I'm reaching out to the asterisk users e-mail list in hopes someone there can provide guidance. A couple of Digium's developers check this e-mail group so they may respond. Unfortunately, they are basically in the get ready for show mode. Major show is next week and they are also releasing the next major version in a matter of days.
Asking if them... What causes asterisk to output the Oooh, got a frame with format of g729 on channel 'PJSIP/121-000001d2' when we're sending 'ulaw', switching to match message? Is it asterisk detecting rtp stream with g729? Why would asterisk change to g729 stream if the codec negotiation is clearly ulaw in the SIP packets? Dan Cropp Senior Software Engineer, R&D Software Dept. AMTELCO, 4800 Curtin Drive, McFarland, WI 53558-9424 608 838-4197 ext. 291 1-800-238-5275 ext 291 www.amtelco.com<http://www.amtelco.com/> Statement of Confidentiality The contents of this e-mail message and any attachments are confidential to American Tel-A-Systems, Inc. (AMTELCO), and are intended solely for the addressee(s). The information contained in this transmission also may be of a legally privileged nature. This transmission is sent in trust and is meant solely for delivery to the intended recipient(s). Receipt of this transmission does not convey any right to reproduce or disseminate any of the information it contains. If you are not the intended recipient, please immediately notify the sender by reply e-mail or telephone and delete this message and any attachments. From: Dan Cropp Sent: Wednesday, October 3, 2018 1:57 PM To: [email protected] Subject: Any idea what causes "Oooh, got a frame with format of g729 on channel 'PJSIP/121-000001d2' when we're sending 'ulaw', switching to match" 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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