VoIP

Hello everyone,

I'm launching a recruitment business based in the UK but operating in Latin America and the USA. I've explored VOIP options like TEAMS, Ring Central, and Skype, but none seem to offer cost-effective packages for both regions, unless I’m understanding the billing incorrectly.

For instance, Ring Central provides 4,000 minutes a month to the USA, but calls to Latin America cost approximately 15p per minute for mobiles. With an estimated 10 hours of weekly calls to both regions, this pricing doesn't seem sustainable.

A couple of key points to note: I will primarily be calling mobile numbers, and I'm also in search of a VOIP service that can provide me with a USA number. While 90% of my business will be in LATAM and the USA, I'll occasionally need to call mobiles across EMEA, but this would be a small amount of phone time.

While considering using separate VOIPs for North and South America, it proves to be neither cost-effective nor practical.

Any suggestions or recommendations for a VOIP service that caters to both regions, includes a USA number, and is suitable for contacting both mobiles and landlines would be greatly appreciated.

Thank you.

How the hell do i transfer calls? and i mean i have someone on the line i put them on hold i call my colleague to see if they are available. What do i do after that? I merged the calls once and the other time i hung up on my colleague it ended both calls. HELP pls

Any alternative to Grasshopper? im trying to signup with them but getting a notification that the country to where i am now is not supported.

Can i Use VPN so that i can use grasshopper or maybe there is an alternative.

Im in the Philippines.

My requirement is I need to call US clients and Have 1800 number on my website.

I have no idea why this is happening, but I'm using FreePBX and i've setup Sangoma talk. It works perfectly fine over WiFi and 4G for some providers (Optus) but when i try to register the phone on 4G on another provider (Telstra), it can't authenticate. It only doesn't work on that network (Telstra).

From my understanding of SIP registration, it should go like:
CLIENT to SERVER: REGISTER
SERVER to CLIENT: 401 Unauthorized
CLIENT to SERVER: REGISTER (containing authentication information)
SERVER to CLIENT: 200 OK

I've setup a wireshark capture on my phone connecting from my Optus SIM and it authenticates without issue. I see in the SIP packet the Authorization, the digest username, the realm, the nonce and URI in the actual packet itself, it looks like the above.

However when i look at the Telstra packet, the authorization part of the packet just isn't there? It's as if it never responds with the Authentication, which is strange, it looks like:
CLIENT to SERVER: REGISTER
SERVER to CLIENT: 401 Unauthorized
CLIENT to SERVER: REGISTER
CLIENT to SERVER: REGISTER
SERVER to CLIENT: 401 Unauthorized
CLIENT to SERVER: REGISTER
CLIENT to SERVER: REGISTER
SERVER to CLIENT: 401 Unauthorized
(and continues)

Does anyone know where i should go for this? Why would my phone behave differently on the 2 networks?

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.

Have a client looking for a cheap USA based VOIP number that will allow 4 users on the road to take inbound calls made into one virtual number, they may also need to dial back missed calls. Simultaneous ringing and unlimited inbound minutes (if possible outbound minutes) included. Nothing fancy needed just the above and maybe voicemail. Users could answer calls via app on their personal phones (Android and Apple). Any recommendations?

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&format=pjpg&auto=webp&s=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!

I used to live in the UK. I still have my UK cell phone number (07) with a traditional provider.

I want to get rid of this and move to a VoIP solution. I very rarely use this number so would prefer to have as minimal cost as possible.

By coincidence, I have a UK landline number (01) with sipgate and that works just perfectly - I top up on their website occasionally and I get my messages in my inbox - great! Unfortunately they don't work with non-landline numbers.

Does anyone have any suggestions or recommendations?

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.

I have been a user of Line2 for multiple years but service quality has steadily declined so I am looking for a cheap alternative.

I spend most of my time outside the US and have been keeping my US phone number active primarily for banking. I just need an app that reliably can handle US calls and SMS for my legacy phone number, nothing more. Very low usage.

Now that my number is no longer with a carrier, some options (like Google Voice) are unavailable for direct porting. Porting to a mobile carrier and then back (which some do) makes me nervous something can go wrong given I have no way to test it with the carrier. I prefer to switch to another VOIP provider in one step. Suggestions?

