[Freeswitch-users] Asterisk canreinvite (directmedia) equivalent in freeswitch

[Ken Rice]

If you are hearing noise then you have a different problem... Post a full
trace of the call...

Sounds more like a broken endpoint.

And don't forget you probably want to do some things with the SDPs to make
sure you have matching codecs... You can do testing against them in the
dialplan and other ways


On 7/1/15, 8:06 AM, "Ričardas Stoma" <ricardas.stoma at gmail.com> wrote:

> I tried bypass_media_after_bridge but calls still get connected with different
> codecs and all i can hear is digital noise. 
> 
> 2015-07-01 15:34 GMT+03:00 Ken Rice <krice at freeswitch.org>:
>> Theres a few ways to accomplish this... But you are probably looking for
>> something like bypass_after_bridge
>> 
>> K 
>> 
>> 
>> On 6/30/15, 12:16 PM, "Ričardas Stoma" <ricardas.stoma at gmail.com
>> <http://ricardas.stoma@gmail.com> > wrote:
>> 
>>> Does freeswitch have anything similar to canreinvite (asterisk sip setting)?
>>> I know there is bypass_media but it does not work in the same way as
>>> canreinvite in asterisk.
>>> 
>>> It looks like bypass_media always tells freeswitch to send media directly
>>> between originator and terminator even though they do not have common
>>> codecs.
>>> 
>>> I want to use bypass_media only when leg A and leg B have common codecs. If
>>> they do not have then freeswitch should stay in media path and handle codec
>>> transcoding.
>>> 
>>> Is there an easy way to achieve this in freeswitch? 
>>> 
>>> 
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-- 
Ken
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