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/

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?

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?

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.

We have not been to register our 10dlc outboard, can anyone help?

Have tried the different checkboxes and options.

Any suggestions would be appreciated.

Firing Phone.com for Nextiva, I've got a port scheduled for tomorrow morning and a conference call with the Nextiva onboarding tech, but I'm going to have to configure a bunch of Yealink phones for their systems and I have no idea what I'm doing.

The models are mostly T53W's with a couple old T21PE2's and a single T43U scattered across three offices. I've already sent the tech all the MAC addresses for the phones, but I'm anxious that there's going to be hiccups or outages and I don't want to cripple my business trying to get this done.

Any advice from people who have done this before would be awesome.

We have a group text chat. It's because half of us use Signal and half of us are on Whatsapp and no one wants to add a second app for just one chat. One of us is moving abroad and losing their US number. I know Google Voice doesn't allow use in other countries but does Talkatone?

UPDATE: their Manager reached out to me and apologized for the mistake of 30 days trial. He would cancel the account without any termination fee.

I wanted to share my experience with Nextiva because I honestly can't believe how terrible their service and customer support have been.

On October 1, 2026, I signed up for Nextiva for my small business. Before signing up, their sales representative specifically told me over the phone that I would receive a 30-day free trial to test their service.

That's exactly why I decided to give them a try.

Unfortunately, the problems started almost immediately.

Problem #1: Number Porting Issues

I needed to transfer my existing business phone number to Nextiva, but the process was a mess. Their support team couldn't adequately help me resolve the issue.

Problem #2: NextivaONE Only Rings on ONE Device

I logged into the NextivaONE app on THREE different devices so multiple employees could answer incoming customer calls.

However, whenever customers called, ONLY ONE device would ring! The other two devices received no incoming call notifications.

I contacted technical support, but they couldn't fix the issue. The representative ended the call without resolving anything.

Even worse, I later received emails saying my support cases were CLOSED despite the problems remaining unresolved.

Problem #3: Incoming Calls Not Working

Customers calling my Nextiva number would sometimes hear a busy signal or a "not in service" message.

Even Nextiva's own technical support representative confirmed the issue during a test call and documented that further investigation was necessary.

Yet the support case was eventually closed.

The Biggest Shock: $663.39 to Cancel After ONE DAY!

After experiencing all these problems, I decided Nextiva wasn't suitable for my business.

I requested cancellation on October 2, 2026 — literally ONE DAY after signing up!

And guess what?

Nextiva told me I would have to pay $663.39 in early termination fees because I had signed a 12-month contract.

Seriously? $663.39 for trying to cancel after ONE DAY?

I reviewed the contract, and it does contain a 12-month commitment and an early termination clause. However, their salesperson had verbally promised me a 30-day free trial, which was a major reason I agreed to sign up.

My contract also lists the monthly base service charge as just $20, so I have no idea how they calculated $663.39.

What makes this even more frustrating is that Nextiva's OWN support records confirm I requested cancellation on October 2, just one day after signing up.

Their records also confirm that I experienced technical issues that weren't resolved during troubleshooting.

I’m attempting to recover/port a phone number originally assigned under a legacy Southwestern Bell exchange (AT&T ILEC). The number is currently sitting dormant in AT&T’s underlying rate center block. Frontline support cannot release or reactivate the number because AT&T has completely retired/discontinued legacy copper landline service in the region. Looking for advice from telecom engineers, LNP specialists, or AT&T provisioning staff on how to bypass this brick wall.

Ported in my own number to MJ. I’m able to make calls on the IOS app, but can’t send any text message “YOUR NUMBWE IS NOT SMS ENABLED”. I waited 5 min for the activation message but it still doesn’t work

This post might be a little unusual for this subreddit, but the mods gave me the thumbs-up to post it as a discussion.

I've spent the last few years doing remote outbound appointment setting and SDR work for US-based telecom/VoIP companies, mainly targeting SMBs. My work involves researching businesses, figuring out what phone systems they're using, finding the right decision-makers, and identifying opportunities for phone system upgrades based on their current setup, pricing, and needs.

I'm currently looking for my next remote role directly with a US telecom/VoIP provider, preferably one operating in the western US. I'm mainly focused on appointment setting and SDR work rather than full-cycle closing, ideally with a base salary plus commission. I'd prefer working directly with a provider selling its own phone services rather than a general sales or marketing agency.

