[Asterisk-Users] SIP to IAX - forcing codec pass thru

Peter Gradwell peter at gradwell.com
Mon May 22 04:49:32 MST 2006


Mark Phillips wrote:
> Hi Peter,
> 
> I don't see any codec allow=blah statements. If your end user has
> something like
> 
> [gradwell]
> disallow=all
> allow=gsm
> 
> Then you'll be forced to send them a GSM coded call. 
> 
> Why not force the codec at your end by only supporting one? If the
> customer then transcodes the call when it gets forwarded to his handset
> there's not much you can do about that but at least you'll have handed
> the call off in the best format you can source.

mmm, but as you've seen, some customers like using multiple codecs. The 
cisco kit is able to support a raft of options - and it does transcoding 
very nicely - so the optimum solution is to have the cisco + customer's 
asterisk agree on the same codec, and then have our asterisk server (in 
the middle) do as little as possible.

cheers
peter

-- 
peter gradwell. gradwell dot com Ltd. http://www.gradwell.com/
  -- engineering & hosting services for email, web and voip --
   -- http://www.peter.me.uk/  -- http://www.voip.org.uk/ --



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