I am running Trixbox PRO.
I don’t know if this is a config issue, since it would seem to be odd that an inbound SIP call into asterisk would answer the call even during ringing. Check out the SIP trace below. It’s a call from the PSTN into an asterisk DID assigned to an ext. On the PSTN side the caller heard ringing before the call was answered…yet on the SIP side there is no 18x response back just: Here is a trace of the INBOUND side (PSTN into SBC) and OUTBOUND (SBC into Asterisk): INCOMING INTO SBC Feb 17 15:01:40 2010 [x.x.x.x] ==> [y.y.y.y] INVITE sip:[email protected]/2.0 m=audio 20828 RTP/AVP 0 18 8 101 c=IN IP4 x.x.x.x a=fmtp:18 annexb=no Feb 17 15:01:40 2010 [y.y.y.y] ==> [x.x.x.x] SIP/2.0 100 Trying Feb 17 15:01:40 2010 [y.y.y..y] ==> [x.x.x.x] SIP/2.0 200 OK m=audio 21642 RTP/AVP 0 c=IN IP4 y.y.y.y Feb 17 15:01:40 2010 [x.x.x.x] ==> [y.y.y.y] ACK sip:y.y.y.y:5060 SIP/2.0 Feb 17 15:01:40 2010 [x.x.x.x] ==> [y.y.y.y] ACK sip:y.y.y.y:5060 SIP/2.0 Feb 17 15:02:31 2010 [y.y.y.y] ==> [x.x.x.x] BYE sip:[email protected]:5060 SIP/2.0 Feb 17 15:02:31 2010 [x.x.x.x] ==> [y.y.y.y] SIP/2.0 200 OK OUTGOING FROM SBC INTO ASTERISK Feb 17 15:01:40 2010 [x.x.x.x] ==> [y.y.y.y] INVITE sip:[email protected]:5060;user=phone SIP/2.0 m=audio 21142 RTP/AVP 0 18 8 101 c=IN IP4 x.x.x.x a=fmtp:18 annexb=no a=sendrecv Feb 17 15:01:40 2010 [y.y.y.y] ==> [x.x.x.x] SIP/2.0 100 Trying *Feb 17 15:01:40 2010 [y.y.y.y] ==> [x.x.x.x] SIP/2.0 200 OK* * m=audio 19282 RTP/AVP 0 8 c=IN IP4 y.y.y.y * Feb 17 15:01:40 2010 [x.x.x.x] ==> [y.y.y.y] ACK sip:[email protected]/2.0 Feb 17 15:02:31 2010 [y.y.y.y] ==> [x.x.x.x] BYE sip:[email protected]:5060 SIP/2.0 Feb 17 15:02:31 2010 [x.x.x.x] ==> [y.y.y.y] SIP/2.0 200 OK
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