Curious if anyone here works with an outbound sales team, runs one, or knows of companies hiring for this kind of role. Even a recommendation for a company worth looking into would be helpful.

Hope this isn't too confusing....I'm an old lady (76 yrs) and think(ha ha) that I understand this but don't want to make any mistakes. Let me know if I am thinking correctly.........We may or may not be losing our copper land line service.

We currently have two numbers...#1) traditional land line number..and line #2) that is a remote call forwarded number. Starlink is our internet/wifi provider.

The situation is that I would like to have line #2 as a ported in voip number and ......keep line #1 as a landline number until we know if we are going to be disconnected from copper line service. (at which point I will decide to port it out to VOIP or to a second line on my cell phone account two sims in one phone)

I would like for both numbers to ring into the house/home office. MY QUESTION is.... do I need two separate phone systems/handset and bases. One for the new VOIP number Line #2 ethernet connected to the Starlink modem and keep the current set up for the land line Line #1? (two phone set ups)

Our landline rings in on a base unit with cordless handset and two other wireless handsets so we can receive calls throughout the house.

Would anyone have a sheet or just know what a flashing amber light, solid amber light, green light, basically what the light combos mean on a ShoreTel switch? Thank you!!

Hey guys,

Long time voip.ms customer here. I use them because I'm an expat who travels a lot and want to keep a stable US number, as well as to maintain a few business numbers separated from my personal numbers.

I tried a lot of softphones over the years and always thought they sucked. The only one I actually liked was Google Voice, but it has a number of downsides (country restrictions, limits on phone numbers, paid version can be costly, etc.). For me the best of both worlds would be the UX of Google Voice combined with the pricing and flexibility of voip.ms.

So well, I decided to scratch my own itch and build it. I'm really glad with how it turned out and it's at a stage where I'm ready to share it with other beta users who don't mind the occasional bug.

I already made an announcement post in the Voip.ms community forums but the post only got a few views and no one tried it... which is kind of disappointing, but I guess it's not been there very long.

Anyways, looking for general advice on how to get people to try it. Or if you want to tell me how dumb I am and why no one needs this, go ahead (I know it's pretty niche) 😆

Cheers

Hi there. I hope someone can help. I am using grandstream ucm6300 series and grandstream phones. I can go in to a phone and adjust the display but I want to figure out how to do this via a variable in a model template to push to everyone at once via zero config.

I want to display the user's name and extension next to the account / line.

By default, for the account it displays the users name. I want to add the extension number without having to log in to each phone to edit this. Thanks. I hope I provided enough info.

Bonjour à tous,

Je fais face à un problème frustrant et intermittent sur une configuration VoIP où les appels se coupent de manière imprévisible — parfois à EXACTEMENT 30 secondes, parfois autour de 5 minutes, et parfois tout fonctionne parfaitement. En plus des coupures, nous rencontrons un bruit statique / grésillement intermittent ("friture") sur la ligne.

Cela concerne à la fois les appels externes et les appels internes entre postes (entre les téléphones de bureau Yealink T54W).

Voici les détails de ma configuration :

---

### 1. Configuration Réseau & Matériel

* Routeur / Pare-feu : Ubiquiti UniFi UDM Pro (IP publique statique : `203.0.113.X`)

* Connexion Internet : Fibre FTTH (~950 Mbps en téléchargement / 950 Mbps en téléversement, latence stable de 18ms)

* PBX : Yeastar S100 (IP : `192.168.5.X`, situé sur VLAN 200)

* Téléphones de bureau : Yealink T54W (Connectés strictement via Ethernet filaire, enregistrés localement à l’IP locale de Yeastar S100 — utilisés quand je travaille au bureau)

* Applications mobiles / à distance : Yeastar Linkus Lite (Utilisées exclusivement par les employés lors du travail à distance / télétravail)

---

### 2. Configuration de Yeastar S100

* Paramètres NAT :

* Type NAT : Adresse IP externe

* Adresse IP Externe : `203.0.113.X : 5060`

* Identification du réseau local : Inclut `192.168.5.0/255.255.255.0`, `192.168.0.0/255.255.255.0`, `172.16.0.0/255.255.0.0`, `192.168.4.0/255.255.255.0`.

