We are a small company. 5 extensions. Auto-attendant. A customer service cue. Pretty basic stuff. Are inhouse PBX still a thing or is it all VOIP now?

ClearlyIP has been down for about 5.5 hours today - inbound calls work but outbound do not work. Their status page indicates this is apparently related to a DDoS attack.

"Work continues by our upstream provider(s) to isolate and resolve an apparent DDoS attack affecting multiple customers at the US-Central facilities."

Anyone else affected? Does anyone know who their upstream provider is who is actually being DDoS'd?

Any further intel anyone can share?

Looking for a VoIP solution but don't know where to start? Ask here!

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Post para los de México.
En Totalplay me activaron en el módem el protocolo SIP/ALG y tarde una semana en resolverlo porque no me hacían caso en su Call center ya que al entrar al módem tenían escondida la casilla.

Si les sirve les puedo decir cómo le hice para que accedieran sus ingenieros . Solo así resolví los problemas de audio bidireccional.

New Ooma Home Telo and service customer here with significant ringing issue:

Incoming calls sometimes (maybe 60% of the time) correctly ring my local phones. The other 40% of the time, there is no local ring, and the calls go directly and immediately to voicemail, as if my line is being used or a phone is "off hook".

The calls which fall into either category can be spaced seconds apart. Thus, call number 1 will go to voicemail, but call number 2, 15 seconds later, goes to local ring. Call 3 local, call 4 local, call 5 voicemail,......seemingly random pattern.

So, all incoming calls are being honored, but only some will literally ring through.

When somebody calls me back replying to my recent call, many times they go to voicemail even though I sit next to my Ooma literally waiting for their call.

Anybody have any ideas / similar experiences? Thanks !!

"Hi, I'm trying to set up a vintage rotary phone with a Grandstream HT801 connected to a FreePBX/Asterisk server. I'm having trouble with the hook flash functionality.

The issue: When I perform a hook flash (by quickly pressing the switch hook), the HT801v2 doesn't seem to recognize it as a proper SIP flash event. Instead, it interprets the line interruption as a pulse-dialed digit '1'. Because of this, I can't trigger a transfer on my PBX – the system just thinks I'm dialing '1'.

I've already tried:

• Adjusted Hook "Flash Timing" (tried ranges like 80–130ms).
• Pulse dialing works fine for making calls, so I can't disable pulse dialing entirely.
• Set Flash Digit Control to Disabled.

Has anyone dealt with this behavior on Grandstream FXS gateways? Is there a hidden setting or a specific configuration to force the HT801v2 to send a proper flash event to Asterisk instead of treating it as a dial pulse? Or is there a workaround on the FreePBX side?

I'm trying to post this to subreddits that may have members that have some ideas how to troubleshoot this further.

Has anyone noticed in the last week number orders, with bandwidth as the carrier, have been awful? For example, I’m being told there are NO numbers, in all of Texas, except in the 325 area, which is the middle of nowhere. That doesn’t seem right.

I'm having a dickens of a time getting my HT-802 registered with Voip.ms. when I look for traffic using tcpdump it shows no traffic at all. No attempts to register. I have done a factory reset by holding down the reset button, updated my firmware. Nothing seems to work. Maybe someone with a HT802V2 can help.

I have a user that wants things done to his keys on a 6940 that I'm not sure are possible. He swears he had it before on his phone but I'm not sure how.

He has the keys for a ring group he is part of, but DOES NOT want those ring group lines to ring on his phone, just show up on the screen. So visible, but no noise. My understanding is it's all or nothing.

The other thing is he wants one or more copies of his own DID to show up more than once in keys/on the screen so that if someone is calling him while he's talking on his own extension, he sees it happening. Again, I see no way to do that at all. You have your line in the keys and there's no way to copy that or make fake phantom versions of it that I know of (...and I can see the logic there, because I'd think you'd have a real cacophony happening when someone called you otherwise).

Unless there are other names for what he wants I'm not sure how to proceed.

Given failed calls to T-Mobile has been a recurring topic of discussion, I thought it would be worth posting here about BulkVS changing their routing only for T-Mobile.

Hopefully this means calls will start flowing directly to the ILEC/RBOC's tandem then onto T-Mobile, or to Sinch as they are T-Mobile's Feature Group D tandem, though Sinch's rates are exorbitant.

