-----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of A J Stiles Sent: Friday, October 15, 2010 11:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP - no audio behind nat problem
On Friday 15 Oct 2010, Zarko Zivanovic wrote: > Hello, > > We have 2 grandstream GX 2000 phones behind NAT and Asterisk outside this > natted network. > > We have the issue with calls to these SIP phones - no audio. > > It is probably the problem with port forwarding on router - but I am not > sure how can I forward same sip ports (5004 to 5100) to two phones (nat > addresses?)? Simple answer, don't run SIP through NAT. Have another Asterisk server on the outside and run IAX2 through NAT instead. Much cleaner :) If you ARE going to run SIP through a NAT, you're going to need to designate a chunk of ports in rtp.conf and poke those in your firewall. We did UDP pokes for 10001-10004 to use 1 line. When you do an Asterisk communication, the "handshake" occurs on 5060; the actual call occurs on 2-4 RTP ports usually in the 10000 range. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
