, 350+440hz {us dialtone} for 940ms v=4000;>=0;+=2;#(60,0);v=2000;%(940,0,350,440) SIT Tone 913.8 hz for 274 ms with no wait, 1370.6 hz for 274 ms with no wait, 1776.7 hz for 380ms with no wait %(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7) ATTN TONE (phone's off the hook!) 1400+2060+2450+2600 hz for 100ms with 100ms wait %(100,100,1400,2060,2450,2600) git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3408 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-11-19 01:05:06 +00:00
size_t sp = 0, len = 0, olen = 0, vtype = 0, br = 0, cpos, block = 128;
char *sub_val = NULL, *func_val = NULL;
if (!strchr(in, '$')) {
return in;
}
olen = strlen(in);
indup = strdup(in);
if ((data = malloc(olen))) {
memset(data, 0, olen);
c = data;
for(p = indup; *p; p++) {
vtype = 0;
if (*p == '$') {
vtype = 1;
if (*(p+1) != '{') {
vtype = 2;
}
}
if (vtype) {
char *s = p, *e, *vname, *vval = NULL;
size_t nlen;
s++;
if (vtype == 1 && *s == '{') {
br = 1;
s++;
}
e = s;
vname = s;
while (*e) {
if (!br && *e == ' ') {
*e++ = '\0';
sp++;
break;
}
if (br == 1 && *e == '}') {
br = 0;
*e++ = '\0';
break;
}
if (vtype == 2) {
if (*e == '(') {
*e++ = '\0';
vval = e;
br = 2;
}
if (br == 2 && *e == ')') {
*e++ = '\0';
br = 0;
break;
}
}
e++;
}
p = e;
if (vtype == 1) {
sub_val = switch_channel_get_variable(channel, vname);
} else {
switch_stream_handle_t stream = {0};
SWITCH_STANDARD_STREAM(stream);
if (stream.data) {
if (switch_api_execute(vname, vval, channel->session, &stream) == SWITCH_STATUS_SUCCESS) {
func_val = stream.data;
sub_val = func_val;
} else {
free(stream.data);
}
} else {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "Memory Error!\n");
free(data);
free(indup);
return in;
}
}
if ((nlen = sub_val ? strlen(sub_val) : 0)) {
if (len + nlen >= olen) {
resize(nlen);
}
Ringback (sponsored by Front Logic) This addition lets you set artifical ringback on a channel that is waiting for an originated call to be answered. the syntax is <action application="set" data="ringback=[data]"/> where data is either the full path to an audio file or a teletone generation script.. syntax of teletone scripts LEGEND: 0-9,a-d,*,# (standard dtmf tones) variables: c,r,d,v,>,<,+,w,l,L,% c (channels) - Sets the number of channels. r (rate) - Sets the sample rate. d (duration) - Sets the default tone duration. v (volume) - Sets the default volume. > (decrease vol) - factor to decrease volume by per frame (0 for even decrease across duration). < (increase vol) - factor to increase volume by per frame (0 for even increase across duration). + (step) - factor to step by used by < and >. w (wait) - default silence after each tone. l (loops) - number of times to repeat each tone in the script. L (LOOPS) - number of times to repeat the the whole script. % (manual tone) - a generic tone specified by a duration, a wait and a list of frequencies. standard tones can have custom duration per use with the () modifier 7(1000, 500) to generate DTMF 7 for 1 second then pause .5 seconds EXAMPLES UK Ring Tone [400+450 hz on for 400ms off for 200ms then 400+450 hz on for 400ms off for 2200ms] %(400,200,400,450);%(400,2200,400,450) US Ring Tone [440+480 hz on for 2000ms off for 4000ms] %(2000,4000,440,480) ATT BONG [volume level 4000, even decay, step by 2, # key for 60ms with no wait, volume level 2000, 350+440hz {us dialtone} for 940ms v=4000;>=0;+=2;#(60,0);v=2000;%(940,0,350,440) SIT Tone 913.8 hz for 274 ms with no wait, 1370.6 hz for 274 ms with no wait, 1776.7 hz for 380ms with no wait %(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7) ATTN TONE (phone's off the hook!) 1400+2060+2450+2600 hz for 100ms with 100ms wait %(100,100,1400,2060,2450,2600) git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3408 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-11-19 01:05:06 +00:00
len += nlen;
strcat(c, sub_val);
c += nlen;
}
