We've gone VoIP here finally, over a "dry DSL" line, using voip.ms as the connection/DID provider (hey, they even have a UK PoP!).

Cheap and a geek's dream. I'm using a PAP2T-NA to adapt all of our existing analog telephones to the VoIP connection, and this also works extremely well.

We've got a couple of DECT handsets here, and really like the feature set on them. But it would be cool if the analog conversion could be skipped somehow. This requires a DECT base unit that has ethernet/IP in/out.

Siemens seems to make them, but apparently not for North America yet. They're available in the UK and Europe, though, and possibly even further afield.

Anyone got one? Know anything more about them?

Thanks

I may need to look into this further. By that I mean, you may have to hand-hold me through a lot of questions I may ask in the future based on your experience with this service.

Right now I'm on a package plan from a voip provider for $20 US per month which gives me unlimited Canada and US. I have an ATA to which I have my home's internal phone network connected. Network being a fancy name for the phone wiring that's built into the house.

If based on my low phone usage I can save some money going this wholesale route it's something I'm going to seriously consider. The ability to set up multiple lines, PBX, etc.. sounds interesting as well. One of the things I'll have to look into more closely is the difficulty/ease of porting my existing 416 number and making sure all the basic calling features (voicemail, call waiting, display, etc..) are all working.

Mark,

I use a Siemens C475 IP for incoming VOIP calls for my business. I have 3 hands sets in total. I have to say they work really really well, the audio quality is excellent and so far I have had no issues at all. My VOIP provider even gave me step by step instructions to get them working, but it wasn't too difficult.

Basically my system can handle 3 calls at once, 2 VIOP and one POTS. In general operation they are a little clunky and slow sometimes, but you can sync you address book between hands sets and there are other pretty cool features.

I would recommend them. I would be happy to forward one onto you if you can't find a source elsewhere.

On a side to the handset question, generally I find VOIP brilliant. My provider supports useful features like if I don't answer my VOIP line then all calls are transferred to my mobile so I never miss any business. I got to control all this via a nice easy web interface which I find very powerful. There are also lots of cool options on the handsets themselves of course.

Cheers

Cris.

Do you know why that phone needs specifically a DSL internet connection? As opposed to any other high speed internet connection...

It'll work with any kind of ethernet connection to the internet. I think the "DSL required" is just a marketing thing, to ease the minds of those who don't know of other options.

EDIT: is CATV internet access even an option in the UK?

Cheers

Perfect. I'll read up on that model (just so I can figure out the right questions to ask). But just to clarify:

This is a DECT system, right?

And I may someday take you up on the purchase/remail offer. Really appreciated, that!

Thanks!

Absolutely. After all, that's what (geeky) friends are for!

Note that even "ordinary" analog DECT-6.0 cordless phones are fairly impressive to use with a full-featured VoIP solution. I'm just wondering if even better voice quality can be had with a pure digital end-to-end solution. Thus the interest in the Siemens IP-enabled DECT systems.

The tricky question is.. does the base unit have to transcode from one digital codec for DECT to another for VoIP? EDIT: no, it doesn't need to transcode!!

Which would kinda negate some of the edge it might otherwise have.

Anyone know off-hand what codec the Siemens DECT phones use? EDIT: Lots of them, including G711 u/a law for VoIP!

Cheers

VoIP.ms provides all of the basic/fancy calling features gratis, except caller-name display which costs USD$0.015/lookup. The included (free) services include caller-number display, voicemail, with handset, web, and email access to recorded messages.

Porting numbers to their service apparently now takes only a couple of weeks, down from 6-weeks last fall. We didn't bother porting, as we wanted to ditch the telemarketers in the move.

Cheers

Chris, can this handset system be tied to any arbitrary SIP provider, or is it somehow locked to only those listed in the technical data part of that link?

Oooh.. the user manual is there.. and it lists codecs.. including G.711 with a *choice* of a-law and u-law, which means it *will* work for us over here! And no transcoding required in most cases!

Thanks

Oh, another semi-important question for Chris:

On the units you have, could you describe the power supplies?

Cord/plug style, voltage + frequency ratings.

Thanks!

Do they charge for caller id for ever call, or just if you want the caller's name?

