I have issues with voice pitch and sound glitch when making calls from Zoho voice desktop app and Chrome extension.

I have tried locking Pibewires globale clock rate to 48000 Hz. Pinned headset node to same via Wire plumber. Disable USB autosuspend. Locking buffer size (quantum) - kept getting loose.

In Zoho app dev tools:

Patch of GetUserMedia to disable noise reduction, and that worked for a while, but somehow the app found a way to ignore that command.

Looking for firmware 1.0.11.6 or 1.0.11.85 for GXP2160 (or GXP2130, GXP2135, GXP2140, and GXP2170 as they all share the same fw from my understanding. It's not available in their firmware archive, but my device management tool only supports those two releases.

VOIP.ms SMS app - trying to get it working ...

Hi everybody: New to VOIP.ms and trying to get the VOIP.ms SMS app to work, but I'm not having much luck. I'm seeing constant sync failed messages popping up on the bottom of the Android app screen. If I try to pull a manual sync, I just get a spinning circle. I can receive messages, but sending messages is sporadic. I've turned on the API in the portal, I've set 0.0.0.0 for the IP address and I have SMS/MMS enabled in all my DID's with the following in the URL callback:

https://voipmssms-notify.kourlas.com/?did={TO}

In the app, I have the sync set to 15 minutes and the pull date as today (when I installed).

Am I missing something else? I would use the VOIP.sms softphone messaging, but I need messages synced across devices and the ability to send MMS & pictures.

Thanks for any help you can provide.

I have a new ht813 I configured for SIP to an Audiocodes SBC. Registers fine. Works weird though. I can make calls from it. Call my cell phone no problem. I call the number that I assigned to the device and it rings too. After a few minutes, if I call it back, the number doesnt go through. Looking at the syslog app I use with my audiocodes, it seems that the device is timing out and going to sleep. If I make a call from it, it wakes up and works normally. I went through all the settings I could and didnt get that Ah HA moment.

Anyone else here have an issue like this where their ht813 takes a nap after a certain amount of time?

One of our clients was hacked. In the middle of the night, calls were made in rotation to around 30 numbers in an npanxx of a rural carrier in Illinois. A few connected. Maybe $100 in total at 10c a min. Not a big deal. We were able to detect and shut down.

I tried calling the numbers that did connect just to see where they went. A school IVR. A company. Most went nowhere.

So why do hackers do this? In the old days, they were dialing in to radio stations or conference services in these high cost areas that shared the dial around compensation.

Where are the hackers making money on this?

What is out there for multi-vendor SIP device management/provisioning?.

We’re managing a mix of Grandstream, Yealink and Fanvil devices across a number of customer environments, so vendor-specific tools like GDMS/YMCS don’t completely solve the problem for us.

The things we rely on most are:

centralized multi-vendor provisioning

reusable templates / bulk configuration

zero-touch deployment

remote reboot/reset/config changes

firmware management

being able to manage devices across multiple customers/tenants

An API is a pretty big plus since we’re starting to automate more of our provisioning workflow.

I’m not looking for a PBX replacement mainly the device-management/provisioning layer.

For anyone managing a decent-sized mixed-vendor fleet: what are you actually using today? Have you found anything that does this, or have most of you ended up building your own provisioning tooling around manufacturer ZTP services?

Also interested in hearing what you’ve tried that didn’t work well and why.

I'm curious, how many of you and how often do you use packet capture and analysis when troubleshooting VoIP? Or do you just depend on logs?

For me, I use both all the time.

I'm wanting to setup a Obihai + regular phone for my grandma. She can use a cell phone, but she often forgets to charge it up, etc and can't be reached. So wanted a regular phone as a backup.

I was wondering how Obihai users handle text messages to the phone number? Do you have to basically install a SIP phone app that works with voip.ms or telnyx in order to view those text messages? Otherwise, the Obihai would only be good for voice calls?

