[Asterisk-Users] Limiting incoming SIP calls & OriginalCallerID on transfer

[David Liu]

Hi Erik,

Can you post your dial plan from incoming PSTN to the receptionist?

David

----- Original Message ----- 
From: "Erik Barker" <erikb at netnation.com>
To: <asterisk-users at lists.digium.com>
Sent: Wednesday, April 21, 2004 4:37 AM
Subject: Re: [Asterisk-Users] Limiting incoming SIP calls & OriginalCallerID
on transfer


> Thanks for the info David,
>
> I'll look at getting the '#' transfer option working again.... I had it
> working at some point where we used it to park calls, however, it does
> not appear to work anymore.
>
>
> -- 
> Erik Barker
>
> On Mon, 2004-04-19 at 11:13, David Liu wrote:
> > Hi Erik,
> >
> > >From my experience with Polycom phones, I can answer you on your
TRANSFER
> > and Caller ID issue.  For Polycom, the transfer behavior is consultation
> > transfer.  In consultation transfer mode, the caller ID of the
transferer is
> > passed to the ringing extension.  To actually pass the caller ID of the
> > incoming caller on the PSTN, you would want to do a blind transfer.  So
far,
> > I have only figured to use the Asterisk transfer option # to do blind
> > transfer.  And this assumes you have the t option enabled on the dial
plan
> > to the receptionist.
> >
> > Hope this helps.
> > David
> > ----- Original Message ----- 
> > From: "Erik Barker" <erikb at netnation.com>
> > To: <asterisk-users at lists.digium.com>
> > Sent: Tuesday, April 20, 2004 6:19 PM
> > Subject: [Asterisk-Users] Limiting incoming SIP calls & Original
CallerID on
> > transfer
> >
> >
> > > I have 2 issues which I need to resolve on our production Asterisk
> > > server:
> > >
> > >
> > > We are currently using Polycom IP600 VOIP phones for our office which
> > > are capable of handling 2 calls per SIP registration. What we're
finding
> > > is when staff are on the phone, Asterisk will pass them a second call
> > > which will show up on their display, and an audible beep is heard over
> > > the phone (regular call waiting). I would like to limit the number of
> > > calls sent to each phone to 1 call only; otherwise respond as being
> > > busy. I have looked at trying to accomplish this in the sip.conf by
> > > using the 'incominglimit' and 'outgoinglimit' parameters, however, the
> > > only one that *seems* to work is the 'incominglimit'. This prevents
> > > further calls from reaching the phones, rings busy, but does not allow
> > > our phones to initiate a 2nd call OR transfer their existing call. The
> > > 'outgoinglimit' parameter does not seem to have any effect on limiting
> > > whatsoever. Is there a way to limit calls passed to the phones from
> > > Asterisk and also allow each phone to initiate 2 calls or transfer
calls
> > > (disable call waiting)??
> > >
> > > I have also looked at the WIKI for the parameters listed above and it
> > > *appears* that 'outgoinglimit' should do what I want, however it also
> > > states that this function has been disabled??
> > >
> > > "The _outgoinglimit__ is currently disabled in the source code of the
> > > SIP channel."
> > >
> >
http://www.voip-info.org/tiki-index.php?page=Asterisk%20sip%20incominglimit
> > >
> > >
> > >
> > > My second problem is that when external calls are transferred by our
> > > receptionist to other staff members, the CallerID of course changes to
> > > her Name instead of the original caller. Is there a way (in the
> > > extensions logic or other) to preserve this CallerID information so
that
> > > staff members receive calls with the proper CallerID information?
> > >
> > >
> > > Thanks,
> > >
> > >
> > > -- 
> > > Erik Barker
> > >
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