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 <da...@debsinc.com> wrote: > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Giorgio > Incantalupo > Sent: Friday, November 19, 2010 9:34 AM > To: asterisk-users@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. > > > -- > _____________________________________________________________________ > -- 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 > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:sai...@gtalk2voip.com <sip%3asai...@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 asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
