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