Quantcast
Channel: FreePBX Community Forums - Latest posts
Viewing all 228149 articles
Browse latest View live
↧

FreePBX Project Leadership Update

June 28, 2024, 10:35 am
$
0
0

End of an era. You have truly been amazing in all faucets. You really helped me on day one (knowing nothing about Asterisk or FreePBX), and never stopped after. Your responses and ideas, and your general encouragement to everyone in the forum moved mountains. You had profound impacts, not only just with us, but also with the customers and people you touched, through us.

There have been many transitions, and while those are never easy, they can also be quite hard. You helped keep things stable and moving during that time. I don’t think things would be where they are without you. Thanks for everything!

↧
Search

FreePBX 17 RC1 Blog

June 28, 2024, 10:40 am
$
0
0

Doubtful, should be find. Ultimately what you need depends on extensions, calls, transcoding, recording, and other jobs.

↧
↧

FreePBX 17 refusing to even try to send mail after VM left

June 28, 2024, 11:09 am
$
0
0

Hello all,

We’re trying to get FreePBX 17 (running on Ubuntu 22.04 LTS) to send mail via Google Workspace when someone calls and leaves a voicemail. We have a script that behaves like sendmail -t, and can successfully send mail using Gmail’s OAuth API with this script (I’ve just tested it; the script still works).

We have been able to get FreePBX to send mail exactly one time. We’ve gone back to the exact state the PBX host was in and tried again, and we still cannot send a second email from PBX. The voicemail inbox works, incoming calls work, the extensions all work, the only thing that’s broken is FreePBX will not send mail.

What do I do? How do I force it to always send mail? Should I just disable the email components of FreePBX and manage it myself?

↧

Sysadmin Port Management CLIc

June 28, 2024, 11:52 am
$
0
0

@pramarajan This was the trick I needed, thank you very much! You wouldn’t happen to know the db entry for the HTTPS Address dropdown on that sysadmin port management page too, would you?

↧

FreePBX Project Leadership Update

June 28, 2024, 11:55 am
$
0
0

Lorne,
Another Big thank you your way… countless others and I appreciate your contributions through the years.

-Taylor Letham

↧
↧

FreePBX Project Leadership Update

June 28, 2024, 12:53 pm
$
0
0

Thank you @lgaetz for all of your insightful interactions on this forum, and for steering the FreePBX ship recently. Hope to continue seeing you here, and best of luck to you in your new endeavors!

↧

FreePBX 17 refusing to even try to send mail after VM left

June 28, 2024, 12:57 pm
$
0
0

mail logs needed and clarification of MTA being used (Debian’s defauilt is exim, but postfix is generally preferred.man sendmail should confirm at the top, what your box is using)

↧

FreePBX 17 refusing to even try to send mail after VM left

June 28, 2024, 1:01 pm
$
0
0

Our MTA is Google Mail (using a Python script that behaves like sendmail as a communication bridge.) I tried generating logs but it’s like the script is never getting called in the first place. Mail command is set correctly (/usr/local/bin/emailproc for reference)

Here’s some additional context on why our setup is the way it is:

  • In our initial research and design phase for our voicemail notifications, we found that FreePBX’s (or perhaps Asterisk’s) built in mail notification system did 99% of the legwork for us; GMail (via this API) expects a Base64 encoded MIME message with minimal JSON overhead. FreePBX/Asterisk generates this MIME message (that we then base64 and wrap in the relevant overhead), and expects to be able to print this into the stdin of sendmail (or something that acts like it). Configuring and maintaining a full mail stack for PBX is not within our capabilities, but a small Python script that acts like “sendmail for Gmail”, combined with a service address our organization already had? Easy.
  • When we first piloted this phone program, FreePBX 17 was in an earlier state in the beta, and ISOs had not been generated yet. Our organization requires a review of software and operating systems before use within the company; Ubuntu 22.04 was already vetted, Debian 12 was not. We chose to deploy to Ubuntu to reduce our internal process overhead, and because of Ubuntu’s origins as a descendant of Debian. We’ve been rolling upgrades out to our existing instance as they get approved.
  • We chose FreePBX 17 knowing it was in beta because our understanding at the time was that by the time our processes were done, FreePBX 17 would be released, or at least in the release candidate stage, and upgrading would be non-Herculean, at the least.
↧
Search

