[Asterisk-Users] Music On Hold + canreinvite=yes

[Matthew Boehm]

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