[cisco-voip] sip trunk to asterisk

[Zoltan.Kelemen at Emerson.com]

Hi Sam,

Do you need to transcode on the Cisco? (i.e. aren’t both endpoints supporting the same set of codecs?)
You need to ensure, that if you use a codec list on one call-leg (dial-peer) you support the same on the other call-leg (or outbound dial-peer) as well.
Trying transparent codec on dial-peers may also work.

For transcoding you’ll need DSCP resources and some configuration:
http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/transcoding.html

Cheers,

Zoltan Kelemen
Emerson

From: cisco-voip [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of s m
Sent: 02 April 2015 12:08
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] sip trunk to asterisk

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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