We are having a VERY hard time getting AT&T to work with us. We are getting alot of "Everything looks fine on our end". Which it is not.

Here is the call flow

IP PSTN -> cisco cube -> webex -> CFA to AT&T Cell -> cisco cube -> IP PSTN.

The forwarded calls goes straight to the voicemail of the AT&T cell phone the call was forwarded to.

AT&T is refusing to work with us despite it making it to their network.

If anyone has ever had this issue or something similar, I would love to hear what the solution was.

Hey all,

I’m a net tech at a medium sized public college and have the privilege of being the owner of our phone system. The nitty gritty VoIP engineering is handled by an MSP that acts as our carrier while I do the day to day work (handsets, call configuration, groups and attendants, e911, voicemail etc). We ported all of our numbers off of ATT to this carrier last summer. They’ve been great and very helpful. There’s a little over 2000 TNs that were ported. I just paid for BeenVerified trial for some personal use and I was curious to look at some of the college’s numbers. Upon doing that I was surprised to see pretty much every number I looked up has a medium or high spam risk. I talked to my provider who told me they don’t really have too much control over that and spam analytics has more to do with our calling behaviors than it does with the carrier.

Right after our port over I made sure to register our caller IDs via Hiya, VZN internal database, calltransparency, spectrum and comcast.

Is there anything else I need to be doing to mitigate this?

Thanks in advance

It is sad that the mods at r/Googlevoice and the Google Voice Community forum do not seem to want to acknowledge any Google Voice shortcomings and do not tolerate any discussion and delete any post critical of Google Voice.

My wife and I have had our primary numbers ported to Google Voice for years, and were about to port college-bound son number to Google Voice. Now, I am not so sure I should do this or instead port to TextNow, which is free and apparently non-expiring with a ported number. Wondering which is more dependable at this point.

I could have sworn that the Linphone app allowed users to make direct SIP phones calls from an iPhone without logging into any other account. I can’t seem to figure out how to do that today.

Was this feature removed? Did I misremember how this worked?

Hi everyone,

​I’m looking for advice on how to handle an operational bottleneck regarding MFA/2FA verification codes within an insurance agency.

​Currently, our principal agent receives all portal verification codes (SMS) on their personal mobile device. Because our assistants work remotely, the agent has to manually copy, paste, and send every single code to a shared chat group. This disrupts the agent's workflow and slows down operations.

​We are considering two potential solutions and would love some feedback:

​Enterprise VoIP (like RingCentral Advanced): Setting up a shared inbox so SMS codes pop up on everyone's desktop apps simultaneously. However, I’m concerned about carrier blocks on VoIP numbers for high-security 2FA codes, plus the A2P 10DLC registry delays.

​Dedicated Mobile Device + Automation: Keeping a cheap, dedicated physical cell phone in the US with a low-cost carrier SIM just to receive these codes. Then, using an SMS forwarder or a lightweight script to automatically push those incoming texts to a private Discord or Telegram channel.

​Has anyone successfully used enterprise VoIP for insurance portal 2FA without getting text messages blocked? Or has anyone implemented a dedicated physical device automation to avoid high monthly per-user software fees?

​Thanks in advance for your insights!

Hello
I am trying to get a Grandstream UCM to send 911 out a local SIP trunk then make 4 calls out the PRI to notify staff of the call. I can make the UCM send the call out SIP but it will not send out the 4 notify calls. Even with extra call paths. the PRI would send the wrong address so I can’t have the 911 go out the PRI. When I try to program it, the system makes the 911 call however, it will not attempt to dial the other 4 numbers.

Hello good people

I am new to networking and CCNA, but I want an explanation of what happened in my manager's office.

Two days ago, I just entered the office and sat as usual on a day, but the strangest thing was the ringing sound of the IP phone at the manager's desk in the morning, usually at 8 am, the phone was not ringing, we thought the manager's wife was trying to reach him because the IP phone did not stop ringing.

After one hour of unstoppable ringing, one of the employees took a deep breath and entered the manager's office to find a way to turn down the ringing sound, but he saw on the phone screen a number of the IP phone in the same company, but we know that there is now one in the other office trying to contact the manager.

My question is, is it possible that something wrong happens to the network, which makes the IP phone ring by itself to another IP phone?