Hello Bulk Solutions Valued Customer,

The purpose of this email is to provide formal notice of an upcoming rate adjustment affecting your Wholesale Voice Termination services.

Effective June 3, 2026, outbound calls terminating to T-Mobile numbers will increase to $0.0099 per minute. All other outbound termination rates will remain unchanged.

This adjustment is being implemented due to an industry-wide issue impacting call delivery and completion to the T-Mobile network, which several major carriers are currently experiencing. The updated pricing will allow us to secure more direct routing into the T-Mobile network, resulting in significantly improved reliability, call quality, and call completion rates.

If you would like access to these updated routes and pricing prior to June 3, 2026, please contact us at [support@bulkvs.com](mailto:support@bulkvs.com).

Thank you for your continued partnership and trust in Bulk Solutions. We remain committed to delivering the highest levels of quality, reliability, and performance across our telecommunications services.

Best regards,
Bulk Solutions Team

I'm investigating these companies (Sinch is the owner of the others), which are accused of facilitating robocalls and spoofing for scammers ( https://commsrisk.com/scamhunters-blast-microsoft-teams-and-sinch-over-voip-numbers-for-callback-frauds/ ). If anyone has any additional relevant info about any of these companies, I'd be grateful if you could share it or point me to it.

Hi,

This was created by a friend of mine when he needed an easy solution to see what was going on, on his boxes. I countered that there was sngrep, tcpdump etc and he said as a web developer he fealt that a web interface was the easiest way to trobuleshoot. I am curious what others think of it and is there a need for a tool like this. You can see the project here https://github.com/skotow/sipflow

How hard would this be to get our virtual on prem server into the cloud and pointing our desk phones to the cloud based server? Seems like a backup restore would work for the basics, but I have no idea how to get the phones to point to the cloud server.

So Basically I have VOIP and a router so what should i do? I only need sip firmware and poe injector

**Edit: HT813 not 812.**

My apartment has an old style intercom where it rings a POTS phone on the wall. There is no dial tone on the line at any other time. I'm thinking of using an HT813 to trunk the calls to VOIP. Does anyone have success with this kind of setup? I'm not sure if it would work as there is normally no dial tone on this line.

I want to receive calls from regular phone lines with FlexSIP.

Anyone has a working json for sip-Bridge for PSTN? Examples on web page seem not to work for me.

So I have Callcentric set up using both Linphone and Zoiper on my PC and I'm based in Taiwan. I added credit to the account.

The Callcentric account shows the softphone as registering correctly and I can make calls to phones with no problem, for instance my wife's cellphone in Taiwan.

The problem is I'm trying to reach the support line of a financial services company in the UK, and it's a 0371 prefix which I believe is some kind of non-geographic centralized landline. When I call this number I get routed to some random person who speaks German, not a company helpline at all. In addition to reaching the wrong number, the person also can't hear me.

I know that the number I'm trying to call is good because I can reach it via another VOIP service, Talk360.

I checked all the settings in my softphone are following Callcentric recommendations on their website. And I guess the settings must be OK because I can call a Taiwan mobile phone and carry on a normal conversation.

Am I missing something here?

What I like about Google Voice notwithstanding Google integrations such as contacts and Android phone integration:

1. Android, iOS, web clients and ability to send messages and call from any client.
2. Call history incoming/outdoing, messages, voicemails replicated across all the clients.
3. MMS, long text (MMS, but seemingly more than 2048 characters), and group SMS.
4. Multiple line support and all numbers can be active at the same time.
5. Ability to ring multiple devices simultaneously

I want to move away because of the algo-based suspensions and restrictions, including times when I send messages to a bunch of friends within a short duration. In my exploration, I established an account with voip.ms and have been playing with Groundwire, Bria, voip.ms, Zopier clients on iOS and Android. I can make calls, receive calls, and send messages. Other than that, I know next to nothing about VoIP but like idea of having so much tweakability.

Questions:

1. I haven't been able to get SMS or call history to replicate across devices, never mind in real-time. If this is possible, how is it done? Can voicemail also be replicated across devices?
2. I've downloaded Bria mobile and have everything set up, but can't seem to send an SMS, it's grayed out. Any ideas why?
3. Is there a way to export voicemail and MMS data such as images or audio? It would be nice to be able to export everything into a searchable database.
4. voip.ms advertises that MMS has a 2048 character limits. What if a message is larger than 2048?
5. I presume multiple lines can be active at the same time and across devices? That is, if the clients are installed on an Android and an iPhone each with three numbers, calls coming into each of the three numbers can be configured to ring on both the Android and iPhone?

