The PJSIP endpoint is configured for ulaw only. Not sure how or why we are seeing the g729 on calls for this endpoint.
Would this be a case that asterisk detects the rtp stream is g729 even though it's negotiated as ulaw? Why would asterisk change the format to g729 when disallow = all and allow = ulaw are the endpoint settings? [121] type = endpoint context = IS transport = transport1 aors = 121 accountcode = 2 dtmf_mode = inband device_state_busy_at = 96 force_rport = no identify_by = username,ip disallow = all allow = ulaw from_user = 121 acl = acl1 [10/03 11:29:34.240] DEBUG[21414][C-00000226] chan_pjsip.c: Oooh, got a frame with format of g729 on channel 'PJSIP/121-000001d2' when we're sending 'ulaw', switching to match INVITE sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bKxqyYl2Jg84158000 To: <sip:2197@ XXX.XXX.XXX.XXX <sip:[email protected]> > From: <sip:[email protected] <sip:[email protected]%20> >;tag=I0n4X7KK Contact: <sip:STUFF @YYY.YYY.YYY.YYY:5060<sip:[email protected]:5060>> Call-ID: OlrmFuyOq-0000-@ YYY.YYY.YYY.YYY <mailto:[email protected]> CSeq: 2223 INVITE Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,OPTIONS,REFER,NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 335 v=0 o=- 11264000 11264000 IN IP4 YYY.YYY.YYY.YYY s=- c=IN IP4 192.168.10.213 t=0 0 m=audio 32768 RTP/AVP 0 2 8 18 110 120 100 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:110 G723/5300 a=rtpmap:120 G723/6300 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-15 a=sendrecv Send SIP/2.0 200 OK Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;received=YYY.YYY.YYY.YYY;branch=z9hG4bKxqyYl2Jg84158000 Call-ID: [email protected] <mailto:[email protected]%20> From: <sip:[email protected] >;tag=I0n4X7KK To: <sip:[email protected] <sip:[email protected]%20> >;tag=b4134118-08f4-4dbc-a145-573d04438092 CSeq: 2223 INVITE Server: Asterisk PBX 13.20.0 Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE Contact: <sip:YYY.YYY.YYY.YYY:5060> Supported: 100rel, timer, replaces, norefersub Content-Type: application/sdp Content-Length: 181 v=0 o=- 11264000 11264002 IN IP4 YYY.YYY.YYY.YYY s=Asterisk c=IN IP4 192.168.11.176 t=0 0 m=audio 18380 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=maxptime:150 a=sendrecv Receive ACK sip:XXX.XXX.XXX.XXX:5060 SIP/2.0 Via: SIP/2.0/UDP YYY.YYY.YYY.YYY:5060;branch=z9hG4bKCIJuiRuH8415a000 To: <sip:2197@ XXX.XXX.XXX.XXX <sip:[email protected]> >;tag=b4134118-08f4-4dbc-a145-573d04438092 From: <sip:STUFF@ YYY.YYY.YYY.YYY <sip:STUFF@%20YYY.YYY.YYY.YYY%20> >;tag=I0n4X7KK Max-Forwards: 70 Call-ID: [email protected]<mailto:[email protected]> CSeq: 2223 ACK Asterisk debugging outputs the following. [10/03 11:29:34.240] DEBUG[21414][C-00000226] chan_pjsip.c: Oooh, got a frame with format of g729 on channel 'PJSIP/121-000001d2' when we're sending 'ulaw', switching to match [10/03 11:29:34.240] DEBUG[21414][C-00000226] channel.c: Channel PJSIP/121-000001d2 setting write format path: ulaw -> g729 [10/03 11:29:34.240] DEBUG[21414][C-00000226] channel.c: Channel PJSIP/121-000001d2 setting read format path: ulaw -> g729 [10/03 11:29:34.240] DEBUG[21414][C-00000226] channel.c: Channel PJSIP/121-000001d2 setting read format path: g729 -> g729
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