[Asterisk-Users] TCP/UDP Ports used by Asterisk

[Dan]

Hi Gary,

> the SIP is using asterisk as a proxy, so therefore is wont/cant
> handoff.
Then how can it make codec conversion?
I have a Cisco 7960 hardware SIP phone (with G.711) and an X-Lite (with
GSM)... and they can talk each other through Asterisk..

Dan

----- Original Message ----- 
From: "Gary" <gary at ausmail.com>
To: <asterisk-users at lists.digium.com>
Sent: Saturday, May 24, 2003 7:05 PM
Subject: Re: [Asterisk-Users] TCP/UDP Ports used by Asterisk


> from experience...
>
> the SIP is using asterisk as a proxy, so therefore is wont/cant
> handoff.
>
>
>
> On Sat, 24 May 2003 18:54:34 +0300, Dan wrote:
>
> >Hi all,
> >
> >I have my Asterisk behind a NAT router, but now it is configured to put
that specific computer in DMZ (directly exposed to Internet).
> >I intend to disable this and to open just the used ports.
> >There is a list of TCP/UDP ports usd by Asterisk in order to connect to
the outside world?
> >
> >One more question: When a call is established between an internal SIP
phone (in LAN) and a phone from another place outside my router/firewall,
using both the same codec (no conversion)... the call is still routed
through  the PBX or the PBX is used only for signaling and then a direct
connection between the two phones is established?
> >
> >I ask this because if the audio stream is passed through the PBX then
there is no need to open other ports on the firewall for the internal
phones.
> >
> >Thanks,
> >Dan
>
> .
>
>
>
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