OK,
I've fixed the '#' transfer problem. We setup a macro for dialing staff
extensions, however, it was missing the 'tr' options on the Dial
application:
[macro-staff-extension]
; Macro for Staff Extensions
exten = s,1,Dial(${ARG2},20,tr) --
exten = s,2,Voicemail(u${ARG1})
exten = s,102,Voicemail(b${ARG1})
exten = s,103,Hangup
I added the 'tr' and we can now perform call transfers while preserving
the correct CallerID information.
Thanks,
--
Erik Barker
Sr. Systems Engineer
NetNation Communications Inc.
http://www.netnation.com | http://www.domainpeople.com
On Tue, 2004-04-20 at 03:16, David Liu wrote:
Hi Erik,
Can you post your dial plan from incoming PSTN to the receptionist?
David
- Original Message -
From: Erik Barker [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
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 [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
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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