[SR-Users] Help with sip balancer

[Bruno Salzano]

Thankyou Alexandru for your suggestions.
I'll give it a try tomorrow and will report my progress here.
It seems that i'm not so far from the result!
Bruno

Il giorno mar 11 ago 2015 alle 23:44 Alexandru Covalschi <568691 at gmail.com>
ha scritto:

> 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/
> _______________________________________________
> 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
>
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