Re: PBX

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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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