[cisco-voip] 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 trace all debug messages in cisco and think it is happened when receiving
call has a codec which differs with first codec priority in cisco router
because my cisco router can not transcode these codecs to each other. am i
right or misunderstand?? if it is true, how can i enable transcoding on my
cisco router?  i have a 2800 router.

any comments ot hints are really appreciated.
SAM
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