[Asterisk-Users] How to control codecs when originating a call?

[Tony Mountifield]

If I call an IAX channel from my SIP phone, the codec used for the call
depends on the codecs available from the SIP phone.

If however I place the IAX call using Originate in the manager API, or
using a .call file, it considers that all codecs are available.

In the latter case, how can I restrict the range of codecs that will
be considered for the call?

Thanks
Tony
-- 
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org