Hello,
Weird thing is the client looks registered but I'm not sure if it really is registered. If I'm not mistaken I should see the peers when I issue 'sip show peers' on asterisk cli. Instead I get this:
*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline]
Also, calling between clients will fail; in Asterisk cli I get:
*CLI>
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [661@default:1] NoOp("SIP/660-00000000", "Testing: Dialed 661") in new stack
-- Executing [661@default:2] Dial("SIP/660-00000000", "SIP/661,3600,rt") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called SIP/661
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [661@default:3] Hangup("SIP/660-00000000", "") in new stack
== Spawn extension (default, 661, 3) exited non-zero on 'SIP/660-00000000'
In this setup I have SIP peers in an asterisk table added like this:
INSERT INTO sippeers (name, defaultuser, host, sippasswd, fromuser, fromdomain) VALUES ('660', '660', 'dynamic', 'password', '660', '
testers.com');
I have Kamailio and Asterisk on the same machine where Kamailio listens port 5060 and Asterisk listens 5070. Things that differ from the guide are Kamailio and Asterisk versions, which in my case are newer. Also, for another testing case I have MULTIDOMAIN enabled in Kamailio, does this interfere with the realtime integration? I'm using only one domain though.
Please let me know if any configs or traces I can provide will help figure out what's going on.
cheers,
Olli