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

[Anton Tonev]

Thank you, for all your replies.

I tried to use the add_path() as it is described in the Spanish tutorial
however
I am still unable to make my sip server pass through the proxy for the
second call leg (the one to the destination).

However I have one question. In the tutorial it is said that Asterisk will
use the path if Asterisk initiates a dialog. What that means ? Are these
dialogs initiated because of a 3th party call control application request
or because Asterisk receives an INVITE from some user behind the proxy and
then Asterisk initiates a dialog for the second leg of the call?

Best regards,

Anton

2016-03-01 11:41 GMT+01:00 Alberto Sagredo <alberto.sagredo at avanzada7.com>:

> 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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>>> sr-users at lists.sip-router.org
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>>>
>>>
>>
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