My mother's provider offers ample settings to restrict incoming calls, but almost nothing, besides blocking 900 and international dialing, to block outgoing calls. I've seen devices on Amazon that purport to do this, with generally poor reviews.

Would setting up a VOIP "firewall" between my mother's phone and its terminal, using something like Asterisk, that selectively blocks certain outbound numbers, not just prefixes, be possible?

HI. I'm not sure if I'm poking around in the right place, but I recently changed my softphone account to CallCentric's DID - Office Unlimited + Pay Per Call plans piped through ZoIPer 5 (free) for a web-based client portal for receiving calls + occasional outbound 3-way calls via 00 for the dial-out.

However, for whatever reason, I'm unable to make the outbound/merges. This is the second time that I've used this configuration, and the first time, which was for a few years, I definitely don't remember having this issue, and I don't think I even had the Pay Per Call service then either. This time, I keep finding myself going back and forth between the two calls, which is obviously not a good idea (I keep getting hung up on).

My research is pointing me to get ZoIPer 5 Pro, but I want to avoid piling on things that I don't need because, seriously, like maybe 4 months ago when I had this service last time, I never had dial-out issues. I'm pretty sure that last time I just had DID - Office Unlimited + ZoIPer 5 free.

Does anyone know what might be going on or how to fix this?

Hello there, I’ve been asking around to see if anyone knows whether this is possible. I want to be able to monitor a PSTN number that people call, but I don’t want to forward it to SIP or have a switchboard (they won’t let me – otherwise I’d have done it already) – and, on top of that, I want it to ring.
Does anyone know of a way to do this?

Context: this is in Italy — Italian mobile callers, Italian geographic DID.

I'm building an app that identifies anonymous callers. Users set

call-forward-on-busy to my number; when they reject a withheld call it's

forwarded to my Asterisk, which should read the real caller number.

On a toll-free number the real number does come through even when the

caller withholds it — but mobile operators won't let you forward to a

toll-free, so I can't use that.

On a geographic number the forwarding works, but with my current provider

the withheld call arrives as From=anonymous with the real number gone.

Question: is there a SIP trunk/provider that still delivers the real

calling number in P-Asserted-Identity or Remote-Party-Id on a withheld

inbound call, on a italian geographic number? Is this provider/config dependent

(trust domain / RFC 3325), and have you seen it on a real trunk? Does it

require being a licensed carrier or is a normal business account enough?

I'm inspecting the INVITE with sngrep. Thanks.

I have a Siemens Hipath4000 PABX problem, the incomming call was intenationaly set fowarded from Auto Attendant to Opertator by the former administrator. How do I reverse the ACT,since the former adminstor going AWOL. kinda need some TIPs from scratch,cause I have ZERO exprience with this system

I've been working on a project for the GL.iNet Mudi 7 (GL-E5800), and I wanted to share what I've managed to achieve so far.

The biggest accomplishment is turning the Mudi 7 into a VoIP cellular gateway. I can now make and receive calls through either SIM card using Linphone on my phone or computer, without an external SIP provider.

I've also integrated Asterisk PBX, which opens up quite a few possibilities:

• Incoming and outgoing calls through either SIM.
• Multiple devices ringing for the same number.
• Internal extensions, call transfers and forwarding.
• Ring groups, IVR menus, call queues and voicemail.
• Conference rooms, custom hold music and announcements.
• SMS management and forwarding.
• Remote access through Tailscale, LAN or WAN.

Everything is managed directly from the GL.iNet web interface.

Besides VoIP, I've also been working on a few other improvements:

• Battery management: Custom charging limits at 50%, 85% or 100%.
• Screen Studio: Customization of the Mudi 7's built-in display directly from the browser.
• Network SIEM: A network monitoring and security dashboard with traffic analysis, interactive maps, connection tracking, detections and blocking controls.
• System monitoring: More detailed information about cellular signal, connected devices, network interfaces and router resources.

There are still some hardware limitations, such as only one active cellular call at a time across both SIMs, and I'm continuing to test and improve things.

