3307 as of Friday night (latest updates as of mid day). Have not updated today. If sip.html isn't used then why when I update the code does the Global SIP Options page update correctly and then set the files with the settings I specify??? Where should I modify the html for Global SIP Options??
Trevor > -----Original Message----- > From: [EMAIL PROTECTED] [mailto:asterisk-gui- > [EMAIL PROTECTED] On Behalf Of bkruse > Sent: Monday, June 23, 2008 12:18 PM > To: Asterisk GUI project discussion > Subject: Re: [asterisk-gui] Patch submission > > http://bugs.digium.com > > Just upload the .patch to the latest svn revision, and get some > comments :) > > The problem with your theory, is that we no longer use sip.html. > > What version of the gui are you really using? > > -bk > > Trevor Benson wrote: > > Should this be done via the bug system, are commits from non auth'd > sources stored for review? I have a patch to sip.html that fixes > issues with bandwidth.com in the specific network configuration I am > running. Without the patch you get 1 way audio unless using externip > and localnet together which kills all sip traffic instead of fixing the > issue. > > > > > > Thanks, > > Trevor > > > > _______________________________________________ > > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > > > asterisk-gui mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-gui > > > > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-gui mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-gui _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-gui mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-gui
