Hello, >From http://www.voip-info.org/wiki/view/Asterisk+new+jitterbuffer, it says that there For Asterisk 1.2 there was no jitterbuffer in the RTP-based channels (i.e. chan_sip).
I am using 1.2 and Ind there is no reason to upgrade. Are there any developments on this? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
