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

[Erik Barker]

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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