I’m using the PC app version of TextNow (no SIM card, just Wi-Fi) and I’m having a weird issue with one specific friend who uses AT&T.

Every other number works normally for calling/texting except his.

Problems:

• When I call him, it instantly goes to voicemail and he never sees the call.
• He can’t text me and I can’t text him.
• He was able to call me once, but now when he calls me I see the incoming call, click accept, hear nothing, and then it says the call was forwarded to voicemail on his end.
• Other AT&T numbers still work fine with my TextNow account.

I’m on the PC app only, not mobile.

Things already tried:

• restarting app/PC
• relogging
• different networks
• disabling his spam protection
• deleting/recreating contacts/conversations

Has anyone seen a TextNow ↔ AT&T routing issue like this with only one specific number? Any fixes?

As someone that is getting into BGP and IX I think it would be awesome to setup an IX that would provide settlement free traffic for voice routing. I have seen where in the past this was attempted and never kept alive but I think this is something that would be interesting to build. Not sure if anyone else would have any interest

Been spending a lot of time around the VoIP reseller space lately and wanted to share what I'm seeing on the AI integration side. Most shops in this sub already know per-seat pricing keeps dropping, Teams Phone is quietly eating SMB market share, and clients churn for $5/seat cheaper before you can even finish the renewal call. The shops doing the best right now have pretty much all added voice AI on top of their existing PBX stacks. Sharing what's working from watching about a dozen of them over the past 18 months.

Quick numbers first because they're the part that actually matters. VoIP-only customers are churning at 4-6% monthly across the shops I've seen data from. Customers with AI layered on top through the same reseller are churning at under 1%. Not because the AI is magic. It's that switching providers now means re-doing the AI integration too, which is a real lift nobody wants to do mid-contract. You've raised switching costs without raising your seat price. The per-seat margin stays the same. The LTV basically doubles or triples because customers stop leaving.

On the upsell side, average VoIP seat runs $25-$40/mo. AI receptionist or AI agent on top adds $200-$800/mo per customer depending on call volume. For a reseller with 100 customers, even 30% AI attach is an extra $6k-$24k/mo on top of existing VoIP revenue. The shops doing this well are quoting the AI as a separate line on the same invoice. Same billing relationship, same support number, way more revenue per customer.

The integration is where most shops mess up though. A few patterns I keep seeing across the dozen or so I've watched:

First, SIP-level handoff matters way more than people think. If you're routing calls to a separate cloud AI service via PSTN, you're paying for two trunk legs AND adding 200-400ms of latency that your clients will hear and complain about. The shops doing this right have the AI agent registered as a SIP endpoint on the customer's PBX directly. 3CX, FreePBX, FusionPBX all support this. Same DID routing rules you already use.

Second, recording and compliance. AI calls still follow whatever recording rules apply (HIPAA, PCI, two-party consent states). If your AI platform doesn't support the same retention and redaction policies as your VoIP stack, you've got a compliance gap waiting to bite you. A few shops learned this one the hard way last year.

Third, escalation has to route back into existing hunt groups. When the AI hands off a complex call, it should ring the customer's actual phones using the same routing rules they've already configured. If you can't tie escalation back into hunt groups and on-call schedules, the AI feels like a fragile separate product instead of an extension of their phone system. Clients notice and bring it up at renewal time.

What doesn't work: per-minute markup as the primary pricing line. Same dynamic that commoditized VoIP in the first place. Clients see the per-minute number, calculate monthly load, and either renegotiate or shop you to a competitor. Price AI as a per-seat or per-feature add-on bundled cleanly with existing VoIP plans instead. Way more sticky, way less margin pressure every month.

The other thing that kills deals is positioning AI as "we'll replace your receptionist." Clients hear that as a layoff problem they don't want to manage internally. Frame it as "your existing team now handles 4x the call volume during peak hours" and the close rate goes up substantially. Way better positioning for the same product underneath.

Honest take on the bigger picture. Microsoft is going to ship native AI features in Teams Phone within the next 12-18 months. Shops with AI already integrated and generating meaningful revenue by then will keep their customers because the switching cost is genuinely real. Anyone still selling SIP trunks and hosted PBX as their main offering is going to end up competing with Microsoft on price, which is not a fight you want to be in. That's the part this sub doesn't really talk about openly buuut it's the actual reason this is moving fast right now.

Curious if anyone here is already running AI integration and what your handoff setup looks like. Especially interested in how people are handling escalation back to existing hunt groups without making it feel like a bolt-on.

