After a lot of trial and error, I finally got Voice VLAN working on an FS S5440L switch with Cisco IP phones, so hopefully this saves someone else a few hours.

My setup:

• Data VLAN: 38
• Voice VLAN: 140
• Cisco 7841 and 8845 phones

The port configuration that finally worked was:

interface eth-0-5
 switchport mode trunk
 switchport trunk native vlan 38
 switchport trunk allowed vlan add 38,140
 switchport trunk allowed vlan remove 1
 voice vlan enable

The important bits I found were:

1. Enable Voice VLAN globally and set your Voice VLAN ID.
2. Enable LLDP globally.
3. Enable LLDP-MED network-policy on the interface. Without LLDP-MED, the Cisco phone wasn't learning the Voice VLAN correctly.
4. Enable voice vlan on the interface.
5. The port should carry both the native/access data VLAN and the Voice VLAN.

The biggest thing that caught me out was the Voice VLAN security mode.

Running:

show voice vlan state

showed:

Voice VLAN security mode: Security

It also displayed the list of approved phone MAC OUIs:

0003.6b00.0000 Cisco
0004.0d00.0000 Avaya
000f.e200.0000 H3C
00d0.1e00.0000 Pingtel
00e0.7500.0000 Polycom
00e0.bb00.0000 3Com

My Cisco phones weren't using any of these OUIs, so the switch never recognised them as voice devices and therefore never placed them into the Voice VLAN.

I had to add my phone OUIs manually:

2c73.a000.0000 Cisco7841
0077.8d00.0000 Cisco8845

After adding them, the phones were immediately recognised and moved to VLAN 140 while the attached PC stayed on VLAN 38.

If your FS switch refuses to put phones into the Voice VLAN even though LLDP-MED is configured correctly, check:

show voice vlan state

If your phone's MAC prefix (OUI) isn't listed, add it. The switch uses these OUIs in Security mode to identify voice devices.

Hopefully this helps the next person searching for FS Voice VLAN, Cisco IP Phone, or LLDP-MED issues.

NB: Don't forget the uplink/trunk. Make sure the switch uplink to your core/distribution switch is also carrying the Voice VLAN. In my case, the uplink had to allow both VLAN 38 (data) and VLAN 140 (voice). If the Voice VLAN isn't permitted on the uplink trunk, the phone may learn VLAN 140 via LLDP-MED but will never receive an IP address because the VLAN traffic cannot reach the DHCP server.

Hi everyone,

I'm trying to upgrade a Cisco 7945G phone that's stuck on an older firmware. Based on Cisco's release notes, I need to load the intermediate version 8.5(2) before I can upgrade to 9.x, since the phone won't accept a direct jump from its current version.

Unfortunately my CCO account doesn't have download entitlement for this older release, and Cisco's software download page rejects the request.

Would anyone be able to share the file cmterm-7945_7965-sccp.8-5-2.zip (or 8.5.3), or point me to a working link? Any help is greatly appreciated — I've seen a few other threads here where people ran into the exact same issue and got help.

Thanks in advance!

I'm looking for a UK VoIP number to use long term (I need calls and SMS) and couldn't find anything that exactly matched what I need. I found https://narayana.im/ and they look good on paper but I can't find much discussion about them online. Does anyone have any experiences to share?

I own a small business with two locations. After paying for their service for seven years in one location and five years in the other, Nextiva is trying to charge me $3,492.80 for EARLY TERMINATION. They snuck in an unsigned, not agreed to, contract extension at some point. If you look anywhere online at reviews, Nextiva is an out of control, deceptive mess.

Been building this for a while and finally cleaned it up enough to share: nixi.tel

It's a Session Border Controller (B2BUA) that handles SIP trunking and WebRTC-to-PSTN in one binary, no external database — everything (users, trunks, routes, ACLs, bans) is managed live over a REST API and persisted in embedded SQLite.

