voip.ms DID Sale ONT/QUE

[death_hawk]

Just got this email from voip.ms


New Reduced Rates on Canadian DIDs

Dear Customer,

For a limited time, VoIP.ms is offering permanent savings of 33% on the monthly fee for DIDs in 120 Canadian Rate Centers.

All the rate centers normally listed at $1.49 monthly fee (on per minute plan) are temporarily lowered to only $0.99 per month. Please note that if you order during the promotion time, your DID will retain the $0.99 monthly fee permanently.

This promotion also applies for new number porting orders of the listed rate centers. Ported-in numbers will be activated with the $0.99 monthly fee.

The offer applies for the following rate centers:

Quebec:
Aylmer, Beauharnois, Beloeil, Boucherville, Chambly, Chateauguay, Chomedey, Coteau du Lac, Cowansville, Gatineau, Granby, Hudson, Hull, Ile Perrot, Lachine, Laprairie, Laval East, Laval Ouest, Les Cedres, Longueuil, Mascouche, Montreal, Pointe-Claire, Pont-Viau, Quebec, Rigaud, Roxboro, Sherbrooke, Sorel, St-Bruno, St-Constant, St-Eustache, St-Jerome, St-Lambert, St-Remi, St. Vincent de Paul, Ste-Genevieve, Ste-Julie-de-Vercheres, Ste-Rose, Terrebonne, Valleyfield, Varennes, Vaudreuil

Ontario:
Amherstburg, Ancaster, Aurora, Barrie, Belle River, Belleville, Binbrook, Bradford, Brampton, Brighton, Burlington, Campbellford, Carleton Place, Clarkson, Cooksville, Cumberland, Dundas, Espanola, Essex, Fort Erie, Frankford, Galt, Georgetown, Gloucester, Grimsby, Guelph, Hamilton, Harrow, Hespeler, Jockvale, Kanata-Stittsville, King City, Kitchener-Waterloo, Lively, London, Maidstone, Malton, Maple, Markham, McGregor, Milton, Mount Albert, Mount Hope, Newmarket, Niagara Falls, Oak Ridges, Oakville, Orleans, Osgoode, Oshawa, Ottawa, Peterborough, Port Colborne, Port Credit, Preston, Richmond Hill, Sarnia, St-Catharines Thorold, Stirling, Stoney Creek, Stoney Point, Streetsville, Sudbury, Thorndale, Thornhill, Thurlow, Toronto, Trenton, Unionville, Waterdown, Welland, Wheatley, Windsor, Woodbridge, Woodslee, Wooler

* Limited time offer . Offer valid for new orders only. All prices are in USD.

Apparently Ontario/Quebec count as "Canada" I want more DIDs in Alberta/BC.

[WildChild]

I would like my ported DID to have a lower price. Currently stuck at $1.99/$0.0149. :(

[barrist]

Just got this email from voip.ms



Apparently Ontario/Quebec count as "Canada" I want more DIDs in Alberta/BC.

Hmm wonder how long this will go on for.. i was planning on signing up in June when we move. Should i just order a DID now?

[akmeikle]

sucks that existing customers don't get this rate as well

[death_hawk]

Hmm wonder how long this will go on for.. i was planning on signing up in June when we move. Should i just order a DID now?

I would. $1 per month isn't exactly bank breaking.
Assuming it goes up, 2 months will cost you $2 and 4 months of regular price will break you even. Anything beyond 4 months = profit.

[barrist]

hm so for 4.95 you get unlimited incoming. i would have to accumulate more than ~400 minutes to make that worthwhile.

[bacid1]

how's the quality of their service?

good enough for a business environment?

anyone with heavy voice usage using these guys?

[bridonca]

I think one would be better off buying sip settings from freephoneline.ca even if it does cost $50. Good for temporary purposes, I suppose.

[RMSimard]

sucks that existing customers don't get this rate as well

+1. Still good service though.


how's the quality of their service?

good enough for a business environment?

One of the best in the industry. Very business friendly, always improving and coming up with new, solid features at no added cost. If they were doing an IPO I'd invest in them; I like their service that much. I do however wish they had SMS pass-trough available like Acanac does, but haven't tested/checked that lately.

[marcelomanzo]

Why pay for it if I can get freephoneline for free?

or for only 60 CAD (only once forever) if you want use your ATA

[X360]

You will have to pay $1.50/month for E911 service with voip.ms... freephoneline.ca cost you nothing for E911 service...

[claudiu_corbea]

Voip.ms has a TON of extra features compared cu freephoneline.ca. For simple calling freephoneline is OK, but if you want extra Voip.ms is better.

[jackie999]

You will have to pay $1.50/month for E911 service with voip.ms

I didn't bother with the e911 fee..you don't have to take it. I figure, we have 911 on our cell phone....
I signed up a month ago so just missed out on the DID sale...I was rather disappointed that they're giving the deal to their existing customers only 'temporary', whereas if you're new, you get it permanently.
Otherwise I have no compliants with the service.

[yakkers]

what type of equipment do you need to hook this up?

[torooo]

....I was rather disappointed that they're giving the deal to their existing customers only 'temporary', whereas if you're new, you get it permanently.
Otherwise I have no compliants with the service.

This was posted on another forum from a voip.ms employee (I e-mailed them and they have adjusted my monthly price to $0.99 per month permanently) http://www.dslreports.com/forum/r24100365-VoIPms-Reduced-Rates-on-Canadian-DIDs:

"You can ask support to apply the new rate to your existing DID's, I'm pretty confident they will give it to you, if it's currently @ 1.49

This kind of promotion is to be able to reach new rates with the carrier (Pricing Plateaus), this then let us reduce prices permanently for ALL customers. Reducing all customer existing DID's now is financially impossible.

Note that it's currently marked as a temporary offer, not a permanent change. This all depend on the volumes we reach. At the time of your purchase, we were unable to offer you the product at the price of this new promotion.

Send an email to get the rate, the last thing we want is to make our customers feel mad.

Hope this help

Regards,
--
Martin - VoIP.ms"

[jackie999]

what type of equipment do you need to hook this up?

I bought a Linksys PAP2T from canadacomputers ($54)...was easy to set up...

[yakkers]

I bought a Linksys PAP2T from canadacomputers ($54)...was easy to set up...

I actually already own one of those, but I think it is branded/locked to Vonage.

Do you know if I can have it unbranded/unlocked?

or will it work as is?

[jackie999]

I think I read, when I was doing my research on setting it up, that it can be unlocked...but you could ask over in the dslreports forum..in the voip threads..you'd find somebody there that would be able to tell you

[iamdman]

I think I read, when I was doing my research on setting it up, that it can be unlocked...but you could ask over in the dslreports forum..in the voip threads..you'd find somebody there that would be able to tell you

don't even bother trying it's serioulsy tedious you would spend an entire day. Just go buy it from CC for $53, that's what i did today actually.

[chaddo]

I think I read, when I was doing my research on setting it up, that it can be unlocked...but you could ask over in the dslreports forum..in the voip threads..you'd find somebody there that would be able to tell you

They can be unlocked but it takes some know how! I have units for $20 if anyone wants one. :)

[X360]

I actually already own one of those, but I think it is branded/locked to Vonage.

Do you know if I can have it unbranded/unlocked?

or will it work as is?
You can PM/ask user DogFace05 at dslreports.com. He does unlock service for $15 US... He connects to your ATA online and unlock it... His method is real unlock so you can update any firmware later... Some other methods, if you upgrade new firmware, it will lock again...

[polka]

How it works with this company ? I mean Is it monthly basis or a price for full year like Acanac ? And how much is it? Do they give you phone number ?
Soryy for so many questions but went to thir web and could find how it works ?

[paulo500]

Even Dell.ca sells em for $52. (free shipping)

***** has a list of stores that sell them. (near you?)
http://www.*****/pp-linksys-pap2t-linksys-price-19321.html

Im currently on Primus Talk BB, but im paying about $30 per month. Unfortunately they dont support phone number porting, the last time i looked into it. :(
I don't know what to do...Pay more, or get a new phone number....it'll be such a pain in the butt to change everything to the new number! :cry:

[WildChild]

How it works with this company ? I mean Is it monthly basis or a price for full year like Acanac ? And how much is it? Do they give you phone number ?
Soryy for so many questions but went to thir web and could find how it works ?

It's a Pay As You Go/Bring Your Own Device VoIP service. You can get a DID (inward number) for a monthly fee and after you only pay for your communication usage (~1¢ per minute depending of the options you choose).

voip.ms offers a lot of free services like voicemail, ring groups, multiple lines, IVR, call forward, DISA and many other. You also have a lot of flexibility on how you can configure your services.

I ported my Vidéotron number to them last June and I've used exclusively their service for my home telephone since. I'm very satisfied with the customer support and with the stability. I use a Linksys SPA2102 ATA and I've also used a few times the softphone "Telephone" on my MacBook with great call quality. My monthly bill went from $20 + taxes to ~$7 per month. It took me approximately 6 months to cover the cost of the ATA.

[death_hawk]

freephoneline.ca is only profitable after 1-2 years compared to this (depending on your call volume), and doesn't come close to the features as voip.ms.
Oh and the restriction on what cities you can call without "long distance"

BUT! If your final objective is to get a dirt cheap phone line for whatever reason, freephoneline is probably a better option.

For me (and a bunch of others, including for business use) voip.ms can't be beat.

[yakkers]

so can someone explain the steps of setting up the service once you have an unlocked PAP unit?

Is it difficult?

Can a phone number from VONAGE be ported into VOIP.MS?

[MXMike]

I'm currently using Comwave but am thinking of jumping ship. I have a Vancouver number and a Montreal number that both ring the same phone ($33/month with tax, unlimited canada LD & several calling features). Can I port both my numbers to voip.ms and have similar functionality?
I see they charge $1.99/month for Vancouver numbers, so my total would be $2.98/month for the 2 numbers + usage? If this is correct, my monthly bill should be cut in half based on my current usage.

Also, does anyone have experience porting numbers from one VOIP provider to another? Can you use your existing provider until the number is ported for minimal interruption?

[d4diesel]

i asked about this last time in a post here as well regarding some other voip service but i still havent found a clear answer to it. I need a dedicated line to connect to a fax machinie. the line will only be used for incoming faxes. would this work? i am currently paying bell about 25 bucks a month to keep that line but i dont mind changing the number. anyone tried using this voip provider for fax?

[Badder]

i asked about this last time in a post here as well regarding some other voip service but i still havent found a clear answer to it. I need a dedicated line to connect to a fax machinie. the line will only be used for incoming faxes. would this work? i am currently paying bell about 25 bucks a month to keep that line but i dont mind changing the number. anyone tried using this voip provider for fax?

Yes, it should work. I currently use 2 separate phone numbers with them. The Linksys SPA2102 supports 2 phone lines. You can dedicate one to your fax number.

[TheRequiem]

i asked about this last time in a post here as well regarding some other voip service but i still havent found a clear answer to it. I need a dedicated line to connect to a fax machinie. the line will only be used for incoming faxes. would this work? i am currently paying bell about 25 bucks a month to keep that line but i dont mind changing the number. anyone tried using this voip provider for fax?

There are also VOIP services designed specifically for faxes: you don;t even need a fax machine. People send faxes to your fax number, and you'll be emailed a PDF of the document. Sounds brilliant, but never tried it myself.

[AlexV]

i asked about this last time in a post here as well regarding some other voip service but i still havent found a clear answer to it. I need a dedicated line to connect to a fax machinie. the line will only be used for incoming faxes. would this work? i am currently paying bell about 25 bucks a month to keep that line but i dont mind changing the number. anyone tried using this voip provider for fax?

No. I haven't checked voip.ms, but it's very rare to find a SIP provider that supports T.38.

[d4diesel]

There are also VOIP services designed specifically for faxes: you don;t even need a fax machine. People send faxes to your fax number, and you'll be emailed a PDF of the document. Sounds brilliant, but never tried it myself.

I saw that as well on vbuzzer's site but i didnt like how the only way to retrieve the fax was via their software installed and running on the computer. they dont email you a pdf file.

anyone who has used or heard of a good fax over IP provider that they can recommend? i would need a local Toronto/Mississauga number to go with that.

[RMSimard]

so can someone explain the steps of setting up the service once you have an unlocked PAP unit?

Is it difficult?

As difficult to set-up as any other BYO-Device service. They do have configuration samples in their user control panel to help you with Linksys/Sipura ATAs (Analog Telephone Adapter), Voxalot and Asterisk, promising to add more guides in the future. I believe you can open an account without purchasing; you'll then be able to check out their user control panel to familiarize yourself with rates and features. :) Then when you're ready you can prepay your account with your CC or PayPal (minimum $25) and pick a phone number (DID). Try using the free X-Lite softphone (http://www.counterpath.com/x-lite.html) from BC-based Counterpath to test things out first without having to shell out for an ATA or IP phone.

Just a note however: Linksys ATA devices are feature-full but a real usability nightmare. I'm pretty comfortable with technology yet had to read a fair deal to understand generally what acronym means what and which form field out of hundreds are important. That to me is a major obstacle to SIP-based VoIP services being accepted by everyday folks ...and why early companies like Vonage supply their own pre-configured units. Still, with an unlocked unit in default state the configurations guides usually hold your hand well enough for you to get the device connected and serving a dial tone.

[barrist]

im new to this stuff as well. are the ATAs from canada computers, dell, etc "unlocked" already?

[WildChild]

im new to this stuff as well. are the ATAs from canada computers, dell, etc "unlocked" already?

If the part number ends with -NA it's an unlocked device. Most web stores sell unlocked devices. I purchased my SPA2102 from DirectCanada last year.

[lcbo45]

i asked about this last time in a post here as well regarding some other voip service but i still havent found a clear answer to it. I need a dedicated line to connect to a fax machinie. the line will only be used for incoming faxes. would this work? i am currently paying bell about 25 bucks a month to keep that line but i dont mind changing the number. anyone tried using this voip provider for fax?

I have used it for fax for about a year and it works fine. I rarely get or receive faxes anymore so this works better than paying Bell for a line that doesn't see much use. I use the premium setting for fax as mentioned in the documentation.

[jackie999]

Here is my experience..from off the top of my head.

Sign up with voip.ms - paypal'd them $25 to start
Buy the ATA (PAP2T canadacomputers $54)
Read the configuration guide on voip.ms site to get the ATA working - it was no more difficult than setting up a router. If you want the ATA to jump thru hoops, then keep reading, it's very powerful. Otherwise, the basic configuration will get you going.
Plug your phone into the ATA. Make a few calls cause you're done.
If you want to receive calls, you can buy a number (DID) or you can *port* your existing phone number over (took about a week for me) ..cost $25..then $.99 cents a month for the DID.

You get all the bells and whistles. Call display..forward the Mother in Laws calls to "this line is no longer in service"...there are so many features, most of which I don't know what they do..so I keep reading :)

The ATA/modem must always be on..the computer does NOT need to be on.
Costs for calls out about 1/2 a cent a minute.
Costs for calls in about 1 cent a minute.
e911 service $1.50 a month - this is optional.
I also have to pay for a 'dry loop' for my teksavvy DSL - this, for me, is $10.22 a month.
All in all, I'm way ahead.
But I won't factor in the new panasonic phones I bought, or the APC backup :)

Hope this helps!

[Sneaks]

Here is my experience..from off the top of my head.

Sign up with voip.ms - paypal'd them $25 to start
Buy the ATA (PAP2T canadacomputers $54)
Read the configuration guide on voip.ms site to get the ATA working - it was no more difficult than setting up a router. If you want the ATA to jump thru hoops, then keep reading, it's very powerful. Otherwise, the basic configuration will get you going.
Plug your phone into the ATA. Make a few calls cause you're done.
If you want to receive calls, you can buy a number (DID) or you can *port* your existing phone number over (took about a week for me) ..cost $25..then $.99 cents a month for the DID.

You get all the bells and whistles. Call display..forward the Mother in Laws calls to "this line is no longer in service"...there are so many features, most of which I don't know what they do..so I keep reading :)

The ATA/modem must always be on..the computer does NOT need to be on.
Costs for calls out about 1/2 a cent a minute.
Costs for calls in about 1 cent a minute.
e911 service $1.50 a month - this is optional.
I also have to pay for a 'dry loop' for my teksavvy DSL - this, for me, is $10.22 a month.
All in all, I'm way ahead.

Hope this helps!

I signed up for the service about 2 months ago (going to email to see about getting dropped to the $0.99, cross fingers, but honestly extremely happy with even the $1.49.)

My process/experience was pretty much exactly the same as this, but will add a couple things.
1. you have full flexibility to change your options around so I didn't opt for the "unlimited" incoming add on. Since they charge in 6 sec increments (with 6 second minimum) I figured I'd just wait and see how much I was using, and then their website allows you to do full call detail reporting, so you can see how much incoming/outgoing, and related charges were. So far I'm nowhere near the $5/month incoming. Other than the $25 initial port I did, I've spent all of $11, that's incoming, outgoing, long distance, the monthly charge, and just signed up for the e911 (was waiting until I moved to register it.)

2. While The ATA does have to be on to make/receive calls obviously (but no computer as mentioned) your voicemail, and other features would still be active if your internet, or power drops, etc. They are configured through the website, and like many providers you can do things like forward it to your cell (this was great on the day of my move, when I was running between houses!) and email your voicemails to your BlackBerry, etc.

Really happy, and recommend them to anyone I've spoken with re: VOIP. BTW, their website is free to create an account for those that want to poke around if you're still not sure. Nice thing they have is a feature request page, where they detail what they've added, whats in development/testing, etc.

[jackie999]

going to email to see about getting dropped to the $0.99, cross fingers,...
I sent in a ticket last night and today they responded by dropping my fee to the new rate. I felt bad about complaining at all. I had compared them to bell, i.e. offering amazing deals to new people and turning their backs on existing customers who have supported them. But I'll admit I was wrong. They are a small company (this surprised me...) and their support is top notch. The live chat on their site has been extremely helpful. I would compare them to teksavvy for customer awareness..and my dig about comparing them to bell was below the belt ..sorry voip.ms...no one is THAT bad...

[aingaran]

I submitted a ticket today to reduce the rate and within a half hour or so, it had been done.

Now, I'm looking to port 2 other DIDs (currently with Unlimitel) and a toll free DID from telcan.net.

Just hoping they'd reduce the cost of porting, as it will cost me $25 x 3 ($75) and my current balance/credit with Unlimitel (non-refundable) of $50.

[shepd]

Hot! Thanks OP, I will jump in on this one!

[jackboot]

I love voip.ms :)

I have 3 DIDs: one business, one personal, and one business fax. The business phone line and business fax was previously through telus and cost $75/month each for a total cost of $150/month. Ripoff! I'm now paying no more than $15 / month for BETTER service overall. Last month my total business bill was $13.19. That's 11x less than what I was paying to Telus!

The only downside is that in my case the fax isn't totally reliable. I didn't expect it to be because VOIP is meant for fax, it's meant for voice. Voip.ms does not currently support T38 fax protocol (which apparently would make fax reliability about 100%), but they have told me that they intend to in the future. Even as it stands, I'd say I get 90% of faxes no problem. When I'm trying to send and it doesn't go through, I just try again. When I'm receiving and it doesn't go through, the sender usually just sends again. I've actually transitioned from sending faxes to doing the same thing via email whenever possible and eventually I'm sure that I'll be doing all my "faxing" this way. For the time being, I'd rather save the ~$850 per year and live with the slightly reduced fax reliability.

The service has been outstanding. I don't think the price can be beat, especially considering the features available and the customer support.

They just added DISA service which is really cool:


Similar to the call-back, The DISA (Direct Inward System Access) let you dial numbers via your VOIP Account from Inbound Calls. The main difference is that a DISA doesn't "call you back". It gives you directly the access to dial a phone number of your choice, similar to what a Calling Card service does. For example, you can create a DISA Entry, and configure one of your DID Number routing to point to a DISA. When you call, you will be prompted a PIN number and the telephone number you want to dial to. You can also use DISA from IVR's (Digital Receptionist), Time Conditions and CallerID Filtering.

[jetway1212]

no fax service is a deal breaker for me :(. I cant believe they dont implement FOIP already.

[apn64]

Fantastic service and truly incredible list of features, even at the old rate of $1.49/mo, but I've also put in a ticket to request the new rate ;)

I ported my RHP number ~15mo ago and haven't looked back.

The flexibility is mind-blowing. Time of day rules allow us to send all calls to voice-mail at certain times of the weekday. CallerID filtering means we get zero telemarketer calls ("we're sorry, the number you have dialed is not in service"). I have a sub-account setup for a 2nd line; outgoing calls right now, but I can also route incoming calls there based on a huge list of criteria.

...and it's all web-based, so I can manage the service from home or away.

Once you've had this level of control over your telephone service, there's no going back to 19th century POTS service, and a cheap UPS ensures the phone still works during short power outages.

Other than toll-free, all outgoing calls are paid, however, note that on the PAYGO option, you only pay for what you use: my InLaws were away for much of last month and their bill was $1.49 for DID + 12 cents for call time :D

In addition to the above, voip.ms also has an "UNLIMITED" incoming plan for ~$4.95 a month, however, it's actually capped at 3,500min/mo

[7-Endless]

Wonder why Ajax/Pickering is still $1.99 where Barrie, Markham, Richmond Hill, etc. are all .99 cents. :(

[bhupeshg]

if you have unlimited incoming on your cellphone, its also great for long distance to most countries thru callback, you pay far less compared to what most voip providers are charging.

[yakkers]

This may be a stupid question, but how does this compare to MAGICJACK (apart from having to keep the computer on for MAGICJACK to work?)

Im interested in this type of thing, but honestly am worried about the set up, configuation etc.

When I look at the website it seems complicated to me, certainly not directed at someone without computer experience.

Im not terrible with computers but I am not a wizzard either.

[plausipo]

:?: newbie question -(I have a 2GB internet bandwidth limit per month)
We talk about 2 hours on the telephone per month, how much bandwidth will it consume, using say a "normal" CODEC :confused:
Thanks in advance for answering.

[bhupeshg]

This may be a stupid question, but how does this compare to MAGICJACK (apart from having to keep the computer on for MAGICJACK to work?)

Im interested in this type of thing, but honestly am worried about the set up, configuation etc.

When I look at the website it seems complicated to me, certainly not directed at someone without computer experience.

Im not terrible with computers but I am not a wizzard either.

Never used magic jack but i have a ata adapter from linksys and x-lite on my computer, works great, very reliable.
its quite easy to setup there are other sites that even give u screenshots on how to setup, if you have pap2t-na. If you want to setup on your computer use x-lite, very easy and straight forward to setup too, there are also plenty of guides for tht online.

[jackie999]

This may be a stupid question, but how does this compare to MAGICJACK (apart from having to keep the computer on for MAGICJACK to work?)

Im interested in this type of thing, but honestly am worried about the set up, configuation etc.

When I look at the website it seems complicated to me, certainly not directed at someone without computer experience.

Im not terrible with computers but I am not a wizzard either.
I asked the same question before I took the plunge..if you're interested here are the threads,
http://www.dslreports.com/forum/r23962054-Teksavvy-Magicjack-quality
http://www.redflagdeals.com/forums/magicjack-teksavvy-867132/

For what it's worth...I have no regrets..my only regret is I didn't do it sooner.
There is a wealth of information on dslreports if you need help...all you're investing is time and <$100 ....which you will recover in no time at all...

[signup]

Here is my experience..from off the top of my head.

Sign up with voip.ms - paypal'd them $25 to start
Buy the ATA (PAP2T canadacomputers $54)
Read the configuration guide on voip.ms site to get the ATA working - it was no more difficult than setting up a router. If you want the ATA to jump thru hoops, then keep reading, it's very powerful. Otherwise, the basic configuration will get you going.
Plug your phone into the ATA. Make a few calls cause you're done.
If you want to receive calls, you can buy a number (DID) or you can *port* your existing phone number over (took about a week for me) ..cost $25..then $.99 cents a month for the DID.

You get all the bells and whistles. Call display..forward the Mother in Laws calls to "this line is no longer in service"...there are so many features, most of which I don't know what they do..so I keep reading :)

The ATA/modem must always be on..the computer does NOT need to be on.
Costs for calls out about 1/2 a cent a minute.
Costs for calls in about 1 cent a minute.
e911 service $1.50 a month - this is optional.
I also have to pay for a 'dry loop' for my teksavvy DSL - this, for me, is $10.22 a month.
All in all, I'm way ahead.
But I won't factor in the new panasonic phones I bought, or the APC backup :)

Hope this helps!

Did you pay activation fee for your dry loop? Also when you port your number are they the one to contact your provider or you would inform your provider to cut the service?

[jackie999]

Did you pay activation fee for your dry loop? Also when you port your number are they the one to contact your provider or you would inform your provider to cut the service?

My home phone was with Teksavvy, I was no contract, month to month. I paid them $20 for my change of service to dry loop ( I still have teksavvy as my DSL provider, via dry loop now).
I also had to pay voip.ms $25 for the port, as I mentioned above.
All I did was contact voip.ms - there is a form you fill out in your profile (re the porting) and they do the rest..it was relatively quick. I had no down time at all switching my DSL to dry loop..maybe I was lucky? I had read there could be no internet for a day....I have a cell phone..I wasn't too concerned.

[Sneaks]

:?: newbie question -(I have a 2GB internet bandwidth limit per month)
We talk about 2 hours on the telephone per month, how much bandwidth will it consume, using say a "normal" CODEC :confused:
Thanks in advance for answering.

Using 729a (I use it fine) about 28Mb, so in essence nothing. :)

[aingaran]

Wonder why Ajax/Pickering is still $1.99 where Barrie, Markham, Richmond Hill, etc. are all .99 cents. :(

It's probably because it's through different carriers.

[Sneaks]

My home phone was with Teksavvy, I was no contract, month to month. I paid them $20 for my change of service to dry loop ( I still have teksavvy as my DSL provider, via dry loop now).
I also had to pay voip.ms $25 for the port, as I mentioned above.
All I did was contact voip.ms - there is a form you fill out in your profile (re the porting) and they do the rest..it was relatively quick. I had no down time at all switching my DSL to dry loop..maybe I was lucky? I had read there could be no internet for a day....I have a cell phone..I wasn't too concerned.

I was nervous about interrupting my DSL as well, (and I was also having to submit a move too, which was the key reason I wanted to port, to keep my number.) So what I did was submitted the port, they contacted bell and cancelled my hard line, my DSL continued to work fine, no charges for what was now effectivly a dry loop (can't imagine that woudl have gone unnoitced for too long.) Once that was done (only took like 4 days for me!, I was surprised how quick it was) though a coupel weeks later I then submitted my move ticket to move my DSL to my new place, and this is where I got dinged the setup/dry loop charge.

So don't know if I just got lucky, or if my DSL would have eventually stopped, but didn't seem like it. :)

[signup]

It looks like a lot of cost involve to jump on, i.e.
ATA - $54
Dry loop - $10
Dry loop activation fee - $20
Porting - $25
Voip.ms deposit - $25USD

[dinesh_zee]

Do you know whether this works if the ATA (Linksys pap2) is connected to the Internet from outside Canada (in Europe)?

I tried vbuzzer (they charged for DID only and not for incoming minutes) and it works, but freephoneline doesn't work most of the time. Not sure about voip.ms.

[djino]

It looks like a lot of cost involve to jump on, i.e.
ATA - $54
Dry loop - $10
Dry loop activation fee - $20
Porting - $25
Voip.ms deposit - $25USD

If you do not require more than a 60GB Cap, then just stick with Bell for your DSL Internet to avoid ALL Dry Loop fees.

I currently pay $24.95 for 7mbps Dry loop internet 60GB Cap (yes + $3.95 Modem Rental).

I am currently with cia for voip ($3.95/month) = unlimited incoming/outgoing local calls, but I'm seriously considering switching to voip.ms as I sometimes make long distance calls which might come out to be cheaper than cia.

djino
"Maybe I'll try out voip.ms for a few months and port later if it works out"

[Sneaks]

It looks like a lot of cost involve to jump on, i.e.
ATA - $54
Dry loop - $10
Dry loop activation fee - $20
Porting - $25
Voip.ms deposit - $25USD

Compared to what I was paying Bell, I'm still way ahead, was able to keep my number in a cross city move, and have more flexibility/features as well, and of course longer I stay further ahead I get.

Slight note: voip.ms deposit $25, -$25 port washes out to $0, but obviously you then have to add some money back in since it's a "pre-paid" service, in the fact that monthly charge for the DID, and then as you use it it decrements from that "deposit." The speed of the port actually caught me off guard, because that's exactly what I did, so was techinically $1.49 in the hole for half a day, got a little reminder email, then added back in another $25 and now am sitting at about $13 in the account. I have it set to remind me via email at $10.

Also for anyone who didn't ntice reminder it is charged in USD, which right now is actually cheaper. ;)

[dinesh_zee]

Minimum deposit is $25USD... but when we have to deposit again? After reducing your balance to $0 or some other amount?

Can this deposit be used to pay for DID as well, or only for in/out calls?

[Sneaks]

Do you know whether this works if the ATA (Linksys pap2) is connected to the Internet from outside Canada (in Europe)?

I tried vbuzzer (they charged for DID only and not for incoming minutes) and it works, but freephoneline doesn't work most of the time. Not sure about voip.ms.

While I've not tried it directly, as long as you can route traffic to their servers, and latency isn't crazy, yes, should be 100% fine.

You can choose whichever server you want to point your pap2 to, so I'd suggest the closest one to where you live, which *usually* would have the least latency. If you're unsure, do a live chat with them to ask your specific question.

From their FAQ:

VoIP Server Locations
Houston, TX
Dallas, TX
Los Angeles, CA
New York, NY
London, UK
Montreal, QC
Toronto, ON

I use TO.

[jackie999]

Compared to what I was paying Bell, I'm still way ahead

+1

I agree...my costs were more than double before (with land line). Granted, there is a learning curve and a cost outlay to set up. But you have to look at the big picture...
Think of how much extra you'll have to buy all of these great RFD deals :)

[jackie999]

I use TO.

I use Toronto as well. As far as I can tell, my phone clarity is same as it was before - with one exception. I was downloading 4...um...linux.iso's last evening and somebody called me. The call was a little choppy so I paused my downloads and it cleared right up. I can go into my router and set up QOS or some such thing, to give voip priority, but so far haven't bothered.
People calling me haven't commented at all so I'm assuming from the other end you can't tell a difference.

[apn64]

Minimum deposit is $25USD... but when we have to deposit again? After reducing your balance to $0 or some other amount?

Can this deposit be used to pay for DID as well, or only for in/out calls?

Deposit is just pre-paying your account (like a PAYGO cellphone). It will count down from usage i.e. monthly DID payment, optional 911, and call time. You can run it all the way to zero, or specify (minimum $1) at what balance they send you an email reminding you to top-up (minimum $25).

BTW, for those jumping on the bandwagon, once you have service up and running to your liking, it might be worth throwing a chunk of money onto your account before the HST lands.

Today they bill GST only when you top-up, so adding $100 is actually billed at $105. Extending the model, it will become $113 after July 1st.

[jackie999]

Minimum deposit is $25USD... but when we have to deposit again? After reducing your balance to $0 or some other amount?

Can this deposit be used to pay for DID as well, or only for in/out calls?
You deposit money into your voip.ms account - this is a prepaid service. Once it's depleted to whatever amount you designate (in my case $10) they will email you to remind you to top up.
All of your costs come out of that pot - DID, e911 (if you want it) your costs per call. Your voip a/c will show you to the 100th of a cent where the money has gone, including phone numbers in and out.

