[c-nsp] sip trunk to asterisk

[s m]

hello everybody,

i want to configure a sip trunk between a cisco router and my system which
has asterisk. this is my scenario:

Freepbx-----my system-----cisco-router----Freepbx

my system acts like a router. in cisco, if i set just one codec in
dial-peers, 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, i can make call
but call is terminated.
i think there is some difference in sip options (maybe sip headers) between
cisco and asterisk which causes to codec negotiation fail. as a result of
it, call terminate.

any body try it before? any comments or hints are really appreciated.

SAM