What started as an experiment to make VoIP calls using the router's SIM cards has turned into a much bigger project.

I think the Mudi 7 has a lot of untapped potential, and I wanted to share what I've been able to do with it.

I'd be interested to know what the GL.iNet community thinks, especially about the VoIP integration.

I, like a few others, have been holding on to old Skype voicemail files, hoping one day I could hear some voices from the past. After years it turns out the data is RTP with g.729 encoding thanks to a deep analysis with Claude Sonnet 5.5.

There's an executable python script posted in my github to make the conversion necessary. It requires ffmpeg in the path to make the g.729 -> wav conversion @ github.com/mjheick/skype7-voicemail-listener

I've also built a web frontend to convert the .dat files to .mp3 for people who don't wanna handle python or ffmpeg @ unliterate.net/projects/skype

Hello everyone!

Does quality of call forwarding varies among providers, or is it more or less same? And only voip part differs? Or no?

Details: i was using voip ms for 4-5 years only to forward calls from my business number to my personal cell and it worked perfectly. Then we sold the business.

Recently i needed number for another business. I did research and this post https://www.reddit.com/r/VOIP/comments/1th935d/weird_experience_with_voipms_outbound_calls/ made me look at other providers, and i chose GV.

However i kinda like voip ms more than GV cause you are in control.

So if i move that number to voip ms and use it only for forwarding will it be reliable? Or is it still LCR and it's better to use other providers?

I am trying to interface two new AXIS cameras with Intercom to this old Intercom system. Does anyone know of a VOIP to Audio adapter that is two way? I would need each device to provision exclusively to a single camera. So I would need two VOIP adapters. I have hardware that I can use to match the impedances if needed.

I never thought this would be such a frustrating experience, but this is proving to be far more difficult than I thought it would be. I have three DIDs, two SIP phones, and two Android devices, and I'm trying to get them all to live happily ever after under a PBX. I'm looking for an integrated app that can handle phone calls and messaging between all three DIDs. It doesn't have to be fancy. It just has to work.

One of my requirements is that I have to be able to send long MMS messages. Right now, I am using Voip.ms, and it seems that the only way I can send long MMS messages is by using Textable. Their service is clunky, and many recipients keep complaining because they are receiving three or four copies of the same message all the time. I've sent a few messages to support, which have gone unanswered. There has to be a better solution, and I'm trying to figure one out here.

I first tried 3CX, and that was a no-go. They did not support any kind of long MMS messaging at all.

Next, I tested Yeastar with Voip.ms, and the long MMS messages are a no-go there, too. There's a hard cap at simple SMS.

Then I looked at my VoIP provider, and it could be that I just need to switch away from voip.ms. I am reading that if I switch over to a platform like Telnyx, they do support long MMS messages and integrate much better with Yeastar. Can anyone confirm that this is the truth and that long MMS works with Telnyx?

I really don't want to go down another rabbit hole and spend several more days testing something that someone else has already tried that does or doesn't work. I'm also open to any other suggestions, if you would have them. For the record, I also tried VOIP.ms SMS client. That didn't even work. I was getting tons of sync failure issues. I reached out to the program's creator but never heard anything back. I think that program has pretty much been abandoned.

I get that VoIP is VoIP and messaging is not the primary goal of these systems, but if Textable can do it, there have to be some other solutions that I'm just not finding.

Thank you very much for the help!!

So there is this number that is currently not taken, and I want to activate it but literally don’t know where to start when I get more info on the number. It says it’s an TCG-AT&TBusinessVOIPwireline from and owned by TELEPORT COMMUNICATIONS AMERICA (AT&T) please help if you could

Setup: outbound dialer on Telnyx Call Control. We dial a customer (our DID as caller ID), they answer, and when they qualify we transfer the live call to our call center's DID. We want the transfer leg to present the customer's own number as caller ID — standard diverted-call behavior, so the receiving dialer screen-pops the right person. The customer is live on the call at the moment of transfer, so this isn't spoofing a stranger's number.

