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@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:: 327fcf07641f80006e962821112a61b5@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@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
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@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@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:: 327fcf07641f80006e962821112a61b5@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@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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Olli Heiskanen