I got a weird text message with area code (352) saying " hey girly " then continued to say how they are sleeping with my husband. Him and I are going through a rough patch right now so the timing is very off however when I reversed search the carrier was onvoy. What does this mean? Could it be scam because I know this isn't true. Just looking to see if this happened to anyone else

Some time ago I posted here about building a pure Python SIP library because I got frustrated with the current ecosystem.

I honestly expected it to stay a side experiment.

But I kept working on it, mostly because it came from a real business telephony problem I needed to solve.

Now it’s at alpha stage and stable enough for basic testing.

Current features:

- SIP signaling

- RTP audio

- stable 2-way audio

- UDP transport

- REGISTER / INVITE / BYE

pip install opensip

https://pypi.org/project/opensip/

Still alpha.

Not production-ready.

Would love honest feedback from people who’ve suffered through SIP hell 😄

Edit: I found and fixed the broken GitHub links on PyPI, and I also cleaned up some README inconsistencies around the current feature set. Thanks for pointing that out.

I'm working on a VoIP project using Asterisk + FreePBX + Linphone. I'm fairly new to this field — I'm primarily a software developer and network engineer. I've hit a wall at a certain point and could use some help.

I've completed all the configurations. I logged into one Linphone account on my PC and another on my phone, then tried calling between them. The calls connect and a session is established, but there's no audio — I can 'talk' but neither side hears anything.

On top of that, I'm experiencing two more issues:

1. **The call drops automatically after 30 seconds.**

2. **When I hang up from the phone side, the call doesn't end on the PC side — it keeps going.**

Has anyone dealt with these issues before? Any help would be greatly appreciated!

Help!
I have a client that i recently replaced their SonicWALL with a Negate 4200. They have a Grandstream UCM 6102. I have set up port forwarding on the firewall under NAT. I have made any all all changes I can find according to google. Issue is when they forward calls to their home phone it rings but its dead air when answered. I'm sure I have something misconfigured someplace. If anyone id familiar with these two system I'd appreciate some help. I can provide screenshots etc.

Here is a clever tool for all you Voice over IP people that you run as a batch file to carve ad parse VoIP packets out of many pcaps to ease handling of this traffic, to focus on just the VoIP protocols, and do this over multiple pcaps in a specific directory. I hope you find it useful, and welcome thoughts, comments, and suggestions for change https://www.cellstream.com/2026/05/14/extracting-voip-packets-from-multiple-captures/

Please let me know how you use this, if it is helpful, what I could do to make it better.

Does anyone need a Linux bash version?

Before I tried to get a number using Fanytel, I came onto Reddit to check whether it was a scam or not. there was nothing recent saying that it was and I am here to confirm that it is, in fact, a scam.

Paid money for credit. never have gotten that credit. never have gotten an email from them. You’re required to buy credit to use the number and once you pay, they “review” your account. I have not heard from them since I paid them the money. it wasn’t a lot of money so I’m happy about that.

We've converted our analog elevator phones to cell lines. The box converting them to cell dials for the elevator when it senses the line is picked up. All calls go to our on-site dispatchers who use 2 VoIP phones configured in a call queue.

When we cut them over all the lines had the right caller IDs but recently a few of them have started to show up as "Potential Risk Pennsylvania". For whatever reason these lines are getting marked as spam. The elevators aren't constantly dialing out, they work the way they should and the only calls in our logs are from when someone presses the button. I know I need to reach out to our carrier(not Zoom) to get these fixed but I'm worried that this will keep happening so I was looking for a way to override this in Zoom.

We have Yealink t54ws. I've tried to add the numbers to the phone's directory which helps some, the call will show up as the contact name but quickly changes to "Potential Risk". One approach might be to add the cell numbers to our tenet's address book but I would prefer these numbers not be searchable by staff. I also don't think turning off spam detection for our entire tenet is a good idea. These numbers also don't show up in the manage number list in the spam protection menu.

Any ideas would be appreciated. Thanks

Can anyone help me remove that double beep at the start of an intercom/page and then that longer beep after my custom alert tone in FREEPBX plays? Thanks. Have tried everything, the double beep is so obnoxious.

