Have it ring and not auto answer.
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Intercom / Auto answer behavior
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Suddenly can't dial out
Your OBI says the line is busy.
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High CPU Usage for an idle box
This would not be improved in PHP 7. It would be improved in PHP 5.6 which is what we are already using.
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Outbound call shows own number on outbound phone instead of dialed number
Not sure how much more explicit the answer can be. Is there some other question you’re not asking?
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Porting 2.11 version 'Wake Up Calls" module to V14
https://wiki.freepbx.org/pages/viewpage.action?pageId=29753755
Unsure what you mean
Everything you need should be in and around the wiki article linked above.
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TTS Engine Custom - Amazon Polly - 24 languages
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If I change the hostname the UCP daemon will not start
If I change the hostname from localhost to something else, the UCP daemon will not start. There is an error in the MySQL database will have issues accessing the new hostname once is changed from localhost to something else. Is there a way to change this during the installation.
Sincerely,
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How to get Inputs from IVR freepbx?
As promised and requested
WHAT IT DOES
This dialplan will ask the user to enter in a pin number, and then it will save it to a file.
Tons more you can do with this and tweaks of course, but this is an example of what you can do.
I’m using all the default recordings in asterisk, so they aren’t perfect prompts, but good enough for you to test with. Happy to try and help with any tweaks or modifications you might have.
Installation
1> Open up /etc/asterisk/extensions_custom.conf
2> Paste the code between the snips into this file.
===== snip ======
[macro-pincollection]
exten => s,1,Answer()
;we set a variable to prevent looping over and over in IVR, either by accident or malicious
exten => s,n,Set(Attempts=1)
;throw a entry in the asterisk log that system is requesting a PIN entry
exten => s,n(begin),NooP(&&&&&&& Collecting PIN info for ${CALLERID(all)} attempt ${Attempts} &&&&&)
exten => s,n,Playback(please-enter-your&pin_number)
;system listens for DTMF and records it in variable “pinnumber”
;the syntax at the end= 7=MAX DIGITS, 1=attempts if no data, 10=timeout. The “,” value i’m not using in this code example)
exten => s,n,Read(pinnumber,then-press-pound,7,1,10)
;system checks for blank value and asks again
exten => s,n,ExecIf($["${pinnumber}"=""]?goto(macro-pincollection,s,1))
;systems plays back the number they entered in to caller
exten => s,n,Playback(you-entered)
exten => s,n,SayDigits(${pinnumber})
;system asks them to press 1 to accept or 2 to retry
exten => s,n,Playback(if-this-is-correct-press)
exten => s,n,SayDigits(1)
exten => s,n,Playback(otherwise-press)
exten => s,n,SayDigits(2)
;system listens for the value entered
exten => s,n,Read(digi,1
;this section we test for invalid entries. On their 4 try we hangup. We dont want someone looping over and over again.
exten => s,n,Set(Attempts=${MATH(${Attempts}+1,i)})
;if they reach their 4th attempt, system plays some messages and hangsup.
;Change the “4” to a different value if you want to increase/decrease chances
;Change the “HANGUP to RETURN” if you want the system to go back to a different location as set in the CUSTOM DESTINATIONS section
exten => s,n,ExecIf($["${Attempts}"=“4”]?playback(sorry-youre-having-problems))
exten => s,n,ExecIf($["${Attempts}"=“4”]?playback(hangup-try-again))
exten => s,n,ExecIf($["${Attempts}"=“4”]?HANGUP())
;if user presses 1 to confirm, system sends the clal on
exten => s,n,ExecIf($["${digi}"=“1”]?goto(writevalue))
;if callers presses any other digit, system will re-ask them to enter in their number
exten => s,n,goto(macro-pincollection,s,begin)
;If they press “1” as being a valid entry, the entry is written into the file.
;system writes the pin collected to the file ‘pincollected.txt’
;saved as "DDMMYYY-HH:MM:SS, CALLERid, UNIQUE NUMBER (to trace call in logs easier if needed), PINNUMBER
exten => s,n(writevalue),system(echo “${STRFTIME(${EPOCH},%d%m%Y-%H:%M:%S)},${CALLERID(all)},${UNIQUEID},${pinnumber}” >> /tmp/pincollected.txt)
exten => s,n,Return
===== snip ======
3> Save these changes
4> Open up FreePBX
5> Go to ADMIN-CUSTOM DESTINATIONS
6> Create a new destination
TARGET = macro-pincollection,s,1
Description= Collect Pin and save
Notes= blank (you can put something in if you want)
Return = YES
DESTINATION= IVR – ivrtype
(the destination in this case is what you want to have happen when they enter their pin number. In my example I wanted the code to continue on, but if you say “no” for RETURN then the system will hangup after the call.