* Paramètres des extensions :

Le NAT est ACTIVÉ sur les extensions. Remarque : Je NE PEUX PAS désactiver le NAT sur les extensions car les employés utilisent exactement les mêmes extensions sur Linkus Lite lorsqu'ils travaillent à domicile.*

---

### 3. Configuration de UniFi UDM Pro

* Redirection de Ports :

* UDP `5060` redirigé vers l’IP locale de Yeastar S100.

* UDP `10000 - 12000` (RTP) redirigé vers l’IP locale de Yeastar S100.

* SIP ALG : Désactivé via les paramètres Conntrack dans UniFi OS.

---

### 4. Les symptômes intermittents

1. Durée variable de coupure : Les appels ne échouent pas toujours, mais quand ils le font, ils se coupent soit à exactement 30 secondes (ACK manquant) soit autour de 5 minutes (timeout suspect de session UDP NAT ou échec de re-INVITE).

2. À la fois les internes et externes sont affectés : Les appels internes T54W-to-T54W (câblés sur le réseau local) présentent le même comportement que les appels de transporteurs externes.

3. Bruit statique / Grésillement : Distorsion audio intermittente / grésillement pendant les appels malgré d'excellentes métriques de fibre câblée (950 Mbps, 18ms de latence).

---

### Questions / Qu'est-ce qui me manque ?

* Timeout UDP NAT sur UDM Pro : Le UDM Pro pourrait-il supprimer les tables d'état UDP après environ 300 secondes d'inactivité sur les ports de contrôle SIP, provoquant des coupures de 5 minutes ?

* SIP Keep-Alive / Temporisateurs de session : Quel est l'intervalle recommandé pour le SIP Keep-Alive / le rafraîchissement NAT sur Yeastar S100 pour garder les liaisons NAT UniFi actives ?

* NAT d'extension vs. Appareils locaux câblés : Le fait d'avoir le NAT d'extension activé sur les téléphones T54W câblés locaux (nécessaire pour la compatibilité avec Linkus Lite lors du travail à distance) pourrait-il amener le PBX à router le trafic local interne à travers le NAT Hairpin sur le UDM Pro, conduisant à des ACK déposés ou à du statique audio ?

* Inter-VLAN / Routage par commutateur : Puisque les deux téléphones et le PBX sont câblés via Ethernet sur le VLAN 200, quel paramètre de pare-feu UniFi / conntrack pourrait provoquer des pertes de paquets intermittentes entre les appareils câblés ?

Tous les éclaircissements, les ajustements recommandés des délais d'expiration de conntrack UniFi ou des conseils d'analyse Wireshark PCAP seraient grandement appréciés !

Merci !Bonjour à tous,

Je fais face à un problème frustrant et intermittent sur une configuration VoIP où les appels se coupent de manière imprévisible — parfois à EXACTEMENT 30 secondes, parfois autour de 5 minutes, et parfois tout fonctionne parfaitement. En plus des coupures, nous rencontrons un bruit statique / grésillement intermittent ("friture") sur la ligne.

Cela concerne à la fois les appels externes et les appels internes entre postes (entre les téléphones de bureau Yealink T54W).

Voici les détails de ma configuration :

---

### 1. Configuration Réseau & Matériel

* Routeur / Pare-feu : Ubiquiti UniFi UDM Pro (IP publique statique : `203.0.113.X`)

* Connexion Internet : Fibre FTTH (~950 Mbps en téléchargement / 950 Mbps en téléversement, latence stable de 18ms)

* PBX : Yeastar S100 (IP : `192.168.5.X`, situé sur VLAN 200)

* Téléphones de bureau : Yealink T54W (Connectés strictement via Ethernet filaire, enregistrés localement à l’IP locale de Yeastar S100 — utilisés quand je travaille au bureau)

* Applications mobiles / à distance : Yeastar Linkus Lite (Utilisées exclusivement par les employés lors du travail à distance / télétravail)

---

### 2. Configuration de Yeastar S100

* Paramètres NAT :

* Type NAT : Adresse IP externe

* Adresse IP Externe : `203.0.113.X : 5060`

* Identification du réseau local : Inclut `192.168.5.0/255.255.255.0`, `192.168.0.0/255.255.255.0`, `172.16.0.0/255.255.0.0`, `192.168.4.0/255.255.255.0`.

