Hi Gustavo quick question: As far as I can see the codec originated on the Huawei is g729 a=fmtp:18 annexb=yes
And on your asterisk > a=rtpmap:18 G729/8000 > > a=fmtp:18 annexb=no Check the options again for annexb also make sure that ont he Huawei side you do not autonegotiate PCMU and PCMA leave only g729 option, also remember that Asterisk is a passthru free for g729 meaning if it does transcoding you need the license for asterisk if not the call passes through, I have noticed that I had to load all the IVR's in g729 in order for them to play correctly thought otherwise it is going to be a silent connection. Kind regards, Saul Bejarano Gustavo A Gonzalez wrote: > HI folks! my topology is: > > > > softswitch (BROADSOFT) -- [sip trunk] -- Asterisk > > > > I need to connect phone calls using g729 codec. Debugging some calls we found > that calls can’t connect because of codec incompatibility. Our Sip provider > send us annexb=yes when a call is comming and our asterisk send annexb=no. > I’m running asterisk 1.4.21.1. Output debug shows: > > > > To: <sip:[EMAIL PROTECTED];user=phone> > > CSeq: 1 INVITE > > Contact: <sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp> > > Supported: 100rel > > User-Agent: Huawei SoftX3000 V300R006 > > Max-Forwards: 69 > > Allow: > INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER > > Content-Length: 274 > > Content-Type: application/sdp > > > > v=0 > > o=HuaweiSoftX3000 194579 194579 IN IP4 189.8.113.170 > > s=Sip Call > > c=IN IP4 XXX.X.XXX.170 > > t=0 0 > > m=audio 49256 RTP/AVP 18 8 0 97 > > a=rtpmap:18 G729/8000 > > a=rtpmap:8 PCMA/8000 > > a=rtpmap:0 PCMU/8000 > > a=rtpmap:97 telephone-event/8000 > > a=fmtp:97 0-15 > > a=fmtp:18 annexb=yes > > > > Via: SIP/2.0/UDP > XXX.X.XXX.170:5060;branch=z9hG4bKo2echm202o5g2dc38701.1;received=XXX.X.XXX.170 > > From: <sip:000@ XXX.X.XXX.170;user=phone>;tag=25f94692 > > To: <sip:7002@ XXX.X.XXX.177;user=phone>;tag=as0de67360 > > Call-ID: [EMAIL PROTECTED] > > CSeq: 1 INVITE > > User-Agent: Asterisk PBX > > Supported: replaces > > Contact: <sip:7002@ XXX.X.XXX.177> > > Content-Type: application/sdp > > Content-Length: 262 > > > > v=0 > > o=root 10183 10183 IN IP4 XXX.X.XXX.177 > > s=session > > c=IN IP4 XXX.X.XXX.177 > > t=0 0 > > m=audio 10772 RTP/AVP 18 97 > > a=rtpmap:18 G729/8000 > > a=fmtp:18 annexb=no > > a=rtpmap:97 telephone-event/8000 > > a=fmtp:97 0-16 > > a=silenceSupp:off - - - - > > a=ptime:20 > > a=sendrecv > > > > Thanks for any help! > > *Gustavo A. González* > Dto. de Infraestructura > Despegar.com, Inc. > [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
