VoIP

Hi, due to a very extensive project, we need to expose FreePBX to the internet. Specifically, we are concerned with the SIP and RTP ports. The purpose of this action is to allow logging into the system using softphones and configured phones without the need for VPN.

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In the past, I noticed that exposing port 5060 results in numerous brute force attacks where the attacker tries to impersonate an extension that exists in the system. However, due to the lack of a password, they are unable to make a phone call. Does an attacker, without knowledge of the extension password, have the ability to make calls at the expense of the client?

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Ports such as 443, 80, 22, etc., will not be exposed to the world, only the ports required for telephony.

Does anyone knows a good VOIP provider that has;

-SIP account option/integration-Good for Privacy & Security- Accepts crypto payments- Has rotating numbers (When you call it will show a number of a country that your calling from)

Thanks!

I am attempting to ready for a TTU. For one, I have questions on understanding the ATT document that shows the IP addresses and what they correlate to elsewhere. We currently have an Avaya IPO 500v2 connected to an AudiCodes Median800b SBC. We have two test phone numbers assigned with the IPflex order and are trying to set it up to be active with our current configuration. I have more questions and can go into greater detail if someone with experience is willing to assist. I would be so grateful!

Hi,

We are seeing issues with our MiCollab system wherein users seem to be presented with an error on both the Mobile App and the Desktop App in the middle of a call, "Sorry, softphone disconnected because the incoming audio stream was interrupted".

We are using Mitel 3330 with MiVoice Business.

There is no specific timeframe or length of call before they are presented with this.
We have turned off SIP ALG on the firewall.

Has anyone seen this before/have any recommendations on what we can check?

Thanks!

Hello everyone,

I am trying to set up in my lab Teams Direct Routing with Audiocodes SBC. I am using the Azure Marketplace image and have been running into issues. I have tried using the Audiocodes Config Wizard using Teams and a SIP trunk that is a trial from SIP.US. I have the domain name and cert working where it shows up inside of Teams Admin Center the TLS connection status is active and SIP option status is active as well. I am able to call the number and teams will ring but when I pick up, no audio is coming in and after 15 seconds, it hangs up.

I was wondering if anyone has run into this issue? I have also tried using the Audiocodes documentation by hand to configure it and TLS and SIP options also shows as active. In that case, when I dial the teams number, it doesnt even rings and gets automatic disconnect.

Any help would be much appreciated.

So, for about 5 days I have been trying to do this, I want to make calls on phone numbers using python script, but I am failing again and again, please help me, My VOIP provider is Fonial

Hi! If you've been paying any attention to all of the new sms bullshit at all, you know that there is a company called "The Campaign Registry" (TCR) that the mobile carriers use to "vet" and "register" "campaigns" for businesses. The first part of the problem is that "campaign" applies to ANY business text regardless of whether it's a marketing message, or a message in a conversation you're having with a customer. The second part of the problem is that TCR rejects entire industries of businesses just because of their industry instead of their sms use case. One industry they block is loan brokers. They say that a broker is a "high risk financial service" which is total bullshit. CAN it be high risk? Sure. Is it high risk just because it's a broker? No. When the blocking goes into effect my business will cease being able to send and receive text messages and our closing rate will likely be cut in half because our customers don't see communications in a timely manner.

My desire: there are thousands of legitimate businesses across the nation that have legitimate sms use cases that are going to be blocked. I want all of these to sue TCR in small claims court in their respective states. If we all do this it will bury TCR in so many lawsuits they might go out of business, which would be ironic since their policies will cause businesses all over the US to suffer, and end up being the reason they go bankrupt. To be clear, I am not advocating for frivolous lawsuits without any merit. I'm advocating that all of us that are going to be materially hurt by the unfair TCR policies sue them because of that.

Also, these carriers that use TCR are common carriers, which requires them to have a reasonable reason for refusing service to anyone. I can't imagine that not applying to businesses as well, and I think my business being a brokerage is not a reasonable reason, esp since we don't spam. Every message is hand typed by my employees.

Anyway, sorry for the rant, but we need to band together and sue them 10,000 times in every state that doesn't award attorney's fees to the winner, in case they win the case.

