<div dir="ltr">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...<br>
<br>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 :(<br>
<br><div class="gmail_quote">On Tue, Sep 9, 2008 at 3:46 AM, Iñaki Baz Castillo <span dir="ltr"><<a href="mailto:ibc@in.ilimit.es">ibc@in.ilimit.es</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
El Monday 08 September 2008 23:22:52 Alex G escribió:<br>
<div class="Ih2E3d">> no matter what i do i cannot seem to make the agent channel change from<br>
> "Not in Use'" when there is a legitimate call on the channel.<br>
<br>
</div>If the user is an user of OpenSIPS then Asterisk has no way to know if he is<br>
currently in a call. Asterisk just knows about him own local users.<br>
<font color="#888888"><br>
--<br>
Iñaki Baz Castillo<br>
<a href="mailto:ibc@in.ilimit.es">ibc@in.ilimit.es</a><br>
</font><div><div></div><div class="Wj3C7c"><br>
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