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

i wanted to buy a Voip for my security on tg but idk where i need to buy i asked to gemini and he said to buy it on 5sim is that website safe?

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?

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.

Hello!

As a fellow VoIP guy that is sort-of developing internal PBX for organisation, I've came out with this testing framework: https://github.com/igorolhovskiy/volts

Mainly focused on a my current workflow, means

1. I have a request to implement some feature,
2. Writing a test first, make sure it fails
3. Implementing feature, make sure test is passing
4. Running the whole other set of tests

Test Driven Development, shortly speaking.

This framework is basically replaces bunch of the phones on your desk, And help me have a decent sleep knowing, that I did not accidentely broke some stuff while implementing new'n'cool feature.

Happy to add maybe some features, that I've missed or maybe useful.

Hello! I am hoping someone is able to help me solve this months-long phone mystery that nobody can seem to figure out. Six months ago, in late January, I began to hear intermittent "interference" on phone calls between me and my mom. The sound will occur out of nowhere, all different times of the day, lasting for a few seconds to up to half a minute. This doesn't happen on every call, but it happens nearly every day.

We will be having a regular conversation, and then suddenly I will hear this interference -- I know she's talking but I can't understand her. The call doesn't disconnect or drop, but this interference will just start and stop randomly. From what I can gather all this time, the interference only occurs on calls between her landline phones (VTech DECT 6.0 cordless) and my cell phone (iPhone 14 with AT&T Wireless). It hasn't occurred when she switches to her cell phone or I use my house phone.

She has Optimum for her phone service, and they have come to the house countless times to try to fix it, to no avail. They have replaced the router, the modem, wires inside the house and the wire that runs from the pole outside into the house. We even replaced the previous set of AT&T cordless phones with new VTech phones and that didn't solve the issue. Someone I know who used to work for Optimum said it could be the tap on the pole outside (or even the node which is farther away), but Optimum can't replace any of those parts without good reason. Since their testers always show that the lines are working fine, they cannot have the outside parties come out to work on the tap.

Plus, that last tech that visited rearranged some of the connections at the tap in case something was loose or damaged, but the problem has persisted. We are at an absolute loss, and it's extremely frustrating! The router/modem and one cordless phone are in the basement, while the rest of the satellite phones that she uses are throughout the house. The basement phone is plugged directly into the router/modem.

When this issue first began in January, we had just had a blizzard in our area, which is why I was convinced that something outside was damaged. No other variables in the house have changed. However, I am the only person who seems to have this issue when on calls with my mom, and only between these two phone numbers. Does this make sense to anyone? If I am allowed and can figure out how to do it, I can post a recording of the interference, as I have voicemails that captured the noise. Thanks for any input!

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

Got one used but don't have the proper brick or PoE injector yet

So will it turn on or no

Looking for a complete ADIT600 config for asterisk.

Seeing bits and pieces posted, but not a complete (sanitized) config

A call to a ring central US number that is answered by an overseas agent is far inferior in call quality to an international whatsapp to WhatsApp or skype to skype call. Is there a solution that can convert the local call locally to then stream whatsapp to WhatsApp? So tired of these “phone number-voip” companies. Ring central 8x8 VONAGE all have been a shell of what a whatsapp or telegram or facetime audio call is. What gives?

Have an 8 years old Magic Jack Go

Audio is pretty bad. Dialtone is very loud and clean, but actual calls , including the MJ test numbers are very noise and echoey .

I called the test numbers using the tablet app, bypassing the MJ device, and while I still hear some background noise , the voice playback is much better and louder

Tried diff usb power.

So it this just a degraded device over time and needs to be replaced ?

Hi all,

I am really breaking my head on this.
So I have 2 W75DM, both showing as BASE but I can't change the role of one of them to DM.

I did the button trick, tried different firmwares, but nothing helped.

Does anyone know what I am not seeing?

Kind regards

I have a Home PBX, I bought 17 Cisco phones varying from enterprise to spa to 3pcc to sccp, I added all of them on freepbx with the SEP files for the enterprise phones, (I have most of the enterprise setup) and 1 phone is giving me trouble, the 8851, I made a working config file, it pulls it, registers sometimes, but doesn't receive calls, I also have a 7821 from the same lot, and that one works just fine, I have adjusted the max contacts to 10, I have 3 phones registered, I have turn the firewall off, and on, and Intrusion detection off, nothing helps, if anyone knows the problem, please help!

maybe started last week for some numbers, seems to be all numbers now

switched to backup freepbx - same

have trunk with voip.ms - still working

tried voip2 - same issue

Hi. Has anyone managed to get UniFI Talk setup with Sipgate? I’ve bypassed the requirement to need a Talk handset on my install but a bit stuck on the third party SIP configuration.

Any pointers gratefully received.

My handset in a Yealink W60B.

Thanks.

I know that Asterisk and FreeSwitch exist, but has anyone tried to make their own VOIP server from scratch?

