[asterisk-users] SIP - IAX Attended transfer

[David Parcerisa]

need help on this..

I have configured all my internal extensions as SIP phones, i have one
ATA-186 and one softphone. When I try to transfer calls between
internalt, I use *1 as it configured on feature.conf.

But my external line configured a IAX.

When there is an incoming call from outside, the line 1 from my ATA186
rings. If I want to transfer my call using *1 is not possible. So I
need to use flash button only when call are from IAX terminal.

This is normal?
I'm using Asterisk 1.2.16 on a mac osx .