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

Gopalakrishnan A.N saigop at gmail.com
Fri Nov 19 09:51:28 CST 2010


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> wrote:

> -----Original Message-----
> From: 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
> 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>
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