Don't forget to include peer type (friend), and The callbacknumber In The table.
It happened to me and asterisk/kamailio behavior was wayyy to weird until made sure both parameters were there. ----- 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 '); ------ El abr 19, 2014 1:17 PM, "Olli Heiskanen" <ohjelmistoarkkite...@gmail.com> escribió: > > Hello, > > One of the tests I've been working with is Asterisk realtime integration > according to Daniel's guide here: > http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb > > 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 > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >
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