<configuration name="switch.conf" description="Modules">
<settings>
- <!--Most channels to allow at once -->
-
<param name="max-sessions" value="1000"/>
+ <!--Most channels to allow at once -->
+ <param name="max-sessions" value="1000"/>
</settings>
<!--Any variables defined here will be available in every channel, in the dialplan etc -->
<variables>
<configuration name="modules.conf" description="Modules">
<modules>
- <!-- Loggers (I'd load these first) -->
- <load module="mod_console"/>
- <!-- <load module="mod_syslog"/> -->
+ <!-- Loggers (I'd load these first) -->
+ <load module="mod_console"/>
+ <!-- <load module="mod_syslog"/> -->
- <!-- Multi-Faceted -->
+ <!-- Multi-Faceted -->
<!-- mod_enum is a dialplan interface, an application interface and an api command interface -->
- <load module="mod_enum"/>
-
- <!-- XML Interfaces -->
- <!-- <load module="mod_xml_rpc"/> -->
- <!-- <load module="mod_xml_curl"/> -->
-
- <!-- Event Handlers -->
- <!-- <load module="mod_cdr"/> -->
- <!-- <load module="mod_event_multicast"/> -->
- <!-- <load module="mod_event_socket"/> -->
- <!-- <load module="mod_xmpp_event"/> -->
- <!-- <load module="mod_zeroconf"/> -->
-
- <!-- Directory Interfaces -->
- <!-- <load module="mod_ldap"/> -->
-
- <!-- Endpoints -->
- <!-- <load module="mod_dingaling"/> -->
- <!--<load module="mod_iax"/>-->
- <load module="mod_portaudio"/>
- <load module="mod_sofia"/>
- <!-- <load module="mod_wanpipe"/> -->
- <!-- <load module="mod_woomera"/> -->
-
- <!-- Applications -->
- <load module="mod_bridgecall"/>
- <load module="mod_commands"/>
- <load module="mod_conference"/>
- <load module="mod_dptools"/>
- <load module="mod_echo"/>
- <!--<load module="mod_park"/>-->
- <load module="mod_playback"/>
-
- <!-- Dialplan Interfaces -->
- <!-- <load module="mod_dialplan_directory"/> -->
- <load module="mod_dialplan_xml"/>
-
- <!-- Codec Interfaces -->
- <load module="mod_g711"/>
- <load module="mod_gsm"/>
- <!-- <load module="mod_ilbc"/> -->
- <load module="mod_l16"/>
- <!-- <load module="mod_speex"/> -->
-
- <!-- File Format Interfaces -->
- <load module="mod_sndfile"/>
- <load module="mod_native_file"/>
-
- <!-- Timers -->
- <load module="mod_softtimer"/>
-
- <!-- Languages -->
- <!-- <load module="mod_spidermonkey"/> -->
- <!-- <load module="mod_perl"/> -->
-
- <!-- ASR /TTS -->
- <!-- <load module="mod_cepstral"/> -->
- <!-- <load module="mod_rss"/> -->
+ <load module="mod_enum"/>
+
+ <!-- XML Interfaces -->
+ <!-- <load module="mod_xml_rpc"/> -->
+ <!-- <load module="mod_xml_curl"/> -->
+
+ <!-- Event Handlers -->
+ <!-- <load module="mod_cdr"/> -->
+ <!-- <load module="mod_event_multicast"/> -->
+ <!-- <load module="mod_event_socket"/> -->
+ <!-- <load module="mod_xmpp_event"/> -->
+ <!-- <load module="mod_zeroconf"/> -->
+
+ <!-- Directory Interfaces -->
+ <!-- <load module="mod_ldap"/> -->
+
+ <!-- Endpoints -->
+ <!-- <load module="mod_dingaling"/> -->
+ <!--<load module="mod_iax"/>-->
+ <load module="mod_portaudio"/>
+ <load module="mod_sofia"/>
+ <!-- <load module="mod_wanpipe"/> -->
+ <!-- <load module="mod_woomera"/> -->
+
+ <!-- Applications -->
+ <load module="mod_bridgecall"/>
+ <load module="mod_commands"/>
+ <load module="mod_conference"/>
+ <load module="mod_dptools"/>
+ <load module="mod_echo"/>
+ <!--<load module="mod_park"/>-->
+ <load module="mod_playback"/>
+
+ <!-- Dialplan Interfaces -->
+ <!-- <load module="mod_dialplan_directory"/> -->
+ <load module="mod_dialplan_xml"/>
+
+ <!-- Codec Interfaces -->
+ <load module="mod_g711"/>
+ <load module="mod_gsm"/>
+ <!-- <load module="mod_ilbc"/> -->
+ <load module="mod_l16"/>
+ <!-- <load module="mod_speex"/> -->
+
+ <!-- File Format Interfaces -->
+ <load module="mod_sndfile"/>
+ <load module="mod_native_file"/>
+
+ <!-- Timers -->
+ <load module="mod_softtimer"/>
+
+ <!-- Languages -->
+ <!-- <load module="mod_spidermonkey"/> -->
+ <!-- <load module="mod_perl"/> -->
+
+ <!-- ASR /TTS -->
+ <!-- <load module="mod_cepstral"/> -->
+ <!-- <load module="mod_rss"/> -->
</modules>
</configuration>
<configuration name="event_multicast.conf" description="Multicast Event">
