[SR-Users] Kamailio as a front-end proxy

[Alberto Sagredo]

You could find something related also on this link

Its in spanish

https://blog.irontec.com/integracion-kamailio-y-asterisk-con-path/



2016-03-01 11:25 GMT+01:00 Jurijs Ivolga <jurij.ivo at gmail.com>:

> Hi,
>
> I would recommend you to take a look on path module:
>
> http://kamailio.org/docs/modules/1.4.x/path.html
> I think this is what you need.
>
> With kind regards,
>
> Jurijs
>
> 2016-03-01 12:02 GMT+02:00 Anton Tonev <anton.tonev at gmail.com>:
>
>> Hello everybody,
>>
>> I am a new user of Kamailio (4.3.1), I am working with it since 1-2
>> months. The thing that I'm trying to do is to build the following system:
>>
>> same LAN
>>
>> 192.168.0.1
>> Alice
>> proprietary SIP Server
>>                          [Public_IP_X]   ------------ [Public_IP_Y]
>> Kamailio [172.26.0.1] ---------- [172.26.0.1]  with
>> 192.168.0.1
>> registrar
>> Bob
>>
>> Obviously Kamailio has to translate the local addresses of Alice and Bob,
>> e.g. to use the Nathelper module.
>> The module is doing well its job because the Contact headers are replaced
>> with the Public_IP_X when a REGISTER message is sent by Alice's or Bob's
>> sip phones (I am using Linphone and Zoiper as clients).
>> Once the incoming sip register was treated by Kamailio it is sent to the
>> proprietary SIP Server. The server sends 200 OK to Kamailio and the proxy
>> relays the message to the clients. So the sip registration for me it is OK.
>>
>> But when it comes to initiate a call from Alice to Bob the things are not
>> as I expect it. The initial request INVITE sent from Alice goes to the sip
>> server but then the server instead of sending the INVITE for Bob through
>> Kamailio, it sends the message directly to Bob's device.
>> Does anyone knows how to "tell" to the sip server, using the SIP
>> protocol, that it must use the proxy?
>> The only thing I have in mind is to force Kamailio to replace the contact
>> of Alice and more precisely the host/ip address by the proxy's host/ip
>> address.
>> I tested this idea and the sip server did what I was expecting but for me
>> this is not a proper solution.
>> To do that I used this discussion -
>> http://opensips.org/pipermail/users/2010-October/014873.html
>> Thank you in advance for your attention !
>>
>> Best regards,
>>
>> Anton
>>
>>
>>
>>
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>>
>>
>
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