[asterisk-users] Moving User Agent To Remote Location

[Nick Khamis]

Oooops yes of course 10004-10007!! Simple math does not come easy
anymore... Anyhow, I singled out Opensips and I have two way audio
form UA(local) -> UA(remote) but not from UA -> Siptrunk. That being
said maybe a small diagram of the architecture. Please don't laugh!!!
:) I know having a block of static IPs would make like easier
however....

UA (Remote) -> Router (Remote) -> Internet -> Router (Local) ->
OpenSIPS+RTPProxy -> Asterisk

Port forwarding (Remote): 5060, and 10000-50000 to UA
Port Forwarding (Local): 5060. and 10000-50000 to OpenSIPS)   "No Audio"
Port Forwarding (Local): 5060. and 10000-50000 directly to Asterisk
"Two Way Audio"

Cheers Guys!

Nick

On 1/3/13, Danny Nicholas <danny at debsinc.com> wrote:
> -----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.
>
>
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