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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