[apn64]

You can choose whichever server you want to point your pap2 to, so I'd suggest the closest one to where you live, which *usually* would have the least latency. If you're unsure, do a live chat with them to ask your specific question.

From their FAQ:

VoIP Server Locations
Houston, TX
Dallas, TX
Los Angeles, CA
New York, NY
London, UK
Montreal, QC
Toronto, ON

I use TO.

Here's another option for those that want to use the 2nd port on a PAP2T;

Setup a FREE sub-account and (w/o buying a 2nd DID) use it for outgoing calls, or to simply manage/route incoming calls e.g. basement, home office, kids room (you can setup a rule that says "send all calls from these numbers to that phone" etc).

Anyway, the point is once you have two active ports, I find it handy to keep my DID reg'd on the TO server, but I have my sub-account reg'd on NYC. Why you ask?

Ever tried to call one of those US toll-free numbers that says "we're sorry, that number is not reachable..." from Canada? Problem solved ;)

[Sneaks]

Minimum deposit is $25USD... but when we have to deposit again? After reducing your balance to $0 or some other amount?

Can this deposit be used to pay for DID as well, or only for in/out calls?

Whatever money you put in will be used for what ever services you use, be it inbound/outbound minutes, getting a new DID (if you're not looking to port,) DID port, e911. So, it's not really a deposit in the traditional sense, but rather a pre-pay, the initial amount you start off with that then starts decrementing as you get a DID, take/make calls etc.

So for my exact scenario.

Created an account to poke around (free.)
Asked a bunch of questions (still free)
Decided to sign up. Threw in $25 (balance $25)
Submitted my port request (balance $0)
Port completed 4 days later $1.49 monthly charge (balance -$1.49 - oops :) )
Added another $25 - you can add more I just figured it was good enough amount to start with (balance $23.51)
Usage started decrementing as I made/received calls.....($23.51 - usage as it happens) and so on, and so on.


So when I added e911, that also decrements the $1.50 from the amount. So right now, I'm at like $13.92726 in my account - yes they really do bill down to sub-cent increments. :)
My Usage over the 2 months: Calls: 265 | Duration: 14:07:17 | Total: $6.59274
I had the $25 port charge, 2 months at $1.49 for the DID, and $1.50 when I setup the e911.

Oh and they were kind enough to drop me to the $0.99 DID going forward yeah! :) Didn't complain was perfectly honest in how happy I was with them, just asked them nicely if they would, since I had just signed up a couple months ago.

They had no obligation to do it, but did and just made me that much happier.

[barrist]

I signed up, and picked a DID

Don't wanna be caught with a 343 area code in Ottawa when 613 runs out!

[d4diesel]

It looks like a lot of cost involve to jump on, i.e.
ATA - $54
Dry loop - $10
Dry loop activation fee - $20
Porting - $25
Voip.ms deposit - $25USD

ATA is a one time investment. much like buying a phone set. as for dry loop charges, doesnt apply to everyone. I have internet over cable and dont have any such charges. and the voip.ms deposit itself is the money that you will be paying for the service which on an average use should last you easily 4-5 months.

[Nraja]

Slightly confused guys

I need a number for business that I can forward to my Rogers cellphone. I've been looking at voip.ms as an option, and have 2 questions that hopefully you guys can answer for me:

1) Do I need an ATA if all I'm doing is forwarding the calls to my cellphone automatically?
2) Can I setup 1 voip.ms account to have more than 1 number forwarded to my Rogers cellphone?

[ChinpokoMon]

1) Do I need an ATA if all I'm doing is forwarding the calls to my cellphone automatically?

No, you do not need an ATA if you just want to forward a DID number to your cell.



2) Can I setup 1 voip.ms account to have more than 1 number forwarded to my Rogers cellphone?

Yes, you can order more than 1 DIDs for your account and have them forward to the same number.

[soblecram]

Slightly confused guys

I need a number for business that I can forward to my Rogers cellphone. I've been looking at voip.ms as an option, and have 2 questions that hopefully you guys can answer for me:

1) Do I need an ATA if all I'm doing is forwarding the calls to my cellphone automatically?
2) Can I setup 1 voip.ms account to have more than 1 number forwarded to my Rogers cellphone?


Nope. I had a DID # in Alberta, and one in Ottawa routed to my cell phone for 6 months before I got an ATA and started making outgoing calls.. Just pay for the incoming minutes (at 1cent/min, billed in 6 second increments....lol)

[signup]

ATA is a one time investment. much like buying a phone set. as for dry loop charges, doesnt apply to everyone. I have internet over cable and dont have any such charges. and the voip.ms deposit itself is the money that you will be paying for the service which on an average use should last you easily 4-5 months.

Yeah some charges are not applicable to everyone, but this is on my case that I have a Teksavvy DSL and Bell phone.

It will take a yr to cover the cost for me and onwards I will save at least $10/mon. Now need to decide...

[Sneaks]

Yeah some charges are not applicable to everyone, but this is on my case that I have a Teksavvy DSL and Bell phone.

It will take a yr to cover the cost for me and onwards I will save at least $10/mon. Now need to decide...

Do it...succomb to the RFD pressure, you'll thank me...and if you don't like it it was the OP's fault. Hehe.. ;)

Serioiusly though, everyones situation's different, so look at your reasons to do so, both cost, and feature/flexibility to determine if there's value for you. I was moving between cities (same calling area still though,) so I had extra incentive, because I wanted to keep my number, and Bell is still in dinosaur times with not being able to do so. My bell line was also costing me like $50/month for less features and while I could have probably tried to get some additional retention deals to reduce that, why bother going to the hassel when I could get it cheaper elsewhere through their regular service, more features, and then keep my number on top.

It was pretty funny when Bell started calling me back to try to win me back to their service, and even after I reconfirmed to the guy that they couldn't port numbers, and they couldn't compete on cost, so what was left for them to offer? :)

[paulo500]

So it sounds like they don't have direct bank withdraw payments? Hopefully I can set up my online banking to transfer money every month automatically? (Im the worst at remembering to pay things monthly.) lol

[Sneaks]

So it sounds like they don't have direct bank withdraw payments? Hopefully I can set up my online banking to transfer money every month automatically? (Im the worst at remembering to pay things monthly.) lol

For how cheap it is, you won't need to pay them every month! :)

[signup]

Do it...succomb to the RFD pressure, you'll thank me...and if you don't like it it was the OP's fault. Hehe.. ;)

Serioiusly though, everyones situation's different, so look at your reasons to do so, both cost, and feature/flexibility to determine if there's value for you. I was moving between cities (same calling area still though,) so I had extra incentive, because I wanted to keep my number, and Bell is still in dinosaur times with not being able to do so. My bell line was also costing me like $50/month for less features and while I could have probably tried to get some additional retention deals to reduce that, why bother going to the hassel when I could get it cheaper elsewhere through their regular service, more features, and then keep my number on top.

It was pretty funny when Bell started calling me back to try to win me back to their service, and even after I reconfirmed to the guy that they couldn't port numbers, and they couldn't compete on cost, so what was left for them to offer? :)

Might consider, now researching for a good ATA.

So far how's the voice quality is it good as POTS, any echo?

[MXMike]

So it sounds like they don't have direct bank withdraw payments? Hopefully I can set up my online banking to transfer money every month automatically? (Im the worst at remembering to pay things monthly.) lol
This is under their "Feature Requests: Planned for Development" section:

Credit Card Auto-replenishment


So it looks like automatic credit card payments are in the works at least. For now like others have said, just set it up to email you when your balance hits a certain level.

I just decided to switch from Comwave to voip.ms. I had a live chat with voip.ms and they confirmed both my BC and QC numbers are eligible to port to them. Then I ordered a cisco PAP2T (http://www.ncix.com/products/index.php?sku=21847&vpn=PAP2T-NA&manufacture=CISCO)from NCIX for $50 (pm'd) with their free shipping promo. I'm going to wait at least until the ATA gets shipped before I port my numbers though so I don't get stuck without an ATA I can use with voip.ms if the port completes soon.

[Tychicum]

After doing much research I ended up with Voip.ms AND Voxalot.com as a PBX service which is layered on top of Voip.ms to get me all of the bells and whistles which I could ever want.

I thought it would be "easier" to make the transition if I had some of the Voxalot services... but it was probably a bit more tricky to get all sorted out and set up. I am happy with it now however.

I also bought Cisco's 3102 ATA not the more common lower end models. There is a guy in the US selling them on eBay for $75 ... but I went the cheap route and bought one in China for $45.

The 3102 is very programmable and as it has the ability to plug PSTN regular land lines in to my little phone network. So I have a multi line service to help me "transition" over to Voip full time. It is also a model which allows me to phone the 3102 (using my cell phone for example) and reroute my long distance over IP through the 3102 instead of through the cell phone company.

So for the time being incoming calls can come in through IP or land line and ring on my portable phones at home. And of course if I don't answer I have that forwarded (Voxalot feature) to my cell phone.

So for the time being anyway I didn't even bother with a DID for my Voip ...

I also am still a fan of Skype. As their system has grown the services just keep getting better and better. Video calls from the kids who live far away are always a treat. And it is impossible to miss a Skype call as I have that forwarded to my cell phone as well ...

When my daughter used to call from Honduras even if I was on vacation in Kelowna my cell phone would ring when I was sitting on the beach !!!

[ur_funneh]

silly question but wouldn't the magicjack serve as a *cheaper* alternative? I mean you get all the bells and whistles with MJ and you get to choose your incoming number.

If your stuck paying $50+ for the ATA Device plus $15+/year for the DID service, you're still better off buying a MJ and that also gives you all the features. Only reason I brought this up was because someone stated the ATA must always be connected to the internet, well so does this.

P.S Fax even works with this...


---I know theirs already a MJ thread but I wanted to input this minor details for those who are nitpicking at pricing.

[jackie999]

I considered Magic Jack originally and decided to go with voip.ms. Here is one site you might want to read if you're considering it,

http://boingboing.net/2010/02/23/magicjack-dials-wron.html

I didn't like the way you had to run your computer 24/7 for it to work, and I didn't like that it spies on your internet activities and it can't be uninstalled. It isn't that much cheaper, if at all. I don't think you can keep your existing phone number with it either..there were just no benefits to it imho. But that's just my 2cents. You may decide otherwise once you weigh the pros and cons...

[tk11]

voip.ms is excellent

switched to them in February

magicjack or any other voip providers dont even come ANYWHERE close to these guys.
Soo many fully customizable features.

Their support is good, but not great. Its not 24 hours support so if you run into problems during their off hours you'll have to wait awhile for a response.

[Nraja]

No, you do not need an ATA if you just want to forward a DID number to your cell.



Yes, you can order more than 1 DIDs for your account and have them forward to the same number.

Sweeet! Just bought a Toll Free US DID....need a local 416 one, but they're all 647 or 289. I know 416's are near depleted, but any way around this?

I just don't want people text messaging the 647 number on the site because they think its a cellphone.

Also, I'm calling my 877 number but it doesn't show the caller id (even though I checked the CallerID box for 1.25 cents a shot). Any idea why it keeps showing that a random 416 number is calling?

thanks again!

[Sneaks]

Might consider, now researching for a good ATA.

So far how's the voice quality is it good as POTS, any echo?

I've been very happy with the quality, and I'm using g729a, which is the lower bandwidth, less quality codec. I also use the half cent/min routes, instead of the "premium" ones. I had one call, where I had what sounded like a brief echo for a few seconds, but it didn't last so not even sure wasn't an anomaly and otherwise I haven't noticed any.

I chose the pap2t, because it was cheap, readily available, and they had configuration guides for it, thus simplifying things. I saw the 2102/3102 at CC as well, but didn't really see any reason for it for what I needed.

Good luck!

[Sneaks]

Oh yeah, and to add, I previously used skype out via a dlink ATA to do unlimited NA calling (actually had both hooked up in sequence before I moved) and quality was much better through voip.ms. Not that the skype out was horrible or anything, just a little more susceptible to echo, etc. I have a year prepaid on it (was kind of by accident as I forgot to cancel it,) but I haven't bothered to even hook it up yet since I moved, cause the voip.ms has been great.

[BillyH]

I don't even know how to search for this and I am not certain if any voip service has this feature.

Is there a service that allows you to call it from another phone, say your cell phone, then have that service make the long distance call for you at the voip rate?

[ganiba]

I don't even know how to search for this and I am not certain if any voip service has this feature.

Is there a service that allows you to call it from another phone, say your cell phone, then have that service make the long distance call for you at the voip rate?

http://www.yak.ca/product/yak-cell/overview

[Hugh]

I also bought Cisco's 3102 ATA not the more common lower end models. There is a guy in the US selling them on eBay for $75 ... but I went the cheap route and bought one in China for $45.

The 3102 is very programmable and as it has the ability to plug PSTN regular land lines in to my little phone network. So I have a multi line service to help me "transition" over to Voip full time. It is also a model which allows me to phone the 3102 (using my cell phone for example) and reroute my long distance over IP through the 3102 instead of through the cell phone company.

So for the time being incoming calls can come in through IP or land line and ring on my portable phones at home. And of course if I don't answer I have that forwarded (Voxalot feature) to my cell phone.

So for the time being anyway I didn't even bother with a DID for my Voip ...


Interesting.

The SPA3102 seems to be in the same family as the PAP2 -- I'm not sure why you'd call it higher quality.

Instead of two FXS ports (like the PAP2), it has one FXS and one FXO. This is good for you: you use the FXO to connect to the incoming land-line.

It also has two ethernet ports and will operate as a router. Perhaps that it can be deployed outside any NAPTing in a typical home installation. Or perhaps that means it doesn't depend on something (say, a wireless router) that the user might wish to turn off when shutting down their computer(s).

Can you tell us how to get SPA3102 units at the good price you did? Or does that require an actual trip to China?

[MXMike]

...Can you tell us how to get SPA3102 units at the good price you did? Or does that require an actual trip to China?

Looks like the SPA3102 is quite readily available in Canada for around $70...
http://www.pricecanada.com/p.php/Cisco-SPA3102-VoIP-Router-SIPV2-1FXO-SPA3102NA-617694/
http://www.pricecanada.com/p.php/Linksys-Voice-Gateway-with-Router--SPA3102NA-431208/

I did a little research into this ATA too but decided to go with the PAP2T. I read in a few places that the 3102 can handle 2 simultaneous calls and the PAP2T can't? Not a feature I'd be likely to use anyway.

[RMSimard]

Is there a service that allows you to call it from another phone, say your cell phone, then have that service make the long distance call for you at the voip rate?

Yes, voip.ms has callback service and now DISA (Direct Inward System Access). I've setup my cellphone so that I can call my own DID and through caller ID filtering I get my own voip outside line (with a PIN for safety) at regular voip fees, while everyone else simply gets my phone ringing instead. Pretty awesome! :)

On the other end, the nice thing about callback service is that if your cellphone has free incoming but low-priced texting and you're really cheap (well, it *is* RFD) you can just send an SMS to your DID and the system will call you back with a dial tone. ;)


Can you tell us how to get SPA3102 units at the good price you did? Or does that require an actual trip to China?

On my end, first I had a SPA1001 (2 voip lines on one port) from ebay, but gave it away to the GF and got myself a SPA3102 (1 voip line on one port, 1 landline on other port, optional built-in router) from DirectCanada last year for $70 while I was transitioning from landline to voip. Convenient if you still need/have a landline, but I now I miss the convenience of having 2 lines and wish i had bought something else.

[soblecram]

Yes, voip.ms has callback service and now DISA (Direct Inward System Access). I've setup my cellphone so that I can call my own DID and through caller ID filtering I get my own voip outside line (with a PIN for safety) at regular voip fees, while everyone else simply gets my phone ringing instead. Pretty awesome! :)

On the other end, the nice thing about callback service is that if your cellphone has free incoming but low-priced texting and you're really cheap (well, it *is* RFD) you can just send an SMS to your DID and the system will call you back with a dial tone. ;)


Do it without texting; if you have free incoming set your DID to CALLBACK your cell#. You call it, get a busy, then it calls you with a dial tone. My DID is setup for callback (caller id filtering) on my cell and my wifes; any other number that calls it gets the PASSWORD prompt

[jackie999]

Also, I'm calling my 877 number but it doesn't show the caller id (even though I checked the CallerID box for 1.25 cents a shot).

From what I understand, the CNAM is hit and miss. I *think* when a call is made, if you've requested the associated Caller ID, then the system must search another database..this 1.25 cent charge you pay is what they pay for the search...once it's searched for the CNAM, it'll retain it for 24 hours so you don't pay again that day for same name.
I read on dslforums you can contact support and they will make sure your number is recorded in this database so future searches will show in the CNAM.
If I'm wrong somebody correct me.
Another nice feature of voip.ms - add all your friends/family to your phonebook on voip.ms, then when a call comes thru it will show that name :)
I love this voip.ms - I'm my own phone company :)

[djino]

Yes, voip.ms has callback service and now DISA (Direct Inward System Access). I've setup my cellphone so that I can call my own DID and through caller ID filtering I get my own voip outside line (with a PIN for safety) at regular voip fees, while everyone else simply gets my phone ringing instead. Pretty awesome! :)

On the other end, the nice thing about callback service is that if your cellphone has free incoming but low-priced texting and you're really cheap (well, it *is* RFD) you can just send an SMS to your DID and the system will call you back with a dial tone. ;)

HOLY CHIT!, Can you really program it to do this? Thats awesome.

djino
"Really considering switching from cia voip to voip.ms for that feature alone"

[dinesh_zee]

Deposit is just pre-paying your account (like a PAYGO cellphone). It will count down from usage i.e. monthly DID payment, optional 911, and call time. You can run it all the way to zero, or specify (minimum $1) at what balance they send you an email reminding you to top-up (minimum $25).

BTW, for those jumping on the bandwagon, once you have service up and running to your liking, it might be worth throwing a chunk of money onto your account before the HST lands.

Today they bill GST only when you top-up, so adding $100 is actually billed at $105. Extending the model, it will become $113 after July 1st.


You deposit money into your voip.ms account - this is a prepaid service. Once it's depleted to whatever amount you designate (in my case $10) they will email you to remind you to top up.
All of your costs come out of that pot - DID, e911 (if you want it) your costs per call. Your voip a/c will show you to the 100th of a cent where the money has gone, including phone numbers in and out.


Whatever money you put in will be used for what ever services you use, be it inbound/outbound minutes, getting a new DID (if you're not looking to port,) DID port, e911. So, it's not really a deposit in the traditional sense, but rather a pre-pay, the initial amount you start off with that then starts decrementing as you get a DID, take/make calls etc.

So for my exact scenario.

Created an account to poke around (free.)
Asked a bunch of questions (still free)
Decided to sign up. Threw in $25 (balance $25)
Submitted my port request (balance $0)
Port completed 4 days later $1.49 monthly charge (balance -$1.49 - oops :) )
Added another $25 - you can add more I just figured it was good enough amount to start with (balance $23.51)
Usage started decrementing as I made/received calls.....($23.51 - usage as it happens) and so on, and so on.


So when I added e911, that also decrements the $1.50 from the amount. So right now, I'm at like $13.92726 in my account - yes they really do bill down to sub-cent increments. :)
My Usage over the 2 months: Calls: 265 | Duration: 14:07:17 | Total: $6.59274
I had the $25 port charge, 2 months at $1.49 for the DID, and $1.50 when I setup the e911.

Oh and they were kind enough to drop me to the $0.99 DID going forward yeah! :) Didn't complain was perfectly honest in how happy I was with them, just asked them nicely if they would, since I had just signed up a couple months ago.

They had no obligation to do it, but did and just made me that much happier.

thanks for the clarification reg. the deposit amt!

[dinesh_zee]

Yes, voip.ms has callback service and now DISA (Direct Inward System Access). I've setup my cellphone so that I can call my own DID and through caller ID filtering I get my own voip outside line (with a PIN for safety) at regular voip fees, while everyone else simply gets my phone ringing instead. Pretty awesome! :)


Accessing your voip account thru local access numbers is the best! I have been using this feature for other voip providers (betamax - only outgoing) as well... MyFav5 local feature in my cell makes this even easier.

The best thing is you don't need ATA at all... u don't even need internet!

[paulo500]

I didn't know we could do this with Voip! How do you guys figure this stuff out?? LOL

Going to pickup the PAP2T at CC tonight. Seems id be silly not to get voip.ms for my needs. :lol:

[mydssbin]

Anyone know where i can get European #'s?

[Tychicum]

Interesting.

The SPA3102 seems to be in the same family as the PAP2 -- I'm not sure why you'd call it higher quality.

Instead of two FXS ports (like the PAP2), it has one FXS and one FXO. This is good for you: you use the FXO to connect to the incoming land-line.

It also has two ethernet ports and will operate as a router. Perhaps that it can be deployed outside any NAPTing in a typical home installation. Or perhaps that means it doesn't depend on something (say, a wireless router) that the user might wish to turn off when shutting down their computer(s).

Can you tell us how to get SPA3102 units at the good price you did? Or does that require an actual trip to China? It wouldn't be a higher quality as I am sure it comes off of the same Chinese assembly line ... just different features for a few bucks difference.

Linksys/Cysco make several configurations.

I got mine on an eBay auction. But they are available from many places nearly as cheap.

[laptop-tech]

Im using voip.ms as a long distance only, without a local DID, just using pre paid balance to call overseas. Amazing quality and no monthly costs. Love it.

If they had a fixed monthly fee like Vbuzzer for local DID, I'd sign up right away. Im using Freephoneline and despite the zero cost, the constant disconnections that a ton of people are experiencing for months now are just too much to handle.

[Tychicum]

Anyone know where i can get European #'s?
Europe. :cheesygri

Every country has different regulations. Just google the country name and the term "Voip". Voip is much more commonly available in Europe as their land line phone companies frequently even charge for local calls.

[Sneaks]

Im using voip.ms as a long distance only, without a local DID, just using pre paid balance to call overseas. Amazing quality and no monthly costs. Love it.

If they had a fixed monthly fee like Vbuzzer for local DID, I'd sign up right away. Im using Freephoneline and despite the zero cost, the constant disconnections that a ton of people are experiencing for months now are just too much to handle.

On their feature accepted for Development page:
Residential Free Zone Calling Plans (Flat Rate Termination Plans)

Perhaps you'll get it yet. :)

[jackboot]

Whoa, I just saw this on their "in development or testing" list:

"Free calls to other VoIP.ms Customers DIDs"

That's pretty rad!

[user007]

Hi Guys,

Just saw this. Thanks for the thread. I read it and still do not understand some of this as I am new to this.

So can one get a new phone number with this.

I have Rogers internet-can I use it with that?

How is the voice clarity? I installed freephoneline but the quality was really poor (a lot of echo).

If I am not satisfied can I cancel it and not have to continue to pay the $1 fee.

Thanks for your help.

[Sneaks]

So can one get a new phone number with this.

Yes, select add DID to get a new one (vs. porting which would move an established number)



I have Rogers internet-can I use it with that?

Yes


How is the voice clarity? I installed freephoneline but the quality was really poor (a lot of echo).

Voice quality is great, didn't use freephoneline, but based off other comments sounds like it's not great, vs. VOIP.ms which is far superior



If I am not satisfied can I cancel it and not have to continue to pay the $1 fee.


Yes, cancel the DID and that will no longer be charged to you, I'm assuming you cant refund the initial $25 though (ask their chat line to be sure,) but hopefully you'll be happy anyways and it won't matter. Otherwise, just use it until the $25 is gone.

[user007]

Thanks for the replies Sneaks.

[apn64]

Anyone know where i can get European #'s?

voip.ms offers numbers in a many European and other countries. Cost is a bit higher than NA @ ~$4/mo per DID, but the option is there.


Whoa, I just saw this on their "in development or testing" list:

"Free calls to other VoIP.ms Customers DIDs"

That's pretty rad!

We can already do this today via SIP to SIP calling :)

[yakkers]

It sounds like this is definitely the way to go if your ultimate goal is to go VOIP and ditch Rogers and Bell completely for home phone service.

After what I read about Magic Jack the only way I would consider that is if I had a dedicated PC that I used just for that phone, I woudnt want to install the magic jack software on my laptop or other PC's for sure.

Altho, the magic Jack is very useful if you are travelling as you have access to your number.

[kamran]

I've read this thread, can someone explain in plain language what's the benefit of this service and what's needed to take advantage of it.

[jackie999]

I've read this thread, can someone explain in plain language what's the benefit of this service and what's needed to take advantage of it.


Because it's a little daunting to set this all up, a few of us have tried to spell out how/what this is all about at the beginning of this thread..my contribution was post #37 I think.
I don't know how I can explain it any better..perhaps if you throw a few questions at us we'll be able to clear it up for you :)

[kamran]

thank you, post 37 does clear things up.

Because it's a little daunting to set this all up, a few of us have tried to spell out how/what this is all about at the beginning of this thread..my contribution was post #37 I think.
I don't know how I can explain it any better..perhaps if you throw a few questions at us we'll be able to clear it up for you :)

[jackie999]

Another place to look..to try and understand exactly what this voip.ms is all about is in the reviews over on dlsforums. There is a review by "mango" that is well written and gives some more detail..here is a link to where you'll find his review (2nd one down on this page) if you're interested:
http://www.dslreports.com/comments/3342#review2

[yakkers]

Because it's a little daunting to set this all up, a few of us have tried to spell out how/what this is all about at the beginning of this thread..my contribution was post #37 I think.
I don't know how I can explain it any better..perhaps if you throw a few questions at us we'll be able to clear it up for you :)

There is probably a huge market for someone to set it all up for you at home. Especially if that person also sold the hardware needed for cheap.

[Sneaks]

It sounds like this is definitely the way to go if your
Altho, the magic Jack is very useful if you are travelling as you have access to your number.

You don't have to use an ATA with voip.ms, you can also use a softphone, etc. So you also can have access while traveling if you so desire. Quite flexible, depending what you want to do. Or you can do both via sub accounts, etc. Since it's all configured via the web, you can also very easily change things, and play around to test out different configurations/options.

So for example, I visited my inlaws in the US, could have forwarded it to their number, or fired up a softphone, if I wanted to use it while there, etc.

Tons of options, the ways to use it are almost limitless. However don't let that scare away people who don't want to do so, for people that just want cheaper phone, and set and forget, you can do that too. No need to tinker, unless you want to. :)

For those that want to get crazy advanced, you can do things like setup sub accounts, and register different ATA's and have your kids phones ring in their rooms, (via caller ID for example) or setup extensions, so you can call room to room or even go as far as setup your own asterisk box (free PBX software) and have full "corporate" type features. Or how about setting up so that unknown numbers have to go through "press 1 to speak to Bob, 2 to speak to Mary, etc." They even have a reseller interface, in case people want to actually really be their own phone company. LOL :)

And if there's a feature you want, but they don't have, request it.

[jackie999]

Another feature I've used...when a voice mail message is received, you can have the .wav emailed to you so you can hear it right away :)
So many toys ...lol...

[Sneaks]

Another feature I've used...when a voice mail message is received, you can have the .wav emailed to you so you can hear it right away :)
So many toys ...lol...

one of my favourite's, as it goes to my BlackBerry, and I can listed to it on there right away.

[cmackie]

I'm pretty tempted to jump in on this - I've been looking to sign up with Voip.ms for awhile now.

My current DID with vbuzzer is active until the end of November that I prepaid for $25 for the year which includes incoming calls. I also have about $5 in credits left there for outgoing calls which I don't use very often.

I'd hate to lose what I have left there but vbuzzer has been seeming more and more sketchy lately. If I wait too long, I'll lose out on the promo rates.

Hmm... time to make a decision.

[jackie999]

Your post made me wonder if voip.ms refunds..
According to their Terms of service:

Refund policy: Swiftvox guarantee the client that its service will be working as offered in its website. A customer can ask for a refund on its unused balance under the present :
-If a customer made a first deposit and did not used its credits for the reason that he can't use the service , for any reason, Swiftvox will refund 100% of the deposited amount to the client.
-If a client made a first deposit and doesn't want to continue the service , for any reason, swiftvox will refund the unused part of the deposit ( initial deposit - usage by the client)
- If a client request for a refund on a deposit that is not its first deposit, swiftvox will refund the unused balance of the client , after charging a US$ 20.00 fee for administration and close client account.
All refund will be done in a delay of 1 week after client request.

I remembered, reading that, that the agent that was answering some of my questions on the livechat had told me that..I could get a refund of my initial deposit if I decided I didn't like the service.

[Sneaks]

Your post made me wonder if voip.ms refunds..
According to their Terms of service:


I remembered, reading that, that the agent that was answering some of my questions on the livechat had told me that..I could get a refund of my initial deposit if I decided I didn't like the service.

Cool, that should remove even more obstacles for those on the fence. If people aren't sure about porting their number, they could always jst try the outbound, without a DID, or get a new DID, play around and then port later, and cancel the original DID (or keep it and have a second #)

[BargainBuff]

Ok, I'm a really noob when it comes to this VOIP stuff. Can somehow explain whether:

1) You pay anything to forward calls from your DID # to your landline or cellphone? (i.e. am I paying per minute for calls that are simply just going THROUGH my DID?)

2) Do I pay anything to access my voicemail? (i.e. if I decide NOT to forward the DID but just use it as a voicemail account, will I incur anything beyond just the monthly fee?)

3) Do you pay to make outgoing calls AND receiving calls too? If so, how is this cheaper than a landline for local calling?

Sorry if my questions seem basic, but I'd just like to know what I'm getting into BEFORE I sign up.

[Sneaks]

1) You pay anything to forward calls from your DID # to your landline or cellphone? (i.e. am I paying per minute for calls that are simply just going THROUGH my DID?)


Yes you pay incoming 1 cent/min (billed in 6 sec increments) + outgoing (since you are both receiving and making a call) <-corrected this after I checked my bill



2) Do I pay anything to access my voicemail? (i.e. if I decide NOT to forward the DID but just use it as a voicemail account, will I incur anything beyond just the monthly fee?)
.

Yes you pay incoming 1 cent/min (billed in 6 sec increments)



3) Do you pay to make outgoing calls AND receiving calls too? If so, how is this cheaper than a landline for local calling?
.
yes, unless you sign up for the $5 unlimited (3500 min actually) incoming.
Cheaper, generally because the monthly charge is only $0.99, and then your exact usage, where as with primas, or Bell etc. would charge you lets say $20 a month (which includes your local incoming and outgoing) + long distance on top. You can do a lot of calling for $20 esp. when billed in 6 sec increments.

[jackie999]

3) Do you pay to make outgoing calls AND receiving calls too? If so, how is this cheaper than a landline for local calling?

I figured it out this way. I was paying $30 a month for my POTS (plain ole telephone service) with bell. Now I pay .99 cents a month. I also have to pay a $10 fee for dry loop since my internet service is DSL (if you are cable then you don't pay dry loop). I now also pay 1/2 a cent for calls going out and 1 cent for calls coming in. That means..to reach that $30 i was paying, if my calls are 50% incoming and 50% outgoing, I can do 34 hours of outgoing calls and 17 hours of incoming calls a month. I don't do anywhere near that amount of calling, so in my case, I'm ahead. You need to dig out a pencil and paper and do your own calculations to see if you'd be ahead.
Also, you aren't paying per minute, its every 6 second increments, and that makes a difference...it's not just the savings..it's also all the extras you get with it!
Hope this helps and doesn't confuse..I think in numbers :)

[polka]

Thanks.Don,t they have like flate rate for north america ?
I do have a device its lynksys which i am using with Acanac.The problem wih Acanac is that they don,t offer (cover)phone number of my city.I am using montreals area code.
I was wondering if i can get phone number with Cornwall area code.
There site is very complicated for me.

