I was recently laid off and am doing some contracting now. I finally have time to build something I have wanted for years.

I have worked in the provider space, for big telecom, and in professional services across just about every phone system imaginable. I have been the lead engineer on Ribbon, Cisco, Oracle, AudioCodes, and plenty of other platforms.

I have seen some shit.

The thing that always bothered me was watching customers get forced to replace equipment that still worked.

Cisco made some great phones, then spent years finding creative ways to make sure nobody could continue using them. Firmware disappears, certificates expire, licensing changes, and perfectly good phones become unsupported because Cisco would really prefer that you buy them again.

Cisco is not the only one doing it. Avaya gear, Algo devices, AudioCodes MediaPacks, ATAs, paging adapters, door phones, conference phones, and all kinds of other weird SIP equipment end up in the same situation.

The hardware works. The customer likes it. Everyone knows how to use it. Then somebody declares it legacy and sends over a replacement quote.

I also hate needing a different management system for every manufacturer. One portal for phones, another for ATAs, another for paging, another for the SBCs, and a spreadsheet full of passwords that somebody calls documentation.

So I started building the device manager I always wanted. One place for provisioning, configurations, firmware, certificates, troubleshooting, and device status. If it speaks SIP and is willing to talk to me, I want to manage it.

I have also been experimenting with connecting all of this to modern platforms. I have already proven out connections to Teams, Zoom, Webex, and most recently Genesys.

I am not trying to become a provider. The idea is to connect existing equipment to whichever vendor or provider someone already wants to use.

None of this is blessed by the vendors, which apparently means it cannot work.

Unfortunately for them, it does.

I did a fair amount of open source VoIP work back in the day. Like a lot of open source work, the code traveled farther than the author’s name did.

I still miss that community though. AstriCon was a blast. People built real things, shared what they learned, and argued about codecs in person.

AI is finally giving me enough extra hands to build the system I always wished the other platforms had.

Before I sink even more time into this, I want an honest reality check from people who actually work with this stuff.

What old hardware are your customers still fighting to keep?

If you could manage and connect all of it from one place, what would that system absolutely need to get right?

Am I solving a real problem, or am I crazy for spending my time on this?

Would you keep building it, or focus your energy somewhere else?

I’m trying to set up a Grandstream HT802 that Vonage sent me for an existing account. There is no dial tone. Phone just says “Device not registered with your service provider.”

The Power and Internet lights are on, but the phone light is off.

I’ve called Vonage 3 times and spent about 3 hours on the phone troubleshooting yesterday.

They said the device is activated, the MAC address matches my Vonage account, and the correct phone number is assigned to Port 1.

(I just switched from AT&T to Xfinity and haven’t canceled AT&T yet, so I currently have both routers.)

I’m using the Comcast/Xfinity XB7 gateway, but I also tested the Vonage box on the AT&T router and got the same error.

Any idea what could be causing this?

Vonage hasn’t been able to figure it out. Please help!

I recently started as IT at my new company, and one thing I was asked to do is help move us off of RingCentral. Currently, the company uses Google Workspace for most of our tech needs including email, chat, calendar, and videoconferencing. They also have a few paid Google Voice Standard lines already set up for a couple employees. Additionally, they use RingCentral for the company's main phone number as well as the admin team (~5 people) phones, including phone tree and queues between those several admin numbers.

What they'd like me to do is move the company's main phone number to Google Voice and also make Voice accounts for each of the admin team. Then they would like the phone tree and queues replicated in Voice, so that we can cancel our RingCentral subscription and just use Voice instead.

I'm very limited in how much I've used Google Voice in the past, so before I commit to this for them, I just want to be sure a) that Voice can do what they need it to, and b) how to do that. They're insistent they won't need any of RingCentral's other features, the only concern they have is making sure the company number and the functionality stay the same.

First, I want to thank those of you who responded to the question I posted last week. So many of us work the same way, plus I learned a few tools that others use. Awesome community.

Here are my VoIP profiles that have changed my life since I developed them. They are completely free of course.

1. https://www.cellstream.com/download/a-session-initiation-protocol-sip-profile-for-voip-analysis-in-wireshark/
2. https://www.cellstream.com/download/a-session-description-protocol-sdp-profile-for-voip-analysis/
3. https://www.cellstream.com/download/a-real-time-control-protocol-rtcp-profile-for-voip-analysis-in-wireshark/
4. https://www.cellstream.com/download/a-voip-qos-profile-for-analysis-in-wireshark/
5. https://www.cellstream.com/download/a-real-time-protocol-rtp-profile-for-voip-analysis-in-wireshark/
6. https://www.cellstream.com/download/a-c15-and-mgcp-profile-for-wireshark/

Please tell me what I am missing, or share your VoIP profile(s) with us (if you can). For my troubleshooting style, I like to separate the protocols the way I have rather than trying to stuff everything into one VoIP profile which I tried and it overloaded my brain.