<settings>
-
<param name="address" value="225.1.1.1"/>
-
<param name="port" value="4242"/>
-
<param name="bindings" value="all"/>
+ <param name="address" value="225.1.1.1"/>
+ <param name="port" value="4242"/>
+ <param name="bindings" value="all"/>
</settings>
</configuration>
<configuration name="event_socket.conf" description="Socket Client">
<settings>
-
<param name="listen-ip" value="127.0.0.1"/>
-
<param name="listen-port" value="8021"/>
-
<param name="password" value="ClueCon"/>
+ <param name="listen-ip" value="127.0.0.1"/>
+ <param name="listen-port" value="8021"/>
+ <param name="password" value="ClueCon"/>
</settings>
</configuration>
<configuration name="iax.conf" description="IAX Configuration">
<settings>
-
<param name="debug" value="0"/>
-
<!-- <param name="ip" value="1.2.3.4"> -->
-
<param name="port" value="4569"/>
-
<param name="dialplan" value="XML"/>
-
<param name="codec-prefs" value="PCMU@20i,PCMA,speex,L16"/>
-
<param name="codec-master" value="us"/>
-
<param name="codec-rates" value="8"/>
+ <param name="debug" value="0"/>
+ <!-- <param name="ip" value="1.2.3.4"> -->
+ <param name="port" value="4569"/>
+ <param name="dialplan" value="XML"/>
+ <param name="codec-prefs" value="PCMU@20i,PCMA,speex,L16"/>
+ <param name="codec-master" value="us"/>
+ <param name="codec-rates" value="8"/>
</settings>
</configuration>
<!-- pick a file name, a function name or 'all' -->
<!-- map as many as you need for specific debugging -->
<mappings>
-
<!-- <param name="log_event" value="DEBUG"/> -->
-
<param name="all" value="DEBUG"/>
+ <!-- <param name="log_event" value="DEBUG"/> -->
+ <param name="all" value="DEBUG"/>
</mappings>
</configuration>
<configuration name="sofia.conf" description="sofia Endpoint">
<profiles>
-
<profile name="mydomain1.com">
- <registrations>
- <!-- <registration name="asterlink">
- <param name="register-scheme" value="Digest"/>
- <param name="register-realm" value=""/>
- <param name="register-username" value="1001"/>
- <param name="register-password" value="nhy65tgb"/>
- <param name="register-from" value="sip:1001@208.64.200.40"/>
- <param name="register-to" value="sip:1001@conference.freeswitch.org"/>
- <param name="register-proxy" value="sip:conference.freeswitch.org:5060"/>
- <param name="register-frequency" value="20"/>
- </registration> -->
- </registrations>
- <settings>
-
<param name="debug" value="1"/>
-
<param name="rfc2833-pt" value="101"/>
-
<param name="sip-port" value="5060"/>
-
<param name="dialplan" value="XML"/>
-
<param name="dtmf-duration" value="100"/>
-
<param name="codec-prefs" value="PCMU@20i"/>
-
<param name="codec-ms" value="20"/>
-
<param name="use-rtp-timer" value="true"/>
-
<param name="rtp-timer-name" value="soft"/>
-
<param name="rtp-ip" value="192.168.1.20"/>
-
<param name="sip-ip" value="mydomain1.com"/>
-
- <!-- this lets anything register -->
- <!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
-
<param name="accept-blind-reg" value="true"/>
-
-
<!--<param name="auth-calls" value="true"/>-->
- <!-- on authed calls, authenticate *all* the packets not just invite -->
-
<!--<param name="auth-all-packets" value="true"/>-->
-
- <!-- optional ; -->
-
<!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>-->
-
<!-- <param name="ext-rtp-ip" value="100.101.102.103"/> -->
- <!-- VAD choose one (out is a good choice); -->
-
<!-- <param name="vad" value="in"/> -->
-
<!-- <param name="vad" value="out"/> -->
-
<!-- <param name="vad" value="both"/> -->
-
<!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
- </settings>
- </profile>
+ <profile name="mydomain1.com">
+ <registrations>
+ <!-- <registration name="asterlink">
+
<param name="register-scheme" value="Digest"/>
+
<param name="register-realm" value=""/>
+
<param name="register-username" value="1001"/>
+
<param name="register-password" value="nhy65tgb"/>
+
<param name="register-from" value="sip:1001@208.64.200.40"/>
+
<param name="register-to" value="sip:1001@conference.freeswitch.org"/>
+
<param name="register-proxy" value="sip:conference.freeswitch.org:5060"/>
+
