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

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?

TL;DR: Go\Trunk is a middleman and isn't a provider and charges you more than higher tier SIP Providers, while routing you to low tier providers that could cause issues with your business.

We have been using Go\Trunk as our SIP Trunk provider for a while (~2-ish years) and we were actually pretty happy with them. But every month I decided to pick up the phone and make sure calls route properly. First i call my ISP provider knowing theyre on a VoIP service so its SIP to SIP to make sure that works and it did. Then i check just going to my cellphone. My personally cellphone (AT&T) wasn't ringing. So my first immediate thought was somehow, my AT&T Spam guard app blocked it. It didn't. Nothing there from the company number. So i called another employee, he didn't get a call. He also has AT&T but doesnt use the spam app. So i called another cell phone that wasn't an iPhone that was also AT&T...didn't go through. On my end, from the desktop phone it would ring, but the calls were not going through to the person i was calling.

We run FreePBX here so i went into a shell and checked the asterix logs and...no STIR/SHAKEN headers attached to the call after it hits Go\Trunk. Which obviously means AT&T is seeing our call is illegitimate and not even letting the call go through to the destination. We get a 200 back, but its fake. I guess its what AT&T does to make malicious callers think their calls are going through, but they aren't? I guess, I don't know that one. So i tried a number I knew that was a T-Mobile line, same thing. Tried calling someone i know who has Verizon, same thing. Now this is a problem because clearly its nothing on our end, its the providers end.

So I submit a support ticket to Go\Trunk and tell them what I just found and basically....what the heck?

The first response was "While they investigate, we have temporarily routed your US outbound calls through a different provider." A "different provider"? Are you not our "provider"? So i said back "Ok, well, what have you found so far that is the problem? Can i please have a more technical answer?" They replied back "I just switched your USA calls to another carrier provider and raised a ticket for the original carrier to investigate further." So their tech support doesn't even know what happened and the fact that our business line was "routed to another provider" was them telling me - "We are not a SIP Trunk Provider, were a middleman between the customer and low tier SIP Trunk providers."

Now, this might be common knowledge to some, but this was news to us and we used them for 2+ years and never had issues until we decided to do this test. NO WHERE on their website do they say this. That they, in essence, are infact, NOT a provider, but they will ROUTE you to a provider they probably have contracts with and clearly, we were moved, unknowingly, to another provider that did not put STIR/SHAKEN headers attached to our outbound calls, which is a red flag to telecom companies, and it drops the calls and (atleast the way AT&T does this), makes our PBX think the call was routed and we get a 200 OK, but its a lie to the phone system. So, this is just a warning to those that DON'T know this because I just caught them doing that, they do not disclaim on their website anywhere they are just a reseller to other providers, in fact they say "Go trunk is a world leading SIP Trunking solution." So they arent saying they are a provider, but they are saying they are a "solution" lol. They also say "Why GoTrunk? Because we set new standards in SIP Trunking!" Well, pretty low standards there.

We have ported our number to a higher tier provider that is infact a "provider", of course I won't say who just to keep to the rules in here as best as I can and its sad to know we were actually paying alot more, to get alot less. We were naive when we implemented having a PBX at the time, we hired better networking specialists, we audit ourselves better now, we have done our research, and its a mistake we won't make again.

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.

Hello. I have a new business that requires an "office number", I do not want this number to be the same as my personal cell phone number. I want to post this office number online and on my website, without exposing my personal cell phone number.

I need to get a new phone number that I can display online, and whoever calls that number would directly call my personal cell phone number, without me having to post my cell number online.

It would also help if I can see texts sent to that "office number", read voicemails for calls I could not answer, or be able to leave a voicemail response to people, asking them to leave a message.

I am trying to move some accounts over to a virtual line but Telegram refuses to accept the number. It gives an invalid number error because it flags the voip carrier block. I do not want to buy a physical sim card for a single login code.

Edit: Swapped to a mobile line from BEE-SMS for the telegram registration instead of fighting with the gvoice settings. It went through instantly.

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

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’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.

I need to call UK mobile numbers from Malaysia with a UK mobile number. I am a UK citizen based in Malaysia and need to call a lot of UK mobile numbers for work related reasons. What’s my best option as zadarma have told me no.

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 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 called, 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.

If I sign up for an account, I can't get past the verification process because no matter what angle the camera is at, it will not accept either my personal information card or my passport. Does anyone else have this problem?. I am in australia in case no one else gets this problem.

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!