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

[Rene Kluwen]

For Asterisk to play MOH, it will need to have an RTP connection, right?
How otherwise, would you want to play MOH?

Rene Kluwen
Chimit

> 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.
>
> Is this resolvable?
>
> Thanks,
>
> Ronald Voermans
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