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

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I have decided to go full fibre which has a planned install date for a weeks time and as a result of that I will lose my landline and phone number. I need to keep my phone number so decided to go with voip. I signed up for voip and have been given a number which I can use. But I need to transfer my old number to replace it.

I am just unsure about the time frame in which it needs to be done. I know I need to transfer my old number before my current landline is cut off or I lose it. That process according to the website takes minimum of 10 working days possibly longer. But does the number transfer just need to be started before my landline is cut off? or does the transfer need to be fully completed? Should I delay the installation?

I do not want to end up in a situation where I lose my old phone number. Any help advice would be greatly appreciated

I’ve recently started a new job that is 80% wfh after induction. I’m half way through and already quite confident so not worried there. One pre-requisite was organising a phone line or voip. I’m not having much joy with my NBN provider even getting a tech upgrade sorted let alone being able to get a number added by them for an extra $10 a month just to receive calls. Is there a provider of just voip fixed line numbers in Australia that I can use that is going to be similar price or even better, free?

We’re currently using Cytracom and have the ability to use our main number as a shared SMS inbox. However, we’re experiencing some growing pains - the desktop / mobile/ web apps aren’t very reliable, often need rebooting to continue viewing updates to a thread, the call history doesn’t show all calls to our main number and by design only shows those you’ve answered individually, among other issues.

Each are minor, but with constantly being upsold on other product lines, we’re ready to switch.

Our workforce is remote about 50% of the time, so a help desk without a permanent home. We’ll need an option with a soft phone and shared sms capabilities on the primary business phone number, not an individuals DID.

Web app is a plus, as are desk phones. Neither required.

Thanks in advance.

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.

I need some advice. We use Twilio mainly and it doesn't support ATA fax lines. We signed up for a voip.ms account to setup a few lines for clients that still have some fax machines but they instantly closed our account and said we broke ToS but wont tell us why.

Our support team is based in India so I am guessing it was due to them detecting an Indian IP logging in as we didn't even setup a single thing yet.

Voip.ms are refusing to re-activate account so I need an alternative that I can setup some lines that just connect to a few ATA fax machines.

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Any suggestions?

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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Hi,

i'm managing a voip system in the company where i work (i inheritted the system). We have a freepbx system running on an old pc which sounds like it's going to give up anytime soon. also i don't like how slow and limited the machine is (takes 5+ min to access the web gui, freepbx not activated, no admin console and overall i never worked with freepbx). first idea was to get a new machine and run 3cx self-hosted, but their yearly subscription went thru the roof recently. i was managing 3cx before and i know to admin it, but pricing is uff.

so option 2 is a hardware dedicated voip central - grandstream ucm6301 sounds like what we need - we have 30 extensions, a few call groups, a simple IVR with 5 selections, work hours and out of work hours recording and routing, plus a few rules for internal calls going out thru the gsm gateway.
all extensions are 3 digit, cisco and yealink ip phones. no analog phones. 2 sip trunks to provider. No mobile phone app needed.

i'd buy 2 and use them in HA mode.
also we have a video door phone that works well with a yealink phone, so not sure if i need a grandstream that supports video, but the price difference between ucm6301 and ucm6300A (audio-only model) is not big...
system is running on it's own subnet on a vlan. trunks are connected to a provider directly with no internet connectivity (dedicated voip port on the ISP gateway).

any advice on this ? IS this a good choice of hardware and overall upgrade?

thanks

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.

I would like to know if there is any service that provide 2nd phone number/Virtuale one from Luxembourg.. I tried to look and do some research but couldn't find any legal/paid service can provide receiveing sms in Luxembourgish phone number.

So we are a small company but very niche. We need to hire more people to help with phone customer support, the problem is all customer support requires very specific niche knowledge as well as company specific knowledge about our processes.

We have it all documented in a database, but the new hired users have to manually search this database for the right answer to the problem which can be problematic.

It would be awesome if there is some VOIP or program solution that has a live transcription with some form of AI while customer and support technician, that knows our database and instantly suggests the best possible answers/articles from our documentation to our support technician.

Does such a solution exist?

Thank you in advance guys for the attention and advice!