Can you get an Ethereal trace that captures the RTP streams going to/from Asterisk? If so, you might look for SSRCs changing mid-stream.
> -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Richard Klingler > Sent: 22 September 2006 15:22 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Forcing Marker bit, because SSRC has changed > > Trying again.... > > > Has anyone an explanation why this error happens? > Only hear my echo and not the other side anymore... > and the other side can't hear me... > > Version asterisk 1.2.9 > > > -- Executing Macro("SIP/1001-9c43", > "stdexten|1010|SIP/1010") in new stack > -- Executing Dial("SIP/1001-9c43", "SIP/1010|40|o") in new stack > -- Called 1010 > -- SIP/1010-8035 is ringing > -- SIP/1010-8035 answered SIP/1001-9c43 > -- Attempting native bridge of SIP/1001-9c43 and SIP/1010-8035 > == Forcing Marker bit, because SSRC has changed > == Forcing Marker bit, because SSRC has changed > > > > cheers > rick > > > _______________________________________________ > --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
