Problem with festival

[system]

I have SIP number which connected to first server (main). From the first server , I have trunk to second server (testserver). SIP number transmitted to second server. In second server I compile , from source , festival ( which reads text passed to it) . Problem:
when user call to SIP number in context with festival - can`t hear the voice acting with festival, but if I make direct call to SIP -everything goes well. Just noticed, when I make an internal (in second server) call to context with festival - I hear the voice acting with festival

ip 192.168.0.1
config first server

[office-pbx]
type=friend
context=office
host=192.168.0.2
port=5060
insecure=invite,port
nat=yes
qualify=yes
disallow=all
allow=alaw

ip 192.168.0.2
config secong server

type=friend
context=incoming_calls
host=192.168.0.1
port=5060
insecure=invite,port
nat=yes
dtmfmode=rfc2833
qualify=10000
canreinvite=no
threewaycalling=no
disallow=all
allow=alaw
fromdomain=192.168.0.2

config SIP namber

[sip]
type=friend
context=incoming_calls
insecure=invite,port
host=x.x.x.x
fromdomain=x.x.x.x
nat=no
qualify=yes
disallow=all
allow=alaw

[david55]

Note that insecure=invite makes no sense unless you have a secret set. Chances are that you don’t need insecure=port, either. canreinvite is deprecated in all supported versions and nat=yes is in the latest. Also the correct forum is Asterisk Support.

I suspect your problem is in the dialplan. It is possible that Festival works in early media mode, which would typically require an explicit call to Progress or Answer.