23 Apr
2014
23 Apr
'14
8:59 a.m.
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(a)gmail.com>om>:
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(a)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(a)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(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users