I use Vitelity which has similar pricing (I didn't actually compare, it's too cheap to be worth switching) and has provided adequate service. Every few months I reload the phone with $40 or whatever their minimum charge is, and don't worry about it until they start emailing me.

Yeah, same idea here with voip.ms. The caller-name lookup fee of $0.015 is charged per-call, if enabled on the account (ordinary calling-number display is *free*). Apparently that's close to what it costs them to do the lookup, so they just pass it on.

They do say they plan to implement caching and stuff at some point, which would reduce those charges. I'm not counting on that actually happening, but.. whatever.

I chose voip.ms because they're Montreal-based, with PoP in Montreal and Toronto, plus the UK and a bunch of USA locations. Two of those are relatively close to Ottawa (here) as far as pings go.

E911 setup was also a breeze. For us Canucks, that's an extra USD$1.50/month drawn from the prepaid funds.

Cheers

Yes this is a DECT system, it handles normal analogue calls too. You can select which service you want to call out on by the time you hold the dial button for, it's switchable so for example if I just press dial I dial out on the analogue but if I press and hold it uses the VOIP service, like I said you can swap that if you wish.

I am not at home right now to check the PSU, it's a wall wart type I know that, but will have to check on the unit for specs.

Setup is easy, and there is the option to select your VOIP SP from a list for auto setup, but you can also enter everything manually too.

I am wondering about DECT from Europe working in Canada, I think we use a different set of frequencies over here ??? Could be wrong.

Cheers

Cris.

Yes, we have a cable operator here, Virgin Media. They only cover about 50% of the population and are no longer investing in their network. If you don't already have coverage you have no chance of getting it at the moment.

Cheers

Cris.

Ahh.. good point.

And some tart went and put digital mobile telephone services on 1850-1910 and 1930-1990 MHz over here. So that scrubs the Siemens for now, until they produce a USA/Canada version.

Thanks!

Mmm.. looks like they've just done so (Jan/2009)!

Good news

They are nice handsets, I agree with most reviews that the sound quality is very good indeed.

Cheers

Cris.

You don't want to go stomping all over someone else's bandwidth? I'm sure the CRTC won't track you down.

Ha! Actually, my mobile phone here uses those exact frequencies, so I'd be stomping on myself!

Correction: that's USD$0.015 per incoming call, and only if one enables "caller-name" lookups (CNAM). Caller-number ID is *free*.

-ml

Sounds good. Anyone know if the Siemens phones will do number matching to its internal addresss/phone book? So that even without CNAM enabled, one can see the details of an incoming call if they match an entry in the phone book.

And if so, does anyone know if it will match on numbers both with or without a leading "1" ?

Mark, any details on the dialing format for your new provider? How many numbers are required to call out? For instance, with my provider, since I have a 416 number, I dial 7 digits for all 416 numbers (omit area code) and dial 11 digits (1 + area codes) for everything else, regardless of where it is. The biggest drawback to this with my provider is that caller ID always comes in as 10 digits for non-416 numbers.

The company that I'm using right now, voip.ms , uses 10-digit dialing, with or without the 1- in front of the 10-digits. They don't care if the 1- is there or not (a Good Thing).

But.. my Linksys PAP2T ATA has user-programmable dialing plans, so I've set up that device for 7-digit local dialing (or 10 or 11 digits: all of those work), and lots of other cool things. Using this "dialing plan" string:

S:10,( <:613>[2-9][2-9]xxxxxS0 | 1900! | 900! | 1.[2-9]xx[2-9]xxxxxxS0 | 18xxxxxxxxxS0 | [34689]11S0 | 011xxxxxx.S5 | 09730021S0 | *xxS0 )

Rather cool, that. Just replace 613 with your local area code (416) to customize for the GTA.

The PAP2T has tons of other cool telephony features too, like CID (Caller-ID) based forwarding, blocking, and/or ring-tones.

Cheers

Oh, I should mention that the dialing plan is almost *always* a function of the ATA, not the VoIP provider.

I use a fully "unlocked" PAP2T-NA ($52 from CanadaComputers), as opposed to the common vendor-locked units.

