[Asterisk-Users] Override Caller ID? Found Answer

Mark Spencer markster at digium.com
Mon Feb 24 07:09:04 MST 2003 

Caller*ID on a PRI has various levels of "trust" associated with it.  It
could be some providers don't trust the callerid.

Mark

On Sun, 23 Feb 2003, Brad Bergman wrote:

> 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
> > >>>>
> > >>>> _______________________________________________
> > >>>> Asterisk-Users mailing list
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> > >>>>
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> > >>>
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> > >>>
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> > >>
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