updates, cleanups, and clarifications to the sample conf files. Uses more of the preprocessor variables for simplicity. FSBUILD-2 from Mark D. Anderson. Thanks Mark.
git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@5034 d0543943-73ff-0310-b7d9-9358b9ac24b2
This commit is contained in:
Michael Jerris
parent
7db63e84db
commit
48ca4cc864
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@ -1,8 +1,8 @@
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<!-- None of these paths are real if you want any of these options you need to really set them up -->
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<configuration name="conference.conf" description="Audio Conference">
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<!-- Advertise certian presence on startup . -->
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<!-- Advertise certain presence on startup . -->
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<advertise>
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<room name="888@sub.mydomain.com" status="FreeSWITCH"/>
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<room name="888@$${server_name}" status="FreeSWITCH"/>
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</advertise>
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<!-- These are the default keys that map when you do not specify a caller control group -->
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@ -29,7 +29,7 @@
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<!--If no profile is specified it will default to "default"-->
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<profile name="default">
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<!-- Domain (for presence) -->
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<param name="domain" value="sub.mydomain.com"/>
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<param name="domain" value="$${server_name}"/>
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<!-- Sample Rate-->
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<param name="rate" value="8000"/>
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<!-- Number of milliseconds per frame -->
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@ -81,9 +81,9 @@
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<!-- Conference pin -->
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<!--<param name="pin" value="12345"/>-->
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<!-- Default Caller ID Name for outbound calls -->
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<param name="caller-id-name" value="FreeSWITCH"/>
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<param name="caller-id-name" value="$${outbound_caller_name}"/>
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<!-- Default Caller ID Number for outbound calls -->
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<param name="caller-id-number" value="8777423583"/>
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<param name="caller-id-number" value="$${outbound_caller_id}"/>
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</profile>
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</profiles>
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</configuration>
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@ -29,7 +29,7 @@
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<!-- Call the FreeSWITCH conference via SIP -->
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<!--<extension name="FreeSWITCH Conference SIP">-->
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<!--<condition field="destination_number" expression="^888$">-->
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<!--<action application="bridge" data="sofia/$${domain}/888@conference.freeswitch.org"/>-->
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<!--<action application="bridge" data="sofia/$${sip_profile}/888@conference.freeswitch.org"/>-->
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<!--</condition>-->
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<!--</extension> -->
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@ -53,7 +53,7 @@
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<!-- Request a certain tone/file to be played while you wait for the call to be answered-->
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<action application="set" data="ringback=${us-ring}"/>
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<!--<action application="set" data="ringback=/home/ring.wav"/>-->
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<action application="bridge" data="sofia/$${domain}/1234@conference.freeswitch.org"/>
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<action application="bridge" data="sofia/$${sip_profile}/1234@conference.freeswitch.org"/>
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</condition>
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</extension>
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@ -67,7 +67,7 @@
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<!-- Start a dynamic conference and call someone at the same time -->
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<extension name="2000">
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<condition field="destination_number" expression="^2000$">
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<action application="conference" data="bridge:mydynaconf:sofia/$${domain}/1234@conference.freeswitch.org"/>
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<action application="conference" data="bridge:mydynaconf:sofia/$${sip_profile}/1234@conference.freeswitch.org"/>
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</condition>
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</extension>
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@ -8,12 +8,13 @@
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<!-- Client Profile (Original mode) -->
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<x-profile type="client">
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<param name="name" value="$${domain}"/>
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<param name="name" value="$${xmpp_client_profile}"/>
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<param name="login" value="myjid@myserver.com/talk"/>
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<param name="password" value="mypass"/>
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<param name="dialplan" value="XML"/>
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<param name="message" value="Jingle all the way"/>
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<param name="rtp-ip" value="auto"/>
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<param name="rtp-ip" value="$${bind_server_ip}"/>
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<!-- <param name="ext-rtp-ip" value="$${external_rtp_ip"/> -->
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<param name="auto-login" value="true"/>
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<!-- SASL "plain" or "md5" -->
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<param name="sasl" value="plain"/>
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@ -23,9 +24,6 @@
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<param name="tls" value="true"/>
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<!-- disable to trade async for more calls -->
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<param name="use-rtp-timer" value="true"/>
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<!-- or -->
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<!-- <param name="rtp-ip" value="auto"/> -->
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<!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/> -->
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<!-- default extension (if one cannot be determined) -->
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<param name="exten" value="888"/>
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<!-- VAD choose one -->
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@ -37,11 +35,10 @@
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<!-- Component (Server to Server Login) -->
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<x-profile type="component">
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<!-- All traffic for *@sub.mydomain.com will come to you -->
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<param name="name" value="$${subdomain}"/>
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<param name="name" value="$${xmpp_server_profile}"/>
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<param name="password" value="secret"/>
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<param name="dialplan" value="XML"/>
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<param name="rtp-ip" value="auto"/>
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<param name="rtp-ip" value="$${bind_server_ip}"/>
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<param name="server" value="jabber.server.org:5347"/>
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<!-- disable to trade async for more calls -->
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<param name="use-rtp-timer" value="true"/>
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@ -4,8 +4,8 @@
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</settings>
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<routes>
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<route service="E2U+SIP" regex="sip:(.*)" replace="sofia/$${domain}/$1"/>
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<route service="E2U+SIP" regex="sip:(.*)" replace="sofia/$${sip_profile}/$1"/>
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<route service="E2U+IAX2" regex="iax2:(.*)" replace="iax/$1"/>
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<route service="E2U+XMPP" regex="XMPP:(.*)" replace="dingaling/$${domain}/$1"/>
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<route service="E2U+XMPP" regex="XMPP:(.*)" replace="dingaling/$${xmpp_server_profile}/$1"/>
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</routes>
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</configuration>
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@ -13,34 +13,115 @@
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Don't modify it while freeswitch is running cos it is mem mapped in most cases =D
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-->
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<!--#set "domain=mydomain.com"-->
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<!--#set "subdomain=sub.mydomain.com"-->
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<!--#set "default_codecs=PCMU@20i"-->
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<!--my domain is $${domain}-->
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<!-- Preprocessor Variables
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These are introduced when configuration strings must be consistent across modules.
