[OpenSIPS-Users] opensips + asterisk + queues

Alex G greekman0000 at gmail.com
Mon Sep 8 23:22:52 CEST 2008

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.

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