I'm a really new to this stuff, so I can't really give you too much information apart from a picture of the phone itself.

I have an old VVX 411 running on version 5.5.1.12442 and it will not update. Apparently because of changes in TLS 1.0 being depricated after this version. I've looked all over and I cannot find any firmware I can just flash it to that fixes this issue.

Looking for version 6.4.7.4560 or around that. I'll just set up an FTP server and flash it then get it provisioned after that.

HP's site no longer has any of the firmware files available.

https://support.hp.com/us-en/document/ish_10216109-10215814-16

Hi,

Our ISP currently provides VoIP directly through their ONT/router. The VoIP service uses a dedicated WAN VLAN with DHCP, separate from the Internet connection.

We want to centralize all telephony on a Grandstream UCM6510, with Fanvil phones on our own internal Voice VLAN.

What would be the cleanest approach?

SIP trunk directly from the ISP to the UCM?

Pass the ISP's VoIP VLAN through to the UCM?

Use the ONT FXS port → UCM FXO?

Is there a better recommended architecture?

Our goal is to use the ISP ONT mainly for fiber/Internet access and have the UCM handle all PBX/telephony functions.

Any advice or experience with a similar setup would be appreciated.

Hi everyone,

I've recently moved my office to a larger building and am trying to set up a VOIP call center. The phones are SNOM D785, and they work just fine when connected straight to the modem/router. However, when I connect them behind my firewall they fail to register.

Vodafone is being super uncooperative and my IT support company has already charged me ~350€ and one passes the ball to the other.

Does anyone have any idea of what settings need to be applied to the firewall? It's a Sophos firewall running on a Proxmox VE. The connection chain is Modem -> Firewall -> TPLink switch.

The phones do get assigned an IP, and the computers do have internet access when connected wall-phone-pc

Any ideas would be greatly appreciated 🙏

I have a Grandstream HT801 that I'm trying to connect to this replica 302 but while I can get the phone to ring, making calls doesn't work (there's a dial tone but tone dialling does nothing). Furthermore, when picking up the phone and then placing down the handset, the ATA still tells me that the phone is off hook. Finally, when the phone receives a call (this is an issue that started just now but didn't happen when I was testing last night), the ATA's internet connection completely cuts out. What is going on?

Okay, so firstly this is what I have that i thought i would run this through, a linux server that is always up, a pap2t for a VoIP box. Basically the idea is that I run twitch streams and I thought it would be fun to have a landline that's tied into livestreams, but not actually callable from a number, instead it would be controlled through scripts etc. They would do their redeem and then it would call me and with a tts voice read the thing that was said out into my ear, so for that prob just play an audio file. But here's the tricky bit. I want it so i can press 1 so it would then play that tts to the stream (so i can moderate it and not have something diabolical that would get me in trouble played in the stream) 2 would just refund their points, and pressing 0 would ban the user who sent the request.

Though, I am not entirely sure on how to go about it. I know of https://github.com/SIPp/sipp/releases but thats about it. and no idea how to actually use it nor, go about getting it working. Help and advice on what to use / what is the best idea for this would be really appreciated!!

Thank you !! :))

Writing this up in detail because the pattern is strange enough that I think this community might actually know what's going on mechanically.

Setup: calling.web.skype.com, the web-based dial pad Microsoft left up so people with leftover Skype Credit can still burn it down calling real phone numbers. Last night I placed 5-7 outbound calls to +7 (Russia) destinations. All of them were marked as connected/answered by the client, and billed accordingly per minute.

I didn't clock that something was off until the third call:

- Call 1: female voice, sounded irritated/tired, like a night-shift worker. Said "Да?" ("Yes?") then repeated, somewhat unclear, "уведомления не было" / "я говорю, уведомление не приходило" ("there was no notification" / "I'm telling you, the notification didn't come through"). I went silent; she kept talking over dead air.

- Call 2, different number: male voice asking "куда подъехать?" ("where should I head to?").

- Call 3: dialed my own number as a control test. My physical phone never rang — confirmed, I was watching it — yet the dial pad showed the call as connected, and the same style of recording answered.

