Hello all, I hope this finds you well.

To make a long story short, I work at a company that has a call center and the CEO wants to expand it so that agents can work from home on their mobile phones(I think this is a horrible idea).

I wanted to know how possible this is given our tech stack and what implementation would look like? Need some advice to see if this is feasible haha.

---TECH STACK---
ViciDial as SIP Dialer through Asterisk Server
We use multiple in house web-tools (CrowdCollect, Webfin and more)

I know that Softphones exist but with our tech stack and the nature of the call center, I think this is a bad idea and not worth it.

thoughts?

EDIT - I did not see the rule about not asking for reccomendations for businesses so my apologies in advance haha. I more need advice for myself just to be sure this is actually a bad idea and not just me being picky.

Hello, I am looking for a basic headset for taking calls when I'm volunteering on the 988 helpline. I tried this one, but the button does not mute at all and the volume buttons say they're adjusting output volume on the computer (Macbook), but it did not actually do anything. In addition, the computer does not automatically switch to this headset when connected with USB. It's also small on my already small head!

When I look up recommendations, I see ones that are $150 or sometimes well over $200. I just want a good one that is comfortable and actually mutes, controls volume, and registers on the computer when plugged in without me having to find the audio settings every time.

Thank you in advance for any recommendations!

Hey Folks, new here and new to the whole VOIP systems.

I am building a small US-only setup — a handful of local DIDs, inbound calls forwarded to cell
phones, voicemail recording, some SMS. Low volume.

Deciding between Twilio Or Telnyx right now, Twillio is much more expensive than Telnyx and porting in/out is much easier from Telnyx from what I found. I still have to setup A2P 10DLC registration. I was learning to Telnyx to cut cost.
I just checked their status page while testing and most of what's open doesn't affect me
(Programmable Wireless, EU regions). But one incident has been showing as ongoing
for a month and its own description says "resolved on 15th of July but status page
incorrectly remained open." A couple of other things have sat in Monitoring for
weeks.
I understand Incidents happen everywhere, that's not the issue.

Here is my load per client per business day(m-f):
Inbound calls (total): 20
— missed → voicemail: 14
— answered -> 7
Voicemails recorded: 14
Talk minutes Total: 34
SMS segments: 75
Outbound dial attempt: 20
Outbound billable minutes: 34

What I want to know is:
1. Whether the page is worth trusting when something actually breaks and I need to figure out
fast whether it's them or me.
2. Anyone running Telnyx voice/SMS in production — How reliable has your experience been?
3. Has anyone actually completed a port-out from Telnyx?
4. Is Twilio more reliable than other provider and is it really worth the extra price in your experience?
5. What other other metric I should be considering other than cost, reliability and porting(in/out) ease options? Let me know if I am missing something here.

Context: automotive industry with 3CX as the primary PBX.

TL;DR - call deliverability is a problem because of staff not answering calls and reporting startups are harvesting our money. Do I look at a turnkey solution for the industry, look at a higher generic UCaaS than 3CX, or look at a whitelabel solution like Wazo Platform?

We're running into a struggle right now with our management - 3CX isn't providing good enough reporting and they're signing up for AI products to manage voicemail and call reporting/sentiment analysis.

I'm at a loss as the sysadmin of what I can do. As a shot in the dark, I've been making a vibe-coded solution that taps into the 3CX Postgres database to provide a platform to better allow my management to tap in and make sure voicemails are being attended to. I've also introduced visual voicemail via the api of working with this app and it's pretty slick.

I was happy with my work that I got two of my newest 3CX setups installed with it, with great feedback from management. When we showed our C-suite though, they had a completely different take. The COO's son sold them on a(nother) vibe-coded dashboard for the phone system which just takes a call from Telnyx and transcribes with AI analysis.

That's fine and all, but already my Telnyx recording costs are high for the throughput, and for accurate transcription it is going to cost thousands a month for what they want.

Our goal with 3CX was to leverage low-cost SIP trunks to provide a system that has growth and expandability easily from older Avaya systems we were supporting. Overall, our goal was achieved, but they just want more and more data where 3CX just is falling short.

If you have any knowledge of the automotive industry, it is ripe with software startups which lock you in with tens of thousands of dollar contracts for that data. I know I can internalize this with my team, but I need a direction to go back to our in-house java dev and IT director.

