[asterisk-users] h323-sip: one way connection

[s m]

hello everybody

i want to have sip connection between two asterisk systems (145 and
146). connection from 145 to 146 is ok but i can not call from 146 to
145.
this is h323.conf file in 145:
[peer146]
host=192.168.0.146
type=friend
context=from-trunk


[to-146]
type=peer
host=192.168.0.146
faststart=yes
tunneling=no
progress_audio=yes
disallow=all
allow=alaw
allow=ulaw

this is mu extensions.conf file in 145:

[from-trunk]
exten=>_1.,1,Dial(SIP/to-231/1${EXTEN:1})
[line-231]
exten=>_2.,1,Dial(H323/to-146/2${EXTEN:1})

i have this error: dropping call because extensions '100', 's' and 'i'
doesn't exists in context default".

if i change "peer146" to "general", every thing is ok and i can call
from two side. my question is: in h323 connection, is it a MUST to
have "general" context in h323.conf? if not, why i have this error and
how i can solve it?
thanks in advance
sam