[SR-Users] Realtime integration: Unregistered clients showing as registered?
Pedro Niño
nino.pedro at gmail.com
Tue Apr 22 20:06:57 CEST 2014
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" <ohjelmistoarkkitehti at 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 at default:1] NoOp("SIP/660-00000000", "Testing:
> Dialed 661") in new stack
> -- Executing [661 at 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 at 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 at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20140422/b7a159bc/attachment.html>
More information about the sr-users
mailing list