I'm working on a solution for myself to give different people different
extensions to reach me; I'm off site quite a bit and want these extensions
to fwd to my cell phone when I have call forwarding on at my desk.
I want to change the caller id sent to reflect the extension dialed, or a
specific caller id - NOT the original callers caller id - i.e. I want to
config in my extensions.conf a Dial/.../callerid=123-456
Is there a way to do this?
It looks relatively easy to patch Dial to do it, however I'm not sure I
follow where all the caller id stuff is being stored/retrieved from and want
to make sure I'm not patching something that can already be done.
Thanks,
Steve Radich - Colocation / Virtual Dedicated / Dedicated Servers
BitShop, Inc. - http://www.bitshop.com - $149/month colo special
-----Original Message-----
From: Steve Radich [mailto:stever at bitshop.com]
Sent: Saturday, February 22, 2003 3:13 PM
To: 'asterisk-users at lists.digium.com'
Subject: [Asterisk-Users] Agressive Echo Cancel Problem..
First let me say the new aggressive echo cancel seems to work wonders.
However in testing I tried a transfer and when I pressed flash on the phone
the caller experienced a horrible squealing sound they said. I transferred
to another phone I could reach, hit flash again to join the calls and heard
this noise myself on the new line joined in - The original line I didn't
hear it (or I may have just failed that section of the hearing test
<grin>).
Anyone else experiencing this?
Other than this transfer issue the new echo cancel sounds great,
Steve Radich - Colocation / Virtual Dedicated / Dedicated Servers
BitShop, Inc. - http://www.bitshop.com - $149/month colo special
-----Original Message-----
From: Klaus-Peter Junghanns [mailto:kpj at junghanns.net]
Sent: Saturday, February 22, 2003 2:58 PM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] inband DTMF in RTP
hi mark,
what's so ugly about this idea? i have modified chan_sip
to support inband dtmf. it's configurable in sip.conf on a
per peer basis.
regards
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET Internet-Services & Software-Development GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon: +49 30 79705392
fax: +49 30 79705391
mobile: +49 160 7503372
email: kpj at junghanns.net
Am Sam, 2003-02-22 um 20.35 schrieb Mark Spencer:> it could be patched to do so but this is an ugly idea in general.
>
> Mark
>
> On Thu, 20 Feb 2003, Ben Clark wrote:
>
> > Is it possible to configure asterisk to understand inband DTMF during
SIP calls?> >
> >
> > ---------------------------------
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