[SR-Users] Routing calls to Asterisk using dispatch

[Olli Heiskanen]

A little progress on this; the double-AOR problem was fixed; I had
callbackextension field in the realtime table, which caused extra REGISTER
messages to be sent from Asterisk to Kamailio and that messed the AORs for
my clients.

Calling between the clients is a problem though, the INVITE gets routed
correctly but for some reason, when the INVITE is returning from Asterisk
to be routed to the called party, location lookup fails. I end up with 404
and no route to destination error. Any clues on to why location lookup
would fail?

kamctl ul show gives correct looking output for both registered clients.

cheers,
Olli







2014-07-12 17:27 GMT+03:00 Olli Heiskanen <ohjelmistoarkkitehti at gmail.com>:

>
> Hello,
>
> I've started playing with an idea to add multiple asterisk servers and
> using dispatcher to balance the sip load between them. I added the code
> according to dispatcher module documentation (
> http://www.kamailio.org/docs/modules/4.2.x/modules/dispatcher.html), but
> I think there's something off in my setup:
>
> kamctl ul output shows 2 AORs for one client:
>
> AOR:: 770 at testers.com
> Contact:: sip:770 at 2.2.2.2:64340;rinstance=c634da314e12385f;transport=UDP
> Q=
>  Expires:: 3221
> Callid:: ZTE1MWYwYzM3NGNjNjMxMmEzM2JjYWNmNzQyZTdiNGI.
> Cseq:: 2
>  User-agent:: Z 3.2.21357 r21367
> State:: CS_SYNC
> Flags:: 0
>  Cflag:: 0
> Socket:: udp:1.1.1.1:5060
> Methods:: 5087
>  Ruid:: uloc-53bfe447-35ae-2a2
> Reg-Id:: 0
> Last-Keepalive:: 1405174150
>  Last-Modified:: 1405174150
> AOR:: 770 at 1.1.1.1
> Contact:: sip:770 at 1.1.1.1:5070 Q=
>  Expires:: 68
> Callid:: 327fcf07641f80006e962821112a61b5 at testers.com
>  Cseq:: 754
> User-agent:: Asterisk PBX 11.10.2
> State:: CS_SYNC
>  Flags:: 0
> Cflag:: 0
> Socket:: udp:1.1.1.1:5060
>  Methods:: 4294967295
> Ruid:: uloc-53bfe447-35af-a82
> Reg-Id:: 0
>  Last-Keepalive:: 1405174477
> Last-Modified:: 1405174477
>
>
> I don't think I should be seeing an AOR for 770 where Contact is the
> public address of my server (here 1.1.1.1) and User-Agent which is
> Asterisk.
>
> I'm using Asterisk Realtime integration, and by what I can tell the sip
> messages are going nicely, client authenticates with Kamailio and sends
> this message to Asterisk (which is on the same machine; Kamailio at 5060
> and Asterisk at 5070):
>
> 1.1.1.1.sip > 1.1.1.1.vtsas: SIP, length: 374
>         REGISTER sip:1.1.1.1:5070 SIP/2.0
>         Via: SIP/2.0/UDP
> 1.1.1.1;branch=z9hG4bKbc8a.f4473947000000000000000000000000.0
>         To: <sip:770 at 1.1.1.1>
>         From: <sip:770 at 1.1.1.1>;tag=4a9c3f1c98b9f1c5704acfd1770d93d2-d0c1
>         CSeq: 10 REGISTER
>         Call-ID: 7ffa0191-13742 at 1.1.1.1
>         Max-Forwards: 70
>         Content-Length: 0
>         Contact: <sip:770 at 1.1.1.1:5060>
>         Expires: 3600
>
>
> Currently I can make calls from 770 to 123 which is an Asterisk extension
> that answers, plays hello world and hangs up. However I can't call another
> sip clients when I route calls through Asterisk, they do work fine if I
> don't use Asterisk for handling calls, but I'd like Kamailio to be in the
> role of proxy/loadbalancer and Asterisk to handle calls.
>
> My config is the simple default config, added with realtime stuff and then
> dispatcher according to the documentation. I wonder if there's something
> wrong in the REGISTER that Kamailio sends to Asterisk, or maybe something
> else going wrong?
>
> Has anyone seen results like this and do you spot something here that
> needs fixing?
>
> Thanks,
> Olli
>
>
>
>
>
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