[SR-Users] kamailio 4 with asterisk

[Sébastien Brice]

Hi Everyone, i like the way this tutorial explain asterisk and kamailio integration.

the only thing i missed is asterisk behaviors'r regarding sip registration ?

I tryed to place a call between 102 and 103 extensions and experimenting an issue

Asterisk tells me that the subscriber is absent and I'm sent directly to voicemail !

 -- Executing [103 at public:1] Dial("SIP/102-00000001", "SIP/103") in new stack
[Feb 14 21:00:15] WARNING[19444][C-00000001]: app_dial.c:2411 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)

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]

apart that my sip.conf and extensions.conf are very minimal:

[LocalSets]
exten => _1XX,1,Dial(SIP/${EXTEN})
exten => _1XX,n,Voicemail(${EXTEN},u)
exten => _1XX,n,Hangup
exten => _1XX,102,Voicemail(${EXTEN},b)
exten => _1XX,103,Hangup

[general]
context=LocalSets                 ; Default context for incoming calls. Defaults to 'default'
rtcachefriends=yes             ; Cache realtime friends by adding them to the internal list

i did INSERT TO the users in mysql tables (sipusers, sipregs and voicemail) and registering extensions from UA works ok (i am using jitsi)

Whats wrong with my setup ?

thank you