r/VOIP

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!

https://preview.redd.it/d8lzpz5yjo0h1.png?width=1918&format=png&auto=webp&s=8c8dc71824c86c02b5db50845c4c29fbdd10d2cb

https://preview.redd.it/lnijibxyjo0h1.png?width=1919&format=png&auto=webp&s=366baa8c0f06ada7d7ad9ff4d64ed47146d90712

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

Hello guys, I’m not sure if there are any Russians in this group, but I need some help.

My fiancée recently moved from Russia to Dubai. She works remotely for a Russian company, and they gave her a Russian phone number with an Android phone. Since she works in marketing, she needs to make and receive calls from Russia using Wi-Fi Calling / VoWiFi.

I’ve been reading online that VoIP or Wi-Fi Calling may not work properly in the UAE. I understand this might apply to UAE carriers, but what about a Russian T2 / Tele2 number while roaming in Dubai?

Is there anyone here in Dubai who came from Russia and uses Wi-Fi Calling or internet calls with a Russian number? If yes, how did you activate it and make it work?

I’d really appreciate your help.

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???

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

Hi, recently moved to VoIP and it's not exactly been a smooth ride. I'm hoping someone can help with a problem that my provider (Phonely) seems unable to fix, their 'solution' is to use their in house call screening, which costs more.

Basically I have a BT cordless phone that uses a call blocking feature, where the caller is asked to state who they are and press # before being connected. Well basically pressing # doesn't seem to be 'heard' by the phone.

The adapter they provided is a Fanvil GA11.

Like I say, I'm very new to VoIP so please bare with me if I don't always understand what you're referring to.

Any help please?

Thanks

Hi everyone,

New to the sub and looking for some technical advice.

I’m currently working from Portugal and I’ve been using my Three UK mobile number for daily work calls to the UK. As expected, Three just sent me a warning that they will suspend my roaming services on the 17th due to their Fair Usage Policy.

I need to keep this exact number for professional reasons, and I’m considering porting it to a VoIP provider so I can stop worrying about roaming rules and just use it via an app on my phone (using my local Portuguese data/Wi-Fi).

I have a few specific questions:

1. Reliability: For those who have ported a UK mobile number to a VoIP service (like Zadarma, Virtual Landline, or VoIPfone), how is the call quality for daily professional use?
2. SMS: My biggest concern is 2FA/Bank codes. Which VoIP providers actually support receiving automated SMS/OTP from UK banks?
3. Alternative: Is there a better "virtual" eSIM or service that allows permanent overseas use without the 60-day limit?

I have a stable fiber connection at home and a local SIM for mobile data, so I'm just looking for the most stable way

So for many years I have built, setup, installed PBX and build/owned a SAAS calling platform. I have ALWAYS setup a redundant outbound calling trunk and these days it's usually primary and secondary SIP trunk providers. So, provider A goes down and outbound calls still flow through provider B and customers don't have any interruption in outbound calls.

Now with all the STIR/SHAKEN regs and rules, providers are no longer allowing you to send calls out with caller id's that don't either match a number in your account or a verified number.

THIS does not work for a SAAS platform where they in most cases the customer has hundreds of numbers that are assigned to their customers and all live on carrier A NOT carrier B. The caller ID must he that of the customer's customer, not some random, single caller ID that lives on carrier B.

How are you handling this? Is there a solution with these regulations in the US?

Is it just the reality now that outbound provider redundancy is just dead and calling is just less reliable as a result?

Appreciate any thought or advice on this. Thanks!

Hi guys, as mentioned in the title.... I'm trying to set up BLF on my new Fanvil phone which I quite like, because I need to track activity of my employees and want to have the overview of overall activity of the entire team.

However, the company I work for uses Nextiva which I absolutely can't stand, but sadly, I can't do anything about it.

Nextiva doesn't provide support for Fanvil and as such, this phone has to be manually provisioned & registered as "generic SIP device", which I am okay with but I it seems to me like these bastards purposefully made it extremely hard to get more functionality out of devices that they're not endorsing.

Good people of reddit, is there a way to troubleshoot this and find some sort of fix. I'm loosing my mind, please help.

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’ve started realizing the most dangerous telecom problems usually aren’t catastrophic failures. It’s the systems that: connect successfully, deliver “most of the time,” pass monitoring, return normal logs, and quietly behave differently depending on carrier behavior, timing, routing, device state, or pure cosmic randomness 😭

Those are the systems that slowly destroy your sanity. Especially once voicemail workflows enter the picture. You’ll have days where: one carrier handles delivery perfectly, another delays it, callbacks happen hours later, timing assumptions drift, retries create weird side effects, and everything technically still appears “healthy” inside the dashboard.

That’s the part I underestimated most about telecom infrastructure:
a huge amount of the complexity isn’t obvious failure.

It’s ambiguous behavior. I started experimenting more with DropCowboy Twilio ringless voicemail workflows recently (didn't want to invest the wheel one more time so went with existed option) and it changed how I think about communication systems in general. The further you get into telecom, the more it feels like distributed systems engineering layered on top of infrastructure that occasionally behaves like folklore.

At some point you stop asking:м“Did the workflow succeed?” and start asking: “What version of reality did the carrier decide to use today?” 😭

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 are a VOIP carrier and looking to implement POTS in a box type services. Our client is looking for boxes similar to granite EPIK and Ooma boxes that have multiple POTS lines and Will accept multiple SIM cards and then do SIP connections over the Internet and output as Analog POTS lines. We can rig something up using grandstream ATAs and LTE modems, but those POTS in a Box solutions are so clean. Is anyone aware of any hardware that we can use with our service that competes with those products?

I'm considering a VOIP number in order to have a secondary business phone number on my actual mobile phone, connected via SIP. I was looking at using voip.ms for this.

I don't want my name and address tied to the VOIP number so that anyone could get my name/address/other personal identifying information just from the phone number alone.

Is there anything I should know about this before I sign up?

Thanks!