The Siemens DECT/IP phones also seem to have user-programmable dialing plans.

cheers

Interesting bit of info. I'm not sure if my Mediatrix device is locked, but it was indeed provided by the VoIP vendor.

What I'm thinking of doing in the next couple of days is try to find out what my usage has been for the past few months and then cost out the same usage with voip.ms rates. That will give me as clear a picture as I'm likely to get as to the monthly base-line financial benefits of switching. I already know I'll be able to do all sorts of extra cool things I can't do now, such as get additional "lines" that I can run concurrently.

Does voip.ms handle storage of voice messaging on their own servers or is that something you have to provision yourself? I noticed the Siemens phones had messaging support built-in, but I'm assuming it's the equivalent of an answering machine (being local only and therefore not necessarily accessible via another telephone or the net).

It's their server entirely. I've configured things so that voice messages are also emailed to me, just in case their server loses them later. I don't know much about their voicemail server -- it's not exactly boldly advertised -- other than that it appears to work.

Messages remain on their server (even if also emailed) until deleted from the handset (or presumably from the web interface which I have yet to explore).

There's gotta be a downside somewhere, but I haven't found it yet. Some people complain about the Customer Service reps being in Mexico when one dials in to them, but the online chat reps have french names..

There are also a few complaints about slow porting of existing DIDs, but I think that's probably a rampant industry-wide issue here in Canada.

Note that your incoming-calls (DID) provider doesn't have to be the same as your outgoing call server..

Cheers

Ahh.. Okay, I was wrong about the web-interface for voicemail: no such thing at voip.ms. But with email and (local/remote) handset access, that's not really an issue.

Cheers

about two months ago I also started with the voip conversion. I'm sick of my household spending over $400/month in monthly phone bills for 3 people. it's amazing what a couple bucks can purchase on the internet. I can actually purchase something for a millionth of a cent now through flowroute.com

I currently have an Snom M3 that I wouldn't recommend to anyone. The buttons are completely flat, tiny and have absolutely no spacing between them or any tactile feedback. You pretty much need to use a stylus to use the keypad without pressing the surrounding 3 keys at once. Also about every 1-50 calls the handset will completely lockup and reboot when I try to answer the call (causing it to drop the call). However, it is a DECT base but I'm pretty sure it won't talk to anything besides Snom headsets. unless you're planning a new port of hijack

I'd also think about setting up something like freeswitch.org to handle the routing of your calls, I wouldn't rely on just the PAP2 for all your telephony needs.

I have a "naked" ADSL service with Internode here in Aus, I picked up a Linksys SPA941 desk phone off ebay very cheaply, and it's pretty good, although it doesn't have an illuminated LCD which is annoying, the SPA942 does though.

Internode give you a "free" account with an incoming number and $10 credit, and the service is integrated within their own network, so they advertise it as being high quality.

I don't know about anyone else, but that's exactly what I'd like to do. I didn't really know about any other free platforms other than Asterisk, but then again I also don't know ****-all about setting either one up. I thought I'd worry about it until after I decide to take the plunge with a new wholesale provider.

Heh.. I had to read through quite a bit of stiff marketing content before "Australia" was mentioned. So Internode isn't really for us here in Canada.

Cool bit about freeswitch.org. But I can't see a use for it in my home though, as the combo of my current provider and the PAP2T already far exceeds our requirements.

Perhaps some day when I decide to build a "do not accept calls from" list of more than eight number patterns (limit of the PAP2T). Then it might come in handy!

Cheers

Speaking of the PAP2T -- the current Cisco (they own Linksys) documentation for it is total crap. For the unlocked version, one needs the "Administrator's Guide", and the Cisco version is rather generic, covering several of their products under a single title.

But I did eventually locate a copy of the original Linksys PAP2(T) admin guide, and that's just so much better! Between it and wikipedia, I can now figure out what all of the settings are used for.

Mind you, it practically worked without a hitch straight from the box, after I entered the SIP user/passwd and turned on NAT traversal. But the extra fun stuff just tends to draw one in..

Cheers

Can I ask which provider you are using ?

Thanks
Mark

I just used my Nokia N800 to make a PSTN call

Denise setup a local Asterisk SIP server using Sein's gear and connected it to the PSTN line a couple of weeks back. We've been using headsets on the PC and normal analogue phones with a PAP2T.

