[SR-Users] Help with sip balancer

[Alexandru Covalschi]

Also, take a look at kamailio-advanced.cfg, there is PSTN GW route already
included. Also you can use LCR for routing calls to different providers, a
simple guide can be found here
http://dopensource.com/least-cost-routing-with-kamailio-v4-1/

2015-08-12 0:41 GMT+03:00 Alexandru Covalschi <568691 at gmail.com>:

> 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
>>
>>
>>
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
>
> --
> Alexandru Covalschi
> ABRISS-Solutions
> VoIP engineer and system administrator
> phone: +37367398493
> web: http://abs-telecom.com/
>



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