VoIP

Apparently in 2023, some people still require you to call physical numbers. (ebay for instance) I'm in Indonesia. I need to make calls to places around the world, usually the USA.

I Googled, most people say Skype. I tried a few other services but they don't seem worth the steep fees.

I wanted to shed some light on a concept that, I feel, is a game-changer for many organizations - network segmentation (NS). At its core, you could look at NS as if compartmentalizing a ship. If one compartment gets flooded, the others remain unaffected, preventing the ship from sinking. Likewise, by dividing the network into segmented zones, we're basically containing the security risks within that specific zone. This means that even if we come across a security incident, its impact remains confined, safeguarding the broader network environment. This line of thinking could beg the question of whether VoIP systems are susceptible to the same threats. Sure, VoIP, like any other networked tech, can have its share of nuances. However, my stance is that integrating it within a segmented network mitigates risks associated with unauthorized access/data breaches, especially if you’re dealing with the top providers in this niche. Getting back to NS. IMO, one of its major perks is the granular control it offers over user access. The way I see it, companies that implement strict access control ensure that only authorized individuals have access to specific network segments. This lessens the risk of insider threats and enhances security around sensitive info. When an organization has multiple departments with varying levels of data sensitivity, they’d probably want to build an infrastructure where department X’s sensitive data is inaccessible to someone from department Y. Now, we’ve all heard or even come across incidents where a (seemingly) harmless device served as a network breach’s entry point. Wouldn’t it be safe to assume that by isolating IoT devices and other endpoints in their respective segments via NS, the risk of them becoming a gateway for malicious actors is substantially reduced? And the implementation stage isn’t even a cumbersome task at this point in time. With the advancements we’ve been enjoying in SDN, creating and managing segments has become smoother. Not saying it’s a be-all-end-all, but it def has a place as part of a layered defense strategy, adding to all the firewalls, intrusion detection systems, etc. Curious to hear how NS has shaped the security posture of any organizations you may be aware of.

I don't know where else to ask this...

My company purchased the wrong software licenses from Ribbon Communications and refuses to accept an RMA even though our sales guy steered us down the wrong path. They won't work for us as they won't play nicely with Transnexus's ClearIP service. We will move forward with a different SBC solution for our use, but are stuck with these costly licenses.

Does anyone know (or want) a reseller that might be interested in these licenses below their cost? While we are a reseller for Ribbon (so the licenses will be supported if transferred), we have never sold their products and, after this experience, certainly do not intend to.

Please DM or comment - I appreciate any help!

https://ribboncommunications.com/products/enterprise-products/session-border-controllers/sbc-software-edition-sbc-swe

I've been an ITSP for almost two decades and the single biggest PITA with respect to rolling out new account over these years has been traversing the firewall. If you're an MSP and have control over the premise firewall, it can still be tricky with some edge equipment. But if you have no control over what that equipment is and no admin level access to it, then it is often a negotiation with the MSP or IT department to modify the firewall.

We are starting to migrate our customer accounts from a variety of platforms over to SkySwitch and am interested to hear from other Skyswitch ITSPs on how they make this as easy as possible. We have some legacy accounts on a Broadworks switch that have Edgemarc on prem but that's not a viable or economical solution going forward. We have some on 3CX and their SBC approach has been great, especially with special firmware for Yealink T5x series that can make any one of them an SBC for up to 10 phones each. The phones register through them and the tunnel it sets up is firewall-proof.

What's solution to get around the firewall issue?

Hi all,

I'm trying to port out my number from Hushed to Lucky Mobile (Bell Canada).

The porting service tried many times to initiale a port but it always failed.

When the search for my phone number, they get Iristel as the actual carrier instead of Hushed (Hushed rep. confirmed me that it should not be the case).

Any tips and/or suggestions would be appreciated.

It's been set up for using certificate before, and now we're going to a new vendor that uses pilot user and I don't known where to put the authorization information since there's pass through / remote and local tables.

Hello, I hope this is not too off-topic, but I suspect there must be many people here who have run into this. I use voip.ms, and have always received voicemails as email attachments with no problem. Since switching to Windows 11 from Windows 10, I can no longer play the files, but have to download them. Windows 11, Chrome, gmail. Error message: 'there was a problem playing this audio file'. Has anyone found a solution? Many thanks in advance.

I want to create many virtual numbers for testing and have seen that Durian is providing virtual numbers cheaper than others but I need an invitation code to even make an account and everybody is selling it. Does anybody have a Durian SMS invitation code? Please send me one in my DMs.