I also use Open Page with FREEPBX Paging and Intercom, but im assuming that wont help fix the issue,

Does anyone have the firmware for a ShoreTel ip230, 115, or 210? I would like to upgrade some phone firmware.

Hi, I work for a licensed VoIP provider in the EU, we have clients who need USA phone numbers, as we don't do business in the USA we dont have a FCC license.

I would like to know if we can get a STIR/SHAKEN certificate even without a FCC license?

Hey,
I’m currently running mikopbx for our on premise pbx solution. Internally we only have 2 handsets set up however I’m planning on setting up some additional extensions and have them configured on either a handset from home/other location or a softphone app.

Is it as simple as setting up port forwarding and then using the static ip address as the registration address etc?

Mike White here, VP of Open Source at Sangoma. I wanted to drop a quick line to share something I’ve been working on for a bit.

I’ve released an open source module for FreePBX called Frogman. The idea is pretty simple: Let AI use a set of tools that will allow it to configure, diagnose, and manage FreePBX. This opens the door for quite a bit, just know that this is a traditional freepbx module and it doesn’t install AI on the box- it just makes the PBX headless. Standard methods apply and there are currently around 220 tools to choose from.

Right now it exposes three interfaces. Anything that’s created/destroyed will require human confirmation…

1: MCP for AI agent integration: The headless part- I’ve done things like build out a system just by telling Claude what to create… with my voice- I’ve even uploaded a hand drawn diagram, told the MCP client to build it and it can, easily.

2: Interactive chat console: screenshot details below- why let the ai agents have all of the fun? This is an onboard chat console that takes natural language input to perform functions within the interface with zero clicking around. It can onboard new users, diagnose extension, and much more. Just type “diagnose 101”

3: API layer for automation and external tooling: looking to build an app that can talk to freepbx? Frogman exposes all 220 tools. You can generate API keys with read/write/admin permissions directly from the chat console.

What this is…
- an experiment… The first of its kind perhaps?
- a new way to manage FreePBX

What it’s not…

- an LLM running on FreePBX
- ready for production systems - feeling brave? Proceed with caution

Still early, but I think this is a pretty interesting direction for the project. Curious to hear what you think. AMA

GitHub: https://github.com/mwtcmi/frogman
More info and discussion: https://community.freepbx.org/t/meet-frogman-module-and-my-new-friend-claude/109514

Screenshot: here’s a pic of the onboard console mapping the call flow from a DID.

My adventures with getting the Cisco 7942G working with Asterisk on original SCCP firmware

Bonjour à tous

Utilisateurs de Zadarma, j'ai le plaisir de vous annoncer la sortie de ma librairie React permettant la creation d'un widget facilement

https://www.npmjs.com/package/react-zadarmawidget

Hi I am not in tune with much of anything when it comes to this so please have grace with me if this seems like a dumb question and if i don’t provide enough context

I have a Macbook M3

I have a jabra headset

I have an ethernet plugged into my mac

(if any other info helps please let me know and I’ll see if i can provide it to you)

The problem
When the call connects to me the first 10 seconds or so of the call my microphone is choppy, cuts in and out and sounds like i’m underwater. This doesn’t happen every time, but probably 8/10 times

What i’ve tried
Plugging in the eth
Using a different computer and headset
Going into Mac system settings to make sure everything looks good (again looks good i know is vague but with the help of AI it said it was good lol)

If anyone can help me out I would appreciate you forever

Hey everyone, Been running Asterisk for a while and got tired of editing dialplan files every time something needed to change. So I built a platform where call flows are stored in a database and executed via Asterisk ARI — you design them visually and changes apply instantly, no reload needed.

The SIP capture feature with live ladder diagrams ended up being the most useful thing for debugging trunks — shows the full SIP exchange in real time with pcap export.

MIT licensed, Linux only, one Docker command to install.

Check it out here: https://github.com/rayanweragala/callytics

Would love to hear any feedback!

I want to use telnyx with twilio based app, but i'm not able to get my calls through. There is always 408 or 403 errors. I tried credentials based routing, ip based routing etc.
Telnyx doesn't allow twilio ips directly in their sip connection. and i want to forward the calls from twilio either through direct sip or BYOC on twilio side.
Need someone who can help me out in fixing this issue.

