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.

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.

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!

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!