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
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.
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
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.
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
augustus 2005 18:02
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [Asterisk-Users] Music On Hold + canreinvite=yes
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
