Argh. That was for chan_pjsip and you are using chan_sip. Be aware that chan_sip is effectively dead.
Richard On Thu, May 14, 2020 at 9:50 AM Richard Mudgett <[email protected]> wrote: > The other end is sending g729 even though it was not negotiated. The > other end should not do this and it usually seems that the other ends that > do send g729. > This was recently fixed. See > https://issues.asterisk.org/jira/browse/ASTERISK-28139 > > Richard > > On Thu, May 14, 2020 at 1:11 AM John Hughes <[email protected]> wrote: > >> I am having a problem with one of my callers who is using either g729 or >> alaw. I can do alaw but not g729 so asterisk should negotiate alaw right? >> In fact from the sip debug it looks like it does, but then I get the >> dreaded "channel.c:5630 set_format: Unable to find a codec translation >> path: (g729) -> (alaw)" and the call hangs up. Why? >> >> Last minute thought: Is it possible that the caller is sending g729 in >> RTP even though the SIP negotiation clearly chooses alaw? Maybe I need >> some RTP debugging. >> >> Asterisk 13.14.1 on Debian, using chan_sip. >> >> Here's the trace: >> >> <--- SIP read from UDP:SUPPLIER:5060 ---> >> INVITE sip:LOCAL@ASTERISK:5060 SIP/2.0 >> Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9 >> From: <sip:REMOTE@SUPPLIER>;tag=gK02498cb1 >> To: <sip:LOCAL@ASTERISK> >> Call-ID: 205665777_90679951@SUPPLIER >> CSeq: 539098 INVITE >> Max-Forwards: 70 >> Allow: >> INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS,MESSAGE,PUBLISH >> Accept: application/sdp, application/isup, application/dtmf, >> application/dtmf-relay, multipart/mixed >> Contact: <sip:REMOTE@SUPPLIER:5060> >> P-Asserted-Identity: <sip:REMOTE@REMOTE-SUPPLIER;user=phone> >> Supported: timer,100rel,precondition >> Session-Expires: 1800 >> Min-SE: 90 >> Content-Length: 282 >> Content-Disposition: session; handling=required >> Content-Type: application/sdp >> >> v=0 >> o=Sonus_UAC 176880 320591 IN IP4 SUPPLIER >> s=SIP Media Capabilities >> c=IN IP4 213.41.124.6 >> t=0 0 >> m=audio 8526 RTP/AVP 18 8 101 >> a=rtpmap:18 G729/8000 >> a=fmtp:18 annexb=no >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=sendrecv >> a=ptime:20 >> <-------------> >> --- (17 headers 13 lines) --- >> Sending to SUPPLIER:5060 (no NAT) >> Sending to SUPPLIER:5060 (no NAT) >> Using INVITE request as basis request - 205665777_90679951@SUPPLIER >> Found peer 'supplier' for 'REMOTE' from SUPPLIER:5060 >> Found RTP audio format 18 >> Found RTP audio format 8 >> Found RTP audio format 101 >> Found audio description format G729 for ID 18 >> Found audio description format PCMA for ID 8 >> Found audio description format telephone-event for ID 101 >> Capabilities: us - (alaw|ulaw|gsm), peer - >> audio=(alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw) >> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 >> (telephone-event|), combined - 0x1 (telephone-event|) >> Peer audio RTP is at port 213.41.124.6:8526 >> Looking for LOCAL in supplier-in (domain ASTERISK) >> sip_route_dump: route/path hop: <sip:REMOTE@SUPPLIER:5060> >> >> So, all looking good here, we've worked out that the combined >> capabilities are (alaw) >> >> <--- Transmitting (no NAT) to SUPPLIER:5060 ---> >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP >> SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9;received=SUPPLIER >> From: <sip:REMOTE@SUPPLIER>;tag=gK02498cb1 >> To: <sip:LOCAL@ASTERISK> >> Call-ID: 205665777_90679951@SUPPLIER >> CSeq: 539098 INVITE >> Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Session-Expires: 1800;refresher=uas >> Contact: <sip:LOCAL@ASTERISK:5060> >> Content-Length: 