I made the same choice, switching from an AT&T land line to the Ooma Telo. You buy the box (currently $119 on Amazon), then you pay taxes and fees monthly based on your location ($3.71 for me) and $10 per month if you want Premium. But otherwise it's free local and domestic long distance for as long as the box lasts. I ported my number over, and have been quite happy with it over the last six months. They have E911 too.
To me the personal and community call blocking feature that comes with Premium is worth every penny. It isn't perfect, but generally works quite well.
I should say that I use the Telo to drive three wired phones through my original home phone wiring, and that works fine. I don't know about the various wireless options they offer.
You do need solid high speed internet for these VOIP devices to work well. I have Cox Cable. I have my Telo immediately behind the cablemodem, and then my router plugged into the Telo. Seems to work fine, but you can also put the Telo behind the router if your router provides QOS. You need to give the Telo priority over other traffic.
The voice quality is quite good, at least at my end. Nobody at the other end has complained.
My advice: If you need this for business, don't switch. Aside from technical issues with VoIP I've seen cases where phone numbers could not be reached with calling cards anymore after they switched. There is no free lunch.
The phone system as we know it when any number could reach any other number no matter what seems to be beginning to unravel.
I've had not a problem with Ooma and number porting... everything went smoothly, and faster than they predicted. ...Jim Thompson
--
| James E.Thompson | mens |
| Analog Innovations | et |
| Analog/Mixed-Signal ASIC's and Discrete Systems | manus |
| San Tan Valley, AZ 85142 Skype: Contacts Only | |
| Voice:(480)460-2350 Fax: Available upon request | Brass Rat |
| E-mail Icon at http://www.analog-innovations.com | 1962 |
I love to cook with wine. Sometimes I even put it in the food.
I do much the same thing. I have a Linksys/Sipura SP921 at home, an SPA 941 in the office, a Linksys PAP2 box for travel, and various "soft phone" programs to run on the laptops and smartphones. They're all on the same phone number. If you call my VoIP number, they all ring. Of course, I can't call OUT on all the phones at the same time, but since I'm the only one using the phones, it's not a problem.
Ummm... do the jitter testing anyway. It's easy enough.
Some of my comments on Comcast bandwidth limiting and jitter.
However, if you have Comcast Xfinity phone service for the home, the VoIP traffic is on a completely seperate RF channel from the internet traffic. The internet can be comatose, and the phone service will still be up. The Arris home internet/phone routers include a built in battery for backup. For business class, you get two separate boxes for internet and phone. With this separation of traffic, Comcast really doesn't care about VoIP traffic on their internet service, when they're in business to sell it to you separately.
No, you need to determine if you need a full service provider, with lots of support, or a cheap provider with an Asterisk box in the closet.
Others have explained SIP quite adequately. If you're going to dive into the VoIP quagmire, be advised that things are very different on the telephone side of electronic fence. For example, everything is an acronym. Every vendor has a different acronym for essentially the same protocol and service. There are many ways to do simple things, most of which are incompatible with each other. A good telecom dictionary will be useful.
Again, others have already explained this. Very simply, if you want to talk to someone, you can do it directly (SIP to SIP), indirectly through a gateway to the PSTN, or through a service providers gateway. Along the way, you'll blunder into firewalls, STUN, STUNT, and TURN servers, as well as RFC's for everything imaginable. (Aren't acronyms wonderful)? To make it easy for users, there are also network to network gateways: and DNS directory servers.
To make outgoing calls, your SIP phone will need a dial plan. Here's mine: (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|xxxxxxxxxxxx.) Note the general lack of an intuitive structure.
However, little of this is really intended to be saved in the SIP phone. My various Liksys boxes have NV memory, but no way to backup or directly load settings into the phone. I would guess my SPA921 has about 200 boxes to fill in. Defaults? Nope. Instead, you're expected to have a TFTP server sitting at home or on the internet with the settings stored. When you connect the phone to an internet accessible LAN, it logs into the TFTP server, downloads the settings, and provisions the phone.
Hint: If you don't want to deal with all this, find a vendor that will do it for you. You may not be able to do tricky things, like my common ring, but at least you'll be sleeping at night instead of hacking the SIP phone settings. I suggest trying Callcentric: I used to have an account with them before I learned enough about VoIP to cause problems for the service provider. About $13/month. Note that call in and call out are separate services.
You're being generous. Their site sucks. The support/faq section is even worse. Like I mumbled... they're for users that can handle the technology with minimum support.
