I would report this as a bug. Firewall is dependent on sysadmin because of hooks it uses but sysadmin should NOT be dependent on it. They should wrap calls in a module status check.
↧
Unable to disable firewall [forked and retitled by mod]
↧
Unable to disable firewall [forked and retitled by mod]
I believe that the OP just wants to disable (turn off) the firewall, so it doesn’t block any traffic.
This can be done at Connectivity → Firewall → Settings → Disable Firewall. You may also want to stop Intrusion Detection.
Disabling the firewall module is something different – it removes the module from the GUI and fwconsole commands.
↧
↧
FTP from FPBX17 (Debian 12 Bookworm) to Windows Server 2019
We have spun up a new FPBX17 server on a Dell ESXi server. We are currently working on setting up our FTP backup to a file share server (also on an ESXI server, on the same VLAN too). In FPBX, FileStore and the backup and restore module are configured correctly. On the Windows server, we do not have IIS blocking any IP addresses. We have also confirmed that our FTP service user account is working and that we have the correct credentials in our FPBX system. I followed the FreePBX deployment process (onto Debian 12) by using the instructions here: https://sangomakb.atlassian.net/wiki/spaces/FP/pages/230326391/FreePBX+17+Installation
Is it possible I am missing an FTP component on the Debian server? The Windows server is configured correctly, as we still use FTP to back up our old FPBX15 server.
Help a noob out, plz.
Thoughts or opinions are greatly appreciated. If more information is needed please feel free to ask. Thanks!
↧
Got locked out on setup
I rebooted, GUI is showing that firewall is not enabled, and everything looks good so far.
But, honestly, it’s not a good start. Shall I expect more surprises like this?
↧
Softphones: Echo Problems / Recommended Softphones
Hello community,
we run a FreePBX with appr. 100 extensions. Most extensions connect via Grandstream-Devices to the FreePBX. Additionally we have a few Fritz!Boxes and 2 softphone-apps.
The 2 softphones (different apps, using udp for on-prem-communication) produce a lot of echo. One started to echo after upgrading from V15 to V17. The other used to echo with V15. We tried different headsets. An OS-Upgrade from Win 10 to Win 11 didn’t change anything.
Going through the FreePBX-extension-settings I did not find any value for e.g. “echo reduction”.
Is there a way to fight echos in FreePBX?
Why would there be an echo after upgrading from V15 to V17?
Is there a list of recommended softphones (Linux and Window-Clients) for FreePBX?
Thanks a lot
Martin
↧
↧
Call deflection 302 Moved Temporarily not working on newly installed FreePBX 17
OK, so, in order to test the “Transfer() needs to be made upon a PJSIP channel” premise, I just created a custom context in FreePBX by the name “mobile_ivan”, and added the folowing to my extensios_custom.conf:
[mobile_ivan-custom]
exten => _.,1,Transfer(PJSIP/sip:mymobilenum@my.sip.provider:5060)
same => n,NoOp(TRANSFERSTATUS=${TRANSFERSTATUS} PROTOCOL=${TRANSFERSTATUSPROTOCOL})
same => n,Hangup()
After that, I made sure the call was not progressing further in the dialplan by sending my DDI directly against the mobile_ivan context. And that made it! I got a 302 message sent over the trunk and a response from my sip provider (not exactly what I expected, but probably some additional variables need to be set in the headers, discussing with them at the moment, this is a whole other).
Now I need to find a way to do that not from the incoming DDI call, but once it reaches the queue and rings its members. I’ll open a new Topic here, as I think this is a whole other other.
Big big Thanks to @blazestudios and specially @jcolp for providing the insights!
↧
Softphones: Echo Problems / Recommended Softphones
Echo when using headset should not be expected.
I would try to DISABLE audio processing inside softphone in this case - acoustic echo cancellation and/or AGC. If you want to keep AGC on, make sure maximum gain is not set to some crazy value.
You could use Audacity to record audio from microphone while playing something from other app - recording should be almost silent.
Back in the Windows XP years I saw some weird cases with audio output routed to audio input through Windows audio mixer. It should be rare these days, but still worth checking.
↧
Dial a number on one extension, and transfer it to another immediately
There is a dialplan implementation of Originate
↧
FTP from FPBX17 (Debian 12 Bookworm) to Windows Server 2019
What is the actual issue?
↧
↧
FreePBX 17 - Sangoma P330/320/310, EPM with DPMA enabled
Crossposted from Reddit:
So I’m not 100% sure where I am screwing up.
HTTP provisioning Port is at 84 HTTPS is at Port 1443 Restful apps is at Port 82 with secure at 3443
DMPA is enabled and configured EPM is configured MAC address is loaded into EPM and everything there is configured
I can provision microSIP and confirm that the line and extension works as intended.
I can configure other phones manually and confirm that they work as intended.
I cannot get the sangoma configuration server to properly configure this phone.
