asterisk users - Jan 2006 - conditional canreinvite

Hi, I have asterisk on public IP and phones in two locations behind 
firewall/nat,
- when I have nat=yes and canreinvite=no, this is working fine, but rtp 
stream must go _always_ through asterisk, even if phones talk inside 
their locations
- when I have nat=yes and canreinvite=yes, phones can speak only inside 
their location and rtp stream is connected directly between phones (this 
is, imho, correct and logical), but,
is possible to combine both, so do reinvite only "within" e.g. one 
context and disable reinvite when connecting phones between two context,
or any better option exist/planned how to solve?
thanks
PJ

Great posting.  I'm keen to learn the answer as well.

Hugh
---------- Forwarded message ----------
From: Pavel Jezek <pavel.jezek@i.cz>
Date: Jan 12, 2006 9:33 AM
Subject: [Asterisk-Users] conditional canreinvite
To: asterisk-users@lists.digium.com


Hi, I have asterisk on public IP and phones in two locations behind
firewall/nat,
- when I have nat=yes and canreinvite=no, this is working fine, but rtp
stream must go _always_ through asterisk, even if phones talk inside
their locations
- when I have nat=yes and canreinvite=yes, phones can speak only inside
their location and rtp stream is connected directly between phones (this
is, imho, correct and logical), but,
is possible to combine both, so do reinvite only "within" e.g. one
context and disable reinvite when connecting phones between two context,
or any better option exist/planned how to solve?
thanks
PJ
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That is the way way SER works. I too am very interested to know if
this can be done with Asterisk.

David

On 1/12/06, Pavel Jezek <pavel.jezek@i.cz> wrote:
> Hi, I have asterisk on public IP and phones in two locations behind
> firewall/nat,
> - when I have nat=yes and canreinvite=no, this is working fine, but rtp
> stream must go _always_ through asterisk, even if phones talk inside
> their locations
> - when I have nat=yes and canreinvite=yes, phones can speak only inside
> their location and rtp stream is connected directly between phones (this
> is, imho, correct and logical), but,
> is possible to combine both, so do reinvite only "within" e.g.
one
> context and disable reinvite when connecting phones between two context,
> or any better option exist/planned how to solve?
> thanks
> PJ
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> Asterisk-Users mailing list
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