[asterisk-users] callerid not forwarded when transferring call from ISDN line to mobile phone via Asterisk

[Gopalakrishnan A.N]

I guess it will not work with PSTN lines since the control is transferred to
the Exchange. I am not too sure, I am just sharing my thoughts....

On Fri, Nov 19, 2010 at 9:28 PM, Giorgio Incantalupo <
gincantalupo at fgasoftware.com> wrote:

> Hi Gopalakrishnan A.N,
>
> I tried it but it seems like my telco is overwriting the value I set as
> callerid.
> Maybe it is possible with Voip providers only.
>
> Giorgio Incantalupo
>
> Gopalakrishnan A.N wrote:
> > Forgot to tell you the version I tried is Asterisk 1.4 with TrixBox, I
> > disabled the caller-id checkbox while creating VoIP trunk then it
> > started working for me..
> >
> > On Fri, Nov 19, 2010 at 9:21 PM, Gopalakrishnan A.N <saigop at gmail.com
> > <mailto:saigop at gmail.com>> wrote:
> >
> >     Please try this in your dialplan
> >     Set(CALLERID(name)=${CALLERID(num)})
> >     Some where I tried and it worked with VoIP account A to B as VoIP
> >     trunk and B forward the call to C whereas in C A's number will be
> >     displayed.
> >
> >     If you could paste more details as Danny said that would help the
> >     list to assist you more.
> >
> >
> >     On Fri, Nov 19, 2010 at 9:11 PM, Danny Nicholas <danny at debsinc.com
> >     <mailto:danny at debsinc.com>> wrote:
> >
> >         -----Original Message-----
> >         From: asterisk-users-bounces at lists.digium.com
> >         <mailto:asterisk-users-bounces at lists.digium.com>
> >         [mailto:asterisk-users-bounces at lists.digium.com
> >         <mailto:asterisk-users-bounces at lists.digium.com>] On Behalf Of
> >         Giorgio
> >         Incantalupo
> >         Sent: Friday, November 19, 2010 9:34 AM
> >         To: asterisk-users at lists.digium.com
> >         <mailto:asterisk-users at lists.digium.com>
> >         Subject: [asterisk-users] callerid not forwarded when
> >         transferring call from
> >         ISDN line to mobile phone via Asterisk
> >
> >         Hi all,
> >
> >         I've got 4 actors on my stage:
> >         Alice calling from outside
> >         Bob transferring incoming calls to Charlie
> >         Charlie who has a mobile phone
> >
> >         My PBX which is connected to my ISDN line.
> >
> >         I want Charlie to see Alice's Callerid after Bob has
> >         transferred the
> >         call as if Charlie is receiving the call from  Alice,
> >         transparently.
> >
> >         Tried to set the callerid but Charlie sees my telco line
> >         number, not the
> >         callerid of Alice.
> >
> >         How can I do this?
> >
> >         Thank you.
> >
> >         Giorgio
> >
> >
> >         --
> >         We know that Alice and Charlie are both on external trunks.
> >          We DON'T know
> >         what flavor of Asterisk you are using, but it probably doesn't
> >         matter your
> >         call is going like this
> >         ID #1 --> asterisk --> destination.
> >         If destination were internal, ID#1 would remain intact, but
> >         since you are
> >         opening a new trunk to forward the call, you lose ID#2 and
> >         replace it with
> >         your Telco ID.  You could "spoof" this depending on your asterisk
> >         version/telco arrangement, but by default, things are as you
> >         described.
> >
> >
> >         --
> >
> _____________________________________________________________________
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> >
> >
> >
> >
> >     --
> >     Thank you  with regards,
> >     Gopalakrishnan A.N.
> >     VoIP call - sip:saigop at gtalk2voip.com <sip%3Asaigop at gtalk2voip.com>
> >     <mailto:sip%3Asaigop at gtalk2voip.com <sip%253Asaigop at gtalk2voip.com>>
> >
> >
> >
> >
> >
> > --
> > Thank you  with regards,
> > Gopalakrishnan A.N.
> > VoIP call - sip:saigop at gtalk2voip.com <sip%3Asaigop at gtalk2voip.com><mailto:
> sip%3Asaigop at gtalk2voip.com <sip%253Asaigop at gtalk2voip.com>>
> >
> >
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Thank you  with regards,
Gopalakrishnan A.N.
VoIP call - sip:saigop at gtalk2voip.com <sip%3Asaigop at gtalk2voip.com>
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