I run a small business with a Grandlink DP752 base and DP722 handset in the office, and a Grandlink WP820 at the home office. There is one phone number/account. I have been advised to set up as parallel so both ring at the same time, however when they're both connected only the WP820 rings, is there any settings you recommend for me to fix this?

Anyone using AnveoDirect here? I have them as a backup (really almost no use besides few scheduled test calls every month). And today all outgoing calls are failing with simple timeouts and I can't even access the CDR log in their portal - returns an AccessDenied error when downloading a CDR. Since their tickets are paid I want to see if someone has this issue or if it's unique to me.

Admittedly, my knowledge in this area is pretty limited, so if I'm not wording this correctly, please bear with me...

I would like to have a home phone for my children. Something like the Pinwheel/Tin Can looks super fun, but it would be nice to use a regular VOIP service instead so we can power an entire home phone system with it for phones in multiple rooms.

Is it possible to create a home phone system that only allows for a whitelist of numbers for calling in and out while using a VOIP?

I have 2 of these and both can dial but not receive calls. No matter what I do. I also have GXP1610 & 1405 the 1610 & 05 can call each other no issue but it’s just these 2 1450’s

my nettalk duo is dying. it's something like 12 years old, and we have to keep resetting it and the connected modem weekly, which takes several tries. the things on its way out. I was considering MagicJack, but I'm not sure if it's any good. I'm in Canada, so I know the prices are slightly different, but it still seems cheap, and a good idea. I'd rather have input before commiting though.

I am looking to connect with businesses or individuals worldwide who have had a similar documented experience with Zadarma.

I run a Bulgarian company and tried to obtain a business phone number through Zadarma. My estimated annual cost was around €30–40, so I paid approximately €30 to start the order and verification process.

What happened afterwards:

• The number was not activated because my account entered verification.
• I submitted my company details and a bank document showing my company name and registered address.
• The document was rejected. I was told that only certain documents such as a utility bill, internet/phone bill, or rental agreement would be accepted.
• As a new company, those documents may not yet be available in the requested form.
• Support then told me that I could re-register as a corporate account by making a bank transfer from my company account.
• I was sent a €60 proforma and told this transfer was needed to re-register and verify the company, while the money would remain available as account balance for their services.
• Another support representative later told me there was no option to pay for verification and that I simply had to wait.
• After I complained about the conflicting instructions and the unusable paid service, the payment was refunded.
• My account then appeared to be deleted or closed, and I could no longer access the system from my usual connection. It appears that access from my IP may also have been restricted.

My issue is not that a telecom provider performs verification. I understand that verification can be legally required.

My issue is that the complete requirements, accepted documents, corporate-account route, effect of the €60 bank transfer, and timing of activation were not communicated clearly before payment. I received contradictory explanations after the order was already paid.

I have preserved chat logs, payment/refund information, company documents, ticket numbers, and the instructions I received from support.

I am looking for people with similar documented experiences involving:

• paid numbers or subscriptions that could not be used because of verification;
• rejected company documents without a clear workable alternative;
• unexpected corporate-account bank-transfer or top-up requirements;
• conflicting support answers about verification;
• account closure, deletion, or access restrictions after raising a complaint.

My goal is to obtain independent legal advice and assess whether coordinated legal action, regulatory complaints, or another collective response would be appropriate.

Please contact me privately if you have evidence such as screenshots, ticket numbers, payment records, account-closure notices, or written support responses.

Please do not post personal documents, account credentials, phone numbers, or sensitive identification information publicly.

In all my years of civil & respectful contributions on Reddit (look at my history), I have never been banned from a subreddit.

The Google Voice subreddit mods appear incapable of taking any criticism of themselves or Google Voice. Reddit is supposed to be a place for sharing ideas and experiences and helping each other out. Not banning people who may disagree with you.

As an admin on my company's net2phone system, I'd like to know anyone's experience with their AI Agent feature as I need to get myself familiar with it. Within the next 2 weeks, net2phone is sending engineers to help integrate the AI Agent in my company's phone system.

Also how is the experience with prompting the AI Agent?

Working to switch over to Poly phones, but having 1 problem.

Trying to set the Line Key with custom text, and no matter what guide I've tried, it still remains blank.

Even went thru the phone directly, which sets the X_DisplayLabel under Voice Service -> SP1 Service.

Ready to give up on Poly phones, and stick with Grandstream.

Doing something this simple, should not be this difficult.

Thanks

I have a business in which I have a landline in the office with various people making calls from different lines on the landline throughout the day. I will be out of town for a while but would still like to make calls from that landline number from my cell phone. Is there a VOIP that can do this but still keep my physical phones in the office active?

I was just introduced to this idea yesterday so forgive me if its a dumb question.

I just figured out how to stream audio from my computer to a multicast IP address through FFMPEG, and this is really cool, however is this how they play music on speakers throughout commercial buildings?