Future-nine has good pricing and fairly good service. I don't care about the web pile.
Hmmm... tracerout to incoming.future-nine.com shows 100 msec latency and 15 hops through 4 server farms. Not the best route for VoIP. Also try sip.future-nine.com and outgoing.future-nine.com. Check it from your location to make sure it will work. This might also help:
--
Jeff Liebermann jeffl@cruzio.com
150 Felker St #D http://www.LearnByDestroying.com
Santa Cruz CA 95060 http://802.11junk.com
Skype: JeffLiebermann AE6KS 831-336-2558
SIP is a protocol for making a connection (dialing), and passing VoIP traffic. More specifically, it's a collection of protocols defined by numerous RFC's that are needed for VoIP to function. The alternatives are H.323, Skype, MGCP, etc. In this case, all SIP does is define a common protocol needed for you to talk on a VoIP service providers system.
A VoIP call that uses the SIP protocol to dial and talk. Lots of things have to be compatible and supported for SIP to work. The most important is the CODEC. You'll probably be using G.711u (64Kbits/sec uncompressed). If you run into a limited bandwidth connection, like dialup, there are CODECs with compression available.
--
Jeff Liebermann jeffl@cruzio.com
150 Felker St #D http://www.LearnByDestroying.com
Santa Cruz CA 95060 http://802.11junk.com
Skype: JeffLiebermann AE6KS 831-336-2558
I appreciate your effort, but the issues of the protocol are not what I'm asking about. The VoIP provider talks about "Free outgoing SIP calls". I need to know what this means in their context. Oddly enough I don't see any contact info and their web site is an odd assortment of pages that are poorly organized. They talk about E911 on one page as not being up yet because they are in "beta" status, but this is dated
2008. Elsewhere they say they have E911. Elsewhere still they say E911 is not free but go to their "rate" page and all you get is a "call simulator" which gives you per minute rates for calls between various locations. No where do they define what they are talking about by SIP calls.
I'm not sure I can deal with a company that has no phone support, no email contact, in fact, no contact info at all! WTF???!!!
Somewhere this is going right past everyone. I'm not asking about any of the technical details of how the VOIP phones work. I'm asking about what the vendor means when they say "Free outgoing SIP calls". Clearly the user doesn't need to know anything about call setup and teardown or what CODEC is being used. It's a phone. You make calls with it. When the user wants to know if he is being billed for a given call how does he know if it is a "SIP" call according to the statement on the web page? Does that mean you are calling another Internet phone and not going through the PSTN?
All in all I am very unimpressed by the Future Nine web site. I don't want to become an IT expert at VOIP. I want to buy a phone, sign up for a plan, have it work and understand what it will cost me. That's all I'm asking. Is that too much? I thought for some reason that this is a place where I might find someone who has gone through all the "stuff" of finding a decent service and save myself a little trouble. I guess that's one advantage of Ooma, it just works and I don't have to learn how it all operates.
Outgoing calls only require them to give you some (IP) network bandwidth. They don't have to "give you a (persistent) presence".
You'll either install some software on a PC (that you will keep running whenever you want to make a call) *or* purchase an ATA (Analog Telephone Adapter/FXS adapter) to which you can connect a traditional handset. In either case, you will then have to configure the device/client to know how to access the SIP service from the VoIP provider (account name, password, SIP port, registrar name, etc.)
You can't, for example, fire up a Skype client (on your PC) and use their service to connect with other skype/PSTN/etc. users.
The big *draw*, of course, is to get you to want other services besides outgoing voice! E.g., to be able to *receive* calls! :>
If they have a (truly) FREE service, why not try it out and see what you can and can't do? At the very least, it will give you an idea of the quality of the service (and support, etc.) that they offer -- should you opt to head down that road formally!
(sigh) This is all too common. It's seen as a "chore" instead of an *asset* by many companies. (and, to be fair, maintaining a web presence *is* tedious -- even if all you are doing is keeping contact info, services and rates "up to date")
I think a good many web sites are offloaded to "web developers" with no real ties to the companies they support. "Build us a web page..." etc. As if they were asking "Build us a parking garage"
Cost of admission. They figure you know this *before* looking into VoIP services :>
SWMBO has a prepaid cell phone for "emergency use". She grumbles about how *impossible* it is to talk to someone about the service, rates, billing information, etc.
