One of the tests I've been working with is Asterisk realtime integration
according to Daniel's guide here:

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

Also, calling between clients will fail; in Asterisk cli I get:
  == 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.

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