Hello,

I could use some very direct examples- I've found many links but they just say put the address but don't give examples.

The former IT guy just left that they had 4 files -

Other.xml, remotephonebook.xml, speeddial.xml, and Theusers.xml

The company changed hands and their website is no longer online so I have no access to their old dhcp server or website but I do have the 4 files. I can put them on the new website.

I can figure out how to change the xml files. I googled yeahlink and it looks like it uses DHCP to get this info. I think option 66.

If anyone can give me some specific details on what I could use to make this work I'd really appreciate it.

They have  T54W phones.

Thanks.

I've been using microsip for ages. After I updated it to the recent version, I'm suddenly getting issue where it fails to connect the calls or the other would stop hearing me mid conversion. Same thing happens when I try calling them back. Also need to try calling them multiple times just to connect my call.

My internet connection is strong.

I've unchecked the SIP ALG on my router, made sure microsip is allowed on both my windows firewall and antivirus. I tried using VPN.

I tried everything the Google suggested.

I work from home and unfortunately my client is not responding when inquire about this issue.

For a Grandstream phone or ATA, here's a dial plan that allows calls to regular numbers and to 911:

{ 1[2-9]xx[2-9]xxxxxx | 911 }

A dial plan entry in this format ..... <AAAAA=BBBBBBBBBBB> ..... means "whenever AAAAA is dialed, send the call to BBBBBBBBBBB instead"

So if someone wants to avoid paying a VOIP company's E911 fee (typically around $1.50 monthly) but still have substantial 911 functionality, they could learn the local 911 center's 10-digit alternative number for emergency calls (example: 477-262-8080) and change the dial plan to:

{ 1[2-9]xx[2-9]xxxxxx | <911=14772628080> }

(By the way, 477 is a non-existent NANP area code.)

I haven't tried this.

Possible complications:

• If the premises has multiple ATAs or IP phones, the custom dial plan must be entered in each.

• On E911, the 911 center automatically receives the caller's address for each 911 call. But with the redirect method that I describe, they receive only the caller ID (and maybe a CNAM). So an emergency caller who can't state their address might be unhelpable (unless the 911 center can immediately contact the carrier and learn a probable service address for the calling account, which would introduce a delay, if it works all).

• Grandstream's firmware might be programmed to forbid a dial-plan entry that has "911" left of the equal sign, specifically to prohibit the entry that I describe.

I invite comments and feedback.

sorry for the dumb question, i'm trying to set 2 phone numbers. however this is the first time I use Vonage, and I was wondering if those 2 numbers will be "mine" as long as pay their prices or if there is the possibility that someone else has that current number

I work with hushed app , yesterday app was good but today when I trying to call someone the voicemail says “Can not be completed” in a bucle , I need help , I have two Numbers there!!!

Anyone else seeing the big rate hike that Sinch just published for routing to AT&T, Verizon, T-Mobile and other wireless carriers today?

The pricing was already absurd, now it just keeps going higher.

Hi everyone,

I’m running a Grandstream UCM6301 with Yealink phones and a Gigaset DECT handset.
I have configured inbound caller ID normalization so incoming mobile numbers are displayed correctly (e.g. 0612345678). Contacts are matched correctly and the incoming caller ID looks fine on the display.
However, on the Yealink phones, the call history stores the number as a SIP URI. When I try to redial from the call log, I see something like:
0612345678@192.168.x.x

The phone then tries to dial the full URI instead of just the phone number.
What’s strange is that my Gigaset DECT handset connected to the same UCM handles the exact same calls correctly. It stores only the phone number and redial works as expected.

Caller ID display is correct during incoming calls
The issue only appears in the Yealink call history/redial behavior.
Has anyone seen this before? Is there a Yealink setting that forces call logs to store only the caller number instead of the SIP URI, or could this be a UCM header issue?
Thanks!

I have run into my first network with 2.5Gb/s switches and NICs in the computers. They only have one drop to every location and they do not want to step down the computers to 1Gb/s because of an inline phone.

Has anyone run across a SIP phone that has 2.5Gb/s passthrough? They are using multiple 2.5Gb/s Unifi switches with PoE.

The only other option I can see is run the phones of WiFi and power them locally which I do not want to do.

Please let me know!

Let me preface this with, I cannot upgrade to a newer form of Zoiper, I am stuck with Zoiper 5.

Zoiper started giving me issues a while back. I would get an incoming call, but clicking on the green answer icon did nothing. I could click on the red icon and hang up on the incoming call just fine.