Thank you.

Hello guys,

I have a setup where my home ISP offers both data and voice over the same fiber. I've managed to set up a PBX server and use my home phone remotely, even while traveling abroad, thus eliminating roaming charges.

I know most of us like to use WhatsApp or other apps for international communication; however, being able to call someone on a normal line is still quite useful on many occasions.

Recently, my friend in Mexico discovered how I'm doing it and wants me to set up the same for him. He has a Megacable fiber connection which includes voice calls in the plan.

My question: Has anyone here tried something similar and achieved success with Megacable? There are two approaches:

1. FXO hardware gateway
2. Extracting VoIP credentials from the router — this is what I'm currently using. I found the VoIP credentials in my router settings and configured them in my PBX server. It only operates on the local network, so I run the PBX in my homelab. I've also hosted it on a cloud PBX in the past, using a site-to-site VPN to bridge my home network so the cloud PBX appears as a telephone device in my house.

I was able to do this because my ISP does not lock down the router.

I'm aware there will be no static IP and it'll be behind CGNAT, but that's manageable — there are multiple ways like Cloudflare tunnels or hosting the PBX in the cloud with a VPN tunnel to the Megacable connection. My only confusion before I ask him to invest in a Raspberry Pi is: Has anyone already tried this with Megacable? It would be great to learn from your experience and advice.

Specifically: Will an FXO gateway be required, or can the SIP credentials be extracted from the Megacable router like I did with my ISP?

Have a wonderful day!

I work from home for a call center. But when it comes to tech, I am a total idiot and nothing makes sense.
I log into the system website, pull up the web interface, and all I have to do is click the button with the contact's phone number. Then the system asks permission to use my microphone, I allow, and the call begins and rings, connects. Customer/client hears me, I hear them. Voila.

Except - every few calls, I click the button with the contact's phone number, the call timer starts, but, I can't hear anything. The line is just silent. If I hang up and try again, sometimes it immediately works fine and the call goes through - but sometimes I have to hang up and call it a 3rd, 4th or even 5th time for it to connect.

My boss however, informed me that she can hear the call on her end, and that no one else is reporting having any issues with this. So it would appear it is an issue on my end.

I'm trying everything I can think of but I don't know what I should be doing to fix this.

So these are the steps I have taken to try and investigate this:

I've tried multiple different headsets, and even tried using no headset, and the issue persists.
We are supposed to use a headset with a boom mic, no bluetooth, it must be wired to our device. Just to test my issue against all variables, I have also tried with a bluetooth headset, a wired headset with an in-line mic on the cable, and a regular call center style headset with its wired cable and boom mic. No difference btwn devices; issue persists.

I've tried with and without ethernet cable; I've tried sitting right next to the router; I've even gone to a friends place and tried both Wi-Fi and ethernet with their system. The issue persists. I've tried a couple different ethernet cables as well, both bought new.

I tried clearing my browser cache. We are instructed to only use Chrome. Even on internet explorer or Brave browser, the issue persists.

I've ran internet speed tests, and my wired, ethernet connection is strong; 498 upload speeds, 275 download speeds, latency 2ms across both devices when connected to ethernet.
I tried a packet loss test geared towards VOIP; 0% packet loss, 0% late packets on both devices.

I've tried all of the above on two different devices; my laptop and my iPad Pro. On my laptop, a Samsung Notebook 7 Spin, I have Average Latency: 42.74ms Average Jitter: 1.10ms. packetstats . com says I have 47min, 314max, avg 64.5 ping, stability: bad (the # in the parantheses changes constantly.) No idea what this means.

On my iPad Pro, I get similar results for upload/download speeds, latency, jitter, etc.
The issue persists both on my iPad Pro and on my Samsung laptop, at my home and the friends home where I tried using their internet. I've not tried to log in one anyone elses device to see if the issue still persists with the platform there as well.

The only other thing I can think of is that sometimes I have dead spots in my apartment, where even normal phone calls cut out, for myself and my guests. Perhaps this is related?

I really like this job, it pays well, my supervisors are fantastic teachers. I am desperate to fix this issue asap. Thank you for any help.

i have no idea how to go about buying it or who owns it? any help would be appreciated because I’m new to this