Hello, Gracias Pedro, kiitos Mikko.
It's good to know I have configured Kamailio correctly. I added the type into my table but so far no luck having asterisk see the clients registered, at least on cli. I do see that asterisk adds registration data into the table. I'll work on this for a bit and ask in the asterisk list on more tricks on asterisk side. I'll post back here if I find out what the problem was, in case someone is having similar issues. Thanks again, Olli 2014-04-22 21:06 GMT+03:00 Pedro Niño <nino.pe...@gmail.com>: > 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 >> >> > _______________________________________________ > 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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