[Asterisk-Users] Override Caller ID? Found Answer

Brad Bergman bradley at bergman.ca
Sun Feb 23 17:58:16 MST 2003 

Perhaps your problem is the telco that is completing your call. I have been 
playing with this myself and have a couple of little observations for calls I 
placed through Asterisk via iconnect: 

1. When I dial my cell phone, I see whatever caller ID I set in Asterisk (with 
a 1 prepended), regardless of how invalid the number is (e.g., 411). If I don't 
set it, I see 1 plus my 8-digit iconnect account number.

2. When I dial my landline, I see a specific number (in area code 646) no 
matter what I set in Asterisk. Perhaps the telco substitutes the ANI number 
when the caller ID it receives doesn't match? I don't really know.

3. Same as (2) when I dialled my sister's cell, which is on a different network 
than mine.


Cheers,
Brad


On Mon, 24 Feb 2003 00:43:27 0 (GMT), Ben Clark wrote:

> 
> 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
> > <a 
href="http://mail.mail1.emailfiltering.co.uk/jump/http://www.djernes.org">http:/
/www.djernes.org</a>
> > 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. - <a 
href="http://mail.mail1.emailfiltering.co.uk/jump/http://www.bitshop.com">http:/
/www.bitshop.com</a> - $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. - <a 
href="http://mail.mail1.emailfiltering.co.uk/jump/http://www.bitshop.com">http:/
/www.bitshop.com</a> - $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. - <a 
href="http://mail.mail1.emailfiltering.co.uk/jump/http://www.bitshop.com">http:/
/www.bitshop.com</a> - $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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> >>>
> >>>
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> >>>
> >>
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> >
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