[Asterisk-Users] sip, call ransfer and call waiting

[trixter http://www.0xdecafbad.com]

On Mon, 2005-09-26 at 11:08 +0200, Daniel ANDRE wrote:
> Hello all,
> 
> I have a very basic question but I haven't found any answer.
> 
> I would like to configure asterisk so that it wil not indicate a call 
> waiting to a SIP phone if it is already on conversation (off hook). But 
> I don't want to loose call transfer, call hold and so on.
> 
> Is there any possibility to do that?

Yup...

exten => 123,1,SetGroup(user1)
exten => 123,2,CheckGroup(1) ; dont let more than 1 call at a time
exten => 123,3,Dial(sip/user1)
exten => 123,103,Busy  ; this is where it goes if CheckGroup indicates
more than X calls
...

see http://voip-info.org/wiki-Asterisk+cmd+SetGroup for more info.

You may have to play games with variables to make a macro perhaps that
would be more generic in this regard, but this should at least get you
started.


-- 
Trixter http://www.0xdecafbad.com     Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
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