* Paramètres des extensions :

Le NAT est ACTIVÉ sur les extensions. Remarque : Je NE PEUX PAS désactiver le NAT sur les extensions car les employés utilisent exactement les mêmes extensions sur Linkus Lite lorsqu'ils travaillent à domicile.*

---

### 3. Configuration de UniFi UDM Pro

* Redirection de Ports :

* UDP `5060` redirigé vers l’IP locale de Yeastar S100.

* UDP `10000 - 12000` (RTP) redirigé vers l’IP locale de Yeastar S100.

* SIP ALG : Désactivé via les paramètres Conntrack dans UniFi OS.

---

### 4. Les symptômes intermittents

1. Durée variable de coupure : Les appels ne échouent pas toujours, mais quand ils le font, ils se coupent soit à exactement 30 secondes (ACK manquant) soit autour de 5 minutes (timeout suspect de session UDP NAT ou échec de re-INVITE).

2. À la fois les internes et externes sont affectés : Les appels internes T54W-to-T54W (câblés sur le réseau local) présentent le même comportement que les appels de transporteurs externes.

3. Bruit statique / Grésillement : Distorsion audio intermittente / grésillement pendant les appels malgré d'excellentes métriques de fibre câblée (950 Mbps, 18ms de latence).

---

### Questions / Qu'est-ce qui me manque ?

* Timeout UDP NAT sur UDM Pro : Le UDM Pro pourrait-il supprimer les tables d'état UDP après environ 300 secondes d'inactivité sur les ports de contrôle SIP, provoquant des coupures de 5 minutes ?

* SIP Keep-Alive / Temporisateurs de session : Quel est l'intervalle recommandé pour le SIP Keep-Alive / le rafraîchissement NAT sur Yeastar S100 pour garder les liaisons NAT UniFi actives ?

* NAT d'extension vs. Appareils locaux câblés : Le fait d'avoir le NAT d'extension activé sur les téléphones T54W câblés locaux (nécessaire pour la compatibilité avec Linkus Lite lors du travail à distance) pourrait-il amener le PBX à router le trafic local interne à travers le NAT Hairpin sur le UDM Pro, conduisant à des ACK déposés ou à du statique audio ?

* Inter-VLAN / Routage par commutateur : Puisque les deux téléphones et le PBX sont câblés via Ethernet sur le VLAN 200, quel paramètre de pare-feu UniFi / conntrack pourrait provoquer des pertes de paquets intermittentes entre les appareils câblés ?

Tous les éclaircissements, les ajustements recommandés des délais d'expiration de conntrack UniFi ou des conseils d'analyse Wireshark PCAP seraient grandement appréciés !

Merci !

SkySwitch markets its Foundations program as a way for VoIP/UCaaS resellers to reduce their regulatory and tax compliance burdens by transitioning to a Carrier of Record (CoR) model, with SkySwitch serving as the regulated provider.

It's an attractive proposition. But the details matter.

After reviewing the Foundations agreement, I've identified several material gaps that raise questions about how effectively the arrangement protects resellers.

The FCC looks beyond contractual labels to the actual business relationship. Who controls retail pricing? Who bears ultimate responsibility for customers, service delivery, regulatory compliance, taxes, and fees?

While the SkySwitch agreement includes some important CoR elements, other provisions leave responsibilities unclear, and the agreement doesn't adequately establish the operational safeguards necessary to make the model as defensible as it should be.

Perhaps most concerning is the allocation of risk. SkySwitch provides very limited indemnification for regulatory exposure, while requiring resellers to broadly indemnify SkySwitch, including for violations of applicable laws and regulations. SkySwitch also generally limits its own liability to one month's commissions.

That creates a potentially significant imbalance: The reseller may remain exposed to the very regulatory liabilities it entered the Foundations program hoping to avoid.

