[OpenSIPS-Users] opensips + asterisk + queues

Alex G greekman0000 at gmail.com
Tue Sep 9 15:38:17 CEST 2008

yeah thats what i figured.. i've tried to fool asterisk by sending calls
from queues out through named trunks for each queue user, but looking at the
sql queries * is still looking to match the channel (the full channel name)
to the sip user table where as when its local users using their own sip
profile, it looks to match only the sip user name...

i looked at * 1.6 and there is a new feature in queues that allow you to
designate the sip channel from sip users that u want to use for channel
state updates. Am going to install today and try, this looks promising. If i
have no luck though looks like this is a huge roadblock in using opensips as
a reg point for extensions when there are queues involved :(

On Tue, Sep 9, 2008 at 3:46 AM, Iñaki Baz Castillo <ibc at in.ilimit.es> wrote:

> El Monday 08 September 2008 23:22:52 Alex G escribió:
> > 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.
> If the user is an user of OpenSIPS then Asterisk has no way to know if he
> is
> currently in a call. Asterisk just knows about him own local users.
> --
> Iñaki Baz Castillo
> ibc at in.ilimit.es
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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