[c-nsp] sip trunk to asterisk

[Brian Turnbow]

Hi

> 
> thank you guys for your answers.
>  i have no problem if i use just one codec as you mentioned Jared. my call
> hangs up when i have voice class codec with more than one codec. i trace all
> debug message and think that my cisco router can not transcode codecs to
> each other. so when the codec of receiving call is different with the first
> priority in cisco router, cisco can not transcode it, therefore call hangs up. do
> you think i am true or not??? if yes would you please tell me how configure my
> router to transcode between different codecs (for example between alaw and
> ulaw)? my cisco router is 2800.

If the call is terminating on the 2800 ( ie ip to tdm) there is no need to transcode....
It would just be a standard sip offer/answer dialog.
As long as one codec matches it would be fine.

If you have 2 ip legs you should look into cisco border element configs as an SBC.
http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book.html

You can then decide how to work the codec selection and how the cisco interacts( interferes may be a better term) with the media session between the 2 legs
Transcoding would only be necessary if you have two IP legs and each leg needs to use a different codec.
Leg 1 g711a and leg2 g729 for example.
The 2800 can do this  
http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-border-element/100480-cube-transcode.html


Regards

Brian



> 
> thank you
> SAM