[OpenSIPS-Users] opensips + asterisk + queues

[Alex Balashov]

This is one of the buggiest aspects of Asterisk, and in any event,
has little to do with OpenSIPS.

On Mon, September 8, 2008 5:22 pm, Alex G wrote:
> anyone have any experience using openser and asterisk queues?
>
> no matter what i do i cannot seem to make the agent channel change from
> "Not
> in Use'" when there is a legitimate call on the channel.
>
> currently using asterisk realtime queues. have tried piping the call out
> of
> a sip channel with call-limit=1 and limit-on-peers=yes with no luck
>
> hope someone has seen this before and has a solution cuz i'm really
> stumped.
>
> Alex
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>


-- 
Alex Balashov
Evariste Systems
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