Ringback (sponsored by Front Logic) This addition lets you set artifical ringback on a channel that is waiting for an originated call to be answered. the syntax is <action application="set" data="ringback=[data]"/> where data is either the full path to an audio file or a teletone generation script.. syntax of teletone scripts LEGEND: 0-9,a-d,*,# (standard dtmf tones) variables: c,r,d,v,>,<,+,w,l,L,% c (channels) - Sets the number of channels. r (rate) - Sets the sample rate. d (duration) - Sets the default tone duration. v (volume) - Sets the default volume. > (decrease vol) - factor to decrease volume by per frame (0 for even decrease across duration). < (increase vol) - factor to increase volume by per frame (0 for even increase across duration). + (step) - factor to step by used by < and >. w (wait) - default silence after each tone. l (loops) - number of times to repeat each tone in the script. L (LOOPS) - number of times to repeat the the whole script. % (manual tone) - a generic tone specified by a duration, a wait and a list of frequencies. standard tones can have custom duration per use with the () modifier 7(1000, 500) to generate DTMF 7 for 1 second then pause .5 seconds EXAMPLES UK Ring Tone [400+450 hz on for 400ms off for 200ms then 400+450 hz on for 400ms off for 2200ms] %(400,200,400,450);%(400,2200,400,450) US Ring Tone [440+480 hz on for 2000ms off for 4000ms] %(2000,4000,440,480) ATT BONG [volume level 4000, even decay, step by 2, # key for 60ms with no wait, volume level 2000, 350+440hz {us dialtone} for 940ms v=4000;>=0;+=2;#(60,0);v=2000;%(940,0,350,440) SIT Tone 913.8 hz for 274 ms with no wait, 1370.6 hz for 274 ms with no wait, 1776.7 hz for 380ms with no wait %(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7) ATTN TONE (phone's off the hook!) 1400+2060+2450+2600 hz for 100ms with 100ms wait %(100,100,1400,2060,2450,2600) git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3408 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-11-19 01:05:06 +00:00
switch_safe_free(func_val);
sub_val = NULL;
vname = NULL;
vtype = 0;
}
if (len + 1 >= olen) {
resize(1);
}
if (sp) {
*c++ = ' ';
sp = 0;
Ringback (sponsored by Front Logic) This addition lets you set artifical ringback on a channel that is waiting for an originated call to be answered. the syntax is <action application="set" data="ringback=[data]"/> where data is either the full path to an audio file or a teletone generation script.. syntax of teletone scripts LEGEND: 0-9,a-d,*,# (standard dtmf tones) variables: c,r,d,v,>,<,+,w,l,L,% c (channels) - Sets the number of channels. r (rate) - Sets the sample rate. d (duration) - Sets the default tone duration. v (volume) - Sets the default volume. > (decrease vol) - factor to decrease volume by per frame (0 for even decrease across duration). < (increase vol) - factor to increase volume by per frame (0 for even increase across duration). + (step) - factor to step by used by < and >. w (wait) - default silence after each tone. l (loops) - number of times to repeat each tone in the script. L (LOOPS) - number of times to repeat the the whole script. % (manual tone) - a generic tone specified by a duration, a wait and a list of frequencies. standard tones can have custom duration per use with the () modifier 7(1000, 500) to generate DTMF 7 for 1 second then pause .5 seconds EXAMPLES UK Ring Tone [400+450 hz on for 400ms off for 200ms then 400+450 hz on for 400ms off for 2200ms] %(400,200,400,450);%(400,2200,400,450) US Ring Tone [440+480 hz on for 2000ms off for 4000ms] %(2000,4000,440,480) ATT BONG [volume level 4000, even decay, step by 2, # key for 60ms with no wait, volume level 2000, 350+440hz {us dialtone} for 940ms v=4000;>=0;+=2;#(60,0);v=2000;%(940,0,350,440) SIT Tone 913.8 hz for 274 ms with no wait, 1370.6 hz for 274 ms with no wait, 1776.7 hz for 380ms with no wait %(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7) ATTN TONE (phone's off the hook!) 1400+2060+2450+2600 hz for 100ms with 100ms wait %(100,100,1400,2060,2450,2600) git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3408 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-11-19 01:05:06 +00:00
len++;
}