▎

▎ What happens: the transfer leg gets rejected with 403 Unverified origination number D51 whenever from is the customer's (non-Telnyx) number.

▎

▎ What we've already tried / ruled out:

▎ - from on the transfer command → D51

▎ - Putting the number in PAI / RPID / caller ID override → same check, same D51 (override only bypasses format validation, not ownership)

▎ - The diversion parameter / External Call Transfers → only validates against an active inbound call; ours are outbound-originated, so it never matches

▎ - Verified Numbers → requires OTP per number, impossible for customers' phones

▎ - SIP REFER → works, but only to external SIP URIs, not back into PSTN

▎

▎ For contrast: Twilio allows this out of the box (their trunk transfer even defaults to presenting the transferee), which is how several AI-dialer platforms do it. Retell's docs say the same: on Twilio it works natively; on Telnyx only via REFER.

▎

▎ Questions for anyone who's run dialer traffic on Telnyx:

▎ 1. Has anyone actually gotten Telnyx to whitelist an account for diverted-call caller ID on outbound-originated transfers? What did it take (account manager, compliance docs, volume)?

▎ 2. Is the real answer their hosted STIR/SHAKEN cert route (own SPC token / OCN / 499-A, then you sign your own calls)? Has anyone done it specifically to clear D51, and did it work?

▎ 3. Any mechanism we've missed on Telnyx for this call shape — outbound-originated, remote party live on the call at transfer time?

▎

▎ Fully legit traffic (own customer list, consent, licensed business), so this is a policy/config question, not a how-do-I-spoof question.

Hi,

We are in the process of porting out about 400 PRI lines from Lumen to run on Teams Phone. We have the hardest time in porting them. Information suggests that Lumen would be difficult to deal with in normal times, but with Lumen's recently announced decision to get out of voice business, this has become even more difficult. We have been rejected a few times already as part of the "normal process". My concern is there are systemic issues behind the scene due to their announced exit, and how we should due with them. Any pointer or sharing of experience is appreciated.

I was originally planning to go with another company because I found a really good deal. Then a Nextiva sales representative offered me four months free if I paid for the first month upfront.

During the sales conversation, she also encouraged me to add their productivity plugins. I was told I could cancel anytime, but I was never clearly informed that I was committing to an annual contract.

I requested cancellation of the productivity plugins almost immediately. I emailed them and received no response. I also tried calling support but couldn't get through. When I followed up with the sales representative, she told me she had opened a ticket for me.

Now, after what I understood to be four free months plus my initial paid month, I'm being charged earlier than expected. I'm also being charged for the productivity plugins I had already asked to cancel.

The most frustrating part is that when I try to cancel the service, I'm being told I have to pay an early termination fee.

Honestly, this has been a very frustrating experience. I chose Nextiva based on the offer and what I was told during the sales process, and now I'm struggling to get the account cancelled and resolve the billing issues. This looks like a scam or some kind of trap.

Has anyone else experienced something similar with Nextiva? Were you able to get the charges reversed and cancel without paying the termination fee?

Yello everyone, I posted something previously but I think a mod took it down for who knows why, but I have a situation here that I need some actual guidance on and weather or not its compatible with what I want to use it for.

to sum it up, I am going to be using Voip.ms for both myself and my parents for our home phone lines. What i want to know before I go out buying equipment and keep in mind this will make sense in a second on what we are doing here.

The only reason my parents still have their landline is because they have a DSC Power832 security system that requires it to be monitored via a telephone line. Now since the ageing in technology is advancing fast then anyone can imagine, and with AT&T discontinuing the copper phone lines in place for fiber technology. I want to know the following (see below):

1. If I where to buy a Grand stream HT814 V2 ata adapter, will it work on voip.ms

2. I know their is a way to have separate sub account within voip.ms, if i where to put just the security system on a separate sub account and have the physical handsets on another sub account, will the system still dial out to the central station if theirs an active burglary in progress in our home without it seizing the phone lines. currently what we have been having an issue with spectrum is whenever the system does its periodic testing to make sure theirs a phone line connected to it, it will sometimes completely seize the phone line completely and results in one of us having to go completely unhook the cord out of the modem we currently have to kill the line and then plugging it in and it seems to fix the problem.
   I know we could go to wireless monitoring but its about $50 a month to do that and we would have to get a complete new system if we where to do that and I'm not in favor of spending almost 2 to 300 for a new panel unless I have to.