Hi! If you've been paying any attention to all of the new sms bullshit at all, you know that there is a company called "The Campaign Registry" (TCR) that the mobile carriers use to "vet" and "register" "campaigns" for businesses. The first part of the problem is that "campaign" applies to ANY business text regardless of whether it's a marketing message, or a message in a conversation you're having with a customer. The second part of the problem is that TCR rejects entire industries of businesses just because of their industry instead of their sms use case. One industry they block is loan brokers. They say that a broker is a "high risk financial service" which is total bullshit. CAN it be high risk? Sure. Is it high risk just because it's a broker? No. When the blocking goes into effect my business will cease being able to send and receive text messages and our closing rate will likely be cut in half because our customers don't see communications in a timely manner.

My desire: there are thousands of legitimate businesses across the nation that have legitimate sms use cases that are going to be blocked. I want all of these to sue TCR in small claims court in their respective states. If we all do this it will bury TCR in so many lawsuits they might go out of business, which would be ironic since their policies will cause businesses all over the US to suffer, and end up being the reason they go bankrupt. To be clear, I am not advocating for frivolous lawsuits without any merit. I'm advocating that all of us that are going to be materially hurt by the unfair TCR policies sue them because of that.

Also, these carriers that use TCR are common carriers, which requires them to have a reasonable reason for refusing service to anyone. I can't imagine that not applying to businesses as well, and I think my business being a brokerage is not a reasonable reason, esp since we don't spam. Every message is hand typed by my employees.

Anyway, sorry for the rant, but we need to band together and sue them 10,000 times in every state that doesn't award attorney's fees to the winner, in case they win the case.

Ever since we set these up at our two stores. One is fine the other store keeps getting calls “test”, 390237920793, 864000963994.

DP752

We run our own FreePBX. It’s our cloud and the phones connect under similar routers, firewall, etc.

Hi guys.

I am rather new to doing VOIP over WebRTC. So be easy on me.

I have CTXsip running in the browser. I have a PortSIP SBC running on AWS acting as the WebRTC WebSocket gateway.

I can connect and register the phone to the Asterisk based PBX in the cloud. However as the title states the calls are only going one way - Out and not in.

Any help will be much appreciated.

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I need it because I am moving out of germany and so many of my accounts use 2FA against my German mobile. I can't port the German Mobile because of German laws - long story.

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SO if I get the VOIP setup now before I leave in 2 days, I can move all those accounts to use the new VOIP number.

I moved as much as possible to Passkeys etc, but some accounts still only use SMS for 2FA :)

Then will get an Esim for data as I will be on the move a lot in different countries. Time for a new IPhone anyway...

Any ideas ? I am a IT systems architect but not a voip person.

I freaking hate Skype btw. There has to be something decent out there that competes. Don't want to hang my hat on MS and Skype.

- voip and sms for international inbound and outbound that 2FA works with. I heard Twillo is refused by some. Some spoofing thing or something I guess.

- app for MAC and IOS for using it that does not suck. Saw a few and they are crappy. WebViews, etc.

I am setting up a new VitalPBX server and I am trying to understand the best way to select the outbound CID from the actual phone. This will probably be using Yealink T54w.

Basically there is a need to select the outbound CID to match one of several DIDs that I have. Each number has it's own context and purpose.

It seems like one way to do this is create an extension for each outbound CID needed. Then I would setup the phone to register multiple extensions with the PBX. The user can then select the extension they need based on CID.

Another way I think would work is dial a prefix before the outgoing number and then create outbound routes to match it and set the CID. This is not as easy to use or very friendly, as the user has to remember the prefix and dial it before each number.

I was wondering if there is a middle ground where you can set up the phone with one extension and then setup buttons on the phone to select a "line". What this would do is dial the prefix for you. I don't think this is possible.

My thinking here is that it would keep things simpler if a person only has to think about one extension for inter-office calling but can still set the outbound CID as needed.

I'm weirded out, I got a text from a twilio number and they said something along the lines of "is this (my name)'s phone?" I don't know who it could be, I don't talk to a lot of people and the people I do talk to know my phone number and text me on it regularly. The caller ID just says the city and state of the area code. What is twilio used for?