<param name="register-frequency" value="20"/>
+ </registration> -->
+ </registrations>
+ <settings>
+ <param name="debug" value="1"/>
+ <param name="rfc2833-pt" value="101"/>
+ <param name="sip-port" value="5060"/>
+ <param name="dialplan" value="XML"/>
+ <param name="dtmf-duration" value="100"/>
+ <param name="codec-prefs" value="PCMU@20i"/>
+ <param name="codec-ms" value="20"/>
+ <param name="use-rtp-timer" value="true"/>
+ <param name="rtp-timer-name" value="soft"/>
+ <param name="rtp-ip" value="192.168.1.20"/>
+ <param name="sip-ip" value="mydomain1.com"/>
+
+ <!-- this lets anything register -->
+ <!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
+ <param name="accept-blind-reg" value="true"/>
+
+ <!--<param name="auth-calls" value="true"/>-->
+ <!-- on authed calls, authenticate *all* the packets not just invite -->
+ <!--<param name="auth-all-packets" value="true"/>-->
+
+ <!-- optional ; -->
+ <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/>-->
+ <!-- <param name="ext-rtp-ip" value="100.101.102.103"/> -->
+ <!-- VAD choose one (out is a good choice); -->
+ <!-- <param name="vad" value="in"/> -->
+ <!-- <param name="vad" value="out"/> -->
+ <!-- <param name="vad" value="both"/> -->
+ <!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
+ </settings>
+ </profile>
</profiles>
</configuration>
<!-- info - informational message -->
<!-- debug - debug-level message -->
<settings>
-
<param name="ident" value="freeswitch"/>
-
<param name="facility" value="user"/>
-
<param name="format" value="${time} - ${message}"/>
-
<param name="level" value="debug,info,warning-alert"/>
+ <param name="ident" value="freeswitch"/>
+ <param name="facility" value="user"/>
+ <param name="format" value="${time} - ${message}"/>
+ <param name="level" value="debug,info,warning-alert"/>
</settings>
</configuration>
<configuration name="woomera.conf" description="Woomera Endpoint">
<settings>
-
<param name="debug" value="0"/>
+ <param name="debug" value="0"/>
</settings>
<interface>
-
<param name="host" value="localhost"/>
-
<param name="port" value="42420"/>
-
<param name="audio-ip" value="127.0.0.1"/>
-
<param name="dialplan" value="XML"/>
+ <param name="host" value="localhost"/>
+ <param name="port" value="42420"/>
+ <param name="audio-ip" value="127.0.0.1"/>
+ <param name="dialplan" value="XML"/>
</interface>
</configuration>
<configuration name="wanpipe.conf" description="Sangoma Wanpipe Endpoint">
<settings>
-
<param name="debug" value="1"/>
-
<param name="dialplan" value="XML"/>
-
<param name="mtu" value="320"/>
-
<param name="dtmf-on" value="800"/>
-
<param name="dtmf-off" value="100"/>
-
<param name="supress-dtmf-tone" value="yes"/>
+ <param name="debug" value="1"/>
+ <param name="dialplan" value="XML"/>
+ <param name="mtu" value="320"/>
+ <param name="dtmf-on" value="800"/>
+ <param name="dtmf-off" value="100"/>
+ <param name="supress-dtmf-tone" value="yes"/>
</settings>
<span>
-
<param name="span" value="1"/>
-
<param name="node" value="cpe"/>
-
<!-- <param name="switch" value="ni2"/> -->
-
<param name="switch" value="dms100"/>
-
<!-- <param name="switch" value="lucent5e"/> -->
-
<!-- <param name="switch" value="att4ess"/> -->
-
<!-- <param name="switch" value="euroisdn"/> -->
-
<!-- <param name="switch" value="gr303eoc"/> -->
-
<!-- <param name="switch" value="gr303tmc"/> -->
-
<param name="dp" value="national"/>
-
<!-- <param name="dp" value="international"/> -->
-
<!-- <param name="dp" value="local"/> -->
-
<!-- <param name="dp" value="private"/> -->
-
<!-- <param name="dp" value="unknown"/> -->
-
<param name="l1" value="ulaw"/>
-
<!-- <param name="l1" value="alaw"/> -->
-
<param name="bchan" value="1-23"/>
-
<param name="dchan" value="24"/>
-
<param name="dialplan" value="XML"/>
+ <param name="span" value="1"/>
+ <param name="node" value="cpe"/>
+ <!-- <param name="switch" value="ni2"/> -->
+ <param name="switch" value="dms100"/>
+ <!-- <param name="switch" value="lucent5e"/> -->
+ <!-- <param name="switch" value="att4ess"/> -->
+ <!-- <param name="switch" value="euroisdn"/> -->
