[Asterisk-Users] Override Caller ID? Found Answer

[Ben Clark]

This is how I have it... I don't understand why it will not work for 
me.  Anyone have ideas on what I could try?

exten => _1NXXNXXXXXX,1,SetCallerID,"Asterisk <3128847514>"
exten => _1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION at deltathree

On Sunday, February 23, 2003, at 12:30 AM, Shawn Djernes wrote:

> to get that to work you need to write it like.
>
> exten => 725,1,SetCallerID "Name Here <8005551212>"
>
> Note if you are calling to PSTN only the number will be transfered
>
>
>
> -- 
> Shawn L. Djernes
> shawn at djernes.org | sdjernes at telerama.com | sdjernes at earthlink.net
> http://www.djernes.org
> 519 Washington Ave. Apt 2, Bridgeville, PA 15017
>
>
> On Sat, 22 Feb 2003, Ben Clark wrote:
>
>> 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?
>>>>>
>>>>>
>>>>> ---------------------------------
>>>>> Do you Yahoo!?
>>>>> Yahoo! Tax Center - forms, calculators, tips, and more
>>>>
>>>> _______________________________________________
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>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>
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>>
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