[asterisk-dev] app_dial.c / stream calls help required

[Tony Howat]

Kevin P. Fleming <kpfleming at digium.com> wrote:
>----- Tony Howat <arhowat at hotmail.com> wrote:
> > Could someone provide some assistance here? I'm working on 1.2.9.1 but > 
>will port and submit any improvements to the trunk.
>
>Native file music on hold with random mode disabled will do exactly what 
>you want; if the MOH class contains only one file, each time that class is 
>played onto a channel, it will start from the  beginning of the file. There 
>should not be any code changes necessary.

Thanks Kevin, it's an option but not quite what I need. I would need to set 
up these MOH channels dynamically and quickly as the backend application 
specifies the sample to play whilst dialling in XML responses and the whole 
thing can be configured on the fly. The thought of rewriting 
musiconhold.conf and reloading the config on demand via AMI is ugly given 
I'm aiming at processing 800+ calls per min on a single box (I'm managing 
600/min at the moment with some heavy optimisation of my fastagi). I'm 
trying to avoid bodges as this needs to be a rock solid system.

Could someone take a look at my queries concerning 
ast_openstream/applystream/playstream in app_dial n the original message 
which is reproduced here :

http://www.i-r-genius.com/astissue.txt

...or asterisk-dev digest vol 25 issue 50 article 6.

It'd help me considerably, and as I've said I'm keen to contribute - a 
starter on this could lead to all sorts of interesting stuff :)

--
Tony