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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 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 _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
