[asterisk-users] pjsip aor stays in status created

[Richard Mudgett]

On Thu, Oct 25, 2018 at 6:58 AM marek cervenka <cervajs2 at gmail.com> wrote:

> hi,
>
> i have webrtc client chrome69/jssip which is connecting to asterisk
> 13.23.1/pjsip
>
> i have strange problem where pjsip aor stays in status "created"
>
> sip trace on asterisk looks ok.
>
>
> do you think if this can be bug?
>

It is not a bug.  The contact has been "created".  It will stay in that
state unless
you are also going to qualify the endpoint.  Asterisk 16 simply renames the
state to
"NonQualified" to be more explicit.

Richard


>
> test*CLI> pjsip show aors
>
>        Aor: <Aor..............................................>
> <MaxContact>
>      Contact:  <Aor/ContactUri............................> <Hash....>
> <Status> <RTT(ms)..>
>
> ==========================================================================================
>
>        Aor:  vr1k50                                               1
>      Contact:  vr1k50/sip:6i2b9766 at 1.1.1.1:34434;tran b2ad914030
> Created       0.000
>
>
>
>
> <--- Received SIP request (566 bytes) from WSS:1.1.1.1:34434 --->
> REGISTER sip:sip.example.com SIP/2.0
> Via: SIP/2.0/WSS v0i0at11ojbn.invalid;branch=z9hG4bK2155317
> Max-Forwards: 69
> To: <sip:vr1k50 at sip.example.com>
> From: "vr1k50" <sip:vr1k50 at sip.example.com>;tag=d56ij3vuo3
> Call-ID: 0mm678kf72bc9b5ur7ea8d
> CSeq: 13 REGISTER
> Contact:
> <sip:6i2b9766 at v0i0at11ojbn.invalid
> ;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:41c3d275-9c22-42ff-aeb3-987cb48902c7>";expires=60
> Expires: 60
> Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
> Supported: path,gruu,outbound
> User-Agent: JsSIP 3.2.9
> Content-Length: 0
>
>
> <--- Transmitting SIP response (484 bytes) to WSS:1.1.1.1:34434 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/WSS
> v0i0at11ojbn.invalid;rport=34434;received=1.1.1.1;branch=z9hG4bK2155317
> Call-ID: 0mm678kf72bc9b5ur7ea8d
> From: "vr1k50" <sip:vr1k50 at sip.example.com>;tag=d56ij3vuo3
> To: <sip:vr1k50 at sip.example.com>;tag=z9hG4bK2155317
> CSeq: 13 REGISTER
> WWW-Authenticate: Digest
>
> realm="asterisk",nonce="1540467808/121f72ae15612cc46a72e2861657a940",opaque="3060464337b28725",algorithm=md5,qop="auth"
> Server: Asterisk PBX 13.23.1
> Content-Length:  0
>
>
> <--- Received SIP request (837 bytes) from WSS:1.1.1.1:34434 --->
> REGISTER sip:sip.example.com SIP/2.0
> Via: SIP/2.0/WSS v0i0at11ojbn.invalid;branch=z9hG4bK9799804
> Max-Forwards: 69
> To: <sip:vr1k50 at sip.example.com>
> From: "vr1k50" <sip:vr1k50 at sip.example.com>;tag=d56ij3vuo3
> Call-ID: 0mm678kf72bc9b5ur7ea8d
> CSeq: 14 REGISTER
> Authorization: Digest algorithm=MD5, username="vr1k50",
> realm="asterisk", nonce="1540467808/121f72ae15612cc46a72e2861657a940",
> uri="sip:sip.example.com", response="376b4ac58b01dde2e043931467bba55a",
> opaque="3060464337b28725", qop=auth, cnonce="v8i7444gio8r", nc=00000001
> Contact:
> <sip:6i2b9766 at v0i0at11ojbn.invalid
> ;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:41c3d275-9c22-42ff-aeb3-987cb48902c7>";expires=60
> Expires: 60
> Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO
> Supported: path,gruu,outbound
> User-Agent: JsSIP 3.2.9
> Content-Length: 0
>
>
> <--- Transmitting SIP response (446 bytes) to WSS:1.1.1.1:34434 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/WSS
> v0i0at11ojbn.invalid;rport=34434;received=1.1.1.1;branch=z9hG4bK9799804
> Call-ID: 0mm678kf72bc9b5ur7ea8d
> From: "vr1k50" <sip:vr1k50 at sip.example.com>;tag=d56ij3vuo3
> To: <sip:vr1k50 at sip.example.com>;tag=z9hG4bK9799804
> CSeq: 14 REGISTER
> Date: Thu, 25 Oct 2018 11:43:28 GMT
> Contact: <sip:6i2b9766 at 1.1.1.1:34434;transport=ws>;expires=59
> Expires: 60
> Server: Asterisk PBX 13.23.1
> Content-Length:  0
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Astricon is coming up October 9-11!  Signup is available at:
> https://www.asterisk.org/community/astricon-user-conference
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20181025/f9364a2c/attachment.html>