rtp debug On 5/15/06, Philippe Lindheimer <[EMAIL PROTECTED]> wrote:
I do a sip debug on the appropriate channel or IP address and look at the SIP messages. Would be great if there were an easier way though? p From: "Brent Torrenga" <[EMAIL PROTECTED]> To: <asterisk-users@lists.digium.com> Date: Mon, 15 May 2006 12:52:19 -0500 Subject: [Asterisk-Users] How to tell if RTP stream is has been reinvited? Howdy, How can you tell if RTP traffic has been reinvited/is bypassing an * server? Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 +1 219 836 8918 x325 Voice +1 219 836 1138 Facsimile www.torrenga.com ________________________________ New Yahoo! Messenger with Voice. Call regular phones from your PC and save big. _______________________________________________ --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
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