+ <!-- <param name="switch" value="gr303eoc"/> -->
+ <!-- <param name="switch" value="gr303tmc"/> -->
+ <param name="dp" value="national"/>
+ <!-- <param name="dp" value="international"/> -->
+ <!-- <param name="dp" value="local"/> -->
+ <!-- <param name="dp" value="private"/> -->
+ <!-- <param name="dp" value="unknown"/> -->
+ <param name="l1" value="ulaw"/>
+ <!-- <param name="l1" value="alaw"/> -->
+ <param name="bchan" value="1-23"/>
+ <param name="dchan" value="24"/>
+ <param name="dialplan" value="XML"/>
</span>
</configuration>
<configuration name="portaudio.conf" description="Soundcard Endpoint">
<settings>
-
<param name="debug" value="2"/>
-
<param name="dialplan" value="XML"/>
+ <param name="debug" value="2"/>
+ <param name="dialplan" value="XML"/>
- <!-- partial string match on something in the name or the device # -->
-
<param name="indev" value="USB"/>
-
<param name="outdev" value="USB"/>
+ <!-- partial string match on something in the name or the device # -->
+ <param name="indev" value="USB"/>
+ <param name="outdev" value="USB"/>
-
<param name="cid-name" value="FreeSwitch"/>
-
<param name="cid-num" value="5555551212"/>
+ <param name="cid-name" value="FreeSwitch"/>
+ <param name="cid-num" value="5555551212"/>
</settings>
</configuration>
<configuration name="zeroconf.conf" description="Zeroconf Event Handler">
<settings>
-
<param name="publish" value="yes"/>
-
<param name="browse" value="_sip._udp"/>
+ <param name="publish" value="yes"/>
+ <param name="browse" value="_sip._udp"/>
</settings>
</configuration>
<configuration name="xmpp_event.conf" description="XMPP Event Handler">
<settings>
-
<param name="#debug" value="1"/>
-
<param name="jid" value="freeswitch@my.jabber.com/me"/>
-
<param name="passwd" value="mypass"/>
-
<param name="target-jid" value="freeswitch@reader.org/him"/>
+ <param name="#debug" value="1"/>
+ <param name="jid" value="freeswitch@my.jabber.com/me"/>
+ <param name="passwd" value="mypass"/>
+ <param name="target-jid" value="freeswitch@reader.org/him"/>
</settings>
</configuration>
<configuration name="dialplan_directory.conf" description="Dialplan Directory">
<settings>
-
<param name="directory-name" value="ldap"/>
-
<param name="host" value="ldap.freeswitch.org"/>
-
<param name="dn" value="cn=Manager,dc=freeswitch,dc=org"/>
-
<param name="pass" value="test"/>
-
<param name="base" value="dc=freeswitch,dc=org"/>
+ <param name="directory-name" value="ldap"/>
+ <param name="host" value="ldap.freeswitch.org"/>
+ <param name="dn" value="cn=Manager,dc=freeswitch,dc=org"/>
+ <param name="pass" value="test"/>
+ <param name="base" value="dc=freeswitch,dc=org"/>
</settings>
</configuration>
<configuration name="dingaling.conf" description="XMPP Jingle Endpoint">
<settings>
-
<param name="debug" value="0"/>
-
<param name="codec-prefs" value="PCMU"/>
+ <param name="debug" value="0"/>
+ <param name="codec-prefs" value="PCMU"/>
</settings>
<!-- *NOTE* change <x-profile></x-profile> to <profile></profile> to enable -->
<!-- Client Profile (Original mode) -->
<x-profile type="client">
-
<param name="name" value="mydomain.com"/>
-
<param name="login" value="myjid@myserver.com/talk"/>
-
<param name="password" value="mypass"/>
-
<param name="dialplan" value="XML"/>
-
<param name="message" value="Jingle all the way"/>
-
<param name="rtp-ip" value="10.0.0.1"/>
-
<param name="auto-login" value="true"/>
+ <param name="name" value="mydomain.com"/>
+ <param name="login" value="myjid@myserver.com/talk"/>
+ <param name="password" value="mypass"/>
+ <param name="dialplan" value="XML"/>
+ <param name="message" value="Jingle all the way"/>
+ <param name="rtp-ip" value="10.0.0.1"/>
+ <param name="auto-login" value="true"/>
<param name="auto-reply" value="Press *Call* to call FreeSWITCH and be sure to come to ClueCon! http://www.cluecon.com"/>
- <!-- SASL "plain" or "md5" -->
-
<param name="sasl" value="plain"/>
- <!-- if the server where the jabber is hosted is not the same as the one in the jid -->
-
<!--<param name="server" value="alternate.server.com"/>-->
- <!-- Enable TLS or not -->
-
<param name="tls" value="true"/>
