Hi,
I have used asterisk. But it's not so stabel. Also you'll have codec
limitations when you start playing from asterisk.
Prashant.
----- Original Message -----
From: "Gentian Bajraktari" <g.bajraktari@xxxxxxxxxx>
To: <openh323gk-users@xxxxxxxxxxxxxxxxxxxxx>
Sent: Wednesday, June 29, 2005 7:29 PM
Subject: Re: PBX
Forget it, use Asterisk for PBX.....
----- Original Message -----
From: "Ramón Montoya Benito" <rmbtelecom@xxxxxxxxxxx>
To: <undisclosed-recipients:>
Sent: Wednesday, June 29, 2005 9:18 AM
Subject: PBX
Hi all!
Al right; I'm trying to implement some PBX abilities to gnugk; but, in
fact, I cannot solve this really right.
The most important thing I'll try first it will be connecting the calling
endpoint to an audioplayer while I find a cute destination for him. This
seems to be difficult.
Let's see: if I used the proxy mode, I can manage everything between
endpoints, but... whenever I wanted to make any change e.g. in addresses,
I must be changing it constantly, (for any H323 message), or endpoints may
fail the call; otherwise, if I connected this calling endpoint to this
audioplayer, and I stored its H245, H225 and Q931 messages, I could see if
I may connect them after both "connect" messages, but... I must implement
a complete H323 protocol for each tried called endpoint.
All this could have been really easier if you had implemented
SoftPBX::MakeCall(...).
Anyway, I found another solution, really worse: transferring calls the
whole time, and, before they hang up, transferring them again. In routed
mode, (not proxy needed). But with this method the calling endpoint never
hears a complete audio message; and, if any of these transferrences fails,
tha call is lost.
I would like to receive some ideas about this.
Thaks a lot in advance,
Ramón Montoya.
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