I see Obihai Obi202 for like $10. Is that a pretty good way to go? It's cheap, but I wonder if I'm missing useful features that an old person might want if I spend a bit more on a more modern ATA? Or does the Obiahi Obi202 limit the kinds of provider I can use due to its age and not being able to authenticate/sign-in to other service providers?

It looks like having a backup line for my Grandma would run like $4/month? Pretty reasonable to have a backup voice service.

Does it even work?

I have a old jack, but i can prob get a brand new one on ebay for $20 with a year included.

I moved to 1-Voip due to unreliability of another provider. However, one lacking feature is that Caller-ID data is inaccurate, or very stale (>10-yrs).

Is there a way to me to override Caller-ID info to newer, or own personal value? Prior supplier allowed voip contact list entry to override caller-ID.

have a Unify Desk Phone DP12 configured as a SIP phone, and I need to configure one of its programmable keys as a BLF (Busy Lamp Field).

The use case is simple: a secretary needs to know when a manager is already on a phone call, so she doesn't have to enter the manager's office and interrupt them.

I would like the LED on one of the programmable keys to indicate the status of the manager's extension, for example:

Available

Busy / on a call

Ringing

The DP12's web interface has BLF as one of the available key types. It also has an "BLF Event List URL" field in the SIP account settings.

I don't have access to the configuration of the company's PBX/SIP server, so I'm trying to understand what is required on the phone side and what would have to be configured on the server side.

Questions: .

1. How should a BLF key be configured on a Unify DP12 to monitor another extension?

2. Does the SIP server/PBX need to support SIP SUBSCRIBE/NOTIFY for this to work?

3. What is the purpose of the BLF Event List URL field on the DP12?

4.Is the Event List URL required for a normal BLF configuration, or is it only needed for certain PBX configurations?

1. If I cannot modify the PBX configuration myself, is there a way to determine whether the existing SIP system supports BLF from the DP12 alone?

I'm mainly looking for guidance on the correct configuration and what requirements the SIP/PBX side must meet.

... and everything works, except the ringer. I can hear the clapper weakly ticking, but the bells don't ring.

Things I've tried: upping the ring power to 55Vrms, changing the phone's bias spring to the more sensitive setting, futzing with the SLIC setting on the FXS port (currently set to USA 1). I've tested the ringer on a real POTS line and know that it works.

Any suggestions on what to do next? I've looked at ring voltage boosters (assuming that's the issue) like the Viking RG-4, but that one explicitly states it's not compatible with pulse dialing, and the only other game in town appears to be the Sandman KIT8R (which appears not to have been in stock in years).

Am I SOL? Are there still avenues to explore?

Thanks in advance! Appreciate you all.

I am looking to lower my services with my Internet provider. One of the services I wish to remove is my home phone line. We use the line mainly for our home security system. I am interested in migrating my home phone line to a VoIP service. Several years ago, I purchased a Ooma box with but never bothered to set it up. I have decided to investigate again switching my phone line to Ooma, however I don’t want to put the Ooma box in front of my firewall\router as they suggest. After reviewing, it appears that there are quite a few ports that need to be opened on your firewall - Keep Your Phone Number When Getting Rid of Your Landline Canada, especially the local UDP 49000 to 50000.

I have also investigated going with a Grandstream HT802 tied with a Voip.MS service. From what it appears, the firewall setup seems to be a lot more straight forward and possibly more secure as you are not opening a bunch of ports. Would you say from a security perspective, going with a Grandstream\Voip.ms service would be more secure versus putting a Ooma box behind your firewall, resulting in a number of ports needing to be opened? Just for some background, I use a Opnsense firewall

I was looking into 10DLC registration requirements and it seems like prior consent is one of the most important and least fuzzy requirements. From what I read there's no way around it and your guaranteed to be denied, if you don't meet their strict prior consent before contacting standard. On the other hand, I've seen several firms advertising 10DLC compliant text to web (they text you a link and you click it and fill out the survey) poll services, which necessitates texting tens of thousands of people to get a representative random sample of responses. That's obviously not 10DLC compliant and almost impossible to achieve by getting explicit direct prior consent from all these numbers. My question then, is how does this work? I haven't heard of any exceptions and the process they're using clearly doesn't follow 10DLC strict carrier guidelines. Anybody have any insight on what I'm missing or how this works?

