Hi, Running Version 0.7.2, I receive the following error when attempting to connect two SIP Devices.
WARNING[16399]:rtp.c : 1204 ast_rtp_bridge : codec0 = 524556 is not codec1 = 524558, cannot native bridge. The bridge is made but the quality of the call is bad, a lot of disturbing noises in background. Oddly enough, both devices are using only one codec G729. I also am using the demo G729 license for Asterisk. I'm not sure how 2 different codecs are being found. I saw in ast_rtp_bridge function, that the get_codec function returned these values. Could anyone tell me where the get_codec function is? Curious as to how this is happening. Should this problem be added to the bug tracker? The SIP calls are very bad, and I did not experience this problem with 0.5.0 . Thanks, Wes _______________________________________________ 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
