Patch is a little harder then I expected. The RTP settings which the one I normally select was missing is not the exact/only resolution to the one way audio with bandwidth.com. When removing every setting I carried over from a working server I had left 2 localnet's defined when I thought it was down to just the canreinvite.
After further testing I realize with my complex network I need multiple Local Network Address (localnet) definitions. Sip.conf shows that localnet is expected to be a multi line variable like extensions, however there is no interface to handle this. Java popup help for Local Network Address says: All RFC 1918 addresses are local networks. This is a bit confusing when requesting data for form field entry. I originally took this to be that all the RFC1918 addresses were being included by default. I can see not wanting to open all the RFC1918's up by default for security, but I think a mechanism to set this should be in the GUI. Delved into the code with my brother and looks like astman.js action=getconfig can handle multiple line returns via variable p for non unique subfields. Just not sure how to set localnet to have variable p set properly or if it is dynamically setting p during the config return. In either case not sure how I would relay that data to the html so it would show multiple Local Network Address prompts based on the existing framework you have built. Sorry this is a bit beyond me. If nobody there has a quick fix to resolve then I will have my brother work with me to submit a patch maybe next week. Posting a bug related to this currently as well. Thanks, Trevor > -----Original Message----- > From: [EMAIL PROTECTED] [mailto:asterisk-gui- > [EMAIL PROTECTED] On Behalf Of bkruse > Sent: Monday, June 23, 2008 1:53 PM > To: Asterisk GUI project discussion > Subject: Re: [asterisk-gui] Patch submission > > Just add the .patch in the issue > > -bk > > Trevor Benson wrote: > > 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 > > > > > _______________________________________________ > --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
