The best help to track this down is to try to identify the specific svn revision that caused the issue and to supply a full freeswitch debug with sip trace.
Mike On Dec 19, 2009, at 3:31 AM, Jason White <ja...@jasonjgw.net> wrote: > Revision 15904 is fine, but after upgrading to revision 16003 I get > the > following. > > 1. No problems with a FreeSWITCH to FreeSWITCH call (Celt codec). > > 2. A PCMU call to a SIP provider is fine for the first 20 to 30 > seconds, then > the audio breaks up completely. > > I have ZRTP compiled in, if that makes any difference. > > Obviously there's a regression somewhere. Let me know if I can > provide further > help. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users@lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
