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

[Daniel ANDRE]

trixter http://www.0xdecafbad.com a écrit :

>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.
>  
>
Thank you for this pointer.

 I have seen that tere is a but in current stable 
(http://bugs.digium.com/bug_view_page.php?bug_id=0003067). In the bug 
report there is a reference to group categories. If I don't use 
categories do I need the patch?

Regards,

Daniel