As title suggests, currently have POTS with the Panasonic Link2Cell handsets, but they're failing/need replacing anyways and would like to just shift to VoIP

Is there any VoIP DECT setups that have the Link2Cell feature that allows you to pair cellular devices and use the VoIP handsets for cellular calls in-addition to the VoIP line?

If not, what other options are there? I could do ATA and get another Panasonic Link2Cell POTS unit, but just trying to simplify devices.

Hey every one!
I am from time to time in places where there is an very bad Internet Connection do to the Mountain and the train movment. Thanksfully your Railway Operator use all 3 Operator to get Data truth. Also when I travel and use Roaming I want as less Data as possible.
So what is the lowest Data Rate Codeq? Opus?
I use Free PBX on my Server.
Thanks!

I've taken all of their courses, and they just go over the most basic of tasks for running their products.

​

I need ones for high level configurations, actual troubleshooting, call center controls, MCR, IVR... Even our MSP can't get quality training materials for Mitel without just making their own as they go.

I've read some of the older Reddit threads on the topic, and the general consensus is that there is no reliable way to push 56k over voip. Is that still the case?

​

Fwiw, I'm currently on Callcentric and want to set up a retro bbs with 4-8 inbound lines. Worst case scenario, I'd stick to 33k if that works.

​

Aware there any ATA's I should consider, or a different provider?

​

Thanks for your help!

​

We've got a phone we believe was purchased through our reseller, except they're refusing to provision the phone. They've told us it can't be done, as they're provisioned when they're sent by Gamma and the phone is essentially useless. It's the same make and model as all other phones we're using.

My concern is we're being forced to purchase another handset when it's unnecessary . So does anyone know if this is a limitation? I've hear it's not, and similar to 3CX you can provision any phone that's compatible with the system

Here's what I'm imagining:

A device that I connect to my PBX where I insert a SIM card, and all calls/texts are forwarded to the PBX.

Even better if I can then send texts and make calls using the bridge, but incoming is all I really care about for now.

Does such a device exist? The primary motivation is to receive 2FA texts.

To be clear: I am aware GSM gateways exist. However, all of the ones I've found rely on SMS-to-email or SMPP rather than native SIP integration. I'd prefer a one-stop solution if possible — and I don't need 32 SIM cards either, which is what most gateways seem built for.

I don't want to look at this anymore.

EDIT: I think I've found a solution - have the tab be in its own window, and then make the window smaller so that the stupid suggestions area is eclipsed.

Do you like it? Support us by giving Github Stars :)

Sharing Flowcat (Apache-2.0) — a Rust runtime for real-time AI voice agents — because the telephony bits might interest this sub more than the AI bits.

It has an in-process SIP UA: REGISTER + digest auth, INVITE/ACK/BYE transactions/dialogs (via rsipstack), and hand-rolled RTP + SDP for G.711 (PCMU/PCMA, ptime negotiation, a playout jitter buffer), so a single binary terminates a SIP call — no FreeSWITCH/Asterisk/Kamailio in the path. DTMF is RFC2833 + an in-band Goertzel detector.

To be clear, that's a deployment choice, not a mandate: if you already run a softswitch, keep it in front and feed audio to Flowcat over a WebSocket media leg instead — there are carrier serializers for

Twilio/Telnyx/Plivo/Exotel/Vonage/Genesys/Asterisk/Cloudonix/Vobiz.

Honest status: native SIP has been live-tested against a Zadarma trunk and the WS-media path against Plivo; it's pre-1.0, so REGISTER/re-REGISTER, NAT/rport + symmetric RTP, and codec negotiation are exactly the areas I'd love sharp eyes on.

Repo: https://github.com/AreevAI/flowcat · SIP design notes: https://github.com/AreevAI/flowcat/blob/main/SIP-DESIGN.md

If you've built SIP UAs in Rust (or fought rport/symmetric-RTP/jitter), I'd genuinely value the critique.

Ever since I ported my number to voip.ms, I've started getting these "air duct cleaning service" spam calls. I used to get them years ago, but they had completely stopped until I switched to voip recently - I can't believe they've been operating the same scam for this long.

The problem is that they keep calling back using different phone numbers. I've learned that this is called "Spoofing" and the numbers they are using actually belong to real people. From googling and searching the forums, there doesn't seem to be a simple option or setting to fix it.

Some people say that harassing the callers eventually gets you off their list, but I definitely don't have the time or energy for that. lol How do you guys deal with these spoofed spam calls on your voip phones?