Obviously there is something going on that I don't understand. I am picturing a small box, about the size of a router, which connects between my phone and an Ethernet port on a router. I pick up the phone, dial a number and the person at the other end says, "Hello", the call has been connected. Where exactly would I enter such an ID?
This is the setup a friend has and pays $10 a month on top of her cell phone bill. She gets unlimited calls and long distance to the best of my knowledge. I'm looking for something like that. $10 a month would be great.
Ok, this is what I'm thinking the Future Nine web site means by "SIP calls", but who knows? Their site is obtuse and incomplete..., VERY incomplete.
I don't get what you mean. What is "terminated to the PSTN"? You mean the call goes to a phone on the PSTN, otherwise known as a land line? How do you do that without a phone number, especially if you are using a phone? Oh, you mean "I" don't have to have a phone number for outgoing calls. Ok, I hadn't considered that since I am looking to replace a standard phone with calls going both ways.
Ok, so that is still a bit unclear. Their pay as you go plan is "free" with a $5/month fee for the phone number while their "Bare Essentials" plan is just $7 a month but with a "free" phone number. So they are only charging $2 a month for 250 minutes of outgoing calls. An odd way of doing it, but I suppose there might be pay as you go customers who don't need a phone number (commercial service) while the other plans exclude commercial service.
Yes, it is *very* complex until someone explains it clearly. I think this is the best I have seen to date. The part that bugs me about Future Nine is that the web site sucks so bad and it looks like I would be left on my own, no phone support, no email, no contact at all. I don't want any headaches.
They talk about taking your phone unit with you on trips. Is it a simple matter to change the E911 setup when you go other places? I would likely be taking it between two locations and would want a *very* simple way to adjust the E911 setup on hooking it up. At one location it would be the most reliable means of reaching emergency services, so I want that to work.
You *do* need to know these things! Else your SIP phone won't talk to their SIP service!! :> You want to just pick up a phone and be able to dial? There's a system for that: it's called POTS!
You can call "whatever can be reached". If they only allow SIP URI's, then your destination will be expressed in a form @. E.g., a google voice phone number is @sip.voice.google.com (similar to an FTP, HTTP, HTTPS, etc. URI)
If they tell you how to configure an ATA for use with their service, then you can pretty much expect to be able to just *dial* a PSTN number (though the quality of the connection may vary greatly over the course of the day or even the course of the *call*!)
Some providers will not have bridges to the PSTN. So, if your destination can't be reached *entirely* over IP, you're SOL. Most full featured providers have provisions to get you back out to the PSTN. (People aren't patient enough to deal with the "we're sorry, the party you are calling is not on our network" crap, anymore.)
Try it and see. There are PC based SIP clients that should allow you to use your PC as a "telephone" (even if you have to resort to using a microphone and speakers instead of a real handset)
You're paying them to sell you a product that is preconfigured to talk to *their* servers! (can you later reconfigure it to use someone *else's* servers?? dunno...)
I hadn't considered having multiple units at the different locations. That could work for me. One unit set up all the time at one location and a second unit for traveling. Yes, that could work great. Can you even call yourself from one location to another or would it just ring busy? lol
My service is through a very small outfit (one guy I believe) off a private tower using a 900 MHz radio link. I"m in the center of nowhere basically and this is much better than satellite. I haven't found out who he deals with for his connections. I'd love to get involved with him. He seems to know how to make it all work, but he is no businessman and especially no salesperson. In fact just the opposite, he drives business away.
No, they have explained SIP as if I were building a system. I'm not, I'm a user... at least for now.
I don't need a dictionary, I need a supplier that knows how to talk to
*people*.
No, that's the problem, they haven't, at least until I read one of David's posts.
Again, this has no meaning to me since it doesn't fit the image in my mind. You seem to have a different picture of making phone calls. I'm guessing you are thinking in terms of some application on a PC or something similar. I'm thinking of a phone that connects to a little box that connects to the Internet.
Again, none of that means anything useful to me as a customer. If I have to do anything more than dial a phone number I don't want it.
Isn't it the job of the provider to figure all that out? Or are you saying these are potential problems my Internet access provider might create?
LOL! How do you punch the special chars into your phone?
Ok, so I assume this is *not* Future Nine?
Ok, I think this is not the company for me. I want a phone, not a gadget that I'll be constantly hacking with to keep it working.
