Hi all,
I’m trying to bulk import extensions via the cli using the command:
fwconsole bulkimport -vvv --type=extensions 'filename.csv'
I don’t see any errors, only the following output:
“Importing bulk extensions”
However the extensions are not being created.
Anyone has an idea? In the full log I also don’t see any errors?
Hi,
Our FreePBX deployment has been around for 8 years.
We don’t usually check the CDR reports. Today, we did.
For some reason, it is empty - meaning that no data gets returned when use the default time range. So we put a CallerID of a number we know called in. Nothing.
Can someone else tell what is wrong or get some steps to troubleshoot?
I have no correction to give, that is what my perusal of the log indicated.
Just starting a topic to say that,
Transcript injection into email works well (GitHub - simontelephonics/transcribe-with-azure) but it would be a whole lot nicer to see the transcript in the visual voicemail list rather than having to go back to the email inbox for it.
I’ve been toying with this idea as well, but haven’t looked at HAProxy. I’ve been looking into Kamailio instead.
I store mine in the database and the users can view them, print them and delete them.
How do they access the transcripts?
Haproxy can filter any tcp or htp/https connection but is not a SIP proxy, kaimalio can’t protect your UCP/provisioning/admin/ssh/mail connections but is my proxy of choice for SIP.
Through a portal I designed for them. I’m not using FreePBX for what I’m doing. I was just giving an overview. Add a new table for the transcripts, update the voicemail/ucp to display the information about the transcripts. I just use a table style for listing CID, time/date, confidence score and view/delete icons. When they view it, a modal style window pops up and they can see the transcript and have options to print or delete the transcript.
@billsimon I currently do this, and a hack job it is… But it works.
First, I am on PBXact Cloud. This is important because one of the “features” as Sangoma calls it (sarcasm) is no root/ssh access to your box. So, I had to find a way to work around this, and still get the data I needed.
First I set each ext to send voicemail emails to a catchall email address. ext+myemail@myemailserver.com
I then have a python script that extracts the attachment, ext, and vm data from the body of the email, and stores it in my own MySQL DB.
Another python script checks for new VMs in the DB, and if detected, runs a transcribe python script. I am using an API call to assemblyai.com to do the VM transcription, and then store the transcription text back in into MySQL.
Once transcribed, we call another script to fetch the now complete MySQL data, send an email to the user that received the voicemail, and do some file cleanup (moving the WAV file to an archive location).
This of course is a simplified expiation, and if you want to know more, feel free to PM me and we can discuss code, etc.
If you do have SSH/Root access to your PBX, you could probably just create a script to transcribe (using assemblyai or another transcription service of your choice) and store text into the freepbx DB, and then have that script call the function to email the transcription to the user.
To anyone else reading this in horror… I am well aware this whole setup is chaos, but it gets the job done given the restrictions I have on the PBX. 
In Advanced settings, do you have Enable CDR Logging set to ON?
Also, did you try to just hit SEARCH without typing a number or any other search parameters? This should display all calls without a filter.
I know sometimes when I try to search for a number, I get no results. It’s usually because I’ve left off the preceding 1, such as 15555555555.
If yes to both of those, have you made any DB changes? CDR data is stored in the freepbx MySQL Database. Do some research on accessing the DB and see if you get any data by actually going into the DB.
what’s in filename.csv ?
@shane8johnson OP @billsimon already linked to a script he created that transcribes using Azure Speech, and it works great. He’s looking for a way to put that information into the visual voicemail list (ucp, sangoma phone, etc.), which I think would be useful as well.
The caller calls into the queue. Operator 1 answers the call. However, after this the call to Operator 2 does not end. This happens sometimes.
The queue consists of only static agents.
Here are my queue settings:
Queue No Answer = No
Call Confirm = No
Call Confirm Announce = Default
Wait Time Prefix = No
Alert Info = No
Ringer Volume Override = No
Ringer Volume Override Mode = No
Restrict Dynamic Agents = No
Agent Restrictions = Call as Dialed
Ring Strategy = ringall
Autofill = No
Skip Busy Agents = Yes
Queue Weight = 0
Max Wait Time = Unlimited
Max Wait Time Mode = Strict
Agent Timeout = 20 seconds
Agent Timeout Restart = Yes
Retry = 5 seconds
Wrap-Up-Time = 30 seconds
Member Delay = 0 seconds
Agent Announcement = None
Report Hold Time = No
Auto Pause = No
Auto Pause on Busy = No
Auto Pause on Unavaliable = No
Max Callers = 0
Join Empty = Yes
Leave Empty = No
Penalty Members Limit = Honor Penalties
Hey @hichem,
Greetings from Steve, As best of my knowledge to configure a new SIP trunk for termination calls in France, you’ll need to choose a reliable SIP trunk provider that offers services in your region. After selecting a provider, they will provide you with SIP trunk credentials, which you will then input into your PBX or VoIP system.
SIP provider in France: Téléphonie SIP pro, trunk SIP | SFR Business AWS
Thanks
Hi @moghimigiti This issue has already been addressed in FreePBX 16.
For now, I have backported the fix and pushed it to the edge module for you to try. However, please note that we are no longer issuing fixes for FreePBX 15. We strongly recommend that you consider upgrading to FreePBX 16, or you might want to try the FreePBX 17 beta.
fwconsole ma downloadinstall qxact_reports --tag=15.0.21
fwconsole chown
fwconsole r
Thanks
Running with: /usr/sbin/fwconsole backup --backup=‘xxxxx-xxx-xxx-xx-xxxx’ --transaction=‘xxxxxx-xxxx-4851-92ff-xxxxxxx’ >> /var/log/asterisk/backup_348725bd-xx-xx-92ff-b968b88deb59_out.log 2> /var/log/asterisk/xxxx-xxxx-4851-xxx-b968b88deb59_err.log & echo $!
Running Backup ID: xxxxxx-xxxx-xxxx-xx-xxxxxxxxxx
Transaction: xxxxxxxxx-xxxx-xxx-xxxxxx-xxxxxxxx
Starting backup backup
This backup will be stored locally and is subject to maintenance settings
Backup File Name: 20240522-085019-1716367819-15.0.17.55-19264.tar.gz
Working with announcement module
Adding module manifest for announcement
Working with arimanager module
In Common.php line 146:
file_put_contents(): Only 0 of 472 bytes written, possibly out of free disk
space
backup [–backup BACKUP] [–externbackup EXTERNBACKUP] [–dumpextern DUMPEXTERN] [–transaction TRANSACTION] [–list] [–warmspare] [–implemented] [–filestore FILESTORE] [–restore RESTORE] [–restorelegacycdr] [–modules MODULES] [–restoresingle RESTORESINGLE] [–backupsingle BACKUPSINGLE] [–singlesaveto SINGLESAVETO] [–b64import B64IMPORT] [–fallback] [–useinfiledb] [–skiprestorehooks]
Thank you Steve
You should also look at the other ISPs: Orange, Bouygues and Free.
There are several independent French VoIP providers, for example ippi.fr .
And, you might consider one of the A-Z providers such as voxbeam or anveodirect .
Hi Stewart, Thanks very much for this, we didn’t had any Outbound Route. It worked fine after we added this.
Best Regards,