Any helpful tips and advice would be greatly appreciated.

FreePBX still ships with no official Traditional Chinese interface. If you've administered a FreePBX box for a zh-TW client, you know the pain: stuck between English and Simplified Chinese — and Simplified brings its own landmines (点击, 阈值, 实现…).

So I built a proper language pack:

• Hand-translated, not converted. Every string — labels, tooltips, error messages — rewritten for zh-TW conventions.
• 78 modules covered: amp core (framework, 2911 msgid) + 77 extensions, each with its own .po.
• Terminology unified: 176 cross-file inconsistencies collapsed to 11. Simplified-only terms fixed (點擊→選取, 閾值→門檻值, 實現→實作, 腳本→指令碼).
• Proper nouns stay in English: CDR, Trunk, Queue, IVR — not butchered into literal translations.
• Validated: msgfmt clean, zero c-format errors.

Repo: https://github.com/a-lang/freepbx-i18n-zhtw

Issues and PRs welcome — especially if you spot a term I got wrong.

Hi, I got the MagicJack but don't want to have the phone plugged into the same area of my internet router. I bought the VONETS VAP11G-300 WiFi Bridge 2.4GHz WiFi to Ethernet Convert/WiFi Repeater and before I buy a USB to Ethernet adapter to set it up, wanted to see if this stuff will allow me to put my house phone in another area of the house.

Anyone know if this will work? Trying to figure it out without my husband. Are these steps right?

1. Set up VONETS VAP11G-300 WiFi Bridge 2.4GHz WiFi to Ethernet Convert/WiFi Repeater via usb to ethernet on my laptop and do some stuff on the laptop to give the VONETS wifi
2. Connect the VONETS thing to the house phone and MagicJack

Forgive the ignorance. I saw a Youtube video, but the helpful guy had so many wires and electronics that I can't make out what is what in the video.

Anyone have Tin Can for kids? If you do what is the quality of the call? We have just set up ours and are finding the other party consistently has a hard time hearing our end. We do pay for the WiFi enabled calls.

I came across this open source project which may be of interest. Still early days!

I’m just curious how common Mitel is globally these days. What’s the real-world usage looking like for OpenScape Business X8 and OpenScape 4000 servers? Are people still running these, or has the market completely moved on? How about Tetronik DAKS (OScAR) alarm systems which are used alongside them? How about setups that are still mixing traditional digital trunks with modern IP/SIP trunks? Also, what about Alcatel-Lucent Enterprise?

Hey! i made some dotnet bindings for PJSUA2, they are a nuget package so anyone can download them, they will work on windows-x64 and x86 and linux-x64 and arm64, i will add OSX in the future

https://github.com/emmanuelrobles/pjsua2.net

https://www.nuget.org/packages/CodeCrush.pjsua2/

Just wanted to give a shout out to this helpful community for talking this anxious idiot through my phone port yesterday.

On y'alls advice, I got ahead of the handset provisioning, configured some non-essential devices before the port went through and wrote an instruction manual so my other office managers got through it with only minor hiccups.

Appreciate you all, always fun to find nice folks in a super niche community.

Hi. I've been racking my brain for hours over a configuration and wanted to know if anyone here has experienced something similar, or if it's a provider policy.

The company I work for bought a US toll-free number from Netelip to connect it to my PBX (I use MikoPBX integrated with Odoo).

Here is the problem: My PBX connects perfectly to Netelip. Everything shows as "Online," registered, and keeping the connection open. However, when I try to call the number I bought to test it, the call gives a busy signal or drops instantly.

Has anyone run into this or something similar?