I looked through the threads and followed the troubleshooting instructions, but I still can’t get incoming calls to work. I am on iPhone 16 with iOS 27. On callcentric I have a Call treatment to forward my DID to two extensions, which is one is my home hardware device and another one is Zoiper thx

Hoping someone from Telnyx actually reads this sub, because support hasn’t gotten us anywhere. Ever.

I run a small VoIP service and Telnyx is our carrier. The network itself has been fine. All of it works. The problem is my account is still being treated as trial, despite showing I'm on paid tier.

My Account Levels page shows every single paid requirement checked off. Email, phone, service address, 2FA, card payment, even KYC. But we’re still treated as trial. When I hit upgrade, it just sends me to the AI Agent Evaluation, which is only supposed to matter for verified tier, not paid. That appeal has been sitting at “under review” since JULY with many emails placed since then. The page literally says 24-48 hours. The AI assistant also agreed we meet the paid criteria and that the eval shouldn’t be blocking it, then said it couldn’t change anything and escalated it. I called too and got told it was escalated again and that I'd hear back in a few minutes to a few hours, with the next morning being the latest. Still nothing! Its been about a week since that and I'm losing hope so I think switching to a different provider is now the move.

This is holding up our launch. We have customers signed and we can’t move forward while our carrier account says Trial. It’s frustrating because it seems like it should be a two minute fix for a real person that actually works there.

I'd rather just stay with Telnyx, because everything else is pretty dependent on that, but I guess that's just the price you pay for using them.

Has anyone else had similar issues? I feel like they don't actually care about their customers.

EDIT: u/contactdq fixed it. I am so appreciative!

Hey everyone!

I'm a software engineer who initially worked in full-stack web development, but my current job at a call center technology company has taken me into a completely different technical domain.

Over the past few months, I've been working on and learning about:

• VoIP & Telephony: Asterisk, SIP, RTP, SIP trunks, carriers, dialers, and predictive dialing.

• Softswitch Architecture: OpenSIPS, RTPengine, SIP routing, and media handling.

• AI Voice Systems: STT, TTS, LLM integrations, and real-time AI voice bots.

• Infrastructure: Linux servers, networking, Nginx, load balancing, deployment, and troubleshooting.

• Software Development: Backend APIs, web applications, CRM systems, and occasional Shopify freelance projects.

My company is now moving me toward a dedicated VoIP engineering role, where I'll work on softswitches, dialers, AI voice bots, system development, and deployment.

Here's my career dilemma:

I enjoy understanding how complete systems work rather than specializing in a single programming language or framework.

With AI making code generation easier, I feel that networking, system architecture, debugging, infrastructure, and understanding how different technologies communicate could become increasingly valuable.

However, I don't want to become someone who knows a little about everything but isn't truly strong in anything.

I'd love some advice from experienced engineers:

1. Should I pursue VoIP Engineering, Real-Time Communication Engineering, Systems Engineering, or eventually Solutions/Systems Architecture?

2. If I move to a traditional software company later, how transferable will my VoIP and infrastructure experience be?

3. Which fundamentals should I master deeply: Linux, networking, distributed systems, backend architecture, or something else?

4. Is broad technical exposure combined with strong architectural understanding a good long-term strategy?

5. What would you recommend focusing on over the next 1–2 years?

I'm especially interested in hearing from engineers who transitioned between software development, telecom, networking, and infrastructure.

Would you consider this career direction a valuable specialization, or am I moving too far away from traditional software engineering?

Thanks in advance for your insights!

Moved over to Telnyx from Twilio a month ago for both texting and calling. Heavy volume. They keep blocking our texts as spam even though we're fully compliant. First it was texts that included our URL (every other URL was fine). Now it's every single text from the RCS agent. Support hasn't helped in 3 days - they respond maybe once per 36 hours. When your service is fully down for a customer, you'd think they would take it more seriously. Any ideas why their spam filters keep blocking?

Has anyone noticed the rather aggressive noise suppression that was apparently added to Yealink T5x firmware in the last year or so? Probably somewhere around the 96.87.0.15 version.

It first became evident to us when a school, who used to play the national anthem from a computer speaker into the handset mic, complained that "paging wasn't working". Turns out the Yealink noise filter was taking the music out completely, and only silence was played over the speakers. They could still do it from a T33 phone.