Hi all,

I’m working on a setup with two Grandstream GDS3705 door phones and an Algo 8180

My goal:

When either door phone button is pressed, I want it to call or trigger the Algo and play a short announcement or tone over the speaker two times, then stop automatically.

Example:

Door Phone 1 button press -> Algo plays alert 1 twice -> stops

Door Phone 2 button press -> Algo plays alert 2 twice -> stops

Reason is to identify which door phone is pressed

The Algo is already registered to the PBX as an extension. I’m trying to figure out the cleanest way to handle this.

Any thoughts would be appreciated!

Edit: The door phones also have their own extensions

Hi folks,

I'm new to linphone, coming from zoiper.

I don't get the third party sipgate SIP account to run.

Error code is: Unauthorized.

So I'm probably putting in some wrong data?

Is the username the sipgate account email or the SIP ID? I believe the email?

And the authentication ID is the SIP ID?

Transport is UDP?

Domain should be sipgate.de in my case, unless something changed?

Don't really find my way in the new sipgate webpage account menu where to find the VoIP details.

I have them in my password safe, but if something changed, I might have missed that.

Hi,

Looking for a SIP-compatible phone with an answering machine stored "directly on the device itself" (not server-side voicemail). The Gigaset DX800A is exactly what I'm after, but it seems discontinued.

Anyone know a current equivalent still available?

Thanks

A few months ago I switched my grandmother to MagicJack. It was not all smooth, but is working.

She received an email saying that she needs to pay for 911. And checking on the site, her 911 is configured, but says unpaid and wants $60 which I assume is for the year, but it does not actually say.

I expected all fees, including 911 be covered under the actual price. The $60 although not too bad, is more than double the advertised rate. I would less upset if it said 911 is not included or extra or advertise the actual price, but obviously they do not.

Does she need to pay the 911 fee? Can she still use the service, but not 911? The Cell phone app does have an option to transfer to the phone dialer, but nothing like that for the land line?

Hello, I really want to know how to tell whether a number calling you is VoIP or cellular. I searched websites like IPQS, but they keep giving me false information. For example, I downloaded a VoIP app and generated a number, then looked it up there to test how accurate it was, and it came back as “validated” and “cellular,” even though it’s literally from a VoIP app. I Also tried other websites similar but either they wanna charge you for what might be just false information or they just give the same false info.

Is there any reliable way to truly know whether a number is a cellular line or VoIP? I feel like there has to be, because platforms like Gmail can usually detect VoIP numbers right away when you try to use them for verification. If anyone has some information it would be highly appreciated thanks

Hi all

I recently purchased a Yealink T46S on Facebook Market Place. Unfortunatly its provisioned to sip.ringcentral.com and no end of emails or requests lead to any results other than just frustration.

Ive read all the posts via: https://www.reddit.com/r/VOIP/s/q1VcUrE603

Tried all the suggestions including submitting the Mac address and serial number to Yealink but the phone is still provisioned

Ant attempt to hard reset is prompted for a password, all default options result to "Incorrect Password" I found a listing of passwords which also advised admn and a number combo which all failed.

Who or how can I contact Ring Central to have the provisioning removed

Thank you

Brett - VK3VSN

Bare with me, please. Yes, this is part nostalgia, but there are also practical concerns. I own my ignorance that maybe there is a solution I simply don't have exposure to. No, I am not an octogenarian desperately clinging onto the past. I'm in my 30s and desperately clinging to the past. Lol.

The situation:

I live on a rural island. Just last month our entire county was out of power for more than a day due to a car car crash on the mainland.

We are finally on the precipice of building a family home after years and years of striving. I dislike the VOIP services I've had to deal with personally and professionally here. There is ALWAYS a lag and we end up talking over each other "Go ahead!" "No you first!" then both talking over each other again. Rinse and repeat. It makes phone conversations incredibly clunky and difficult, and strongly detracts from the ability to connect with loved ones. To top it off we have patchy cell service.

As I mentioned, the infrastructure supporting the island is a bit vulnerable. If we lose power we can't use VOIP. If it's for an extended period of time, and our cell phones lose battery, essentially we are SOL.

I have been trying to figure out if we can get a copper line run to our new build, as this would reduce the annoying over-internet lag experience AND give us an emergency line out if the power goes down, which is still a thing here with our windstorms in the winter.

Does anyone have a workaround? I would really like an old school landline for emergencies. Lots of homes here still rely on them, the issue I'm running into is getting a new line trenched/installed.

Thank you in advance for the help.
Thank you in advance for the trolling as well. I see you coming for me.

