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@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