I did file a support case I’m waiting from them to replay back to me
@comtech Version FreePBX 14.0.2.14 , Asterisk 13.19.1, Appointment Reminder 14.0.1.4 .
Thank you for your time
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Appointment reminder is calling before the day start
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Extend time for user to enter extension for transfer
Assuming your Yealink phones:
After pressing Transfer, the phone should wait for the transfer-to number the same as for a normal call. Set the Inter Digit Time as desired. Better, set it very long and set up Dial-now Rules (*xxx for this case) so most calls get sent as soon as the complete number has been entered. (For international calls, you’ll still have to press Send or wait for the timeout, because the phone doesn’t know how many digits to expect.)
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Queue Report [Resolved]
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
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Moving to a temp server while existing being rebuilt (licensing)
Nope. It will simply download a new license file if one is available. Licensing is checked locally, not remotely.
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Moving to a temp server while existing being rebuilt (licensing)
And if licensing was removed for an instance - or rather, moved to a new ID, wouldn’t the nightly downloaded license then be invalid on that box?
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Moving to a temp server while existing being rebuilt (licensing)
The problem is in the license file itself. When you buy a 25 year module from us the license is set for 25 years. We don’t ever verify after the 25 years as people screamed at us when we use to that if I have something for 25 years it should not phone home.
We can’t just say hey when you phone home deactivate the licnese as you can just not phone home and have the license work for 25 years.
You will be amazed the number of cheats we have that try and cheat the system already.
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Callback - Specifc Outbound Route with Inbound CID
Ok. So, i had another problem but i was able to solve it…
Here’s what happened:
On Callback Number i inserted: 0070${CALLERID(num)}
When i configured the destination to a Extension, it worked like a charm.
But when i directed to a Queue, it did’nt work.
I decided to take another look today…
The problem is:
The queue ANSWERs the call… So the call wont go to the Caller ID.
You have to enable Queue No Answer.
Also, even if u enable Queue No Answer, it’ll still not work if u have Music on Hold.
You have to set the queue also to “Ring Only”. Otherwise it’ll not work.
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Moving to a temp server while existing being rebuilt (licensing)
Just thinking of possible solutions: You could just require the phoning home and explain that it’s for good reason. Or, to make everyone happy, you could make the phoning home an opt-in thing so that people can choose either a two-transfer limit or an unlimited limit with phoning home.
Like I said though, advanced cheaters can simply crack whatever protection you build in, so please prioritize the needs of your paying customers over the never-ending effort of blocking the cheaters. I’m afraid to do a clean install because of the current limits. When I play video games, I never use my potions or single-use scrolls since I don’t want to waste them.
Thank you
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Combining FreePBX with Panasonic KX-NCP500
It very well could be not defined properly. What makes you question it on the FreePBX side?
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Total Active Calls from Dashboard Far More Calls than Exist
Yes, but I was hoping to avoid that. Thanks.
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Moving to a temp server while existing being rebuilt (licensing)
All you have to do is message support and they will take care of you. Even past the 2 limit. It’s the same for Microsoft. You have a limit of a certain number of activations (was 2) if you go past that just call them and explain and they give you another one.
Scanning through this thread it appears you haven’t even asked support about the issue you are going to face. Which means they could have just reset the limit for you to proceed with this. This is why you should ask them instead of speculating.
To be honest. This isn’t a priority. Instead we are slowing upgrading our backend licensing system to work with PHP 7+ which zend doesnt support. So perhaps Zend resets won’t exist at all. But that is speculating as well.
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Total Active Calls from Dashboard Far More Calls than Exist
Assuming you are logged in as root, in which case you definitely don’t want to start asterisk this way. Use:
fwconsole restart↧
Inbound calls stop working, but outbound calls work
FINALLY figured out why Inbound calls weren’t working! Drum roll please!!!
You can’t set E164 to True in VoicePulse. You must set it to False. I knew this because I watched a tutorial saying not to do it and yet they were set to true anyway; I had to go back to that same tutorial to think to look at that variable. Man IT is stressful, and it’s always the smallest things.