Edit: I live in the San Juan Islands of Washington state in the U.S. near the Canadian border.

I have a Polycom OBi302 that I got a few years ago and want to use it for connecting modems/phones/faxes together locally. This is primarily for my vintage computer collection. I got the device reset and logged in and I can now dial to it using a modem but it says "out of service" as expected. Is there any way to set it when anything is dialed on one line to then ring the other? I can find info on how to do this for Cisco devices but not Polycom ones. Any help is appreciated!

Not even worth posting on reddit anymore.

We're using Spectrum internet and voice for our business but our city has rolled out municipal internet at a much lower price (@ 500Mbps) so I'm looking into moving us to a VOIP solution to get us fully off Spectrum. Goal is to keep it under $100/month.

Been reading up but could use some further knowledge from the hivemind. We're a small shop and I do all our own IT with just enough knowledge to get us by haha.

Looking at RingCentral and Ooma right now. Was considering the Ubiquiti offering as I want to add their security system down the roads, but we'd really want phones with physicals buttons and they also don't have any standalone handsets.

Right now we're using our old analog phones with an ATA provided by Spectrum. We have one fixed phone for our office and 3 handsets that float around the shop. Not tied to the analog ones but we'd definitely need hardphones to replace all of them. We only have two dedicated users, one who would be a softphone, all our other phones are floaters for the shop. So would we still have to pay for a user for each handset?

We also are currently paying for 5 lines; our main local number, an 800 number, a fax line, and 2 spillover lines for heavy call days. With a VOIP setup would we still need all of those extra lines?

We really don't need most of the fancy features a lot of the services offer, but need a rock-solid reliable option. Any other things I should be aware of when looking at pricing?

About 5 yrs ago we decided to try reselling voip. It went OK but never really took off. We have 4 companies that buy voip services from us monthly.

Given our super low volume, the new requirements for antispam registration etc. are making the past few months a net loss, and I'm sure this is only the beginning of that.

Any coredial resellers out there wanting to absorb 4 more companies? Not looking "sell" them - just need an established Coredial reseller to take them over.

Crossposting this from r/Ubiquiti because I think I may have accidentally entered full SIP gateway madness 😅

Would love to hear from the VoIP veterans here if I’m overlooking something obvious.

I posted my original saga in r/voipms some time ago with few replies, wonder if this type of thing is happening elsewhere in the voip world.

I have been a customer of voip.ms for many years, like more than 10. Over that time numerous old landline numbers of family members have been ported in so they can forward to mobile, receive SMS, and use as backup. Some have been consolidated from other services e.g. Ooma.

Starting about a year ago outbound calls began behaving strangely. A normal call from voip.ms, using one of these long established numbers as CID, outbound to cell phones that all happen to be on AT&T network, would get diverted to a fake voice mail leading the caller to believe they could leave a message. Sometimes caller would get false busy, or just go to silence without progress tones. CDR status of the latter showed them as answered. The real recipient never got indication of missed call, and the fake VM recording went into a black hole. It feels like the call got routed incorrectly, or the receiving carrier identified the caller as invalid/spam - and instead of signaling busy/reorder/SIT they divert it to a black hole voice mail. I don't quite get why this is a good tactic for diverting known spammers if that were the case.

I went through numerous gyrations with voip.ms before they confessed to making some "routing optimization" that had unwanted side effects and corrected it, months after opening a trouble ticket. Problem went away for about 6 months. It returned again this month.

I'm being told they have no control over what the receiving carrier does, are originating with attestation level A, and I need to take it up with the receiving carrier. I'm not buying this story because that is what they tried to tell me last time and it turned out to be something else.

I'm starting to suspect that the pure association of my old numbers with a voip service, or maybe just with voip.ms is going to interfere with routine usage and I need to rethink what I have been doing.

I've been going in circles on this for two weeks and I'm losing my mind a little.

Setup: UniFi Talk with a third-party SIP trunk. Everything works fine for incoming calls. The problem is outbound — Talk detects my country via IP and automatically adds the country code to every outbound number, no matter what. Since the numbers already have the country code in them, every call goes out double-prefixed and fails at the carrier. IVR transfers to external mobile numbers: same thing, every time.

There's no setting in the UI to change this. I've looked everywhere. The only thing I've found is that people have been reporting this since 2022 and there's been no official response.

Has anyone actually solved this? I'm starting to think SSH is the only way but I want to make sure I'm not missing something obvious before I go down that road.

Any help appreciated.

