Tue Sep 5 14:32:44 MST 2006
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.
----- 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
> I have 2 issues which I need to resolve on our production Asterisk
> 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."
> 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?
> Erik Barker
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
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