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-->
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<!-- sip_profile
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Must be a domain name if you are being a registry server; otherwise
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can be any string.
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used by: sofia.conf.xml enum.conf.xml default_context.xml directory.xml
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-->
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<!--#set "sip_profile=mydomain.com"-->
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<!-- xmpp_client_profile and xmpp_server_profile
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xmpp_client_profile can be any string.
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xmpp_server_profile is appended to "dingaling_" to form the database name
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containing the "subscriptions" table.
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used by: dingaling.conf.xml enum.conf.xml
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-->
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<!--#set "xmpp_client_profile=xmppc"-->
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<!--#set "xmpp_server_profile=xmpps"-->
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<!-- bind_server_ip
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Can be an ip address, a dns name, or "auto".
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This determines an ip address available on this host to bind.
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If you are separating RTP and SIP traffic, you will want to have
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use different addresses where this variable appears.
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Used by: sofia.conf.xml dingaling.conf.xml
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-->
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<!--#set "bind_server_ip=auto"-->
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<!-- external_rtp_ip
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Used as the public IP address for SDP.
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Can be an ip address or a string like "stun:stun.server.com"
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If unspecified, the bind_server_ip value is used.
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Used by: sofia.conf.xml dingaling.conf.xml
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-->
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<!--#set "external_rtp_ip=stun:stun.server.com"-->
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<!-- server_name
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A public ip address or DNS name that is used when advertising conference
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presence or registering sip.
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Used by: conference.conf.xml
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-->
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<!-- outbound_caller_id and outbound_caller_name
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The caller ID telephone number we should use when calling out.
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Used by: conference.conf.xml
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-->
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<!--#set "outbound_caller_name=FreeSWITCH"-->
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<!--#set "outbound_caller_id=8777423583"-->
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<section name="configuration" description="Various Configuration">
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<!--#include "switch.conf.xml"-->
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<!--#include "modules.conf.xml"-->
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<!--#include "spidermonkey.conf.xml"-->
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<!-- Order doesn't matter, but for clarity these are in same order as modules.conf.xml.
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If they aren't loaded by modules.conf.xml, then they are ignored.
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-->
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<!-- Loggers -->
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<!--#include "console.conf.xml"-->
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<!--#include "syslog.conf.xml"-->
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<!-- Multi-Faceted -->
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<!--#include "enum.conf.xml"-->
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<!-- XML Interfaces -->
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<!--#include "xml_rpc.conf.xml"-->
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<!--#include "xml_curl.conf.xml"-->
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<!-- none for mod_xml_cdr -->
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<!-- Event Handlers -->
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<!--#include "cdr.conf.xml"-->
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<!--#include "event_multicast.conf.xml"-->
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<!--#include "event_socket.conf.xml"-->
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<!--#include "xmpp_event.conf.xml"-->
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<!--#include "zeroconf.conf.xml"-->
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<!-- Directory Interfaces -->
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<!-- none for mod_ldap; dialplan_directory.conf.xml has ldap connection info -->
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<!-- Endpoints -->
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<!--#include "dingaling.conf.xml"-->
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<!--#include "iax.conf.xml"-->
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<!--#include "console.conf.xml"-->
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<!--#include "sofia.conf.xml"-->
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<!--#include "syslog.conf.xml"-->
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<!--#include "woomera.conf.xml"-->
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<!--#include "wanpipe.conf.xml"-->
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<!--#include "portaudio.conf.xml"-->
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<!--#include "alsa.conf.xml"-->
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<!--#include "zeroconf.conf.xml"-->
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<!--#include "xmpp_event.conf.xml"-->
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<!--#include "dialplan_directory.conf.xml"-->
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<!--#include "dingaling.conf.xml"-->
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<!--#include "xml_curl.conf.xml"-->
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<!--#include "xml_rpc.conf.xml"-->
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<!--#include "rss.conf.xml"-->
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<!--#include "sofia.conf.xml"-->
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<!--#include "wanpipe.conf.xml"-->
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<!--#include "woomera.conf.xml"-->
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<!-- Applications -->
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<!-- none for mod_bridgecall, mod_commands, mod_echo, mod_park, mod_playback -->
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<!--#include "conference.conf.xml"-->
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<!--#include "enum.conf.xml"-->
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<!-- ivr.conf is used by mod_dptools -->
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<!--#include "ivr.conf.xml"-->
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<!-- Dialplan Interfaces -->
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<!--#include "dialplan_directory.conf.xml"-->