You don't. You are limited to telephone numbers which that little box ("ATA") will translate into a SIP URI of the form +@telnum (or, if it is a dedicated/proprietary device, it may convert it to a special URI intended for use explicitly by it's own proxy server)
Yes. "Somewhere", there is a bridge located *near* (whatever that means) the destination that you are trying to reach (ideally, so it's a "local call" from that bridge to your destination). The SIP protocol (messages) eventually results in that bridge *dialing* a PSTN "phone number" on your behalf. And, connecting you to the person who picks up the receiver! (it does this using regular SIP messages to track the progress of the call/connection/disconnection)
Exactly. You can't *receive* calls! You exist in IP-land and not in PSTN-land ("But, for a small monthly fee, we will register your PSTN phone number (or, SIP URI) so that you, too, can receive calls!")
They are letting you try to find the right price/service point based on your expected needs.
Also note the fine print regarding the QoS quarantees that they may (or, likely, may NOT) apply to each of these services! E.g., you might get an inexpensive (price) service only to discover it is a *cheap* (quality) service -- high latency, one-way audio, dropped connections, etc.
It's akin to the days of WATS lines where you literally had to
*pick* how you wanted to make each particular call to get the best value.
Or, nowadays, having a "calling card" to handle certain long distance calls while other LD calls may be handled by your regular carrier (and "local" calls handled differently).
E911 is how emergency services are notified of "where to find your body". :> The phone also has to register itself (?) in its new location for inward dialing (i.e., so your provider knows how to send IP packets *to* your phone regardless of what your current IP address happens to be)
The county switched to VOIP for all their phone lines around a decade ago. Their IT people manage and maintain it, along with all their servers, routers, switches and every computer used by the county. Their phone service is excellent.
--
Anyone wanting to run for any political office in the US should have to
have a DD214, and a honorable discharge.
We do lay caregiving as one of our volunteer activities. Some of that happens in person, some over long distance phone. So we have lined up discount options because those sessions can be 1-1/2h regularly. One person has signed up for a "deal" where the landline was dropped. Since that time the discount carriers can no longer dial into that number.
Another example is at a large enterprise. They have a 1-866 service for phone conferencing. Probably also some supposedly "good deal". 1-866 should be toll-free, right? Well, one fine day I got a message "Deedle-deedle-dee ... We are sorry, this number requires a toll connection".
Some intrepid soul on the Win98 group [I think 98Guy] made all his answering machines answer with that irritating tone that your phone is no longer an active telephone line, and then would transfer on through to ring you.
The only issues I've heard of are recalcitrant land-line companies refusing the porting, or uncooperative and slo-o-o-ow.
You're in Californica... need I say more ?>:-} ...Jim Thompson
--
| James E.Thompson | mens |
| Analog Innovations | et |
| Analog/Mixed-Signal ASIC's and Discrete Systems | manus |
| San Tan Valley, AZ 85142 Skype: Contacts Only | |
| Voice:(480)460-2350 Fax: Available upon request | Brass Rat |
| E-mail Icon at http://www.analog-innovations.com | 1962 |
I love to cook with wine. Sometimes I even put it in the food.
Badly worded. It means that there are no additional charges up to the number of outgoing minutes specified in the plan. Exactly like a cell phone plan. For "Bare Essentials", that's 250 mins/month. For "America Free", it's 2000 min/month.
Sorry, but those days are long gone. Today, the phones are full of complex features that will take some expertise to use. For example, I have my Future-Nine voicemail messages delivered by email as a WAV file. I've also taken the time to try every feature available to see how it really works.
If the user has a Future-Nine account, he can login and review his call and billing history. The charges per call are itemized. Of course, that's after making the call, so the user needs to know how calls are billed. Since *ALL* calls are SIP calls, that description doesn't help. A better way is that all calls made through the service providers SIP to PSTN gateway are billed with the first 250 or 2000 minutes pre-paid depending on the plan. If you go over, you go online, add money to the account, and continue. No penalties for going over.
For international calls, additional charges are involved, depending on which route is used: For example, when I call Israel, it's an additional $0.0107/min for the slow route that usually reverts to half duplex, or $0.0154/min for the premium route.
SIP is to VoIP as SMTP is to Email.
When you call another SIP phone, or call through one of the numerous online gateways, you are NOT going through Future-Nine's system. There's no way Future-Nine can bill you for the call because the call is not going through their Asterisk box. All Future-Nine does is provide the ENUM lookup for the call. Your SIP phone logs into the Future-Nine Asterisk server (with a login and password that's different for each phone). That way, they know the IP address of the phone so that you can move around the internet. Things get a bit complexicated for ENUM lookups and such, but when it's done, the call is directed to your phone's: SIP:device-name@your-ip-address:port-number That's just a simple DNS lookup and is free. The actual call connection is made directly between your phone (or computah) and the destination SIP phone.
