On 2014-09-07 17:07, Joshua Colp wrote:
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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 Thanks Regards, Hans -- _____________________________________________________________________ -- 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
