87 lines
3.7 KiB
XML
87 lines
3.7 KiB
XML
<configuration name="sofia.conf" description="sofia Endpoint">
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<profiles>
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<profile name="$${sip_profile}">
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<!--aliases are other names that will work as a valid profile name for this profile-->
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<aliases>
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<alias name="default"/>
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</aliases>
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<!-- Outbound Registrations -->
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<gateways>
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<!--<gateway name="asterlink.com">-->
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<!--/// account username *required* ///-->
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<!--<param name="username" value="cluecon"/>-->
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<!--/// auth realm: *optional* same as gateway name, if blank ///-->
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<!--<param name="realm" value="asterlink.com"/>-->
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<!--/// account password *required* ///-->
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<!--<param name="password" value="2007"/>-->
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<!--/// replace the INVITE from user with the channel's caller-id ///-->
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<!--<param name="caller-id-in-from" value="false"/>-->
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<!--/// extension for inbound calls: *optional* same as username, if blank ///-->
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<!--<param name="extension" value="cluecon"/>-->
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<!--/// proxy host: *optional* same as realm, if blank ///-->
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<!--<param name="proxy" value="asterlink.com"/>-->
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<!--/// expire in seconds: *optional* 3600, if blank ///-->
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<!--<param name="expire-seconds" value="60"/>-->
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<!--/// do not register ///-->
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<!--<param name="register" value="false"/>-->
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<!--</gateway>-->
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</gateways>
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<domains>
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<!-- indicator to parse the directory for domains with parse="true" to get gateways-->
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<!--<domain name="$${domain}" parse="true"/>-->
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</domains>
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<settings>
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<param name="debug" value="1"/>
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<param name="rfc2833-pt" value="101"/>
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<param name="sip-port" value="5060"/>
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<param name="dialplan" value="XML"/>
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<param name="dtmf-duration" value="100"/>
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<param name="codec-prefs" value="$${global_codec_prefs}"/>
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<param name="codec-ms" value="20"/>
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<param name="use-rtp-timer" value="true"/>
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<param name="rtp-timer-name" value="soft"/>
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<param name="rtp-ip" value="$${bind_server_ip}"/>
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<param name="sip-ip" value="$${bind_server_ip}"/>
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<!-- if you want to send any special bind params of your own -->
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<!--<param name="bind-params" value="transport=udp"/>-->
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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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<!--If you have ODBC support and a working dsn you can use it instead of SQLite-->
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<!--<param name="odbc-dsn" value="dsn:user:pass"/>-->
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<!--Uncomment to set all inbound calls to no media mode-->
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<!--<param name="inbound-no-media" value="true"/>-->
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<!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok-->
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<!--<param name="inbound-late-negotiation" value="true"/>-->
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<!-- this lets anything register -->
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<!-- comment the next line and uncomment one or both of the other 2 lines for call authentication -->
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<param name="accept-blind-reg" value="true"/>
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<!--TTL for nonce in sip auth-->
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<param name="nonce-ttl" value="60"/>
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<!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec
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that the originator is using-->
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<!--<param name="disable-transcoding" value="true"/>-->
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<!--<param name="auth-calls" value="true"/>-->
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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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<!-- <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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<!-- <param name="vad" value="out"/> -->
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<!-- <param name="vad" value="both"/> -->
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<!--<param name="alias" value="sip:10.0.1.251:5555"/>-->
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</settings>
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</profile>
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</profiles>
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</configuration>
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