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 <sai...@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 <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> > > > -- 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