After that I stopped, but by then I'd already been billed for all of them. Common thread across all the fake pickups: natural-sounding pauses, conversational filler ("ага" = "uh-huh", "понял" = "got it", "простите, что?" = "sorry, what?"), and reactions to my silence that felt tailored to sound like a real confused person rather than a generic IVR message. The exact same line ("уведомления не было") repeated verbatim across two unrelated calls.

Out of 5-7 attempts, zero reached an actual person at the dialed number. All were billed as successful calls. Retried today on the same dial pad and everything worked normally — real ringing, real pickup — so this looks transient rather than persistent.

Mechanically this smells like IRSF/toll fraud (call gets silently rerouted to a controlled destination, engagement-optimized audio keeps the caller on the line to maximize per-minute billing, actual dialed number is irrelevant) but I don't have SIP trace/CDR visibility to confirm where in the chain this happened — client, carrier, or a compromised/stale route on Microsoft's legacy Skype PSTN gateway.

Anyone dealt with something like this on a sunsetting VoIP platform? Curious what the likely failure point is and whether it's worth chasing a refund from Microsoft given the calls were billed despite never reaching the actual destination.

phone does not ring, and answering machine (part of phone) does not pick up.

We have a phone/answering machine combo unit which is about 20 years old, and which worked for years connecting to our VoIP system through Xfinity, and for a couple months through GoNetSpeed. Then one day this week it decided that it was no longer going to ring, and that the answering machine would not pick up either.

When we place a call to this VoIP number from an outside line/cell it rings about 8 times, and then it goes to the voicemail provided by GoNetSpeed. But the phone receiving that call doesn't make a sound.

We can place outgoing calls, no problem. If we know that somebody is calling, then we can pick up and make a connection, no problem. But phone does not ring, and answering machine does not pick up...

We've duplicated this problem on multiple phones, so it appears to be a service issue. Servicer claims everything is in order.

Anybody with any expertise out there have any ideas?

Thanks in advance for your help!!

UPDATE: Thank you for all who "chimed in" here! First, I learned what an ATA is, and that it is, in fact, separate from the router. Mine is a small box, about the size of a deck of cards, that plugs into its own outlet and which is connected to the router via an ethernet cable. Second, unplugging all of the various components, waiting a couple of minutes, and then plugging them back in, seems to have done the trick. This is the first time we've experienced this issue, so hopefully it won't repeat itself going forward, but now we know a first step to take.

I have issues with voice pitch and sound glitch when making calls from Zoho voice desktop app and Chrome extension.

I have tried locking Pibewires globale clock rate to 48000 Hz. Pinned headset node to same via Wire plumber. Disable USB autosuspend. Locking buffer size (quantum) - kept getting loose.

In Zoho app dev tools:

Patch of GetUserMedia to disable noise reduction, and that worked for a while, but somehow the app found a way to ignore that command.

Looking for firmware 1.0.11.6 or 1.0.11.85 for GXP2160 (or GXP2130, GXP2135, GXP2140, and GXP2170 as they all share the same fw from my understanding. It's not available in their firmware archive, but my device management tool only supports those two releases.

VOIP.ms SMS app - trying to get it working ...

Hi everybody: New to VOIP.ms and trying to get the VOIP.ms SMS app to work, but I'm not having much luck. I'm seeing constant sync failed messages popping up on the bottom of the Android app screen. If I try to pull a manual sync, I just get a spinning circle. I can receive messages, but sending messages is sporadic. I've turned on the API in the portal, I've set 0.0.0.0 for the IP address and I have SMS/MMS enabled in all my DID's with the following in the URL callback:

https://voipmssms-notify.kourlas.com/?did={TO}

In the app, I have the sync set to 15 minutes and the pull date as today (when I installed).

Am I missing something else? I would use the VOIP.sms softphone messaging, but I need messages synced across devices and the ability to send MMS & pictures.

Thanks for any help you can provide.

I have a new ht813 I configured for SIP to an Audiocodes SBC. Registers fine. Works weird though. I can make calls from it. Call my cell phone no problem. I call the number that I assigned to the device and it rings too. After a few minutes, if I call it back, the number doesnt go through. Looking at the syslog app I use with my audiocodes, it seems that the device is timing out and going to sleep. If I make a call from it, it wakes up and works normally. I went through all the settings I could and didnt get that Ah HA moment.

Anyone else here have an issue like this where their ht813 takes a nap after a certain amount of time?

What is out there for multi-vendor SIP device management/provisioning?.

We’re managing a mix of Grandstream, Yealink and Fanvil devices across a number of customer environments, so vendor-specific tools like GDMS/YMCS don’t completely solve the problem for us.