Spam telemarketing has pretty much been made illegal barring some specific exceptions carved out by the TCPA. So, long story short, every time I manage to identify one of the spam callers calling my phone... I sue them and get paid. It's turning out to be a pretty nice gig. =)

It's really hard to identify some of these spam callers sometimes though. As you might imagine, they hang up when they sense any kind of danger, like asking for company names or website addresses. Right now for example, I have a call back number for this company, it works, but it has a vanilla voicemail and they are never going to call me back.

How could I get the owner of this phone number? Any advanced methods that aren't the whois sites?

Does anyone have experience with freeswitch esl?

I am hooked into freeswitch esl through node. I do this to greatly extend the functionality of freeswitch.

What I am trying to do is catch the event when freeswitch processes an SMS. I can see it on fs_cli (console) but I am not finding ANY event header for it and trying to avoid writing a js script fs side in mod_sms.

I googled for a bit and found that the event header was 'CUSTOM' (Where sofia events fire) and the subclass was 'sms::receive-message'. But this subclass does not exist.

I even looped through all object properties of the event::custom object. I searched for sms and SMS but turned up nothing.

Does anyone have any experience with this?

I'm using the latest version of sip.js (0.21.2) in a nodejs app.  I have an external device that sends and accepts packets of G.711 ulaw encoded audio using RTSP.  I am able to instantiate my SIP client, and able to  call that SIP client and accept the call.  However, I'm floundering in what to do next in order to handle the RTP packets to and from the caller.  My intent is to act as a proxy between the caller and the external device shunting audio packet between the two.

I'm really struggling to figure out what to do next.  It feels like creating a custom MediadHandlerFactory is the way to go,  but beyond that I really don't have any good ideas about what to do there.

Would anyone out there be able to offer any sort of help or information as to how I should proceed?

Thanks in advance

UpCentral.io offers a range of solutions specifically designed to address the needs of small business owners. These solutions include:

1. Employee Scheduling: UpCentral helps in streamlining the process of employee scheduling. It provides real-time, transparent scheduling which aids in communication, prevents scheduling conflicts, and enhances overall productivity. This leads to smoother operations and more efficient teams​​.
2. Business Communication and Organization: The platform offers advice and tools for effective business communication and organization. This includes tips on phone and text messaging, employee scheduling, and online invoicing, all aimed at assisting small business owners in focusing on growth while managing the complexities of day-to-day operations​​.
3. Business Phone System: UpCentral also features a next-generation business phone system. This system includes advanced features such as easy call transfer and seamless integration, ensuring reliable and effective communication within the business. It's designed to revolutionize how small businesses handle their communications​​.
4. Comprehensive Mobile Application: UpCentral provides an all-in-one app which allows small businesses to manage not just scheduling and communication, but also time off requests. The app can turn any iPhone or iPad into a fully functioning business phone system, enabling businesses to stay connected with customers and vendors through text messages (SMS) and calls​​.

These features collectively address several core challenges faced by small business owners, such as efficient employee management, effective communication, and streamlined organizational processes. By providing a suite of tools in one platform, UpCentral.io aims to simplify these aspects of business management, allowing owners to focus more on growth and less on administrative complexities.

Can someone guide me on where to post questions on a recommendation

I have a client who has an existing Toshiba IPedge PBX, with 35 extensions and four pots lines connected via an audiocodes FXS gateway, the IPedge has been fine for them except now it’s saying that its perpetual license has expired because the host ID has changed, is there any way to fix that licensing issue, and if not any amount of guidance would help tremendously as these guys have been on the same pbx for almost 15 years now and the IPcentric equivalent from cox is an absolutely ridiculous addition of 1,800$ a MONTH to our existing service.

We also would have to address switching out phones either for a soft phone app with a headset or a desk phone (both would be super nice to have) that is under 100$ per unit.

Hello all,

As the title states, I'm looking for a new VoIP provider for my company and FC reached out to me and we have a demo scheduled in two weeks.

