<div dir="ltr"><br><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Jan 13, 2016 at 12:58 PM, Trey Hilyard <span dir="ltr"><<a href="mailto:kctrey@gmail.com" target="_blank">kctrey@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">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.<div><br></div><div>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.</div><div><br></div><div>Any thoughts?</div><div><br></div><div>SIP Logger:</div><div><div>INVITE sip:+18165116504@12.4.240.200:5060;user=phone SIP/2.0</div><div>v: SIP/2.0/UDP 10.77.27.103:20065;branch=z9hG4bK0020C575A392E895C39051;oc-accept</div><div>Max-Forwards: 70</div><div>t: <<a href="mailto:sip%3A%2B18165116504@12.4.240.200" target="_blank">sip:+18165116504@12.4.240.200</a>;user=phone></div><div>f: <<a href="mailto:sip%3A%2B18165116504@10.77.27.103" target="_blank">sip:+18165116504@10.77.27.103</a>;user=phone>;tag=000010847511385389740959</div><div>i: <a href="mailto:117620342110831512016142@10.77.27.103" target="_blank">117620342110831512016142@10.77.27.103</a></div><div>CSeq: 1 INVITE</div><div>d: no-fork</div><div>Privacy: none</div><div>P-Asserted-Identity: <sip:+18165116504;oli=62;rn=+1229218@10.77.27.103:20065;user=phone></div><div>Require: 100rel</div><div>Accept: application/sdp</div><div>k: histinfo,resource-priority</div><div>c: application/sdp</div><div>m: <sip:<a href="http://10.77.27.103:20065" target="_blank">10.77.27.103:20065</a>></div><div>Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,UPDATE</div><div>l: 228</div><div><br></div><div>v=0</div><div>o=PVG 1452710812870 1452710812870 IN IP4 10.77.160.55</div><div>s=-</div><div>c=IN IP4 10.77.160.55</div><div>t=0 0</div><div>m=audio 37700 RTP/AVP 0 101</div><div>b=AS:80</div><div>b=RR:0</div><div>b=RS:0</div><div>a=rtpmap:101 telephone-event/8000</div><div>a=fmtp:101 0-15</div><div>a=ptime:20</div><div>a=maxptime:20</div><div><br></div><div><--- Transmitting SIP response (495 bytes) to UDP:<a href="http://10.77.27.103:20065" target="_blank">10.77.27.103:20065</a> ---></div><div>SIP/2.0 421 Extension Required</div><div>Via: SIP/2.0/UDP 10.77.27.103:20065;rport=20065;received=10.77.27.103;branch=z9hG4bK0020C575A392E895C39051;oc-accept</div><div>Call-ID: <a href="mailto:117620342110831512016142@10.77.27.103" target="_blank">117620342110831512016142@10.77.27.103</a></div><div>From: <<a href="mailto:sip%3A%2B18165116504@10.77.27.103" target="_blank">sip:+18165116504@10.77.27.103</a>;user=phone>;tag=000010847511385389740959</div><div>To: <<a href="mailto:sip%3A%2B18165116504@12.4.240.200" target="_blank">sip:+18165116504@12.4.240.200</a>;user=phone>;tag=z9hG4bK0020C575A392E895C39051</div><div>CSeq: 1 INVITE</div><div>Require: 100rel</div><div>Supported: 100rel, timer, replaces, norefersub</div><div>Server: Asterisk PBX 13.3.0-rc1</div><div>Content-Length: 0</div></div><div><br></div></div></blockquote><div><br></div><div>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.<br></div></div><br>-- <br><div class="gmail_signature"><div dir="ltr"><div><div dir="ltr"><div>Matthew Jordan<br></div><div>Digium, Inc. | Director of Technology<br></div><div>445 Jan Davis Drive NW - Huntsville, AL 35806 - USA</div><div>Check us out at: <a href="http://digium.com" target="_blank">http://digium.com</a> & <a href="http://asterisk.org" target="_blank">http://asterisk.org</a></div></div></div></div></div>
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