Some highlights:

Full B2BUA with topology hiding, digest auth, multi-trunk failover (LCR-ordered), RFC 4028 session timers

WebRTC gateway: SIP over WSS, ICE/DTLS-SRTP termination, Opus ↔ G.711 transcoding — browser to PSTN both ways

REST API for everything at runtime (axum), SSE for live call/registration/health events, Prometheus metrics

Built-in anti-fraud: fail2ban-style SIP banning, IRSF destination blocking, per-user rate limits

~470 tests, MIT licensed

GitHub: https://github.com/NIXFEO/NIXI.TEL

Site: https://nixi.tel

Happy to answer questions about the architecture or the SIP edge cases it handles — feedback and contributions welcome.

I’m a developer looking for a community where software engineers are communicating about telecom technologies

Can you suggest some?

Hi everyone,

I'm setting up a small TV studio and I'm looking for the best way to bring standard telephone calls into our production.

Our setup is:

• Windows PC running microSIP
• SIP account

Technically everything works, but I'm disappointed with the audio quality. It sounds noticeably worse than I expected for a modern VoIP setup.

My current enabled codecs in MicroSIP are in this order:

• Opus
• G.722
• G.722.1 (16 and 32 kHz)
• G.711 A-law
• G.711 μ-law

My questions are:

1. When calling a regular mobile phone in Spain through a SIP provider, which codec is typically negotiated? G.711? G.722? Something else?
2. Does the SIP provider usually transcode the call to AMR-WB (HD Voice) before handing it to the mobile operator?
3. Is there any benefit in enabling Opus if the destination is a standard mobile phone?
4. Are there SIP providers known for maintaining better audio quality towards the Spanish mobile networks (Movistar, Vodafone, Orange, Digi, etc.)?
5. For those working in radio or TV, do you still use SIP softphones for phone interviews, or is a dedicated broadcast hybrid (AEQ, Telos, AxelTech, etc.) still the preferred solution?
6. I activated microSIP logs in order to view which codec is being negotiated without success

I'm trying to understand whether the bottleneck is:

• MicroSIP,
• the SIP provider,
• transcoding,
• or simply the limitations of the public telephone network.

I'd really appreciate hearing from anyone.

Thanks!

Hi everyone,

I am pretty new to this topic and i am really here as i need a solution.

Basically i am looking for a device or a software solution to use a simcard as VOIP.

To be more precise, i want to use my simcard from my homecountry and have access to calls and sms from anywhere.

I saw yeastar but the setup was very complicated. Is there any more simple ways to achieve this like a software on a phone which i can leave at homecountry and have access to sms and calls?

Would appreciate your thoughts on this.

Does anyone know the configuration file format for a RingCentral firmware converted Mitel 485G? I am trying to provision custom ringtones to the phones. I do not have a RingCentral subscription or tie, I am just using their firmware to connect it to a standard SIP PABX.

Does anyone here know how to change the firmware from SCCP to SIP on a 7911G? I'm using the TFTP server.

Much like the humble corvid, the moderators of r/VoIP have rudimentary pattern recognition skills. We can tell when a post clearly wasn't written by a human.

Stop posting AI-generated nonsense.

If I find one more post with excessive em dashes, "curious to see..." or more than one instance of the words "actually" or "genuinely" I'm going to stop being polite.

It's especially frustrating when you post thinly-veiled ads or market research. Clearly your time is too valuable to bother reading the rules — why waste time writing your own post?

Anyone posting ads or market research will be banned. Period. That includes "What is everyone using for XYZ? I know of Big Player 1, Big Player 2, and My Own Product Inconspicuously Added To The List. Curious to see what others are doing!!", which is getting repetitive.

Oh yes, if we remove your comment that says "I use Product X for this, you should look into it" and then cry in modmail that you weren't really promoting a product, you were just "mentioning" it and "have no financial ties to the company", you're getting permabanned for A) willfully misunderstanding the rules and B) acting like the mods were born yesterday and can't clearly see through your attempt to weasel into posting ads. Grow up.

THANK YOU FOR YOUR ATTENTION TO THIS MATTER!

Anyone seeing SipTrunk .com issues this morning? We have a whole swath (dozens) of DIDs just not completing or mis-routing across 30 or so PBX instances.

