[Asterisk-Users] Override Caller ID? Found Answer

[Shawn Djernes]

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
> >>
> >> _______________________________________________
> >> Asterisk-Users mailing list
> >> Asterisk-Users at lists.digium.com
> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
>