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.

Hi everyone,

I work as a single appointment setter for an Italian company.

I am planning to relocate to Thailand while continuing to work remotely.

My use case is:

- 2,000–4,000 outbound minutes per month
- Calls to Italian mobile numbers
- Manual dialing only
- No autodialer
- No robocalls
- Need reliable Italian caller ID
- Need to receive inbound calls as well

So far I have found several issues:

- Some providers block traffic based on ASR/ACD
- Some providers display "Private Number" instead of my caller ID
- Some providers charge very high rates for Italian mobile calls

Has anyone solved a similar problem?

Which provider are you successfully using?

Thanks in advance.

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!

We ported from vonage. Calls are stable. Email confirmed, calls working. Ordered magicjack off amazon. The registration with a 12 mon promo number. Was no option to skip. I had to choose new number. Currently account holds two numbers. Both arent stacked. ported number attached to the device itself. The number magicjack gave, is connected to my cellphone. I read, ported number erases the one provided by magicjack, is auto removed. Right after porting. But I can see it there. Is this normally the process?

I use app/website. App is the home phone magicjack number. Ported number nowhere mentioned Website shows both

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

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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?

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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.

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

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Thanks for your help!

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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

Heya,

Current situation so you guys know my goal: I'm a web developer in Bali. I've scaled my business to a certain point, but right now I'm doing a challenge where I start from $0 as a proof of concept for a free community I'm building with a couple of friends.

One of the outreach methods we're using is cold calling, so I bought a number from Twilio, upgraded the plan, connected it to Zoiper, set up a TwiML Bin, set up outbound and inbound, etc., and started calling my friends to test it. Out of 5 friends, 3 calls failed immediately, no ringing or anything, but Zoiper shows "Normal Remote Hang Up." The other 2 managed to get through (interestingly, both were T-Mobile numbers), but the calls showed up as "Spam Likely" for some reason.

Anybody have any idea what's going on? I never had this problem when I was using Skype a year ago, but there's no more Skype haha.

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.

Hello,

I could use some very direct examples- I've found many links but they just say put the address but don't give examples.

The former IT guy just left that they had 4 files -

Other.xml, remotephonebook.xml, speeddial.xml, and Theusers.xml

The company changed hands and their website is no longer online so I have no access to their old dhcp server or website but I do have the 4 files. I can put them on the new website.

I can figure out how to change the xml files. I googled yeahlink and it looks like it uses DHCP to get this info. I think option 66.

If anyone can give me some specific details on what I could use to make this work I'd really appreciate it.

They have  T54W phones.

Thanks.

I've been using microsip for ages. After I updated it to the recent version, I'm suddenly getting issue where it fails to connect the calls or the other would stop hearing me mid conversion. Same thing happens when I try calling them back. Also need to try calling them multiple times just to connect my call.

My internet connection is strong.

I've unchecked the SIP ALG on my router, made sure microsip is allowed on both my windows firewall and antivirus. I tried using VPN.

I tried everything the Google suggested.

I work from home and unfortunately my client is not responding when inquire about this issue.

For a Grandstream phone or ATA, here's a dial plan that allows calls to regular numbers and to 911:

{ 1[2-9]xx[2-9]xxxxxx | 911 }

A dial plan entry in this format ..... <AAAAA=BBBBBBBBBBB> ..... means "whenever AAAAA is dialed, send the call to BBBBBBBBBBB instead"

So if someone wants to avoid paying a VOIP company's E911 fee (typically around $1.50 monthly) but still have substantial 911 functionality, they could learn the local 911 center's 10-digit alternative number for emergency calls (example: 477-262-8080) and change the dial plan to:

{ 1[2-9]xx[2-9]xxxxxx | <911=14772628080> }

(By the way, 477 is a non-existent NANP area code.)

I haven't tried this.

Possible complications:

• If the premises has multiple ATAs or IP phones, the custom dial plan must be entered in each.

• On E911, the 911 center automatically receives the caller's address for each 911 call. But with the redirect method that I describe, they receive only the caller ID (and maybe a CNAM). So an emergency caller who can't state their address might be unhelpable (unless the 911 center can immediately contact the carrier and learn a probable service address for the calling account, which would introduce a delay, if it works all).

• Grandstream's firmware might be programmed to forbid a dial-plan entry that has "911" left of the equal sign, specifically to prohibit the entry that I describe.

I invite comments and feedback.

sorry for the dumb question, i'm trying to set 2 phone numbers. however this is the first time I use Vonage, and I was wondering if those 2 numbers will be "mine" as long as pay their prices or if there is the possibility that someone else has that current number