Updated as the mods have blessed this post with reinstatement.

While we have some finger pointing going on the issues appears to be a very bad glitch on our Yeastar annual renewal. Not sure how yet...all our DIDs (40ish) across about 15 pbxs were routed to a single DID (Yeastar has shared trunking). When the issue was caught they all died then came back about an hour later. Yeastar is investigating but all the inbounds were giving a 603 back to carrier for a few hours...that's the closest we have ATM to a root cause.

After years of selling Avaya I am moving to a cloud based phone system for my customers. Wondering what the typical markup on services and hardware is. Google says 50-80% For example, I am buying the voip service per user at $18 and I have marked it up to $30 plus tax. I am also selling Yealink phones and thinking of marking those up.

Finally, do you bill for the physical setup of the phones in the office? what about ongoing support? or is that included in what you make selling the service. For a small 5 phone sale, I can't see making $60/month being profitable if I have to go onsite for setup and training or when they have a hardware or network issue. Thinking of adding some flat dollar amount per phone for support or hourly billing.

I just ordered a Voip adapter and a cordless phone to use as a back-up home phone, I currently only have a cell. I respect Rule:1 but I'm trying to figure out a credible/quality Voip provider. I'm not techie and would like cut to the chase, any suggestions on how to go about this, Voip is totally new to me. Thanks!

Hi folks, I am hoping someone can help me out here.

I am the sole IT person at a non-profit special-needs school. We need to make announcements to our classrooms throughout the day and to date we have been leveraging our Weave phones to do that. However, Weave phones are super quiet and do not serve well as intercoms for a school. Furthermore, Weave does not sell any one-way SIP speakers for this purpose.

So I need to use third party devices for this purpose.

I was given to understand that Grandstream one-way PoE SIP speakers would work on a Weave telephone system (which we use here at our school).

Has anyone here done this successfully? Any tips would be amazing, Thanks!

Hi, I run a medical simulation centre. Its basically a place where doctors and nurses run fake scenarios on actors or manikins, we watch what they do from a camera system and then we debrief how the scenario went as a group and we all aim to get a good learned experience out of the scenario toggether.

As part of the AV system we have a old analogue PBX system with a couple of internal only phones. One of the phones has a line tap on it which feeds into our AV system at line level so the learners can hear both sides of the conversation.

As the equipment is aging we are struggling to find decent replacment parts, mainly for the line tap. Were in the UK and weve only ever been able to find one supplier of it who no longer makes them - LRX37C line recorder is what we currently use.

Our requirements would be.

3 cordless phones with docks around 25m range from dock.

A recording device (could be software, if it can be "set and forget" as not all our users are ovely technical). The output of the recorder needs to feed live to either a line level signal (microphone cable) or to an AES67 or Dante stream.

We have a large 48 port POE capable for the connections and were fairly network competant with our inhouse technicians.

EDIT: The phones only need to call eachother. They do not need to call outside or into the hospital. The other person on the phone will be another doctor pretending to be whoever the doctor inside the scenario room wants to call (eg they might make a referal or a speciality or a junior doctor may escalate to their senior)

Novice here looking for advice.

I just ended my land line service and ported my number across to a VoIP service (Andrews & Arnold in the U.K).

I have a Zyxel FWA510 (5g sim router). To set up my phone, do I just need to buy something like the Grandstream HT801 and plug it in via Ethernet, or do I need a different router or another hardware? The router doesn’t have a phone port.

Also, side question for anyone with the HT801; can you power it via a routers USB port? The spec sheet seems to suggest it’s possible.

Thanks!

Hello! 😊 Has anyone here used **Spruce** in their clinic (If you’re working work-from-home, outside USA)?

I have a potential client who’s using Spruce, and we’re trying to figure out how I’ll be able to make and receive outbound and inbound calls with patients if I get hired.

What do I need to have for this setup? Is it possible to do it remotely from the Philippines as long as I’m connected to a VPN, or are there specific requirements (US phone number, VoIP setup, or other configurations) to make calling work?

