Sorry for the top post, i've only got a few seconds to respond. The Async patch (-I) has nothing to do with packetization. The poster that added that information to the bug notes under 5162 was confused.
As to why it is not working, you said you set it on a peer. Did that 'peer' call Asterisk, or did another device on Asterisk call it? Is the second device also using 30ms? Do you have re-invites enabled? A re-invite to/from a device not told to use 30ms won't use 30ms. I use type 'friend' and get 30ms to/from my endpoints, and since my server is primarily for MeetMe, I do not have reinvite enabled. Dan -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Neubauer Sent: Friday, May 19, 2006 7:27 AM To: Asterisk Users Mailing List Subject: [Asterisk-Users] RTP Packetization Hi all, I need to be able to adjust packet sizes and found the patch at http://bugs.digium.com/view.php?id=5162 Thus, I checked out and compiled http://svn.digium.com/view/asterisk-old/team/group/5162_rtp_packetizatio n I added the line "packetization = 30" for one peer in my sip.conf and started asterisk with the "-I" switch for async RTP. That's all it takes according to the 5162 issue page. Nevertheless, asterisk still keeps sending it 20ms packets, even though a "sip show peer foobar" shows Packetization: 30. What could be wrong? What about that ztdummy thing for internal timing? Is this necessary to run asterisk properly? Is it important for packetization? Regards, Patrick _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
