this one (written by Daniel)
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
maybe a bit outdated but still consistent ?
the thing i am really stuck with (and concerning real-time) is that none of my
extensions (from asterisk CLI) are online:
ns3325046*CLI> sip show peers
Name/username Host Dyn
Forcerport Comedia ACL Port Status Description
Realtime
102/102 (Unspecified) D Auto (No)
No 0 Unmonitored Cached RT
103/103 (Unspecified) D Auto (No)
No 0 Unmonitored Cached RT
2 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 2 offline]
that drives asterisk crazy ! and logger reports: app_dial.c:2411
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber
absent) every time i place a call.
The tutorial written by daniel mention a channel configuration pretty minimal:
INSERT INTO sipusers (name, defaultuser, host, sippasswd, fromuser, fromdomain,
mailbox)
VALUES ('102', '102', 'dynamic', '102', '102', 'yoursip.com', '102');
and since there's no context associated to the 102 extension i cant figure out
where that channel enter the dialplan ? [public] [LocalSet] [default] ????
and a dialplan
exten => _1XX,1,Dial(SIP/${EXTEN})
exten => _1XX,n,Voicemail(${EXTEN},u)
exten => _1XX,n,Hangup
exten => _1XX,101,Voicemail(${EXTEN},b)
exten => _1XX,102,Hangup
i am sorry to bother you with issues more asterisk oriented than kamailio.
By the way i took a good start with kamailio as it seems to work flawlessly on
my system.
thx you.
On Mon, Feb 15, 2016 at 12:26:06PM +0100, Sébastien Brice wrote:
> Hi Everyone, i like the way this tutorial explain asterisk and kamailio
> integration.
Which tutorial?
> the only thing i missed is asterisk behaviors'r regarding sip registration ?
That was a part of a tutorial I once saw. In essence asterisk uses the
kamailio database, UA registers on kamailio and is stored there,
asterisk sees the same data (realtime).
Sébastien BRICE VoIP, Support et Intégration
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