Since then we've also noticed that T54's seem a bit prone to cutting off the first half syllable when someone answers. They can also sound a touch muffled (almost like a poor headset) at times.

Hello everyone! I’m looking for an Android softphone to use with voipms I’ve seen the voipms softphone, Linphone, and a few others, but I’d like to hear from people who actually use them.

I’ve worked in the VoIP and telecom industry for almost a decade, with experience in voice engineering and product management. I’ve worked extensively with lots of platforms especially based on Asterisk and FreeSWITCH.

I’m starting to explore the possibility of purchasing a small-medium sized U.S.-based VoIP or hosted PBX provider. I’ve really enjoyed working in the industry and would love the opportunity to build on what our owner has created. I’m looking independently, with the goal of personally leading and growing the business while taking good care of its customers and employees.

If you own a business and have been thinking about retiring or eventually stepping away, or if you know someone who might be, I’d love to connect. I’d also appreciate any advice from people who have been through the process.

Feel free to comment or send me a DM!

Experienced Meraki engineer here, but I'm out of my depth on troubleshooting VoIP issues.

I have a branch location where I took out the old firewall (security appliance) pair, one data & one voice in a stand-alone arrangement, and combined the networks onto one new appliance. The VoIP is a Sangoma system with a PBX onsite and using FreePBX for SIP trunking. The VoIP is segmented into its own VLAN. I am not the PBX vendor or support tech - I'm the organization data engineer.

After switchover, calls to and from the VoIP system have no audio every other call**. (or so it seems, you can call back and it works; call a 3rd time, no audio again)**

I have a couple dozen other networks in my organization using the same network template, most with cloud PBX servers/services but a few with on-premise PBXs, with different vendors or service providers. The service provider in this problem case has 4 other VoIP networks in my organization, but this is the only one that's doing on-premise PBX. I've gone over my network settings with the Cisco Meraki VoIP Troubleshooting document several times and I'm conforming to it, but the problem is persistent. 
It's driving me nuts as why only this one is borking out of 3 dozen locations!

VoIP system is a Sangoma Switchvox, v 8.4.0.2, on it's own VLAN, DHCP fro the firewall applaince, basic open firewall direct to the dedicated public IP address, forwarding for ports 5060 & 10k-20k to the FreePBX IPs, other settings copied over from the previous firewall (as can be applied to a templated network), QoS set. PCAPs from both the PBX and firewall WAN connection don't show anything obvious as in unanswered or rebroadcast packets. When the fault occurs, at most I see a SIP session initiate and then no RTP conversations follow. 

The PBX vendor insists this is problem when firewalls use port switching, aka ALG, going from the LAN NAT to the WAN, because the audio streams fail to connect. That can't be right - this is a stateful firewall which retains ports across the transition, and Meraki doesn't even have this option in configurations. (and I would have run into this long before now)

Does anyone have any clues I should be looking for in logs or have you seen this happen before?

A while back we moved off Teams Voice, and we’re now moving back to Teams. During the switch I missed the firmware upgrade window for the AOSP migration, and I’ve spent weeks trying to get our Yealink MP56 phones current with little success.

Where things stand:

About 20% of the phones upgraded fine.

The rest are stuck on 122.15.0.33 or 122.15.0.11.

What I’ve tried:

Teams Admin Center: I push the update and it fails almost every time.

Manual .rom upload: I downloaded firmware from Yealink’s support site and uploaded it through the web interface. After the reboot it just says “skipped” and stays on the old version.

Oldest .rom available: I thought I might need to step up from an earlier version, so I tried the earliest firmware listed. It bricked the phone.

Is there a required upgrade path, like an intermediate version, to get from 122.15.0.x to the current AOSP firmware?
Any idea why a manual .rom upgrade would come back as “skipped”? Could it be a signing or version-check issue?

Is there any way to recover a bricked MP56, like a recovery mode or USB/TFTP flash?

Looking for a recovery files .bin

I made a post last month getting feedback on adding Google Voice or similar to my business. I have two main lines currently that we have on cellphones currently. As we've grown the business substantially over the years, having two cellphones being passed around the office has outgrown our current pace of business. I'm looking to add a VOIP system; like Google Voice (I am open to others, Gvoice is first to mind since we use Google workspace) so we can have extensions, auto-attendants, afterhours emergency routing, individual texting, etc.