- <!-- disable to trade async for more calls -->
-
<param name="use-rtp-timer" value="true"/>
- <!-- or -->
-
<!-- <param name="rtp-ip" value="my_lan_ip"/> -->
-
<!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/> -->
- <!-- default extension (if one cannot be determined) -->
-
<param name="exten" value="888"/>
- <!-- VAD choose one -->
-
<!-- <param name="vad" value="in"/> -->
-
<!-- <param name="vad" value="out"/> -->
-
<param name="vad" value="both"/>
+ <!-- SASL "plain" or "md5" -->
+ <param name="sasl" value="plain"/>
+ <!-- if the server where the jabber is hosted is not the same as the one in the jid -->
+ <!--<param name="server" value="alternate.server.com"/>-->
+ <!-- Enable TLS or not -->
+ <param name="tls" value="true"/>
+ <!-- disable to trade async for more calls -->
+ <param name="use-rtp-timer" value="true"/>
+ <!-- or -->
+ <!-- <param name="rtp-ip" value="my_lan_ip"/> -->
+ <!-- <param name="ext-rtp-ip" value="stun:stun.server.com"/> -->
+ <!-- default extension (if one cannot be determined) -->
+ <param name="exten" value="888"/>
+ <!-- VAD choose one -->
+ <!-- <param name="vad" value="in"/> -->
+ <!-- <param name="vad" value="out"/> -->
+ <param name="vad" value="both"/>
</x-profile>
<!-- Component (Server to Server Login) -->
<x-profile type="component">
<!-- All traffic for *@sub.mydomain.com will come to you -->
<param name="name" value="sub.mydomain.com"/>
-
<param name="password" value="secret"/>
+ <param name="password" value="secret"/>
<param name="dialplan" value="XML"/>
<param name="rtp-ip" value="208.64.200.42"/>
<param name="server" value="jabber.server.org:5347"/>
- <!-- disable to trade async for more calls -->
-
<param name="use-rtp-timer" value="true"/>
+ <!-- disable to trade async for more calls -->
+ <param name="use-rtp-timer" value="true"/>
<!-- "_auto_" means the extension will be automaticly set to the called jid -->
-
<param name="exten" value="_auto_"/>
-
<!--<param name="vad" value="both"/>-->
+ <param name="exten" value="_auto_"/>
+ <!--<param name="vad" value="both"/>-->
</x-profile>
</configuration>
<configuration name="xml_rpc.conf" description="XML RPC">
<settings>
- <!-- The port where you want to run the http service (default 8080) -->
-
<param name="http-port" value="8080"/>
- <!-- if all 3 of the following params exist all http traffic will require auth -->
-
<param name="auth-realm" value="freeswitch"/>
-
<param name="auth-user" value="freeswitch"/>
-
<param name="auth-pass" value="works"/>
+ <!-- The port where you want to run the http service (default 8080) -->
+ <param name="http-port" value="8080"/>
+ <!-- if all 3 of the following params exist all http traffic will require auth -->
+ <param name="auth-realm" value="freeswitch"/>
+ <param name="auth-user" value="freeswitch"/>
+ <param name="auth-pass" value="works"/>
</settings>
</configuration>
<configuration name="rss.conf" description="RSS Parser">
<feeds>
- <!-- Just download the files to wherever and refer to them here -->
- <!-- <feed name="Slash Dot">/home/rss/rss.rss</feed> -->
- <!-- <feed name="News Forge">/home/rss/newsforge.rss</feed> -->
+ <!-- Just download the files to wherever and refer to them here -->
+ <!-- <feed name="Slash Dot">/home/rss/rss.rss</feed> -->
+ <!-- <feed name="News Forge">/home/rss/newsforge.rss</feed> -->
</feeds>
</configuration>
<!-- Profiles are collections of settings you can reference by name. -->
<profiles>
-
<profile name="default">
- <!-- Sample Rate-->
-
<param name="rate" value="8000"/>
- <!-- Number of milliseconds per frame -->
-
<param name="interval" value="20"/>
- <!-- Energy level required for audio to be sent to the other users -->
-
<param name="energy-level" value="300"/>
- <!-- TTS Engine to use -->
-
<!--<param name="tts-engine" value="cepstral"/>-->
- <!-- TTS Voice to use -->
-
<!--<param name="tts-voice" value="david"/>-->
-
- <!-- If TTS is enabled all audio-file params not beginning with -->
- <!-- '/' or with drive: (i.e. c:) will be considered text to say with TTS -->
-
- <!-- File to play to acknowledge succees -->
-
