<div dir="ltr"><div><br></div><div>Thanks, there are no register lines in my sip.conf, but I have defined callbackextension fields in the realtime table, to be the same value as the extension name. In this case, extension 771 has callbackextension value 771. I tried replacing those with null values but that had no effect on the outcome.<br>
</div><div><br></div><div>Currently when I register clients in, after some seconds Asterisk starts sending REGISTER messages, at which point Kamailio sees 2 AORs, here's an example: </div><div>(here 1.1.1.1 is the public ip of my server that houses Kamailio at 5060 and Asterisk at 5070, and 2.2.2.2 is the public ip of the network clients are in)</div>
<div><br></div><div><div> AOR:: <a href="mailto:771@testers.com">771@testers.com</a></div><div> Contact:: sip:771@2.2.2.2:5060;rinstance=c8447637c890c010;transport=UDP Q=</div><div> Expires:: 3470</div>
<div> Callid:: NDQ5Njk4ZmUxZGJhNzRjMzUwMTA2OThmOGFjYzc4Zjk.</div><div> Cseq:: 2</div><div> User-agent:: Z 3.2.21357 r21367</div><div> State:: CS_SYNC</div>
<div> Flags:: 0</div><div> Cflag:: 0</div><div> Socket:: udp:<a href="http://1.1.1.1:5060">1.1.1.1:5060</a></div><div> Methods:: 5087</div>
<div> Ruid:: uloc-53bfe447-35b0-608</div><div> Reg-Id:: 0</div><div> Last-Keepalive:: 1405429865</div><div> Last-Modified:: 1405429865</div>
<div> AOR:: <a href="mailto:771@1.1.1.1">771@1.1.1.1</a></div><div> Contact:: <a href="http://sip:771@1.1.1.1:5070">sip:771@1.1.1.1:5070</a> Q=</div><div> Expires:: 105</div><div>
Callid:: <a href="mailto:3e946958322b1e2d6bfa564d46bf8e00@testers.com">3e946958322b1e2d6bfa564d46bf8e00@testers.com</a></div><div> Cseq:: 133</div><div> User-agent:: Asterisk PBX</div>
<div> State:: CS_SYNC</div><div> Flags:: 0</div><div> Cflag:: 0</div><div> Socket:: udp:<a href="http://1.1.1.1:5060">1.1.1.1:5060</a></div>
<div> Methods:: 4294967295</div><div> Ruid:: uloc-53bfe447-35b0-708</div><div> Reg-Id:: 0</div><div> Last-Keepalive:: 1405429980</div>
<div> Last-Modified:: 1405429980</div></div><div><br></div><div>I guess there should be only one AOR, so Asterisk might get wrong kind of data to begin with or it's configured incorrectly. In my sip trace the REGISTER flow from client to Kamailio to Asterisk seems ok, I could be wrong though. </div>
<div><br></div><div>In my setup clients authenticate with Kamailio and Kamailio sends a REGISTER to Asterisk according to guide I used: <a href="http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb">http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb</a></div>
<div><br></div><div>How would I fix this double-AOR problem, can it be fixed on Asterisk configuration? </div><div><br></div><div>thanks,</div><div>Olli</div><div><br></div><div><br></div><div><br></div><div><br></div><div>
<br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">2014-07-15 16:00 GMT+03:00 Joshua Colp <span dir="ltr"><<a href="mailto:jcolp@digium.com" target="_blank">jcolp@digium.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Olli Heiskanen wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div class="">
Hello,<br>
<br>
Thanks for your response, I actually verified that the Zoiper setting is<br>
not the reason for Asterisk to start sending REGISTERs, it only looked<br>
like it as I checked the Kamailio output before Asterisk sent the first<br>
REGISTER to Kamailio, right after I had played with that setting.<br>
(sorry, my bad!)<br>
<br>
However, _something_ is causing these REGISTERs, here's an example of a<br>
REGISTER message sent from Asterisk to Kamailio:<br>
<br></div>
REGISTER sip:<a href="http://testers.com" target="_blank">testers.com</a> <<a href="http://testers.com" target="_blank">http://testers.com</a>> SIP/2.0<div class=""><br>
Via: SIP/2.0/UDP my_ip:5070;branch=<u></u>z9hG4bK7477f754;rport<br>
Max-Forwards: 70<br>
From: <sip:771@my_ip>;tag=as7a88c4c6<br>
To: <sip:771@my_ip><br>
Call-ID: <a href="mailto:3e946958322b1e2d6bfa564d46bf8e00@testers.com" target="_blank">3e946958322b1e2d6bfa564d46bf8e<u></u>00@testers.com</a><br></div>
<mailto:<a href="mailto:3e946958322b1e2d6bfa564d46bf8e00@testers.com" target="_blank">3e946958322b1e2d6bfa56<u></u>4d46bf8e00@testers.com</a>><div class=""><br>
CSeq: 121 REGISTER<br>
User-Agent: Asterisk PBX<br>
Expires: 120<br>
Contact: <<a href="http://sip:771@91.221.66.61:5070" target="_blank">sip:771@91.221.66.61:5070</a><br></div>
<<a href="http://sip:771@91.221.66.61:5070" target="_blank">http://sip:771@91.221.66.61:<u></u>5070</a>>><div class=""><br>
Content-Length: 0<br>
<br>
Is there any other reason - other than client settings - why this would<br>
happen?<br>
</div></blockquote>
<br>
If Asterisk was configured to do so, yes. Do you have any register lines in sip.conf or do you have the "callbackextension" option set for any peers?<div class="HOEnZb"><div class="h5"><br>
<br>
-- <br>
Joshua Colp<br>
Digium, Inc. | Senior Software Developer<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - US<br>
Check us out at: <a href="http://www.digium.com" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a><br>
<br>
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