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.sagr...@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....@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.to...@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 >>> >>> >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users@lists.sip-router.org >>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.org >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >> >> > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >
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