Hi everyone
I did follow the tutorial Kamailio 4.0.x and asterisk written by Daniel-C and I
need a hand :-)
after a successful install (kamailio and asterisk on the same server), i did
setup 2 sip extensions (102 and 103) and tryed to place a call between them 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/101-00000001", "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]
Actually none of my extensions are online and i am wondering if Kamailio
forward register with asterisk ?
transport between K and Asterisk seems ok (at least i did setup bindport in
both sip.conf and kamailio.cfg)
apart that my sip.conf and extensions.conf are very minimal:
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 am a bit new with Kamailio and I don't know if this behavior is normal since
i have a sipregs mysql table that could do the job on purpose?
Could someone point me to the right direction and enlight my knowing, thanks you
thx you
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