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.pedro@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.

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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');

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El abr 19, 2014 1:17 PM, "Olli Heiskanen" <ohjelmistoarkkitehti@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

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