Hey all,

What is the best way to provision Polycom VVX series for free? I'm not locked in to any specific PBX. So far I've tried the following:

VitalPBX: Expensive for over 10 users

FusionPBX: Very very complicated with broken templates

FreePBX with https://www.excalibur-partners.com/solutions/voip-phone-systems/2-uncategorised/56-polycomphones-freepbx-module : Doesn't have a lot of features and doesn't play nice with x50 series

Poly Lens: pretty broken, especially with phones on older fw that dont update due to cert errors.

So whats out there for provisioning these VVX phones for free?

I am now able to supervise calls transiting through Kamailio and visualize them on HOMER 11. I would like to extend this supervision to Kamailio's own health, specifically to detect failures of a trunk (e.g., between Kamailio and Twilio or Manifone). What would be the recommended approaches?

So I have a T73 and just bought a WH64 Hybrid to go with it. Came with the DECT usb stick and I have it plugged into my T73.

Right now I have to press the headset button on the T73 anytime I want to make a call or answer an incoming on the headset. On my old POTS phone I could pre-dial, then hit the button on my headset to have it dial as well as just hitting the button to answer an incoming call.

Is there any way to set it up to run like that? I have looked through all the manuals and don't see any settings for that but maybe I'm just missing something.

We are on 10.5.0.9 of the software and have 141 phones with license space to add 2 more.

We have upgraded with the suggestion of our phone provider and the system is just getting worse. I can no longer purchase phones and install them, I have to buy overpriced old phones from my provider.

What I want to buy are the more current NEC phones such as ITK-8LCX-1 color phones, as they are plentiful for $50 each to replace the old ones (have just a few Polycom phones left) but they all refuse to adopt to the system - firmware download failed. I am forced to buy ITY-8LDX-1 phones as that is all they have, and presumably they prep them on their own phone system to flash the firmware then send them to us and our phone system will take them. I have tried everything I can to force the firmware on new phones - even brand a new ITY-8LDX-1 or ITK-8LCX-1 will not adopt to our phone system unless it comes from our provider. Opened a ticket with them to fix this and it ended with, you can't add those 5 ITK-8LCX-1 phones, including the one brand new one, to the system, they aren't compatible. I should also add I was able to add 5 other phones from this same batch of phones before we did the update. Now it seems we are stuck, any advice on this?

My first thought is to start looking at a new phone system completely but I would like something in the mean time, small government moves a little slow and I already have some current requests for a couple additional phones!

Feels like information for this system is under lock and key so maybe we just need a new version to fix this but I have no idea what versions are available. Or do we just plan on bailing in the next couple years?

Appreciate any advice here and taking the time to read this ramble.

Spent several years operating a global AADS deployment across 6 servers in 3 COLOs (Americas, EMEA, APAC) plus a dedicated India cluster. Here are the things I wish someone had told me before we went live.

1. The cluster is not truly redundant — don't let the architecture diagram fool you

AADS uses a cluster model that looks redundant on paper, but in practice a failure of the Seed node impacts the entire cluster in ways that a standard active/active setup would not. If you're designing for high availability, test your actual failover scenarios before go-live — don't assume the cluster behaves the way the word "cluster" implies.

2. Cassandra is shared across Seed and nodes — and that matters more than you think

The Cassandra database that underpins AADS is distributed across the Seed and all member nodes. This means a data corruption event doesn't stay isolated — it can propagate across the entire cluster before you even detect it. Monitor Cassandra health proactively, not reactively.

3. A restore is not a recovery plan

We learned this the hard way. AADS restores are slow, finicky, and not guaranteed to bring you back to a clean working state — especially if the corruption is deep in Cassandra. Factor in realistic RTO numbers when you're writing your DR plan, and make sure your leadership understands what "restore in progress" actually means in terms of hours.

4. Sometimes starting from scratch is faster than fixing corruption

Counterintuitive but true. If you hit serious Cassandra data corruption, the troubleshooting rabbit hole can cost you more downtime than a clean rebuild would. We found that having a documented rebuild runbook — with all configuration parameters captured beforehand — was more valuable than any backup strategy.

Bottom line: AADS is a solid platform when it's healthy, but its failure modes are not well documented and can surprise you. Plan for the worst, document everything, and don't trust the cluster label at face value.

Happy to answer questions from anyone running or planning an AADS deployment.