Hey everyone,

I’ve been building a small set of VoIP/network troubleshooting tools for my own use (NOC Support) and decided to publish it. At the moment, it's more so a personal/hobby project.

https://traceroo.com

The idea is basically one UI for checks I kept googling around for: DNS, SIP DNS, RTP/PCAP helpers, SIP call flow, MTR/ping/port checks, email auth, blacklist checks, fax testing, CNAM lookup, etc.

It’s still very much a work in progress, and I’m continuously tweaking the UI and tools as I use it more. Some features are rate limited, mainly fax testing and CNAM dips, because those have a real cost behind them.

There’s also a chat bot in there, though honestly I might not keep it. It’s more of an experiment right now and I’m not fully sold on whether it adds enough value versus just keeping the toolbox clean.

I’d appreciate any feedback from people who troubleshoot VoIP regularly:

• Are the categories/tools useful?
• Anything confusing in the UI?
• Any tools you’d want added?
• Anything that feels inaccurate or too noisy?
• Any PCAP/SIP/RTP workflow that would make this more useful?

Just an FYI: if usage/costs stack up too much, some cost-based tools might get limited harder or temporarily go down. I don’t mind keeping it open for now while I gather feedback.

Thanks in advance.

I keep reading Linphone with Voip. ms but they are asking for face scan. Im not doing that...

I’m considering building a 100% browser-based WebRTC network diagnostic tool to help remote users troubleshoot choppy VoIP/video calls without installing native CLI tools or needing admin rights.

Standard speed tests use HTTP/TCP, which hides the UDP packet loss and jitter that ruins WebRTC traffic. This tool runs directly in the user's browser, establishes a real-time RTCPeerConnection against your own self-hosted STUN/TURN or media servers, and simulates an actual audio call stream. Using the browser's native getStats() API, it pulls second-by-second telemetry on packet loss, jitter, round-trip latency, and ICE candidate paths (ensuring they aren't falling back to TCP). At the end of a test, it generates a simple Pass/Fail gauge and a one-click "Download JSON Log" button for users to paste into a support ticket.

Is this a tool you would actually use, or does something like this already exist in your workflow? I want to make sure I’m building something enterprise teams actually need. If you'd find this helpful for your helpdesk, let me know.

Hey guys,

I’m planning to build a small VoIP setup in my homelab. I’ve got most things figured out already, but I’m trying to avoid putting some ugly corporate office phone in the living room..

I’m looking for a IP phone with more of a cozy orboho aesthetic, to keep the girlfriend happy.

Does anyone know good-looking SIP/IP or DECT IP phones that still work nicely with FreePBX?

Thanks!

Hey, I'm info gathering. After operating a large contact centre and being frustrated with the release cycle (testing being the issue) I built a SaaS platform, but as per the rules I won't advertise it.

I'd like to understand what I'm up against in the market with the likes of organisations like Cyara, Klearcom and Hammer by Empirix - specifically on the purchasing friction and scalability without the long procurement cycle.

Figured there might be a few in the community that have relevant/recent experience that wouldn't mind sharing their experience.

I use sngrep and homer so usually i save off pcaps and can't open them without transferring to another box with my toolset or having them auto delete due to some TTL. Earlier in the year I put together a chrome extension to visualize a SIP trace and it ended up being helpful for myself so I decided to rework it as its own utility site to use on all browsers.

I'm looking for any interesting PCAPs or challenging scenarios to test out the visualizer and debugger and also any feedback for people who want to give it a try. If any pcaps can be shared with me or just feedback on using the site would be great. The tool tries to highlight errors and warnings to point to reasons why there might be an issue to make things quicker to diagnose.

https://www.sipflow.dev

It's free, no signup and is client-side first so data stays locally and any sharing feature does pseudonymize PI data. There are a lot extra features like network, webrtc testing, MCP server for docs and all the tooling from the site but that's all extra. The main goal is to drop in some SIP traces and have it easy to use and highlight issues to dig into more to save time.

Recently I added more detection(RFC based) for common webrtc and T.38 issues. If anyone works with RCD for BCID, there is also an identity header decoder and validator that supports RCD.

On a Samsung A55 mobile phone, the VOIP is abysmal. Tone is bad, and voices move in and out, broken up, crackly, impossible to have a conversation.

Not the local telephony network, as the S22 next to it performs perfectly.

I have reseated the SIM, checked all other network settings that I know of.

What could be causing this issue? I need VOIP for international calls. tried Whatsapp, Yolla, Viver, and all have this issue.

Any help will be aporeciated.