[Asterisk-Users] Moving call control to a second server

Stephan Wik
Fri, 10 Dec 2004 07:02:13 -0000

We've got the following set up:

Local Phone <-SIP(no reinvite)-> Local * <-IAX-> Central *
<-SIP(no
reinvite)-> Remote Phone

I've got calls working just fine between Local and Remote phones.

All of the outgoing calls / voicemail / Music on Hold are done on the
Central * server. I would like to configure it so that the Local Phone
can use the Transfer facilities on the Central * server.

No matter what I do it seems that the Local * server always intercepts
the # key. Is there anyway to transfer 'control' of calls to the
Central * server if a call is placed via the Local server?

I've searched in vain for info on Authenticated Transfer (is that what
is needed?).

Thanks,

Stephan Wik
ANU Galway