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
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