[asterisk-users] PJSIP Returning 421 Extension Required

[Matthew Jordan]

On Wed, Jan 13, 2016 at 12:58 PM, Trey Hilyard <kctrey at 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 at 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 at 12.4.240.200;user=phone>
> f: <sip:+18165116504 at 10.77.27.103;user=phone>;tag=000010847511385389740959
> i: 117620342110831512016142 at 10.77.27.103
> CSeq: 1 INVITE
> d: no-fork
> Privacy: none
> P-Asserted-Identity:
> <sip:+18165116504;oli=62;rn=+1229218 at 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 at 10.77.27.103
> From: <sip:+18165116504 at 10.77.27.103
> ;user=phone>;tag=000010847511385389740959
> To: <sip:+18165116504 at 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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