Mmmm... all right, but only for last chance: do you know if that
SoftPBX::MakeCall() function could ever be implemented, or if there's
someone yet working at it?
Regards,
RMB.
From: "Gentian Bajraktari" <g.bajraktari@xxxxxxxxxx>
To: <openh323gk-users@xxxxxxxxxxxxxxxxxxxxx>
Subject: Re: PBX
Date: Wed, 29 Jun 2005 14:59:06 +0100
Reply-To: openh323gk-users@xxxxxxxxxxxxxxxxxxxxx
Forget it, use Asterisk for PBX.....
----- Original Message -----=20
From: "Ram=F3n 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, that call
is lost.
I would like to receive some ideas about this.
Thaks a lot in advance,
Ramón Montoya.
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