On Wed, Jan 13, 2016 at 12:58 PM, Trey Hilyard <kct...@gmail.com> wrote:
> I am turning up a PJSIP Endpoint and am having problems when they send an > INVITE to my server. Asterisk is returning a 421 Extenstion Required. Since > "extension" means different things in the SIP stack versus Asterisk, I > don't know what it is complaining about. > > I have attached the trace below. Nothing else shows up with core verbose > or core debug enabled, so I am assuming it has to be dying at the PJSIP > module. The INVITE does come from an abnormal UDP Port, which is also shown > in the Via header, but the fact that the PBX is responding makes me think > that isn't the culprit. > > Any thoughts? > > SIP Logger: > INVITE sip:+18165116504@12.4.240.200:5060;user=phone SIP/2.0 > v: SIP/2.0/UDP 10.77.27.103:20065 > ;branch=z9hG4bK0020C575A392E895C39051;oc-accept > Max-Forwards: 70 > t: <sip:+18165116504@12.4.240.200;user=phone> > f: <sip:+18165116504@10.77.27.103;user=phone>;tag=000010847511385389740959 > i: 117620342110831512016142@10.77.27.103 > CSeq: 1 INVITE > d: no-fork > Privacy: none > P-Asserted-Identity: > <sip:+18165116504;oli=62;rn=+1229218@10.77.27.103:20065;user=phone> > Require: 100rel > Accept: application/sdp > k: histinfo,resource-priority > c: application/sdp > m: <sip:10.77.27.103:20065> > Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,UPDATE > l: 228 > > v=0 > o=PVG 1452710812870 1452710812870 IN IP4 10.77.160.55 > s=- > c=IN IP4 10.77.160.55 > t=0 0 > m=audio 37700 RTP/AVP 0 101 > b=AS:80 > b=RR:0 > b=RS:0 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > a=maxptime:20 > > <--- Transmitting SIP response (495 bytes) to UDP:10.77.27.103:20065 ---> > SIP/2.0 421 Extension Required > Via: SIP/2.0/UDP 10.77.27.103:20065 > ;rport=20065;received=10.77.27.103;branch=z9hG4bK0020C575A392E895C39051;oc-accept > Call-ID: 117620342110831512016142@10.77.27.103 > From: <sip:+18165116504@10.77.27.103 > ;user=phone>;tag=000010847511385389740959 > To: <sip:+18165116504@12.4.240.200 > ;user=phone>;tag=z9hG4bK0020C575A392E895C39051 > CSeq: 1 INVITE > Require: 100rel > Supported: 100rel, timer, replaces, norefersub > Server: Asterisk PBX 13.3.0-rc1 > Content-Length: 0 > > PJSIP is rejecting the inbound INVITE request as 100rel is required, but is not in the Supported header of the inbound SIP INVITE request. I would suspect that the UAC is doing things incorrectly by placing 100rel in the Require but not in the list of option tags in the Supported header. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org
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