Hi, all, I have buy 5 Digium's G.729A codec(it just support G.729A license) When I calll with 2 SIP UA that support G.729A and G.729B, its rtp frame have some problem when softswitch with Asterisk.
The voice frame have been drop, so sometime I can't hear voice. If I want to fix the problem when softswitch G.729A and G.729B codec. What source code I must to modify ? Or some people have finished the issue, Could you show me how to do? -- Jacky _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
