[Asterisk-Dev] Not for 1.2, was SIP RTP JitterBuffer in Asterisk - now aimed for 1.3 dev

Beau Hargis beauh at bluefrogmobile.com
Thu Aug 25 16:17:42 MST 2005


I just subscribed to the list and posted up a bug for AEL, but I can say
that it is not yet production worthy. It has some issues with its
parsing of various things (especially if-else) that are not obvious to
anyone trying to use it. It works well enough for me to create some
complex things with it, but I am paying close attention to its
limitations. 

I do have to say that AEL is mandatory as far as my work is concerned.
The way that extensions and flow is defined in extensions.conf is not
suitable for creating and maintaining large IVR applications, which I am
currently working on.

I will be hammering away at AEL and making it work for me, but I would
not consider it stable at all.

On Thu, 2005-08-25 at 15:37 -0500, Tilghman Lesher wrote:

> Speaking of which, has there been any discussion on postponing AEL into
> 1.3?  It looks like something that will be a great tool in the future,
> but it still seems to be a bit too beta for stable.
> 



More information about the asterisk-dev mailing list