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

[Stephan Wik]

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