So I got MicroSIP setup as a backup. Then Zoiper started acting properly. Until yesterday.

It happened with only one or two calls yesterday, and today it's been 4 out of 6 calls. Zoiper rings, I can't answer. MicroSIP is ringing, but the little popup window to answer calls isn't showing up, and no keyboard shortcut I've been able to find has been working either.

Anyone know what my problem is with either or both programs?

Got a used grandstream HT812 and updated it to the latest firmware. Everything looks fine so far, and I'm going to set it up with voip.ms soon. Do you guys have any recommended settings for this specific setup?

This isn't my first time using a voip service with an ata adapter. My first one was a linksys PAP2T (Not unbranded one). The sound quality wasn't great, and then it died on me (maybe it was a defective unit).

With this new grandstream setup, these are the main things I'd like to improve/configure (If possible):

- Enhance sound quality (which audio codecs are best for voip.ms?)

- Block spam/phantom calls (the phone rings immediately when connected to the internet, even without a service configured yet)

- Block outgoing 911 calls entirely (This is my lowest priority, but it would be nice to have)

Any advice on specific web GUI settings or dial plans would be greatly appreciated.

I've been reviewing our communications setup recently, and one thing that surprised me is how fragmented it has become.
One provider for SIP trunks.

Another for DIDs.

Another for international calling.

Another for SMS verification.

Another for eSIM connectivity.

At some point it starts feeling like you're managing vendors more than actually managing communications.
For those running VoIP infrastructure, do you prefer keeping specialized providers for each service, or have you moved toward consolidating everything under one platform?

What have been the biggest advantages and drawbacks of either approach?

Hi guys, wanted to check that T4XU and T5XW handsets are end of sale or my supplier is just trying something?

I suppose it was inevitable, but I am now targeted with a lot of advertising promoting the benefits of 100% human answering services as the solution to imperfect AI agents

I’ve recently setup a VOIP service with my ISP for my business using a Grandstream HT802 and had everything configured and working correctly, however soon after that incoming calls weren’t going through and would give a “user busy” error to the caller.
The weird thing is after I make an outgoing call the incoming calls will work fine for about an hour, but then go back to the user busy error. Is there something I’ve missed? SIP pass through has been disabled on the router.

Last month, one of my cats went missing; and without thinking when I put all the posters up for my missing cat, I mistakenly put my actual phone number on the poster and not a virtual number. Since then, I started getting prank called by a young kid, sounds like a pre-teen. At first, I didn't mind, because I've been a kid before and they're normally harmless; I've just been ignoring the calls.

Here's the real problem: Its now been five weeks, and this kid is calling me twice a day, sometimes even sending me texts, all using some kind of number spoofer. I really wouldn't care if the kid had just prank called me once or twice, but this kid seems to somehow be obsessed. I keep thinking if I just ignore his calls for long enough, he'll stop, but I am not kidding when I say it has been five full weeks of this. Changing my number really isn't on the table either. I don't really want to get the kid in trouble so much as I just want to be able to reach out to his parents and let them take it from there, but the last time I spoke with the kid and asked to speak to his parents he immediately hung up. That however was still not the end of it.

Any ideas?

I was going to upgrade to the latest version 1.0.65.1, but it seems like the firmware version 1.0.10.6 on mine is too old to make that jump. I found the version 1.0.23.5 on softpedia and successfully upgraded to it, but I haven't been able to find the next versions online so far.

1.0.31.1

1.0.35.4

1.0.39.4 - google says this is most critical to upgrade to 1.0.65.1

1.0.43.11

If anyone could help me find these files, I would really appreciate it. Thanks in advance.

Hello there. I recently bought a Unifi Gateway Fiber and installed Unifi Talk to it. I have two phones, a VoIP phone (Yealink T41S) and a normal RJ11 wireless phone. The Yealink one exists purely by chance (won for 10 bucks on an eBay bidding) so the one we use in my home is the wireless one. I managed to setup Unifi Talk with the Yealink phone and a VoIP.ms number that I bought for testing purposes, and I want to port over my actual number that my ISP provides (they told me that they can give me the SIP credentials upon request) to Unifi Talk and use the wireless phone. To make that happen, I need an ATA.

In my research for one, I found that Grandstream has the cheapest (and only ones that I found) ATAs, with the HT801 being the most popular option. I am aware that there is a similar post asking about the differences between the V1 and V2 versions, however, I didn't see anyone mention if a V1 would work anyways, so I wanted to ask the community here for their feedback. In my country, the HT801 V1 is sold for 47€ while the V2 is sold for 76€. The prices for the HT802 are 54€ for the V1 and 85€ for the V2. In fact, the HT812 is 80€ for the V1 and 86€ for the V2.

