[Asterisk-Users] Override Caller ID? Found Answer

[Ben Clark]

Does this work?  I've tried this same config with no luck... Each time 
I call out from asterisk on iconnect it is a "private number" on the 
remote caller id.


On Saturday, February 22, 2003, at 05:52 PM, Steve Radich wrote:

> 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?
>>>
>>>
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>>
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>
>
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