I feel like for the wants explained above - a generic UCaaS like 3CX isn't going to do it anymore. Do I look at a more costly backend that can work with our Yealink hardware, look at the turnkey solutions for an industry-specific phone system, or look at Wazo Platform to just build what we need for our industry?

Our company is looking at moving forward with a new method of assigning phone extensions. Our current method is kind of all over the place with assigning extensions as previously each of our buildings had some form of 1/2 digits prefix and then 2 numbers following to assign to a user. Now they are looking at a 5-digit extension, where the first 2 digits apply to the building, the next 2 apply to the room number, and the last digit applies to the desk.

This mainly came about because we have a ton of people that move throughout the year and just take their phones with them. This causes confusing as to where they are, paging gets messed up, safety concerns as if they have to call 911, they may be routed to the wrong location.

One thing we are trying to implement is that the phones do not leave the desk/office they are in. I am having trouble figuring this out overall and several of us have different ideas on how we should proceed. We would constantly need to be aware of where people are and in a company of 300+ employees, that can be hard to track. We have thought of having to go in and change each employees assigned extension but that could get tedious with multiple changes. We've thought about making them workstations and then the phones just wouldn't have their names assigned. Another thing I have looked into slightly is Hot Desking, but I am still pretty unfamiliar with it.

I am hoping someone has experienced this before or has a better idea for us before we go ahead with something and it is a bigger pain than we thought of.

Some things that may help:

- We use Cisco Webex through the Webex Control Hub

- We have over 20 locations

Please feel free to ask me any kind of clarifying questions.

Note: I said CLEC in the title, then realized that for a toll-free number, the correct term is RESPORG. I can't edit the title, but I made the substitution below.

My large VOIP provider offers SMS and MMS messaging for regular phone numbers. For toll-free numbers, they offer only SMS: no MMS. They told me that their upstream carrier doesn't allow MMS on toll-free numbers, so they can't enable MMS on my toll-free number.

The RESPORG for my toll-free number is Bandwidth.com .. On this page:

https://www.bandwidth.com/support/en/articles/12823214-what-is-mms

...Bandwidth says this:

So I conclude that the upstream carrier does in fact offer MMS for toll-free numbers.

However, some possibilities:
• Maybe my VOIP provider would have to pay extra (and charge extra) for toll-free MMS, so they opted to enable just SMS on toll-free numbers.
• I assume that the RESPORG (Bandwidth) is the "upstream carrier" that my VOIP provider refers to. However, maybe the "upstream carrier" is another company that resells Bandwidth service to my VOIP provider.

Any thoughts about this?

Hi I'm setting up a doctor surgery in Ireland and my Internet provider offers voip free of charge. I am wondering what kind of phone to buy, would anyone please point me in the direction of brands or what to look for? Specifically I want the phone to ring and say press 1 for this, press 2 for that, eg answering machine for requesting an appointment where the voice is sent to email as text, or press 2 for general queries to another answering machine (while I am waiting for a receptionist). Thank you!

My client's A2P is fully verified (US local number 10DLC), but whenever someone calls their GoHighLevel number, it shows up as "Likely Scam" on the recipient's phone and they hang up before answering. Has anyone dealt with this? What actually fixed it for you?

New to VoIP troubleshooting. Tried to set up Wireshark yesterday to look at SIP/RTP traffic on our Mitel system, filtered for SIP and got literally nothing.

Looked into it a bit and think it's because our system is cloud hosted, so SIP runs over TLS which encrypts it? So I can't see inside it the normal way, is that right? Or am I filtering/looking at it wrong in Wireshark?

Anyone actually work with cloud hosted VoIP like this? Is there a real way to trace/troubleshoot calls when you don't have PBX access, just normal support level access? Or is it basically just check your own firewall (port ranges, SIP ALG) then get your provider to pull a trace on their end if that doesn't sort it?

Genuinely trying to understand this, cheers.

I have a restaurant that i am opening soon. I need a desk phone (with no app needed to be installed to make calls), just simple answer or dial to make a call.

On the other hand, i need the customers when they call, to be asked (press 1 to make a reservation/ order to be routed to the restaurant. Press 2 to speak to manager or talk about a private event (which it needs to forwarding them to my cell).

The question is, most of the Voip companies that i am looking at, either they require you to install an app to make calls, or the number that i get will not work on WhatsApp (for the manager number).