FreePBX 17 refusing to even try to send mail after VM left

June 28, 2024, 1:10 pm
$
0
0

Won’t be easy helping you with your home brewed python script :wink: , as a first point of strategy install postfix, it’s probably more robust and /var/mail/mail.log will likely be instructional

If your are still not feeling up to it, there are at least 101 postfix recipes for you to simply\ relay to your personal gmail account

↧
↧

FreePBX 17 refusing to even try to send mail after VM left

June 28, 2024, 1:28 pm
$
0
0

We can’t.

Postfix isn’t approved by our information security department, and we do NOT have the resources to maintain a full sized mail stack like Postfix or exim. The script itself isn’t even the problem; FreePBX is just pretending it doesn’t exist.

Do I have to rename it to /usr/bin/sendmail?

This isn’t at my house. This is at the office of a large (1000+ employee) organization.

↧

FreePBX 17 refusing to even try to send mail after VM left

June 28, 2024, 1:37 pm
$
0
0

Any binary needs to have execute permission for the asterisk user, if it is not in the $PATH just fully qualify it , which sendmail you get depends on the users and Asterisk will look in home directory first

which sendmail
echo $HOME
echo $PATH
 cat /etc/passwd|grep asterisk
`
↧

FreePBX 17 refusing to even try to send mail after VM left

June 28, 2024, 1:46 pm
$
0
0

Support for Google Business accounts is probably not best found here.If your proprietary script doesn’t work have you considered mal-chimp mail-gun or whatever

↧

Upgrading VVX Phones with Old Firmware

June 28, 2024, 5:25 pm
$
0
0

I discovered that one has to manually create an XML file for the phone to read when updating from a local server. To help others in this same situation, here’s what it’s supposed to look like:

<?xml version=”1.0″ encoding=”utf-8″ standalone=”yes”?>
<PHONE_IMAGES>
<REVISION ID=””>
<PHONE_IMAGE>
<VERSION>1.2.3.4</VERSION>
<PATH>http://localip/directory/</PATH>
</PHONE_IMAGE>
</REVISION>
</PHONE_IMAGES>

The 1.2.3.4 is the version that is reported in the drop-down box when the phone checks for firmware on a custom server. The path is exactly that - where the firmware is on your LAN. This file MUST be saved with DOS newlines - phone doesn’t recognize the file when it’s saved with unix/linux newlines.

I tried this on a single phone and found that I had to copy the 3111-46162-001.sip_57x.ld to 3111-46162-001.sip.ld because the phone was looking for that file. After that, the phone got the file and even though it looked like it wasn’t updating, it did! The one phone I left connected on my bench that was running 5.4.something is now running 5.7.1.2205 :slight_smile:

Unfortunately, when I tried to update it from the Polycom server to make the final jump to 5.9.8.5760, it fails - “Failed to fetch available software from the Polycom Hosted server. Please try again later or contact your Network Administrator if the problem persists.” :man_facepalming: Maybe they’re having network problems and it’ll fix itself… :man_shrugging: (Regardless, I’m done with this for the week. :laughing: )

While I’m fairly confident that I can dump the 5.9.8.5760 into a directory accessible to the web server on my PC, I would rather update directly from Polycom’s server if possible.

