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: lists.digium.com/pipermail/asterisk-users/attachments/20050823/1d84bff0/attachment.htm
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 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.Umm.. "DUH!" If you remove the RTP stream from asterisk, asterisk can't send audio (the rtp stream) to the phones. -Matthew
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@lists.digium.com [mailto:asterisk-users-bounces@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 _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: lists.digium.com/mailman/listinfo/asterisk-users
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 > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > lists.digium.com/mailman/listinfo/asterisk-users