[asterisk-gui] Patch submission
Trevor Benson
tbenson at a-1networks.com
Mon Jun 23 23:38:21 CDT 2008
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: asterisk-gui-bounces at lists.digium.com [mailto:asterisk-gui-
> bounces at lists.digium.com] 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: asterisk-gui-bounces at lists.digium.com [mailto:asterisk-gui-
> >> bounces at lists.digium.com] 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
> >>>
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