Some of the terminology baffles me so I'll ask. I want setup a gateway on my home ADSL network. The idea is that if I'm out of town I can somehow hit my own PC/network at home from my laptop at clients or hotel room, with my speakers and microphone, and make an outgoing phone call from my own phone.
(I have the software in place so I know what IP address is at my home network. I can currently Terminal Server in quite nicely.)
Yes, yes, I know I could use Skype or whatever but I want to do this all myself. Among other things the phone number on the callerid will be my own.
Well, this is trivial if you have a voip service provider that lets you bring your own phone. You just set up your computer (or take another ATA along on the trip with you) and configure the same account-name/password into it as your home unit. Both units will ring for incoming calls and either unit (or both) can make outgoing calls and show the same caller-id number. The voip service providers that charge by the minute usually couldn't care less how many ATA's you provision for the same account. There are quite a few that charge in the 2-3 cent per minute range, so we aren't talking about a lot of money.
A list of mostly per minute providers can be found at this next URL. I use both Gefachi and Teliax. Both have no monthlies and have 2 cent minutes. There are probably others that popped up since the last time I checked the field. If you find others, please speak up.
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The voice service providers that charge you a flat-fee per month usually don't even let you look at the settings in your ATA. They have no interest in letting you program several units for the same account. Unless you make over 1000 - 2000 minutes of calls per month you'd probably want to avoid these guys anyway. The numbers for the per minute service will be much more favorable.
Now if you are asking about using your normal analog phone line via a voip call over the net, you can do that too. Some ATA's like the Sipura SPA-3000 will let you call out on your analog phone after calling into the unit via VOIP. I have one of these and that trick does work, but the quality of the connection isn't that great. I ended up just using the SPA-3000 as a souped-up answering machine (along with asterisk). I ask people to just call my voip number and ignore my analog line all together.
Oh yeah, when I'm not travelling I spend a fair bit of time in a friends coffee shop so it'd be nice to make phone calls from there instead of using my more expensive cell phone.
ATA probably stands for "analog telephone adaptor". FXO is a telco word that means "connects to a switch" in contrast to FSO which "connects to a telephone". Aren't you glad you asked. ;-)
I don't think there are any other ATA's that connect to incoming phone-line side of things (fxo). All the other ones I know of have one or to telephone interfaces (FSO's). You can try your luck googling for "ATA FXO", but I suspect all roads will lead back to the SPA-3000.
The only other possibilities I know of are running asterisk and getting a cheap fxo PCI card. The official asterisk card by itself is about the same price as the spa-3000, so you don't save any money going that route, but you do get the flexibility of using asterisk as a really tricked-out fully-programmable answering machine.
ATA probably stands for "analog telephone adaptor". FXO is a telco word that means "connects to a switch" in contrast to FXS which "connects to a telephone". Aren't you glad you asked. ;-)
You can try your luck googling for "ATA FXO", but the pickings are fairly slim. I just looked and in addition to the Sipura SPA-3000 there is now the Grandstream Handytone ATA-488.
The only other possibilities I know of are running asterisk and getting a Digium FXO PCI card. The official asterisk card by itself is about the same price as the SPA-3000, so you don't save any money going that route, but you do get the flexibility of using asterisk as a really tricked-out fully-programmable answering machine. Thats more or less what I do, but I use a SPA-3000 to keep the phone line at least one black-box away from my computer.
Ahhhh, Makes sense from what use I'd seen of those terms. Thanks muchly.
Thanks for browsing for me.
I already have some software running on my home system that does voicemail and fax using an el cheapo PCI voice modem. And it emails those to me. Besides Asterisk, from what I've been able to glean runs under Linux thus requiring it's own machine. That's something I'm not at all sure I want to get into right now.
Mind you I really like the concept of open source PBX systems and so forth. I'll have to fire these links off to my buddies who would think these kinds of things are really, really cool. (Done. Along with an editorial comment on how you could likely have a backup PBX as well as a primary PBX for the same price as the commerical systems.)
So an stand alone black box with an RJ 45 and POTS connections will do me just fine. All I'd need to do is configure my router to route the incoming packets on the port(s) to the black box.
Sorry as I am not sure did I get your meaning exactly. You want to call on SIP via internet thru laptop on software while you are out of town in something like hotel?
If so, you can try to use my net fone, as they have a software version which u can call via SIP services to any australian phone no.. I never try that as I did purchase a phone style VoIP using SIP Services.
If you only want to call back to home via internet to save the money, I can suggest you this:
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If you want further information you can check it out from them.
You have expressed an interest in not signing up with a VoIP service, but at the risk of incurring your considerable wrath, allow me to suggest you at least consider it as it may be the easiest solution.
You can sign up for prepaid pay-as-you-go service (no monthly fees), pay 2 cents a minute to USA/Canada/western Europe/developed Asia/Australia, and configure your caller ID to be whatever you like (including your home number). Then you can either use a standalone IP phone, or a soft phone on your computer, and not mess around with anything at home. You save one piece of equipment, you avoid tying up your home phone line, and you eliminate a lot of complexity.
What would you have used instead? I'm considering a similar sort of setup, along with asterisk, and wonder which units to consider. Using an SPA-3000 (or similar) would allow bypassing the VoIP setup should power go out, granted only on the analog handset attached to it but I can live with that. I'm looking to add at least two other analog 'lines' internally to use as extensions with normal telco handsets. But only the one outside POTS line and, of course, a DSL connection.
with the two phone ports feeding into a London 16 PABX, which then feeds all the phones in the house. The POTS line also connects into the Fritz!Box as a backup in case of power failure, although only the phones designated as power-fail phones will work in this case. The other advantage is that dialling rules can be set to route certain numbers such as emergency calls and premium rate prefixes via the PSTN rather than over a VoIP line where they are either not supported or more expensive. It's also possible, using the PABX, to have a single DECT handset ring for all incoming lines.
