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.
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: (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) AOR:: 7...@testers.com Contact:: sip:771@2.2.2.2:5060;rinstance=c8447637c890c010;transport=UDP Q= Expires:: 3470 Callid:: NDQ5Njk4ZmUxZGJhNzRjMzUwMTA2OThmOGFjYzc4Zjk. Cseq:: 2 User-agent:: Z 3.2.21357 r21367 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:1.1.1.1:5060 Methods:: 5087 Ruid:: uloc-53bfe447-35b0-608 Reg-Id:: 0 Last-Keepalive:: 1405429865 Last-Modified:: 1405429865 AOR:: 771@1.1.1.1 Contact:: sip:771@1.1.1.1:5070 Q= Expires:: 105 Callid:: 3e946958322b1e2d6bfa564d46bf8...@testers.com Cseq:: 133 User-agent:: Asterisk PBX State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:1.1.1.1:5060 Methods:: 4294967295 Ruid:: uloc-53bfe447-35b0-708 Reg-Id:: 0 Last-Keepalive:: 1405429980 Last-Modified:: 1405429980 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. In my setup clients authenticate with Kamailio and Kamailio sends a REGISTER to Asterisk according to guide I used: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb How would I fix this double-AOR problem, can it be fixed on Asterisk configuration? thanks, Olli 2014-07-15 16:00 GMT+03:00 Joshua Colp <jc...@digium.com>: > Olli Heiskanen wrote: > >> Hello, >> >> Thanks for your response, I actually verified that the Zoiper setting is >> not the reason for Asterisk to start sending REGISTERs, it only looked >> like it as I checked the Kamailio output before Asterisk sent the first >> REGISTER to Kamailio, right after I had played with that setting. >> (sorry, my bad!) >> >> However, _something_ is causing these REGISTERs, here's an example of a >> REGISTER message sent from Asterisk to Kamailio: >> >> REGISTER sip:testers.com <http://testers.com> SIP/2.0 >> >> Via: SIP/2.0/UDP my_ip:5070;branch=z9hG4bK7477f754;rport >> Max-Forwards: 70 >> From: <sip:771@my_ip>;tag=as7a88c4c6 >> To: <sip:771@my_ip> >> Call-ID: 3e946958322b1e2d6bfa564d46bf8...@testers.com >> <mailto:3e946958322b1e2d6bfa564d46bf8...@testers.com> >> >> CSeq: 121 REGISTER >> User-Agent: Asterisk PBX >> Expires: 120 >> Contact: <sip:771@91.221.66.61:5070 >> <http://sip:771@91.221.66.61:5070>> >> >> Content-Length: 0 >> >> Is there any other reason - other than client settings - why this would >> happen? >> > > 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? > > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users