Hello all, I am having problems with a D-Link Horstbox Professional DVA-G3342SB which is behind a router and is not receiving any calls when registered in OpenSIPs; when I register him on a Asterisk it does receive calls.
My 2 different configurations are: a) Asterisk 1.4.26.2 b) Opensips 1.6.4 + Cisco Media GW As you can see in the pastebin - http://pastebin.com/LXPf3mXe , in both Request of Register by the client (Asterisk and OpenSIPs) the source-port is different then the Via-port ( e.g. in Asterisk source-port=1024 and via-port=1026 ), which by itself is very strange but Asterisk handles well the situation. When placing a call to the client ( first Invite in the pastebin ) it uses the same IP:port for both placing the SIP-message and the Via-header delivering correctly the call. In OpenSIPs: since the ports are different between the source and the via-header, OpenSIPs identifies the user as behind NAT ( which is true ). When placing a call to the client (the Last Invite on the Pastebin ) the via and the destination-port are different because it uses both contact and the received information from DB to generate the SIP Request and the call never gets to the client. So for the question to maybe solve my problem: Is it possible to clone the Asterisk-behaviour, i.e. compare both via-IP and destination-IP and if they are equal just change the via-port with the destination-port? Or by any reason I am thinking this all wrong and should rethink this? Thanks for your time - Best Regards, Joel Oliveira -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Change-Via-port-based-on-Destination-port-tp6454274p6454274.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users