I was so underwhelmed by the quality of a voip call made via my pots line that I just gave up on the whole idea. The problem was that the volume was so low I had a hard time hearing the other side. When I bumped the gain up in the SPA-3000 the echo was quite evident and much to loud to ignore. I'd hear my own voice echo after half a second or so. The volume of the echo was roughly the same as the person talking at the other end.
I briefly thought about getting a digital line such as ISDN. That should get rid of the electrical echo sources at my end. Unfortunately, unlike Europe where ISDN often costs the same as analog (POTS), PacBell wanted quite a bit more.
(It wasn't even possible to find out exactly what they offered and how much it cost since their web site is so screwed up. "What, you want us to post a price list where people can see it??? Unthinkable!")
I now use two low-cost voip providers to essentially "host" my phone connection for me. It is a little bit more expensive to make calls to my neighbors (2cents/min), but calling even a few miles away it is cheaper than what I would be paying on the analog line. Echo also isn't much of an issue. Sometimes there is a bit of echo initially, but the echo cancelers in the phone system itself manages to tune that out after a few seconds of training.
Are the two ports independent? Can each one have it's own conversation going (assuming a route is available, of course). Can they share the same conversation (conference call)?
I don't see a US supplier listed for this product.
So you had two main problems, volume and latency. The volume question would certainly be dependent on the quality of the ATA in the SPA-3000. The echo might be more affected by available network bandwidth and it's latency. Having a line with enough bandwidth is important but it has to be a responsive link as well as 'fast enough'. I'd wonder if your setup introduced more latency than it could overcome?
Here's the thing, right now I can depend on the ILEC to provide a consistent level of performance. That and since I'm on a shared DSL line (with POTS) I'm going to have the analog connection anyway. Since I'm paying for it I might as well make use of it.
I'm not sure how often I'd bother using the POTS for VoIP redirects. Being 'able' to use it effectively would be nice, but not unless the echo and volume were at reasonable levels. I'd certainly want to use at least one of the inside handsets as a direct-out POTS connection should power go out, and most external (non-PCI) FXS/FXO ATA devices seem to support this capability. The question remains, which one 'sucks the least'?
Yes they are completely independent and can be set up for different providers. The unit contains a PBX so theoretically I think you could link the two ports in a conference call, but having a PABX already I've never tried it.
It's still fairly new, I don't think it's being aggressively marketed outside of Germany, even in the UK there are only a couple of suppliers.
I could make enquiries for you if you want, but I'd need to know a little more about the specs for ADSL on your line.
That's good to know. I'm not sure we've got the need to use a PBX here. I've dealt with them before (spec'ing, setup and management) so I'm well aware of their benefits. But for the handful of extensions we'd use here at the house it'd be overkill. Besides, I'd be more likely to go with smart VoIP phones like Cisco (to get the graphic display features).
Does it plug directly into the ADSL line? I was under the impression it'd use a regular wired ethernet line from whatever connection is already active (cable, xDSL, whatever). I ask this as there's an outside chance we might switch to using the local telco's fiber service. So being capable of using the ADSL line is a nice feature, I wouldn't want to get stuck with it. Dunno what our ADSL line uses other than to say Westell and Zyxel ADSL modems are interchangeable on it.
My theory is that the volume is low because Sipura knows they will have an echo problem if they bump the volume to be normal phone volume. The way to prevent an echo (which is mathematically just a dampened oscillation) is to cut the gain of the loop. That is exactly what they did and why there isn't any audible echo with the factory settings. The volume at factory settings is quite inferior to the volume one would get on a normal analog POTS phone.
From the reading I've done, echo is a major problem in the analog phone system. The only reason it isn't annoying normally is that the ear is very good about removing echo if it occurs within a very short time of the primary signal and with a signal-volume below a certain level. The longer the echo delay the lower the volume has to be for the ear to remove the echo. (Talking in a room with reflective walls would be very annoying if the ear wouldn't automatically filter the
10ms - ~100ms echos one normally gets from walls that are 5ft to 50ft away.)
The normal phone system simply punts on the echo removal for local (short echo delay) calls, assuming that the user's ear will be able to sort the matter out even if echo volume is quite high. Thats all nice and good until one adds this echo to the packetization delays (20ms at least for voip codecs) and jitter buffer delays (in the 100ms+ range). At that point the ear refuses to filter the echos since they are too loud for that long a delay.
Setting the jitter buffer to lower delay values will certainly cut the delay, but it also cuts how much jitter the unit can mask. I opted to cut the jitter buffer to the lowest value. I don't think that has caused any problems.
The phone call that "broke the camels back" and caused me to stop using the spa3k for anything but answering machine use was an LD call from someone behind an older PBX in the Washington DC area calling me in the San Francisco area on my residential POTS line. The echo was clearly far-end echo from some mismatch at their PBX. Calls from LD POTS users were usually a bit better, but still far from perfect.
Internally everything here runs to a cheap gigabit switch that ties the other voip phones to the spa3k. I don't expect the 5 voip boxes each connected to the switch with a 10Mbit/sec NIC at their end to be able to saturate the switch's switch fabric.
I'm forced to have the POTS line too. I just called and had all the extra cost features removed and had it changed to per-minute billing. I just think of it as part of the cost of getting ADSL. I will of course keep this in mind when comparing the total ADSL price to cable.
I encourage you to give it a try. You may have better luck than I did due to better external wiring to the CO etc. I can't remember if my initial tests were with the pots on the 14kft loop to the CO or if it was already with the 1kft loop to the local RT.
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