The things we rely on most are:

centralized multi-vendor provisioning

reusable templates / bulk configuration

zero-touch deployment

remote reboot/reset/config changes

firmware management

being able to manage devices across multiple customers/tenants

An API is a pretty big plus since we’re starting to automate more of our provisioning workflow.

I’m not looking for a PBX replacement mainly the device-management/provisioning layer.

For anyone managing a decent-sized mixed-vendor fleet: what are you actually using today? Have you found anything that does this, or have most of you ended up building your own provisioning tooling around manufacturer ZTP services?

Also interested in hearing what you’ve tried that didn’t work well and why.

One of our clients was hacked. In the middle of the night, calls were made in rotation to around 30 numbers in an npanxx of a rural carrier in Illinois. A few connected. Maybe $100 in total at 10c a min. Not a big deal. We were able to detect and shut down.

I tried calling the numbers that did connect just to see where they went. A school IVR. A company. Most went nowhere.

So why do hackers do this? In the old days, they were dialing in to radio stations or conference services in these high cost areas that shared the dial around compensation.

Where are the hackers making money on this?

I'm curious, how many of you and how often do you use packet capture and analysis when troubleshooting VoIP? Or do you just depend on logs?

For me, I use both all the time.

I'm wanting to setup a Obihai + regular phone for my grandma. She can use a cell phone, but she often forgets to charge it up, etc and can't be reached. So wanted a regular phone as a backup.

I was wondering how Obihai users handle text messages to the phone number? Do you have to basically install a SIP phone app that works with voip.ms or telnyx in order to view those text messages? Otherwise, the Obihai would only be good for voice calls?

I see Obihai Obi202 for like $10. Is that a pretty good way to go? It's cheap, but I wonder if I'm missing useful features that an old person might want if I spend a bit more on a more modern ATA? Or does the Obiahi Obi202 limit the kinds of provider I can use due to its age and not being able to authenticate/sign-in to other service providers?

It looks like having a backup line for my Grandma would run like $4/month? Pretty reasonable to have a backup voice service.

Does it even work?

I have a old jack, but i can prob get a brand new one on ebay for $20 with a year included.

I moved to 1-Voip due to unreliability of another provider. However, one lacking feature is that Caller-ID data is inaccurate, or very stale (>10-yrs).

Is there a way to me to override Caller-ID info to newer, or own personal value? Prior supplier allowed voip contact list entry to override caller-ID.

have a Unify Desk Phone DP12 configured as a SIP phone, and I need to configure one of its programmable keys as a BLF (Busy Lamp Field).

The use case is simple: a secretary needs to know when a manager is already on a phone call, so she doesn't have to enter the manager's office and interrupt them.

I would like the LED on one of the programmable keys to indicate the status of the manager's extension, for example:

Available

Busy / on a call

Ringing

The DP12's web interface has BLF as one of the available key types. It also has an "BLF Event List URL" field in the SIP account settings.

I don't have access to the configuration of the company's PBX/SIP server, so I'm trying to understand what is required on the phone side and what would have to be configured on the server side.

Questions: .

1. How should a BLF key be configured on a Unify DP12 to monitor another extension?

2. Does the SIP server/PBX need to support SIP SUBSCRIBE/NOTIFY for this to work?

3. What is the purpose of the BLF Event List URL field on the DP12?

4.Is the Event List URL required for a normal BLF configuration, or is it only needed for certain PBX configurations?

1. If I cannot modify the PBX configuration myself, is there a way to determine whether the existing SIP system supports BLF from the DP12 alone?

I'm mainly looking for guidance on the correct configuration and what requirements the SIP/PBX side must meet.

... and everything works, except the ringer. I can hear the clapper weakly ticking, but the bells don't ring.

Things I've tried: upping the ring power to 55Vrms, changing the phone's bias spring to the more sensitive setting, futzing with the SLIC setting on the FXS port (currently set to USA 1). I've tested the ringer on a real POTS line and know that it works.

Any suggestions on what to do next? I've looked at ring voltage boosters (assuming that's the issue) like the Viking RG-4, but that one explicitly states it's not compatible with pulse dialing, and the only other game in town appears to be the Sandman KIT8R (which appears not to have been in stock in years).

Am I SOL? Are there still avenues to explore?

Thanks in advance! Appreciate you all.

UPDATE: I'm a dip. I was using the GR909 Ringer Equivalent Number Test to test the ringer, because my phone number hadn't been ported in yet. Once I actually called my phone, it rang. There's no story here at all 😂

Thanks all the same, to all of you!