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Paging & Intercom - Code Red Alert
I have a PBXact with Sangoma S405 phones installed at a school and they added a phone to every class room. They had us add a button that a teacher can press that plays a “Code Red need Assistance” message which works just as they like, however the phone that activates the page also has the page come over it even though it is not in the page group.
What they are asking is the phone that activates the page stays silent so people in the room are unaware the activation has been done. Is this possible and if not is anyone out there want to give me a quote to develop this?
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We are hiring an Onboarding Customer Engineer for our Cloud Services
And best of all you get to work directory for the company behind FreePBX and a team of awesome dedicated people.
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Call Control & Calendars
Before I shoot myself in the foot, does anyone have input on the best way to manage call schedules with the calendar module?
I’ve read the documentation, I’ve linked my calendar, I’ve figured out a solution but I’m not sure if it is the best one.
What I currently have is a single calendar (ical format) with time blocks to specify the doctor on call - simply with a name in the calendar time slot.
I can control time conditions now based on having an appointment in a time slot. I can also make groups and select specific appointments within that calendar. But it looks like I have to login to freepbx and reselect which the appointments when they change.
I could also make 3 calendars, one for each person. And then put an event when they are on call.
Is there a better way to go about this? If someone has a link I’d happily read it. I wasn’t successful in finding a solution with web searching.
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Voip.ms Registration timed out, trying again
Hi,
I have a system running FreePBX 14.0.1.36.
I have 4 sip trunks setup, 3 from Voip.ms and 1 from voipmuch.
The voipmuch line always registers, never have problem with it.
One of the voip.ms line always works but is very lightly used (5 calls per day)
the other 2 voip.ms (about 300 calls per day) lines had lots of problems with registration failing when i first setup (1 month ago) then I set the udp timeout on my nat rules to 120, and tadaa, issue resolved. (mikrotik router) UNTIL… rebooted the freepbx system and then the disconnects started happening again.
I have contacted voip.ms and they say that everything looks fine on their end. nothing wrong with my connection string peer details etc… they say it must be something in my router or network.
I will also add that i have another freepbx system on the same network with 2 lines from voip.ms and never have a problem with those but they are also light use… (maybe 30 calls per day.) Obviously this is urgent so any help from anybody is greatly appreciated.
I realize i have not provided any details so what ever you want to see let me know.
I will also mention that i had another setup where i had similar issue and i replaced his rogers cable modem/router with a tp-link router and hes never had a problem since. (I dont think there is anything wrong with my router because i have other lines which are not used as much work fine all the time, although i do think some settings need to be changed possible due to the 2 lines being used more than the rest)
My problem is pretty much identical to this one: Dropping ip registrations
If i restart asterisk then it stays registerd for 1 - 15 minutes
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Ipdect db20n
I am having a terrible time getting our DB20N registered… I am trying to use the End Point Manger and it keeps telling me: “Please enter proper MAC address before proceeding with submission…” However, it lists the basestation and its MAC address is correct… What am I missing???
Thanks,
Jim
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UCP - Unable to connect to the UCP Node Server because: 'Error: xhr poll error'
Yep, I didn’t have 8001 or 8003 opened. Thanks!
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Announcements and off-site micellaneous destinations
Thanks. I have not had a chance to try out your suggestions until today. In DAHDI Config - Global Settings I have increased the rx and tx gain by 0.1 and will see how much of an improvement that brings. I am not familiar with the insides of Asterisk and so, while I understand that the r option on the outgoing trunk will generate ring tones, per your recommendation, I cannot with assurance determine where this is actually set in FreePBX. I would appreciate be pointed to the correct module and setting location.
In the Global Settings there is a field called: Other Global Dahdi Settings: , but this has an implied syntax of option = value. Is this where the r option is set? Do I just place ‘r’ in the left hand side and leave the value empty? Or is this set somewhere else entirely?
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