<configuration name="sofia.conf" description="sofia Endpoint"> <profiles> <profile name="$${domain}"> <registrations> <!-- <registration name="asterlink"> <param name="register-scheme" value="Digest"/> <param name="register-realm" value=""/> <param name="register-username" value="1001"/> <param name="register-password" value="nhy65tgb"/> <param name="register-from" value="sip:1001@208.64.200.40"/> <param name="register-to" value="sip:1001@conference.freeswitch.org"/> <param name="register-proxy" value="sip:conference.freeswitch.org:5060"/> <param name="register-frequency" value="20"/> </registration> --> </registrations> <settings> <param name="debug" value="1"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="enum,XML"/> <param name="dtmf-duration" value="100"/> <param name="codec-prefs" value="$${default_codecs}"/> <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="auto"/> <param name="sip-ip" value="auto"/> <!--Uncomment to set all inbound calls to no media mode--> <!--<param name="inbound-no-media" value="true"/>--> <!--Uncomment to let calls hit the dialplan *before* you decide if the codec is ok--> <!--<param name="inbound-late-negotiation" value="true"/>--> <!-- 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"/> <!--TTL for nonce in sip auth--> <param name="nonce-ttl" value="60"/> <!--Uncomment if you want to force the outbound leg of a bridge to only offer the codec that the originator is using--> <!--<param name="disable-transcoding" 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>