It was fun having extensions and setting up local voicemail to be stored on the IMAP server.

Cool!

Glad you're having fun with it.

Sure, I use VOIPTalk I have an incoming freephone number with them.

Cheers

Cris.

Thank you.

Progress report.

The Bell telephone line got disconnected this past Monday, and as a result our "dry DSL" loop showed an immediate improvement in S/N characteristics. It's been quite rock solid since then.

Which bodes well for our VoIP telephone service that relies on the DSL now.

We've got E911 set up on the account now (painless), so recurring charges tally in at US$3/month + calling fees, plus the CDN$9.10/month for the dry loop itself. Internet access is CDN$34/month w/static IP.

The Siemens DECT/IP sets are not yet widely available over here, so I've gone and added three handsets (total of five now) to our existing conventional DECT 6.0 Vtech system. Really, it could hardly be any better suited. It has dedicated voice-mail access buttons and indicators, and no local answering machine. Perfect for use with voip.ms!

Cheers

Mark, what ISP are you with? And what are your down/up published rates and monthly bandwidth allowance?

I just hooked my parents up with TekSavvy for both a line and DSL. Didn't go voip, so it's a full-blown line. I'd love to save them some money, but I'm not sure if I can rely on voip for them.

We get internet from ncf.ca (National Capital Freenet), who resell Teksavvy at pretty much the same rates as Teksavvy themselves. So, really, we're with Teksavvy, just indirectly. No complaints whatsoever there. Bandwidth allowance is 200GB monthly, with explicit permission to host servers and share the connection with neighbours.

Data rates are for up to 5mb/s down and 800kb/s up. But we are a very long way from the C.O. (5.4km, I believe), so we're lucky to have DSL at all really. Our line is currently configured with a 2496/640 profile (2.4mb/s down, 640kb/s up), interleaved.

This has worked very well for us, with a sync loss once or thrice a day over the past few years (S/N ratio of 6dB or so). But when the Bell line got removed on Monday, the dry loop suddenly became a lot more stable (S/N ratio is now 13dB). It could easily do fast-mode 3mb/s down now, but perhaps with the odd sync loss here and there again. I'm trying to decide whether to ask for a slightly faster line profile, or to leave it alone in its new rock-solid state.

We're also thinking of setting up voip.ms for the mother-in-law in Toronto, perhaps. She already has Teksavvy DSL (and excellent line quality, 5.5mb/sec), but would need a new dry DSL loop installed. Then have the DSL service moved to that from the Bell "wet loop" currently in use, transfer the DID, and disconnect the Bell loop finally.

Trickiest bit would be getting her Bell DID transfered over.. it's a nice number, and she'd like to keep it. Piece of cake if we actually lived in Toronto, where we could just handle it all for her. But from Ottawa.. dunno.

EDIT: Oh, wait. She also has a telephone-based alarm system there. Have to check if it has a dedicated line, or if it shared the house telephone line.

Cheers

One cool trick I discovered today, was a setting in the PAP2T unit to have it use euro-style ETSI FSK signalling for the CID functions on the handsets.

Normally over here, CID info gets sent to the handset between the first and second rings, but with the ETSI method it is sent immediately before the first ring. Much better for screening calls that way. The DECT sets have no issues with it (as expected), and neither does another CID-capable handset we were using.

-ml

I think you are quite brave going for a dry loop. This isn't something widely available yet here, but it could be just around the corner.

In the UK I wouldn't recommend that to anyone, retain dial tone on your line. I know the state of our network in the UK inside out, and with pairs at a premium the last thing you want is to not have dial tone. Before you know it "someone" will pinch it away as engineering techniques in general haven't moved on for 50 years and most don't think to check for a digital signal before assuming it's a spare.

Of course this is against company policy, but what would you do yourself? I bet a high percentage wouldn't be following the company line!

I hope practices in Canada are a little more advanced than here.

Cheers

Cris.

They have been doing it in Australia for maybe 6 or 12 months now. It's called "Naked ADSL" here. I've not heard of anything like what you're suggesting Cris where your pair would get taken away from you.