It's a Pay As You Go/Bring Your Own Device VoIP service. You can get a DID (inward number) for a monthly fee and after you only pay for your communication usage (~1¢ per minute depending of the options you choose).

voip.ms offers a lot of free services like voicemail, ring groups, multiple lines, IVR, call forward, DISA and many other. You also have a lot of flexibility on how you can configure your services.

I ported my Vidéotron number to them last June and I've used exclusively their service for my home telephone since. I'm very satisfied with the customer support and with the stability. I use a Linksys SPA2102 ATA and I've also used a few times the softphone "Telephone" on my MacBook with great call quality. My monthly bill went from $20 + taxes to ~$7 per month. It took me approximately 6 months to cover the cost of the ATA.

[ender78]

I have signed up for the service. Setup is a little difficult [would be really nice to have a FAQ that takes you through the setup of all the various service elements]. This is a process to have the "kids" and/or a computer savvy friend set up but once done, usage appears seamless.

I'm using a soft phone [Sip Communicator] for my Mac and so far it works great. Will pick up the Linksys ATA tomorrow.

WRT voicemail, is there a way to set up the VM so that I dont need to remember my mailbox number to retrieve VM from my cell phone and/or work line ? I would like to call my DID, enter password and listen to VM.

[abu_sme]

This is a whole level of cheapness in this thread. People are excited that they are saving $.50 for a service that only costs $1.50 to begin with...:lol:

Wow. $6 in savings per year, that's more than 40 minutes of work for a minimum wage employee!

[BargainBuff]

Thanks Jackie999 and Sneaks----your replies clarified all my questions. I'm gonna sign up as well. :)

[jackie999]

I was wondering if i can get phone number with Cornwall area code.


If you register on the site (it's free) then you can look around - see what numbers are available. If you decide to try it, you can either buy a DID (a phone number that people call you on) or you can keep your current phone number and *port* it over.

[kingfencer]

is long distance to other countries free?

[jackie999]

is long distance to other countries free?
No..the rates are listed on their main page
http://www.voip.ms/rates.php

[jackie999]

WRT voicemail, is there a way to set up the VM so that I dont need to remember my mailbox number to retrieve VM from my cell phone and/or work line ? I would like to call my DID, enter password and listen to VM.
When I first signed up..I was playing with the voicemail and I had to input mailbox/password. When I got my DID ported over and I went in and set up the voicemail and now my phone gives me my VM at the push of a button. This is something you could ask their live chat when it's available..the staff is amazingly helpful.

[kingfencer]

Here is my experience..from off the top of my head.

Sign up with voip.ms - paypal'd them $25 to start
Buy the ATA (PAP2T canadacomputers $54)
Read the configuration guide on voip.ms site to get the ATA working - it was no more difficult than setting up a router. If you want the ATA to jump thru hoops, then keep reading, it's very powerful. Otherwise, the basic configuration will get you going.
Plug your phone into the ATA. Make a few calls cause you're done.
If you want to receive calls, you can buy a number (DID) or you can *port* your existing phone number over (took about a week for me) ..cost $25..then $.99 cents a month for the DID.

You get all the bells and whistles. Call display..forward the Mother in Laws calls to "this line is no longer in service"...there are so many features, most of which I don't know what they do..so I keep reading :)

The ATA/modem must always be on..the computer does NOT need to be on.
Costs for calls out about 1/2 a cent a minute.
Costs for calls in about 1 cent a minute.
e911 service $1.50 a month - this is optional.
I also have to pay for a 'dry loop' for my teksavvy DSL - this, for me, is $10.22 a month.
All in all, I'm way ahead.
But I won't factor in the new panasonic phones I bought, or the APC backup :)

Hope this helps!

the paypal 25 bucks to start was to port your existing number right?

[Nraja]

no more 416 numbers!

[jackie999]

the paypal 25 bucks to start was to port your existing number right?
I put the $25 in just to try the service..then after an evening with it I was sold. So I applied to have my number ported and immediately got an email saying my account had gone into 'low balance' due to the port fee. It had in fact gone into overdraft since I'd used up .10 cents and the port costs $25 ..I fired off another paypal payment of $25 and I expect that will last me a year or more :)

[Sneaks]

the paypal 25 bucks to start was to port your existing number right?

You can use it for any services, if you're going to port, then yes.

[jackie999]

no more 416 numbers!

The 416 numbers have been on the endangered list for a long time now. That's one of the benefits of voip.ms..that they will port your existing number over..I don't think all voip companies will do that..so you can retain that rare 416 area code.

[kingfencer]

how did you port your number? say i have service with bell at the moment, does the company arrange it with bell? and does bell remove my telephone services? hate calling those morons.

[Sneaks]

how did you port your number? say i have service with bell at the moment, does the company arrange it with bell? and does bell remove my telephone services? hate calling those morons.

Go to DID numbers and select DID portability and then check to confirm your number is portable, and if so, start the procedure.

Basically you end up having to put in your information from your most recent bill, then they will do all the legwork and cancel Bell for you.

Do you have a DSL line as well? If so, then technically you need to contact your ISP and get it switched to a dry line. If you don't have that, and have rogers, for example then this step isn't necessary.

No need to call Bell to port. They'll just try to call you to win you back. ;)

[mrken]

I submitted a ticket today to reduce the rate and within a half hour or so, it had been done.

Now, I'm looking to port 2 other DIDs (currently with Unlimitel) and a toll free DID from telcan.net.

Just hoping they'd reduce the cost of porting, as it will cost me $25 x 3 ($75) and my current balance/credit with Unlimitel (non-refundable) of $50.
Just wondering, why are you porting away from Unlimitel?

By the way, Unlimitel has T.38 support for fax and routes international calls through Bell (if you choose the premium routes) so international faxes work fine too.

[Groove]

On the other end, the nice thing about callback service is that if your cellphone has free incoming but low-priced texting and you're really cheap (well, it *is* RFD) you can just send an SMS to your DID and the system will call you back with a dial tone. ;)


how does that work? when i text my voip.ms DIDs i get a thingy from Rogers telling me i need to use their text 2 voice service

p.s. voip.ms rocks

[jetway1212]

Just wondering, why are you porting away from Unlimitel?

By the way, Unlimitel has T.38 support for fax and routes international calls through Bell (if you choose the premium routes) so international faxes work fine too.

are you sure about the T.38 fax support? On their website they say they currently only support incoming fax. They will receive fax then email you the pdf file.

So this sounds just like voip.ms

[polka]

If you register on the site (it's free) then you can look around - see what numbers are available. If you decide to try it, you can either buy a DID (a phone number that people call you on) or you can keep your current phone number and *port* it over.

thanks
unfortunatly Cornwall number is not available.

[apn64]

Thanks.Don,t they have like flate rate for north america ?

At these rates, I don't think you WANT a flat-rate (outgoing) plan.

For example, Vonage Basic is $20/month, capped at 500mins usage. Yet on voip.ms, I average about 2,000min to Canada/USA/UK for about $12/month, so I get 4x the call time for nearly half the price. Capiche?

Why pay more than you have to?

BTW, they do have a flat-rate incoming plan; $4.95, but it only makes sense if your receiving >500mins incoming/month vs. the paygo rate of 1cent/min

[jackie999]

No need to call Bell to port. They'll just try to call you to win you back. ;)

Heh...and with your new voip.ms you can route these annoying spam phone calls from bell to a recording of your choice, or my favorite, have the system hang up on them :)

[BargainBuff]

Will I be able to port my DIDs on voip.ms BACK to Bell or Rogers if I decide to switch back?? I don't want to go through the expense of listing such DID numbers on my business cards and then not be able to keep the numbers if I want to use a PTNS line again. Does anyone know?

[Tinku]

I think you can port back the numbers but there will be a fee which is waived by Bell or Rogers.

[jackie999]

Will I be able to port my DIDs on voip.ms BACK to Bell or Rogers if I decide to switch back??

Yes, the CRTC passed a number portability tariff a few years back that insures you keep your number. There are some circumstances where I've read it isn't possible, but switching back and forth is fine, although you'll likely have to pay a fee.
My fees were $25 for voip.ms to set it up..and $20 for teksavvy (my DSL provider) to switch me to dry loop (so that really wasn't a porting expense).


I think you can port back the numbers but there will be a fee which is waived by Bell or Rogers.

Not surprising when you figure they're charging you $25 a month minimum.....my voip.ms is now costing me $12 a month (including my dry loop fee)...so the $25 fee I paid was well spent.

[djino]

I think the only case where you couldnt port back is if you had no money in your voip.ms account and your account went idle (unused) for a certain amount of months.

djino

[jackie999]

I think the only case where you couldnt port back is if you had no money in your voip.ms account and your account went idle (unused) for a certain amount of months.

djino

You're right..good point. The same thing happens with your cell phone number. If you choose a new provider and let the old number lapse, you loose your number there as well. I suppose this is why you let the new provider contact the old provider to port the number, so it remains active until it's ported over...

[BargainBuff]

Thanks for the replies. I suppose there really isn't much risk then. I was afraid to go through the expense of marketing a VOIP.ms DID and then not be able to use it if the company changes its services, prices, or worse yet, goes under.

[Jeffers]

Does voip.ms support ip phone? If so, do you still need to buy an ATA. Is there any advantage to one over the other?

[luckydog2006]

Does voip.ms support ip phone? If so, do you still need to buy an ATA. Is there any advantage to one over the other?

Yes. Advantage is that ATAs can be used with regular analog phones and are generally less expensive, and more available vs IP phones.

[SSingh]

I don't understand why they charge LESS for International calling using premium vs value ?

[jackie999]

I don't understand why they charge LESS for International calling using premium vs value ?

I downloaded both value and premium rate plans and compared side by side on one spreadsheet...at a quick scan, the premium is higher...where are you seeing it less?

[SSingh]

I downloaded both value and premium rate plans and compared side by side on one spreadsheet...at a quick scan, the premium is higher...where are you seeing it less?

Hi Jackie,

I just compared India Mobile at that time and didn`t compare other countries.

I also downloaded xls sheets now and compared few other countries and saw that rates on premium are higher but for calling India it`s vice-versa.

[Ben07]

dry loop

What is a "dry loop" is it something like using voip.ms without Shaw cable or Telus?

Thanks.

[djino]

What is a "dry loop" is it something like using voip.ms without Shaw cable or Telus?

Thanks.

Click here -> http://lmgtfy.com/?q=dry+loop

Dry loop is DSL Internet without an active telephone line.

For example, if you want Internet from Telus but you do not wish to get a Telus telephone line, then its infact a dry loop DSL connection.

djino

[jackie999]

Hi Jackie,

I just compared India Mobile at that time and didn`t compare other countries.

I also downloaded xls sheets now and compared few other countries and saw that rates on premium are higher but for calling India it`s vice-versa.

Yep..I see that now, you're right! I think, if it was me, I'd contact livechat before making any calls to India, and verify the rate beforehand. I saw in another post that some of the rates quoted weren't accurate...

Either value or premium to India mobile appears to be either .02 or .03 cents a minute..it doesn't sound like much does it..what does bell charge I wonder?

[jackie999]

What is a "dry loop" is it something like using voip.ms without Shaw cable or Telus?

Thanks.

My dry loop charge is $10.22 a month. Because my internet is via my phone line (DSL with teksavvy) I need to keep my phone line active. When I cancelled the telephone service, I still had to pay bell a fee just for using the 'copper' that they run into my house so I don't loose my internet.
You may say "Well why not get rid of teksavvy and get cable internet...then you can tell bell to take a flying leap..." But being an RFD'er I compared...
Teksavvy = $29.95 + dry loop $10.22 = $40 for 200 gig bandwidth and speeds that are fine for me.
Rogers = $36 for 36 gig bandwidth :( nope, not enough to keep me in linux distros...

[Ben07]

Dry loop is DSL Internet without an active telephone line.

For example, if you want Internet from Telus but you do not wish to get a Telus telephone line, then its infact a dry loop DSL connection.

djino


Thanks.

Currently I have Shaw cable for internet and cable TV, Telus for local phone. I 'm assuming by using Voip.ms I can ditch Telus and tag Vopi.ms to Shaw without any additional fees...what will happen should I cancel my Shaw internet in the future but still keeping Shaw for the cable TV service, will I still be able to use Voip.ms without any internet service? (or now I need a "dry loop" from Shaw, even I have the Shaw cable with TV service?)

Thanks again.

[jackie999]

what will happen should I cancel my Shaw internet in the future

VOIP=Voice Over Internet Protocol

[yakkers]

Thanks.

Currently I have Shaw cable for internet and cable TV, Telus for local phone. I 'm assuming by using Voip.ms I can ditch Telus and tag Vopi.ms to Shaw without any additional fees...what will happen should I cancel my Shaw internet in the future but still keeping Shaw for the cable TV service, will I still be able to use Voip.ms without any internet service? (or now I need a "dry loop" from Shaw, even I have the Shaw cable with TV service?)

Thanks again.

you NEED some type of high speed internet service to use ANY VOIP

[DragOnT]

Europe. :cheesygri

Every country has different regulations. Just google the country name and the term "Voip". Voip is much more commonly available in Europe as their land line phone companies frequently even charge for local calls.

if you don't know, keep it to yourself.

Try skype outcall and incall.

By the way, voip is common just about anywhere now - such as many "home phone" offerings by Rogers, Yak, Comwave, Freephoneline, etc etc.

[Ben07]

Thanks jackie999, yakkers!

http://www.redflagdeals.com/forums/staples-linksys-pap2-voip-9-99-a-353927/38/

I bought this Linksys PAP2 v2.0/Vonage for about $18 1/2 yrs back (I'm sure many RFD members also got this deal) which was successfully unlock with CYT-Unlocker and I tested this unit with the free 1 yr Acanac voip service 1/2 yrs back and it was perfect.

Now, if I'm to pay $25 for Voip.ms and get a new DID line for Vancouver @$1.99 and once I'm familiar with Voip.ms and want to port my Telus landline number to Vop.ms, can I keep both numbers/lines at the same time? (one is the new DID line I paid $1.99 and one is the ported Telus number/line, so now will be $1.99 X 2 with Voip.ms).

Thanks.

[SSingh]

Yep..I see that now, you're right! I think, if it was me, I'd contact livechat before making any calls to India, and verify the rate beforehand. I saw in another post that some of the rates quoted weren't accurate...

Either value or premium to India mobile appears to be either .02 or .03 cents a minute..it doesn't sound like much does it..what does bell charge I wonder?

Bell only charges 23c per min lol. http://www.bell.ca/web/wireline/en/all_regions/pdfs/LD/LD_InternationalExtra.pdf

[input]

I just bought a Linksys PAP2T-NA from Canada Computers and the setup up took me literally less than 30 minutes.

Once I signed up for an account at voip.ms, I deposited $30 into my account and setup the ATA as per voip.ms' configuration sample. I made a call to test it out and it had worked as planned.

To me, I couldn't hear a difference between voip.ms' premium and value rates.

Anyhoo,

Correct me if I'm wrong, but for me to get incoming calls, I would need to either buy a DID or port my own home phone number that is attached to Bell, right?

Thanks

[djino]

Question: I currently haven't made my initial $25 deposit, am I able to check voicemail and make SIP to SIP calls with no funds in the account?

djino

[luckydog2006]

Question: I currently haven't made my initial $25 deposit, am I able to check voicemail and make SIP to SIP calls with no funds in the account?

djino

No. They don't run a free business else they would not be able to pay their employees.

[jackie999]

Correct me if I'm wrong, but for me to get incoming calls, I would need to either buy a DID or port my own home phone number that is attached to Bell, right?

Right, the DID (Direct Inward Dialing) allows people to call you...once you have a DID you are assigned a mailbox for your voicemail.

[Kwirky]

Either value or premium to India mobile appears to be either .02 or .03 cents a minute..it doesn't sound like much does it..what does bell charge I wonder?

Voip.ms premium rates are occasionally cheaper than the value rates - I think it just comes down to what their providers charge, and in some areas their 'premium' provider can get better rates than their 'value' provider. Another example is their UK landline rates; premium is $0.007 and value is $0.0079


Now, if I'm to pay $25 for Voip.ms and get a new DID line for Vancouver @$1.99 and once I'm familiar with Voip.ms and want to port my Telus landline number to Vop.ms, can I keep both numbers/lines at the same time? (one is the new DID line I paid $1.99 and one is the ported Telus number/line, so now will be $1.99 X 2 with Voip.ms).

Yes, you can have as many DID numbers as you like with voip.ms, and doing it this way is a good lower-risk way of testing the service.

[signup]

Once you buy a DID is it possible to deactivate it when you decided to port your number? Just want to try their service first before I would port my number.

[apn64]

Once you buy a DID is it possible to deactivate it when you decided to port your number? Just want to try their service first before I would port my number.

Yes you can do that. However, there are easier ways to test the service w/o having to temporarily change your phone #;

1. Using a PC/other softphone application, you don't need to buy an ATA.

2. If you do buy an ATA ($50), you can use the service for outbound calls to check voice quality etc.

Unless you're making only toll-free calls, both the above options require you to fund your account @ $25 to get started and if you like what you see, you initiate a port request. Simple as that.

[signup]

Yes you can do that. However, there are easier ways to test the service w/o having to temporarily change your phone #;

1. Using a PC/other softphone application, you don't need to buy an ATA.

2. If you do buy an ATA ($50), you can use the service for outbound calls to check voice quality etc.

Unless you're making only toll-free calls, both the above options require you to fund your account @ $25 to get started and if you like what you see, you initiate a port request. Simple as that.

Great, thanks!

[apn64]

Correction: I just checked and learned that you do in fact have to fund the account to make ANY outgoing calls (even toll-free).

However, the voip.ms rep reiterated that they'll happily refund anyone not satisfied with the service.

[micron15]

Yes, you can have as many DID numbers as you like with voip.ms, and doing it this way is a good lower-risk way of testing the service.

Quick question...say i get 2 DID numbers from voip.ms under same acct. Are calls free between the two DID numbers?
Also, I have a fax machine that we occasionally use. Does voip.ms support faxing (sending/receiving) 100%?

[websavvy]

Quick question...say i get 2 DID numbers from voip.ms under same acct. Are calls free between the two DID numbers?


That doesn't make sense. DID numbers are for incoming calls.

[spamjack]

Just wondering how was everyone's experience with the system reliability? I had Acanac viop back in the 2006-2008 period and it was horrible. The server would go down around once every one or two months and that's when I had noticed the downtime. Who knew how many more times did it go down when I wasn't trying to make a call. On top of that they also made all kinds of billing mistakes and it was very diffiult to get it corrected. Tech support from India would blame billing in Canada and billing would correct it but needed tech support to perform the actual correction, etc. I guess you can imagine the amount of delays and mistakes. That made me nervous about low cost voip providers and i'm on a bell line now (thru teksavvy) and enjoying the same solid service i expect. I am sure a premium voip provider like vonage will be much better but then there isn't much or any saving over a bell line so i didn't want to go that route.

[spamjack]

I am also wondering if I buy an UPS power backup (like one of those APC units that goes on sale on dell.ca for around 50-70 dollars every other week) and hook up my dsl modem, wireless router, ATA unit, and a cordless phone; how long can the backup battery sustain?

[jackie999]

Just wondering how was everyone's experience with the system reliability?
It's pretty obvious I couldn't be happier with the service. I just looked at my account..I've been with them 1 month yesterday. My costs for one month (excluding $25 set up) were
"Calls: 108 | Duration: 06:24:37 | Total: $2.11449415"
So usage was $2.11 + DID $1.49 (which is now .99cents a month) = $3.60
My $25 investment is sitting at $21..should last me a while :)
Take THAT bell :lol:

As for the APC backup..I have one on order. All of my voice mail is to voip.ms, so no worries there if power goes out here. I expect it will run off a 350 APC for quite awhile..there isn't much draw - maybe somebody else will let us know.

I'm with teksavvy and connection is nice and stable. Only problem I have found..when I'm on phone, and I'm downloading something large, the connection is a little choppy..so I pause my download. There are ways to go into your settings in router to give voip priority over downloads..but I've been too lazy to look into setting that up.

[micron15]

That doesn't make sense. DID numbers are for incoming calls.
Forgive my ignorance, I'm still grasping this whole VOIP thing...I guess i was thinking along the lines of when line#1 makes a call to line#2 (both lines under the same voip.ms acct)....since the call originates and terminates on same network (voip.ms) and not routed through an external network say rogers/Bell...wouldn't the call be free? Something like free skype-to-skype calls, free Mj-to-MJ etc etc...doesn't this apply here?

[Webslinger]

doesn't this apply here?

No. You pay for everything with voip.ms

[Kwirky]

Forgive my ignorance, I'm still grasping this whole VOIP thing...I guess i was thinking along the lines of when line#1 makes a call to line#2 (both lines under the same voip.ms acct)....since the call originates and terminates on same network (voip.ms) and not routed through an external network say rogers/Bell...wouldn't the call be free? Something like free skype-to-skype calls, free Mj-to-MJ etc etc...doesn't this apply here?

- You can setup sub accounts on the voip.ms control panel and place calls between sub accounts by dialing an extension number you select (e.g. 101, 102). Calls between sub accounts are free.
- DID aren't lines in the traditional sense, but are numbers that can be routed to any or all sub accounts with different routing options for time of day, originating caller ID, failover, voicemail, etc. - they aren't limited to ringing just one line as with a traditional phone service.

[ARK]

I am with voip.ms for the last 2 years and never had any problems. Ported my bell number.

I also have freephoneline.ca line for everyday use and voip.ms is my second line. Did not choose e911 (which is extra $1.5/month) since freephoneline.ca number includes e911 service.

With my usage, deposit of $25 lasts for 8-10 months. I remember I was paying approx $40 to bell and now on average I am paying $5 per month( paid one time charge of $50 for freephoneline).

I would recommend this service to anyone.

[apn64]

No. You pay for everything with voip.ms

Actually, you DON'T pay for calls between sub-accounts on the same (master) account, nor do you pay for SIP URI calls (IP calling) to other voip.ms subscribers.

Today, if you call another account via the PSTN, then you will be charged standard rates, but the company has indicated that all calls to other voip.ms subscribers will be free in the near future.

[a2vr6]

Question for Previous Vonage customers that have ported there number from there to Voip.ms. I have a vonage locked motorola VOIP router, I plan on getting the Cisco adapter form CC's but I was wondering, are you notified by VOIP.MS that your number has been ported and the service has started?

[fogo]

Hi everyone,
I'm a Newb regarding VOip and anything related to it.

I have a few simple questions which I was wondering if someone
could answer.

I've read the thread but I'm confused.

I essentially want to decrease my LD charges and maybe replace my Bell line.

I live in Barrie and make most of my calls to people in Barrie, Newmarket and Toronto, with the occasional call overseas to Europe or South America.

I use Skype through the computer which is a pain in the a s s.

I've heard of Freephoneline.com and voip.ms but don't understand how they work.

Can anyone tell me what my options are?

Thanks in adavance

fogo

[death_hawk]

It looks like a lot of cost involve to jump on, i.e.
ATA - $54
Dry loop - $10
Dry loop activation fee - $20
Porting - $25
Voip.ms deposit - $25USD
Dry Loop only counts if you're on DSL.
And you don't NEED to port your number.
"Deposit" is more like depositing into a bank rather than a deposit on a bottle for example. The money is yours to spend.



Do it...succomb to the RFD pressure, you'll thank me...and if you don't like it it was the OP's fault. Hehe.. ;)

NO U!
lol


So it sounds like they don't have direct bank withdraw payments? Hopefully I can set up my online banking to transfer money every month automatically? (Im the worst at remembering to pay things monthly.) lol
Personally... I just shove $100 in there until it emails me complaining there's no money. It's easier than paying monthly.


Might consider, now researching for a good ATA.

So far how's the voice quality is it good as POTS, any echo?
I find it on par with POTS. Even better if you account for features.
PAP2T is what I like.


silly question but wouldn't the magicjack serve as a *cheaper* alternative? I mean you get all the bells and whistles with MJ and you get to choose your incoming number.

If your stuck paying $50+ for the ATA Device plus $15+/year for the DID service, you're still better off buying a MJ and that also gives you all the features. Only reason I brought this up was because someone stated the ATA must always be connected to the internet, well so does this.

MJ doesn't have any of the "advanced" features. Any idiot can offer CID,VM,conf,forwarding,etc but almost no one offers DISA,IVR,blacklisting,etc.

No one ever said this is a budget option. It's cheap and functional. The MJ is just cheap.
Oh and you don't need a computer on (for those who actually turn them off) for an ATA to be used. Assuming your computer uses 100W, that's an extra $7.20 per month assuming $0.10 per kWh (what? I pay a lot for power... stupid lack of hydro)

Oh and you don't get cool stuff like toll free or multiple DIDs.



From what I understand, the CNAM is hit and miss. I *think* when a call is made, if you've requested the associated Caller ID, then the system must search another database..this 1.25 cent charge you pay is what they pay for the search...once it's searched for the CNAM, it'll retain it for 24 hours so you don't pay again that day for same name.
I read on dslforums you can contact support and they will make sure your number is recorded in this database so future searches will show in the CNAM.
If I'm wrong somebody correct me.

Assuming I'm reading this right, cell providers don't pass on a proper CID but pass on a "general rate center" for billing purposes.
I haven't been able to get toll free to pass a proper CID.
Then again... I don't pay the $0.0125.
I should flip it on to see what happens.


Europe. :cheesygri

Every country has different regulations. Just google the country name and the term "Voip". Voip is much more commonly available in Europe as their land line phone companies frequently even charge for local calls.
As mentioned, you can get DIDs around the world.



Correct me if I'm wrong, but for me to get incoming calls, I would need to either buy a DID or port my own home phone number that is attached to Bell, right?

Thanks
Yup. Unlike "traditional" phone systems (including other VoIP) this is a professional solution. This means that outgoing and incoming calls are run separately. Unless you're on the residential unlimited plan, you can receive unlimited* simultaneous incoming calls (res = limit 2) on the same number.

*Bandwidth pending


Voip.ms premium rates are occasionally cheaper than the value rates - I think it just comes down to what their providers charge, and in some areas their 'premium' provider can get better rates than their 'value' provider. Another example is their UK landline rates; premium is $0.007 and value is $0.0079

Yeah I noticed that too.


Once you buy a DID is it possible to deactivate it when you decided to port your number? Just want to try their service first before I would port my number.
There's a 1 month minimum commit.
I accidently ordered 2 DIDs and cancelled one. Apparently it's not refundable, but I could cancel it easily.


Just wondering how was everyone's experience with the system reliability?
Billing is spot on.

Reliability is awesome too. They have very little downtime. Hell... if you set up redundant routing you can actually have ZERO downtime.


I am also wondering if I buy an UPS power backup (like one of those APC units that goes on sale on dell.ca for around 50-70 dollars every other week) and hook up my dsl modem, wireless router, ATA unit, and a cordless phone; how long can the backup battery sustain?
Depends on the UPS.
I have a 1500VA one and I can sustain communications for about 8 hours. I could extend this if my routing equipment wasn't a computer.


Forgive my ignorance, I'm still grasping this whole VOIP thing...I guess i was thinking along the lines of when line#1 makes a call to line#2 (both lines under the same voip.ms acct)....since the call originates and terminates on same network (voip.ms) and not routed through an external network say rogers/Bell...wouldn't the call be free? Something like free skype-to-skype calls, free Mj-to-MJ etc etc...doesn't this apply here?
Technically no. EVERYTHING is billable. But apparently they did roll out free voip.ms to voip.ms calls.


Question for Previous Vonage customers that have ported there number from there to Voip.ms. I have a vonage locked motorola VOIP router, I plan on getting the Cisco adapter form CC's but I was wondering, are you notified by VOIP.MS that your number has been ported and the service has started?
Yes. They send you an email saying "port complete".

[kcormier]

Question for Previous Vonage customers that have ported there number from there to Voip.ms. I have a vonage locked motorola VOIP router, I plan on getting the Cisco adapter form CC's but I was wondering, are you notified by VOIP.MS that your number has been ported and the service has started?

I switched from Vonage a couple years ago...

Once you initiate the porting process on voip.ms, your account will be updated with a cutover date and you will see progress updates online (also sent to your email) as the port takes place and "moves through the system" to completion.

[death_hawk]

Oh I forgot to mention, once you pair up a PBX to this system, the awesomeness goes up 10 fold.
voip.ms lets you use their PBX but running your own is like putting some more icing on your bucket of icing. I am literally my own phone company.

Long story short, I have my PBX in the same DC as voip.ms so my latency for non "VoIP" calls is literally 1ms. (IE PSTN to VoIP to PSTN when for example someone calls me and my number is forwarded)
You don't NEED to have your PBX in the same DC, but it's nice. For those of you who live near one of voip.ms servers, you definitely don't.

Oh and running a PBX means literally FREE "calls" between your ATA/Softphones. Instead of dialing 123-456-7890 I can dial extensions (IE. 200) to make free calls completely outside the voip.ms network.
So if you have a friend in let's say Britain, give them an ATA or softphone and connect them to your PBX. Now you can dial 201 and reach them directly. Hell you could purchase let's say a Toronto DID and route that to 200 so calls to "Toronto" reach your ATA in Britain.

[jackie999]

Great service getting even better :)

copied from dslreports,

The servers are on their way, they are all in testing, the changes you see in the portla is to "harmonize" the order process and to be able to add additional servers on the fly in the future. We'll also add additional pops to existing locations instead of load balancing them. We're getting delayed, i'm sorry if I advertised them for last week, but we're trying to get them online as fast as possible.

Incoming Phase 1 (In testing):
Chicago
Seattle
Tampa
Singapore
Atlanta

Incoming Phase 2:
Toronto 2
Montreal 2 and 3
Houston 2 and 3
New York 2

The point "hidden from the user" is always quick and top notch. We don't use crappy providers, and if we found out that a route to a SIP TRUNK to a specific trunk takes more than a certain amount, we usually proxy with one of our OpenSER servers to create a shortcut.

I know this can sound very vague but I know for sure our main sys admin is a perfectionist, every trunking with the providers are tested for latency, jitter, number of hops, and more before getting pushed live.

Again sorry for the delay, I hope the new servers goes online this week!

Regards
--
Martin - VoIP.ms

[vint]

I would like to know if the following scenario is possible. Pros only please! :cheesygri

Can I purchase a DID from a neighboring country B and then use it to make calls to call country A?

I know what DID and what the purpose is. However I would like to check the rate calling from closer country because the current rate is really high.

[nosnoop]

I gather voip.ms is pretty good in terms of reliability.
But what concerns me more is my internet connection.

I can't recall my POTS has ever had any down time, ever. But my Rogers cable extreme can go offline once a while. Sometimes for a few minutes, sometimes longer.

How are you guys coping with this? Does your broadband connection have rock solid reliability? Or do you have back up/redundant connections?

I am very tempted to switch my phones to voip, but the worry about reliability (especially regarding internet connections, and also in case for my business use which cannot afford any downtime at all) hold me back.

[jackie999]

But my Rogers cable extreme can go offline once a while. Sometimes for a few minutes, sometimes longer.

That surprises me..I thought cable was stable...if you're watching TV via your cable does the picture go blank now and then?
My teksavvy DSL is solid, sure..there may be downtimes...I can't remember one offhand...but $hit happens. My VM is on voip.ms - so even if I did loose my phone during an outage, I'd still have VM backup.
I'd think if your internet connection isn't good - then VOIP isn't for you...

[death_hawk]

I would like to know if the following scenario is possible. Pros only please! :cheesygri

Can I purchase a DID from a neighboring country B and then use it to make calls to call country A?

