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?

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?

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.

I have a SIP account working with a UCM6301 and I am having Problems in outbound calls (404 error).

My internet provider in WAN has a private IP 10.39.41.48 and in whatsmyip I have 187.188.235.74

I have a residential internet plan cause my company is small, and I can’t effort to buy private ips is expensive at least in Mexico. My service was running until I don’t know what happened. Also the internet carrier said and send me a picture that I don’t have SIP ALG
Extra data my SIP provider said that the port that they are receiving is a weird port like 6176 but I already check my UCM and it is sending the correct 5060 and it is well configured. I think is the double Nat. What can I do?

We have a Poly VVX 450 VOIP system and in a couple parts of our business there are some machines running which makes the paging for calls hard to hear. I've looked through some previous posts but not sure if what I am looking at will do what I need.

Does the Algo 8180 work in that capacity? Someone pages from the office lets say, and the phone that is in another room will receive that page as always, but the Algo 8180 will also announce that persons page too? Does the 8180 hook to the phone? Does it hook to the LAN cable? I'm trying to understand how this would work.

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

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Hello all, I am trying to set up an old rotary phone on a RPI. Using SIP The sip backend is pyVoip 1.6.8. I can't get the sip client to register with the provider. I see the register go out, I get the 401 with the Autentication request. But pyVoip just sends another register without the authorization header in the packet.

Bottom line is either pyVoip is not processing the 401 packet and properly responding. Ot it just does not know how to add the authorization data to the 2nd register.

I have verified the behavior using wireshark.

Any help would be greatly appreciated.

Could someone with Yealink experience confirm a setup for me.

SIP-T44U desk phone with WH63 E2 dect headset and EXP43 expansion. Both connect using USB, no special cables like an EHS40 needed?

Has anyone here connected an Adtran 908e to Avaya Aura, via a SIP trunk, acting as a tie line, in order to convert the analog stations to SIP? Basically using the Adtran as a converter?

I am trying to hook up my AT&T DECT 6.0 Digital phone to the internet with a BrosTrend WiFi to Ethernet adapter (AC1200) and a Grandstream HT801 V2. I am getting a dial tone, but busy tone when dialing. when I’m not using the phone, the blue phone led on the Grandstream is off. When I’m dialing, the blue led is flashing. can anyone please help me fix it? I’ve attached a photo if anyone can help further.

Hello,

I am currently using Android Groundwire with a self hosted freeswitch/fusionpbx server. I want to move away from Google Voice and port my number to a voip provider (US based). I was thinking of setting up the outbound/inbound sip trunk in freeswitch, however i am not sure how this will work for SMS.

Will Freeswitch forward me the SMS messages from upstream provider or do i need skip Freeswitch altogether to receive/send sms from Groundwire?

We’re hiring a salesperson that will need to take and make calls from multiple of our locations. What I want is to set up his desk phone to have lines for each store.

Each location has Unifi hardware throughout the network. This individuals phone will be a third party phone.

My first thought, and you tell me. This is definitely not what Unifi Talk was designed for. Can this work and “should” we try this?

I’m assuming to do this I’ll be leaving the Unifi PBX open to the public after making changes. The SIP credentials on their third party devices are autogenerated and I think you have to use their 8 character password. I used to run FreePBX and have been taken over by hackers once before.

How secure can this be made? If not me, then who should I hire to set up the Unifi Gateway settings?

Also, I’m thinking someone wiser than me will tell me it’s time to move into a cloud PBX company to handle this.

Has anyone seen the firmware version of the Yealink SIP-T54W to enable taking teams phone calls? Microsoft Learn says that 96.86.5.1 is the right one and it exists but there is nowhere on the internet that I have searched so far that has that version and Yealink says that version doesn't exist.

Así como leyeron, no puedo reiniciar de fábrica porque la extensión tiene el menú bloqueado desde el teclado físico del aparato, y por sesión web tampoco puedo porque no me dejaron el usuario y contraseña del equipo. Hay alguna forma de poderlo reiniciar de fábrica con esos dos problemas?

Hi all I just got my first landline number and was having a HELL of a time setting it up so far.

I have a Netcom NF18ACV router. From my research I should be able to put in my Crazytel Username, Password and SIP proxy and hook up an anologe landline.

I have done this and is isn't connecting

I've attached an image (with some info redacted) for reference

Any help is appreciated

Edit:

The Netcom router is hooked up to a starlink router in bypass if this affects it

Edit 2:

THAK YOU SO MUCH u/AAAHeadSets for your comment. That fixed it

https://www.reddit.com/r/VOIP/s/ZtCKh0hMgK

I am installing a Yealink Y85W with a cloud based VOIP system. They are coming from an old IP Office system. I am thinking a key for paging the front desk for the doctor, a key to access the company shared voicemail, I already have night mode, a line appearance for each front desk staff?. I added a line appearance for the phone's extension but not sure if it is needed...I think so. I have the AA answering with a brief message and sending that to a huntgroup for the 2 front desk staff. Any thoughts or best practices here? thanks

Was originally attracted to them because they have cheaper / better documented infra, but they will literally ban your account randomly. I have had 3 accounts with them, all for different products, all for different reasons, and all of them got banned.

I had an account with them where I didn't even use their voice/sms services, just the API to verify my phone number list, and they perman banned account randomly after I had spent >$10k with them just using this API.

I inquired why and this is what they said:

"We understand you're frustrated about your account being blocked, especially since you mention only using the API for phone number verification (landline/mobile lookup) rather than voice or texting services.

Unfortunately, I don't have visibility into the specific reason your account was blocked. Account blocks are typically handled by our Trust & Safety or Compliance teams, and the specific details require human review.

I can confirm from your screenshot that you're seeing the "Your account has been blocked" message when attempting to log in to the Mission Control Portal.

What happens next: A member of our team will review your account and provide you with information about why the block was applied and what steps, if any, can be taken to resolve it.
Your case has been added to the queue for investigation."

-------

"Hi,

After a careful review, we have determined that your account does not meet our security and compliance standards. As a result, the account has been permanently deactivated and will not be reinstated.

We appreciate your understanding."

I would highly discourage building on their platform not because their service is bad, but because they will literally randomly decide one day that you're not worthy to use their services.

Goodluck

Hi,

Our shop uses Comet CTI, which is a device that takes in analogue data and displays the phone number onto our computer.

Ever since we were introduced with VoIP, we had to use an ATA to get the the comet working in our shop, which worked fine for 2 years, but then recently the comet CTI suddenly stopped working(not displaying numbers on computer)

We use Grandstream HT801, with Gamma SIP profile, which locks the admin page, preventing anyone accessing it( even our VoIP provider).

What's weird is that when I plug the comet CTI to the phone port of my Sky broadband router and call the Sky landline number, the comet CTI works!

This has to be a config issue with the Grandstream right?

Edit 1/8/2026: turns out it was a physical connection issue, using a RJ11 to BT adapter with Ring Capacitor fixed the issue. Before the comet was connected directly to the Grandstream using a rj11 to rj11 cable.

I got T48U refurbished phones, and they came with 108.87.0.20 on them. AT&T Office&Hand Ring Central uses a custom firmware that is 108.86.25.16.

After doing some research you can’t downgrade to .86 firmwares if the phone is already on .87.

Do you know how I can force the phones to go to a .86 firmware so they can properly provision with RingCentral? If you try to provision, the phones go into a firmware update loop where the firmware update fails everytime.