It is a similar system though I'd guess and the original issue was always you had to have a POTS service before you could get ADSL so that the line stayed allocated.

In my case to actually go to the naked service it would actually cost me more and get me less since I have a pretty good deal already. Plus the phone call rates I get are quite reasonable for the odd local or international call (international ones are very good actually I think).

We had an SDSL line put in years ago, and when the engineer was looking for a spare pair on our trunk to put it on he inadvertently pulled a KiloStream to a branch office out because he couldn't hear a dial tone on the line. Needless to say, our IT director tore strips out of his manager.

We've had "dry DSL" (as opposed to "wet DSL") offered locally for at least the past 2.5 years. Should have switched to it sooner, but I think waiting helped a bit while things stabilized around the engineering/deployment side of it.

Cheers

Yep, that is exactly what I am talking about. Openreach's (BT) continued under investment in it's staff training and lack of discipline within the engineering community will only make this worse over time. The worst part of it is that it really doesn't matter which SP you go with as they all have to use the Openreach Network for delivery.

There is another side to this in the UK. If it truly is a "dry" line, ie missing even the 50v DC wetting then on the network in the UK that isn't going to last too long fault free.

I think my point really only applies to the UK, as our network is so old and over stretched, along with very unhappy engineers, short cuts are always taken to deliver new service.

I can say all of this of course now I'm not part of that problem anymore. I would point out that I was never a "pair pincher"

Cheers

Cris.

Heh, hardly! It's a real eye-opener touring around the world, and discovering that backwoods Canada really is backwoods in so many many ways.

The installer guy shows up, with a belt hung line test unit. If he wants to find an unused pair, he selects a likely candidate from the bundle, and checks for existing signals with the line test unit. Thankfully, it will see DSL as well as voice equipment, so things should be safe enough. Time will tell.

You are one step ahead of us here then!

The equipment currently widely in use here pre dates the digital era, whilst most engineers will carry a a test unit capable of detecting a digital signal (they are the blue ones you see) most, myself included, used to hide the old yellow equipment come inspection time as they are a whole lot better to use. I used to use an amplifier that held up to the pair would detect the digital signal before I broke the pair down. Now try showing that to someone who has always waded in and cut a pair down to test it.

I must admit, the company bought me £1200 worth of digital tester for finding faults, but it was so poor to use I ended up using my old trusty £50 needle multimeter. f you know what to look for you can even get distance to fault off one of those things, and experience tells you what a DSL DSLAM looks like on an analogue scale. And again most engineers don't have the training backup to use the equipment so it just sits in the back of the van will the batteries leaking all over the place. Management are only interested in the quickest solution to the problem in hand. So hence my advice, for the UK at least, to keep your dialtone even if it's out going calls barred or something.

Ah them were the days.

Cheers

Cris.

I remember the days when the test units didn't detect DSL, and lots of DSL customers were getting disconnected because a tech thought he'd found a free pair he could use...

No, back in those days the DSL lines also always had voice service. So the tech probably just didn't check at all.

A customer of mine lost their DSL connection while I happened to be onsite recently. I noticed that a BT engineer was working in the building's cabinet and I mentioned to him that the downstairs company had lost their broadband. He insisted that nothing he was doing could be responsible and offered to show me his test meter thingamajig as proof!
Suffice it to say that when my client's broadband was restored a few days later upstairs lost theirs (or their fax -- they lost *something BT wise). I sortof *knew* they were stealing pairs and I now feel vindicated that you guys (Chris and Andy) are confirming that!

Another thing while I'm thinking about BT... I see a lot of cases that go like this, the fault is: DSL present on line (according to router) but no Internet, no "bad logon" reported by router:
1. Customer reports fault to their IT provider (my company)
2. We we report problem to BT reseller
3. They check for DSL logins etc
4. We check (and sometimes replace) router
5. Reseller reports fault to BT and BT supposedly test the line as NFF
6. BT threaten a fee for a wasted site visit if we make them attend
7. BT attends site, line NFF
8. Customer's DSL starts working
9. No charge.

It's happening more and more these days. So much so that we're starting to 'pretend' we've replaced the router etc. until BT have been to site. I know that's wrong but over 95% of the time we get a NFF and the DSL starts working again.