↧
↧

Upgrading VVX Phones with Old Firmware

June 28, 2024, 5:45 pm
$
0
0

Scratch that - I tried bringing a production phone up to 5.7.1 & it was able to grab the final update from Polycom no problem. Must be something goofy with the phone on my bench; I rednecked it into our network last minute the other day, manually setting the network without going through the entire provisioning process, so likely just missed something while doing that. :man_facepalming:

So while I don’t have an answer as to why the phones with old firmware won’t upgrade directly from the polycom servers other than it’s something to do with SSL, I found a viable workaround. :slight_smile:

↧

Configuring FXS Module TDM2400P on FreePBX 13.0 server

June 29, 2024, 2:30 am
$
0
0

I am running a FreePBX 13.0 for IP telephony service. The 35 extensions are working well.The system used to work perfect before it crashed and I have had to reinstall with my little knowledge on FreePBX. I have an asterisk TDM2400P FXS card with analog PSTN already set. I need assistance on configuring the ports of this card to extension Lines in order to make outbound calls and receive the calls from the external customers. Kindly assist

↧
Search

Forward phone call to first available Stasis-based extension

June 29, 2024, 2:54 am
$
0
0

I have a set of extensions all configured at the moment as follows:

[from-did-direct]
exten => 200X,1,NoOp()
 same => n,Stasis(myAppX)
 same => n,Hangup()

where X is a number in {1, …10}. For example, when one calls 2005, the app ‘myApp5’ will take care of that call. That app interacts with the caller in an automated way (no human interaction).

On the other hand, I have successfully configured a SIP trunk and an inbound route so that I can have a call from that phone number being routed to one of these extensions, but only one.

My goal would be to implement the following scheme when someone calls that phone number:

- call routed to 2001 -> myApp1 starts. If 2001 is busy then:
- call routed to 2002 -> myApp2 starts. If 2002 is busy then:
...
- call routed to 2010 -> myApp1 starts. If 2010 is busy then return busy tone.

I was wondering what is the best way to achieve this? Eventually I want to be able to scale this to hundreds of extensions. I’ve been reading a bit and I have sketched this solution:

  • I set the inbound route from that phone call to 2001. In the configuration of 2001 (Advanced → Optional Destinations → Busy) set 2002 when 2001 is busy. Repeat: set 2003 when 2002 is busy, …, set 2010 when 2009 is busy.

I have read some solutions based on ring groups, but I think it would not work for me, because of the Stasis nature (since all extensions can automatically start).

However, I am not entirely convinced of my solution, since it is hard to scale to more extensions.

I appreciate any hints or ideas.
Thank you.

↧

Forward phone call to first available Stasis-based extension

June 29, 2024, 4:41 am
$
0
0

I’m not sure why you are using FreePBX, as you seem to have effectively completely bypassed it.

Also what do you mean by by Busy. I’m guessing you mean your stasis app is only serially reusable, not re-entrant, in which case I think you could do this using group counts

↧
↧

Installing Zerotier on Raspberry Pi 3

June 29, 2024, 6:18 am
$
0
0

I would like to install Zerotier on my raspberry pi 3 that i have had FreePBX running for years. I followed an example found on here but it keeps failing so does anyone have
a different way. GPg2 doesn’t work on my version FreePBX 14.0.17 on my pi 3
I tried google to find different ways and tried different installs but none have worked.
I have Zerotier working on several raspberry pi 3B with allstar hamvoip with great results.
Dave
wa2kjc

↧

Forward phone call to first available Stasis-based extension

June 29, 2024, 6:22 am
$
0
0

Well, I needed to forward the audio stream to a different server and the only solution I found for that was the Asterisk RESTful Interface (ARI) and the concept of External Media Server (External Media and ARI - Asterisk Documentation).

Is there a way how this can be done with FreePBX? I don’t think so, right?

On the other hand I need to rely on FreePBX functionalities such as the bulk handler.

By busy I mean that a call has been already established between two parties.

↧

Forward phone call to first available Stasis-based extension

June 29, 2024, 8:13 am
$
0
0

How does this differ from a normal intracompany trunk?

Nothing in your original question indicated that this had actually happened, and the normal time to treat a callee as busy, is just before making the initial setup request, not on answer.

At the moment, I think my original answer is correct, assuming the ARI is needed.

↧
Viewing all 228149 articles
Browse latest View live
Search
<script src="https://jsc.adskeeper.com/r/s/rssing.com.1596347.js" async> </script>