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

[Ronald Voermans]

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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050823/1d84bff0/attachment.htm