Purely OOC, any idea what's actually going on behind the scenes here?

That sounds kind of like what I'm going through with my Dad's ADSL line at the moment. It still has sync but appears to be failing to authenticate. The ISP (Plusnet) also can't authenticate with his details on their test line and yet BT have still been sitting on the fault seemingly doing nothing for the last three days.

I doesn't help that I couldn't talk my Dad through using the BT test login. I don't suppose there is a router out there with a web page that has a simple "switch to BT test login and then switch back to my login" option ?

I suppose I should defend slightly what the guys are doing, I would say in the vast majority of times the engineer is not cutting off one customer to feed another, at least not in his eyes. He sees that as a spare and uses it, quite often his system will be telling him it is spare but the records are wrong. This doesn't excuse poor engineering practices of course.

Sounds like your common problem is the card locking in the exchange, the very fact that they remote test it will likely free it up again and bingo it starts working. Do you find it happens on the same brand of router you are using? If so it could be some kind of conflict with certain cards, and that is what is causing the locks? That is the sort of thing we used to see back in the early days where people didn't like using the BT supplied router and put their own on there.

Cheers

Cris.

Thanks for the explanation Chris.


We mostly use Zyxel routers such as the Zyxel P-660H model so by default, the problems we see are pretty much all with one brand of router. I must admit I hadn't given any thought to the possibility of bugs/lockups etc. Do BT publish details of 3rd party routers known to have compatibility problems etc. that you are aware of?

Zyxel have totally stupid default settings on their routers. SNMP, telnet, FTP and web to the router are all set to be allowed from the outside.

Not that I have any experience of this but with Zyxel routers you can configure them by issuing command lines over telnet. I suppose a clever person could script something to automate the switch over.

Alternatively, you could post him a cheap (2nd user?) router pre-configured with the BT test login.

I actually used to use a script just like that to switch my Efficient Networks (BT supplied) router to the BT login and back at 6am every morning. In the early days of BT ADSL it was the only way to ensure a reliable line. Thankfully things have improved at lot since.

Unfortunately while my Dad's Zyxel router has telnet access it is telnet access into a menu system, making it pretty much non scriptable.



Hmmm, good point. I have two old BT EN5861 routers sat here doing nothing. Though of course you play russian roulette every time you power cycle them, their fragile PSUs only last so long.

However I fear that having my Dad unplug one router and then plug another in could well end up with him not having everything plugged together correctly afterwards.

I know they used to, at least internally. But I think they stopped doing this a long time ago once ADSL routers flooded in from all sorts of places. My knowledge of routers that caused problems used to be pretty good, but it is getting less and less relevant now.

I would suggest that next time you have a problems like this, preferably one that repeats, to replace the router with a different brand before reporting to BT and see if the problem goes away. If it does it could be likely that you have an incompatibility going on.

As for the old BT routers blowing up all the time, that kept me fed and watered and in good overtime for a couple of years, loved those jobs

Cheers

Cris.

Over here, where the loops are long and heavily attenuated, we swear by Thomson SpeedTouch xDSL gear. The cheap/simple ST516 in particular is rather legendary for its performance.

Cheers

Yeah, I used expect for my previous script. It would however be a nightmare doing it for the menu based ui on my dads router. I'd be much better off swapping it with one with a simple web ui.

I've written scripts in Python to handle those menus before. My current modem is a Zyxel that does have a CLI though.

I'm sure it is possible, but in the time it would take me to do it could earn far in excess of the cost of a new router with a simple web interface that he could actually use to swap account detail

If I remember rightly my Dad's Zyxel also has a CLI, but you can't access any of the ADSL or firewall settings from it, doh !

He could do with a new router anyway. He currently has a router, hub and WAP, the last two of which he only actually needs when we go to visit.

And after years of working with DSL everyday I would have to agree. The Thomson routers are by far the best for long line performance. For example I get an extra 1Mbps over any other modem, and it hardly ever drops sync. Even when my cctv PSU was failing recently and chucking out loads of REIN it still kept sync.

Cheers

Cris.

I just came to the realization that my cordless Uniden phones operating at 5.8GHz completely trample my 5GHz 802.11n WiFi signal.

