On Thu, Oct 25, 2018 at 6:58 AM marek cervenka 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:
>
> Contact:
>
>
> ==
>
>Aor: vr1k50 1
> Contact: vr1k50/sip:6i2b9766@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:
> From: "vr1k50" ;tag=d56ij3vuo3
> Call-ID: 0mm678kf72bc9b5ur7ea8d
> CSeq: 13 REGISTER
> Contact:
> ;transport=ws>;+sip.ice;reg-id=1;+sip.instance="";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" ;tag=d56ij3vuo3
> To: ;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:
> From: "vr1k50" ;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=0001
> Contact:
> ;transport=ws>;+sip.ice;reg-id=1;+sip.instance="";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" ;tag=d56ij3vuo3
> To: ;tag=z9hG4bK9799804
> CSeq: 14 REGISTER
> Date: Thu, 25 Oct 2018 11:43:28 GMT
> Contact: ;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
--
_
-- 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