I know what DID and what the purpose is. However I would like to check the rate calling from closer country because the current rate is really high.
DID = Incoming
Technically you don't even need a DID to call Country A (or C or Y)
voip.ms termination rates apply to any server they have.
So let's say you're in Country C and connect to Dallas, you can call Country B at $0.01 per minute. If you decide to move to Country A, you still connect to Dallas (or Toronto) and call Country B for the same rate.

What I just said sounded good in my head, so I hope it makes sense.


Seattle
F'in EH!
I can't wait. I can finally colo a server in Seattle so my VoIP calls don't have Dallas type latency.



Or do you have back up/redundant connections?

I am very tempted to switch my phones to voip, but the worry about reliability (especially regarding internet connections, and also in case for my business use which cannot afford any downtime at all) hold me back.

I'm redundant.

For business use, (depending on volume) you could contract out a T1 line. It runs about $500 per month, but there's a SLA attached.
With the proper codecs, you can hit a bucket of calls at one time that justifies the $500 per month.

If it's just a store that has a line or two, redundant internet is still cheaper than two lines.

[fogo]

Hi everyone,
I'm a Newb regarding VOip and anything related to it.

I have a few simple questions which I was wondering if someone
could answer.

I've read the thread but I'm confused.

I essentially want to decrease my LD charges and maybe replace my Bell line.

I live in Barrie and make most of my calls to people in Barrie, Newmarket and Toronto, with the occasional call overseas to Europe or South America.

I use Skype through the computer which is a pain in the a s s.

I've heard of Freephoneline.com and voip.ms but don't understand how they work.

Can anyone tell me what my options are?

Thanks in adavance

fogo

Can anyone hold my hand or point me to the right direction?

[jackie999]

I'll give it a shot..although I'm not sure where to begin. First, if you have an internet connection on 24/7 you can use that for your phone. Cost will be .99 cents a month and you pay 1/2 a cent to call out and 1 cent a minute to receive calls..that's in Canada, international rates are on the voip.ms site. You can get a new phone number or pay to retain your existing number. Easiest way is to buy an ATA (about $50)..plug it into your existing router (you have one?) and plug your phone into that. If you have wireless phones throughout your house you're all set. You can feed the voip throughout your existing phone lines in your house wiring but I don't know how complicated you want to get...
That help?

[vint]

DID = Incoming
Technically you don't even need a DID to call Country A (or C or Y)
voip.ms termination rates apply to any server they have.
So let's say you're in Country C and connect to Dallas, you can call Country B at $0.01 per minute. If you decide to move to Country A, you still connect to Dallas (or Toronto) and call Country B for the same rate.

What I just said sounded good in my head, so I hope it makes sense.


Thanks for reply. However what I meant is lets take the following example: calling LD from country A to country B is really expensive. I mean like 30c/min expensive even with VOIP. Now if I could somehow use country C which is much closer to make calls (then again I'm not 100% sure it will be cheaper but I would like to find out) or it doesn't really matter? I hope it makes sense.

[vint]

I'll give it a shot..although I'm not sure where to begin. First, if you have an internet connection on 24/7 you can use that for your phone. Cost will be .99 cents a month and you pay 1/2 a cent to call out and 1 cent a minute to receive calls..that's in Canada, international rates are on the voip.ms site. You can get a new phone number or pay to retain your existing number. Easiest way is to buy an ATA (about $50)..plug it into your existing router (you have one?) and plug your phone into that. If you have wireless phones throughout your house you're all set. You can feed the voip throughout your existing phone lines in your house wiring but I don't know how complicated you want to get...
That help?

connecting other phones is as easy as plugging the phone cord from ATA to the wall given your phone line is disconnected and there is is only silence.

[SSingh]

Is Linksys PAP2T the best thing in the market for VOIP for two lines ?

[BargainBuff]

So, let me get this right----I will only pay 0.5 cents/min to call OUT and 1 cent/min to RECEIVE calls? I thought I somehow was paying 1.5cents a minute----something about receiving and sending data at the same time because of the way VOIP conversations work...

Can someone please explain this and what a "termination rate" is?

[ganiba]

Can anyone hold my hand or point me to the right direction?

Hi fogo,

I think that you only want to call people for cheap and not have a local number from different city to receive calls from others.

So, if you are a real newbie and happy with Skype but don't like each time to use the desktop, i suggest to buy this one:

Belkin Desktop Internet Phone for Skype ~108$
http://shop.skype.com/intl/en-ca/phones/belkin-desktop-internet-phone-for-skype.html

http://shop.skype.com/i/images/products/Phones_Belkin_Desktop-230x230.jpg

And subscribe to Unlimited calls to United States and Canada landlines and cell phones for 2.95$/month
http://www.skype.com/allfeatures/subscriptions/?region=uscanada#mexicanSubscriptionTab

You will also be able to talk with Skype users if they are online.

Good luck.

[death_hawk]

Thanks for reply. However what I meant is lets take the following example: calling LD from country A to country B is really expensive. I mean like 30c/min expensive even with VOIP. Now if I could somehow use country C which is much closer to make calls (then again I'm not 100% sure it will be cheaper but I would like to find out) or it doesn't really matter? I hope it makes sense.

Country B is $X.XX per minute regardless of the originating Country.
Check the termination rates to figure out exactly how much it costs to call Country B.



So, let me get this right----I will only pay 0.5 cents/min to call OUT and 1 cent/min to RECEIVE calls? I thought I somehow was paying 1.5cents a minute----something about receiving and sending data at the same time because of the way VOIP conversations work...

Can someone please explain this and what a "termination rate" is?

You place a call and it's $0.01 (I pay $0.015 for Prem)
You receive a call and it's $0.015
The above costs are per minute and billed "once" aka not for both you talking and you hearing.

"termination rate" is how much you're paying to "terminate" (aka call) that particular country.

[ganiba]

Thanks for reply. However what I meant is lets take the following example: calling LD from country A to country B is really expensive. I mean like 30c/min expensive even with VOIP. Now if I could somehow use country C which is much closer to make calls (then again I'm not 100% sure it will be cheaper but I would like to find out) or it doesn't really matter? I hope it makes sense.


So, let me get this right----I will only pay 0.5 cents/min to call OUT and 1 cent/min to RECEIVE calls? I thought I somehow was paying 1.5cents a minute----something about receiving and sending data at the same time because of the way VOIP conversations work...

Can someone please explain this and what a "termination rate" is?

Rates of VOIP calls depend only on destination not the call origin, even if you are in Canada, and you want to call someone in Canada you will be paying the same rate as someone calling Canada from Australia.

That's why, we call it "termination rate".

[ganiba]

So, let me get this right----I will only pay 0.5 cents/min to call OUT and 1 cent/min to RECEIVE calls? I thought I somehow was paying 1.5cents a minute----something about receiving and sending data at the same time because of the way VOIP conversations work...

Can someone please explain this and what a "termination rate" is?

If you are a newbie like me, with voip.ms. On their website, there are two options to Route your calls: Over a standard link (Value) or a premium link (Premium) with better calls quality, i guess.
So, you have to go to "Main menu> Account Setting", there you will find Routing options.
If you select for instance "USA48/Canada Routing" as 'Value', you will pay 0.5cents to make calls, but if you select premium you'll pay 1.25cents.

On my account the default was "Premium".

Thanks to OP.

[fogo]

Thanks ganiba

I know that option exists, better yet a wireless version of that phone would be a better option. ganiba, I'm looking to have people call me and I call others in the 416 - Toronto 705 - Barrie and 905 - New market area codes. On top of that I want to get rid of my Bell line. That's why voip.ms sounds interesting.



Thanks jackie999

Let me ask you or anyone else a question about voip.ms.

Can I buy a Barrie Did number, a Toronto Did number and a Newmarket Did number from voip.ms, connect my ATA device to my router and dect phones and be able to receive and call Barrie, Toronto and Newmarket numbers for less than $20 per month?

[Sneaks]

So, let me get this right----I will only pay 0.5 cents/min to call OUT and 1 cent/min to RECEIVE calls? I thought I somehow was paying 1.5cents a minute----something about receiving and sending data at the same time because of the way VOIP conversations work...


For normal calls, no you only pay once.

If receiving a call, then 1c/m
If placing a call 0.5c/m (assuming Can/US - other country rates posted)

Your confusion may have come from when I was answering someones question about forwarding costs. Only time you pay both is in the case of forwarding to another number, for example a cellphone, etc.

In that case you are both receiving the call to your DID, and then placing one to the forwarded number, you cell, for example

[Sneaks]

Can I buy a Barrie Did number, a Toronto Did number and a Newmarket Did number from voip.ms, connect my ATA device to my router and dect phones and be able to receive and call Barrie, Toronto and Newmarket numbers

Yes, you can get 3 DID's, and then set it such that all three would ring your phone(s) - You can set it all to ring the same phone, but our outbound caller ID though you would typically set to only one in that case. If you have a multi-line phone and then could set each one to a different outbound DID caller ID - just would need to have an ATA that had 3 ports - the PAP2T only has 2 so you'd need 2 of those in that case (others may know some more advanced configs though.)


for less than $20 per month?
Yep (depending on usage)
3x DID =$2.98/month +usage.

[fogo]

Thanks Sneaks for responding.

So let me get this straight.

I need a ATA device with three ports, one port for every phone which would handle each Did??

In other words, I'd have to have a phone to make/receive Barrie calls, one phone to make/receive Newmarket calls and one phone to make/receive Toronto calls.
Seems excessive!

Couldn't I just have a regular ATA device hooked to a dect phone system like this http://www.costco.ca/Browse/Product.aspx?Prodid=10323240&whse=BCCA&Ne=4000000&eCat=BCCA|79&N=4008779&Mo=22&No=2&Nr=P_CatalogName:BCCA&cat=82&Ns=P_Price|1||P_SignDesc1&lang=en-CA&Sp=C&topnav=
hooked to my router in the basement with handsets all over the house allowing me to make and receive calls from all 3 Dids?

Cheers and thanks for you help.

fogo

[jackie999]

make and receive calls from all 3 Dids?


Fogo..I think you're making it more confusing than it actually is..or maybe I'm not understanding you. Why do you want 3 DID's? Do you have 3 bell telephone numbers now? Can't you just buy one DID (or port your existing number over to voip.ms) and make all your calls with it? Call Newmarket, Barrie, Toronto.. all for 1/2 a cent a minute. If people call you, it will cost you 1 cent a minute. If you call overseas..it's a little more..but no where near the cost of Bells rates....
Hope this helps!
Oh, and yes, that DECT phoneset will work just find. I bought the panasonic DECT handsets a month ago - thanks to an RFD hotdeal :)

[BargainBuff]

Many thanks to all that clarified the incoming/outgoing rates confusion; I now understand, especially why forwarding would be paid 2 times. This VOIP stuff truly is pretty cool; I'm quite impressed by all the features that voip.ms has---I wish their website had a online manual so that newbs like me can get better acquainted with all the bells and whistles. Thanks again to all who've helped me out!

[fogo]

jackie999

Again thanks so much for responding.

The Toronto Did and Newmarket Did would be for inlaws in order for them to call without incuring liong distance fees

My parents and inlaws are old school and cheap. By having a Toronto and Newmarket number we hope they would call more.

jackie999

The thing that I can't get believe is that I can replace my Bell line for next to nothing. Wow, but for some reason I'm a bit hesitant.
Is having my landline replaced by a voip line that much of a jump?

[RMSimard]

how does that work? when i text my voip.ms DIDs i get a thingy from Rogers telling me i need to use their text 2 voice service

For a while my gf used to send texts to get callbacks from her voip.ms DID using Telus mobile paygo; she's now using DISA. I guess Rogers behaves differently.


I need a ATA device with three ports, one port for every phone which would handle each Did??

Nope. You can have them all pointed to one ATA device with one phone.

As for stability of professional-grade VoIP in general: call centers and larger businesses have been using that for years, costs less and scales better. They wouldn't if it wasn't reliable.

As the for low price, it's because underneath it all telephony service really is that cheap. The $30-50 a month you're paying Bell & co. is partly for legacy hardware maintenance, personnel and substantial profits. That archaic business model of theirs is definitely doomed ...whenever SIP-based VoIP gets *much* easier to configure and understand for the common folks. Now that might take a while!

[Netcruzer]

A sale for new customers only is the kind of thing you expect for the 'regular' telco players, not Voip.ms. :(

I tried to get the new rate, here's my support ticket response:

"I\'m sorry this only apply to the new carrier numbers we started selling a few months ago. Allstream numbers (Your Ottawa) are not eligible due to the higher rate we pay for those. I apologize for the inconvenience.

Regards

__
Steve Poirier
Dir. Technical Operations
VoIP.ms"

[jackie999]

Is having my landline replaced by a voip line that much of a jump?

It's just progress.
I too am 'old school'. My first phone was a 'party line' (you shared with your neighbours). There were no cell phones then. Your phone line was your life line, it was as important as heat,hydro, water etc. Making the switch to voip.ms was difficult for me too. I don't accept change easily. I started looking into this voip to recoup the costs the HST is going to make to my budget and decided I had no choice. I had to take make the leap and now I'm so glad I did.
The grass IS way greener over here :)

[KA24DE]

I received the email regarding this promotion as well. I emailed them and they adjusted the rate down to $0.99 for my existing DID. Excellent Customer Service!

[Sneaks]

Thanks Sneaks for responding.

I need a ATA device with three ports, one port for every phone which would handle each Did??


fogo

No you don't need to have multiple ATA's, for each DID, you could receive all three via one. Sorry if confusing, the slight nuance is that if you want to set your outbound callerID via one port, this is where I'm not sure you can do without reconfiguring your ATA, or putting in a code, or something to select which one to display. Perhaps someone smarter than I can suggest an alternative. :)



The Toronto Did and Newmarket Did would be for inlaws in order for them to call without incuring long distance fees


Very posbile: what I suggest for this scenario

Get your 3 DID's
Configure your ATA for your "primary" number, including your CallerID.
Route the other 2 numbers to your main account
You could then recieve calls from all 3 DID's, giving out yout Newmarket and TO numbers to the respective people, so they don't incurr LD charges.
However your outbound callerID would display your primary number, that's all.

Again, others may know some more tips and tricks, to do a selectable CallerID - or just don't set one.

[jackie999]

One thing that springs to mind here...you could program your inlaws phone to display your name instead of a number - do all new phones do this - my new DECT phones do.
So, when you call the in-laws it will say 'fogo' calling, instead of some number that they don't recognize.

[Webslinger]

Actually, you DON'T pay for calls between sub-accounts on the same (master) account, nor do you pay for SIP URI calls (IP calling) to other voip.ms subscribers.

I could be mistaken, but I doubt that's exactly what micron15 was referring to.


Today, if you call another account via the PSTN, then you will be charged standard rates, but the company has indicated that all calls to other voip.ms subscribers will be free in the near future.

That I wasn't aware of. Thanks for the info

[thedonald]

Can anyone comment on how the voice/connection quality is compared to Acanac? I know that price/features would beat Acanac...but I was wondering about the quality of the connection...

Thanks,

--TheDonald

[jackie999]

This thread is getting pretty long..I guess people don't want to read all through it. Many asked about the quality, many responded that it's good. I saw a few other posts about people unhappy with acanac and thinking of switching..if you search the thread you'll find your answer ;)

Search over at dslreports shows this,
http://www.dslreports.com/forum/r22094889-Acanac-VOIP-Good-or-Bad

Is acanac 'home phone' voip?

Nice thing about voip.ms is you can keep your existing number.

[thedonald]

No you don't need to have multiple ATA's, for each DID, you could receive all three via one. Sorry if confusing, the slight nuance is that if you want to set your outbound callerID via one port, this is where I'm not sure you can do without reconfiguring your ATA, or putting in a code, or something to select which one to display. Perhaps someone smarter than I can suggest an alternative. :)



Very posbile: what I suggest for this scenario

Get your 3 DID's
Configure your ATA for your "primary" number, including your CallerID.
Route the other 2 numbers to your main account
You could then recieve calls from all 3 DID's, giving out yout Newmarket and TO numbers to the respective people, so they don't incurr LD charges.
However your outbound callerID would display your primary number, that's all.

Again, others may know some more tips and tricks, to do a selectable CallerID - or just don't set one.


Why not just get a toll-free number from VOIP.MS? Instead of three seperate DIDs?

--TheDonald

[paulo500]

So im signing up to their website, and im activating my account. Did everyone have to email them a scanned image of some form of ID? The online chat guy said for me to scan my driver's licence or passport...(I can block out any personal info he said.) So this is just to verify my address?

"The system has set your account on mode "Manual Validation Required". Please contact our staff by Live Chat or Email to have your account activated."

[wally_walrus]

Why not just get a toll-free number from VOIP.MS? Instead of three seperate DIDs?

--TheDonald

+1... best advice so far

[jackie999]

So im signing up to their website, and im activating my account. Did everyone have to email them a scanned image of some form of ID? The online chat guy said for me to scan my driver's licence or passport...(I can block out any personal info he said.) So this is just to verify my address?

I had to send a scan of my most recent phone bill with a signature on it...this was to port my number over from teksavvy home phone....maybe some people have tried to port someone elses number over?

[Shaf]

So im signing up to their website, and im activating my account. Did everyone have to email them a scanned image of some form of ID? The online chat guy said for me to scan my driver's licence or passport...(I can block out any personal info he said.) So this is just to verify my address?

"The system has set your account on mode "Manual Validation Required". Please contact our staff by Live Chat or Email to have your account activated."

I had to scan a copy of my DL and send it to them. I just took a pic and attached it....

[Sneaks]

Why not just get a toll-free number from VOIP.MS? Instead of three seperate DIDs?

--TheDonald

They only currently offer US DID's, and while DID is only .99, inbound is .029/min and you'd just have to ask support to unlock canada calling, which is apparently charged at .08/min.

[Sneaks]

I had to send a scan of my most recent phone bill with a signature on it...this was to port my number over from teksavvy home phone....maybe some people have tried to port someone elses number over?

Ditto, didn't do DL from what I recall. Process might be different if you're not porting like I was, but requesting a new DID.

[thedonald]

I had to scan a copy of my DL and send it to them. I just took a pic and attached it....

Yeah, I just got off a chat with them. They said it was because I was using a CAD address but signed up from Italy. Which is true. So, I sent him a copy of my passport and they said it wasn't clear enough... I sent 4 different files... apparently they all needed to be higher resolution!?!?!??!? A 500k jpg of my passport pic is not good enough? WTF!

They activated my account but no paypal or CC access to reload! So, I have a pointless account! Until I send them a higher resolution jpg!

Anyway, I have given up with them. Any higher resolution and it would have been real life!

--TheDonald

[fogo]

A big thank you goes to all who helped out by posting.

I really do appreciate your help.

I'm almost there but I have a few more questions.

What kind of dect phone system do I need that would allow me to answer all 3 Dids through one phone?

What ATA device would you recommend?

Correct me if I'm wrong, but a Did allows people to call you through a local number. For example, a Newmarket Did would allow people to call me at no cost to them but I have to pay the 99 cent per month fee and an incmong fee of $0.0100 per minute. Correct?

Using the same Did which I'm paying 99 cents per month I could call friends in Newmarket and pay the additional termination fee of 0.00503118 per minute. Correct?


How am I doing?

[BillyH]

This may have already been addressed... excuse if so.

Does callback double charge like DISA does?
(once to call you back and once to route your call?)

Thanks

[sharjeel]

i had an ATA i bought earlier, but was still looking for a good provider. setting up voip.ms was a breeze, and their customer service was great (i had issues with the proxy). highly recommended. i think people should consider freephoneline.ca as well.

[Sneaks]

What kind of dect phone system do I need that would allow me to answer all 3 Dids through one phone?

Its not a function of DECT per se, that's really just the wireless technology used to broadcast between your base and the handsets. Technically you could do it with any phone (rotary might put you out of luck though on dialing at least) ;) However to answer all 3 lines at the same time, then I believe you would need a multi line phone (and this is where you potentially get into the need for multiple ATA ports.) I don't have a DECT phone system at this time though so I'll let others pipe up on any recommendations/feature commentary



What ATA device would you recommend?


I didn't do a ton of research, but I chose the Linksys PAP2T-NA because it was cheap, popular and readily available ATA that many seem to use and as a major bonus configuration samples right on the voip.ms site, thus reducing guesswork on the setup (not that I couldn't figure it out, but hey why re-invent) :)



Correct me if I'm wrong, but a Did allows people to call you through a local number.

For example, a Newmarket Did would allow people to call me at no cost to them but I have to pay the 99 cent per month fee and an incoming fee of $0.0100 per minute. Correct?

A DID is just an inbound number people call you on. So what they pay is a technically a function of their own phone plans. However generally yes if you get a DID in their local calling area, then it's free for them to call it and the charges in your example is right on the money for what $ you would pay. :)



Using the same Did which I'm paying 99 cents per month I could call friends in Newmarket and pay the additional termination fee of 0.00503118 per minute. Correct?


Technically doesn't matter which Ontario DID you choose, local and long distance would be the same, but yes again correct in your example.



How am I doing?


Just fine. :)

[theSPOOLER]

Do you get to pick your DID from a bunch of options, or does it just generate one and assign it to you?

[Sneaks]

because I was using a CAD address but signed up from Italy.

Must of been afraid you were part of the mafia up to some good old scheming. ;)

Seriously though, sorry to hear your troubles, but I guess not always as easy for every situation.
Did sound a bit odd though why they kept having problems reading it.

[Sneaks]

Do you get to pick your DID from a bunch of options, or does it just generate one and assign it to you?

You can view the available numbers first and choose, but depending on the area may limit how many numbers you have available to choose from. I just randomly sampled a couple, and Lively, ON you can only pick from 7, Brampton, like 20, and TO a few screens worth.

[ganiba]

Do you get to pick your DID from a bunch of options, or does it just generate one and assign it to you?

bunch of options, but not too many in my city.

[jackie999]

For the $25 fee..I think it's money well spent to just port your existing number over. My 905 area code is becoming rare..so I'll hang on to it as long as I can.

[destinet]

Thanks to all, just start the procedure too port my number from vonage to them. I calculate last month vonage charged me 23$ and with them it will have cost 6$...ridiculus.

Can't wait to test all the option now;)

[Sneaks]

Thanks to all, just start the procedure too port my number from vonage to them. I calculate last month vonage charged me 23$ and with them it will have cost 6$...ridiculus.

Can't wait to test all the option now;)

Cool, good luck, curious how you find it compares in the end to vonage.

[jackie999]

A great company just keeps getting better and better :)

Introducing lower pricing and more accurate results for CNAM queries:

We have implemented a new CNAM Provider, in an effort to provide more accurate results to the customer base at a lower price

After intense testing of the new CNAM provider, we're glad to announce that we'll be implementing the new provider on May 1st,, and the new rate for CNAM Queries will go down from $0.0125 to $0.008, to be more precise, from 1.25 cent to 0.8 cent (eight tenth of a cent)

Please take a moment to read the following points:

- New Cnam-query price, with savings of 36% over the previous query price (CNAM Queries have been lowered from $0.0125 to $0.008)

- The new provider offers more accurate results.

- We've not been able to find a reliable provider that would allow us to cache CNAM Queries on a 24 hours period due to their contractual obligations. For this reason, we can no longer cache queries on a per 24 hours period.

- The 24 hours caching change will go live on May 1st, 2010.

- The pricing change is going live now.


If you have a certain list of contacts calling you a frequent basis, and wish to avoid repeated cnam-query charges on your account, we advise to add the caller's name and number to your VoIP.ms phonebook, there are no charges for queries made from your phonebook.

Thank you for your understanding.

Regards,

VoIP.ms Staff

[tenchi]

for people who have successfully ported from Bell Landline.
What Service Provider Address did you specify in the request form?
The address that is on the payment stub of the bill (P.O. Box 9000 Stn. Don Mills) or some other one?

Thanks

[George W. Bush]

Dudes - why not stick with a CANADIAN company - Acanac- pay $9.95 a month for unlimited Canada/US use with all kinds of features?

I don't get this...

[signup]

Dudes - why not stick with a CANADIAN company - Acanac- pay $9.95 a month for unlimited Canada/US use with all kinds of features?

I don't get this...

Still too much for casual user...

[RMSimard]

Dudes - why not stick with a CANADIAN company - Acanac- pay $9.95 a month for unlimited Canada/US use with all kinds of features?

I don't get this...

Dude, voip.ms *is* a Canadian company. It is based around Montreal and has bilingual staff.

[mondriano]

As a brand-new customer, I can only hope that their voip service is equal to their customer service. I had a number of questions regarding many different aspects of voip in general, as well as their specific service. Detailed responses in less than 5 minutes?! Wow. Impressive.

[Hugh]

This sounds good.

In my mind, this kind of service is designed for folks that understand VoIP. All the working parts are exposed, a la carte. That's a good thing but rather a shock for folks accustomed POTS (Plain Old Telephone Service). RFD members may just be motivated enough to get over the learning curve.

If you ignore the difficulty learning about this system, I think that this is a great bargain compared with POTS or Bell's VoIP service or Rogers VoIP service. POTS is bound to be more reliable.

I like the idea of it much better than Skype. Skype uses a proprietary protocol so they "own" you in some sense. It has also meant that it has been difficult to get Skype phones that did not require a computer. But I imagine Skype is packaged well and has an easier learning curve (after all, most of the choices are made for you).

One of the great things about VoIP as voip.ms does it is that there is a whole competitive marketplace. Various vendors produce interchangeable parts and therefore compete on price and service. unlimitel.ca has been very well thought of as a VoIP provider. I currently use les.net in Canada and vitelity.com in the US. I suspect my ISP, telnetcommunications.com does this well too.

I learned about this stuff partly through the local Asterisk group http://taug.ca/. Asterisk is open sourcs PBX sofware which typically runs on Linux. One of the telnetcommunications people is quite active in TAUG. The guy running Unlimitel makes the trek to talk to TAUG once or twice a year.

Many of the TAUG work in the telephone business. Often replacing corporate phone systems for new ones with more features for a lot less money. With Asterisk it is easy to customize phone systems for the user. No longer are you tied to big vendors. All this is overkill for most RFD readers (and for me). But there is a real revolution rolling out in telephony.

Warnings:

the learning curve is steep. If you value your time and don't count this as fun, you might not want to become an expert. It may be the case that you can fairly quickly follow a recipe and get satisfactory results.
reliability of VoIP is lower that POTS at this time. Broadband connections are the main culprit, but you also depend on your ITSP (the technical term for wayt voip.ms is)
if you want your phone to work through a power failure, get POTS or Rogers Home Phone or a carefully-deployed UPS
typically SIP is not encrypted. If you value privacy there are complicated issues. End-to-end encryption (necessary but not sufficient for privacy) is quite awkward to arrange with any of the normal choices, including POTS.

[BargainBuff]

Can someone explain this CNAM-Query thing? From what I read via a google search, it has something to do with caller id. I'm glad to hear the price has dropped, but do I need this? I didn't check it as an option on my account settings because I really had no clue what it's for-----is it for ME to be able to see who's calling ME or is it for those that I call to be able to see its ME calling?

[ganiba]

Can someone explain this CNAM-Query thing? From what I read via a google search, it has something to do with caller id. I'm glad to hear the price has dropped, but do I need this? I didn't check it as an option on my account settings because I really had no clue what it's for-----is it for ME to be able to see who's calling ME or is it for those that I call to be able to see its ME calling?

"Be able to see who's calling ME". BUT, if you use the "phonebook" option, you can add some phones with their corresponding names, this way you won't pay for this service.

[BargainBuff]

"Be able to see who's calling ME". BUT, if you use the "phonebook" option, you can add some phones with their corresponding names, this way you won't pay for this service.

Thanks for clearing that up. I'll definitely be using the phone book option then. Thanks for the tip!

[jackie999]

My understanding of the CNAM is that it will associate a name with a number when you RECEIVE a call. If you don't pay for the service, I think all you see is a number.
If the persons number that is calling you is in your phone address book or voip.ms phonebook, then that is the name you'll see on your telephone display-regardless if you've paid for CNAM or not.
It's not foolproof though. I have enabled the feature but I still get 'unavailable' in my display...which means voip failed in it's lookup..for whatever reason...the name/number isn't in a database. I asked this just recently elsewhere and was given these reasons, amongst others, why you'd get 'unavailable':

These tend to fall into various groups:

a) Telemarketers and such.

b) Cellphone callers.

c) Callers who themselves are using a VoIP service.

d) Perfectly legitimate POTS landline callers, who don't even know how to hide a number, yet my telco (Verizon in this case) is screwing up.

Hope this clears it up a little....this voip is a learning curve for all of us.

[BargainBuff]

Am I restricted to only have 5 Voice mail boxes with Voip.ms? I was hoping to create a IVR where each 'person' would have their own voicemail, but 5 just won't cut it----especially if 5 is the maximum for ALL your DIDs. Is it really 5 voicemail boxes per account or per DID or is there a way to get more (without paying of course!)?

[ganiba]

Am I restricted to only have 5 Voice mail boxes with Voip.ms? I was hoping to create a IVR where each 'person' would have their own voicemail, but 5 just won't cut it----especially if 5 is the maximum for ALL your DIDs. Is it really 5 voicemail boxes per account or per DID or is there a way to get more (without paying of course!)?

I didn't try it but you can buy DID numbers on different voip.ms accounts, so you will have 5 per account if it is the limit of course.

[paulo500]

Dudes - why not stick with a CANADIAN company - Acanac- pay $9.95 a month for unlimited Canada/US use with all kinds of features?

I don't get this...IIRC Acanac won't port your existing number. This was a deal killer for me. (This was last year sometime...im not sure if things have changed.) Currently im on Primus TBB right now...Service has been good, but price has been about $30/mth. Too much)

Just sent in my ID to voip.ms for account validation. My cisco pap2t is ready to go :cheesygri

[Nraja]

Started off a few days ago with just a US Toll free number...now I have a new local Toronto number...and now looking into another Toronto number...this is getting addictive lol

My question though, is if I get a 647 number, can I have the calls forwarded to a UK number?

So basically the setup would be that in my cellphone phonebook I would assign my cousin 'Bob' who lives in the UK with a 647 Toronto number, would it allow me to fwd the calls internationally? Or would the fwd'ed numbers only be allowed to be within North America

[BargainBuff]

I didn't try it but you can buy DID numbers on different voip.ms accounts, so you will have 5 per account if it is the limit of course.

The downside with that option is that you have to prepay $25 for each account set up. I wonder if there is an easier way to get more than 5 voice mail boxes, since I already ordered 3 DIDs.

[Sneaks]

The downside with that option is that you have to prepay $25 for each account set up. I wonder if there is an easier way to get more than 5 voice mail boxes, since I already ordered 3 DIDs.

Why don't you try Posing your question either to their live support chat line, or file a ticket outlining exactly what you're looking to do. They may be able to suggest something. Their supports be great for the Q's I've asked, and is free.

[Sneaks]

for people who have successfully ported from Bell Landline.
What Service Provider Address did you specify in the request form?
The address that is on the payment stub of the bill (P.O. Box 9000 Stn. Don Mills) or some other one?

Thanks

Here's exactly what I used, which worked fine:
Service Provider Name: Bell Canada
Service Provider Address: P.O. Box 9000
Service provider address / Continued (Optional, e.g: suite 343): STN Don Mills
City: North York
State or Province: ON
Country: Canada
ZIP or Postal Code: M3C 2X7

[d4diesel]

For the $25 fee..I think it's money well spent to just port your existing number over. My 905 area code is becoming rare..so I'll hang on to it as long as I can.

initially, I couldnt find any 905 number in the Mississauga region and i didnt want a 289 number. I decided to wait a little while to read more reviews and port my current number if need to still be able to keep a 905 number. Yesterday I saw some 905 number posted on the site and I quickly snatched 2 DIDs. both DID differs by 1 as the last digit. I dont know where I will be using them but for 99 cents a month, i will keep them till i can decided what I want to do with them. so, if you dont like the 289 number, keep checking regular. they do post some 905 numbers every now and then but get taken up quickly.