<!--<param name="ack-sound" value="/soundfiles/beep.wav"/>-->
- <!-- File to play to acknowledge failure -->
-
<!--<param name="nack-sound" value="/soundfiles/beeperr.wav"/>-->
- <!-- File to play to acknowledge muted -->
-
<!--<param name="muted-sound" value="/soundfiles/muted.wav"/>-->
- <!-- File to play to acknowledge unmuted -->
-
<!--<param name="unmuted-sound" value="/soundfiles/unmuted.wav"/>-->
- <!-- File to play if you are alone in the conference -->
-
<!--<param name="alone-sound" value="/soundfiles/yactopitc.wav"/>-->
- <!-- File to play when you join the conference -->
-
<!--<param name="enter-sound" value="/soundfiles/welcome.wav"/>-->
- <!-- File to play when you leave the conference -->
-
<!--<param name="exit-sound" value="/soundfiles/exit.wav"/>-->
- <!-- File to play when you ae ejected from the conference -->
-
<!--<param name="kicked-sound" value="/soundfiles/kicked.wav"/>-->
- <!-- File to play when the conference is locked -->
-
<!--<param name="locked-sound" value="/soundfiles/locked.wav"/>-->
- <!-- File to play to prompt for a pin -->
-
<!--<param name="pin-sound" value="/soundfiles/pin.wav"/>-->
- <!-- File to play to when the pin is invalid -->
-
<!--<param name="bad-pin-sound" value="/soundfiles/invalid-pin.wav"/>-->
- <!-- Conference pin -->
-
<!--<param name="pin" value="12345"/>-->
- <!-- Default Caller ID Name for outbound calls -->
-
<param name="caller-id-name" value="FreeSWITCH"/>
- <!-- Default Caller ID Number for outbound calls -->
-
<param name="caller-id-number" value="8777423583"/>
- </profile>
+ <profile name="default">
+ <!-- Sample Rate-->
+ <param name="rate" value="8000"/>
+ <!-- Number of milliseconds per frame -->
+ <param name="interval" value="20"/>
+ <!-- Energy level required for audio to be sent to the other users -->
+ <param name="energy-level" value="300"/>
+ <!-- TTS Engine to use -->
+ <!--<param name="tts-engine" value="cepstral"/>-->
+ <!-- TTS Voice to use -->
+ <!--<param name="tts-voice" value="david"/>-->
+
+ <!-- If TTS is enabled all audio-file params not beginning with -->
+ <!-- '/' or with drive: (i.e. c:) will be considered text to say with TTS -->
+
+ <!-- File to play to acknowledge succees -->
+ <!--<param name="ack-sound" value="/soundfiles/beep.wav"/>-->
+ <!-- File to play to acknowledge failure -->
+ <!--<param name="nack-sound" value="/soundfiles/beeperr.wav"/>-->
+ <!-- File to play to acknowledge muted -->
+ <!--<param name="muted-sound" value="/soundfiles/muted.wav"/>-->
+ <!-- File to play to acknowledge unmuted -->
+ <!--<param name="unmuted-sound" value="/soundfiles/unmuted.wav"/>-->
+ <!-- File to play if you are alone in the conference -->
+ <!--<param name="alone-sound" value="/soundfiles/yactopitc.wav"/>-->
+ <!-- File to play when you join the conference -->
+ <!--<param name="enter-sound" value="/soundfiles/welcome.wav"/>-->
+ <!-- File to play when you leave the conference -->
+ <!--<param name="exit-sound" value="/soundfiles/exit.wav"/>-->
+ <!-- File to play when you ae ejected from the conference -->
+ <!--<param name="kicked-sound" value="/soundfiles/kicked.wav"/>-->
+ <!-- File to play when the conference is locked -->
+ <!--<param name="locked-sound" value="/soundfiles/locked.wav"/>-->
+ <!-- File to play to prompt for a pin -->
+ <!--<param name="pin-sound" value="/soundfiles/pin.wav"/>-->
+ <!-- File to play to when the pin is invalid -->
+ <!--<param name="bad-pin-sound" value="/soundfiles/invalid-pin.wav"/>-->
+ <!-- Conference pin -->
+ <!--<param name="pin" value="12345"/>-->
+ <!-- Default Caller ID Name for outbound calls -->
+ <param name="caller-id-name" value="FreeSWITCH"/>
+ <!-- Default Caller ID Number for outbound calls -->
+ <param name="caller-id-number" value="8777423583"/>
+ </profile>
</profiles>
</configuration>
<configuration name="enum.conf" description="ENUM Module">
<settings>
-
<param name="default-root" value="e164.org"/>
+ <param name="default-root" value="e164.org"/>
</settings>
<routes>
- <route service="E2U+SIP" regex="sip:(.*)" replace="sofia/test/$1"/>