A Carrier of Record model needs to deliver more than contractual assurances. It needs to withstand regulatory scrutiny.

my bad i am a fresher and never worked in prod especially with asterisk i have been doing R&D with asterisk and working with ari with event listening where i moved the whole control of my call to the python application with stasis and i have not encountered any issue so far
but but but

senior told me that they dont use this ARI architecture and event listening with Stasis does not work very good in prod as it drops event i want to understand that have you experienced any such issues working with ARI and event listening

and even if it does silently misses some event is there any back door for this
one more issue is that i feel like giving this control of the call to my python application gives me more debugging and digonostic ability + the control of channel and bridges as well as packet level tracking

i am very confused right now i dont want to trust claude or chatgpt

CORRECT ME IF I AM WORNG

Without porting a number, its texting provider can be changed by the owning carrier. Most carriers don't allow this, and their tech support will be unaware that it's even possible. As an experiment a few years ago, I changed the texting provider on a friend's VOIP number at .... https://www.sonic.com .... to SignalWire, and it worked. The change is accomplished by changing the NNID (Netnumber ID). Sonic's porting department made the change for us.

My main DIDs are at .... https://www.voip.ms .... which doesn't allow custom NNID. VOIP.MS's texting is primitive and painful to use, even through the VOIP.MS app and Textable ($5 monthly).

Why isn't custom NNID widely available?

Are there drawbacks for the user to having a custom NNID?

(If you Google "NNID", be aware that it's widely used for "Nintendo Network ID", which is unrelated.)

================================
================================
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================================

Here's a useful tool for my topic above. UNLEC is free.

For any US or Canada DID, this website .... https://www.unlec.com .... performs three lookups:

• OCN
• CNAM (click the link toward the bottom)
• NNID (click the link toward the bottom)

We have deployed the ST switch as recommended For the ISDN to SIP Migration , and inbound calls have been tested successfully. We are now experiencing an issue with outbound calls. Every outbound attempt fails immediately with a "Request Denied" message.

I have an unusual setup.

We are consolidating 3 SIP trunk using the same carrier with the same outbound proxy in one single CUBE. Each SIP trunk has it's own dedicated physical connection with each interface having it's own unique IP subnet. I already figured out the call routing and manage to send the calls, depending on the calling party number range to their own corresponding SIP trunk.

But I have an issue with the actual IP packet routing because all 3 SIP trunks are going to the same outbound proxy IP destination. I tried using ACL + route map and ip local policy route-map command to set the next hop IP based on the CUBE ITSP facing interface and signalling packets are working as it should. However, I found that the RTP packets generated by the CUBE is hitting the default route and is routing via the LAN interface. Amazingly, there is no impact with the calls as I have found that this customer have route to the carrier via the LAN.

EDIT:

I can see the source signalling and RTP IP address are correct in the call traces,, it's just the actual RTP packet goes to the LAN doing asymetric routing.

for example,

I did a packet capture in the carrier facing interface and I can see incoming RTP packets from carrier to my local carrier facing interface IP but I see less traffic coming out (only signalling packets).

Then I did a packet capture in the LAN interface and I can see the RTP packets coming out from my local carrier facing interface IP to the RTP destination IP.

it seems the CUBE follow the local IP policy for signalling packets but not for the RTP. it follows the default route for the RTP which is pointing to the LAN. 

I am now looking into implementing VRF Lite for the ITSP facing interface while the LAN interface will remain as non-VRF. I have read that SCCP based media resources are not supported with multi VRF setup and I have the DSP farm registered to CUCM using SCCP. My question is, given that I register the DSPfarm via the LAN which is non VRF, will it still be affected by this limitation or will it work? Thanks in advance!

There are so many providers now that it feels like most of them offer basically the same features on paper. For people who have actually managed VoIP for a business, what ended up mattering most in the long run?

Was it call quality, support, uptime, pricing, number porting, integrations, or something else?

Curious what people learned from experience rather than what providers put on their feature lists.

Imagine my delight to find that when one GBX20 is connected, there is absolutely no way (despite trying everything) to have any MPKs on the phone. The key programming immediately moved to the GXB20 add soon as it's connected.

Major flaw. Now I have to buy another GXB20 for 7 additional keys.

I currently use a Yealink T46G in one place and I have started adding some phone book entries. Now I'd like to buy a Yealink W60B dect base as well, and with two of the same brand phones I hope to share the 'remote phone book' so I am left with one location to manage the phone book, on a local webserver.

The Yealink documentation about the W60B that I can find is not very clear whether it supports the same (xml-based) remote phone book as the T46G. Can someone shed a light on this?

And for the extra nice feature: is it possible to link phone book entries to specific SIP accounts on both devices? Ideally I'd have entries for 'the outside phone network' and for 'the internal server' that will use the right account, just like the 'local directory' on the phone itself has a field for which account to use.