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 ohjelmistoarkkite...@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:: 7...@testers.com
Contact:: sip:770@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@1.1.1.1
Contact:: sip:770@1.1.1.1:5070 Q=
Expires:: 68
Callid:: 327fcf07641f80006e962821112a6...@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.f4473947.0
To: sip:770@1.1.1.1
From: sip:770@1.1.1.1;tag=4a9c3f1c98b9f1c5704acfd1770d93d2-d0c1
CSeq: 10 REGISTER
Call-ID: 7ffa0191-13742@1.1.1.1
Max-Forwards: 70
Content-Length: 0
Contact: sip:770@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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