If you listen to the last quarertly call we had around minute 17 you will hear Bill our CEO mention that Nick has sold the last of his stock.
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3CX trend has me concerned as a new user of FreePBX
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Cannot upgrade freepbx 13 to 14
Yep that was it. Thanks so much for the help!
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After adding BLF's to phones ring groups acting strange. They all keep ringing even after call is picked up
Looks like that was it! New switch resolved the issue.
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Inbound call from fxo always busy in IP phones
/var/log/asterisk/full.
Double check that DAHDI is still running and is configured correctly.
A “pretty OK practice” (I won’t say ‘best’) is to get your DAHDI configuration set and disable the DAHDI control modules from FreePBX. It does a good job as a starting point, but once you get it set up, you don’t need to (pretty much) ever mess with it again.
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Rest API not working correctly
Pop some details into an Issues Ticket. I’m pretty sure there are people for that.
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Streaming MOH on FreePBX
Streams in 64 bit format is different than 32-bit/character URLs. Which question are you trying to get answered.
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Rest API not working correctly
The rest api module is not supported.
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FreePbx Iso Vs FreePbx Instalación Manual
Yo lo venia haciendo manual hasta que probé con la ISO, mucho más fácil y menos posibilidades de errores.
El único problema que tuve con la ISO fue cuando lo quise instalar en un VPS de OVH , me volví loco, opte por pasarme a Vultr y va de 10!
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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
Yes - don’t specify the ‘9’ and don’t require it for an outside line. With FreePBX, there is no need to specify a digit for getting an outside line. The system is smart enough to get the calls where they need to go when correctly configured.
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Fresh install sng7 on System 60 via USB
The “hardware” problem isn’t a failure (although there’s no reason to believe that hardware can’t fail after two years). The problem is that the kernel doesn’t support your APIC configuration. As the error message recommended, try booting with the “noapic” option turned on.
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Inbound Call - We hear caller, caller can't hear us
Check the /var/log/asterisk/full log and you’ll probably see that inbound calls drop after 30 seconds because of RTP failures.
One part of that is that your register string points to a network and not a specific server. I doubt that you are using 16 addresses on this server for your VOIP system. Set that register string so that it works with your specific server and you might see some relief.
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Streaming MOH on FreePBX
Are there any versions of FreePBX that supports 64-bit/character URLs for streaming MOH?
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Streaming MOH on FreePBX
I can’t think of a reason why FreePBX wouldn’t support 64-bit URLs. The internationalization code in the system works pretty well for most things; this this is passed intact, not working would fail the Rule of Least Astonishment for me. @xrobau would be a good resource for this (being a guy that pushes buttons and knows the guts of the system). Perhaps he can provide some insight?
If you don’t get any satisfaction here, you can drop an Issues ticket into the mix with the specific URL you are trying to hit so the code monkeys can look into it.
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Streaming MOH on FreePBX
Thanks for the reply Dave. It seems we can only get our 32-bit URLs to work with FreePBX. Whenever I have provided a 64-bit URL to the end user they have silence when callers are placed on hold.
I think I’ll open an Issues ticket to see where that gets me.
Thanks for the help.
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FreePBX, pfSense, Site to Multi-Site VPN - Calls Immediately Hang Up
@PitzKey might be on to something.
FreePBX has an integrated firewall. Among other things to look at, you should check the internal blacklist to make sure that your phones haven’t been blocked during your testing. Also, make sure that the phone system has a clean route (check the from console log in) to all of the phones in the remote network. Double check the internal firewall setting to make sure that all of the networking in the local network are whitelisted.
If you “tail -F /var/log/asterisk/full” from the console login (login as ‘root’) and see what happens with the call. If the call isn’t getting processed, it’s either a routing or firewall problem. If the call gets through but rejected, let us know the error message and we can troubleshoot from there.
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Streaming MOH on FreePBX
OK - just to make sure we are saying the same thing:
- A 64-bit URL involves things like Kanji-language web site addresses.
- A URL to a 64-bit stream is like “http://www.netbsd.org/kanji-code”.
There’s a HUGE difference, between those. Be sure to ask for the right thing.
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Misc destinations not working
I have this problem with “one-armed” calls (calls coming from my provider and going back to through that provider). My “ham-fisted” solution is to set up a second “emergency” outbound route through another provider. I use a pre-paid “outbound only” provider for this. For the dozens of minutes I use every year, the $20 I put in back in 2015 is still being whittled away.
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Streaming MOH on FreePBX
Here is an example of our stream:
https://streaming.easyonhold.com/034d7636a89b6579f89f6556fb90e168f972251e3f95802453a309ae72d9d87a
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Streaming MOH on FreePBX
That is a URL to a 64-bit datastream. The URL is not the problem.
This is probably as Asterisk problem (rather than a FreePBX problem). Using a 64-bit data stream for a 16-bit telephone connection seems like a real waste of processing power. Even with a 32-bit stream, Asterisk has to transcode that down to an 16-bit mono on-hold channel.
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Phone ring inactive network
Check the /var/log/asterisk/full log and see what’s happening.
I don’t know of anything in the system that would do this, but the logs will tell the tale.
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