[SR-Users] Help with sip balancer

[Alexandru Covalschi]

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 at 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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>


-- 
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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