If that helps, here's the debug seen from asterisk when I try to call
from testteop...@domain.corp to sipte...@domain.corp: (.188 is the IP
of testteopad2 and .181 the ip of siptest2)
<--- SIP read from TCP:192.168.14.25:44622 --->
INVITE sip:siptest2@192.168.14.25 SIP/2.0
Record-Route:
Via: SIP
Thanks Olle, it helped a lot
Now, calls come through Asterisk and voicemail is working but it's
"working too well" ;)
When I try to call someone, Asterisk tells me that the subscriber is
absent and I'm sent directly to voicemail:
app_dial.c:2433 dial_exec_full: Unable to create channel of typ
15 nov 2012 kl. 11:58 skrev Christophe ROY :
> Hi everyone
>
> I'm trying to integrate Asterisk with Kamailio for voicemail.
> I tried to follow this tutorial:
> http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
> BUT:
>
> - I had to adapt it because I use LDAP authen
