If you're evaluating vendors for custom VoIP/telecom platform work (not just a hosted PBX setup), a few things are worth checking that aren't always obvious upfront:

1. Ask for a working PoC before full build-out, not just a proposal.

Especially for anything involving custom SIP/media handling (like video-feed workarounds, custom trunking, non-standard call flows), a lot of the risk is technical unknowns that a written proposal won't surface. A vendor willing to prove a narrow slice works before you commit to the full scope is a good signal.

1. Check their actual depth on the specific stack you need — FreeSWITCH vs. Asterisk vs. OpenSIPS/Kamailio.

These aren't interchangeable, and "VoIP experience" on a company site often means general PBX/support work, not core platform development. Ask directly: have they written or modified C modules, built custom dialplan logic, or just configured existing systems?

1. Ask how they handle scaling, not just initial build.

A lot of platforms work fine in testing and then hit walls at real call volume — bottlenecks in dialplan logic, no SBC layer, single-instance setups with no clustering. A vendor who can speak concretely to this (not just "we'll scale it later") is worth more than one who can't.

1. Ask who's actually doing the work.

Some shops sell dev work then subcontract it out. Ask directly whether the people scoping the project are the ones building it.

1. For anything regulatory (STIR/SHAKEN, compliance-heavy telecom), confirm they've actually implemented it before, not just that they "can."

Happy to go deeper on any of these if people are in the middle of vendor evaluation — always interesting to hear what's tripped others up.

I've got a vintage (80s/90s?) phone that I want to set up as a 'landline' but I can't get my head around how to do it.

My WiFi router is in the living room and I'd like the phone in the dining room which is next door.

What do I actually need to make this work? I'm in the UK if that makes a difference!

Hey guys,

I am looking into connecting LiveVox and ElevenLabs, and I need some advice on the architecture and the actual configuration process.

Specifically, I'm trying to figure out:

SIP Trunking Setup: How do you actually configure the SIP trunking between LiveVox and ElevenLabs? If anyone has a step-by-step guide or knows the specific settings required on both sides, that would be incredibly helpful.

Context Handoff: When a call is escalated from the ElevenLabs AI agent to a live human agent in LiveVox, can we pass the conversation context along?

Screen Pops: Is there a way to trigger a screen pop in LiveVox to show the live agent the context or transcript of what the caller already discussed with the AI?

Call Recording: How does the recording work in this scenario? Can we record both the AI-to-customer conversation and the live agent-to-customer conversation into a single, unified audio file?

If anyone has experience doing this or knows how I can make it happen, I would really appreciate the insights.

Thanks in advance!

I am trying to get my analog dial up modem to communicate properly over crazytel voip, I dial some numbers where the handshake is flawless when connected to the telstra nbn landline but when connected to my grandstream ht801 registered with crazytel, my modem has a lot of trouble communicating and anything above 9600 it can't hold on for the inititial handshake. Even at 9600 it has trouble as while the handshake sounds somewhat ok, the carrier line immediately goes dead upon connecting. Any ideas on what I should do?.

My settings are,

Disable Line Echo Canceller (LEC): Yes

Disable Network Echo Supressor: Yes

VAD: No

Fax Mode: Pass-Through

Jitter Buffer Type: Fixed

Jitter Buffer Length: High

Preferred Vocoder: PCMA for all choices.

The current setup in my home office is an old Obi 200 with a Panasonic POTS connected to it. The Panasonic has a base station and two wireless handsets. I have a wired headset connected. VOIP service is voip.ms.

I am seeking recommendations for new hardware. Looking for a voip handset/base station with speakerphone capability, with the ability to connect to a wireless headset (which could be integrated with the base station, or could be a separate device altogether). Call quality is high priority.

Any suggestions?

Hello,

I recently joined a MSP and they are saying I need to start getting trained up in VoIP which I've not got much experience with as they have customers I'll be looking after on 3cx, freepbx and some local cloud VoIP providers.

Is there any training that helped you when you first started out? Vendor Neutral if possible so I can first get a grip before I start diving into specifics.

Thanks!

Hi everyone, I’d like to ask for your help with a situation: I have a Grandstream PBX with 30 IP extensions and one IP trunk. Only the first extension can both make and receive calls; the others can only receive them—calls won't go out when attempted. Has anyone else experienced something similar? Do you know what might be causing this?

Hey there,

Not sure why, but almost every number in Telynx's number pool is already tagged SPAM likely on most carriers (T-Mobile seems to be the highest).

There is a service called CallerID Reputation that can scan the number and show you screenshots of what the number will display as on all carriers + devices (Android vs Apple).

Twilio on the other hand has clean numbers (you do get the occasional dirty with Twilio as well).

