Hi everyone! I've been working on setting up an Asterisk server and my two Digium cards I ordered will arrive tommorow, so I'm excited to plug some 'real' old-school lines into it.
But tonight I've been testing with some of our staff around the world, and while handing off 'real' (PSTN - over VoIP using Voicepulse) calls to SIP and SIP calls to VoIP PSTN works fine, SIP to SIP calls provide no audio, and just this message on console: -- Executing NoOp("SIP/322-3edd", ""call for "331") in new stack -- Executing Dial("SIP/322-3edd", "SIP/331|60|tr") in new stack -- Called 331 -- SIP/331-fd48 is ringing -- SIP/331-fd48 answered SIP/322-3edd -- Attempting native bridge of SIP/322-3edd and SIP/331-fd48 My sip.conf has nat on and canreinvite=no, and those were the only suggestions I could find. Help would be greatly appreciated! We are really excited about the potential of Asterisk. Sip.conf: [322] type=friend context=softphone ; match with the outgoing context in extensions.conf host=dynamic ; This device needs to register callerid="Michael Furdyk" <XXXX> nat=yes ; X-Lite is behind a NAT router canreinvite=no ; Typically set to NO if behind NAT allow=all ; codec choice: GSM consumes far less bandwidth than ulaw dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users