[Lulz]

just got an email from them.

can anybody translate it to plain english please?


Introducing lower pricing and more accurate results for CNAM queries:

We have implemented a new CNAM Provider, in an effort to provide more accurate results to the customer base at a lower price

After intense testing of the new CNAM provider, we're glad to announce that we'll be implementing the new provider on May 1st,, and the new rate for CNAM Queries will go down from $0.0125 to $0.008, to be more precise, from 1.25 cent to 0.8 cent (eight tenth of a cent)

Please take a moment to read the following points:

- New Cnam-query price, with savings of 36% over the previous query price (CNAM Queries have been lowered from $0.0125 to $0.008)

- The new provider offers more accurate results.

- We've not been able to find a reliable provider that would allow us to cache CNAM Queries on a 24 hours period due to their contractual obligations. For this reason, we can no longer cache queries on a per 24 hours period.

- The 24 hours caching change will go live on May 1st, 2010.

- The pricing change is going live now.


If you have a certain list of contacts calling you a frequent basis, and wish to avoid repeated cnam-query charges on your account, we advise to add the caller's name and number to your VoIP.ms phonebook, there are no charges for queries made from your phonebook.

Thank you for your understanding.

Regards,

VoIP.ms Staff

Note: Do not reply to this email, you will not receive a response. You can contact us regarding this update by sending an email to support@voip.ms




EDIT: looks like somebody already raised the question on previous page...sorry but i couldn't read 18 pages to make sure it wasn't posted.
I still don't quite understand what it is for since the question wasn't answered...but for someone who uses DID only for forwarding purposes to real cell phone, do i need to care? (i don't use DID for outgoing)

[destinet]

Am I restricted to only have 5 Voice mail boxes with Voip.ms? I was hoping to create a IVR where each 'person' would have their own voicemail, but 5 just won't cut it----especially if 5 is the maximum for ALL your DIDs. Is it really 5 voicemail boxes per account or per DID or is there a way to get more (without paying of course!)?

I just try and i was able to create more then 5 voice mail...I try to create 6 and never receive an error

[coolspot]

EDIT: looks like somebody already raised the question on previous page...sorry but i couldn't read 18 pages to make sure it wasn't posted.
I still don't quite understand what it is for since the question wasn't answered...but for someone who uses DID only for forwarding purposes to real cell phone, do i need to care? (i don't use DID for outgoing)

voip.ms lowered their caller id rates.

If forwarding, yes it matters, voip.ms may do a caller id lookup prior to transferring the call.

You can disable caller id lookup in the control panel if you don't want to pay the 0.8c charge ;)

[jackie999]

just got an email from them.

can anybody translate it to plain english please?


It's a notification of a change in the CNAM rate they're charging, they're dropping the price :)
I tried to explain (in post #248) what the CNAM is ..what you're paying for..it is an optional fee.
HTH's

[BargainBuff]

I just try and i was able to create more then 5 voice mail...I try to create 6 and never receive an error

I wonder what's going on. When I go to Account Settings to create voicemail accounts it says you have to designate a mailbox number between 1 and 5 (max). I never tried entering more numbers but I will now.

[destinet]

I wonder what's going on. When I go to Account Settings to create voicemail accounts it says you have to designate a mailbox number between 1 and 5 (max). I never tried entering more numbers but I will now.

ahh thats the sub number you assign like [11111](customer number) + a number from 1 to 5 digit long
ex: 1111122222

edit:
but you can have has many you wants that mean you can have 5expo9 number of voicemail

[tenchi]

Thanks Sneaks.


Here's exactly what I used, which worked fine:
Service Provider Name: Bell Canada
Service Provider Address: P.O. Box 9000
Service provider address / Continued (Optional, e.g: suite 343): STN Don Mills
City: North York
State or Province: ON
Country: Canada
ZIP or Postal Code: M3C 2X7

[BargainBuff]

ahh thats the sub number you assign like [11111](customer number) + a number from 1 to 5 digit long
ex: 1111122222

edit:
but you can have has many you wants that mean you can have 5expo9 number of voicemail

Doh! It never dawned on me that THAT's what they meant! I thought I could only enter a number from 1 to 5!

Many thanks for clearing that up!

[jackie999]

Heh....I don't think you're the only one that thought that you could only have 1~5 boxes :)

[barrist]

So am i right in that you can't have your voicemail say your name ("you have reached the voicemail box of _____" when someone who calls gets it?

[jackie999]

So am i right in that you can't have your voicemail say your name ("you have reached the voicemail box of _____" when someone who calls gets it?

No, I called and asked them about that. I made a recording then didn't know *where* to put it. They told me I could email in the .wav file and they would set it up for me, but suggested it would be easier if I just used my phone to make/save the recording. FYI, I also turned off the answer machine built into my phone so that the voip.ms will take the voicemail messages for me (in the event of a power outtage this seems best) ..the system emails me if I have a voice message - you set that up in your voip a/c.

My setup is very basic...if you have multiple voicemail boxes..I suggest you ask them in livechat, then post back here how you do it :)

[wally_walrus]

Did anyone here succesfully used a standalone IP phone with voip.ms? I am looking specifically into the Aastra 480i. What would be the advantage of using an ATA device vs a dedicated IP phone (aside from price)? If you need a full-featured good quality phone (ie Nortel, Panasonic, etc) they are quite expensive, so why not buy an IP phone instead? Am I missing anything here?

Thanks,

[apn64]

Did anyone here succesfully used a standalone IP phone with voip.ms? I am looking specifically into the Aastra 480i. What would be the advantage of using an ATA device vs a dedicated IP phone (aside from price)? If you need a full-featured good quality phone (ie Nortel, Panasonic, etc) they are quite expensive, so why not buy an IP phone instead? Am I missing anything here?

Thanks,

IMHO, the primary advantage of using an ATA is that you can continue to use standard phones (mine are all Nortel and Panasonic), plus you maintain centralized control and the ATA only takes a single router port. The ATA may also have a few features not found in all IP phones.

IP phones (BTW, Aastra purchased Nortel's phone business) give a LOT more flexibility and provide individual control, plus you're no longer limited by the physical L1/L2 phone lines in the house; you want an extension, just run a CAT5e to the location and you're set. However, the disadvantage is that each phone needs a router/switch port.

Even though they're pricey, if I was starting from scratch, I'd think about IP phones. However, the cost to replace three high-end Nortel and two Panasonic cordless phones meant the $50 ATA was a way more cost effective route for me. YMMV.

[sternem]

I have a linksys ATA (3102). I tried to setup an outgoing dial plan retaining my existing setup, so I could give their service a try and see how the quality was.

I tried something like <7,:>x.<:@gw1>, where I defined gw1 using the proxy, userid and password from voip.ms. This should've allowed me to dial '7' and then whatever number I wanted to connect to. This didn't work for me (works with another VOIP provider I have).

I could only get dialout working if I changed my primary provider settings on the ATA. Do you guys know if voip.ms only supports registered dialout? Or is there something else I missed?

[Shaf]

I have a linksys ATA (3102). I tried to setup an outgoing dial plan retaining my existing setup, so I could give their service a try and see how the quality was.

I tried something like <7,:>x.<:@gw1>, where I defined gw1 using the proxy, userid and password from voip.ms. This should've allowed me to dial '7' and then whatever number I wanted to connect to. This didn't work for me (works with another VOIP provider I have).

I could only get dialout working if I changed my primary provider settings on the ATA. Do you guys know if voip.ms only supports registered dialout? Or is there something else I missed?

I use FPL as my primary provider on teh SPA3102 and use Voip.ms only for outgoing (set up as gateway 1).

This is my dial plan:

(911S0|[2-9]xxxxxxxxxS0|1[2-9]xx[2-9]xxxxxxS0|011[2-9]xx.<:@gw1>|<#1,:>x.<:@gw1>|<:18004664>411S0|<:416> 310xxxxS0||1900!|*xx)

I don't need to dial #1 for 011 calls only for LD that is not free with FPL.

[wally_walrus]

Did anyone here succesfully used a standalone IP phone with voip.ms? I am looking specifically into the Aastra 480i. What would be the advantage of using an ATA device vs a dedicated IP phone (aside from price)? If you need a full-featured good quality phone (ie Nortel, Panasonic, etc) they are quite expensive, so why not buy an IP phone instead? Am I missing anything here?

Thanks,


IMHO, the primary advantage of using an ATA is that you can continue to use standard phones (mine are all Nortel and Panasonic), plus you maintain centralized control and the ATA only takes a single router port. The ATA may also have a few features not found in all IP phones.

IP phones (BTW, Aastra purchased Nortel's phone business) give a LOT more flexibility and provide individual control, plus you're no longer limited by the physical L1/L2 phone lines in the house; you want an extension, just run a CAT5e to the location and you're set. However, the disadvantage is that each phone needs a router/switch port.

Even though they're pricey, if I was starting from scratch, I'd think about IP phones. However, the cost to replace three high-end Nortel and two Panasonic cordless phones meant the $50 ATA was a way more cost effective route for me. YMMV.

Thanks! My setup won't be too complicated, all I want is one extra phone in the house to use for long distance and / or as a secondary line when the main POTS line is busy. But I want a good quality phone with features, which are quite pricey and comparable to an IP phone (think Nortel / Aastra). Plus with an ATA there would be one more box with a power brick and cables to worry about.

So in my case it likely doesn't make sense to use an ATA, unless I want to fall back on POTS when VoIP fails. I'll think about that...

Thanks again for your thoughts

[apn64]

Thanks! My setup won't be too complicated, all I want is one extra phone in the house to use for long distance and / or as a secondary line when the main POTS line is busy. But I want a good quality phone with features, which are quite pricey and comparable to an IP phone (think Nortel / Aastra). Plus with an ATA there would be one more box with a power brick and cables to worry about.

So in my case it likely doesn't make sense to use an ATA, unless I want to fall back on POTS when VoIP fails. I'll think about that...

Thanks again for your thoughts

voip.ms is a carrier-grade VoIP service, which most people use as a replacement for POTS service (it provides 911 and many, many more features than POTS/Cable-VoIP services). However, if your intention is to only use VoIP for cheap LD and maintain a POTS line, then there might be better/cheaper alternatives.

I believe that Skype offers a $2.95/month unlimited North America plan. IIRC, D-Link and others make adaptors that accept both POTS and IP connections, allowing you to use a standard phone. Taking it a step further, Belkin and Netgear make a Skype phones, so you're not stuck with using a PC to make phone calls.

Although I'm not a big fan of it, Freephoneline.ca might be another option, but IMHO, neither of these services come close to the quality/service/flexibility of voip.ms, but again what's best for you depends on your requirements.

[sternem]

I use FPL as my primary provider on teh SPA3102 and use Voip.ms only for outgoing (set up as gateway 1).

This is my dial plan:

(911S0|[2-9]xxxxxxxxxS0|1[2-9]xx[2-9]xxxxxxS0|011[2-9]xx.<:@gw1>|<#1,:>x.<:@gw1>|<:18004664>411S0|<:416> 310xxxxS0||1900!|*xx)

I don't need to dial #1 for 011 calls only for LD that is not free with FPL.

Thanks Shaf, confirms that it's not voip.ms but my configuration issue. Turns out I had to have 111XXX@sip.ca2.voip.ms for my gateway2 setting. I only had the actual server at first. All working now.

For my other provider on gateway1, just the server name was enough:confused:.

[Shaf]

Thanks Shaf, confirms that it's not voip.ms but my configuration issue. Turns out I had to have 111XXX@sip.ca2.voip.ms for my gateway2 setting. I only had the actual server at first. All working now.

For my other provider on gateway1, just the server name was enough:confused:.

Perfect!

[airstream]

My question though, is if I get a 647 number, can I have the calls forwarded to a UK number?

So basically the setup would be that in my cellphone phonebook I would assign my cousin 'Bob' who lives in the UK with a 647 Toronto number, would it allow me to fwd the calls internationally? Or would the fwd'ed numbers only be allowed to be within North America

Yes, you can have calls forwarded overseas. When I travel, I forward my cell to my voip.ms number as I board the plane. When I land and pick-up a local prepaid SIM, i just login and forward my voip number to the local assigned SIM. In most of Europe and Asia, the local carriers have free incoming calls. So when friends and family from home call my Toronto mobile number, the get forwarded to my overseas number, and the cost to me is very little in voip LD rates.

[Nraja]

Yes, you can have calls forwarded overseas. When I travel, I forward my cell to my voip.ms number as I board the plane. When I land and pick-up a local prepaid SIM, i just login and forward my voip number to the local assigned SIM. In most of Europe and Asia, the local carriers have free incoming calls. So when friends and family from home call my Toronto mobile number, the get forwarded to my overseas number, and the cost to me is very little in voip LD rates.

God damn ingenious! bad thing is now that you told me that I need another line!

Now thats 3 DID's plus a Toll Free US DID all in 1 week

I need to stop reading this thread..there's too many ways to use these numbers!

[Nraja]

Yes, you can have calls forwarded overseas. When I travel, I forward my cell to my voip.ms number as I board the plane. When I land and pick-up a local prepaid SIM, i just login and forward my voip number to the local assigned SIM. In most of Europe and Asia, the local carriers have free incoming calls. So when friends and family from home call my Toronto mobile number, the get forwarded to my overseas number, and the cost to me is very little in voip LD rates.

Wait...doesn't Rogers generally charge to fwd your cell to another number?

[websavvy]

Wait...doesn't Rogers generally charge to fwd your cell to another number?

I'm pretty sure that Rogers isn't involved in this scenario. Once you buy a SIM card from a local carrier, you're dealing with their network (and pricing) and your phone.

[plg_cp]

Wait...doesn't Rogers generally charge to fwd your cell to another number?

I believe call forwarding is a "feature" for Rogers Wireless, so if your plan includes it, then you could use it at will.

[digdeep]

I believe call forwarding is a "feature" for Rogers Wireless, so if your plan includes it, then you could use it at will.

I guess he meant the forwarded calls are charged against air time of your plan. ;)

[airstream]

I guess he meant the forwarded calls are charged against air time of your plan. ;)

Call fowarding is included in my plan (up to 3500mins), so it wasn't a problem.

For an example, I spent the month of December in Thailand. I paid a total of $7CAD in prepaid SIM/top-up minutes on Happy (one of their carriers) and a total of $2.26 on my voip.ms account (for a total of 2:08:15 calling time). I also have a SIP client on my iPhone, so anytime I had wi-fi there, I could call home for next to nothing!

[wally_walrus]

Thanks! My setup won't be too complicated, all I want is one extra phone in the house to use for long distance and / or as a secondary line when the main POTS line is busy. But I want a good quality phone with features, which are quite pricey and comparable to an IP phone (think Nortel / Aastra). Plus with an ATA there would be one more box with a power brick and cables to worry about.

So in my case it likely doesn't make sense to use an ATA, unless I want to fall back on POTS when VoIP fails. I'll think about that...

Thanks again for your thoughts


voip.ms is a carrier-grade VoIP service, which most people use as a replacement for POTS service (it provides 911 and many, many more features than POTS/Cable-VoIP services). However, if your intention is to only use VoIP for cheap LD and maintain a POTS line, then there might be better/cheaper alternatives.

I believe that Skype offers a $2.95/month unlimited North America plan. IIRC, D-Link and others make adaptors that accept both POTS and IP connections, allowing you to use a standard phone. Taking it a step further, Belkin and Netgear make a Skype phones, so you're not stuck with using a PC to make phone calls.

Although I'm not a big fan of it, Freephoneline.ca might be another option, but IMHO, neither of these services come close to the quality/service/flexibility of voip.ms, but again what's best for you depends on your requirements.

These might be cheaper but not necessarily better or simpler. I'd like to keep everything as close to the appearance of a POTS line as possible. I don't mind paying a little extra for a rock-solid VoIP line, and like you mentioned the flexibility is incredible

Thanks again

[Dave98]

I'm looking into signing up to get rid of my RHP service but my VOIP knowledge is fairly limited.

Have any of you ported your number from Rogers Home Phone?

I see some of you mentioning you use both freephoneline.ca and voip.ms. How come?

As of right now with RHP, I don't even have call waiting so the thought of cutting my phone costs so much with VOIP and yet still have more features than RHP is tempting. I just have to get all this VOIP terminology and everything down.

Most calls placed in my household would be domestic, to the USA, and China. I'll be looking into getting the PAP2T.

[MXMike]

I just received my Linksys PAP2T today from NCIX. Set up was very easy using the config example on voip.ms and I confirmed it works for outgoing calls.

So now that I know my ATA works properly I started the number porting procedure to port my 2 Canadian numbers from Comwave to voip.ms. voip.ms charges $25 per number which if fine, but I had no idea Comwave also charges $50 per number I want to port out to another carrier (reverse LNP)! Is there normally a fee like this to port a number out? (I didn't think there was). It's going to cost me $150 total just to transfer my 2 number to voip.ms.

[signup]

Does anyone with voip.ms configure their Linksys ATA using SIP Proxy Redundancy?

[Badder]

... but I had no idea Comwave also charges $50 per number I want to port out to another carrier (reverse LNP)! Is there normally a fee like this to port a number out? (I didn't think there was).

I also ported my number to voip.ms (from Bell) a few weeks ago. I did it by the book and paid my account in full before initiating the port. Today I just received a bill from Bell with $25.56+ for "line cancellation fees". :mad:

Did anyone get the same thing? I don't recall having to pay any such fees when I ported my wireless number from Rogers a few years ago.

[WildChild]

I also ported my number to voip.ms (from Bell) a few weeks ago. I did it by the book and paid my account in full before initiating the port. Today I just received a bill from Bell with $25.56+ for "line cancellation fees". :mad:

Did anyone get the same thing? I don't recall having to pay any such fees when I ported my wireless number from Rogers a few years ago.

Don't know about Bell but I didn't have any fees when I ported my number from Vidéotron.

[RMSimard]

I also ported my number to voip.ms (from Bell) a few weeks ago. I did it by the book and paid my account in full before initiating the port. Today I just received a bill from Bell with $25.56+ for "line cancellation fees". :mad:

Did anyone get the same thing? I don't recall having to pay any such fees when I ported my wireless number from Rogers a few years ago.

No Bell fees for me either when I ported out a number last year. Maybe they figure that since you're leaving them anyway there's no harm in taking money from you one last time for frivolous reasons.

Regarding such porting fees in general, my understanding is that most carriers charge a fee to leave them for a competitor. That competitor usually takes the financial hit, hoping you'll stay with them long enough for them to recoup the porting loss they incurred. I thought that the $25 fee from voip.ms was to cover such charges since VoIP carriers' profit margins are much thinner than POTS. Sucks tough if Bell takes said money ...then turns around and asks you for more. :evil:

[jackie999]

Little info I dug up (in the small print)

Comwave small print,


3.08 Telephone numbers
If you were assigned a telephone number from Comwave, this number may at Comwave sole and absolute discretion and subject to applicable law be assigned to another Subscriber. Transferring phone numbers to Comwave (LNP) is possible only in Comwave serving territory. If you decide to LNP your existing phone number to Comwave from another telephone provider; it is possible that your service will be interrupted in certain circumstances while the transfer is in process. Specific LNP instructions are listed in the LNP web page under "My Account". To transfer any number to another company (Reverse LNP) will incur an administration fee of $50 per number.

Voip small print,

15. At VoIP.ms sole discretion, you may incur a port away fee for any DID number(s) leaving our network as this is a pass-through charge from VoIP.ms carrier(s).

I paid a $20 charge for Teksavvy to change my DSL to dry loop service. There was no 'port away' charge.

I'm like most people thou, I never read the small print ...

[death_hawk]

(Note, this is not directed at anyone in general, but rather an observation.)

I find it hilarious that those people quoting "reliability" on POTS vs VoIP most times have cordless phones that depend on power to survive and don't have any "hardwired" phones for failover.
Sure you can make calls in an emergency while the power is out, but not with a cordless phone.

A properly set up VoIP system CAN make phones during a power outage.

[ganiba]

(Note, this is not directed at anyone in general, but rather an observation.)

I find it hilarious that those people quoting "reliability" on POTS vs VoIP most times have cordless phones that depend on power to survive and don't have any "hardwired" phones for failover.
Sure you can make calls in an emergency while the power is out, but not with a cordless phone.

A properly set up VoIP system CAN make phones during a power outage.

How? where is the internet connection without a powered modem during a power outage? Is there another way I'm missing here?

Thanks for clarifying your point.

[death_hawk]

That's why I used the qualifier "properly setup"

Basically you need your entire network infrastructure to be UPSed as well as your phone network (assuming it's different, IE PoE etc)

In my particular setup, my main networking infrastructure has a UPS dedicated to it which allows me to use the internet (as well as VoIP) for about 12 hours after a power outage.

[ganiba]

That's why I used the qualifier "properly setup"

Basically you need your entire network infrastructure to be UPSed as well as your phone network (assuming it's different, IE PoE etc)

In my particular setup, my main networking infrastructure has a UPS dedicated to it which allows me to use the internet (as well as VoIP) for about 12 hours after a power outage.

Dear Sir,

So, why not connect your cordless phone to your UPS and it should stay working for days, no? It is not fair to compare cordless phone without UPS with a VOIP service with UPS.

The only reliable way is a hardwired phone, as you said before.

[d4diesel]

Dear Sir,

So, why not connect your cordless phone to your UPS and it should stay working for days, no? It is not fair to compare cordless phone without UPS with a VOIP service with UPS.

The only reliable way is a hardwired phone, as you said before.

The point originally being made was that people have a false sense of security thinking that that can make calls in an outage on their POTS line without realizing that they will need a UPS to make their cordless phone work.

The summary in the end should be that if you want to make calls in a power outage, get a UPS and hook up your phone (and ATA+router+modem in case of voip) to the UPS.

I, personally, will stick to making calls using my cell phone for now in a power outage. The last time it happen in my area, most of the eastern side of continent was out so even if i wanted to make a call, they was hardly anyone else with power to receive it that time.

[death_hawk]

Dear Sir,

So, why not connect your cordless phone to your UPS and it should stay working for days, no? It is not fair to compare cordless phone without UPS with a VOIP service with UPS.

The only reliable way is a hardwired phone, as you said before.

d4diesel pretty much nailed it.

My entire point was that most people with cordless phones will be without phone service because they don't have a failover phone. If they had a UPS it wouldn't be an issue, but 99.99% of users don't have a UPS, or don't think to hook their cordless phones to it. Hell, I bet 99.9% of users think a UPS isn't necessary or in the realm of high tech corporations etc.

Hell, d4diesel has another point: Just use your cell phone. Not that it matters since anyone you call that's not on a cell can't pick up anyway...

[Elusive]

If a tower went down, say your neighbourhood power goes down, even with your UPS, wouldn't you be out of connection anyway since the neighbourhood is down?


That's why I used the qualifier "properly setup"

Basically you need your entire network infrastructure to be UPSed as well as your phone network (assuming it's different, IE PoE etc)

In my particular setup, my main networking infrastructure has a UPS dedicated to it which allows me to use the internet (as well as VoIP) for about 12 hours after a power outage.

[djino]

If a tower went down, say your neighbourhood power goes down, even with your UPS, wouldn't you be out of connection anyway since the neighbourhood is down?

Assuming the Internet (Cable or DSL) was still active, then no, your connection would be alive as long as your Battery on the UPS is running.

djino

[Dave98]

I can see the termination rates on the official page and I'm interested in the rates for calling China. They seem to range from 0.0101 to 0.0124.

Anyone know if these rates are better than those offered on those $5 calling cards? I ask because my parents make calls to China every so often using phone cards.... and you never know with those things whether you actually get what you pay for.

Regardless, it looks like I'll be making my first deposit to voip.ms. I'll hopefully make as smooth a transition to voip as the rest of you seem to have.

[Lulz]

I have a quick question about forwarding while roaming.

Am I understanding how it works:

I have a DID number thru VOIP.ms

I forward my Toronto cell phone (which has 2500CF minutes) to VOIP.ms number.

Then, i forward VOIP.ms number to my Russian SIM card which has free incoming.

So when somebody calls my Toronto number, they will be forwarded to DID, which will get forwarded to prepaid SIM card.

How much will I pay for each minute? I know it varies by area code to which Russian area code number I am forwarding, but in general, how much will I be paying? Area code is 7926 and when I go to VOIP.ms, the rates for 7926 is $0.05480000

So let's say I talk for a minute exactly...will I be paying half a cent for each minute? (i understand it's 6 second increment...but for sake of simplicity, lets just say 1 minute)
Half a cent a minute sounds just wayyyy too good to be true, is it really like that?

[djino]

I have a quick question about forwarding while roaming.

Am I understanding how it works:

I have a DID number thru VOIP.ms

I forward my Toronto cell phone (which has 2500CF minutes) to VOIP.ms number.

Then, i forward VOIP.ms number to my Russian SIM card which has free incoming.

So when somebody calls my Toronto number, they will be forwarded to DID, which will get forwarded to prepaid SIM card.

How much will I pay for each minute? I know it varies by area code to which Russian area code number I am forwarding, but in general, how much will I be paying? Area code is 7926 and when I go to VOIP.ms, the rates for 7926 is $0.05480000

So let's say I talk for a minute exactly...will I be paying half a cent for each minute? (i understand it's 6 second increment...but for sake of simplicity, lets just say 1 minute)
Half a cent a minute sounds just wayyyy too good to be true, is it really like that?

You will be paying I believe 1 cent/minute on the incoming to the Toronto DID as well as 5.48 cents/minute on the forwarding to your Russian termination.

So on a 60 second call, your cost is $0.0648

djino

[Sneaks]

Area code is 7926 and when I go to VOIP.ms, the rates for 7926 is $0.05480000

So let's say I talk for a minute exactly...will I be paying half a cent for each minute? (i understand it's 6 second increment...but for sake of simplicity, lets just say 1 minute)
Half a cent a minute sounds just wayyyy too good to be true, is it really like that?

Actually you'd be 6 c/min (based off the rate looked up for that area code) for the voip.ms part, as you'd have the in essence: incoming call to your DID from your forwarded Cell (1c/min), and then the outgoing call to Russian prepaid SIM (5c/min.)

total 1+5=6c/min

[djino]

Actually you'd be 6 c/min (based off the rate looked up for that area code) for the voip.ms part, as you'd have the in essence: incoming call to your DID from your forwarded Cell (1c/min), and then the outgoing call to Russian prepaid SIM (5c/min.)

total 1+5=6c/min

As I posted above :)

djino

[Lulz]

just occurred to me that $0.05480000 is 5 cents, not half a cent.
lol...figures i hated math and failed calculus and had to take summer school.
thanks djino and sneaks.

so was I correct in how setup works (whats forwarded to what)...meaning is this what people always referred to when they said: "forward to voip and have cheap roaming", or is there another secret way to talk out of country, without having your friends dial foreign number?

[jackie999]

Since I don't call overseas I took a quick look at bells rate..am I right, is a call to Russia normally over $3? And we're tossing around whether voip is 1/2 a cent or 1 cent + 1/2 a cent?

[djino]

Since I don't call overseas I took a quick look at bells rate..am I right, is a call to Russia normally over $3? And we're tossing around whether voip is 1/2 a cent or 1 cent + 1/2 a cent?

technically, he/we were discussing whether the cost is 1/2 a cent or 5 cents on the minute.

djino

[jackie999]

I'm just amazed at the difference in rates .... if a call to Russia is over $3 via bell..then after making, say, a 1/2 hour call to a family member in Russia using voip, you've recouped the cost of a $50 ATA and have some spare change left to cover your Canadian calling for the year...maybe my decimals are mixed up too...

[Ginzu]

So I admit, I jumped on board with this thinking it was a good way to use them as local forwarding numbers for folks outside of your local calling area.

Example: my in-laws live in Leamington - I purchased a Windsor DID (a local call for them) and setup call forwarding to my 905 TN here in Markham. For them, it's a local call and I pay a much lower rate for the LD rates than they would have through Bell/Cogeco.

I also setup a Toronto DID with DISA so I can use that as my own local calling card - dial the Toronto DID, get the prompt to enter a new number, then dial from there.

Are there other use cases (other than porting my home TN - not ready for that - yet...) that people use the DID numbers for?

[airstream]

just occurred to me that $0.05480000 is 5 cents, not half a cent.
lol...figures i hated math and failed calculus and had to take summer school.
thanks djino and sneaks.

so was I correct in how setup works (whats forwarded to what)...meaning is this what people always referred to when they said: "forward to voip and have cheap roaming", or is there another secret way to talk out of country, without having your friends dial foreign number?

Yes, your forwarding sequence is correct.

If you have a SIP client on your cell phone, and are in a WI-FI friendly area (and your phone has WI-FI), you can then use it to call home as well for local rates. I configured my VOIP.ms caller ID to display my cell number. So when I called my folks, they would not ignore a call from a unrecognized number.

[Mykolas]

This is an amazing thread. Just read all 21 pages and will be going with voip.ms instead of acanac as I had planned.

Two questions though:
1) DISA: a couple of you have set this up so that a call from your cell phone presents you with a dial tone for onward calling. Does this mean that if actually want to talk to someone at your house,using the cell phone, you need to enter your home phone number a second time at the dial tone?

2) QoS: I'd be interested in hearing anyone's experience with setting up QoS on their router. I have DD-WRT running on my Linksys, and have an SPA2102 in the mail. I use my internet connection quite heavily, and should really set up QoS to make this work I think. From general reading, it seems most people somehow set up 'services' on their routers and then perhaps link those to specific port numbers; I'm not clear on how this is done. Last step is to prioritize the services. Anyway, any info in this area would be appreciated.

[luckydog2006]

This is an amazing thread. Just read all 21 pages and will be going with voip.ms instead of acanac as I had planned.

Two questions though:
1) DISA: a couple of you have set this up so that a call from your cell phone presents you with a dial tone for onward calling. Does this mean that if actually want to talk to someone at your house,using the cell phone, you need to enter your home phone number a second time at the dial tone?

2) QoS: I'd be interested in hearing anyone's experience with setting up QoS on their router. I have DD-WRT running on my Linksys, and have an SPA2102 in the mail. I use my internet connection quite heavily, and should really set up QoS to make this work I think. From general reading, it seems most people somehow set up 'services' on their routers and then perhaps link those to specific port numbers; I'm not clear on how this is done. Last step is to prioritize the services. Anyway, any info in this area would be appreciated.


ANS#1 - You can setup an IVR and use that virtual receptionist to route your call to DISA, fwd to your home phone, or fwd to your cellphone etc. So, you don't need to enter your phone numbers at all other than when choosing the DISA option.

ANS#2 - Click on LINK (http://www.linksysinfo.org/forums/showthread.php?t=60304&highlight=QoS)

[RMSimard]

1) DISA: a couple of you have set this up so that a call from your cell phone presents you with a dial tone for onward calling. Does this mean that if actually want to talk to someone at your house,using the cell phone, you need to enter your home phone number a second time at the dial tone?

2) QoS: I'd be interested in hearing anyone's experience with setting up QoS on their router. I have DD-WRT running on my Linksys, and have an SPA2102 in the mail. I use my internet connection quite heavily, and should really set up QoS to make this work I think. From general reading, it seems most people somehow set up 'services' on their routers and then perhaps link those to specific port numbers; I'm not clear on how this is done. Last step is to prioritize the services. Anyway, any info in this area would be appreciated.

