[SR-Users] kamailio 4 with asterisk

[Sébastien Brice]

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maybe a bit outdated but still consistent ?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