jarvisswope,
The way that the "Provision" option works is that it will first fetch for the config file. The phone will check the file for changes. If there are any parameters that warrant a reboot, e.g. Advanced settings, network, maintenance etc, it will reboot to apply the changes.
Thanks.
Why not open a feature request for this and provide patches for this along with other feature codes. issues.freepbx.org
D&U mode has not been supported in years. I suggest not using it
I'm running FreePBX Distro 13.0.74. I have all of the latest updates for all modules installed.
I'm trying to edit the basefile for a Grandstream GXP2160 and am getting the following error:
Whoops\Exception\ErrorException
Undefined index: description
File:/var/www/html/admin/modules/endpoint/views/basefileEdit.php:305
I was able to successfully edit the basefile about a week ago.
I tried a different browser in case there was trouble with Chrome, but had the same problem. I tried deleting and recreating the template. I've reboot the PBX of course. I have the same problem with the same phone on another PBX running the same version of FreePBX.
Any suggestions? In case this proves hard to fix, can someone tell me where to find the actual basefile so I can download it with WinSCP and do a manual edit, if necessary?
Thanks!
Happy to. But, I'm guessing that you don't want a few hundred lines of _custom.conf, right 
The right way, assuming that there's not something I'm missing, would be to have the conf generation 'engine' include the two key lines right into the correct place in the dial plan for each code. That would seem to be neater. That would mean modifying the _additional.conf files and I've not checked into how those are generated. Is there something on the wiki giving a good overview or is it more a case of find a weekend and a bottle of scotch and start digging?
Cheers!
As far as I know, it's up to you. There's no "systemic" retention policy. The driver is going to be drive space (recordings are pretty big unless you convert them to MP3).
There used to be a QueueMetrics commercial module that handled most of that in the form of reports. It didn't do any kind of dashboard, though, as I recall.
The problem is that different people need different things from the system, so this might be a project that you want to hire someone to build.
Hello...
First thing's first:
PiAF Installed Version: 3.0.6.7
FreePBX Version: 13.0.73 1
Asterisk Version: 13.2.0
OS: CentOS 6.7
I installed the Extension Routes Commercial Module. Now, before you go ahead and tell me I need to register, I already did. I purchased the FREE 1 YR License. The order on the portal is COMPLETE. I see the License displayed on the portal under the Deployment ID for the server. On SysAdmin, it says that the machine has been activated. And, from there is where I clicked the link to get the free Extension Routes Module license.
However, when I go to Outbound Route Settings, or Extension Settings, it keeps saying that "Extension Routes is not registered". But, that's just not true.
Is there anything I can do for the system to refresh the licenses from the portal? My guess is that it just hasn't replicated. But, it has already been a few hours, and I've never had to wait this long before. Specially considering that the Portal shows the order as Complete. Any help will be greatly appreciated. Thanks!
Installed version of Extension Module: 13.0.6
I'm not surprised to hear that PJSIP is still not ready for primetime. It's a complete re-write of the SIP stack, and there's going to be a very long testing period. I doubt that Chan_SIP will go away until PJSIP is stable, if it does, everyone will just stick with an older version of Asterisk that still has it.
I understand why the FreePBX devs included such language in the FreePBX UI, but I think you're worried about something you needn't worry about.
For now, I'm continuing to use Chan_SIP.
That sounds like a custom context. There are lots of examples of "pin based" services out there. Set your IVR up so that "#" (or 9 maybe) goes to the "Assistants Selector" and set up the context so that the pins you want to hit are "extensions" in the context. These then point to actual specific extensions. Sounds pretty straight-forward to me. With a little creativity, you could even use database entries and add a simple database update page that associates PINs with extensions.
I'd design it using custom-contexts and make it a miscellaneous destination (so you can hit from an IVR choice).
http://wiki.freepbx.org/display/FPG/Outbound+Routes+Module talks about the outbound route module, including using restricted routes. I'm pretty sure I sent you to this the last time you asked, since I looked it up about 10 days ago.
One would think that doing that would help, but from the endless problems we've been seeing from people that tried to do that, I'd suggest that you aren't likely to save more than the hour or two the system will take to do it for you.
I'd let the system handle the upgrade. That way, you end up fwconsole in the right place and all of the database changes are handled.
Feel free to do it the hard way, though. We'll be here to answer your questions about problems trying to do that can cause.
Is that an error?
With your "fix", if I'm using 4-digit extensions, I could set up a call to go to "day-night mode" or some other similar silliness. Also, your change invalidate the ability for me to send a wake-up call to my daughter's cell phone (so she knows she's just busted her curfew).
I'm not sure I like your suggested change.
Since you are using both 5060 and 5061, you have to get into the guts of your configuration to set up the incoming so that it listens on more than one port. It's doable - we've been doing it for years to support our "roaming, random-IP" phones.
If you are using PJ-SIP and SIP, you HAVE to use two different ports. It took me several tries on the one machine that I AM using PJ-SIP on to get it all set up correctly. For now (until PJ-SIP is something more than a "Beta with a plan"), I'd recommend setting your incoming to use CHAN-SIP and configure each "trunk" with its own port.
I'm not sure how you would route those to your phone. They are files (as are your voicemail), but not in the same contextual framework as a voicemail.
How would the interface work? You click on the download link and ...
If you can elucidate how you want it to work, someone might be able to suggest a way to help you make it happen. Failing that, you could (or should) submit a feature request and see what happens.
On the other hand, remember that this is Open Source software. Submitting a feature request with working code could make the process go a lot faster.
I was able to submit the ticket now, thanks TheJames!
Dave Burgess, the fix is simply correcting a problem with their code - in FreePBX, if you set the max digit length in the WakeUp calls module, then the module will not allow you to set an extension greater than the length.
With their original code however, it was the opposite: It wouldn't accept any LESS than the Max digit length. How many digits you want to send depends on your setting. This code fix just makes it correct again.
Oh sure, I'll be working on making it happen, it's something we're going to need due to our particular environment.
As for how it would work, exactly the same as voicemail. Click the phone icon, phone rings, lift receiver, listen to message/call recording. The interface is already there, conceptually at least.
I wanted to ask about it incase this was something it already did and I simply hadn't turned it on!
The UCP is quite an interesting chunk of code from what I've seen so far, so this will have to wait a few weeks while I get up to speed with the rest of things. If this does not already exist, then it will do soon!
Easy. Send the Playback() application the full path to where the file is. As long as Asterisk can read the file then it'll playback.
Yeah it's pretty easy to choose, and nice to have the choice!
Downgraded to Asterisk Ver. 11.19.0 working fine:innocent: