Asterisk does not support that yet. Zoa
rachid wrote: > Hello, > > I have some problems to use G722, when my client sent an invite request > to asterisk using G722/16000 codec > asterisk respond with G722/8000 codec. > > I dont know exactly if Asterisk supports G722/16000 codec?? > If yes how can I activate It?? > > Thanks. > > Rachid. > > Below wireshak trace: > > <--------------------------------------------------------------------------------------------------------------------> > > My Invite: > > INVITE sip:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 192.168.10.12:5060;branch=z9hG4bK2600322761 > From: Manager <sip:[EMAIL PROTECTED]>;tag=3871604470 > To: <sip:[EMAIL PROTECTED]> > Call-ID: [EMAIL PROTECTED] > CSeq: 21 INVITE > Contact: <sip:[EMAIL PROTECTED]:5060> > Authorization: Digest username="Manager", realm="asterisk", > nonce="1c8c3fd9", uri="sip:[EMAIL PROTECTED]", > response="5d32f87fa423cd2f1bf9aefb8cf920b6", algorithm=MD5 > Max-Forwards: 70 > User-Agent: wengo/v1/wengophoneng/wengo/rev54/trunk/ > Expires: 120 > Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE > Content-Type: application/sdp > Content-Length: 365 > > v=0 > o=userX 20000001 20000001 IN IP4 192.168.10.12 > s=A call > c=IN IP4 192.168.10.12 > t=1202402970 1202406570 > m=audio 10600 RTP/AVP 0 8 109 3 101 > a=rtpmap:0 PCMU/8000/1 > a=rtpmap:8 PCMA/8000/1 > a=rtpmap:109 G722/16000/1 > a=rtpmap:3 GSM/8000/1 > a=rtpmap:101 telephone-event/8000/1 > m=video 10702 RTP/AVP 34 31 > a=rtpmap:34 H263/90000/1 > a=rtpmap:31 H261/90000/1 > > <--------------------------------------------------------------------------------------------------------------------> > > Asterisk response: > > SIP/2.0 200 OK > Via: SIP/2.0/UDP > 192.168.10.12:5060;branch=z9hG4bK2600322761;received=77.203.231.140 > From: Manager <sip:[EMAIL PROTECTED]>;tag=3871604470 > To: <sip:[EMAIL PROTECTED]>;tag=as5c1447b6 > Call-ID: [EMAIL PROTECTED] > CSeq: 21 INVITE > User-Agent: Asterisk PBX SVN-trunk-r102777 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces, timer > Contact: <sip:[EMAIL PROTECTED]:5060> > Content-Type: application/sdp > Content-Length: 397 > > v=0 > o=root 1999706631 1999706631 IN IP4 91.121.31.80 > s=Asterisk PBX SVN-trunk-r102777 > c=IN IP4 91.121.31.80 > b=CT:384 > t=0 0 > m=audio 18950 RTP/AVP 109 0 8 101 > a=rtpmap:109 G722/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > m=video 18692 RTP/AVP 34 > a=rtpmap:34 H263/90000 > a=sendrecv > > <--------------------------------------------------------------------------------------------------------------------> > > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
