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

[Matthew Boehm]

Ronald Voermans wrote:
> For canreinvite=yes to work, I think I need to remove the t argument in 
> the Dial(SIP/ext|60|t) application. Otherwise, Asterisk will allways 
> stay in the middle. I don't want that, so I removed the 't' argument. 
> That works. Now, when two UA are calling, Asterisk gets out of the RTP 
> stream. However, when removing the 't' argument, the Music On Hold 
> doesn't work anymore between these two UA. If I put one UA on hold, 
> Asterisk states that it is starting Music On Hold, but the holding party 
> doesn't hear the audio stream.

	Umm.. "DUH!" If you remove the RTP stream from asterisk, asterisk can't 
send audio (the rtp stream) to the phones.

-Matthew