DISA: You can set up a Digital Receptionist (IVR) that filters and answers only to your Caller ID, then have it offer you option #1 for an outside line, option #2 to ring a phone. Best of both worlds! :) (damn, luckydog2006 beat me to it!)

QoS: Can't help with DD-WRT specifically, but I'm using a D-Link DIR-825 and gave low-priority data transfer ports and protocols a QoS setting of 255 (lowest), and set priority 1 to VoIP UDP ports. Never had any problem with choppy VoIP since, no matter how large the bandwidth usage.

[jackie999]

T

2) QoS: I'd be interested in hearing anyone's experience with setting up QoS on their router. I have DD-WRT running on my Linksys, and have an SPA2102 in the mail. I use my internet connection quite heavily, and should really set up QoS to make this work I think. From general reading, it seems most people somehow set up 'services' on their routers and then perhaps link those to specific port numbers; I'm not clear on how this is done. Last step is to prioritize the services. Anyway, any info in this area would be appreciated.

I've been looking for a way to set up the QoS on my router. The link luckydog2006 gave you is for tomato (I think the DD-WRT is different) but that forum is probably the best place to research.
I have yet to try toastmans tomato QoS tutorial, most seem to deal with torrenting, which doesn't affect me..so I'm still reading..maybe I should switch my tomato to DD-WRT if TCP Vegas works..let me know :)

[destinet]

Just finishing my porting from vonage to voip.ms. It toke 5 days to port my number, so fare everyting works fine and the quality is really good.

Now i need to fight with vonage because they want to charge me 275$ in fees for cancelling event if i was a member since 4years...another battle.

[mondriano]

Having come from Primus' voip, which was as easy as plugging in the ATA, and getting a dial tone, I am finding setting up the voip.ms incredibly frustrating. Once connected through the router, cannot get a dial tone, and screws up the internet connection for the laptop. When running the modem directly into the ATA (no router, no computer...only modem, ATA, phone), still cannot get a dial tone, despite following voip.ms configuration samples. I had been assured by customer support that this was going to be a painless plug-and-play installation; it has been anything but so far...now hearing that configuring the router is going to be necessary...that is far from plug-and-play from my perspective.

On the plus side, their customer service is still amazing :D

[jackie999]

Are you sure your ATA is working properly? Maybe you got a bad one? My experience was, within 15 minutes of plugging it in I was making calls. I use a router, but there were no changes needed in my WRT54GL running tomato..what does their support suggest is the problem?


now hearing that configuring the router is going to be necessary

You don't have to configure the router..that's only if you want to "push the envelope" :)

[Sneaks]

Having come from Primus' voip, which was as easy as plugging in the ATA, and getting a dial tone, I am finding setting up the voip.ms incredibly frustrating. Once connected through the router, cannot get a dial tone, and screws up the internet connection for the laptop. When running the modem directly into the ATA (no router, no computer...only modem, ATA, phone), still cannot get a dial tone, despite following voip.ms configuration samples. I had been assured by customer support that this was going to be a painless plug-and-play installation; it has been anything but so far...now hearing that configuring the router is going to be necessary...that is far from plug-and-play from my perspective.

On the plus side, their customer service is still amazing :D

Silly q, I'm assuming you've ruled out, but will confrim anyway: your ATA wasn't "locked" to primus or anything was it? Or did you purchase it yourself, and is open.

[Badder]

QoS: Can't help with DD-WRT specifically, but I'm using a D-Link DIR-825 and gave low-priority data transfer ports and protocols a QoS setting of 255 (lowest), and set priority 1 to VoIP UDP ports. Never had any problem with choppy VoIP since, no matter how large the bandwidth usage.

I don't even bother setting up QoS on my WRT54GL router because my ATA is behind the router, in which case the QoS should be relevant only for upstream data (it's too late for downstream). In order to have true QoS, you're supposed to put the routing-capable ATA (e.g. SPA2102) before your usual router and setup QoS on the ATA. I haven't tried it though.

[kenj]

I have a quick question about forwarding while roaming.

Am I understanding how it works:

I have a DID number thru VOIP.ms

I forward my Toronto cell phone (which has 2500CF minutes) to VOIP.ms number.

Then, i forward VOIP.ms number to my Russian SIM card which has free incoming.

So when somebody calls my Toronto number, they will be forwarded to DID, which will get forwarded to prepaid SIM card.

How much will I pay for each minute? I know it varies by area code to which Russian area code number I am forwarding, but in general, how much will I be paying? Area code is 7926 and when I go to VOIP.ms, the rates for 7926 is $0.05480000

So let's say I talk for a minute exactly...will I be paying half a cent for each minute? (i understand it's 6 second increment...but for sake of simplicity, lets just say 1 minute)
Half a cent a minute sounds just wayyyy too good to be true, is it really like that?

I'm doing this exact thing right now. I'm in Ukraine and have it forwarded to my prepaid cellphone (Beeline, of course, free incoming!) As well, I purchased another european card which I forward to when I am in Europe. Works great. There's been a few times where I've had stuttering, but more often than not it works great.

[jackie999]

I don't even bother setting up QoS on my WRT54GL router because my ATA is behind the router, in which case the QoS should be relevant only for upstream data (it's too late for downstream). In order to have true QoS, you're supposed to put the routing-capable ATA (e.g. SPA2102) before your usual router and setup QoS on the ATA. I haven't tried it though.

I'm sorry if this is a painfully basic question...but everyone here is so helpful, I thought I'd ask.
Are you saying, since my set up is:
internet/modem ---> router---> to ATA and PC
My ATA is therefore considered to be 'behind' the router?
VoIP information is up and downstream data? So setting up a QoS would help a bit (for the up) during heavy downloading..no?

[mondriano]

Silly q, I'm assuming you've ruled out, but will confrim anyway: your ATA wasn't "locked" to primus or anything was it? Or did you purchase it yourself, and is open.

No, I am trying to connect through a new, unlocked PAP2T adapter purchased from Canada Computers. I was assured at the time that it was not locked to any provider. What I find even more frustrating is that even when the router is taken out of the equation (so I am going directly from modem to ATA), I am getting zilch (no dial tone, no incoming calls).

[jackie999]

I find, it can take about a minute for the PAP2T to 'wake up' after starting router etc. Do you see 3 green lights on the front?
Also, if you are bypassing your router, did you put the modem into routing mode, or is it in bridge mode?
Could be you got a defective one :(

Also, what is the make/model of the router...perhaps that will help troubleshoot...

[mondriano]

I find, it can take about a minute for the PAP2T to 'wake up' after starting router etc. Do you see 3 green lights on the front?
Also, if you are bypassing your router, did you put the modem into routing mode, or is it in bridge mode?
Could be you got a defective one :(


I have yet to see three green lights. Only "Power" and "Internet". Never "Line 1". As far as "routing" vs. "bridge" mode, I'm sorry, but I don't know. I removed the router just to get the most direct route, and to see if that worked. Since it didn't, I don't know what the next step should be.

[mondriano]

Also, what is the make/model of the router...perhaps that will help troubleshoot...

Router is Linksys WRT54G2.

As far as the ATA being defective, is this likely? I hear a fair bit about this, but are new-in-box ATAs that likely to be shipped defective?

Thanks for all the suggestions :D

[jackie999]

Heh..this is like the blind leading the blind...hopefully somebody else will step up and help.
When you are running a modem and a router, one of them must handshake with your internet provider with username/password. Normally that PPPoE handshake is done with the router (that's with a DSL connection - cable would be different). Then the modem is set in a more simple 'bridge' mode - it's just a pathway. If you've removed the router, then the modem must be set up with the connection information - it's easy to go back and forth, once you have the steps printed out.
Does this help at all?

[paulo500]

I've got the same scenario as mondriano.
I just got a message from Voip.ms saying my number has been successful ported from Primus. I called home to tell my wife this but the phone is dead (busy signal)..Guess it worked..so far so good... Ohh the wife is going to kill me. >:(

Anyway, when i get home tonight i need to setup my Cisco PAP2T-NA (from Canada computers) right away. I'll be using the voip.ms configuration sample for this ATA. Crossing my figures.

[jackie999]

Yep, once the number is ported the land line won't work. I had my ATA all set up ahead of time so it just started taking incoming calls on it's own.
I spent time reading over on dslreports.com in the Voip Tech Chat forums - they are very helpful if you have any problems...
Good luck!

[death_hawk]

Assuming the Internet (Cable or DSL) was still active, then no, your connection would be alive as long as your Battery on the UPS is running.

djino

Plus I have redundant internet connections so even if one or the other went down, I'd still be live.

Oh and I have a backup generator. I'm not going down without a fight.

[luckydog2006]

I have yet to see three green lights. Only "Power" and "Internet". Never "Line 1". As far as "routing" vs. "bridge" mode, I'm sorry, but I don't know. I removed the router just to get the most direct route, and to see if that worked. Since it didn't, I don't know what the next step should be.

It's rare that a brand new ATA is defective. You should get a dial tone if all your settings are fine.

Make sure your ATA settings and dial string is correct. Example;

Dial string: (*xx|<411:18004664411>S0|[2-9]xx[2-9]xxxxxxS0|1[2-9]xx[2-9]xxxxxxS0|011[2-9]x.|1900xxxxxxx!)

Check for other dial string examples, but usually the default dial string you have should work fine, and you can customize it to fit your taste. Would not recommend to make it long-winded or complex. Keep it simple and short.

Hard to trouble shoot without detailed info. I would recommend you check out the voip websites for help. This should be a walk in the park to setup. All you need is an account setup, login and password.

[paulo500]

I've got the same scenario as mondriano.
I just got a message from Voip.ms saying my number has been successful ported from Primus. I called home to tell my wife this but the phone is dead (busy signal)..Guess it worked..so far so good... Ohh the wife is going to kill me. >:(

Anyway, when i get home tonight i need to setup my Cisco PAP2T-NA (from Canada computers) right away. I'll be using the voip.ms configuration sample for this ATA. Crossing my figures.
OK all systems go!
Disconnected the Primus ATA box and replaced it with the new Cisco one. Powered it up and checked all my phones in the house. Dial tone was good! Dialed into the ata box using cisco's keypad codes and retrieved the IP it's using. Plugged the ip and other info (from the voip.ms config page) into the cisco ata via web interface. Everything registered as expected on the voip.ms user portal too.

So, i'll be testing out this new setup and if all's good, ill be on the phone with Primus TBB canceling the account. ;)

[Badder]

I'm sorry if this is a painfully basic question...but everyone here is so helpful, I thought I'd ask.
Are you saying, since my set up is:
internet/modem ---> router---> to ATA and PC
My ATA is therefore considered to be 'behind' the router?
VoIP information is up and downstream data? So setting up a QoS would help a bit (for the up) during heavy downloading..no?

Yes, in this setup QoS should help for upstream data, but not by much during heavy downloading because you're, well... downloading (downstream). It'd help during heavy uploading, which is not the case for most residential consumers.

[Lulz]

another question.
great thread!

the question is:

is there any way to change the number your DID is being forwarded to using internet on cell phone?

Because the way it works right now, is when you click on login, it logs you in fine. But then, when you move your mouse over the corresponding links, it has a drop-down menu.

That menu is problematic because Opera Mini does not have drop-down feature, so you can not click on those links using your phone.

When I will get to Europe, I will buy a SIM card, and will have to forward my DID to that prepaid European SIM card using my phone and WiFi connection.
What can I do in this situation?

[MXMike]

...That menu is problematic because Opera Mini does not have drop-down feature, so you can not click on those links using your phone.

When I will get to Europe, I will buy a SIM card, and will have to forward my DID to that prepaid European SIM card using my phone and WiFi connection.
What can I do in this situation?
You could create bookmarks to link directly to the appropriate DID features in your account. So use this link to enter your new European phone number:
https://www.voip.ms/m/callforwarding.php

Then create a bookmark to the link to manage your DID, where you can then edit the DID to route calls to your new forwarding number.
https://www.voip.ms/m/managedid.php

I agree that their menu interface works great on a full desktop browser, but the javascript drop-down menus could cause problems on a mini-browser. I would prefer that their site included a sidebar menu linking to all the features. It would also be nice if there was a way to activate call-forwarding by dialling your DID from another phone and entering a code or something.

[randomroyalty]

Thanks to this thread I switched from my long-standing voip provider (Téliphone) to voip.ms. While Téliphone's CS was very responsive, call quality was inconsistent and they have been dragging their heels in adding services (IVR, which I needed).

So far extremely impressed with voip.ms call quality and CS but be warned...configuration is not for the faint of heart. Having said that, this is a great example of how being tech savvy can save you money!

I went from 2 DIDs to 4, and it costs me $20 less per month!

[Sneaks]

That menu is problematic because Opera Mini does not have drop-down feature, so you can not click on those links using your phone.


Won't help you at the moment, but I know they have listed some improvements planned for mobile phones like BlackBerries, etc. in their feature page.

I'm hoping for good things, a I can do *some* stuff from my BB, but it is a bit cumbersome.

[mondriano]

I'm happy to report that after all my problems (detailed above), I am connected and able to make/receive calls on my voip.ms line. Looking forward to $4/month phone bills.

As bizarre as it sounds, it seems that the problems I was encountering was the order of powering on my devices. When I powered on the router, followed by the ATA, no connection. When I powered on the ATA, followed by the router, I instantly connected to the voip.ms servers. I have since repeated both processes, and found that this is, indeed, the cause of the problems.

Either way, thanks to everyone who made suggestions...much appreciated! And the to OP, a huge thanks!

[Sneaks]

I'm happy to report that after all my problems (detailed above), I am connected and able to make/receive calls on my voip.ms line. Looking forward to $4/month phone bills.

As bizarre as it sounds, it seems that the problems I was encountering was the order of powering on my devices. When I powered on the router, followed by the ATA, no connection. When I powered on the ATA, followed by the router, I instantly connected to the voip.ms servers. I have since repeated both processes, and found that this is, indeed, the cause of the problems.

Either way, thanks to everyone who made suggestions...much appreciated! And the to OP, a huge thanks!

Great to hear, enjoy!

[Lulz]

Every day I find this thread addicting more and more.

We currently pay $55/month taxes included for our Bell home line. We so much? Because we need Call Forwarding, CallerID, VoiceMail, Call Waiting and 3 Way Calling. It sucks paying that much, but there's no competition for land line.

So with that in mind, I am thinking of transferring Bell to VOIP.ms

But there are few issues though. I'd be glad if any of you can answer:

1) i have internet with TekSavvy, and I heard of something called Dry-Loop and that if there's no Bell line active, I think TekSavvy charges $10 more. Am I understanding it correctly?

2) With VOIP, if I am not mistaken, when internet goes down, phone goes down as well...correct?

3) If the above is correct, what are some ways to avoid having lines cut off...I really don't want to be left without home phone.

4) How heavy is the usage of VOIP during the call? If somebody is downloading torrents and newsgroups, will voice quality suffer?

5) And vice-versa, if somebody is using the phone line, will latency suffer with games and YouTube etc...

6) I always hear you guys talk about routers when talking about phone lines. Is it required? I never used a router in my life, and only had 1 PC connected to internet without any firewalls or routers. If I do need it, what would be my starting cost to get transferred from Bell to VOIP.ms? Aside from the port fee that VOIP.ms charges you...I am talking about hardware costs.

[Candeeman]

2) With VOIP, if I am not mistaken, when internet goes down, phone goes down as well...correct?

you are not mistaken

[RMSimard]

1) i have internet with TekSavvy, and I heard of something called Dry-Loop and that if there's no Bell line active, I think TekSavvy charges $10 more. Am I understanding it correctly?

Yup. Do check with them for exact amount. Dry-Loop is essentially paying Bell through TekSavvy for copper-wire connection service.


2) With VOIP, if I am not mistaken, when internet goes down, phone goes down as well...correct?

Yes.


3) If the above is correct, what are some ways to avoid having lines cut off...I really don't want to be left without home phone.

Use a reliable internet service provider (TekSaavy is one), buy a good new UPS for your networking equipment should there be a power failure, use a cellphone (you can get cheap cellphones and service for occasional/emergency use from Petro Canada and 7-Eleven)


4) How heavy is the usage of VOIP during the call? If somebody is downloading torrents and newsgroups, will voice quality suffer?

VoIP bandwidth usage is low compared to available bandwidth on modern hi-speed internet. Read about Quality of Service (QoS) for your router, or limit bandwidth usage directly inside file sharing applications.


5) And vice-versa, if somebody is using the phone line, will latency suffer with games and YouTube etc...

Shouldn't. See QoS above.


6) I always hear you guys talk about routers when talking about phone lines. Is it required? I never used a router in my life, and only had 1 PC connected to internet without any firewalls or routers. If I do need it, what would be my starting cost to get transferred from Bell to VOIP.ms? Aside from the port fee that VOIP.ms charges you...I am talking about hardware costs.

Wow. No router, no firewall, running a PC, downloading stuff off newsgroups and torrents!? That's like sleeping with cheap prostitutes and using no condoms. :rolleyes: Do download Avira antivirus for a free checkup or pony up cash for Kaspersky or some other quality suite, in case your computer is now a zombie. I wouldn't do any kind of banking or credit card purchases on that possibly compromised computer. A good basic router will run about $50. Search for threads about Asus routers running Tomato or DD-WRT firmwares.

[djino]

1) i have internet with TekSavvy, and I heard of something called Dry-Loop and that if there's no Bell line active, I think TekSavvy charges $10 more. Am I understanding it correctly?

Correct. If you sign up with Bell Directly on dry-loop internet, there is no Dry loop fee, so you may want to keep this in mind.


2) With VOIP, if I am not mistaken, when internet goes down, phone goes down as well...correct?

Yes. but ask yourself, how often does this occur, and in an emergency, do you usually have a cell phone?


3) If the above is correct, what are some ways to avoid having lines cut off...I really don't want to be left without home phone.

Internet going down isn't really something you can control, but how likely is it to happen. Have you had many outages in your area when it comes to Internet service?

As for a power outage, one thing that can protect you is using a UPS device. But again, I'd prefer to get an emergency cell phone first or in an addition to UPS device.


4) How heavy is the usage of VOIP during the call? If somebody is downloading torrents and newsgroups, will voice quality suffer?

Yes, if someone is a heavy into downloaded P2P/torrents, then this will effect call quality not less you can setup QOS within your router settings.


5) And vice-versa, if somebody is using the phone line, will latency suffer with games and YouTube etc...

Voip traffic has very little impact on anything else. You should not concern yourself with this here.


6) I always hear you guys talk about routers when talking about phone lines. Is it required? I never used a router in my life, and only had 1 PC connected to internet without any firewalls or routers. If I do need it, what would be my starting cost to get transferred from Bell to VOIP.ms? Aside from the port fee that VOIP.ms charges you...I am talking about hardware costs.

You will need a router if you connect more than one device on the internet. If you have more than one computer, home networking with a printer, gaming consoles like the Wii, X-box, etc. A router is basically there to split your internet connection among many networking devices (including a voip terminal).

If your current setup is just one computer, then you will need to purchase a router. They are pretty inexpensive and easy to setup.

What you will need:

Routers can be priced anywhere from $20-$100
A voip terminal - $50 to $100 (perhaps look for a router/voip combo)
Voip.ms porting fee to port your number to Bell - $25

Not necessary, but if you want UPS - $40-$200

djino
"That should get you started"

[Lulz]

Thank you djino and RMSimard. You've been great help.

1) How about when VOIP servers go down on their end? I just read on their front page about downtime for Los Angeles servers. How often does it happen when your internet works fine, but your phone doesn't because of VOIP.ms servers?

2) And when your internet is down, is there any way to automatically forward the call to your cell phone when line is unreachable? Because we don't use home phone that many times, so it would suck if phone line was not working, but we did not know about it because nobody tried to call anybody to understand that there's no dial tone.

3) Does Bell offer any holds for home lines? We will be going away for 2 month starting in June, so if Bell has any hold services where they disconnect the phone temporarily, I won't have to worry about transfering to VOIP until after the trip. Then when I come back in August, I will start the process of moving to VOIP.ms
Tried calling, but it seems I need to wait until Monday 9AM to call.

[spudz2000]

I have had a voip.ms DID for about 6 months now and really happy with them and moved from another provider (les.net).

However lately for the last few weeks I'm noticing that quite often when I make an outgoing call after I dial the number on my SIP phone it makes the usual sounds until the call connects and then I just get dead air. No ring tone sound on my end. The thing is the call IS GOING THROUGH on the other end. It's just that i can't hear anything. I know this because I have often called back again and once it rings on my end and they pick up I asked them if the previous time I called them if it rang on there end, and they always say yes. They also say that they could here me as well but i could here nothing.

This is with the value routing of .5c per min. I did send in a ticket and they suggested a few things and also asked if I was using the value routing.

So I have now switched to the premium routing (1.25c per min) and am going to see if this is still happening but it is annoying. I want to know if anyone here using voip.ms has experienced this type of issue before?

[signup]

I have had a voip.ms DID for about 6 months now and really happy with them and moved from another provider (les.net).

However lately for the last few weeks I'm noticing that quite often when I make an outgoing call after I dial the number on my SIP phone it makes the usual sounds until the call connects and then I just get dead air. No ring tone sound on my end. The thing is the call IS GOING THROUGH on the other end. It's just that i can't hear anything. I know this because I have often called back again and once it rings on my end and they pick up I asked them if the previous time I called them if it rang on there end, and they always say yes. They also say that they could here me as well but i could here nothing.

This is with the value routing of .5c per min. I did send in a ticket and they suggested a few things and also asked if I was using the value routing.

So I have now switched to the premium routing (1.25c per min) and am going to see if this is still happening but it is annoying. I want to know if anyone here using voip.ms has experienced this type of issue before?

Same here 2 days ago, other party can't hear me and when I tried to call my number the line is busy, but now everything back to normal and also using the value rate.

[djino]

1) How about when VOIP servers go down on their end? I just read on their front page about downtime for Los Angeles servers. How often does it happen when your internet works fine, but your phone doesn't because of VOIP.ms servers?

I can't speak for voip.ms, but I've had voip from cia.com and in the 3 years I've had service, I'd say I've had about 3 outages that lasted only a few hours in each occurrence.


2) And when your internet is down, is there any way to automatically forward the call to your cell phone when line is unreachable? Because we don't use home phone that many times, so it would suck if phone line was not working, but we did not know about it because nobody tried to call anybody to understand that there's no dial tone.

yes. but you'd have to do it via voip.ms website. There is a section you could tell it to forward numbers (even conditional forwarding options).


3) Does Bell offer any holds for home lines? We will be going away for 2 month starting in June, so if Bell has any hold services where they disconnect the phone temporarily, I won't have to worry about transfering to VOIP until after the trip. Then when I come back in August, I will start the process of moving to VOIP.ms
Tried calling, but it seems I need to wait until Monday 9AM to call.

Perhaps they do, I'm not sure.

What you could do is setup voip.ms now (port your number over to them) and you'd just be paying $0.99/month while you are away and even have voip.ms foward your number to a cell phone you have while on your trip.

djino

[death_hawk]

1) How about when VOIP servers go down on their end? I just read on their front page about downtime for Los Angeles servers. How often does it happen when your internet works fine, but your phone doesn't because of VOIP.ms servers?
They rarely have outages. If you're REALLY concerned, you can set up failover to a different PoP. Or just use a cell phone.


2) And when your internet is down, is there any way to automatically forward the call to your cell phone when line is unreachable? Because we don't use home phone that many times, so it would suck if phone line was not working, but we did not know about it because nobody tried to call anybody to understand that there's no dial tone.
There is a failover option in the portal, but I hear it's broken.
I personally have a PBX so I handle all that stuff myself. Hell, most of my DIDs go to a ring group to simultaneously ring all my phones.


3) Does Bell offer any holds for home lines? We will be going away for 2 month starting in June, so if Bell has any hold services where they disconnect the phone temporarily, I won't have to worry about transfering to VOIP until after the trip. Then when I come back in August, I will start the process of moving to VOIP.ms
Tried calling, but it seems I need to wait until Monday 9AM to call

I'd just port it now.

[xrvr]

Just signed for a service, got 2 DID, for myself and my kid.
Installed Acrobits SIP client for iPhone ($8) and started making calls right away.
They have option to enable calls over 3G, but it's extra 10 $.
Great web control panel, tons of options, so glad didn't go with freephoneline.ca :)
Setup was very easy, pretty straightforward.

[enon]

What does DID stand for?

[xrvr]

http://en.wikipedia.org/wiki/Direct_Inward_Dialing

[fuzzy]

Does anyone know if Rogers blocking voip over 3G or not? Thanks,


Just signed for a service, got 2 DID, for myself and my kid.
Installed Acrobits SIP client for iPhone ($8) and started making calls right away.
They have option to enable calls over 3G, but it's extra 10 $.
Great web control panel, tons of options, so glad didn't go with freephoneline.ca :)
Setup was very easy, pretty straightforward.

[xrvr]

From Acrobits site:

Note for users in Greece, Poland and Hungary: Acrobits Softphone is not currently available on the App Store in these countries because Apple does not allow VoIP applications that work over 3G to be sold in these countries. We are working on creating a Wi-Fi only version of Acrobits Softphone for these countries.

***VOIP OVER 3G - IMPORTANT NOTICE - PLEASE READ***

Be aware that some mobile network operators prohibit or restrict the use of VoIP (Voice over Internet Protocol) over their network. They may prohibit the use of VoIP over their network or impose additional fees and/or charges when using VoIP over their network.

They don't mention Canada, so I guess it should work.
Only one way to find out :)

[airstream]

Does anyone know if Rogers blocking voip over 3G or not? Thanks,

I don't believe so, I can place voip calls on Fido over 3G.

[Kwirky]

I don't believe so, I can place voip calls on Fido over 3G.

Same here; I can place 3G calls over Rogers, just don't expect to be able to receive calls over 3G since you can't forward the needed ports to receive calls over the 3G internet.

[xrvr]

Sweet :)
With free incoming, you basically get unlimited calls on your smartphone. (Except cost of actual cellphone plan).

[tk11]

HUGE problems on the weekend with these guys.
Saturday they had server problems and calls were getting cut off and then the line went dead for a good 3+ hours.
Sunday was ridiculous everybody who called the line received a busy signal, the number couldn't even be forwarded. It would just stay busy, this was basically the entire day until like 8pm (no incoming calls). Absolutely ridiculous service this weekend.
Been with them since Feb, and its happened on 1 or 2 other occasions as well alot of times on weekends.
If they pull this ish again I might have to cancel.

[a2vr6]

Just signed for a service, got 2 DID, for myself and my kid.
Installed Acrobits SIP client for iPhone ($8) and started making calls right away.
They have option to enable calls over 3G, but it's extra 10 $.
Great web control panel, tons of options, so glad didn't go with freephoneline.ca :)
Setup was very easy, pretty straightforward.

Does anything need to be done on the control panel once you get the DID to have the iphone working (other than purchasing the app)?

[paulo500]

HUGE problems on the weekend with these guys.
Saturday they had server problems and calls were getting cut off and then the line went dead for a good 3+ hours.
Sunday was ridiculous everybody who called the line received a busy signal, the number couldn't even be forwarded. It would just stay busy, this was basically the entire day until like 8pm (no incoming calls). Absolutely ridiculous service this weekend.
Been with them since Feb, and its happened on 1 or 2 other occasions as well alot of times on weekends.
If they pull this ish again I might have to cancel.No problems here. Weekend worked flawlessly. You sure your internet wasn't down?

[a2vr6]

HUGE problems on the weekend with these guys.
Saturday they had server problems and calls were getting cut off and then the line went dead for a good 3+ hours.
Sunday was ridiculous everybody who called the line received a busy signal, the number couldn't even be forwarded. It would just stay busy, this was basically the entire day until like 8pm (no incoming calls). Absolutely ridiculous service this weekend.
Been with them since Feb, and its happened on 1 or 2 other occasions as well alot of times on weekends.
If they pull this ish again I might have to cancel.

No issues this weekend either, I am using the Toronto servers.

[apn64]

HUGE problems on the weekend with these guys.
Saturday they had server problems and calls were getting cut off and then the line went dead for a good 3+ hours.
Sunday was ridiculous everybody who called the line received a busy signal, the number couldn't even be forwarded. It would just stay busy, this was basically the entire day until like 8pm (no incoming calls). Absolutely ridiculous service this weekend.
Been with them since Feb, and its happened on 1 or 2 other occasions as well alot of times on weekends.
If they pull this ish again I might have to cancel.

No problems here either, and I'm using the Toronto (L1=DID) and New York (L2=Outgoing) servers.

This weekend we called to/from;

- local numbers: received and made calls
- Canada long-distance (London & Vancouver): received and made calls
- England: received voice mail and called back

No issues whatsoever.

[input]

I'm trying to call the Philippines right now, but I think I'm doing something wrong.

Don't I just put in the area code number + the phone number? All I'm getting is like a dead line tone.

[apn64]

I'm trying to call the Philippines right now, but I think I'm doing something wrong.

Don't I just put in the area code number + the phone number? All I'm getting is like a dead line tone.

If "area code" means what I think it does, then yes, it sounds like you messed up. You need to dial something like 011+country code+area code+number

Some countries e.g. UK require you to remove a 1-prefix from the "area code" to dial from outside the country.

Finally, if you're getting dead air, it's likely that your dial plan is not setup to accept the number format that you're dialing, so the ATA is waiting for additional digits to match the number to a designated pattern.

[vint]

I had a problem not being able to call certain country. After they replied that they do block certain area codes until you send them ID I have requested a refund. I mean come on what is wrong with them?

[jackie999]

Wouldn't you need to precede the area code with the country code?

[input]

If "area code" means what I think it does, then yes, it sounds like you messed up. You need to dial something like 011+country code+area code+number

Some countries e.g. UK require you to remove a 1-prefix from the "area code" to dial from outside the country.

Finally, if you're getting dead air, it's likely that your dial plan is not setup to accept the number format that you're dialing, so the ATA is waiting for additional digits to match the number to a designated pattern.

Yeah, that's what I pretty much figured out. According to my Account Settings, my dialing mode is set to the standard North American style. I just thought that since it was VOIP, I could just type the number in and it would connect.




North America (Recommended): You can dial to countries part of the North American Numbering Plan Administration by dialing 10 or 11 digits. (with or without the 1 prefix). You'll need to use the 00 or 011 prefix to call international numbers.

E164: You dial numbers by using Country Code first. You don't need to send the 00 or 011 prefix when using this mode, but it's still supported.

Anyways, thanks everybody.

[xrvr]

Just got email from voip.ms:


For a limited time, VoIP.ms is waiving the setup fee on orders for local international numbers.

Countries Included:
ARGENTINA, AUSTRALIA, BELGIUM, BRAZIL, BULGARIA, CHILE, CROATIA, CYPRUS, CZECH REPUBLIC, FINLAND, FRANCE, GEORGIA, GERMANY, GREECE, HUNGARY, IRELAND, ISRAEL, ITALY, JAPAN, LATVIA, LITHUANIA, MEXICO, NETHERLANDS, NEW ZEALAND, NORWAY, PERU, POLAND, SLOVAKIA, SLOVENIA, SPAIN, SWEDEN, SWITZERLAND, UNITED KINGDOM.

* Limited time offer . Offer valid for new orders only. All prices are in USD.
** This offer does not apply to non-geographical numbers (national) and toll-free numbers.

[BillyH]

Does anyone know if callback double charges like DISA does?
(once to call you back and once to route your call?)

If so, is there an option to use your cellphone with voip.ms without getting charged twice?


