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Problems module admin updates

hi, when running module admin all plugins say : Enabled; Not available online yet all connectivity seems to be working. I have no problems connecting to schmooze servers.

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Cannot enable the followme/conference/queue

Hello:I test with FreePBX-13 and centos-7. After check and enable the modules from modules management. the GUI can not show the modules and no action for further process. -------One more thing, when i...

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Realtime CDR records

Thanks dicko, I get the following from the ssh client:ERROR 1045 (28000): Access denied for user 'freepbxuser'@'localhost' (using password: YES)

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Asterisk cmd vs freepbx

I've read in the asterisk's manual that the setup of guy interface, have a bad impact on the quality of calls..( but It's true that this book is out of date..) for this reason i've opened a question.

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Cannot enable the followme/conference/queue

None of the modules you have shown above are installed. You should install them first by installing the missing dependencies that are stated in red Also if you are a sangoma reseller you should be...

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FreePBX 13 NO GUI Web Access

Same error as before. Freepbx can't talk to asterisk

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Numerious Errors due to Certificate Manager update failure

Would sure be great to have some errors.

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Bad Destinations after Update

I am using Asterisk 13 Beta , and 4 days ago I was with full updates and now I had like 30 modules for update and I make a backup and update it, Update went perfect but then I saw on the Dashboard...

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FreEPBX 13 Follow Me time plan

I'm new to Freepbx and have managed to get everything up and runing without to many issues but the one thing I can't seem to find is a time plan for "Follow Me". Follow Me is working great but I want...

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FPBX13 RC, G722 and 1-way audio - even to voicemail

kristiandg: Is there anything I could look at to find out why the "13" system doesn't like it, but the "12" system does? one-way audio problems are always an Asterisk configuration issue. Are you sure...

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FPBX13 RC, G722 and 1-way audio - even to voicemail

Phone was set to NO NAT, and it's an internal call to voicemail. Local subnet is listed in FPBX so it's aware that's local (not to mention the server's in the same scope as the phones so it shouldn't...

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FPBX13 RC, G722 and 1-way audio - even to voicemail

What happens when you use *43?

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FPBX13 RC, G722 and 1-way audio - even to voicemail

For me, the echo test is *90, but that is also dead air. But as I said, if I call voicemail, I can leave messages, just not hear the prompts. Basically, anything that requires the server to be in the...

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FreePBX 13 NO GUI Web Access

I want to think that there is some communication between freePBX and Asterisk as every change I make is being reflected on the system. As mentioned above.. the only menu option not working is the...

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FPBX13 RC, G722 and 1-way audio - even to voicemail

The echo test literally returns the data that was sent to it. Have you tried a different phone? But stop messing around with voicemail and stuff - echo test is the first thing to get working, and it's...

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FPBX13 RC, G722 and 1-way audio - even to voicemail

There's really not a lot to screw up in that though.External IP: my static IPLocal Networks: 192.168.1.0/24, 192.168.2.0/24, 192.168.101.0/24, 192.168.102.0/24 (the one actually involved here is the...

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FPBX13 RC, G722 and 1-way audio - even to voicemail

kristiandg: Port Ranges (49152-49999) - Woah.. That looks amazingly and awfully wrong. Put that back to 10000-20000. There's no reason to change it, unless you have another SIP server behind the same...

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FPBX13 RC, G722 and 1-way audio - even to voicemail

That's precisely why I changed it. Its now in the dynamic port range instead of a range that other systems occasionally use. I'll switch it back for giggles, but again - internal call - no NAT -...

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FPBX13 RC, G722 and 1-way audio - even to voicemail

As the evidence suggested, port range back to 10000-20000 had no effect. I've put it back since that didn't have any effect.

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FPBX13 RC, G722 and 1-way audio - even to voicemail

I've already defaulted a phone - no effect. Please explain how changing a codec impacts NAT? Also, why NAT would be in-play on same-subnet calls? And lastly, if it were an issue on the phone, why I...

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