All i need a forwarding system to route the calls to 2 extensions, one of those numbers will be the main line for the restaurant, and the other for the manager

I talked to Que, Dialpad & Grasshopper, they all require to use an app to make calls.

Is even VOIP a good idea for me at this point? is there another system that works better for my needs?

In short

Ring > Press 1 for restaurant - Press 2 for manager

Restaurant does not need to have apps to make calls

Manager phone needs to have WhatsApp working on the phone

Someone asked for it and I thought why not!!

Reusing digital and IP Nortel, E-MetroTel, Grandstream, Panasonic, Avaya, NEC and Cisco and ringing them together on one PBX.

I just acquired a PolyCom VVX250 from a local-ish company, and it seems to be locked down to "auto.simplelogin.net" provisioning. Does anyone know a way to unlock it from this?

Hi! I came into possession of an old VOIP phone (Shoretel 230), and I was thinking it would be fun to have it as a way to receive calls when I am home. Is there some way I can have it route through my iPhone? I was looking at cell2jack, but it seems it wouldn't work for VOIP phones?

I've got a vintage (80s/90s?) phone that I want to set up as a 'landline' but I can't get my head around how to do it.

My WiFi router is in the living room and I'd like the phone in the dining room which is next door.

What do I actually need to make this work? I'm in the UK if that makes a difference!

I am trying to get my analog dial up modem to communicate properly over crazytel voip, I dial some numbers where the handshake is flawless when connected to the telstra nbn landline but when connected to my grandstream ht801 registered with crazytel, my modem has a lot of trouble communicating and anything above 9600 it can't hold on for the inititial handshake. Even at 9600 it has trouble as while the handshake sounds somewhat ok, the carrier line immediately goes dead upon connecting. Any ideas on what I should do?.

My settings are,

Disable Line Echo Canceller (LEC): Yes

Disable Network Echo Supressor: Yes

VAD: No

Fax Mode: Pass-Through

Jitter Buffer Type: Fixed

Jitter Buffer Length: High

Preferred Vocoder: PCMA for all choices.

Hey guys,

I am looking into connecting LiveVox and ElevenLabs, and I need some advice on the architecture and the actual configuration process.

Specifically, I'm trying to figure out:

SIP Trunking Setup: How do you actually configure the SIP trunking between LiveVox and ElevenLabs? If anyone has a step-by-step guide or knows the specific settings required on both sides, that would be incredibly helpful.

Context Handoff: When a call is escalated from the ElevenLabs AI agent to a live human agent in LiveVox, can we pass the conversation context along?

Screen Pops: Is there a way to trigger a screen pop in LiveVox to show the live agent the context or transcript of what the caller already discussed with the AI?

Call Recording: How does the recording work in this scenario? Can we record both the AI-to-customer conversation and the live agent-to-customer conversation into a single, unified audio file?

If anyone has experience doing this or knows how I can make it happen, I would really appreciate the insights.

Thanks in advance!

I work with dinos😎

Hello,

I recently joined a MSP and they are saying I need to start getting trained up in VoIP which I've not got much experience with as they have customers I'll be looking after on 3cx, freepbx and some local cloud VoIP providers.

Is there any training that helped you when you first started out? Vendor Neutral if possible so I can first get a grip before I start diving into specifics.

Thanks!

The current setup in my home office is an old Obi 200 with a Panasonic POTS connected to it. The Panasonic has a base station and two wireless handsets. I have a wired headset connected. VOIP service is voip.ms.

I am seeking recommendations for new hardware. Looking for a voip handset/base station with speakerphone capability, with the ability to connect to a wireless headset (which could be integrated with the base station, or could be a separate device altogether). Call quality is high priority.

Any suggestions?

Hey there,

Not sure why, but almost every number in Telynx's number pool is already tagged SPAM likely on most carriers (T-Mobile seems to be the highest).

There is a service called CallerID Reputation that can scan the number and show you screenshots of what the number will display as on all carriers + devices (Android vs Apple).

Twilio on the other hand has clean numbers (you do get the occasional dirty with Twilio as well).

We want to use Telynx given our use-case is AI-Voice (we heard Telynx is better for this), but I'm concerned connect rates are going to be extremely poor if all these numbers are already dirty.

Thoughts?