Thanks

[Mazda323]

OK all systems go!
Disconnected the Primus ATA box and replaced it with the new Cisco one. Powered it up and checked all my phones in the house. Dial tone was good! Dialed into the ata box using cisco's keypad codes and retrieved the IP it's using. Plugged the ip and other info (from the voip.ms config page) into the cisco ata via web interface. Everything registered as expected on the voip.ms user portal too.

So, i'll be testing out this new setup and if all's good, ill be on the phone with Primus TBB canceling the account. ;)Hi Paulo500, did you port your Primus number over to Voip.ms? If yes, how much did Primus charge you?
I'm planning to switch to Voip.ms but I don't know if it's worth to keep my existing number or not.

[paulo500]

Hi Paulo500, did you port your Primus number over to Voip.ms? If yes, how much did Primus charge you?
I'm planning to switch to Voip.ms but I don't know if it's worth to keep my existing number or not.As of yet, Primus hasn't charged me anything. I called them up after everything was switched over to cancel, and they said that im no longer in the system and everything is paid. So Voip.ms not only ported my number, but they closed my Primus account! LOL. I suppose this is normal once you ported a number. The losing company automatically closes that account because the number is gone.
Maybe that $25 fee voip.ms charges helps cover any residual fees the losing company charges? Im not sure.

So i emailed Primus to see when/if they are going to contact me regarding the return of their ATA. (as i was renting it from them.) Im assuming they will send me a mail bag and an address sticker to send them their unit back. (Did this when my first ATA from them crapped out and I needed a new one). I haven't heard back from them yet. I'll give it another day, then ill call them. I don't want to be stuck with a fine for 'keeping' their ATA...Even though it's useless to me because it's locked to the Primus specs.

[Mazda323]

As of yet, Primus hasn't charged me anything. I called them up after everything was switched over to cancel, and they said that im no longer in the system and everything is paid. So Voip.ms not only ported my number, but they closed my Primus account! LOL. I suppose this is normal once you ported a number. The losing company automatically closes that account because the number is gone.
Maybe that $25 fee voip.ms charges helps cover any residual fees the losing company charges? Im not sure.

So i emailed Primus to see when/if they are going to contact me regarding the return of their ATA. (as i was renting it from them.) Im assuming they will send me a mail bag and an address sticker to send them their unit back. (Did this when my first ATA from them crapped out and I needed a new one). I haven't heard back from them yet. I'll give it another day, then ill call them. I don't want to be stuck with a fine for 'keeping' their ATA...Even though it's useless to me because it's locked to the Primus specs.Thanks a lot Paulo500. I don't want to keep their ATA either. Hopefully they'll give you a letter sooner.

[signup]

Anyone in Mississauga area have problem today in their voip? I have this intermittent issue that the other party can't hear me. This problem occurs trice already since I signed up to them last month.

[jackie999]

You should put in a ticket with voip.ms..note the times of the calls from your CDR list...they are very good at resolving issues.

[Mazda323]

Anyone in Mississauga area have problem today in their voip? I have this intermittent issue that the other party can't hear me. This problem occurs trice already since I signed up to them last month.Any comment?

[BargainBuff]

I have 2 questions that hopefully can be resolved (support is taking quite long to respond----even their chat feature is down):

1) What do subaccounts do? How are they useful?

2) Is this scenario possible---person calls DID and IVR picks up, then they make a selection (e.g. 1), should transfer to ring my phone (connected via Pap2), but if I don't pick up after say 3 rings, goes to voicemail. Can I do this with just 1 DID (and perhaps subaccounts) or do I need a separate DID to forward the person from the IVR to my phone?

Any insight would be much appreciated.

[WildChild]

Hello BargainBuff,
here are some answers:

1) Subaccounts are used to connect devices. You need subaccounts for every devices/lines/softphones you want to connect.

In my setup, each line of my SPA2102 (same thing with your PAP2T if you want) are connected with the main account and one subaccount. I also have a soft-phone on my MacBook (the software is Telephone) connected with its own subaccount.

One DID can ring all those devices at the same time, the first to answer gets the call. You have to use "ring groups" for this (up to 5 subaccounts in one group).

2) You only need one DID for that. The DID is associated to the IVR entry. If it works like I think, when the person press 1 to ring your ATA (PAP2T subaccount or ring group if you configure like I said in 1) after the timeout configurable in the main account/subaccount screen, the call should fall in the "Routing if Destination No Answer" category configurable for the DID. You can set this failover to reach the voicemail then.


I have 2 questions that hopefully can be resolved (support is taking quite long to respond----even their chat feature is down):

1) What do subaccounts do? How are they useful?

2) Is this scenario possible---person calls DID and IVR picks up, then they make a selection (e.g. 1), should transfer to ring my phone (connected via Pap2), but if I don't pick up after say 3 rings, goes to voicemail. Can I do this with just 1 DID (and perhaps subaccounts) or do I need a separate DID to forward the person from the IVR to my phone?

Any insight would be much appreciated.

[RMSimard]

1) What do subaccounts do? How are they useful?

2) Is this scenario possible---person calls DID and IVR picks up, then they make a selection (e.g. 1), should transfer to ring my phone (connected via Pap2), but if I don't pick up after say 3 rings, goes to voicemail. Can I do this with just 1 DID (and perhaps subaccounts) or do I need a separate DID to forward the person from the IVR to my phone?

1) You can have a sub account for a softphone on your laptop and another connected to a sip-enabled app on your smartphone. You then can setup a simultaneous ring group that includes your PAP2 and your subaccounts so that when someone calls your DID your ATA will ring and so will your softphone if you're working in a coffee shop and *possibly* your smartphone anywhere you are unless incoming VoIP traffic is blocked by cell operator. :-) [not tested the last one, as don't have a smartphone yet seeing as I loose 1-2 cellphones a year to ever-present gravity. I just simultaneously ring my cellphone instead.] And you can also dial from any of those devices/locations too, so you can be free from your ATA's limited range.

2) Yes, only one DID required, not even a subaccount. Link your DID to the IVR, set selection 1 to ring your main account/ATA for 3 rings tops, then activate failover to voicemail.

[BargainBuff]

Thank you both for the detailed answers. I'll go back to tweaking my settings now that I know my goal can be achieved with 1 DID. Many thanks for all the help! :)

[signup]

Any comment?

Yesterday I chatted to their support and did some adjustments and also used the softphone but to no avail. He opened a ticket for a 2nd level support but never receive an email or advice. I'm losing confidence on this voip thing.

This morning everything is back to normal. Don't know when it is going to occur again.

[Mazda323]

Are there anybody who find your PAP2T-NA too hot to be hold?
I just set up mine and find it so hot even than my computer video card.
Is it normal or I got a faulty one?

[paulo500]

Are there anybody who find your PAP2T-NA too hot to be hold?
I just set up mine and find it so hot even than my computer video card.
Is it normal or I got a faulty one?Mine seems ok. I fastened mine to my wall panel the other day, I didn't notice any heat issues...

[mondriano]

Are there anybody who find your PAP2T-NA too hot to be hold?
I just set up mine and find it so hot even than my computer video card.
Is it normal or I got a faulty one?

I've often wondered the same thing. I have just changed from the SPA2102 to the PAP2T-NA, and both have been ridiculously hot. They literally feel as if they are going to explode. Weird.

[jackie999]

Mine doesn't get hot at all...it's mounted on the side of my computer desk and isn't hot..just warmish...

[apn64]

Mine doesn't get hot at all...it's mounted on the side of my computer desk and isn't hot..just warmish...

Ditto.

[Mazda323]

I've often wondered the same thing. I have just changed from the SPA2102 to the PAP2T-NA, and both have been ridiculously hot. They literally feel as if they are going to explode. Weird.Mondriano, can I ask where you bought your PAP2T-NA from? Canada Computers? I got mine from China.

[mondriano]

Mondriano, can I ask where you bought your PAP2T-NA from? Canada Computers?

Yes, Canada Computers.

[jackie999]

I was surprised to see that some of the PAP2T's are counterfeit..possibly that's why yours are hot..they're clones?

Found this site that will tell you is the MAC addy is correct for the device...
http://www.coffer.com/mac_find/

[xrvr]

Here's my settings for Nokia E63 SIP client for Voip.ms:

STEP 1

Go to Tools / Settings / Connection / SIP settings. Create the following profile
Profile Name: Voipms
Profile: IETF Default
Access point: (your WLAN AP name)
Public user name: (your 6 digit voip.ms main username)@174.137.63.202 (or toronto.voip.ms)
Use Compression: No
Registration: "Always on "
Use Security: No

STEP 2

You do not need to enter Proxy Server settings
Proxy Server Address: none
Registrar Server:none
Realm: none
User Name: none

STEP 3

Enter the following for Registrar Server settings
Registrar Server: toronto.voip.ms
Realm: toronto.voip.ms
User Name: (your 6 digit voip.ms main username)
Password: main account password
Transport Type: AUTO
Port: 5060

Hope this helps someone, as it took me couple days to figure out settings. As I mentioned before, for iPhone it was just a matter of buying SIP client (Acrobits), and I was done in 2 minutes.

[akmeikle]

I had a problem not being able to call certain country. After they replied that they do block certain area codes until you send them ID I have requested a refund. I mean come on what is wrong with them?

So you're objecting to their security measure? I'm sure some area codes are problematic to them which is why they want some sort of verification before letting you call them /shrug.

Shame you want a refund cuz you're missing out on a great service.

[xrvr]

One of the reasons why they do this Mumbai underworld now using VoIP from New Jersey (http://www.dslreports.com/forum/r24241014-Mumbai-underworld-now-using-VoIP-from-New-Jersey)

Mumbai : The probe into the murder of 26/11 defence lawyer Shahid Azmi has revealed that underworld gangs are getting tech-savvy in their communication, resorting to methods traditionally used by terror outfits.

According to the Mumbai Police Crime Branch which investigated the murder, fugitive gangsters Bharat Nepal and Vijay alias Bala Shetty, who are wanted in the case, spoke to their shooters in the city using Voice over Internet Protocol (VoIP), calling platforms based in different cities across the world. Sources revealed that Nepali and Shetty used VoIP platforms based in Los Angeles, London, Toronto and Hong Kong to pass on orders to their men in Mumbai and coordinate the hit on Azmi.

“Some of the VoIP services used were Teleglobe, RAITEE, and Del Telecom Holding. The numbers used by Nepali, Shetty and the arrested accused were connected to these platforms. Even the Pak handlers involved in the 26/11 attacks had used a similar modus operandus to pass on instructions,” said an officer who didn’t wish to be named.

It's for your own good :)

[4public]

Today toronto.voip.ms SIP server does not work (down)
No news from the main www.voip.ms page
I have reconfigured SIP to montreal.voip.ms and service is up again

This is a main problem with VOIP phones - you will not know that your phone is not responding, till you check the tone. How many incoming calls were missed? :evil:

[jackie999]

Mine is going thru Toronto and I've been using my phone today. As far as I know, if there is a server issue you're rerouted (by the service) to an alternate so there should be no down time. Did you speak to support and ask why you were down?

[Moe Hunter]

Are there anybody who find your PAP2T-NA too hot to be hold?
I just set up mine and find it so hot even than my computer video card.
Is it normal or I got a faulty one?

All ours run very hot also, dont have them laying flat on their rubber feet, there is no vents on top. attach a stickie hook & have it hang down the side of your computer case by a fan if possible, or on a wall on their side with the vent slots running up & down, this seams to keep them cooler, not cool but not cooking hot either.

[Lulz]

This is a main problem with VOIP phones - you will not know that your phone is not responding, till you check the tone. How many incoming calls were missed? :evil:

I agree
it's very annoying and is probably main reason why i wont get it for home line

but perhaps you can have failover forward option so that whenever there is a failure to reach you on your home VOIP line, it will get automatically forwarded to your cell phone. it will only work when there's a failure and not all the time.

[paulo500]

All ours run very hot also, dont have them laying flat on their rubber feet, there is no vents on top. attach a stickie hook & have it hang down the side of your computer case by a fan if possible, or on a wall on their side with the vent slots running up & down, this seams to keep them cooler, not cool but not cooking hot either.Yeah I just used the two slots at the back for mounting...and hung it on my wall like so:
http://imgur.com/wJPLw.jpg

[destinet]

Today toronto.voip.ms SIP server does not work (down)
No news from the main www.voip.ms page
I have reconfigured SIP to montreal.voip.ms and service is up again

This is a main problem with VOIP phones - you will not know that your phone is not responding, till you check the tone. How many incoming calls were missed? :evil:

Why you dont configure de failover! If your Internet is down or if they can't reach your ata then IT Will be forward to your cell phone.

[Dave98]

What do you guys do with phones you can't connect to the adapter? Is it possible to use multiple adapters for a single number?

I currently have RHS so I don't know if simply replacing the rogers voip modem with the PAPT2 will do the job to feed the entire house.

Otherwise, the simplest seems to be just using a single wireless/dect phone system I suppose.

[4public]

Mine is going thru Toronto and I've been using my phone today. As far as I know, if there is a server issue you're rerouted (by the service) to an alternate so there should be no down time. Did you speak to support and ask why you were down?

Yesterday I got the answer from the support - good fast service
They recommended to use IP instead of DNS name. My SIP device used dns server name and could not reach the server to be registered.
I think the issue was deeper - my dynamic outgoing IP address was changed and server could not register it for some reason.

[jackie999]

Yesterday I got the answer from the support - good fast service
They recommended to use IP instead of DNS name. My SIP device used dns server name and could not reach the server to be registered.


Ah, okay, makes sense now. I had same problem a few weeks ago. I got an email suggesting a change in the server name, which I did. Then next day I was checking my voip.ms portal and noticed it wasn't staying registered..and I contacted them and they suggested the same thing to me and it's been solid ever since. Hopefully this will save others from having the same problem by just changing to the IP.

[jackie999]

...replacing the rogers voip modem with the PAPT2 will do the job to feed the entire house.

Otherwise, the simplest seems to be just using a single wireless/dect phone system I suppose.

I use the PAP2T and plugged in my main panasonic DECT phone into it...then I scattered the remaining panasonic handsets (5) around the house.
I have a wireless modem that has a couple of ports that I use..or a stand alone modem (in bridge mode) with a WRT54GL.
I don't know about a rogers voip modem...can the voip be disabled in it? Does it have an extra port for the PAP2T? Could try asking here (http://www.dslreports.com/forum/voip).

[paulo500]

Ah, okay, makes sense now. I had same problem a few weeks ago. I got an email suggesting a change in the server name, which I did. Then next day I was checking my voip.ms portal and noticed it wasn't staying registered..and I contacted them and they suggested the same thing to me and it's been solid ever since. Hopefully this will save others from having the same problem by just changing to the IP.Noob question: Where in our PAP2T do we see IP or DNS name? I set up everything they requested via their setup page.
Proxy for toronto is toronto.voip.ms right?

I guess i don't need to do anything, as i've been running fine since the beginning. (Atleast I think i have. lol)

[jackie999]

Noob question: Where in our PAP2T do we see IP or DNS name? I set up everything they requested via their setup page.
Proxy for toronto is toronto.voip.ms right?

From the voip.ms configuration for PAP2T


Step 2

Using your favorite web browser from a computer on the same network, point the address to the IP address of your adapter.
(example http://192.168.1.2) Replace 192.168.1.2 by the IP Address your device is currently using.

Step 3

You should now see the web interface of your Linksys/Sipura.

click on the link "Admin", and once the page has reloaded, click again on the link "Advanced View".

Step 4

Under the LINE 1 Tab, Find the following fields and fill them with the following information

Proxy: newyork.voip.ms

We were suggesting, instead of the newyork.voip.ms, use the IP. You'll see that on voip.ms account settings ( i.e. Toronto, ON (toronto.voip.ms / 174.137.63.202 ) I use 174.137.63.202
HTH's

[Moe Hunter]

Yeah I just used the two slots at the back for mounting...and hung it on my wall like so:
http://imgur.com/wJPLw.jpg

Awesome install,
This is how all installs should be. At the demarkation Point or on the pannel board. Battery back-up for the Modem, VoIP & Router so when there is a loss of hydro both the internet & Voip Phone lines still all work. Set-up also allows VoIP 1st crack at the bandwidth which would stop VoIP Line cut outs & Line noise. This install also keeps the space around your computer neat & tidy (No Wire Hell) with much less need mega power at one outlet.

Very Nice install

[gottagetta]

I wasn't able to read the entire thread but I'm interested in getting this setup for my parents who have just moved.

Noob question but does your internet cpnnect speed impact your calls. Would Rogers Ultra Lite be enough to support VOIP? Is there a basic requirement for internet speed?

[Moe Hunter]

I wasn't able to read the entire thread but I'm interested in getting this setup for my parents who have just moved.

Noob question but does your internet cpnnect speed impact your calls. Would Rogers Ultra Lite be enough to support VOIP? Is there a basic requirement for internet speed?

To make sure, call your parents intended provider they will tell you right away so there is no confusion or miss information.

[X360]

I wasn't able to read the entire thread but I'm interested in getting this setup for my parents who have just moved.

Noob question but does your internet cpnnect speed impact your calls. Would Rogers Ultra Lite be enough to support VOIP? Is there a basic requirement for internet speed?
Instead of slow speed 500KB/s Rogers Ultra Lite 2GB bandwidth / month... You can venture with faster Rogers mobile internet HUB 3GB bandwidth / month... You can add unlimited 3G voice plan... with lowest package = $50 / month... With this HUB you can have the internet mobile, simplify a lot for your parents...

http://img39.imageshack.us/img39/5479/hubh.jpg


Just plug the Rocket™ Hub into any power outlet and create a Wi-Fi spot connected to the Rogers Wireless Network and enjoy internet connectivity for up to 10 computers at one time.

Choose from data only plans for email and internet access, or data and voice plans for additional phone connectivity starting from only $35 per month!

No technicians needed or software downloads required – just plug the Rocket™ Hub into a power outlet to enjoy wireless services. Check your coverage to get started!
http://www.rogers.com/web/content/internet-mobile/hubplans

[jackie999]

INoob question but does your internet cpnnect speed impact your calls. Would Rogers Ultra Lite be enough to support VOIP? Is there a basic requirement for internet speed?

I suggest you go to the voip.ms site and use the live chat link and ask them. I'm pretty sure the rogers lite will be fine, but voip staff will tell you for sure. My phone is about $3 a month on voip.ms ..I use teksavvy for my internet...(they are getting into 'cable' soon..).
I wouldn't even consider X360's suggestion
You can add unlimited 3G voice plan... with lowest package = $50 / month.. unless your parents have money to throw away :)

[Mazda323]

:cry:Tried to use IP address instead of toronto.voip.ms. And then my PAP2T seems to be dead. The power light keeps blinking (green-orange-red-green), the Internet light is solid, the two phone lights don't work. Since I can't use brower to log on it, had to ried to dial **** and then 73738# to reset it. But nothing happened. I hate this stuff. It even doesn't carry a RESET key.

[jackie999]

dial **** and then 73738# to reset it. But nothing happened.

I have read that the reset doesn't always work if the PAP2T was previously locked to a provider. Did you buy from an authorized dealer? Perhaps contact them and ask for a replacement...

[Mazda323]

Thanks for reply. It's a PAP2T-NA. I just checked web and found that there are other people had same problem as me.

[gottagetta]

thanks for all the advice. I think I have to get a dialog started with voip.ms. I doubt I'll try that option that X360 suggested but thanks for the idea.


I suggest you go to the voip.ms site and use the live chat link and ask them. I'm pretty sure the rogers lite will be fine, but voip staff will tell you for sure. My phone is about $3 a month on voip.ms ..I use teksavvy for my internet...(they are getting into 'cable' soon..).
I wouldn't even consider X360's suggestion unless your parents have money to throw away :)

[jackie999]

Thanks for reply. It's a PAP2T-NA. I just checked web and found that there are other people had same problem as me.

I was reading back over this thread and came across another post of yours where you mentioned your PAP2T was overheating...if you have a counterfeit/clone device..that could be why it's not doing a reset.

[Mazda323]

I was reading back over this thread and came across another post of yours where you mentioned your PAP2T was overheating...if you have a counterfeit/clone device..that could be why it's not doing a reset.My PAP2T-NA seems like a clone unit. But even those from Canada computers have overheating issues. Where can I get a REAL one? I read the posts and found yours is OK. May I ask where you got it?
Also I'm tring to find ATA with physical reset buttons. D-link does but their ATA discontinued in North America.

[Panda Bear]

I got my PAP2T-NA from DELL.ca a month ago! so far! zero issue.

I was thinking that at least Dell's suppliers won't sell a fake one.

The price is OK too with free shipping!

http://accessories.us.dell.com/sna/productdetail.aspx?c=ca&l=en&s=dhs&cs=cadhs1&sku=A0703190&dgc=SS&cid=38279&lid=1344936

[X360]

I wouldn't even consider X360's suggestion unless your parents have money to throw away :)
Are you kidding? only $50 / month for 3G mobile internet 3GB/month + unlimited mobile local incoming phone with Caller ID... and you say people don't have money to pay... It's way below the average people paying for internet + phone...

http://www.rogers.com/web/content/internet-mobile/hubplans

and look like his/her parents currently pay over $50/month for internet + phone with Rogers...

[jackie999]

It's way below the average people paying for internet + phone..

It may be below the average that people are paying..but not this peep :)
Their "The Government Regulatory Recovery Fee ranges from $2.46-$3.46/line/month (varies by province and plan selected)" alone is more than I paid for my voip.ms last month..granted, I'm minimum usage at about 6 hours of calls a month. And your Rogers charge of $50 for 3gig limit? Oh no..that would never do for me..but to each his/her own..that's why we're here on RFD..to compare and get the best deal possible.

Mazda323, I got my PAP2T at Canada Computers..I saw where the other poster, who also bought from Canadacomputers, mentioned the overheating issues. It would be interesting if the Mac addy verified their device as a clone ..would all counterfeit/clones have the same erroneous Mac addy I wonder?
On, and I don't think there is a reset button..the reset thru the phone works fine here..I know cause I did it to check :)

[4public]

Attention
Hidden charge from Bell if you do your local phone porting.

Bell has to get 30 days notice for cancellation, otherwise you will be charged by Bell 28.95$
I had month to month Bell plan and I was successfuly ported to VOIP for 25$
Now Bell is charging additional 28$ (I did not let them know about that 30 days before)

Do not do this mistake and let them know if you want to keep some money.

This is a link from Bell about early termination:
http://support.bell.ca/Documents/HomePhone/Legal/utos%28en%29.pdf

[jackie999]

Attention
Hidden charge from Bell if you do your local phone porting.

Bell has to get 30 days notice for cancellation, otherwise you will be charged by Bell 28.95$
I had month to month Bell plan and I was successfuly ported to VOIP for 25$
Now Bell is charging additional 28$ (I did not let them know about that 30 days before)

Do not do this mistake and let them know if you want to keep some money.

Thanks for the warning 4puplic...I remember when I was switching from bell to teksavvy a while ago, I was told this and I made sure to call bell and ask before I made arrangements. I didn't want to call bell but had to suffer thru all of the "but teksavvy uses inferior equipment" "bell is better" "teksavvy is much slower" on and on and on....it just makes you all the more certain you want to get clear of these underhanded tactics dished out by bell and rogers. It's a refreshing change to sign up with a company like voip.ms or teksavvy where every charge or fee is right there on their website..not hidden under nearly invisible links and tiny unreadable print.

[Phils]

I ported one of my bell lines over last month (April 23, actually) and I just received my latest bill from them. There was no $25 charge. I was not on any sort of plan or contract with them.

Hopefully, it won't appear next month.

[apn64]

Attention
Hidden charge from Bell if you do your local phone porting.

Bell has to get 30 days notice for cancellation, otherwise you will be charged by Bell 28.95$

This is complete bullcrap. IIRC, when the LNP rules were established one of the conditions was that you DID NOT have to contact your old provider and listen to their whiny retention staff try to get you to change your mind.

Hopefully, those on automated payments (bank or credit card) have the opportunity to cancel the transaction or at least fight an unauthorized charge after the fact.

[4public]

This is complete bullcrap. IIRC, when the LNP rules were established one of the conditions was that you DID NOT have to contact your old provider and listen to their whiny retention staff try to get you to change your mind.

Hopefully, those on automated payments (bank or credit card) have the opportunity to cancel the transaction or at least fight an unauthorized charge after the fact.

It is not the same. My phone was ported from Bell without any problem during 5 days.
Bell has their own service policy for their subscribers (they can change it without your agreement). That policy has description about "early termination"
I disagree with this charge, but we cannot ignore it.

[plausipo]

Are there anybody who find your PAP2T-NA too hot to be hold?
Got mine from ebay (China, US35, shipping took 27 days). Not hot. Mac address shown in case's label the same as read from router's display (through browser), translated to be Linksys-Cisco as post#385 suggested
:arrow:Temperature measured @ 42C (room temperature @23C)
1) subscribed to voip.ms
2) plug-in and set up ATA (using SIP, G729a) as voip.ms instructed for Montreal
3) paypal US25+tax (CAD31) - Got dial tone and made trial call successfully with a wired telephone (with slight echo)
4) ordered DID (0.99+setup), got a number, changed voip.ms "account setting/inbound setting" to ATA - successfully got incoming call.
:arrow:The whole process took about an hour from start to finish.
Happy with that!:)...thanks OP and all those contributed positively to this thread!

[paulo500]

Hi Paulo500, did you port your Primus number over to Voip.ms? If yes, how much did Primus charge you?
I'm planning to switch to Voip.ms but I don't know if it's worth to keep my existing number or not.
Ok just got my last bill from Primus. They are charging me:
$20.35 (TBB Termination Charge)
$9.95 (Hardware Handling Fee)

Thank goodness voip.ms is alot cheaper...I'll be recovering my losses in a few months :rolleyes:

[Mazda323]

Ok just got my last bill from Primus. They are charging me:
$20.35 (TBB Termination Charge)
$9.95 (Hardware Handling Fee)

Thank goodness voip.ms is alot cheaper...I'll be recovering my losses in a few months :rolleyes:So they still charge you $30. I checked my Primus terms and conditions. It doesn't mention these things. Have you returned them the rental ATA? What does the Hardware Handling Fee mean? We pay them the rental fee every month already. B-------------S!

[robinto]

Attention
Hidden charge from Bell if you do your local phone porting.

Bell has to get 30 days notice for cancellation, otherwise you will be charged by Bell 28.95$
I had month to month Bell plan and I was successfuly ported to VOIP for 25$
Now Bell is charging additional 28$ (I did not let them know about that 30 days before)

Do not do this mistake and let them know if you want to keep some money.

This is a link from Bell about early termination:
http://support.bell.ca/Documents/HomePhone/Legal/utos%28en%29.pdf

So how can we avoid this ?
Tell them we'll port the number in 30 days ?
If you tell them to cancel your landline service, will they also cancel your number ?

[paulo500]

So they still charge you $30. I checked my Primus terms and conditions. It doesn't mention these things. Have you returned them the rental ATA? What does the Hardware Handling Fee mean? We pay them the rental fee every month already. B-------------S!
I know.
But unfortunately their online General Terms & Conditions (http://www.primustel.ca/en/residential/legal/termsofuse.html) has a blurb about paying for the Termination Charge and the Handling fee. I think the main issue is that voip.ms is able to port your number in a week or two and cancel your loosing carrier at the same time. Primus says they need 30 days notice...:cry:
We can't win!

Edit: Yeah Primus sent me a letter with a pre-paid return letter to attach to my rental box. So i sent it off last week.

[Badder]

So how can we avoid this ?
Tell them we'll port the number in 30 days ?
If you tell them to cancel your landline service, will they also cancel your number ?

Exactly what happened to me as well. It's a lose-lose situation for the customer. Either you lose your number or you contribute to Bell's profit margin. :mad:

[skategoat]

Some newb questions here:

- can I tie in my cellphone somehow to my voip.ms account? That is, can I have the voip number forward to my cellphone and have it listed as one of my MY5 thus not having the calls use up my minutes?

- is the Linksys PAP2T the best adapter for hooking up regular phones? Where's the best place to get one in Canada?

- does voip.ms work for fax numbers? I see no mention of it on their website.

Thanks.

[RobberBarron]

Some newb questions here:

- can I tie in my cellphone somehow to my voip.ms account? That is, can I have the voip number forward to my cellphone and have it listed as one of my MY5 thus not having the calls use up my minutes?

- is the Linksys PAP2T the best adapter for hooking up regular phones? Where's the best place to get one in Canada?

- does voip.ms work for fax numbers? I see no mention of it on their website.

Thanks.

1). Absolutely. You can forward to any number, including your cell. However, you will incur the minutes/rate on your Voip.ms, unless you get the unlimited local calls option---otherwise, it's about 1 cent per minute.

2) PAP2 is one of the best and most reliable--it's also one of the most used so easier to get tech support, especially on how to configure. I suggest you buy through eBay (I got mine new for under $40), or buy from CanadaComputers (usually around $50-$55).

3) If you set the settings to use "premium" line quality, fax transmission should work. However, it doesn't seem to be very reliable if say you are forwarding to another voip based fax (eg e-Fax) because of the multiple interferances. However, within about 2 months, Voip.ms is expected to be releasing fax capabilty---- hopefully that will mean that faxes will go to email---but those details are not yet clear.

[skategoat]

Just to clarify #1, when voip.ms forwards to call to my cell, my cell provider recognises the call as coming from my voip number as opposed to the original caller?


1). Absolutely. You can forward to any number, including your cell. However, you will incur the minutes/rate on your Voip.ms, unless you get the unlimited local calls option---otherwise, it's about 1 cent per minute.

2) PAP2 is one of the best and most reliable--it's also one of the most used so easier to get tech support, especially on how to configure. I suggest you buy through eBay (I got mine new for under $40), or buy from CanadaComputers (usually around $50-$55).

3) If you set the settings to use "premium" line quality, fax transmission should work. However, it doesn't seem to be very reliable if say you are forwarding to another voip based fax (eg e-Fax) because of the multiple interferances. However, within about 2 months, Voip.ms is expected to be releasing fax capabilty---- hopefully that will mean that faxes will go to email---but those details are not yet clear.

[jackie999]

- is the Linksys PAP2T the best adapter for hooking up regular phones? Where's the best place to get one in Canada?

Be warned there are clone/counterfeit PAP2T's being sold. We discussed that here..just a few posts back.
I got my PAP2T-NA (be sure it's NA) from Canadacomputers (http://www.canadacomputers.com/advanced_search_result.php?keywords=PAP2T&sid=mburefim2qaki7apbgvpbbca83&x=0&y=0)

There were posts that dell.ca was selling (free shipping) although I couldn't find a link to that..is it still available?

[claudiu_corbea]

Just to clarify #1, when voip.ms forwards to call to my cell, my cell provider recognises the call as coming from my voip number as opposed to the original caller?

That's a very good question. With any regular provider you should normally see your DID number, but a smart feature would be to actually see the original caller ID; I think this is called CID pass-through. Personally I want CID pass-through and see the original CID.

[pipolchap]

Ok just got my last bill from Primus. They are charging me:
$20.35 (TBB Termination Charge)
$9.95 (Hardware Handling Fee)

Thank goodness voip.ms is alot cheaper...I'll be recovering my losses in a few months :rolleyes:

The $9.95 is the return shipping charge for hardware.


I know.
But unfortunately their online General Terms & Conditions (http://www.primustel.ca/en/residential/legal/termsofuse.html) has a blurb about paying for the Termination Charge and the Handling fee. I think the main issue is that voip.ms is able to port your number in a week or two and cancel your loosing carrier at the same time. Primus says they need 30 days notice...:cry:
We can't win!

Edit: Yeah Primus sent me a letter with a pre-paid return letter to attach to my rental box. So i sent it off last week.

