On Thu, Oct 25, 2018 at 6:58 AM marek cervenka <[email protected]> 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:[email protected]: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:[email protected]> > From: "vr1k50" <sip:[email protected]>;tag=d56ij3vuo3 > Call-ID: 0mm678kf72bc9b5ur7ea8d > CSeq: 13 REGISTER > Contact: > <sip:[email protected] > ;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:[email protected]>;tag=d56ij3vuo3 > To: <sip:[email protected]>;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:[email protected]> > From: "vr1k50" <sip:[email protected]>;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:[email protected] > ;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:[email protected]>;tag=d56ij3vuo3 > To: <sip:[email protected]>;tag=z9hG4bK9799804 > CSeq: 14 REGISTER > Date: Thu, 25 Oct 2018 11:43:28 GMT > Contact: <sip:[email protected]: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
-- _____________________________________________________________________ -- 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