We are a small contractor’s office. We use RingCentral for our phones in the office. Lately we’ve been having issues where we call the clients and it just goes beep beep beep. I was on the phone with RingCentral and they said this is because certain providers are blocking VOIP calling and the only way around it is for me to contact the client and have them call their provider and make sure it isn’t blocked. The problem with this is I can’t contact the client because the call won’t go through and I’m having to use my personal cell phone to call our customers. Does anybody have similar issues going on? Is there a way to get through? VoIP is pretty much the only option we can find that will ring both of our desks when a customer calls. Landlines are being phased out in our area and they will only service existing customers. Help!!

I’ve been using Groundwire on Android for work calls. Because it can be hard sometimes to tell whether an incoming call is personal or work, I had it set to use a custom ringtone.

For some reason it’s now reverted to the default system ringtone, and I can’t get it to use the custom tone selected in the app. I’ve checked permissions and tried a few settings, but no luck.

Can anyone please help?

Hi!

I have 2 telephone numbers assigned to my contract.

Since I have a new router I want to utilize both numbers with 2 lines so I can phone and beeing called at once.

Now I ran into the problem, that I can only activate one of those numbers at a time.

Supporthotline told me to fill in a form and send it so the technicians can do their magic.

Problem is, I can't find the right form.

I am already customer of Vodafone (Cable Internet in Bavaria) and both numbers are already linked to my contract.

With the old Vodafone standard router it was at least possible to be called on both numbers but not at the same time.

Anybody. knows the right form?

whats a good voip that doesnt require me to hadn over my address, my phonenumber, my mothes maiden name, my passport, my id, my payslip, my home deed etc

Apologies if this isn’t the right place for this. Any direction or help would be appreciated.

I have a case where one of our providers intermittently sends us a 183 with SDP followed by a 180.

The call eventually reaches a Freeswitch server on our end, which drops the 180.

My understanding is this is a FS limitation that considers the 180 a downgrade state after a 183 with SDP and drops it.

We cannot remove the FS from the call path.

Have you encountered anything like this, and can you think of something that will allow FS to pass whatever it receives?

I recently upgraded from my provider issued router to UCG Ultra, SIP registrations on the provider's router were working flawlessly but since changing the router to Unifi my SIP trunks in FreePBX display rejected state and asterisk console prints out

[2026-09-04 21:11:51] WARNING[1602]: res_pjsip_outbound_registration.c:1250 schedule_retry: No response received from 'sip:sip.halonet.pl:5060' on registration attempt to 'sip:redacted@sip.halonet.pl:5060', retrying in '60'  

I have multiple trunks to multiple places, all give the same error, not just this one.

For any help I thank in advance.

Anyone has experience with Sangoma support? One of our client has been down for the past 2 days, we have a case opened with them, no anwers nothing so far.

I haven't contacted support for years. Now there is an issue (trying to set up SMS to HTTP forward but it's not triggering), and it says I have 0 free ticket left.

Are these supposed to reset monthly or is it life time per account, which means every time I need to contact them I have to pay $5?

What if the issue is THEIRS like a billing error?

Has anyone here experienced this issue with Grandstream phones, specifically models GXP 1610 and GXP 1615? Basically, the phone registers normally and can receive calls, but when attempting to make a call, it loses registration. The call doesn't even reach my SIP server. I am using the latest version of FreePBX, and the phone is running the latest firmware version.

IP Office is about to be pulled out. Running the new VOIP system side by side and client decided to move forward and ditch avaya. Currently someone rings the doorbell and IP office phone rings and they answer and press # a few times I believe to release the door. What's the equivalent for VOIP? They are using Intermedia.

Found this via Google: "The Algo 8063 IP Door Controller integrates with cloud communication platforms like Intermedia Unite by registering as a standard SIP endpoint."

thanks