On Tue, Feb 18, 2014 at 10:53 PM, Gholamreza Sabery <gr.sab...@gmail.com> wrote: > Hello, a few days ago I sent a question: > > http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html > > but no one answered me! I just want to know is it possible or not?
Hi! As many others mentioned, if you don't get an answer, first go googling then try the #asterisk IRC channel, or maybe the forums at forums.asterisk.org. I noticed your first post today and was going to answer it there, before I saw this new post as well... To attempt answering your question... I believe so. The NAT section of the sip.conf sample contains a lot of helpful options, including: ;directmedia=nonat ; An additional option is to allow media path redirection ; (reinvite) but only when the peer where the media is being ; sent is known to not be behind a NAT (as the RTP core can ; determine it based on the apparent IP address the media ; arrives from). That is for chan_sip in Asterisk 11, and should also be available in Asterisk 1.8 I've not used a config with this option before, but it sounds like the intent is what you may need. A link to the sample file (that is also included with your source files) http://svnview.digium.com/svn/asterisk/branches/11/configs/sip.conf.sample?view=markup -- Rusty Newton Digium, Inc. | Community Support Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct: +1 256 428 6200 Check us out at: http://digium.com & http://asterisk.org -- _____________________________________________________________________ -- 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