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 < gincantal...@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 <sai...@gmail.com > > <mailto: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 > > <mailto:da...@debsinc.com>> wrote: > > > > -----Original Message----- > > From: asterisk-users-boun...@lists.digium.com > > <mailto:asterisk-users-boun...@lists.digium.com> > > [mailto: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 > > <mailto: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> > > <mailto:sip%3asai...@gtalk2voip.com <sip%253asai...@gtalk2voip.com>> > > > > > > > > > > > > -- > > Thank you with regards, > > Gopalakrishnan A.N. > > VoIP call - sip:sai...@gtalk2voip.com <sip%3asai...@gtalk2voip.com><mailto: > sip%3asai...@gtalk2voip.com <sip%253asai...@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 > -- 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
