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 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
