[asterisk-users] Moving User Agent To Remote Location

[Danny Nicholas]

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at 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.