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 type 'SIP'
(cause 20 - Subscriber absent)
if I take a look in the asterisk CLI, I have that:
rtpproxy1*CLI> sip show peers
Name/username Host Dyn
Forcerport ACL Port Status Description
Realtime
kamailio (Unspecified)
a A 0 Unmonitored
siptest2/siptest2 (Unspecified) D
0 Unmonitored
Cached RT
testteopad2/testteopad2 (Unspecified) D
0 Unmonitored
Cached RT
3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 3 offline]
rtpproxy1*CLI> sip show registry
Host dnsmgr Username Refresh
State Reg.Time
0 SIP registrations.
And I'm not sure I understand correctly the line "Be sure you
configure Asterisk to not authenticate SIP requests coming from
Kamailio." in the tutorial:
I've tried to add in sip.conf these lines:
[kamailio]
type=friend
permit=192.168.14.0/24
secret=
transport=tcp
outboundproxy=192.168.14.25
(Kamailio is 192.168.14.25)
Thanks for your help
Christophe
2012/11/15 Olle E. Johansson <o...@edvina.net>
>
>
> 15 nov 2012 kl. 11:58 skrev Christophe ROY <christophe.roy.tha...@gmail.com>:
>
> 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 authentication with Kamailio
> - I had problems with Asterisk 10.7 (problems with chan_sip module crashing)
> so I've installed Asterisk 11 on another VM
> - we have high-availability with 2 Kamailio servers, with Kamailio listening
> on TCP (constraint from our SSL gateway in front of Kamailio) on a "virtual
> IP" (created by keep-alive): this VIP is not visible with ifconfig, but you
> can see it with the command "ip addr sh eth0"
>
> For now, we use Linphone on Windows as SIP clients to test.
> If I don't define WITH_ASTERISK, calls work, I can call some...@domain.tld
> However, if I define WITH_ASTERISK, calls fail (even with destination
> registered and available) and I have these errors in the logfile:
>
> Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [ut.h:333]: no
> corresponding socket for af 2
> Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:424]:
> ERROR: can't fwd to af 2, proto 1 (no corresponding listening socket)
>
>
> Seems like Kamailio and ASterisk is not using the same transports. Check
> sip.conf in Asterisk so that you enable the proper transports
> that you are using for forwarding. If Asterisk is ONLY listening to udp, add
> ";transport=udp" to the forwarding URI. To force TCP, use
> "transport=tcp".
>
> Now since the error message indicates proto 1, which in Kamailio-speak is
> UDP, it seems like you have an issue with that.
>
> Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm
> [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches
> Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: INFO: <script>: reply
> error
> Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: sl
> [sl_funcs.c:371]: ERROR: sl_reply_error used: I'm terribly sorry, server
> error occurred (1/SL)
>
> It seems to happen on the if (!t_relay()) line in ROUTE[RELAY]
>
> 192.168.14.25 is the real IP of the Kamailio server,
> 192.168.14.24 is the VIP of the Kamailio "cluster"
> 192.168.14.28 is the IP of the Mysql server
> 192.168.14.32 is the IP of the Asterisk server
>
> I can't find why the relay doesn't work. I've tried to bypass the VIP and
> have Kamailio listen on the real IP, but it still doesn't work: I don't seem
> to have the same errors as above, but I don't see any traffic between
> Kamailio and Asterisk.
>
> What could be the problem? Thanks for your help
>
> If you forward register to Asterisk, you have to configure outboundproxy in
> sip.conf in asterisk so that you get messages back from Asterisk. Or use one
> of my branchces with support for the SIP Path header in Asterisk (using the
> PATH module in Kamailio).
>
> Using the onsend route you can check IP, port and transport used to deliver a
> message from Kamailio. CHeck the Kamailio cookbook on the wiki for more
> information about that.
>
> /O
>
>
> --
> * Olle E. Johansson - o...@edvina.net
> * Kamailio & SIP Masterclass Miami FL December 2012
> * http://edvina.net/training/
>
>
>
>
>
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