For some background, my company, roughly 500 users distributed between two main facilities and 35 distribution centers all on the east coast, migrated to BroadView in 2017 from individual NEC systems. At first, we really liked their service as their support was great, their feature set allowed us to mimic the old NEC system's functionality, (static park/retrieve slots specific to each facility to mimic hard lines for ex) which was a boon for our older users. Minimal training was a big plus.

Fastforward a bit and WindStream aquired Broadview and promptly purged their knowledgable local support staff in favor of outsourced teams. They filed for chapter 11, service reliability went down the tubes, and it's been a slow uphill climb to get back to a point of reliability that we once had.

Now that I've got the reigns, I'd like to move to a new system that pairs with Teams as we're married to 365. Fusion Connect reached out to me and I learned that many employees of FC were former BroadView people, including a large number of the engineers and the CEO.

I don't see much about FC, other than their own chapter 11 filing in 2019 that they recovered from in 2020. Has anyone out there had experience with them and can offer any insight?

Thank you all!

There was a really funny one out there a while ago. Can anyone remind me please?

So I have a bit of a difficult situation. I did a small install in an office with only one phone line at first. As a hardware phone we opted for a Yealink W60b Base with a W56H Handset. The setup is just working fine with no issues.

A couple months later the customer got a second line from the provider (German Telekom) which we setup in freePBX and got the customer a Yealink W52H as a mobile device. Strangely this phone is making and receiving calls just fine but if it gets called from outside the caller isn’t able to hear anything while the receiver is hearing anything normal. Also the call gets ended after around 30 seconds ago. I tried switching the lines around so the “old” line would lead to the new handset and vice versa and now the problem moved to the old line. So I thought it must be the handset. But both are configured the same (nearly defaults) and also the problem doesn’t exist internal calls.

Has anyone got an idea what the problem could be?

Hello, i'm an IT&C student and i have to create a simple Session Border Controller project, basically two or three virtual machines (i use Ubuntu 22.04 for all of them) and two VPNs, one for each VM and the middle one to route between them. In the first virtual machine i use a Voip server like Kamailio which is working, Asterisk but my recomandations were to use FREESWITCH, and for the second VM, the middle one with interfaces in both networks i should use a SBC to protect the server and internal possible Softphones from the third machine in the second network.

My problem is that the informations i read across the Internet are vast and i'm a beginner in VoIP, although i like the ideea and i want to learn it, BUT my time for this project is limited and i need something working to learn faster.

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In present, i succesfully created a Kamailio server with working softphones that can communicate, but after SO MANY TENS OF HOURS, i can't establish a connection between it and a Session Border Controller using Kamailio as SIP server and Freeswitch as SBC or vice versa. I mainly used this guide: https://developer.signalwire.com/freeswitch/FreeSWITCH-Explained/Auxiliary-Knowledge-and-Utilities/SBC_Setup_13174198/ or https://tvveaks.wordpress.com/2012/10/16/kamalio-sip-proxywith-freeswitchsbc-configuration/ but it doesn't work, cause Kamailio won't load with those configurations and i think it's pretty outdated. I also used Kamailio as SBC , LibreSBC but cant get over that Ansible and signal token, SEMS SBC which i think it's outdated or i can't understand how to install it.

I'm feeling a bit overwhelmedbecause of internet's different informations and I'M ASKING FOR A STARTING GUIDE because I just want to make this project as simple as possible working succesfuly because i have a lot of time to understand and learn it by analyzing it after it will deliver some results, like protection DOS attacks or NAT topology hiding, AND I WOULD BE GRATEFUL FOR EVERY TIP AND TRICK, again i have to say that i'm certainly not informed enough and i don't understand many aspects in this domain.

sorry for my English

How was my comment "advertising". Clearly someone works for bonline.
Guess you will just delete this post too.

https://preview.redd.it/wdyqz1icaj0c1.png?width=737&format=png&auto=webp&s=a28005b9dfeb2e257e9acf34fe3f61c45474748e

Anyone know if an ata exists that you can whitelist numbers? Prefer not to do on pbx side and have ata connect directly to Sip provider. Looking to setup for an elderly person and only allow specific people to call.