I've been on hold with some company now for 10 minutes or so ad noticed that I was getting really long load times on some webpages. Then the browser timing out. Even just connecting to my base station via Airport Utility or my Linksys router (acting as QoS gateway only) was taking forever.

Switch over to the 2.4GHz network and poof. Everything's going relatively smooth again.

I just hung up the phone and switched back to the 5GHz network and confirmed, everything flies.

When are those Siemens DECT handsets going to be available again?

It could be some time yet, I think.

And they'll be very expensive initially too, I imagine.
So for a stop-gap measure, that could equally be a permanent measure, go to Costco and pick up a 3, 4, or 5 handset DECT 6.0 system (non IP) now for a touch more than $100 or so.

Cheers

Look like I'll be moving to voip.ms as well. Lingo, while they have been decent, are currently unable to update my CC info. That tied with them disconnecting me without notice last year when my CC expired - I'm partially to blame, but...voice quality has gotten worse, lots of echo currently, and coupled with their parent declaring bankruptcy last month I'm going to walk.

I gather all I need to do is pickup a PAP2T-NA, register with voip.ms, configure and all will be better?
Are there any other ATAs I should consider? This would just before typically home usage with a run of the mill cordless phone.
Off to voip.ms for further reading.

The PAP2T is the first gen ATA from Linksys. I am (still) *very* happy with it, and with the ability to totally customize it (without needing to customize it).

But it lacks T.38 FAX support, so if you want slightly better odds of a fax machine working, then look for one with that feature. The very similar SP2102 has T.38. There are others, too.

If your internet arrives over DSL, then consider the SpeedTouch 780wl DSL modem / wireless / VoIP unit. It can probably do a really good job of QoS without fuss, since it "knows" about the underlying ATM transport. Plus, it's a *Speedtouch*, which means it can work wonders on an otherwise flaky DSL loop.

Internet is shiny fiber from Verizon FIOS, into a Linksys router running WRT firmware.
I'll read up on the SP2102.

Before considering VoIP.ms as the provider, first look into how close their servers are to your particular locale. This probably isn't a big deal, but some people do care about it.

They have access servers in Montreal, Toronto, New York, Houston, Dallas, Los Angeles, and London (UK). If any of those are "internet close" to you (extremely likely), then there's unlikely to be an issue with it. They say their main servers are the Dallas and Houston ones.

Ping times can be an indication, but a traceroute will really tell the story. Eg.

Code:

traceroute to sip.ca2.voip.ms (24.102.60.67), 30 hops max, 40 byte packets
 1  zippy.localnet (10.0.0.2)  0.328 ms (my local firewall)
 2  lo-100.erx05.tor.packetflow.ca (206.248.154.105)  54.735 ms
 3  2120.ae0.bdr02.tor.packetflow.ca (69.196.136.66)  54.803 ms
 4  gw-mountaincable.torontointernetxchange.net (198.32.245.17)  58.049 ms
 5  colo4500-to-core-main.bgp.mountaincable.net (24.102.5.13)  60.621 ms
 6  cn45-core.mountaincable.net (24.102.5.18)  61.529 ms
 7  ca2.servnet.ca (24.102.60.67)  66.465 ms

Note that our DSL line is "interleaved", which means those times are 35ms longer than they would be on a regular "fast mode" DSL line.

-ml

Lucky bugger.

Some of the best reading at voip.ms is available only after login. So create an account (free!) and login and poke around some more.

If you like it, then deposit the $25 minimum via Paypal and grab a DID or two. Actually, request several, and then return the ones that aren't catchy numbers. It's only $2-3 or so per DID to do that.

And there are lots of cool tricks you can play with extra DIDs..

Cheers

More reading: voip.ms discussion forum.

Ordered a PAP2T from Amazon on Saturday, with standard shipping it arrived on Tuesday - apparently there is an Amazon warehouse just over the MA/NH border.
Upgraded the firmware, painless as long as you know the IP which can be gotten via phone. Getting the IP via phone is about all the Linksys manual is good for. I pulled the IP from my router's DHCP client list.
Configuring the PAP2T isn't simple, there are many many options, the vast majority of which can be ignored. The more important setting are under Line1 or 2 and Mark has posted a PDF of his settings in the voip.ms discussion forum linked above.

voip.ms is very advanced user friendly, lots of things to play with and tweak, unlike Lingo - this is both good and bad, but I'm finding it interesting to peak into.
I do like that I can select Premium or Basic call quality and cost settings independently for International and USA/Canada calls, as Premium quality calls to Irish landlines are actually cheaper then Basic quality.