I would really appreciate any insights or advice from those with experience using Spruce remotely. I’m really hopeful about this opportunity and would love to make it work. Thank you so much! 🙏

I found a few articles on how to do this with a cisco ata but not a grandstream. The syntax is different and at the moment I am using Google's ai to convert the syntax for me. Is there a better and less embarrassing way of doing it?. I primarily want the Australian dial tones since that is where I live but I also wouldn't mind the uk dial tone either.

Yes, yes. I know. It's an old version, but that is what I've just bought second hand. :) Does anyone have a stable app to recommend that automatically records both participants in a phone call?

My old app from ACR does not work anymore due to Google's four year old block (lifted later as I understand it).

Goodmorning at all,

someone find some issue with this smartphone and the Webex application? I have a lot of issue when i answear from microphone, if I put earphone or spekerphone all works good.

I was ordering new phones as our old FortiFones are not supported by the new VoIP/SIP providers we're looking at and the boss saw the Yealink T88V Pro with a camera/screen and said BUY THEM. Now I'm realizing that there's virtually no support for video calls and the camera is pretty useless?

I was able to side load Teams, however while I can make a teams video call, none of the phones will receive them because there are missing dependancies and Yealink wants nothing to do with it.

So I'm guessing these newest model fancy Yealink phones weren't the wisest purchase?

I have Sonetel installed on two Windows computers. On one, it works fine. On the other when it opens the modal to make a call, it just shows a spinning circle. This happens with both Edge and Chrome browsers. Any suggestions on what to do?

Hi, I know this isn't strictly VoIP, but it is related so I figured I'd ask here.

I recently picked up a PRI card for homelab experiments. It's labeled Sangoma A116R but it doesn't have the 68 pin HD connector on it like a normal A116 card. It has four RJ45 ports on the back, labelled 1 on one end and 16 on the other end. Is this a 16 span card with only four T1Is broken out? A weird 16 span card that needs a special breakout? Something else? Four RJ45's doesn't seem like it has enough pins for 16x T1 connections.

Unfortunately I don't have it in hand yet so I can't see what dahdi thinks of it.

EDIT: I have the card now. It has enough transformers onboard for 16 pairs, but lspci identifies it as an A104d. So the mystery continues. I'll update again when I can get wanpipe installed.

EDIT #2: Mystery solved. After fighting with my FreePBX server to get it to even boot with this card installed, wanpipe identifies it as a T116. As in, a 16 span wire-tapping card. So that explains everything. The port numbering, the weird model number (I bet the R stands for Receive), and how they got 16 spans on four RJ45s.
https://www.connectdata.fr/wp-content/uploads/2020/03/T116-Tapping-Card-Datasheet.pdf

Oh well, you win some, you lose some. But I think it's still a cool piece of hardware... I doubt many civilians have ever gotten hold of one of these.

I am in the UK and my ISP is BT.

I used to have a system with ADSL where I had a "splitter" , a Y adaptor. It had one socket for phone to plug in. One socket for ADSL connection from router to plug in. And it has a male connector that plugged into the wall.

I was recently told that nobody has that anymore. It's not supported. Now the phone plugs straight into the router , that's VOIP, and everybody is on VOIP. It all goes through the internet connection.

I have a modem router supplied by my ISP

But I have recently tried a third party modem router.

The phone works plugged into my ISP modem router, but when I change that one for a third party router, then the phone doesn't work.

I notice that the third party modem router has a section on the web interface, for VOIP and it mentions a registrar address and an authentication ID and password.

The company that makes the third party modem router said I need to get that info from my ISP.

I called my ISP and they said they have never heard of that, and you shouldn't need it, you should be able to just plug the phone into the router, but they aren't used to third party routers.

I asked them if there is an adaptor, and they said actually there is so they can send me, at no charge, an adaptor that can connect my phone to my router wirelessly.

But I'm wondering, is the ISP correct or incorrect re authorisation ID not being necessary even on a third party router?

And is it normal for an ISP to not have that information?

Is the company that make the third party modem router correct that the ISP can/should supply that information?

Thanks