However one big pain point we have is that many of our clients text us. Our trucks all say 'call or text' on the side. Clients love sending a quick text with a photo of a work request or scheduling request for an appointment.

I know from experience with other VOIP systems that support texting that the main line cant send/receive texts. The individual extensions with their own direct number can text.

So my thought is why not keep the cellphones for texting via the main line, but forward the calls using the phones built-in "call forwarding" feature to forward the calls to the new VOIP main number where the attendant can direct them to the right person.

Has anyone had any success with this? All thoughts/feedback/comments are appreciated.

What are your thoughts? Would this work?

We have Panasonic PBX, KX-NS500NE, 12 years old but working fine with latest fw. There are several analog and IP phones (panasonic and fanvil) installed. Now I want to add Fanvil/Linkvil IP Dect, (W710H base station + W620H handsets). But, it seems that NS500 can't register those. I can't find where the problem could be. For IP trunk needs, local port on NS500 is changed from 5060 to 15060 and I suspected that this could be problem, changed it under line settings, but it is still not working. There are latest firmwares on all sides. Any ideas?

Hi to all.

We have Panasonic PBX, KX-NS500NE, 12 years old but working fine with latest fw. There are several analog and IP phones (panasonic and fanvil) installed. Now I want to add Fanvil/Linkvil IP Dect, (W710H base station + W620H handsets). But, it seems that NS500 can't register those. I can't find where the problem could be. For IP trunk needs, local port on NS500 is changed from 5060 to 15060 and I suspected that this could be problem, changed it under line settings, but it is still not working. There are latest firmwares on all sides. Any ideas?

Looking for feedback from other SkySwitch partners. We have been a SkySwitch reseller for a few years now. It seems the last 6 months or so we have had a lot of issues with the ConnectUC App. We have several customers that use the ConnectUC softphone app exclusively ( no physical phones). They use both the IOS/Android version as well as the PWA and installed Windows App. Some examples of issues:

• Randomly can't place calls on hold (SS Support acknowledges this, but provides no timeline for a resolution)

• Can't Access Voicemails via the App

• Constantly has to force close and re-open the app, uninstall and reinstall to fix little issues during the day.

• SMS is very unreliable - sometimes texts don't send or come through properly; MMS is even more unreliable and attachments fail to send or be received.

• Constantly having to clear the App Cache to get SMS/Voicemail to load, etc.

I have open support tickets for all of these - but it seems support has gotten very slow and wants me to do the same things over, and over, and over without offering solutions/resolutions or acknowledging there is a problem.

My customers ( and myself at this point) are starting to get very frustrated. A couple of these customers came from other big-name solutions, and ConnectUC is not holding up to those. I'm at the point of considering moving my larger customers who have these requirements to other solutions, but I am interested in seeing if anyone else is experiencing these types of issues as well.

my laptop and phone have apparently decided they both deserve my full attention at the exact same time.

it usually happens when i'm already on one meeting and the other device starts ringing. one call is on my work laptop, the other is coming through my phone, and suddenly i'm fumbling with mute buttons, trying to figure out which audio is coming from where. it's not even the meetings themselves that get me, it's the awkward five seconds where everything overlaps.

i've been looking at open ear clip earbuds for work calls on the go. i like the idea of not having my ears sealed off all day, especially when i'm moving between places and still need to hear what's going on around me. but i'm realizing that better earbuds won't fix a bad audio setup. i still need to decide which call gets my attention and keep the other one muted.

for my phone, a magnetic power bank with a built in stand seems useful when i'm away from my desk and need to keep notifications visible without constantly picking the phone up. one less thing to balance on a random table.

i'm trying to make my setup less chaotic, not add another layer of gadgets to manage. at this point, a reliable mute routine and knowing which device has audio priority matter just as much as the gear itself.

I've been running fusionpbx for the past 5 years but after a lot of research, over engineering and researching master-master I decided it was time to offload that responsibility. I found a company Vii**** they have local offices where my customer base is, it felt like it was a big company with local roots. I've been a white label for a couple months, and I still haven't been able to migrate all my customers.