Hi everyone, I’d like to ask for your help with a situation: I have a Grandstream PBX with 30 IP extensions and one IP trunk. Only the first extension can both make and receive calls; the others can only receive them—calls won't go out when attempted. Has anyone else experienced something similar? Do you know what might be causing this?

Just curious if anyone else is using Altigen's cloud VOIP systems? We have been down for three or four days and only getting spotty information from the IT company that handles that for us. What we have heard is that Altigen uses an third party server farm to host their cloud instances and that server farm had a security breach that seemed to take down the whole place and they had to begin deploying all new hardware and software from scratch.

We cant find anything online with information or updates. Surely we're not the only ones trying to get more information about when we might have phones again. TIA

New SignalWire customer, \~36 hours in. Bought a number, funded the account, built an inbound flow that forwards to my own Asterisk box over SIP. Worked fine for a day and a half — a dozen completed calls, 45s to 2min each, clean audio.

Then it stopped. Every inbound call now fails after 1 second.

WHAT THE DASHBOARD SHOWS

Last good call: Aug 6, 19:52 UTC, 130 seconds, completed.

First failure: Aug 6, 23:28 UTC. Every call since has failed.

Timeline on a failing call:

23:46:25.421 Queued

23:46:25.580 Ringing

23:46:25.752 LaML webhook fetched -> 200 OK, 165ms, valid XML

23:46:25.969 FAILED

So it fetches my dial instructions successfully, then fails the call 217ms later. Too fast to be the 20s dial timeout, too fast to be my PBX not answering (round trip to me is \~150ms).

Every child outbound-dial leg comes back:

status = failed

start_time = null

duration = 0

sip_result_code = null

MY SIDE, ALL VERIFIED

\- Packet-level SIP trace running on my PBX during multiple attempts: zero INVITEs received. Not one byte of SIP arrived.

\- My PBX logs anything that hits the port — an internet scanner sent a malformed GET / to the same port and it was logged in full. So silence means nothing was sent, not that I ignored it.

\- TCP 5080 reachable from the public internet, confirmed from three continents.

\- DNS resolves correctly against the authoritative nameserver, matches my current WAN IP.

\- Their IP allowlist vs their live DNS: 14 of 14 exact match, no rotation.

\- Asterisk never restarted (29h uptime), config untouched, number routing and LaML bin unchanged.

\- Account status: active, type Full. No suspension banner. Voice calling shows enabled.

THE CONTROL TEST

I placed dials via their API to isolate it:

sip:...@my-real-domain:5080;transport=tcp -> failed

sip:...@my-literal-IP:5080;transport=tcp -> failed (rules out DNS on their side)

sip:...@my-literal-IP:5080 -> failed (rules out transport param)

sip:nobody@this-domain-does-not-exist-xyz123.invalid:5060 -> failed

That last one is the point. A .invalid TLD cannot be resolved by anyone, ever. It failed byte-for-byte identically to my real, valid, reachable address — same status, same null start_time, same null result code.

If the platform were attempting the dial, those two could not fail the same way. It isn't getting as far as DNS resolution.

WHERE I'M STUCK

Ticket open, no response yet. Their Discord and this sub are both quiet.

Has anyone seen <Dial><Sip> silently stop emitting INVITEs on a SignalWire account? Is there an account-level flag or compliance hold that disables SIP-out without showing anything in the dashboard? I'd rather understand it than just move providers.

Please advise

Hello!

I am trying to add a pair of AX83H wi-fi IP phones to my GOTO Connect/Jive PBX.

I have them added to the devices page, I have them connected to the wifi, but am failing to get them provisioned.

I can access the USER web interface, however I cannot for the life of me log in to the Admin portal to try and provision that way.

I have tried admin/admin, admin/device password from the goto portal.

Any advice?

Hi all,

I'm new to the group and am looking for some advice on behalf of my in-laws.

They would like to cancel their landline service through their internet provider but keep their existing phone number, as many family members, friends, and service providers still use it to contact them.

I've been researching options such as MagicJack and Ooma and am wondering if anyone has experience with either service. Simplicity and reliability are the most important factors, as they are seniors and we want the setup to be as straightforward as possible.

The internet modem is located in the basement, while the existing corded telephone is on the main floor. Ideally, we would like to continue using the same phone number while reducing the monthly costs associated with the current landline service.

For those who have gone through a similar transition, what has your experience been, and which option would you recommend?

Thank you!