[Asterisk-Users] Already on the phone?

[Michael Ulitskiy]

Paul,

Thanks. Unfortunately your patch doesn't work reliably.
I specified incominiglimit=1 and placed a call to extension 12125551111.
Now I have:
pbx1*CLI> sip show inuse
Username        incoming        Limit           outgoing        Limit
12125550011     0               N/A             0               N/A
12125559999     0               N/A             0               N/A
12125552222     0               N/A             0               N/A
12125550029     0               N/A             0               N/A
12125550012     0               N/A             0               N/A
12125551111     1               1               0               N/A
12125550028     0               N/A             0               N/A
12125550014     0               N/A             0               N/A

I put it on hold and placed a few other calls. Then I see:
pbx1*CLI> sip show inuse
Username        incoming        Limit           outgoing        Limit
12125550011     0               N/A             0               N/A
12125559999     0               N/A             0               N/A
12125552222     0               N/A             0               N/A
12125550029     0               N/A             0               N/A
12125550012     0               N/A             0               N/A
12125551111     0               1               0               N/A
12125550028     0               N/A             0               N/A
12125550014     0               N/A             0               N/A

So it looses status of existing call somehow. Now callwaiting is
there again. It seems that the status is lost after calling chanisavail
application, although I'm not sure about that.
Also if I can make a suggestion it would be great not to have
incominglimit set statically per client, but have an application
to change it from dialplan (have no idea how hard it is to implement).
If there are other ways to check if the line is already in use or
turn on/off callwaiting on SIP clients, that would also be very
nice and desirable feature.
Thanks.

Michael

On Tuesday 28 October 2003 07:20 pm, Paul Liew wrote:
> Michael,
> 
> I've added a patch a week ago on to bugtracker to fix this - feel free to
> try it and let me know
> 
> http://bugs.digium.com/bug_view_page.php?bug_id=0000408
> 
> Paul
> ----- Original Message ----- 
> From: "Michael Ulitskiy" <mulitskiy at acedsl.com>
> To: <asterisk-users at lists.digium.com>
> Sent: Wednesday, October 29, 2003 10:31 AM
> Subject: [Asterisk-Users] Already on the phone?
> 
> 
> > Hi,
> >
> > I'm wondering if there's a way within a dialplan or AGI to find out
> > if an extension (SIP client) is already in use and the
> > person is already on the phone?
> > By default the channel is assumed available and callwaiting tone
> > is transmitted to the called extension. AFAIK there's no way to turn
> > off callwaiting from within the dialplan.
> > I need to avoid the callwaiting behavior in some cases and pass the
> > call to another extension if called extension is already in use. Is this
> > possible with asterisk?
> > I've tried chanisavail application, but since callwaiting is enabled it
> > always returns true.
> > Thanks.
> >
> > Michael
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>