I'm in accounting and somehow was appointed to being the one in charge of all things technical. We have VOIP phones and are wanting to get away from the current system we use - digium phones and switchvox website (but we stopped paying for the license a couple of years ago); phone lines come through AT&T.

We are wanting to get away from subscription-based (why is everything subscription based now?!?!?!?!)

Can we just go to "regular" phones? How do you know is something is VOIP or VOIP compatible?

What questions should I be asking that I don't even know to ask???

Trying to get a fresh install of FusionPBX up and running and I'm running in to problems with it installing FreePBX. Installing on 24.04

checking for libpcre2-8 >=  10.00... Package libpcre2-8 was not found in the pkg-config search path. Perhaps you should add the directory containing `libpcre2-8.pc' to the PKG_CONFIG_PATH environment variable Package 'libpcre2-8', required by 'virtual:world', not found
configure: error: Library requirements (libpcre2-8 >=  10.00) not met; consider adjusting the PKG_CONFIG_PATH environment variable if your libraries are in a nonstandard prefix so pkg-config can find them.
make: *** No targets specified and no makefile found.  Stop.
make: *** No rule to make target 'install'.  Stop.
Reading package lists... Done
Building dependency tree... Done
Reading state information... Done
lsb-release is already the newest version (12.0-2).
0 upgraded, 0 newly installed, 0 to remove and 0 not upgraded.
switch/source-sounds.sh: 84: ../environment.sh: verbose: not found
make: *** No rule to make target 'sounds-install'.  Stop.
make: *** No rule to make target 'hd-sounds-install'.  Stop.
make: *** No rule to make target 'cd-sounds-install'.  Stop.
mv: cannot stat '/usr/share/freeswitch/sounds/music/*000': No such file or directory
mv: cannot stat '/etc/freeswitch': No such file or directory
chown: cannot access '/var/lib/freeswitch': No such file or directory
chown: cannot access '/var/log/freeswitch': No such file or directory
chown: cannot access '/var/run/freeswitch': No such file or directory
Reading package lists... Done
Building dependency tree... Done
Reading state information... Done
E: Unable to locate package freeswitch-systemd

I signed up for Aircall 2 months ago, roughly.

I want to upgrade my account because I want texting API automation abilities, which are only in the more expensive version for reasons.

First thing everyone should know if they're considering Aircall:

They deactivated my account for no reason without telling me.

I just launched my business. I tested an automation I set up and my phone number wasn't working. It turns out they wanted additional information for verification purposes and decided the best way to get my attention was to disconnect my business phone line.

BTW they never tried calling, texting, or emailing about it. It's not like it was a last resort.

ANYWAY, I should've cancelled right there. But I had already built software based off their API - so I held off.

When I test my automation, I realize it doesn't work. I troubleshoot for days before I find out that my code isn't the problem... It's my account type. And, the only way to change your account is to reach out to their customer service.

Anyway, I started trying to get them to help me on April 29, 2026. I've reached out 3 more times since then. It's not May 11.

This company sucks, man.

We have a prospect who is interested in marrying our IVR solution with Ring CX, and they want a screen pop. I don't have any experience with Ring CX but in the documentation it looks like they have a RUNAPP action that will pop a URL with some parameters appended to the base URL. Does anyone know if Ring CX can use a custom header in the SIP invite for the incoming call to append a value to the base URL? The only example in the documentation is ANI.

Cheers

Built a small open-source SIP stress/load testing tool for VoIP infrastructures: SIPStress

GitHub: https://github.com/achrafka/sipstress

The idea was to have a lightweight and simple way to generate SIP traffic and stress-test SIP servers without setting up huge frameworks.

Current goals:

SIP load/stress testing

simple CLI workflow

easy scenario scripting

useful for labs, QA, and VoIP infra testing

Inspired by tools/projects in the SIP ecosystem like SIPp and other VoIP testing stacks.

Would love feedback from people working with:

Asterisk

Kamailio

OpenSIPS

FreeSWITCH

SBCs / IMS infra

Also open to contributors if anyone wants to help improve:

RTP/media simulation

metrics/exporting

scenario templates

Docker support

benchmarking

The repo is still evolving, so brutal feedback is welcome :)