Thank You

I want to create many virtual numbers for testing and have seen that Durian is providing virtual numbers cheaper than others but I need an invitation code to even make an account and everybody is selling it. Does anybody have a Durian SMS invitation code? Please send me one in my DMs.

Thank You

Hey!
I want to create a "Phone" using a raspberry pi and some buttons. I was looking for a way to get the button input into some sort of client but had no luck finding a good solution. I was looking for command line tools that I could use in a python script but all I found were deprecated or undocumented :(

Since I'm not a VOIP Pro, I wanted to ask here. Maybe someone has a nice tip or solution?! As I said the goal is not necessarily to find a command line tool, it is to get the button input from the PI's GPIOs into a Software that can process audio only calls. I am fairly experienced in scripting, but I don't know how much effort it would be to write a tool from scratch

Thanks already if someone has some kind of Information for me <3

My org has two separate VoIP providers. One provider that is set up with our corporate PBX system (a 3rd party cloud system we don't have any control over) and another that is tied to an Asterisk instance we set up in the AWS cloud.

A few months back we ported a local number from the 1st provider to the 2nd.

Based on some issues we've been trying to track down, it seems that even though the number was ported to carrier 2, it seems like some carriers are still routing calls to the number in question to carrier 1.

Is there some kind of tool I can use to check on the "routing" for this number from various carrier viewpoints? In the networking world, I would use a looking glass server at various ISPs to get info on routing for a given IP address. Is there something similar I can use for a phone number?

Have a super small office that uses a step up from a home 2.4Ghz phone setup. They have been using Spectrum's phone system that provides 2 analog phone lines straight from the ISP modem.

The office is switching to AT&T fiber but they can not offer their similar phone system to the area in any reasonable amount of time.

So I am looking for a basic VoIP provider that offers an analog handoff for the 2 lines they need. I see Vonage offers a box that does this https://support.vonage.com/articles/answer/Vonage-Box-VDV22VDV23-Telephone-Adapter-1041 but not sure if that is for business or residential service, not that it really matters to the office.

Is there some other provider that I should be looking at? This is very basic, nothing past having the line ring is needed.

Looking to establish phone access to the video doorbell stream on our hosted VOIP phones.

We have the Ubiquiti Doorbell Pro and Yea-Link T57U phones. We were hoping to be able to interface the two and wondered if anyone had any suggestions. Ubiquiti provides an API and the camera has HTTP access if that helps.

Background

We used Comm Quotes to get our phone service through Ring Central with some explicit requirements. The most important of which was that we needed to be able to access the video door intercom system for controlled access. We wanted to be able to click a button and interact hands free with the front door. Then pressing a button would open the maglock.

We purchased the phones that we were told would work and they have not worked yet. We purchased the door camera ($1600) and it only works through a terrible dated web interface.

We can finally press the 6 button to ring someone in, but not with video.

We are moving to a new office now and still have a year on our contract with RingCentral. They have not been any help at all getting this to work and quite frankly, we are ready to move on when our contract expires.

We are considering moving to Ubiquiti Talk, but not sure yet.

Meanwhile, we have the Ubiquiti Pro Doorbell access. control system. It doesn't work with 3rd party VOIP, so we are looking for alternative ways to manage the intercom from our phones. Ubiquiti has an API which would allow HTTP access to the door, does anyone have any ideas on how to get the YeaLink T57w phones to display an http page or video?

https://www.yealink.com/en/product-detail/ip-phone-t57w

I am in the process of doing my Zoho PhoneBridge integration and I am trying to get Click to Dial working. Well, I do have it working, but only for physical phones and the mobile app. WebRTC (Sip.js) is having an issue.

I am calling the originate command through ESL using node. Click to dial works great if I have a Yealink or my mobile app registered, but if I only have the web app registered (Sip.js), then Freeswitch console throws:

2023-11-28 14:04:42.951275 97.17% [NOTICE] switch_ivr_originate.c:3049 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
2023-11-28 14:04:42.951275 97.17% [DEBUG] switch_ivr_originate.c:4045 Originate Resulted in Error Cause: 806 [USER_NOT_REGISTERED]

The funny thing is. The webapp is registered. Makes and receives phone calls just fine and even sofia shows the extension is registered. I have reactive components that actively show the registration state in the web app too.

I have no idea why Freeswitch is throwing this error.

Anyone have any leads?