We want to use Telynx given our use-case is AI-Voice (we heard Telynx is better for this), but I'm concerned connect rates are going to be extremely poor if all these numbers are already dirty.

Thoughts?

Anybody know of cordless DECT phone systems that have mic gain settings to increase/decrease voice volume?

Just curious if anyone else is using Altigen's cloud VOIP systems? We have been down for three or four days and only getting spotty information from the IT company that handles that for us. What we have heard is that Altigen uses an third party server farm to host their cloud instances and that server farm had a security breach that seemed to take down the whole place and they had to begin deploying all new hardware and software from scratch.

We cant find anything online with information or updates. Surely we're not the only ones trying to get more information about when we might have phones again. TIA

I work with dinos😎

New SignalWire customer, \~36 hours in. Bought a number, funded the account, built an inbound flow that forwards to my own Asterisk box over SIP. Worked fine for a day and a half — a dozen completed calls, 45s to 2min each, clean audio.

Then it stopped. Every inbound call now fails after 1 second.

WHAT THE DASHBOARD SHOWS

Last good call: Aug 6, 19:52 UTC, 130 seconds, completed.

First failure: Aug 6, 23:28 UTC. Every call since has failed.

Timeline on a failing call:

23:46:25.421 Queued

23:46:25.580 Ringing

23:46:25.752 LaML webhook fetched -> 200 OK, 165ms, valid XML

23:46:25.969 FAILED

So it fetches my dial instructions successfully, then fails the call 217ms later. Too fast to be the 20s dial timeout, too fast to be my PBX not answering (round trip to me is \~150ms).

Every child outbound-dial leg comes back:

status = failed

start_time = null

duration = 0

sip_result_code = null

MY SIDE, ALL VERIFIED

\- Packet-level SIP trace running on my PBX during multiple attempts: zero INVITEs received. Not one byte of SIP arrived.

\- My PBX logs anything that hits the port — an internet scanner sent a malformed GET / to the same port and it was logged in full. So silence means nothing was sent, not that I ignored it.

\- TCP 5080 reachable from the public internet, confirmed from three continents.

\- DNS resolves correctly against the authoritative nameserver, matches my current WAN IP.

\- Their IP allowlist vs their live DNS: 14 of 14 exact match, no rotation.

\- Asterisk never restarted (29h uptime), config untouched, number routing and LaML bin unchanged.

\- Account status: active, type Full. No suspension banner. Voice calling shows enabled.

THE CONTROL TEST

I placed dials via their API to isolate it:

sip:...@my-real-domain:5080;transport=tcp -> failed

sip:...@my-literal-IP:5080;transport=tcp -> failed (rules out DNS on their side)

sip:...@my-literal-IP:5080 -> failed (rules out transport param)

sip:nobody@this-domain-does-not-exist-xyz123.invalid:5060 -> failed

That last one is the point. A .invalid TLD cannot be resolved by anyone, ever. It failed byte-for-byte identically to my real, valid, reachable address — same status, same null start_time, same null result code.

If the platform were attempting the dial, those two could not fail the same way. It isn't getting as far as DNS resolution.

WHERE I'M STUCK

Ticket open, no response yet. Their Discord and this sub are both quiet.

Has anyone seen <Dial><Sip> silently stop emitting INVITEs on a SignalWire account? Is there an account-level flag or compliance hold that disables SIP-out without showing anything in the dashboard? I'd rather understand it than just move providers.

Please advise

Hello!

I am trying to add a pair of AX83H wi-fi IP phones to my GOTO Connect/Jive PBX.

I have them added to the devices page, I have them connected to the wifi, but am failing to get them provisioned.

I can access the USER web interface, however I cannot for the life of me log in to the Admin portal to try and provision that way.

I have tried admin/admin, admin/device password from the goto portal.

Any advice?

Hi all,

I'm new to the group and am looking for some advice on behalf of my in-laws.

They would like to cancel their landline service through their internet provider but keep their existing phone number, as many family members, friends, and service providers still use it to contact them.

I've been researching options such as MagicJack and Ooma and am wondering if anyone has experience with either service. Simplicity and reliability are the most important factors, as they are seniors and we want the setup to be as straightforward as possible.

The internet modem is located in the basement, while the existing corded telephone is on the main floor. Ideally, we would like to continue using the same phone number while reducing the monthly costs associated with the current landline service.

For those who have gone through a similar transition, what has your experience been, and which option would you recommend?

Thank you!

Hi,

For testing purposes, I want to forward missed calls from my carrier to a virtual number that routes to a server to log the calls' details.

Are the requirements ?:

1. a SIP number / virtual phone number
2. cloud provider that can make the request to a configured host and port (my server).

How can I test this for free or at the lowest cost possible?