- <route service="E2U+IAX2" regex="iax2:(.*)" replace="iax/$1"/>
- <route service="E2U+XMPP" regex="XMPP:(.*)" replace="dingaling/jingle/$1"/>
+ <route service="E2U+SIP" regex="sip:(.*)" replace="sofia/test/$1"/>
+ <route service="E2U+IAX2" regex="iax2:(.*)" replace="iax/$1"/>
+ <route service="E2U+XMPP" regex="XMPP:(.*)" replace="dingaling/jingle/$1"/>
</routes>
</configuration>
<configuration name="ivr.conf" description="IVR menus">
- <menus>
- <menu name="main" greet-long="/soundfiles/greet-long.wav" greet-short="/soundfiles/greet-short.wav" invalid-sound="/soundfiles/invalid.wav" exit-sound="/soundfiles/exit.wav" timeout ="15" max-failures="3">
- <entry action="menu-exit" digits="*"/>
- <entry action="menu-sub" digits="2" param="menu2"/>
- <entry action="menu-exec-api" digits="3" param="api arg"/>
- <entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
- <entry action="menu-back" digits="5"/>
- <entry action="menu-call-transfer" digits="7" param="888"/>
- <entry action="menu-sub" digits="8" param="menu8"/>>
- </menu>
- <menu name="menu8" greet-long="/soundfiles/greet-long.wav" greet-short="/soundfiles/greet-short.wav" invalid-sound="/soundfiles/invalid.wav" exit-sound="/soundfiles/exit.wav" timeout ="15" max-failures="3">
- <entry action="menu-back" digits="#"/>
- <entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
- <entry action="menu-top" digits="*"/>
- </menu>
- <menu name="menu2" greet-long="/soundfiles/greet-long.wav" greet-short="/soundfiles/greet-short.wav" invalid-sound="/soundfiles/invalid.wav" exit-sound="/soundfiles/exit.wav" timeout ="15" max-failures="3">
- <entry action="menu-back" digits="#"/>
- <entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
- <entry action="menu-top" digits="*"/>
- </menu>
- </menus>
+ <menus>
+ <menu name="main"
+ greet-long="/soundfiles/greet-long.wav"
+ greet-short="/soundfiles/greet-short.wav"
+ invalid-sound="/soundfiles/invalid.wav"
+ exit-sound="/soundfiles/exit.wav" timeout ="15" max-failures="3">
+ <entry action="menu-exit" digits="*"/>
+ <entry action="menu-sub" digits="2" param="menu2"/>
+ <entry action="menu-exec-api" digits="3" param="api arg"/>
+ <entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
+ <entry action="menu-back" digits="5"/>
+ <entry action="menu-call-transfer" digits="7" param="888"/>
+ <entry action="menu-sub" digits="8" param="menu8"/>>
+ </menu>
+ <menu name="menu8"
+ greet-long="/soundfiles/greet-long.wav"
+ greet-short="/soundfiles/greet-short.wav"
+ invalid-sound="/soundfiles/invalid.wav"
+ exit-sound="/soundfiles/exit.wav"
+ timeout ="15"
+ max-failures="3">
+ <entry action="menu-back" digits="#"/>
+ <entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
+ <entry action="menu-top" digits="*"/>
+ </menu>
+ <menu name="menu2"
+ greet-long="/soundfiles/greet-long.wav"
+ greet-short="/soundfiles/greet-short.wav"
+ invalid-sound="/soundfiles/invalid.wav"
+ exit-sound="/soundfiles/exit.wav"
+ timeout ="15"
+ max-failures="3">
+ <entry action="menu-back" digits="#"/>
+ <entry action="menu-play-sound" digits="4" param="/soundfiles/4.wav"/>
+ <entry action="menu-top" digits="*"/>
+ </menu>
+ </menus>
</configuration>
</section>
<!-- *NOTE* The special context name 'any' will match any context -->
<context name="default">
<extension name="tollfree">
- <condition field="destination_number" expression="^(18(0{2}|8{2}|7{2}|6{2})\d{7})$">
+ <condition field="destination_number" expression="^(18(0{2}|8{2}|7{2}|6{2})\d{7})$">
<action application="enum" data="$1"/>
<action application="bridge" data="${enum_auto_route}"/>
- </condition>
+ </condition>
</extension>
<!-- Call the FreeSWITCH conference via SIP -->
- <!--<extension name="FreeSWITCH Conference SIP">-->
- <!--<condition field="destination_number" expression="^888$">-->
-
<!--<action application="bridge" data="sofia/test/888@conference.freeswitch.org"/>-->
- <!--</condition>-->
- <!--</extension> -->
+ <!--<extension name="FreeSWITCH Conference SIP">-->
+ <!--<condition field="destination_number" expression="^888$">-->
+ <!--<action application="bridge" data="sofia/test/888@conference.freeswitch.org"/>-->