Hi, so my company decided to make a proxy for their internal and external phone call. so every call goes through the server (I'm using kamailio for it). then if the number called was internal, it send calls them. now for the external call, we had several SIP providers. in the past we use microsip to do external call with all of them. until the management are getting some problems like different dashboard for each providers, and too many calls to one provider while others are not being used that much. so here i am making a round robin for it. and would love any kind of help or pointers to make it.

my questions are :
- is there a way to ping the sip provider?, i saw a ping using OPTION but it doesn't seems to be reliable since most people would drop OPTION instead of replying them. i was thinking of console ping but not sure how to connect it to dispatcher database, how do i weight and prioritize them. along with when their server is on, but their asterisk are having problem.

- each providers had their own username and password, which they called an extensions. i think some might have a number prefix or suffix that needed to be added to phone number before i could use them. now how do i connect to it?, i tried UAC module but still unsure if it works

- when i called using JSSIP, i am getting sip uri like `123456789@sip.localhost.com` how do i replace the `sip@localhost.com` using uri that was used in the dispatcher that probably connected to UAC?.

my config the cfg is heavily adopting ChatGPT, i really had no idea what I'm doing.
the plan was to receive phone call only from TCP / websocket (we making our own softphone), then forward external phones to the real SIP provider. while rejecting UDP call. but it seems i need UDP port to send data to heplify / homer.

I have a weird problem with my system. Whenever I dial other people menu tree/handler numbers, it will connect to their voicemail system. Most are using Cisco unity but there are others too. If I dial their direct line, it is fine. It doesn't happen to every menu tree/handler number or maybe they are not VoIP or something. What should I check?

I need recommendations for a VoIP monitoring tool that will work on a Windows computer to mainly show jitter, MOS, packet loss. This is for my home VoIP network so I would prefer something that is free.

Here is my current setup and equipment.

I would prefer something on Windows; however, if there are no options for Windows, I do have a spare PC that I can install another OS on.

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https://preview.redd.it/anjxr7bz5z2c1.jpg?width=3151&amp%3Bformat=pjpg&amp%3Bauto=webp&amp%3Bs=cdd42565e17d76be22c03052b68c7e0ae6c87ca4

Hello, I run a home inspection/pest control business that requires communications with clients from my support staff. Outgoing/inbound calls and texts.

Currently using ring central and paying $300 a month for 3 members of my support staff which are based in the Philippines.

The pest control business has a different phone number from the home inspection side.

Any recommendations on comparable services that don’t cost an arm and a leg?

Thank you!

Looking to get a new home phone for my VoIP service and I'm wondering if anyone knows of a chart that compares all of the Panasonic phone models available cause there are quite a few models and no easy way that I've found to actually compare them on the Panasonic website.

1. Is PSTN for 2G and 3G calls and possibly 4G VoLTE calls in the early days and VoIP would be calls made through the PDN?
2. I am a little lost of after the user is authenticated on the IMS and the SIP connection is made at the SCSCF, how does the PGW know to forward User A call traffic to User B. Does the IMS use the SCSCF to tell the Media Gateway to tell the PGW to allocate radio resources for the VoLTE call?

Long story short, my business bought a commercial building and we're moving our company there. It's out of town, fibre optic isn't available, and the previous owners said VoIP is horrible through the available internet.

My question is, what are my options? I'm in Canada, is there a service provider over 4G/LTE where there's a hub for SIM, and I can still keep my extensions? Or do I have to switch back to traditional landline?

Hi all, how can I find out the latest telecommunications tax in my area (Snohomish, WA)? I got a quote from a VoIP recommended by colleagues in September and they quoted 17% estimated telecommunications tax. I got a new quote from them last week and now it's 19%. They said it's because the tax increased even though it's still the same calendar year? Thanks

https://i.imgur.com/TjbtpVl.png

helllo.

Yes I know the phone system is old (no need to mention that).

My provider is migrating and is now requiring "

When sending calls you must put the pilot number in the p-asserted-identity sip header in full E164 format (+CC). All other phone numbers must be in full E164 format (+CC).

"

so I am unable to place outgoing calls. Incoming is fine. Accounts are registered and reachability is ok. The devices I have are yeastar u100s and some SOHOs. How can I correct for this in the gui? or patching the code through ssl?

I tried appending +1 to the dial string, which is something I need to do anyway. No difference.

Aybody have any ideas? Thanks. Im stuck.

8x8 won't offer SMS for 800's and e-fax cannot apply to the same number.

I want a solution where I will receive fax and SMS for my vanity number.

Talking to Ring Central was a good experience and they seem to be a big player in the space.

Review on the web and here on Reddit indicate Nextiva is horrible but I can't get a good reading on other worthwhile competitors.

Any recommendations for 800# hosts other than Ring Central and how is your experience with RC?

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What VOIPS are free for Android that let you change your area code to another states that are in Android stores for download for free?

Anyone who knows, please help.