0 >> >> >> <------------> >> >> <--- Transmitting (no NAT) to SUPPLIER:5060 ---> >> SIP/2.0 180 Ringing >> Via: SIP/2.0/UDP >> SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9;received=SUPPLIER >> From: <sip:REMOTE@SUPPLIER>;tag=gK02498cb1 >> To: <sip:LOCAL@ASTERISK>;tag=as4502927f >> Call-ID: 205665777_90679951@SUPPLIER >> CSeq: 539098 INVITE >> Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Session-Expires: 1800;refresher=uas >> Contact: <sip:LOCAL@ASTERISK:5060> >> Content-Length: 0 >> >> >> <------------> >> Audio is at 13948 >> Adding codec alaw to SDP >> Adding non-codec 0x1 (telephone-event) to SDP >> >> <--- Reliably Transmitting (no NAT) to SUPPLIER:5060 ---> >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP >> SUPPLIER:5060;branch=z9hG4bK02B5ab9c8e55f864da9;received=SUPPLIER >> From: <sip:REMOTE@SUPPLIER>;tag=gK02498cb1 >> To: <sip:LOCAL@ASTERISK>;tag=as4502927f >> Call-ID: 205665777_90679951@SUPPLIER >> CSeq: 539098 INVITE >> Server: Asterisk PBX 13.14.1~dfsg-2+deb9u4 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, >> PUBLISH, MESSAGE >> Supported: replaces, timer >> Session-Expires: 1800;refresher=uas >> Contact: <sip:LOCAL@ASTERISK:5060> >> Content-Type: application/sdp >> Require: timer >> Content-Length: 264 >> >> v=0 >> o=root 227409966 227409966 IN IP4 ASTERISK >> s=Asterisk PBX 13.14.1~dfsg-2+deb9u4 >> c=IN IP4 ASTERISK >> t=0 0 >> m=audio 13948 RTP/AVP 8 101 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:20 >> a=maxptime:150 >> a=sendrecv >> >> <------------> >> >> >> And that's good to, we've sent the OK for the INVITE saying that we want >> alaw. >> >> >> <--- SIP read from UDP:SUPPLIER:5060 ---> >> ACK sip:LOCAL@ASTERISK:5060 SIP/2.0 >> Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5bc037285f864da9 >> From: <sip:REMOTE@SUPPLIER>;tag=gK02498cb1 >> To: <sip:LOCAL@ASTERISK>;tag=as4502927f >> Call-ID: 205665777_90679951@SUPPLIER >> CSeq: 539098 ACK >> Max-Forwards: 70 >> Content-Length: 0 >> >> <-------------> >> --- (8 headers 0 lines) --- >> [May 13 13:46:58] WARNING[7245][C-000031da]: channel.c:5630 set_format: >> Unable to find a codec translation path: (g729) -> (alaw) >> >> What's this nonsense! Why is set_format trying to use g729! >> >> Scheduling destruction of SIP dialog '205665777_90679951@SUPPLIER' in 32000 >> ms (Method: ACK) >> set_destination: Parsing <sip:REMOTE@SUPPLIER:5060> for address/port to send >> to >> set_destination: set destination to SUPPLIER:5060 >> Reliably Transmitting (no NAT) to SUPPLIER:5060: >> BYE sip:REMOTE@SUPPLIER:5060 SIP/2.0 >> Via: SIP/2.0/UDP ASTERISK:5060;branch=z9hG4bK156fd67d >> Max-Forwards: 70 >> From: <sip:LOCAL@ASTERISK>;tag=as4502927f >> To: <sip:REMOTE@SUPPLIER>;tag=gK02498cb1 >> Call-ID: 205665777_90679951@SUPPLIER >> CSeq: 102 BYE >> User-Agent: Asterisk PBX 13.14.1~dfsg-2+deb9u4 >> X-Asterisk-HangupCause: Normal Clearing >> X-Asterisk-HangupCauseCode: 16 >> Content-Length: 0 >> >> >> --- >> >> <--- SIP read from UDP:SUPPLIER:5060 ---> >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP ASTERISK:5060;branch=z9hG4bK156fd67d >> From: <sip:LOCAL@ASTERISK>;tag=as4502927f >> To: <sip:REMOTE@SUPPLIER>;tag=gK02498cb1 >> Call-ID: 205665777_90679951@SUPPLIER >> CSeq: 102 BYE >> Content-Length: 0 >> >> <-------------> >> --- (7 headers 0 lines) --- >> SIP Response message for INCOMING dialog BYE arrived >> Really destroying SIP dialog '205665777_90679951@SUPPLIER' Method: ACK >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk community forum at: >> https://community.asterisk.org/ >> >> New to Asterisk? 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