[Asterisk-Users] Override Caller ID? Found Answer

[Steve Radich]

I see there's a SetCallerID app I can call; sorry didn't see that until
after I sent the mail.

If anyone else is interested:

exten => 725,1,SetCallerID,725
exten => 725,2,Dial,SIP/1(cellnumber)@iconnect

As has been discussed already - IConnect works relatively well and doesn't
tie up a line for the outgoing.  If that fails I can fall through to a dial
by Zap, but that's not a concern as voicemail is always an option.

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 5:58 PM
To: 'asterisk-users at lists.digium.com'
Subject: [Asterisk-Users] Override Caller ID?

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?
> >
> >
> > ---------------------------------
> > Do you Yahoo!?
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> 
> _______________________________________________
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> Asterisk-Users at lists.digium.com
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