Hello Richard,

thank you for the reply. My comments are below between the lines.

2016-03-01 23:09 GMT+01:00 Richard Brady <rnbrady@gmail.com>:
You have several options:

1. If your SIP server supports an "outbound proxy" configuration, set that parameter to the IP:port of Kamailio, then it will send all INVITEs to Kamailio. You will need to configure RR and you might still need some mangling in Kamailio to work around NAT, depending on your SIP client run by Alice and Bob.


A: Unfortunately my server does not have such option.

 
2. If you can't set outbound proxy then you could use Kamailio to edit the contact header.  Again you will need to configure RR and you might still need some mangling in Kamailio to work around NAT, depending on your SIP client run by Alice and Bob.


A: When you say here, edit the contact header. Do you mean that Kamailio has to replace the IP/host by its ip/host ? For example if the caller has the following contact 1001@192.168.0.0.1 and  Kamailio has a public_IP_Kamailio, than the replacement should result as 1001@public_IP_Kamailio ?

At his stage of my work, Kamailio is configured and has the ability to detect that the caller is NATed. For the moment when Kamailio detects this situation, it changes the contact header of the caller by replacing his local ip by the public ip of caller's network.


3. Only if you have multiple Kamailio proxies sharing one SIP server and the SIP server supports RFC3327 (i.e. the Path header) then you could could try get this to work with the Kamailio path module. This is the most difficult approach and the least likely to work in my opinion.


A: At the moment I am not very sure but I think that my sip server does not implement the Path header.

 
So if you can choose an option then we can try to help you with that. 

Regards,
Richard


Best regards,

Anton


 
On 1 March 2016 at 16:41, Anton Tonev <anton.tonev@gmail.com> wrote:
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@avanzada7.com>:
You could find something related also on this link

Its in spanish 




2016-03-01 11:25 GMT+01:00 Jurijs Ivolga <jurij.ivo@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@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.
Thank you in advance for your attention !

Best regards,

Anton




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