Kevin P. Fleming wrote:
Matthew Boehm wrote:
Umm.. "DUH!" If you remove the RTP stream from asterisk, asterisk
can't send audio (the rtp stream) to the phones.
Umm. "DUH!" Yes it can.
When a SIP endpoint is placed on hold, Asterisk will re-INVITE the audio
stream back to itself for precisely that reason.
Hmm..I stand corrected. And now that I think about it, it seems I jumped
the gun without thinking.
-Matthew
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