+ <!--</condition>-->
+ <!--</extension> -->
<!-- Call the FreeSWITCH conference via IAX -->
- <!--<extension name="FreeSWITCH Conference IAX">-->
- <!--<condition field="destination_number" expression="^8888$">-->
-
<!--<action application="bridge" data="iax/guest@conference.freeswitch.org/888"/>-->
- <!--</condition>-->
- <!--</extension>-->
+ <!--<extension name="FreeSWITCH Conference IAX">-->
+ <!--<condition field="destination_number" expression="^8888$">-->
+ <!--<action application="bridge" data="iax/guest@conference.freeswitch.org/888"/>-->
+ <!--</condition>-->
+ <!--</extension>-->
<extension name="testmusic">
- <condition field="destination_number" expression="^1234$">
+ <condition field="destination_number" expression="^1234$">
<!-- Request a certain tone/file to be played while you wait for the call to be answered-->
<action application="set" data="ringback=${us-ring}"/>
<!--<action application="set" data="ringback=/home/ring.wav"/>-->
-
<action application="bridge" data="sofia/test/1234@conference.freeswitch.org"/>
- </condition>
+ <action application="bridge" data="sofia/test/1234@conference.freeswitch.org"/>
+ </condition>
</extension>
<!-- Enter an existing conference -->
<extension name="1000">
- <condition field="destination_number" expression="^1000$">
-
<action application="conference" data="freeswitch"/>
- </condition>
+ <condition field="destination_number" expression="^1000$">
+ <action application="conference" data="freeswitch"/>
+ </condition>
</extension>
<!-- Start a dynamic conference and call someone at the same time -->
<extension name="2000">
- <condition field="destination_number" expression="^2000$">
-
<action application="conference" data="bridge:mydynaconf:sofia/test/1234@conference.freeswitch.org"/>
- </condition>
+ <condition field="destination_number" expression="^2000$">
+ <action application="conference" data="bridge:mydynaconf:sofia/test/1234@conference.freeswitch.org"/>
+ </condition>
</extension>
<!-- extensions starting with 4, all the numbers after 4 form a numeric filename -->
<!-- the desired file. This is for demo purposes both actions -->
<!-- could have been under the same <extension> tag as well. -->
<extension name="playsound1" continue="true">
- <condition field="source" expression="mod_sofia"/>
- <condition field="destination_number" expression="^4(\d+)">
-
<action application="playback" data="/var/sounds/beep.gsm"/>
- </condition>
+ <condition field="source" expression="mod_sofia"/>
+ <condition field="destination_number" expression="^4(\d+)">
+ <action application="playback" data="/var/sounds/beep.gsm"/>
+ </condition>
</extension>
<extension name="playsound2">
- <condition field="source" expression="mod_sofia"/>
- <condition field="destination_number" expression="^4(\d+)">
-
<action application="playback" data="/root/$1.raw"/>
- </condition>
+ <condition field="source" expression="mod_sofia"/>
+ <condition field="destination_number" expression="^4(\d+)">
+ <action application="playback" data="/root/$1.raw"/>
+ </condition>
</extension>
<!-- send everything with a certian RDNIS to Wanpipe ISDN -->
<extension name="To PRI">
- <condition field="rdnis" expression="8881231234"/>
- <condition field="destination_number" expression="(.*)">
-
<action application="bridge" data="wanpipe/a/a/$1"/>
- </condition>
+ <condition field="rdnis" expression="8881231234"/>
+ <condition field="destination_number" expression="(.*)">
+ <action application="bridge" data="wanpipe/a/a/$1"/>
+ </condition>
</extension>
<!-- Call *MUST* originate from mod_iax and also be dialing ext 9999-->
<extension name="9999">
- <condition field="source" expression="mod_iax"/>
- <condition field="destination_number" expression="9999">
-
<action application="playback" data="/var/sounds/beep.gsm"/>
- </condition>
+ <condition field="source" expression="mod_iax"/>
+ <condition field="destination_number" expression="9999">
+ <action application="playback" data="/var/sounds/beep.gsm"/>
+ </condition>
</extension>
</context>
projects / thirdparty / freeswitch.git / commitdiff