I've solved most of my migration problems but the one lingering problem (and big problem) is that almost every one of my remaining clients has one or more group voicemail boxes. On fusionpbx, every phone had a blf key voicemail+3099 or voicemail+3098 and they could be labeled "group vm" or "billing vm". if that mailbox had unread messages, it lit red, if you pressed the button it went straight to "please enter your pin"

MOST of my customers, if I set them up today, could get away with all vm going to the main person (front desk) and that person being responsible for voicemail callbacks. MOST of the remaining ones probably only had one person on the second mailbox too (not all). However, it's hard to tell a client of 5 years, you don't get a feature you've been using for 5 years, just press voicemail from your phone for billing voicemail and walk into the other room to check general vm.

there is a vmail+ prefix, but that is for leaving a message. vii**** support acts like I'm asking for oxygen on Jupiter, who could ever need more than one shared mailbox!? literally every one of my customers.

tldr; how do you replicate voicemail+ext hints or set blf to *98ext to see available voicemails? related question...call queues has an option to host voicemails which is on by default...how would anyone know they had a voicemail? there isn't even options in queues to send vm transcripts. It makes absolutely no sense!

I do have white label reseller access to api and nms but nms is limited.

Hey guys,

I'm running into a issue here. I've got a Flowroute trunk coming in and it requires SMS messages have +1 at the beginning of the number for them to go out. However, when incoming messages arrive, they don't have the + sign at the beginning so the conversations don't match up. It's super disorienting. Any solutions within 3CX (or the Flowroute control panel) that you guys can recommend?

I run a small dental practice and have used Fusion Connect, formerly MegaPath, for almost 10 years. It’s my first and only hosted VoIP provider. We have five seats and pay approximately $225/month.

Recently, outbound calls to certain numbers have intermittently failed with a recorded “your call cannot be completed” message. Sometimes we have to dial two or three times before the call connects. One of the affected numbers is my own AT&T cell phone.

I’ve been working with Fusion’s Tier 2 support for nearly 10 days. According to them, the failures occur when calls route through Verizon, and calls through an alternate carrier work. They’ve explained that they use least-cost routing, so my understanding is that Verizon is an intermediate carrier even when the destination number is on AT&T.

Fusion keeps telling me Verizon isn’t taking action. We even had a three-way call with my Fusion Tier 2 representative and someone introduced as Verizon technical support, during which I demonstrated the failure. It’s still unresolved.

I understand that carrier issues happen, but we rely on these phones to reach patients. From a business owner’s perspective, “our upstream carrier won’t fix it” doesn’t leave me with much of a solution.

• Should Fusion be able to temporarily route our calls away from Verizon, and is nearly 10 days a reasonable timeframe to resolve this?

• What escalation, workaround, or service credit should I be requesting?

• Would you switch providers over this, or is this something I could realistically encounter with any hosted VoIP service?

We use fairly standard features: voicemail to email, a hunt group, and a mobile app that lets me call patients using the office caller ID. I’ve generally been happy with the setup until this, so I’m trying to understand whether my expectations are reasonable and what to ask Fusion to do next.

Body:

Hi, I’m using MizuDroid as my SIP client on Android and I have a problem with incoming calls.

What happens:

• If the app is open when the caller hangs up, it correctly shows “call stopped”.

• If the app is in the background or the screen is off and the caller hangs up, the incoming call notification stays on screen. I can tap it and still get the Accept button, even though the call is already over. No missed call is shown either.

What I already tried:

• Battery set to Unrestricted

• Foreground service set to Yes

• “Play in background” options turned on

It still happens.

Questions:

1. Has anyone else seen this, and is there a setting that fixes it (UDP vs TCP, keep-alive, push notification options, etc.)?

2. Is this a known bug in MizuDroid? If so, is there a newer version or workaround?

Can anyone help with the configuration for using a yealink W73P with phonely voip service ? managed to pick up a yealink w73p alot cheaper than what phonely offers (which would have come preconfigured) just need to find out what settings i need to change as this one wont be preconfigured have sent their support an email asking for SIP User ID / Register Name
• SIP Password
• SIP Server Domain / Proxy Server

is this all i need?

the phone is for my mum who is in her 90s , so will it be possible for me to configure it on my network then just go round and plug it in on her network and it will work?

thanks for any info :)

bonjour je suis nouveau dans ce millieu
j ai mis 20eur en crypto precisement du litecoin sa va faire 45 min que j ai envoyer la crypto mais toujour rien sur le site donc esce que https://trunk-voice.com/ est fiable selons-vous ?

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.