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