----- Original Message ----- From: Douglas Garstang [mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:[EMAIL PROTECTED] Sent: Tue, 25 Jul 2006 16:31:13 -0300 Subject: RE: [asterisk-users] Caller ID on Transfers
> > I thought the new SIP invite had a 'Diverted' field or something in it? If > that's true, this is really bad because I need some way in my AGI script to > determine that it's a transferred call, and not a new call. In the case of a > transferred call, we want to set the caller id to the original calling party > info, not the transferring party info. I swear that last week when I was > doing this, the RDNIS agi variable was being set and I could use that to set > the caller id information as needed. However, now it's suddenly stopped > working and I don't know why. > > > Is this documented somewhere? No, the new INVITE does not have that info... I even just tested it from my Polycom IP600, it was a regular normal INVITE. As for documented about the call flow... probably somewhere on the internet, it's a standard SIP REFER transfer with a replaces. Joshua Colp Digium _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
