2 msgUNICALL MFC/R2 + Asterisk 1.4
3 msgAuthenticate() application and CDR
2 msgSIP over TCP
5 msgAsterisk 1.2.13 and presence
1 msg'HiarPinning' via TDM400 in the UK ...

call limit
\ Vieri (21 Sep 2007)
. \ Nicholas Blasgen (22 Sep 2007)

4 msgAsterisk and MS Exchange 2007
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2 msgDialing an external number and then passing it ...
7 msgAsterisk 1.2.24 simultaneous call limits.
4 msgGROUP() issues for me
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Subject:Re: call limit
Group:Asterisk-users
From:Nicholas Blasgen
Date:22 Sep 2007


 


Setting call-limit=1 in sip.conf will limit the number of incomming (and
outgoing?) channels on your SIP device to the number you specifiy (1 in this
case). If you want to allow more outgoing, but only 1 incomming, you could
do that with some GROUP() checking. Problem is that when there isn't an
available channel, Asterisk will return CHANUNAVIABLE or something like
that. It's not very helpful. GROUP() checking will allow you to provide a
more informative answer to the person who's calling.

On 9/21/07, Vieri <rentorbuy> wrote:
>
> Hi,
>
> I would like to know if the following is possible:
>
> * how to accept only one call at a time on a
> particular SIP extension (softphone). I'm referring to
> incoming calls. Can it be done on the server side or
> just on the client? ie. all other incoming calls will
> just be dropped while the extension is busy. In other
> words I would like to simulate having just one phone
> line available. I tried using call-limit=1 in
> sip.conf. Is this the right way?
>
> * how to accept only one incoming call at a time for a
> whole group? If there's an active call on any one of
> the extensions, drop the other incoming calls.
>
>
>
>
>
>
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--
/Nick


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