The error I get is as follows every time the phone goes in queries this server:
SSL 6 [SSL_ERROR_ZERO_RETURN] (Read ret: 0 Len: 65535)
Some confirmations:
Confirmed not self signed certificate - sourced free from Cloudflare
Confirming that I have added the MAC into EPM
confirming I also added the extention and account to the Mapping, along with template and configured by server is selected.
↧
Call deflection by endpoint (302) not working, probably due to Diversion Header
Hello community,
i am pretty new to FreePBX.
I already figured out some things but with one problem I need some help.
I try to use call forwarding enabled by a phone/endpoint (not using die Asterisk Featurecode).
Every time there is a diversion header added with my extension. Lets say my extension is 15 than the header looks like
Diversion: "Name" <sip:15@12.10.x.20;user=phone>;reason=unknown
My provider wants a real number or no diversion header at all but I could not figure out a way to change oder remove this header.
This is something else i tried:
[macro-dialout-trunk-predial-hook]
exten => s,1,NoOp(Executing custom predial hook to remove Diversion)
same => n,Set(PJSIP_HEADER(remove,Diversion))
same => n,Set(PJSIP_HEADER(add,Diversion)=<sip:+4xxx123456789@sipconnect.ipfonie.de>\;reason=unconditional\;screen=yes\;privacy=off)
same => n,Return()
It just keeps the header i mentioned above.
Forwarding a call with asterisk feature Code ist working.
Any ideas?
Thank you.
NetRacer
↧
Call deflection by endpoint (302) not working, probably due to Diversion Header
You are fighting the system when you explicitly set Diversion headers. It sounds like you are trying to undermine the automated processing, rather than using REDIRECTING - Asterisk Documentation to work with the system.
Sorry, I have only used this sort of thing on raw Asterisk, and that was many years ago, so I can’t tell you how to do it cleanly with FreePBX.
↧
Users are unable to flag administrator posts
Nothing will change, chris Maj is employed to silence dissent and dictate to us, he loves the extreme power sangoma have granted him, allowing his power tripping and abuses and gets off on it, he has free reign to make up his own rules, ghosts so he can flag posts from them all making posts seemingly hidden by community when in fact are not.
Open source advocate not, he is a sangoma advocate.
↧
↧
Users are unable to flag administrator posts
Now there you go, you did the ad-hominem thing, now he has an excuse to ban you for any amount of time between one day and infinity 
To give him his credit, Mussilini at least made the trains run on time ( for a few years (Legally it would be hard to define that as ‘ad hominem’ I think even if the intension was there))
↧
FTP from FPBX17 (Debian 12 Bookworm) to Windows Server 2019
In the Linux world, you will find that scp/(rsync over scp) is a preferred method over FTP/SFTP for many reasons. Windowsnow has WSL to make that easy
↧
Users are unable to flag administrator posts
Staying on topic and using my opportunity for my once a day comment on this post. I think everyone should message their account reps and everyone they know to ask that admin flagging be restored and slow mode be disabled.
Also since every feature of discourse is the next potential passive aggressive jab there should be a clear policy in place when features are used, why they are used and time frames. You know transparency. Show the community there is a method to this and it isn’t all just made up and arbitrary
↧
Support, Quality, Cutting corners
really? I thought we made our money on time for handling, configuring and installing, I make no money on freepbx its unsalable, about time you got a clue, but your history on here shows that unlikely
↧
↧
FreePBX 17 - Sangoma P330/320/310, EPM with DPMA enabled
Are you seeing this error on asterisk cli ?
SSL error means phone is not able to make HTTPS connection with the PBX /Asterisk.
Try to enable DPMA logging to see if this gives you any clue.
https://sangomakb.atlassian.net/wiki/spaces/PG/pages/32669947/EPM-DPMA+for+Digium+Phones#Troubleshooting
↧
FTP from FPBX17 (Debian 12 Bookworm) to Windows Server 2019
First of all, try to use Linux ftp command to confirm you can connect to your windows FTP server from Freepbx server.
If FTP connectivity is fine then something could be from Freepbx level.
As @BlazeStudios asked , please share the issue detail and if its backup related then what does the /var/log/asterisk/backup.log file says?
Regards
Kapil
↧
FreePBX 17 - Sangoma P330/320/310, EPM with DPMA enabled
I’m following this error in the following log
/var/log/asterisk/full
The biggest problem that I’m running into seems to be a discrepancy of whether or not I can provision using TLS.
I’m also not able to get a fully functional provision even if I use TCP as I’m not sure that endpoint manager is actually creating a functional provisioning file.
I don’t seem to have a means to be able to log into the web portal of a P330 phone and tell it where the server is either which is kind of odd. As soon as it connects over TCP the password of the phone changes and the web portal is disabled on the phone itself. But then the GUI on the phone also becomes useless saying that nothing’s been activated or anything.
↧