7> Save changes
8> Open up your IVR and put in an entry in it to go to this custom destination
example:
If they press “2” in your ivr, select “CUSTOM DESTINATION” then select “Collect Pin and save”
9> Save your changes, then submit.
10> TEST!
Dial the number that will trigger this. In my example I all the IVR then pressed “2”
System will ask you to enter your pin, then will play it back to confirm and then if “YES” is selected, it will save it to /tmp/pincollected.txt DDMMYY-HHMMSS, CLID info, unique ID, pin#entered
02042018-11:51:38,John Smith <123-555-1165>,1522680683.98551,123
02042018-11:54:49,Jane Doe <123-555-4455>,1522680864.98560,1234
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If I change the hostname the UCP daemon will not start
You shouldn’t be changing localhost. localhost should always be 127.0.0.1
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High CPU Usage for an idle box
Ok.
Just recalled where I read this and that was that Freepbx 16 “might” be using php7, were speed improvements are significant:
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If I change the hostname the UCP daemon will not start
tm1000,
I can’t have a system just called “localhost” when we have different FreePBX servers in different locations.
I believe if we just get a way to add the hostname to the installer or perhaps it is a way to change it in the MySQL server database.
Options?
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Suddenly can't dial out
Really strange. The line definitely isn’t busy, and when I dial in it picks up the call and works as normal, even hanging up properly so the Obi does know that it’s free. The Grandstream gateway - which is in a different physical location and connected to a totally different line - also still appears as being engaged.
I can’t work out why this would happen. I might have to factory reset one of the units and see what happens!
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If I change the hostname the UCP daemon will not start
In addition, I am getting this error when pressing the refresh button on the dashboard > see attached image. 
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If I change the hostname the UCP daemon will not start
Is this error due to re-installation?
This is the second time installing the system, I selected the same IP address though, but I am leaving the default localhost.localdomain < name on the system for the moment.
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How to delete voicemails
I did not get a change to check it
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Ftp in active mode , backup
Hi,
We have changed our ftp server to accept connection in Active mode and ever since pbx gets ftp connection error while we want to save backup on the ftp server.
and ftp from ssh connection gives me 421 connection error.
There is no firewall set and I see Syc/Syn-Ack/Ackj sequence in pcap, however , I get RST from ftp server when ftp client ( pbx ) is sending Syn from a port more than 45000 to ftpserver:21.
Since we have different pbxs, I can see if pbx is sending request to port 21 from any port under 45000, we get access to the ftp server, but if ports are more than 45000 , we get RST from tfp server.
Since we can not change our ftp server, any idea how to change ports in our pbx? for our ftp client?
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Backup using ftp in active mode
gitim
1m
Hi,
We have changed our ftp server to accept connection in Active mode and ever since pbx gets ftp connection error while we want to save backup on the ftp server.
and ftp from ssh connection gives me 421 connection error.
There is no firewall set and I see Syc/Syn-Ack/Ackj sequence in pcap, however , I get RST from ftp server when ftp client ( pbx ) is sending Syn from a port more than 45000 to ftpserver:21.
Since we have different pbxs, I can see if pbx is sending request to port 21 from any port under 45000, we get access to the ftp server, but if ports are more than 45000 , we get RST from tfp server.
Since we can not change our ftp server, any idea how to change ports in our pbx? for our ftp client?
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Elastix Outbound Caller ID
Hi all.
i have such trouble.
i want to call some numbers and have their self numbers in their phones.
i need something like set outbound CLI = Dilaled number.
please advise,
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Voice Payload size with ATT to FreePBX (asterisk 13) to Riedel intercom payload error expected 120 get 240
I’ve seen after doing a distro and asterisk upgrade that some settings under SIP Settings were confused after the upgrade.
Go over your SIP Settings and [i assume you use] ChanSIP settings to make sure everything is in place.
After making changes please reboot your system.
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Why i get this when creating extension
Not sure which router you use, but most routers you can open/forward ports from specified destinations only. Open only from your SIP Provider and trusted remote locations
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