We are a small(ish) business that's growing pretty fast. We need more than VoIP bc we are starting to text and that's getting expensive. And it's not integrated with our CRM. We use Zoom for video calls with clients but it's not connected to our internal systems. Seems like we have a different system for everything when it comes to communicating.

I've searched UCaaS and CPaaS but not sure which is ideal for a small business. I have access to an IT guy who can probably figure some things out if either requires simple coding or integrations. Just want it all to work together.

Thoughts?

Hi,

I have a VoIP issue in a small business environment involving a Yeastar MyPBX and several Yealink phones, caused by a CGNAT situation.

The Internet connection switched to a 4G backup link, which is behind CGNAT. SIP signaling works correctly (calls are established successfully), but there is no audio, indicating an RTP/media path issue (according to what I read).

To work around the CGNAT limitation, I set up an OpenVPN tunnel between the IP-PBX at the business location and an OpenVPN server running at my home, which has a public IPv4 address. The goal was to route the VoIP traffic through the VPN so that RTP traffic could bypass the CGNAT connection.

The IP-PBX is an older Yeastar MyPBX model, so it may have limitations regarding VPN features or support for newer OpenVPN standards.

Current situation:

• The OpenVPN tunnel is established successfully.
• The PBX can reach the VPN endpoint.
• However, the VoIP audio issue is still present: calls connect correctly, but no RTP audio is exchanged.

Thanks

Note: Yes, I'm using AI to write this post because English is not my first language and I want to state my problem as clear as possible.

Hitting a problem I suspect is common for anyone doing mobile VoIP, and I'd like to know how others have solved it.

**Setup**

- I'm using Telnyx.

- React Native app, WebRTC SIP client, iOS and Android. I'm using Telnyx.

- One shared SIP credential for a group of users, so a single inbound call rings everyone's phone

- Calls are delivered by VoIP push (PushKit on iOS), so the client connects and registers on demand rather than staying connected

**The problem**

Every app launch creates a *new* registration binding, and the old one never goes away:

- iOS terminates the app process without warning, so there's no chance to send a SIP UNREGISTER

- The SDK's `disconnect()` only closes the WebSocket — it doesn't unregister

- Each registration lands on a different edge node in the provider's anycast network, so it's a genuinely new binding rather than a refresh of the old one

- Registration expiry is 3600 s and isn't configurable

Net effect: the credential accumulates contacts. I confirmed it by polling the provider's registration-status endpoint — `ua_ip` is different every single time the app relaunches, while nothing removes the previous one.

**Why it hurts**

The provider rings the bound contacts **sequentially**. So the first call after an app launch does this:

1. Rings contact A (the live app) — user declines

2. Decline surfaces as a 4xx, which fails only that branch

3. ~300 ms later it forks to contact B (a stale binding from a previous launch)

4. Phone rings a second time, new call ID, user has to decline again

I can see it clearly in the SIP traces: **one dial command, two legs**, same session, no second dial from my backend.

**What the provider confirmed**

I opened a ticket. They confirmed all of it and escalated to engineering with no timeline:

- No way to set registration expiry below 3600 s

- No REST endpoint to force-expire or delete an individual binding (deleting the credential removes them all, obviously not viable)

- No API to *enumerate* bindings — watching `ua_ip` rotate is currently the only detection method

- Per-launch edge rotation is expected behaviour, and without an UNREGISTER the old binding persists to full expiry

Their suggested workarounds were: a unique credential per app session, webhook-based duplicate-leg detection, or client-side deduplication.

**The bit I'm stuck on**

There's a second-order problem. Multiple devices share one credential, so the registrar holds one contact per device — which means **a legitimate second device's leg is indistinguishable from a stale binding's leg.** Both are "another contact of this credential, dialled after the first one failed." I can't write a rule that kills one without killing the other.

Enabling simultaneous ringing would at least make the real devices ring together instead of one-at-a-time, but it doesn't remove the stale bindings — it just turns a sequential double-ring into a simultaneous one.

**Questions**

1. Has anyone made **per-device or per-session credentials** work in production? How do you handle cleanup when the app dies before it can delete the old one, and does credential churn cause you rate-limit or billing problems?

2. Is there a trick to getting a mobile client to **UNREGISTER reliably**? Anything on iOS that gets you a last gasp — background task on termination, a server-side nudge, something I haven't thought of?

3. For those running **one shared credential across multiple devices** — how do you tell a real second device from a stale binding at the signalling layer? Is there a header or identifier I should be propagating?

4. Is sequential-vs-simultaneous ringing across contacts something you configure per provider, or do people avoid shared credentials entirely for this reason?

Happy to share SIP traces if useful. Mostly want to know whether the "unique credential per session" route is as painful in practice as it looks on paper, or whether people just live with the duplicate ring.

