Johann Steinwendtner wrote:
On 2014-09-07 17:07, Joshua Colp wrote:
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3981/
Testing
Originated a call to a UnicastRTP channel and sent it to a Playback.
Confirmed that RTP was sent to the provided IP address/port with the
given format.
Hello, can you please explain what you mean by "with the given format".
There is a patch from John R. Covert which adds the capability of
selecting the codec. Or is this not necessary in trunk.
http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/48495
The UnicastRTP dial string allows specifying the format. I did not touch
MulticastRTP.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-dev
