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

[Ronald Voermans]

I found the problem. The ztdummy wasn't loaded. So it had no timer
there. When the RTP stream was going through asterisk, I think * used
the stream for timing. 

Ronald

-----Oorspronkelijk bericht-----
Van: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] Namens Matthew Boehm
Verzonden: dinsdag 23 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) 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

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