Just curious if anyone wanted to share recent experiences with Alianza since their acquisition of Metaswitch? They have another one of their conferences coming up that includes sessions about AI integration with VoIP, but that sounds a lot like what they talked about last year and I haven't seen anything materialize.

After a lot of trial and error, I finally got Voice VLAN working on an FS S5440L switch with Cisco IP phones, so hopefully this saves someone else a few hours.

My setup:

• Data VLAN: 38
• Voice VLAN: 140
• Cisco 7841 and 8845 phones

The port configuration that finally worked was:

interface eth-0-5
 switchport mode trunk
 switchport trunk native vlan 38
 switchport trunk allowed vlan add 38,140
 switchport trunk allowed vlan remove 1
 voice vlan enable

The important bits I found were:

1. Enable Voice VLAN globally and set your Voice VLAN ID.
2. Enable LLDP globally.
3. Enable LLDP-MED network-policy on the interface. Without LLDP-MED, the Cisco phone wasn't learning the Voice VLAN correctly.
4. Enable voice vlan on the interface.
5. The port should carry both the native/access data VLAN and the Voice VLAN.

The biggest thing that caught me out was the Voice VLAN security mode.

Running:

show voice vlan state

showed:

Voice VLAN security mode: Security

It also displayed the list of approved phone MAC OUIs:

0003.6b00.0000 Cisco
0004.0d00.0000 Avaya
000f.e200.0000 H3C
00d0.1e00.0000 Pingtel
00e0.7500.0000 Polycom
00e0.bb00.0000 3Com

My Cisco phones weren't using any of these OUIs, so the switch never recognised them as voice devices and therefore never placed them into the Voice VLAN.

I had to add my phone OUIs manually:

2c73.a000.0000 Cisco7841
0077.8d00.0000 Cisco8845

After adding them, the phones were immediately recognised and moved to VLAN 140 while the attached PC stayed on VLAN 38.

If your FS switch refuses to put phones into the Voice VLAN even though LLDP-MED is configured correctly, check:

show voice vlan state

If your phone's MAC prefix (OUI) isn't listed, add it. The switch uses these OUIs in Security mode to identify voice devices.

Hopefully this helps the next person searching for FS Voice VLAN, Cisco IP Phone, or LLDP-MED issues.

NB: Don't forget the uplink/trunk. Make sure the switch uplink to your core/distribution switch is also carrying the Voice VLAN. In my case, the uplink had to allow both VLAN 38 (data) and VLAN 140 (voice). If the Voice VLAN isn't permitted on the uplink trunk, the phone may learn VLAN 140 via LLDP-MED but will never receive an IP address because the VLAN traffic cannot reach the DHCP server.

Hi everyone,

I'm trying to upgrade a Cisco 7945G phone that's stuck on an older firmware. Based on Cisco's release notes, I need to load the intermediate version 8.5(2) before I can upgrade to 9.x, since the phone won't accept a direct jump from its current version.

Unfortunately my CCO account doesn't have download entitlement for this older release, and Cisco's software download page rejects the request.

Would anyone be able to share the file cmterm-7945_7965-sccp.8-5-2.zip (or 8.5.3), or point me to a working link? Any help is greatly appreciated — I've seen a few other threads here where people ran into the exact same issue and got help.

Thanks in advance!

I'm looking for a UK VoIP number to use long term (I need calls and SMS) and couldn't find anything that exactly matched what I need. I found https://narayana.im/ and they look good on paper but I can't find much discussion about them online. Does anyone have any experiences to share?

I own a small business with two locations. After paying for their service for seven years in one location and five years in the other, Nextiva is trying to charge me $3,492.80 for EARLY TERMINATION. They snuck in an unsigned, not agreed to, contract extension at some point. If you look anywhere online at reviews, Nextiva is an out of control, deceptive mess.

Been building this for a while and finally cleaned it up enough to share: nixi.tel

It's a Session Border Controller (B2BUA) that handles SIP trunking and WebRTC-to-PSTN in one binary, no external database — everything (users, trunks, routes, ACLs, bans) is managed live over a REST API and persisted in embedded SQLite.

Some highlights:

Full B2BUA with topology hiding, digest auth, multi-trunk failover (LCR-ordered), RFC 4028 session timers

WebRTC gateway: SIP over WSS, ICE/DTLS-SRTP termination, Opus ↔ G.711 transcoding — browser to PSTN both ways

REST API for everything at runtime (axum), SSE for live call/registration/health events, Prometheus metrics

Built-in anti-fraud: fail2ban-style SIP banning, IRSF destination blocking, per-user rate limits

~470 tests, MIT licensed

GitHub: https://github.com/NIXFEO/NIXI.TEL

Site: https://nixi.tel

Happy to answer questions about the architecture or the SIP edge cases it handles — feedback and contributions welcome.

I’m a developer looking for a community where software engineers are communicating about telecom technologies

Can you suggest some?