First of all I'd suggest to use http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb guide in combination with http://saevolgo.blogspot.com/2011/11/how-to-increasing-voip-services.html But, assuming your platform is behind NAT, you need: 1st. Use rtpengine instead of rtpproxy. You can read about how to advertise your external public adress on rtpengine git page. 2nd. In Kamailio configuration when you define listen, you should use listen - advertise construction ( http://www.kamailio.org/wiki/cookbooks/4.0.x/core#listen). 3d. Be sure to leave "secret" column empty on asterisk database, otherwise all users registered on asterisks won't have OK status, what can cause problems with queues etc.
2015-08-12 0:19 GMT+03:00 Bruno d4rkstar@gmail.com:
Hello, i'm on my first try with kamailio. I need to build a SIP balancer that should keep SIP registration from VoIP provider and route the calls to the asterisk boxes where an IVR will take care to answer.
Here's my network topology:
+---> [asterisk1][public_ip] | 10.50.10.131 [router] <---NAT---> [kamailio] <---+ 10.50.10.1 10.50.10.120 | +---> [asterisk2] 10.50.10.132
In my setup i planned to use UAC and DISPATCHER modules. I started from the "kamailio-basic.cfg" and added some extra lines to handle UAC and DISPATCHER.
All is working fine when i do a test call from a softphone inside network 10.50.10.0/24.
When a call is coming from the sip carrier, troubles occurs because asterisk boxes are sending their internal ip in SDP.
I understand that i need to rewrite SDP in that case, but i actually don't know how/where.
I've attached kamailio configuration and a sip trace taken with sngrep where the problem is visible.
For security reasons, i would like to force the RTP through RTPProxy.
I'm missing something, and need your help me to understand my errors.
Best Regards, Bruno
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