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<!-- mod_dialplan_xml is configured in the separate "dialplan" section. -->
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<!-- Codec Interfaces -->
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<!-- no configuration needed -->
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<!-- File Format Interfaces -->
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<!-- no configuration needed -->
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<!-- Timers -->
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<!-- no configuration needed -->
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<!-- Languages -->
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<!--#include "spidermonkey.conf.xml"-->
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<!-- none for mod_perl -->
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<!-- ASR /TTS -->
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<!-- none for mod_cepstral -->
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<!--#include "rss.conf.xml"-->
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<!-- Say -->
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<!-- none for mod_say_en -->
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<!--#include "mod_cdr.conf.xml"-->
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</section>
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@ -48,6 +129,8 @@
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<!--#include "default_context.xml"-->
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</section>
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<!-- mod_dingaling is reliant on the vcard data in the "directory" section. -->
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<!-- mod_sofia is reliant on the user data for authorization -->
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<section name="directory" description="User Directory">
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<!--#include "directory.xml"-->
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</section>
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@ -66,3 +149,4 @@
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</document>
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@ -1,6 +1,6 @@
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<configuration name="sofia.conf" description="sofia Endpoint">
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<profiles>
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<profile name="$${domain}">
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<profile name="$${sip_profile}">
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<!-- Outbound Registrations -->
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<gateways>
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<!--<gateway name="asterlink.com">-->
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@ -28,12 +28,12 @@
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<param name="sip-port" value="5060"/>
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<param name="dialplan" value="enum,XML"/>
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<param name="dtmf-duration" value="100"/>
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<param name="codec-prefs" value="$${default_codecs}"/>
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<param name="codec-prefs" value="PCMU@20i"/>
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<param name="codec-ms" value="20"/>
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<param name="use-rtp-timer" value="true"/>
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<param name="rtp-timer-name" value="soft"/>
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<param name="rtp-ip" value="auto"/>
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<param name="sip-ip" value="auto"/>
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<param name="rtp-ip" value="$${bind_server_ip}"/>
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<param name="sip-ip" value="$${bind_server_ip}"/>
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<!--If you don't want to pass through timestampes from 1 RTP call to another (on a per call basis with rtp_rewrite_timestamps chanvar)-->
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<!--<param name="rtp-rewrite-timestampes" value="true"/>-->
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@ -60,9 +60,8 @@
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<!-- on authed calls, authenticate *all* the packets not just invite -->
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<!--<param name="auth-all-packets" value="true"/>-->
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<!-- optional ; -->
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<!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>-->
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<!-- <param name="ext-rtp-ip" value="100.101.102.103"/> -->
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<!-- <param name="ext-rtp-ip" value="$${external_rtp_ip}"/>-->
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<!-- <param name="ext-sip-ip" value="100.101.102.103"/> -->
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<!-- VAD choose one (out is a good choice); -->
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<!-- <param name="vad" value="in"/> -->
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@ -0,0 +1,49 @@
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#!/usr/bin/perl -w
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use strict;
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=head1 NAME
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combineconf.pl - expand #include PIs in a freeswitch conf file
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=head1 SYNOPSIS
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# cd conf
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# ../scripts/combineconf.pl freeswitch.xml > freeswitch_combined.xml
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=head1 DESCRIPTION
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This is recursive, and will take multiple input files on the command line.
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You need to run it from the working directory that the relative include paths
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except to be resolved from.
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=head1 AUTHOR
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Mark D. Anderson (mda@discerning.com)
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Released under same terms as Perl, or alternatively the MPL.
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=cut
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use IO::File;
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sub filter_file {
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my ($f) = @_;
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my $fh = $f eq '-' ? \*STDIN : IO::File->new($f, 'r');
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die "ERROR: Can't open $f: $!\n" unless $fh;
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while(<$fh>) {
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if (m/<!--#include\s+"(.*?)"/) {
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filter_file($1);
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}
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else {print;}
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}
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undef $fh;
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}
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sub main {
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die "Usage: $0 file1 ...\nCombined output goes to stdout. Use '-' as the filename to use stdin." unless @ARGV;
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for(@ARGV) {
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filter_file($_);
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}
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}
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main();
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