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 <j...@calva.com> 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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