hello everybody,
i want to configure a sip trunk between my system which has asterisk 11.5.1
and a cisco router. this is my scenario:
Freepbx-----my system-----cisco-router----Freepbx
my system acts like a router. if i set just one codec in dial-peers on
cisco router, every thing is ok and i can make a call. but if i set
different codecs in a voice class codec and assign it to dial-peers in
cisco router, i can not make calls.
if i change my scenario like:
Freepbx------cisco-router------Freepbx
calls are succeed without any problem. Freepbx are asterisk-base too, so i
think something is wrong in my system (my asterisk configuration is not
correct or something is missing).
any body knows how should i fix this problem? any comments or hints are
really appreciated.
P.S: my sip.conf:
[peer-1]
host=X.X.X.X
type=peer
context=from-trunk
allow=all
qualify=yes
insecure=port,invite
[peer-2]
host=Y.Y.Y.Y
type=peer
context=from-trunk
allow=all
qualify=yes
insecure=port,invite
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