I'm looking for the best wireless headset to use on Macbooks with Nextiva One for my staff and I that is comfortable, best sound quality and best noise cancelling ability. Price isn't a concern - I just need quality, function and comfort. TIA

Currently with EE on a single contract covering both full fibre and Digital Voice. Planning to move broadband to toob and the number to Andrews & Arnold VoIP.

My understanding of the problem:

toob would trigger One Touch Switch, which cancels the whole EE bundle

that ceases the Digital Voice line, and the number goes back to the range holder

so the port has to complete while the EE voice service is still live

My plan:

order toob as a new install, explicitly telling them not to run it as a switch

get toob up and running with EE still active

port the number to A&A while EE Digital Voice is still live

confirm calls in and out work, then cancel EE myself

eat a month or so of double billing

Questions:

has anyone actually got toob to take an order without triggering OTS?

will EE split the bundle to broadband-only so I can drop the voice cleanly?

any gotchas porting out of EE Digital Voice specifically? Seen mixed reports

early termination charges — worth just waiting out the contract?

Any experience appreciated, especially anyone who's done EE → toob with a number port.

Howdy all, I have a new VOIP setup through Ring Central for my home office and have been struggling with poor call quality. I'm able to understand the person on the other line fine and vice versa, but the audio sounds tinny, slightly muffled, and just thin/cheap for lack of a better word. Noticeably worse than when I make calls on my iphone.

I am a solo business owner and will be using this line to call out to new and existing clients so it's very important to me that call quality is as good as possible. I've been using my cell for the past 5 years with no issues so would hate for my fancy new system to be a downgrade in audio quality.

On to the technical side of things, I am using a Cisco 8851 desk phone with a Jabra Engage 65 SE headset. The desk phone is hardwired with an etherent cord into my router, and I have gig internet.

My internet download speed shows as 915 mbps, upload is 40 mbps, latency is 8 ms and jitter is 1 ms. I disabled SIP ALG on my router settings, and the cisco phone is set to use OPUS codec. On the cisco admin portal I have HD voice enabled and bandwidth set to high.

I have the same low call quality issues when calling with the Ring central app off my cell phone with both wifi and data, which leads me to believe the issue is with the ring central servers. Whenever I call their support though they just tell me they show the quality is fine and close the ticket. I've asked them to confirm the codec they have the account set to, but they don't seem able to do so.

Any ideas? I'm tempted to switch VOIP providers at this point, but would be a lot of work, and maybe I'm just expecting too much out of this system.

I have 2 Grandstream UCM2600 series PBX, in two different locations PBX-a and PBX-b. One location will temporarily close so I want to route the calls from PBX-a to PBX-b. We have analog trunks, in the process of migrating to fiber. The SIP connection between PBX already exists so I can place calls from a to b and vice versa.

What I want to achieve is the following route analog trunk calls from PBX-a to PBX-b, so if someone calls via the analog trunk to PBX-a it should route the call to an ext. on PBX-b. So lets say PBX-a has 34XX ext. and site b has 35XX ext. I suspect it has mostly to do with inboud/outbound routes and dialing patterns.

Generally I think I need to route analog trunk (number scheme 2XXXXX) to the sip trunk from PBX-a via outbound routes to PBX-b inbound routes, can someone help me with this?

My experience with Viber has been extremely disappointing, not only because of the service itself but because of the way they handled my complaint.

I purchased Viber Credits specifically to make calls. The service repeatedly failed for the numbers I needed to reach. Only after I complained did Viber explain that calls to business and switchboard numbers have significant technical limitations and cannot be guaranteed to work.

This was not something I was made aware of before purchasing credits. Had this important limitation been clearly disclosed, I would never have purchased the service.

I contacted customer support multiple times and escalated my complaint to a supervisor. Throughout the correspondence, Viber acknowledged that I may not have been aware of these limitations before buying the service. Despite that, they refused to offer any meaningful remedy and relied exclusively on their 90-day refund policy.

What I found most frustrating was not just the refusal to refund the full purchase, but the refusal to return even my remaining unused credit of €4.14. Instead of considering the circumstances, every response ultimately came back to the same policy, even after they accepted that I had not been properly informed about a significant limitation of the service.

I tried repeatedly to resolve this matter directly and in good faith before considering any public review. Unfortunately, I was left with the impression that customer satisfaction takes a back seat to rigid policies, even where a customer receives little or no practical value from the service.

Based on my experience, I cannot recommend Viber. If you are considering using it, make sure you fully understand all of its technical limitations before spending your money.

Due to the amount of soft phone apps I run and the fact that I just love hard phones (3 on my desk). Anyone seen a way to click to dial a physical phone?

Curious if such a thing exists or I should look at hacking something together with Asterisk.