Ringback (sponsored by Front Logic) This addition lets you set artifical ringback on a channel that is waiting for an originated call to be answered. the syntax is <action application="set" data="ringback=[data]"/> where data is either the full path to an audio file or a teletone generation script.. syntax of teletone scripts LEGEND: 0-9,a-d,*,# (standard dtmf tones) variables: c,r,d,v,>,<,+,w,l,L,% c (channels) - Sets the number of channels. r (rate) - Sets the sample rate. d (duration) - Sets the default tone duration. v (volume) - Sets the default volume. > (decrease vol) - factor to decrease volume by per frame (0 for even decrease across duration). < (increase vol) - factor to increase volume by per frame (0 for even increase across duration). + (step) - factor to step by used by < and >. w (wait) - default silence after each tone. l (loops) - number of times to repeat each tone in the script. L (LOOPS) - number of times to repeat the the whole script. % (manual tone) - a generic tone specified by a duration, a wait and a list of frequencies. standard tones can have custom duration per use with the () modifier 7(1000, 500) to generate DTMF 7 for 1 second then pause .5 seconds EXAMPLES UK Ring Tone [400+450 hz on for 400ms off for 200ms then 400+450 hz on for 400ms off for 2200ms] %(400,200,400,450);%(400,2200,400,450) US Ring Tone [440+480 hz on for 2000ms off for 4000ms] %(2000,4000,440,480) ATT BONG [volume level 4000, even decay, step by 2, # key for 60ms with no wait, volume level 2000, 350+440hz {us dialtone} for 940ms v=4000;>=0;+=2;#(60,0);v=2000;%(940,0,350,440) SIT Tone 913.8 hz for 274 ms with no wait, 1370.6 hz for 274 ms with no wait, 1776.7 hz for 380ms with no wait %(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7) ATTN TONE (phone's off the hook!) 1400+2060+2450+2600 hz for 100ms with 100ms wait %(100,100,1400,2060,2450,2600) git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3408 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-11-19 01:05:06 +00:00
if (*p == '$') {
Ringback (sponsored by Front Logic) This addition lets you set artifical ringback on a channel that is waiting for an originated call to be answered. the syntax is <action application="set" data="ringback=[data]"/> where data is either the full path to an audio file or a teletone generation script.. syntax of teletone scripts LEGEND: 0-9,a-d,*,# (standard dtmf tones) variables: c,r,d,v,>,<,+,w,l,L,% c (channels) - Sets the number of channels. r (rate) - Sets the sample rate. d (duration) - Sets the default tone duration. v (volume) - Sets the default volume. > (decrease vol) - factor to decrease volume by per frame (0 for even decrease across duration). < (increase vol) - factor to increase volume by per frame (0 for even increase across duration). + (step) - factor to step by used by < and >. w (wait) - default silence after each tone. l (loops) - number of times to repeat each tone in the script. L (LOOPS) - number of times to repeat the the whole script. % (manual tone) - a generic tone specified by a duration, a wait and a list of frequencies. standard tones can have custom duration per use with the () modifier 7(1000, 500) to generate DTMF 7 for 1 second then pause .5 seconds EXAMPLES UK Ring Tone [400+450 hz on for 400ms off for 200ms then 400+450 hz on for 400ms off for 2200ms] %(400,200,400,450);%(400,2200,400,450) US Ring Tone [440+480 hz on for 2000ms off for 4000ms] %(2000,4000,440,480) ATT BONG [volume level 4000, even decay, step by 2, # key for 60ms with no wait, volume level 2000, 350+440hz {us dialtone} for 940ms v=4000;>=0;+=2;#(60,0);v=2000;%(940,0,350,440) SIT Tone 913.8 hz for 274 ms with no wait, 1370.6 hz for 274 ms with no wait, 1776.7 hz for 380ms with no wait %(274,0,913.8);%(274,0,1370.6);%(380,0,1776.7) ATTN TONE (phone's off the hook!) 1400+2060+2450+2600 hz for 100ms with 100ms wait %(100,100,1400,2060,2450,2600) git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@3408 d0543943-73ff-0310-b7d9-9358b9ac24b2
2006-11-19 01:05:06 +00:00
p--;
} else {
*c++ = *p;
len++;
}
}
}
free(indup);
return data;
}
/* For Emacs:
* Local Variables:
* mode:c
* indent-tabs-mode:nil
* tab-width:4
* c-basic-offset:4
* End:
* For VIM:
* vim:set softtabstop=4 shiftwidth=4 tabstop=4 expandtab:
*/