<div dir="ltr"><div dir="ltr"><br><br><div class="gmail_quote"><div dir="ltr">On Thu, Oct 25, 2018 at 6:58 AM marek cervenka <<a href="mailto:cervajs2@gmail.com">cervajs2@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">hi,<br>
<br>
i have webrtc client chrome69/jssip which is connecting to asterisk <br>
13.23.1/pjsip<br>
<br>
i have strange problem where pjsip aor stays in status "created"<br>
<br>
sip trace on asterisk looks ok.<br>
<br>
<br>
do you think if this can be bug?<br></blockquote><div><br></div><div>It is not a bug.  The contact has been "created".  It will stay in that state unless</div><div>you are also going to qualify the endpoint.  Asterisk 16 simply renames the state to</div><div>"NonQualified" to be more explicit.<br></div><div><br></div><div>Richard</div><div><br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<br>
<br>
test*CLI> pjsip show aors<br>
<br>
       Aor: <Aor..............................................> <MaxContact><br>
     Contact:  <Aor/ContactUri............................> <Hash....> <br>
<Status> <RTT(ms)..><br>
==========================================================================================<br>
<br>
       Aor:  vr1k50                                               1<br>
     Contact:  vr1k50/<a href="mailto:sip%3A6i2b9766@1.1.1.1" target="_blank">sip:6i2b9766@1.1.1.1</a>:34434;tran b2ad914030 <br>
Created       0.000<br>
<br>
<br>
<br>
<br>
<--- Received SIP request (566 bytes) from WSS:<a href="http://1.1.1.1:34434" rel="noreferrer" target="_blank">1.1.1.1:34434</a> ---><br>
REGISTER sip:<a href="http://sip.example.com" rel="noreferrer" target="_blank">sip.example.com</a> SIP/2.0<br>
Via: SIP/2.0/WSS v0i0at11ojbn.invalid;branch=z9hG4bK2155317<br>
Max-Forwards: 69<br>
To: <<a href="mailto:sip%3Avr1k50@sip.example.com" target="_blank">sip:vr1k50@sip.example.com</a>><br>
From: "vr1k50" <<a href="mailto:sip%3Avr1k50@sip.example.com" target="_blank">sip:vr1k50@sip.example.com</a>>;tag=d56ij3vuo3<br>
Call-ID: 0mm678kf72bc9b5ur7ea8d<br>
CSeq: 13 REGISTER<br>
Contact: <br>
<sip:6i2b9766@v0i0at11ojbn.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:41c3d275-9c22-42ff-aeb3-987cb48902c7>";expires=60<br>
Expires: 60<br>
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO<br>
Supported: path,gruu,outbound<br>
User-Agent: JsSIP 3.2.9<br>
Content-Length: 0<br>
<br>
<br>
<--- Transmitting SIP response (484 bytes) to WSS:<a href="http://1.1.1.1:34434" rel="noreferrer" target="_blank">1.1.1.1:34434</a> ---><br>
SIP/2.0 401 Unauthorized<br>
Via: SIP/2.0/WSS <br>
v0i0at11ojbn.invalid;rport=34434;received=1.1.1.1;branch=z9hG4bK2155317<br>
Call-ID: 0mm678kf72bc9b5ur7ea8d<br>
From: "vr1k50" <<a href="mailto:sip%3Avr1k50@sip.example.com" target="_blank">sip:vr1k50@sip.example.com</a>>;tag=d56ij3vuo3<br>
To: <<a href="mailto:sip%3Avr1k50@sip.example.com" target="_blank">sip:vr1k50@sip.example.com</a>>;tag=z9hG4bK2155317<br>
CSeq: 13 REGISTER<br>
WWW-Authenticate: Digest <br>
realm="asterisk",nonce="1540467808/121f72ae15612cc46a72e2861657a940",opaque="3060464337b28725",algorithm=md5,qop="auth"<br>
Server: Asterisk PBX 13.23.1<br>
Content-Length:  0<br>
<br>
<br>
<--- Received SIP request (837 bytes) from WSS:<a href="http://1.1.1.1:34434" rel="noreferrer" target="_blank">1.1.1.1:34434</a> ---><br>
REGISTER sip:<a href="http://sip.example.com" rel="noreferrer" target="_blank">sip.example.com</a> SIP/2.0<br>
Via: SIP/2.0/WSS v0i0at11ojbn.invalid;branch=z9hG4bK9799804<br>
Max-Forwards: 69<br>
To: <<a href="mailto:sip%3Avr1k50@sip.example.com" target="_blank">sip:vr1k50@sip.example.com</a>><br>
From: "vr1k50" <<a href="mailto:sip%3Avr1k50@sip.example.com" target="_blank">sip:vr1k50@sip.example.com</a>>;tag=d56ij3vuo3<br>
Call-ID: 0mm678kf72bc9b5ur7ea8d<br>
CSeq: 14 REGISTER<br>
Authorization: Digest algorithm=MD5, username="vr1k50", <br>
realm="asterisk", nonce="1540467808/121f72ae15612cc46a72e2861657a940", <br>
uri="sip:<a href="http://sip.example.com" rel="noreferrer" target="_blank">sip.example.com</a>", response="376b4ac58b01dde2e043931467bba55a", <br>
opaque="3060464337b28725", qop=auth, cnonce="v8i7444gio8r", nc=00000001<br>
Contact: <br>
<sip:6i2b9766@v0i0at11ojbn.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:41c3d275-9c22-42ff-aeb3-987cb48902c7>";expires=60<br>
Expires: 60<br>
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO<br>
Supported: path,gruu,outbound<br>
User-Agent: JsSIP 3.2.9<br>
Content-Length: 0<br>
<br>
<br>
<--- Transmitting SIP response (446 bytes) to WSS:<a href="http://1.1.1.1:34434" rel="noreferrer" target="_blank">1.1.1.1:34434</a> ---><br>
SIP/2.0 200 OK<br>
Via: SIP/2.0/WSS <br>
v0i0at11ojbn.invalid;rport=34434;received=1.1.1.1;branch=z9hG4bK9799804<br>
Call-ID: 0mm678kf72bc9b5ur7ea8d<br>
From: "vr1k50" <<a href="mailto:sip%3Avr1k50@sip.example.com" target="_blank">sip:vr1k50@sip.example.com</a>>;tag=d56ij3vuo3<br>
To: <<a href="mailto:sip%3Avr1k50@sip.example.com" target="_blank">sip:vr1k50@sip.example.com</a>>;tag=z9hG4bK9799804<br>
CSeq: 14 REGISTER<br>
Date: Thu, 25 Oct 2018 11:43:28 GMT<br>
Contact: <<a href="http://sip:6i2b9766@1.1.1.1:34434" target="_blank">sip:6i2b9766@1.1.1.1:34434</a>;transport=ws>;expires=59<br>
Expires: 60<br>
Server: Asterisk PBX 13.23.1<br>
Content-Length:  0<br>
<br>
<br>
-- <br>
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Astricon is coming up October 9-11!  Signup is available at: <a href="https://www.asterisk.org/community/astricon-user-conference" rel="noreferrer" target="_blank">https://www.asterisk.org/community/astricon-user-conference</a><br>
<br>
Check out the new Asterisk community forum at: <a href="https://community.asterisk.org/" rel="noreferrer" target="_blank">https://community.asterisk.org/</a><br>
<br>
New to Asterisk? Start here:<br>
      <a href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started" rel="noreferrer" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Getting+Started</a><br>
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