Will have to get SWMBO to make some calls to see how it sounds to her before porting our number - unfortunately we only have one handset in the house, so it may be a while before we fully move over.

Yup, I know I am a lucky bugger to have FIOS. We've had it for the past 3 years, I think our town was a test town. We get solid 3mbit down and 700kbit up.
I did tracert to their four servers. While NY is geographically and one hop closer, I was getting timeouts on the 9th of 11 hops. So I went with their us1 server.

Well it only took 8 months, but I finally got around to doing something about this, the call quality has gotten too bad with Lingo, plugged in our only phone and did a test call and verified the issue was with Lingo.

So I got the ball rolling by ordering a new cordless phone from Amazon to replace our aged one and let us have both phone lines running so that we'd be able to test voip.ms and have a seamless change over when the time came, and then started reading the 90 page thread that Mark links to above.
About 20 pages into that someone suggests using grabbing a temporary number and using Call Forwarding while transferring from ones old provider - a serious light bulb moment for me, and I checked to see if Lingo offered call forwarding- they did.

Within a few minutes Iwas able to move everything over. I'd ordered and received a new number, I skipped getting a number from my local exchange for one a few towns away that ended in 1984, this was processed and available instantly. A few clicks on the Lingo site and incoming calls were forwarded to the new number. A few more clicks on the voip.ms site, and our out going calls on the voip.ms network show our Lingo phone number. Finishing up with a quick test call to my cell phone to verify things were working correctly. All that is left to do is to have our number ported over.

Initial feed back from SWMBO on sound quality is positive. I'll be calling Ireland tomorrow so we'll see how that goes.

One of the great features that we'll be able to use is CallerID Filtering. While the Do Not Call Registry has done a decent job of reducing the amount of telemarketers we get, we still get some, eg we dropped DirectTV last year,so they call legally call us for I think 18 months to woo us back, but almost always from the same number, that number will soon begetting routed to "I'm sorry, the number you have dialed is not longer in service".

Feature wise, voip.ms has more knobs and dials to play with then any home phone system should, and a lot more then I'll ever need. If you can think it, they probably already offer it.

Now if only I could get a similar deal for our cell phones.

Great.

And here, no change from before. Except we now have two DSL "dry loops", each at 3008/800, bonded with MLPPP to give us decent (nowhere near FIOS) speeds (6000/1600).

Still with voip.ms, and see no reason to look elsewhere.

Cheers

Mark, should I do QoS on my router (Asus 520 running Tomato) given my internet speed (FiOS 10/2) and don't typically don't consume the bandwidth? If so can you point me at a reasonable how-to? I've googled many, but I'm no a good judge. David

Yeah, should should set up the QoS, at least for VoIP. A more elaborate config would speed up web browsing, too.

For just VoIP, you need to go to QoS->Basic and enter your max outbound bandwidth (measure it on a speedtest site, and enter the result there), and then your max inbound bandwidth (same thing). Since our connection here is dodgy, I fudged the numbers to be somewhat lower than the true max throughput.

Then, on the same screen, set up the traffic classes (Highest, High, Medium, Low, and Lowest) with whatever percentages you feel like. I'll attach a screenshot of mine here shortly.

Then, on the QoS->Classification screen, just create a rule for your PAP2T to place all traffic from it into the "Highest" class. Then add other lower priority classes to suit for other stuff.

Screenshots coming shortly.

Thanks Mark, the screenshots make it very easy to understand.

Yeah, except there's no screen for setting INBOUND classifications.. duh. I wonder where that went to?

You'd need to set that up on the ISPs router at the other end.

I know that.

But there is a way to gain some of that effect with back-pressure,
which is how Linux (and thereby Tomato) does it.

Just kind of odd they don't provide settings for it.
Especially since it's already half there on the QoS->Classification page.

Cheers