What about an indication on the dashboard that shows if the VPN is active or down.
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What would you like to see added in FreePBX 15?
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BLF - Not showing ringing and not allowing pick up - shows ** instead
For some reason I had **8 but I’ve change it to *8 - testing now.
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Streaming MOH on FreePBX
It is my understanding that the audio files uploaded to the stream are 16-bit, 128k, mp3 mono files. The length of the URL is for security purposes only, we could have a stream with 2 characters. So when I refer to a 64-bit URL it is just the amount of characters for the stream.
Could it be that either FreePBX or Asterisk has a maximum character length size for URLs?
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Asternic stats stopped working on August 8, 2018
Hi everyone.
We’ve had FreePBX operating normally for months now, but just last Wednesday, possibly after one module update or another (which ones? How do I find out after the fact?), it appears that new calls simply are no longer being logged in a way that Asternic understands. Asternic has no trouble connecting to the database it keeps, we can still see old call logs and Asternic works perfectly well with those old call logs, but we run a call center with 20+ employees in it. Everything else works perfectly fine, but we need our call statistics back.
I also just noticed that we can still see the CDR records by using the FreePBX Dashboard, and going to Reports->CDR Reports.
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Streaming MOH on FreePBX
Doubt it, but you never know. The URL is stored in the FreePBX database and is translated into one of the configuration files for Asterisk. If the problem is the length of the URL, it should be a FreePBX problem (based on the fact that all of the FreePBX settings are stored in a MySQL database).
If you have a test environment, you could try dropping your URL into the field on the FreePBX screen, store it, and restart your browser and reopen the screen. If the URL field is truncated, you have your answer.
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Inbound Calls Possibly Not Reaching the Server
So far still so good.
Is there a Responsive Firewall log that we can monitor?
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Disable OPTIONS trunk request
We did a packet capture on the port (non/standard / not 5060) being used for pjsip which is what we use to connect to our SIP trunk provider. We are seeing many of these:
“Request: OPTIONS” to our SIP provider’s address, and then a “404 Not Found” response.
Is the OPTIONS request required, and if not, how do we disable it?
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Asternic stats stopped working on August 8, 2018
maybe a connection issue between asternic and your mysql database. asternic should be able to assist with this.
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Asternic stats stopped working on August 8, 2018
No, I tried that already. Besides, if I wasn’t able to connect to the MySQL database, then Asternic wouldn’t work at all and I wouldn’t get the old stats either.
I consider this to be a problem in FreePBX because it’s part of the whole package. I haven’t done anything to Asternic, or to the MySQL backend, or any of that. I’ve only interacted with it through the GUI, and mostly left it alone except for going in and doing the module updates once a week. If that breaks things, then whatever it breaks isn’t the problem, it’s FreePBX breaking things.
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Asternic stats stopped working on August 8, 2018
We do not manage, develop or maintain asternic software. They know how their software works so they can look at commits. In general we try not to make any breaking changes so 3rd party software should generally work across the board on the same version. typically if there is a change no a new release it is an addition so old stuff typically works. In any case the correct course of action is to contact them. I wouldn’t even know which modules they call to give an answer. If they are not using native functions then it is possible a change could break a bunch of stuff as changes that would be non-breaking would happen by way of the internal functions.
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[Solved] Disable OPTIONS trunk request
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Calls dropping
I have a major issue. A client has a huge call flow. When 3 or more calls come into the office, the phones will drop the calls. 2 of the CSR’s will be on the phone and a third call will come in causing all the calls, the CSR’s were on to drop. It is a ring group with 4 extensions.
Where do I start looking for issues or adjusting settings.
(greenhorn, on this system for 1 year)
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Ringing is "static-y" on incoming calls and dialing on keypad
We are running a Sophos XG Firewall and S705 Phones in this setup. I also have another setup with S705 phones but not XG.
When calling in or dialing on the phone pad itself they are “static-y” when dialing or ringing. This has happened with both of these deployments. I have added an S700 to the mix and its doing it as well. They are all on the latest firmware for the phones.
Once connected the call quality is pretty much just fine. No issues there. These are up-to-date Cloud hosted PBX’s in the CyberLync Datacenter.
Any Ideas?
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Users dial 9 to get out but when trying to Redial from History it won't allow them to dial out because there is not a 9
Also due to a new law in place, not sure if its a state one or federal, you MUST make sure you can dial 911 without needing to dial anything before hand.
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[Solved] Disable OPTIONS trunk request
That did it. Thank you.
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Users dial 9 to get out but when trying to Redial from History it won't allow them to dial out because there is not a 9
Yup. It was discussed here long ago.
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Ringing is "static-y" on incoming calls and dialing on keypad
What is the display timeout set to? I’ve seen this happening only when the phone is “waking up from sleep”
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Firewall Questions
There’s nothing magic with pulling from git - just pull! There’s a bunch of instructions here: https://wiki.freepbx.org/display/FOP/Developer+Corner+Home
I, personally, prefer to clone to /usr/src/freepbx/modulename and then ln -s that folder to /var/www/html/admin/modules/modulename, but it’s totally up to you how you do it.
To create the module.sig, you just use the Devtools repo to generate the signature - see https://wiki.freepbx.org/display/FOP/Signing+your+own+modules#Signingyourownmodules-LocalKeyWalkthrough for the walkthrough on how to do it.
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Streaming MOH on FreePBX
I must say, I was pretty confused myself.
What, exactly, is the error you are encountering, and what are you doing when this error is encountered?
Also, you should be using UUID’s. It’s a standard format for something that is guaranteed to be universally unique.
“034d7636a89b6579f89f6556fb90e168f972251e3f95802453a309ae72d9d87a” is actually 64 BYTES, which is 512 bits.
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Calls dropping
/var/log/asterisk/full would be where I’d start.
Next, I’d try having one CSR call another, then have the third try to call his mom. If all the calls drop in that scenario, check your power supply for the phones (assuming the phones are running on POE from the switch).
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