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
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
Also retest with no zrtp
send a full console debug log with sip trace
On Dec 19, 2009 8:33 AM, Michael Jerris m...@jerris.com wrote:
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
I tried a patch out of pure deduction and speculation from your post.
Can you update and test it for me please?
On Sat, Dec 19, 2009 at 9:19 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
Also retest with no zrtp
send a full console debug log with sip trace
On Dec 19, 2009 8:33
