-----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker Sent: Thursday, January 03, 2013 2:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Moving User Agent To Remote Location
On 01/03/2013 02:23 PM, Markus Weiler wrote: > Am 03.01.2013 21:21, schrieb Nick Khamis: >> Oh that's so smart!!! So, if I did not misunderstand you, for this >> one call, have: >> rtpstart=10004 >> rtpend=1008 The rtpend should be 10008 and rtpstart should be 10005. A SIP call in Asterisk originates on 5060 (5061 for TLS) and then spawns two RTP channels for audio. AFAIK the odd channel is send and the even channel is receive (smarter folks than me like Tzafir can give you the specifics; this was covered at least twice in 2012 threads). If you open 5060 on your NAT/firewall, but open no RTP channels, you will establish a call with no sound. -- _____________________________________________________________________ -- 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
