[Asterisk-Users] Already on the phone?

Paul Liew pliew at atp.org.au
Wed Oct 29 15:02:49 MST 2003


Michael,

Thanks for the feedback, I'll check it out and let you know...

Paul
----- Original Message ----- 
From: "Michael Ulitskiy" <mulitskiy at acedsl.com>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, October 30, 2003 6:17 AM
Subject: Re: [Asterisk-Users] Already on the phone?


> 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
> > >
> >
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
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