[Freeswitch-users] Codecs and Freeswitch features

[trixter]

On Wed, 2006-07-19 at 10:31 +0100, AmberVoIP wrote:
> Hello,
> 
> would be nice to hear some resume, if someone have tested freeswitch.
> especially i interesting in:
> 
> h323 -> sip and sip -> h323;
you cna do that via woomera easily.  


> codecs (especially g723 and g729 - is available (testing, commercial?);
g.723.1 and g.729 can be written there are plans to provide those codecs
soon, however there is a little bit of stuff that has to be done
regarding the patent licenses and liability.  That should be taken care
of shortly however.

> ivr?
You can do this using a variety of methods, so far I personally have
only dont this type of stuff in javascript.

> something like AGI in asterisk (external dialplan applications).
> 

everything is intended to be an external dialplan function, I have done
some stuff using mod_spidermonkey as have others, and that works quite
well.

Asterisk tries to get you to do everything in the dialplan by default,
and freeswitch is designed to make that hard on purpose.  Once the opal
stuff is done you will have even more languages to choose from in
freeswitch as well.