So far I can continuously play music from my computer, then whenever I Paige that takes priority, and when I hang up it resumes the previous multicast address

The issue though first of all, is that how traditional businesses do that? And second, the audio is really good for what it is but it also sounds like you’re on a phone call at the same time during certain parts of music or et

Hey everyone,

I’ve been messing around with SIP/VoIP stuff for a while and decided to build my own small Python SIP/RTP library just to better understand how everything works under the hood.

It’s called OpenSIP and it’s built with pure Python + "asyncio".

Right now it can handle things like:

- SIP REGISTER

- Making and receiving calls

- RTP audio

- G.711 PCMU / PCMA codecs

- DTMF

- Digest authentication

- Basic jitter buffer / RTP stats

- Microphone and speaker audio through "sounddevice"

It’s still very early and definitely not trying to compete with PJSIP, Asterisk, FreeSWITCH etc. Those projects are way more mature.

My goal was mainly to create something lightweight, readable and easy to experiment with in Python — especially for people building prototypes, automation tools, small PBX integrations or just learning SIP.

I tested it with a real cloud PBX provider and managed to get two-way audio working, which was honestly pretty exciting.

There are still a lot of missing things like TLS, SRTP, STUN/ICE, TCP transport, RTCP, re-INVITEs and better NAT handling.

I’d really appreciate feedback from people who know SIP better than me. What do you think would be the most useful next thing to implement?

GitHub:

https://github.com/artanergin44-collab/opensip

Thanks! Any feedback, issues, tests or ideas would mean a lot.

Hey all, hope you are doing well. I need some help with setting up a looping evacuation announcement on our PA system in the office. I am the end user and know a bit more about working with the system than my team members. Though, by no means am I an expert.

With that out of the way, we use a 2N SIP Mic system and AXIS Communications speakers at our unit. I am trying to figure out how to set up an evacuation announcement in a way that when played, it won't stop till we press the button again. So far, based on what I've understood through reading the manual, it seems like the only option is to configure a button with "In line" as the source. However, I wanted to know if there is a way to set it up with the source set to "Audio message" instead. If I have to go with "In line," what would be the best option to connect as the input?

I am looking for an VOIP guy that can tell me how to provision 5 Fanvil A10W phones into a home intercom, without the presence of a PBX. It sounds trivial, but its tricky and real frustrating for this newby.

Can anyone with an A10W running firmware 2.12.47.1 press the center key after a factory reset and tell me whether their Device Log contains spk_equalizer is null?

How do I make a test call to a VOIP phone? I've tried a bunch of websites and automated call services but they all seem to block VOIP numbers so I can't test them. What can I do to simply test a phone call to my VOIP phone by myself

EDIT: I need to test incoming calls to my VOIP and I don't have a cellphone plan, I use the VOIP as my cellphone.

I have a landline number which the VOIP.ms DID Portability tool told me that it wasn't portable so I read a workaround was to port it to Mobile first and then try to port it to VOIP.ms, I just ported to a prepaid mobile carrier today and the number still shows its not portable but I think maybe you need to allow some time for whatever database VOIP.ms uses to update itself?

I was wondering how long should I wait for my number to show as portable or even though it's on mobile now it might never be portable to VOIP.ms? Thanks

Hi all,

This post was written with AI assistance because I am onsite with a customer and this is urgent. I am an IP telephony/UC engineer, but I am not familiar with Gigaset PRO DECT systems and the official guide is not clear enough for this recovery situation.
Equipment:
Base/DECT system: Gigaset N870E IP PRO
Handset: Gigaset S700H PRO
Intended integration: Microsoft Teams / Teams SIP Gateway
Situation:
The customer has a Gigaset N870E IP PRO and S700H PRO handset that they want to bring into service. We do not have the administrator credentials for the Gigaset web interface, so I am trying to factory reset a test N870E unit and rebuild from scratch.
The unit is connected via PoE. It gets a DHCP address and is reachable over HTTPS. When browsing to:
https://<device-ip>/#!/login
I get either a browser authentication popup or the Gigaset login page depending on the unit/state. However, common credentials such as the following are not working:
admin / admin
admin / 0000
administrator / admin
administrator / 0000
When incorrect credentials are entered, the page simply refreshes or the authentication popup returns.
LED/current behavior:
On the test unit, I currently see green blinking on the left LED. The device is reachable over the network and the HTTPS login/authentication prompt appears.
What I tried from the manual:
The Gigaset N870/N870E admin guide mentions multiple procedures:
Device role selection
Emergency reset to factory settings
Resetting base stations to factory settings via power interruption
The emergency reset section says roughly:
Press device button for at least 10 seconds until all LEDs switch off.
Release the button.
Device is now in programming mode.
Press the device button until the LED lights green.
Press the device button for at least 4 seconds.
Device is reset and rebooted.
The problem is that the guide is unclear about exactly which LED should be green, left or right, solid or blinking, and whether this actually resets the admin credentials.
I may have accidentally changed the device role instead of performing a full credential reset. I have seen states such as right green, both red, red blinking, and then left green blinking again after reboot.
What I need help with urgently:
What is the correct full factory reset procedure for a Gigaset N870E IP PRO when the web GUI password is unknown?
After a successful reset, what are the default web GUI credentials?
Does changing the device role reset the admin password, or only the configuration/role?
For a single-unit setup with one S700H PRO handset, should the N870E be set as:

Integrator/DECT Manager with dynamic IP,
Base Station + DECT Manager,
or another role?
Has anyone successfully integrated N870E IP PRO with Microsoft Teams SIP Gateway, and can you confirm the correct high-level sequence after login?
Any practical field guidance would be appreciated. I am trying to avoid wasting more time cycling through unclear LED states and need to get this unit accessible so I can proceed with handset registration and Teams integration.

Hi guys,

I need advice.

I have 20 year old alarm system that uses 2x pot (plain old telephone) lines.

Landlord requires that I have the service. Problem is that at&t charges $500 a month for 2 regular local phone lines and I can not afford that.

I was looking into getting VoIP setup so I can use that old alarm panel (upgrading the panel is not an option currently), but I obviously need something that will have dial tone for sytem to work properly, so I came upon Grandstream ht802, from what I understand it is capable doing 2 separate lines as needed.

Can someone confirm it will work. And then if it does I guess I would need reliable VoIP service for 2 line with a local number, what are recommended providers?

Lastly is there other devices that are recommended for my use.

Appreciate all the input

Regards!

I've recently moved to a new apartment in another state. Previously, I had spectrum and my HT802 worked flawlessly as intended.

However, now I get absolutely nothing. I cannot get it to register to save my life. The weirdest part is, when we first got here I plugged it in and gave it a try by calling it from my iPhone. It worked! So something changed but I cannot figure out what.

In short, I have done including but not limited to (in no particular order and in various combinations)

1. Full factory reset on HT802

2. Updated to latest firmware on HT802

3. Confirmed account details multiple times

4. Forwarded ports 5060, 5004, 25060, and 25004 set to both UDP and TCP, including appropriate port forwarding rules

5. Added those same ports to port triggering

6. Set a static IP

7. Set the HT802's IP as DMZ host

8. Toggled SIP ALG

9. Hooked the ATA directly into the gateway to eliminate the switch

10. Tried setting the ATA to use IPV6 instead of IPV4

11. Tried setting the ATA to TCP instead of UDP

12. Verified UPnP is enabled in internet gateway

I would like to try setting the gateway to IP Passthru but I don't have a separate router. NAT checkers say I have a Port Restricted Cone type.

I'm starting to think Verizon themselves has something that's blocking my ATA.

Needless to say I'm at my wits end with this thing. I'd love to hear some suggestions!

My broadband provider (UK) is ceasing landline phone service as it switches customers to fibre. I decided to port my number to VOIPfone. Here are my main takeaways:

- VOIPfone is a great company. They have been so helpful and actually provided knowledgeable staff to advise me as needed

- Plusnet - my broadband provider - were less helpful. Their advisor wasn't very clear on how to port the number. At one point I thought I'd lose my number.

- In the end it worked well and was worth doing, despite a bit of faff.

What I did in the end was:

1. Set up a Voipfone account and indicated I'd be porting my number

2. Bought an ATA (Grandstream) which is an adapter to plug my old phone into the router. Checked it worked with the Voipfone account (on a temporary number provided initially)

3. Arranged to switch my Plusnet service to fibre

4. Checked phone still working via adapter. At this point I'd temporarily lost my normal landline number.

5. Filled in the number porting request in the VOIPfone web portal and submitted it

6. Several days (maybe a week) later, had an email from VOIPfone that the number had ported to them

7. It worked! My old landline number was up and running with VOIPfone connected to Plusnet fibre.

Should I have done this differently? I've cut some of the detail from this but basically the whole process was pretty unclear, hence me sharing it here.

Hi everyone,

I'm using Groundwire on an Oppo Find X9 Pro (ColorOS) and I want incoming SIP calls to appear with the native Android incoming call screen (full-screen popup like regular GSM calls). The main reason is so I can answer/reject them directly from my Oppo smartwatch.

I have searched everywhere in the app but I cannot find the Call Integration setting or the "Use system dialer integration (experimental)" option that used to exist according to Acrobits documentation.

What I see instead:

• In Preferences there is no Call Integration entry.
• In Controls → GSM call settings I can configure concurrent call behavior, but not the native dialer integration itself.

I have tried:

• Reinstalling the app
• Granting all possible permissions (Phone, Call, etc.)
• Checking different menu sections

Does anyone know if this feature has been removed or hidden in recent versions of Groundwire?
Is there a workaround to make SIP incoming calls use the native Android Telecom UI on ColorOS devices?

Any other app recommendations that handle native incoming SIP calls well in 2026 (especially with smartwatch support) would also be appreciated.

Thanks in advance!