However, if the call is made through the Future-Nine SIP to PSTN gateway, it gets billed. If international, add some more billing.
Agreed. Incidentally, I also use several private services run by friends, which have no web page beyond the account admin page and offer zero user support. However, I can do things with them that are difficult with Future-Nine.
Incidentally, when you login to the account, there's a box for sending email to the admin. I get an answer in less than a day. Most of my email are requests for changes in service for various customers.
The alternative is one of the plug-n-play type of services. Ooma, Callcentric, 8X8, Vonage, etc will provide you with the necessary equipment and expertise, for a price, of course. I can offer recommendations based mostly on customer experiences, but I'll need to know the expected usage pattern.
Understanding the cost may be a problem, as different VoIP providers have different rate structures depending on customer type and use. You may find the pay-by-minute plan a good way to start because you can changes easily. If you get a 4-line phone like my SPA941, you can have 4 different VoIP service providers programmed into the phone (which will be 4 different phone numbers). Save one for direct access. If one is down, use another. If one is too expensive, use the cheaper one.
The requirement to understand the cost might be difficult. I still cannot reliably predict the initial and continuing costs of cable and satellite TV service. My phone bill is full of odd charges that a difficult to analyze. VoIP service is little better, unless you opt for a flat monthly free.
I had a ladyfriend that had that attitude about computers and automobiles. Her computers were constantly getting infected by viruses because she didn't want to take the time to understand how they worked so that she could make an intelligent decision on what to click. Her new automobile almost blew up when she didn't bother to change the oil for something like 10,000 miles. "I just want something that I can drive without becoming a mechanic" was her mantra. Maybe it will work for you.
--
Jeff Liebermann jeffl@cruzio.com
150 Felker St #D http://www.LearnByDestroying.com
Santa Cruz CA 95060 http://802.11junk.com
Skype: JeffLiebermann AE6KS 831-336-2558
Like I said before, not the cheapest, but works great...
First of May I changed from Centurylink (nee Qwest) to Ooma. I went Premier, so I'm effectively $12.28/month (each line, I have two, prepaid rate).
Nice features, including a personalized reject list, since the Federal DoNotCall is a farce.
Internet down? Automatically forwards to my cellphone. ...Jim Thompson
--
| James E.Thompson | mens |
| Analog Innovations | et |
| Analog/Mixed-Signal ASIC's and Discrete Systems | manus |
| San Tan Valley, AZ 85142 Skype: Contacts Only | |
| Voice:(480)460-2350 Fax: Available upon request | Brass Rat |
| E-mail Icon at http://www.analog-innovations.com | 1962 |
I love to cook with wine. Sometimes I even put it in the food.
No. E911 is a MAJOR problem. Your VoIP phone number is registered to a specific location (physical address). When you call 911, the call goes to the PSAP that handles calls for that area. If you are calling from out of the area, it still looks like it's coming from the registered location.
With Future-Nine and other VoIP providers, E911 service is ugly. To start with, the service cannot be cancelled once it's setup. That's $1/month permanently. You can change the address, but it's not easy. For example, when I login to my account, this is what I see: Instead, I have a collection alternative access numbers written on the phone and one programmed into the dial plan. Note that most PSAP's consider the direct numbers to be confidential, so you will have difficulties obtaining them. The SIP phone can trap the 911 dialing sequence, and convert it to a different number as part of the dial plan. For example, if the dial plan includes: 911S0|12345678 It will dial 12345678 if 911 is dialed. (The S0 sets the interdigit dialer delay to zero, which is a good idea if you're in a hurry). I don't think it's practical to do all this for every trip.
Note that you may also need to register your phone for the local emergency notification system. It's much the same as E911, where once registered, it's a pain to change or unregister. Here's the local page for Santa Cruz CA.
In my case, I have the same phone number ringing at 3 different locations, in 3 different jurisdictions. Which location do I use for E911 registration? Dunno.
--
Jeff Liebermann jeffl@cruzio.com
150 Felker St #D http://www.LearnByDestroying.com
Santa Cruz CA 95060 http://802.11junk.com
Skype: JeffLiebermann AE6KS 831-336-2558
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