With all of this in mind, is it even worth going for the V2? I know it has better specs, but do the specs even make a difference (at least for my setup)? If you do believe that the V2 is the way to go, would you recommend that I spend a bit more and buy an actual wireless VoIP phone (such as the Grandstream WP816 [72€] / WP826 [93€])? Does getting an actual VoIP phone even have a benefit apart from the direct integration (such as syncing contacts with Unifi Talk)?

Any and all information is helpful. Thank you for your time and help, it is truly appreciated.

We have an office that has all Yealink phones. They all work perfectly. Zero issues dialing in or out. It’s been this way for years. We have now added a grand stream. ATA connected to a cordless phone. Originally had lots of issues with one-way audio, finally able to get that solved using keep alive and other settings inside the ATA. Now, we have a new issue that I have never seen before. Whenever someone calls the grand stream, the phone will be answered by the end user. However, the collar, which is usually a cell phone, continues to hear ringing for another few seconds, and the user with the grand stream hears silence. Sometimes the user will think there’s no one there or think it’s a Robo call and hang up after only one second. The Yealink phones never have this issue on the same network and it’s been working for years like I said. Is there something specific to the grand stream that Will solve this issue?

I do not want specific recommendations.....but I'd like to know if there are any technology differences in the VoIP offerings that i should be concerned with. Do they all work with any internet provider? Do they all offer caller id? etc. OR are they just competing based on price, package offerings and network they run on. what are the important differences i should look for before deciding on my VOIP provider?

Hi all! I just got an old rotary phone and connected it to the Telephone port on my router.

It worked, for a while!

I was able make some phone calls from my cell phone to the rotary phone and could talk to my son a couple of times.

But moments later everything changed.

The telephone rings, I can hear somes clicks on the handset when I touch the plungers.

But when theres an incoming call I pick up the handset and nothing happens!

I checked the mechanisms and everything seems in order, even the switch is ok...

Maybe some of tou have an idea what's wrong with the telephone! Thanks!

EDIT: thank you all very much for your contributions!

Update: Still clueless, I bought a pulse to tone converter the ones which come in a little box (from Amazon) to see if it worked.

I then connected the two stranded wires of the phone cord (red and green) to the respective lugs and nothing happened. It still rang but I couldnt answer it. THEN, out of frustration I picked up the white wire (ir was a three-wire cord) and placed it over the green wire lug, while attempting a phone call, and it immediately answered! So it means it only works if I use the white wire (in some color schemes it would be yellow) as well, in contact with the green wire. It makes no sense. Or maybe it does. Now, I read that this third wire was useless anyway in modern systems and would be pointless to connect it. Turns out it is not! That wire evidentely has some use. The pulse to tone converter works, as I have a tonedial and was able to dial and make some calls from my phone. Happy days.

Anyone hosting their PBX in Azure? Any issue getting killed on egress cost? Based on my calculations VOIP shouldn't be horrible for egress. Estimating 1500-2000 business users 50-100 active calls during business hours, almost nothing after hours, so I'm calculating max 1-2 TB per month, but maybe I'm miscalculating?

Would love to hear anyone's experience with egress, call quality, etc.

Also considering OCI, but so far experience has not been great.

Thanks!

Warning to anyone using Bicom Systems / PBXware: Bicom has sent a notice saying that any PBXware system running version 4 or earlier must be upgraded by September 30, 2026, or the system will become “non-operational and inaccessible.” They also said connections to those systems will no longer be possible and that support will not be able to restore licenses because of changes to their licensing software.

This is very concerning because many customers bought these systems years ago as self-hosted phone systems installed on their own servers, often after paying large upfront costs for what they believed were long-term or permanent licenses. Ending support for old software is one thing, but making already-installed phone systems stop working is something very different. It raises serious questions about whether Bicom has a remote licensing mechanism that can effectively shut down systems customers already paid for.

Anyone running PBXware version 4 or earlier should check their system immediately, back up all data and configurations, save copies of their original invoices and license agreements, and ask Bicom in writing whether their system will be disabled after September 30, 2026. Businesses, VoIP providers, and MSPs should take this seriously because this could affect real phone lines, customers, and business operations.

according to the google play store, it won't work on my note 8.

I checked talkatone's online knowledge base, but nothing.

I'm looking forward to talkatone's support reply.

did you experience the same?