Yup, I'm with Primus. I asked them about porting my number out and the CSR told me I need to give them 30 days. Once you give them notice, the porting process must occur between day 31 and day 60, otherwise you have to give them additional notice after 60 days. This second notice is an extension, thus you DO NOT have to wait another 30 days. If you follow these steps, you should avoid the $20 charge.

Hope this helps others.

[skategoat]

I'm just about to pull the trigger on this and there are no 416 or 905 numbers. I think I will get Rogers Home Phone and then port over later. Rogers has some 416 numbers available.

[jackie999]

I'm just about to pull the trigger on this and there are no 416 or 905 numbers. I think I will get Rogers Home Phone and then port over later. Rogers has some 416 numbers available.

There was a post about this earlier..someone mentioned if you keep checking back you may see some 416's or 905's become available. If you already have a 416 or 905 number..you can just port it over.

[robinto]

Be warned there are clone/counterfeit PAP2T's being sold. We discussed that here..just a few posts back.
I got my PAP2T-NA (be sure it's NA) from Canadacomputers (http://www.canadacomputers.com/advanced_search_result.php?keywords=PAP2T&sid=mburefim2qaki7apbgvpbbca83&x=0&y=0)

There were posts that dell.ca was selling (free shipping) although I couldn't find a link to that..is it still available?

http://miniurl.com/38924
$55.99 with free shipping

[Chaseme]

Don't know if it's been covered, but will my alarm to central be OK with voip.ms? I'm currently with Videotron and looking to switch ASAP!

Seems this is an ongoing discussion around the web. Anyone with experience here with alarm and voip?

[tariaki]

http://miniurl.com/38924
$55.99 with free shipping

Ordered. I will be switching from Vonage to voip.ms. I will provide update on the transition.

Thanks to OP for bringing this up and all the contributions form RFDers.

[SNES_Chalmers]

Thanks to the OP for this thread. I got setup with VOIP.MS after seeing how easy it is from all these posts. Took me no more than 1 hour to get setup with my DID and everything. Was going to go with Freephoneline.ca, but I like all the options that VOIP.MS provides and the pay as you go for long distance as I would never rack up the $13.95 a month with VOIP.MS that Freephonline charges for its world unlimited.

I really like the fact that it is so configurable with so many options and probably more coming. I've been told by a friend who's really into this stuff that they will be fully supporting fax...

[ginko123]

sorry all lil late into the game but just wanted to quickly ask: All I need to set this VOIP line at home would be:
PAP2NA either from dell or CC
Buy voip.ms .99 monthly canadian line

Is that it? Set it up per their instructions and will they invoice everymonth or take from my $25/credit that they initially ask?

thanks all.


p.s. I bought one of those pap2na from ebay (hongkong) and it only worked for like 4 months with VBUZZER and i simply got rid of entire thing 2 yrs back.

Using MagicJack rightnow for about a year.

[KingofMay]

Ordered. Now all my LD goes through voip.ms, and I am happy with the call quality.

Next step is to port my bell line.

Thanks, OP.

[Matrx]

sorry all lil late into the game but just wanted to quickly ask: All I need to set this VOIP line at home would be:
PAP2NA either from dell or CC
Buy voip.ms .99 monthly canadian line

Is that it? Set it up per their instructions and will they invoice everymonth or take from my $25/credit that they initially ask?

thanks all.


p.s. I bought one of those pap2na from ebay (hongkong) and it only worked for like 4 months with VBUZZER and i simply got rid of entire thing 2 yrs back.

Using MagicJack rightnow for about a year.

Yup. They'll deduct your monthly usage costs from the balance in your account.

[AAA]

If I want to create an account to take a tour, is it ok to enter fake info or is it a no-no?

[luckydog2006]

Don't know if it's been covered, but will my alarm to central be OK with voip.ms? I'm currently with Videotron and looking to switch ASAP!

Seems this is an ongoing discussion around the web. Anyone with experience here with alarm and voip?

Not recommended as its alarm company dependent. Some might work with your voip provider choice, some might not. I would stick to using voip for your LD calls since you might need to keep Videotron for your local home phone calls.

Another way is to keep your Videotron until you know the alarm to central is working fine on voip before you cancel your Videotron account.

[robinto]

sorry all lil late into the game but just wanted to quickly ask: All I need to set this VOIP line at home would be:
PAP2NA either from dell or CC
Buy voip.ms .99 monthly canadian line

Is that it? Set it up per their instructions and will they invoice everymonth or take from my $25/credit that they initially ask?

thanks all.


p.s. I bought one of those pap2na from ebay (hongkong) and it only worked for like 4 months with VBUZZER and i simply got rid of entire thing 2 yrs back.

Using MagicJack rightnow for about a year.

To be more precise:
Per minute (Inbound):
$0.99 per month / $0.0100 per minute
Setup Fee: $0.50
Billing Increment: 6 seconds

[AAA]

Here is my experience..from off the top of my head.

Sign up with voip.ms - paypal'd them $25 to start
Buy the ATA (PAP2T canadacomputers $54)
Read the configuration guide on voip.ms site to get the ATA working - it was no more difficult than setting up a router. If you want the ATA to jump thru hoops, then keep reading, it's very powerful. Otherwise, the basic configuration will get you going.
Plug your phone into the ATA. Make a few calls cause you're done.
If you want to receive calls, you can buy a number (DID) or you can *port* your existing phone number over (took about a week for me) ..cost $25..then $.99 cents a month for the DID.

You get all the bells and whistles. Call display..forward the Mother in Laws calls to "this line is no longer in service"...there are so many features, most of which I don't know what they do..so I keep reading :)

The ATA/modem must always be on..the computer does NOT need to be on.
Costs for calls out about 1/2 a cent a minute.
Costs for calls in about 1 cent a minute.
e911 service $1.50 a month - this is optional.
I also have to pay for a 'dry loop' for my teksavvy DSL - this, for me, is $10.22 a month.
All in all, I'm way ahead.
But I won't factor in the new panasonic phones I bought, or the APC backup :)

Hope this helps!


So the calls in/out 1c and 1/2c are local call rates? Why incoming call is more expensive? We pay for calls made by marketers. :mad: I looked at the rate table and not quite sure I understand it. Is the rate for outgoing call? If that's the case, does it mean calls in rate is always the same, 1c regardless of local or International?

You paid over $10 for a dry loop + voip, is it worth it? I heard you have to pay like $40 setup for dry loop fee as well. I'm still debating if I should switch to voip or stay in land line.

[RMSimard]

So the calls in/out 1c and 1/2c are local call rates? Why incoming call is more expensive? We pay for calls made by marketers. :mad: I looked at the rate table and not quite sure I understand it. Is the rate for outgoing call? If that's the case, does it mean calls in rate is always the same, 1c regardless of local or International?

Yup, flat rate incoming no matter where the caller is from; the rate table is for outgoing calls. And outgoing "local call rates" of 0.5c to 1.5c are for all of North America, from your local pizzeria all the way to the Playboy mansion. ;)

If you don't like telemarketers, join the national do not call list (https://www.lnnte-dncl.gc.ca). It's free and valid for 5 years (renewable).

[ToChicka]

To be more precise:
Per minute (Inbound):
$0.99 per month / $0.0100 per minute
Setup Fee: $0.50
Billing Increment: 6 seconds

So basically .99 for DID, e911 is $1.50/month optional, outgoing is 1/2 cent, incoming 1cent anywhere NA>
Did I get it right? :)

Main cost would be the ATA + $25 initial credit I assume. Not bad also considering if you can use the SIP details on your FRING application while travelling in US or abroad?

[Chaseme]

Not recommended as its alarm company dependent. Some might work with your voip provider choice, some might not. I would stick to using voip for your LD calls since you might need to keep Videotron for your local home phone calls.

Another way is to keep your Videotron until you know the alarm to central is working fine on voip before you cancel your Videotron account.

I've found out from the installer of my alarm system that he uses only Vonage with no cell backup for his own DSC based system. So he's telling me there should be no issue. I'll let you guys know how it works out!

[death_hawk]

That's a very good question. With any regular provider you should normally see your DID number, but a smart feature would be to actually see the original caller ID; I think this is called CID pass-through. Personally I want CID pass-through and see the original CID.
CID passes through.

The only place CID is broken is with Toll Free DID.


So the calls in/out 1c and 1/2c are local call rates? Why incoming call is more expensive? We pay for calls made by marketers. :mad: I looked at the rate table and not quite sure I understand it. Is the rate for outgoing call? If that's the case, does it mean calls in rate is always the same, 1c regardless of local or International?

You paid over $10 for a dry loop + voip, is it worth it? I heard you have to pay like $40 setup for dry loop fee as well. I'm still debating if I should switch to voip or stay in land line.
If you don't like the rate structure, you're free to go with any other company.
Besides... if you do receive an obscene amount of incoming calls, there's always the 3500 minute incoming option.

I have dry + voip and I wouldn't look back.



If you don't like telemarketers, join the national do not call list (https://www.lnnte-dncl.gc.ca). It's free and valid for 5 years (renewable).
DNC list is a joke. No telemarketer will call you from Canada, but that doesn't prevent the list from being sold to foreign countries. Nor does it stop non telemarketer calls (IE Surveys).

Besides... with a PBX you can have some fun with these people.


So basically .99 for DID, e911 is $1.50/month optional, outgoing is 1/2 cent, incoming 1cent anywhere NA>
Did I get it right? :)

Main cost would be the ATA + $25 initial credit I assume. Not bad also considering if you can use the SIP details on your FRING application while travelling in US or abroad?

Yup. Except outgoing is 1.5 cents (1 cent if you're premium) and incoming I think is 1.5cents (premium).

If you just want to "test drive" it, you could soft phone it. If you're happy with it, then an ATA is the only other hard cost.
The $25 credit I don't really count as a "cost" as it's refundable minus a couple bucks for usage.

[lpsaad]

NOOB question: If I get a Linksys PAP2 device and regular home phones, I wont be needing a softphone on my computer right??:confused:

[techfox]

NOOB question: If I get a Linksys PAP2 device and regular home phones, I wont be needing a softphone on my computer right??:confused:

Yes, this is correct.. (as long as it's an unlocked PAP2).

[RMSimard]

Main cost would be the ATA + $25 initial credit I assume. Not bad also considering if you can use the SIP details on your FRING application while travelling in US or abroad?

Yup. Fring 4.1 now supports DTMF after placing a call, so navigating an IVR or your voicemail is finally possible. Still waiting for an Apple app update for that though.


NOOB question: If I get a Linksys PAP2 device and regular home phones, I wont be needing a softphone on my computer right??:confused:

Right. With an ATA and a regular phone you do not even need a computer, just high-speed internet access. I still use a softphone on my desktop PC to act as a caller-id/visual notification popup.


DNC list is a joke. No telemarketer will call you from Canada, but that doesn't prevent the list from being sold to foreign countries. Nor does it stop non telemarketer calls (IE Surveys).

Besides... with a PBX you can have some fun with these people.


Well, YMMV but since joining the do not call list the amount of telemarketers have dropped to just one every few months. And then I simply politely asked these remainders to remove me from their list. As for surveys, for having worked in that field for 6 months I believe they're a good thing. Your answers actually do influence policy making and TV/radio content. As long as these are real surveys; disguised telemarketing usually begins with "hello, how are you?" ;)

[nkwu]

really easy - except my calls are dropping after the call is picked up on the other hand

there's just enough time to say "hello" and then the call gets cut off...any one else experience this?

[tdott]

I read the first 7 pages. Sorry if this was answered after that.

Can anyone confirm this works with a Home Alarm system? Mine is ADT if that matters.

Thanks.

[jackie999]

really easy - except my calls are dropping after the call is picked up on the other hand

there's just enough time to say "hello" and then the call gets cut off...any one else experience this?

Mine doesn't drop calls...but I followed the tips in this blog to set up my PAP2T ..maybe it would help you?
http://www.toao.net/25-linksys-pap2t-voip-adapter-review

[shepd]

I read the first 7 pages. Sorry if this was answered after that.

Can anyone confirm this works with a Home Alarm system? Mine is ADT if that matters.

Thanks.

No, it probably won't.

You should test out the response time on your alarm company, though. I had ADT for my store and for the two mistake alarms that happened while I wasn't there, I arrived at the store *first* and had to wait a half hour (total response time 45 minutes) for the cops.

We dumped ADT after that and didn't get another alarm company because, after speaking with the cops, alarms are only serviced if someone happens to be nearby. The cops said I was lucky they got there in 45 minutes! Otherwise the cops basically seem wait for the siren to turn off (which is what happened both times for us, after 15 minutes it turns itself off). Yay tax dollars at work. And they want to charge for false alarms... *sigh*

We left the system in, though, since the siren is the only thing that could put any sort of fear into a robber, anyways. Although if they knew that I knew about the response time...

[a2vr6]

I think everyone knows of the response time to home alarms. Waste of money IMO since they will be long gone by then. Not sure how well it will work when there is a fire if the response is the same.

[Mazda323]

I made a call by Voip.ms DID, but the people that I called told me that he only saw my DID number without displaying my name. I did put my name in my PAP2T settings. Can anybody help me fix it?
Thanks in advance.

[Kwirky]

I made a call by Voip.ms DID, but the people that I called told me that he only saw my DID number without displaying my name. I did put my name in my PAP2T settings. Can anybody help me fix it?
Thanks in advance.

Caller name is not transmitted by the originating carrier - only the number is. The name displayed on the handset comes from a CNAM database lookup done by the callee's service provider, and as far as I know voip numbers are not in this database.

[BargainBuff]

I have 2 DIDs with VOIP.ms under 1 account, but I can't seem to get my PAP2 to configure correctly with 2 Lines. Here's what I'm trying to do: have 1 DID ring my phone; the other DID to use with my fax machine.

Whenever I enable both lines in the PAPS2, only the second DID picks up (i.e. it goes straight to fax, regardless of whatever DID I dial!)----that is, if I call either DID, the fax machine picks up. When I disable the 2nd DID, the original (1st DID) rings my phone, as I'd like. But, I'd like BOTH lines working independently!

Do I need to have each DID under a separate Voip.Ms account (so as to have a unique username/password combo for each!) or can both be under the same account? After all, I suspect the problem may be due to only having 1 account since the settings for both Lines are the exact same (e.g. username/password)-----am I missing something here?

Any help would be much appreciated it.

[4public]

So how can we avoid this ?
Tell them we'll port the number in 30 days ?
If you tell them to cancel your landline service, will they also cancel your number ?

The best way is to port the phone without notice - you will get a month of cheap VOIP service. 30 days notice means for you an additional expensive month of Bell service. In both cases you will loose some money. But you will loose with bell 30 days notice the same or more.
Decide what is better

[fuzzy]

You may need to create a "sub account" and associate one of your DIDs to this "sub account". So, this DID will have its own login name and password.


I have 2 DIDs with VOIP.ms under 1 account, but I can't seem to get my PAP2 to configure correctly with 2 Lines. Here's what I'm trying to do: have 1 DID ring my phone; the other DID to use with my fax machine.

Whenever I enable both lines in the PAPS2, only the second DID picks up (i.e. it goes straight to fax, regardless of whatever DID I dial!)----that is, if I call either DID, the fax machine picks up. When I disable the 2nd DID, the original (1st DID) rings my phone, as I'd like. But, I'd like BOTH lines working independently!

Do I need to have each DID under a separate Voip.Ms account (so as to have a unique username/password combo for each!) or can both be under the same account? After all, I suspect the problem may be due to only having 1 account since the settings for both Lines are the exact same (e.g. username/password)-----am I missing something here?

Any help would be much appreciated it.

[robinto]

1. Do you need to notify Bell (confirm that they allow to port out) before porting ?
2. Do you have to arrange Dry Loop service in advance, if you are using DSL ?
3. How long is the interruption time during the transition ?
4. Is ident-a-call number also portable ?

Thanks in advance.

[Sneaks]

I can comment on my experience YMMV.


1. Do you need to notify Bell (confirm that they allow to port out) before porting ?
I didn't, no problems, no extra charges, etc. May be worth a call though.



2. Do you have to arrange Dry Loop service in advance, if you are using DSL ?
My understanding is you are supposed to do that, but again, I didn't but because I was moving I had a time crunch and wanted to ensure that I didn't lose my number (even if meant my DSL got interrupted.) The port can only complete if the number is active and you shouldn't perform any other cancellation/change of service before it completes.



3. How long is the interruption time during the transition ?
I had no interruption personally, only interruption was that it happened faster than I expected, and I hadn't setup my PAP2T yet. :)



4. Is ident-a-call number also portable ? can't comment, not familiar with this.


Hope that helps, good luck!

[destinet]

Hi All,

There is my final bill after porting from vonage to voip.ms
After one month of normal use.

Outgoing (including 1800# which are free): 4h15min56sec for a total of: 0.901$
Incoming: 2h44min43sec for a total of: 1.679$
DID: 0.99$

Total for 1 month: 3.57$

I need to do the "amortization" of the Adapter (50$) + the transfert of my did (25$)
Total amortization: 75$
Total by month before with vonage: 23$
So i less then 4 month i will make money in my pocket;)


Thanks OP!!!

[djino]

Hi All,

There is my final bill after porting from vonage to voip.ms
After one month of normal use.

Outgoing (including 1800# which are free): 4h15min56sec for a total of: 0.901$
Incoming: 2h44min43sec for a total of: 1.679$
DID: 0.99$

Total for 1 month: 3.57$

I need to do the "amortization" of the Adapter (50$) + the transfert of my did (25$)
Total amortization: 75$
Total by month before with vonage: 23$
So i less then 4 month i will make money in my pocket;)


Thanks OP!!!

Did you not add e911 to your services? :confused:

djino

[robinto]

Hope that helps, good luck!

Thank you for your feedback.

I called Bell and voip.ms and got the answers, YMMV:

1. Do you need to notify Bell (confirm that they allow to port out) before porting ?
No. You can just tell your new provider to contact Bell and give them 30 days notice before porting, otherwise you will be charged for the whole month.

2. Do you have to arrange Dry Loop service in advance, if you are using DSL ?
Yes. The original service (Bell) will be terminated immediately after porting.

3. How long is the interruption time during the transition ?
From a few hours to the whole day.

4. Is ident-a-call number also portable ?
No. Only the main number on you bill is portable.

[BargainBuff]

You may need to create a "sub account" and associate one of your DIDs to this "sub account". So, this DID will have its own login name and password.

Many thanks!

[destinet]

Did you not add e911 to your services? :confused:

djino

Nope i configure the 911 in my dialing plan to dial my local police. And if i need 911 i will use my cellphone.

[kosodo]

I live in Mississauga however I don't see Missisauga number's listed on voip.ms site. Would it work if I choose a Toronto number? Thanks!

[death_hawk]

That's up to you really. You can get a phone number anywhere in the world and have it "work".

In your case, if you don't know anyone in Toronto (or it's Long Distance for Mississauga to call Toronto) that number will most likely be useless for you.

I personally have numbers in my home town, Calgary and Vancouver.

[dP21]

I currently subscribe to Bell Entertainment Services IPTV (with no phone service). Dell provided a dry loop when I signed up for the service. If I want to sign up for the voip.ms service, do I need to :


Port over my dry loop number
Or Create a new DID number with voip.ws (and leave the dry loop)


I'd prefer to get a 416 and my dry loop is 416 right now.

[althoma1]

dP21 Bell Entertainment Services IPTV

--------------------------------------------------------------------------------
I currently subscribe to Bell Entertainment Services IPTV (with no phone service). Dell provided a dry loop when I signed up for the service. If I want to sign up for the voip.ms service, do I need to :

Port over my dry loop number
Or Create a new DID number with voip.ws (and leave the dry loop)

I'd prefer to get a 416 and my dry loop is 416 right now.

You'll need a new number and to keep the dry loop if you're staying with BES. You'll still need a dry loop for as long as you have any kind of internet service that uses a wireline - if you switch to cable in the future then you won't need a dry loop anymore, but I still don't believe you can port a dry loop number.

[death_hawk]

For those who don't have an ATA, H3LL has one on the Days of Deals that's not bad:

http://accessories.dell.com/sna/products/VoIP_Telephony/productdetail.aspx?c=ca&l=en&s=bsd&cs=cabsdt1&sku=A0947981

It's dead basic and EOL, but not a bad price for what you get.
For those advanced users, it's missing a few features but for a simple ATA it's damn near perfect.

I'm personally skipping it as I want PoE.
Here's a comparison to a few SPA ATAs:
http://www.voip-info.org/wiki/view/SPA-941

[SNES_Chalmers]

I read the first 7 pages. Sorry if this was answered after that.

Can anyone confirm this works with a Home Alarm system? Mine is ADT if that matters.

Thanks.

It does work with a home alarm system. I have Reliance Protectron and it works fine. My alarm system was installed after I had voip.ms setup, so the alarm guy tested it and everything to make sure it worked fine.

[Longobongo]

I live in Mississauga however I don't see Missisauga number's listed on voip.ms site. Would it work if I choose a Toronto number? Thanks!

I would like to know this too. Wanna make the jump to voip.

[death_hawk]

I would like to know this too. Wanna make the jump to voip.

Read my post directly below the one you quoted.

[djino]

For those who don't have an ATA, H3LL has one on the Days of Deals that's not bad:

http://accessories.dell.com/sna/products/VoIP_Telephony/productdetail.aspx?c=ca&l=en&s=bsd&cs=cabsdt1&sku=A0947981

It's dead basic and EOL, but not a bad price for what you get.
For those advanced users, it's missing a few features but for a simple ATA it's damn near perfect.

I'm personally skipping it as I want PoE.
Here's a comparison to a few SPA ATAs:
http://www.voip-info.org/wiki/view/SPA-941

I read the tech specs and I thought I read that it has PoE?

Question: What about this makes it 4 line. Whats the use of 4 Lines on 1 phone?

djino

[death_hawk]

SPA942 has PoE

And upon further thought, I'm gonna skip PoE and order a couple of these.
PoE needs an injector/switch and are $30-40 more than these.
I can't seem to find a decent non-consumer switch to power these. I wouldn't trust a consumer switch.

Assuming the 4 line is like any other 4 line phone, you can link up 4x SIP accounts and switch between them. It's useless for most people, but for a business for example, it'd be handy.

[tdott]

It does work with a home alarm system. I have Reliance Protectron and it works fine. My alarm system was installed after I had voip.ms setup, so the alarm guy tested it and everything to make sure it worked fine.

Thanks! That is what i thought, because i currently have Rogers Home phone and that is basically the same as having the Linysys convert to anolog (What the alarm system uses).

I will set it up and try it out for the first month, worst case i won't lose much, since i won't be porting any numbers.

[FAQ]

Can you configuer an ATA as well as a SIP phone with the same number ? they are on different network (shaw and telus ) , is it possible to set up one voip.ms number with both devices ? ( i.e. both phone ring when the number gets call ).

thanks,

[RMSimard]

Can you configuer an ATA as well as a SIP phone with the same number ? they are on different network (shaw and telus ) , is it possible to set up one voip.ms number with both devices ? ( i.e. both phone ring when the number gets call ).

thanks,

Sure! Just assign one of the devices to your main account and create a sub-account for the second one. Then assign both the main account and the sub-account to a ring group. Finally assign the DID to the ring group. Et voilà! :)

[Lulz]

what did you guys put for Service Provider Info for Bell during port steps?


Customer Account #
(Your account with your current provider)

*
Service Provider Name

*
Service Provider Address

*
Service provider address / Continued
(Optional, e.g: suite 343)


City

*
State or Province

*
Country

*
ZIP or Postal Code

[ryehigh17]

I used the BELL address shown on my monthly statement. No problems for me.


what did you guys put for Service Provider Info for Bell during port steps?


Customer Account #
(Your account with your current provider)

*
Service Provider Name

*
Service Provider Address

*
Service provider address / Continued
(Optional, e.g: suite 343)


City

*
State or Province

*
Country

*
ZIP or Postal Code

[bahbuu]

Hi,

I have a random/newbie question and hope you guys can provide me with some insight!!

I reside in Richmond Hill, and I want to get a DID (haven't had a home phone ever, so don't have a number to port)

However, on voip.ms, they only offer (at the moment anyway) 289 area codes.

However, in other regions (Also part of Greater Toronto) i.e. Markham/Thornhill/Toronto, they provide some 905 and 647 area codes.

If I use those, can I still use the number when I reside in Richmond Hill? And is there no long distance charges? (My cell phone starts with 647, but I can still use it in Richmond Hill with no long-d charges)

Thanks!

[donvaz]

Hi,

I have a random/newbie question and hope you guys can provide me with some insight!!

I reside in Richmond Hill, and I want to get a DID (haven't had a home phone ever, so don't have a number to port)

However, on voip.ms, they only offer (at the moment anyway) 289 area codes.

However, in other regions (Also part of Greater Toronto) i.e. Markham/Thornhill/Toronto, they provide some 905 and 647 area codes.

If I use those, can I still use the number when I reside in Richmond Hill? And is there no long distance charges? (My cell phone starts with 647, but I can still use it in Richmond Hill with no long-d charges)

Thanks!

Yes you should not have to pay long distance charges. I am in Vancouver and have had a 416 VOIP number for the past 6 years. Have never paid a cent in LD charges to any 416 number.

[Jeffers]

I have been using voip.ms for 1.5 months now and really like it so far. My monthly cost is reduced by 90%. I have been using a temporary DID (I ordered one that I don't plan on keeping) to test the service and am ready now to port my Bell number over to voip.ms. Before I port the number, I was wondering if anyone here knows whether I can port my number back to Bell or another provider at a later date if I choose to. Thanks.

[shadow_cruiser]

Hey guys, I've learned quite a bit about voip just by reading this thread. I want to port my cell number to my voip account, can this be done?. Once that is done, will Rogers let me transfer my plan to another individual without a number?

[death_hawk]

I was wondering if anyone here knows whether I can port my number back to Bell or another provider at a later date if I choose to. Thanks.
Yes.
At least I'm pretty sure you can. I honestly can't see why you wouldn't be able to.




If I use those, can I still use the number when I reside in Richmond Hill? And is there no long distance charges? (My cell phone starts with 647, but I can still use it in Richmond Hill with no long-d charges)

Thanks!
Assuming that other carriers recognize that X is still a local call to Y, you won't incur any LD.


Hey guys, I've learned quite a bit about voip just by reading this thread. I want to port my cell number to my voip account, can this be done?. Once that is done, will Rogers let me transfer my plan to another individual without a number?

AFAIK, once you port a number out, that account ceases to exist.
If you want to retain the plan, I would talk to someone at CS and see if they can duplicate the plan onto another (temporary) number while you port your number out.

If you port without contacting the outgoing carrier, the plan attached to the number will just disappear.

[RMSimard]

Before I port the number, I was wondering if anyone here knows whether I can port my number back to Bell or another provider at a later date if I choose to. Thanks.

death_hawk is right, you can definitely port your number in and out. That was my concern when I signed-up a while ago and VoIP.ms reassured me that I could port out my number back to Bell or any other carrier any time I want. I really like such freedom and no-hassle attitudes; definitely don't feel like leaving VoIP.ms anytime soon. :)

[Jeffers]

death_hawk is right, you can definitely port your number in and out. That was my concern when I signed-up a while ago and VoIP.ms reassured me that I could port out my number back to Bell or any other carrier any time I want. I really like such freedom and no-hassle attitudes; definitely don't feel like leaving VoIP.ms anytime soon. :)

Thanks guys. I'm going to go for it.

[andrewhaji]

I recently got myself an account with these guys. I purchased a new DID in the same area code as my landline, and I forwarded my landline to the new DID. That has allowed me to test out the VoIP connection as though it were actually my landline--but without paying the $25 fee prematurely.

In a few weeks, if all continues to work well, I'll port my existing landline over.

[duckdown]

Hi all, quick question, I don't suppose it's possible to use any SMS related functions with this service? IE: send text messages or receive text messages from friends with cellphones?

I plan on signing up tomorrow.

Also, is the PAP2 the best choice? I don't mind spending a bit more for some of these fancy phones on the CC page, if they're worth it:

http://www.canadacomputers.com/index.php?cPath=30_414&sid=oiufsfdo3ddga8i1mbp0plirs0


Thanks!

[duckdown]

Oh sorry, and how long does it take to become activated? If I sign up and go to CC to buy the equipment then come home, will it be ready to rock?

Thanks again

[andrewhaji]

The PAP2T is the best choice if you have any existing phones around the house. It's what I bought.

If you're activating a new DID (phone number), it's instantly activated. Deposit money using PayPal, order the DID, and you're done. It takes maybe 10 minutes to set up everything on the PAP2T.

[RMSimard]

Hi all, quick question, I don't suppose it's possible to use any SMS related functions with this service? IE: send text messages or receive text messages from friends with cellphones?

Currently no SMS passthru. Do ask them for it though: I'm hopeful that if we're enough asking for this they will implement it! :idea:


Oh sorry, and how long does it take to become activated? If I sign up and go to CC to buy the equipment then come home, will it be ready to rock?

Once your account is open and you put money in, it only takes a minute for the DID number to start resolving to your ATA/Softphone. So open an account now (free), add funds and just use the free X-Lite softphone to get you started, then buy your ATA or IP Phone later.

[Worldy]

I bought a Sipura SPA 2002 when I used vbuzzer a few year ago, is it compatible with voip.ms DID?

Thanks.

[duckdown]

Thanks so much, a person on IRC just reccomended signing up right away and using XLITE until I get my PAP2 tomorrow

About to pull the trigger, just to clarify one more time

My own DID is 99 cents a month, and its just 1c/minute to people within the GTA?? How is this so cheap? Am I missing some kind of fee or something?

25 minimum deposit is no problem but am I missing a part of the monthly equation here?

thanks again for the responses!

[RMSimard]

I bought a Sipura SPA 2002 when I used vbuzzer a few year ago, is it compatible with voip.ms DID?

Thanks.

Yes. Any unlocked SIP or AIX device is compatible with VoIP.ms. I've used SPA-1001 and SPA-3102: so far, so good.

[RMSimard]

My own DID is 99 cents a month, and its just 1c/minute to people within the GTA?? How is this so cheap? Am I missing some kind of fee or something?

It's a "pay for what you use" service. $0.99 gets you a phone number for people to reach you. About $0.015 a minute lets you call anywhere in USA/Canada, be it local or traditional long distance. $0.01/minute to receive calls from anywhere too. It's cheap because it uses the net's infrastructure and low-overhead from VoIP.ms, likely running on well-tuned open source software. No huge staff, no legacy hardware to maintain. And they throw all those "star services" in for free too. :) And yes, it is a business-quality voice service you're getting: once hooked you won't ever go back to those overpriced Bell/Telus dinosaurs.

You can add 911 service if you want for a fee. Or can upgrade to $5/month per DID for free incoming (up to x hours of talk time, above 50h I think.)

[duckdown]

Do they ever send anything important in snail mail? Like invoices, account usage, or other important stuff?


If not, technically couldn't I just sign up with an alias and fake address and pay via PayPal business account as well to have a bit of anonymity?

Thanks again!

[duckdown]

Hey all, ok I signed up, and bought the PAP2 today.

I can pickup my cordless phone and dial out with the VOIP account but when I call my voip.ms 289 phone number, it just hangs up once I call it. My phone never rings like a regular house phone, it just straight up hangs up on me when I call it

What am I doing wrong?

Thanks all

[andrewhaji]

I can pickup my cordless phone and dial out with the VOIP account but when I call my voip.ms 289 phone number, it just hangs up once I call it. My phone never rings like a regular house phone, it just straight up hangs up on me when I call it

In the voip.ms control panel, does your account show as "registered" at the bottom?

You might have issues with NAT, or maybe you didn't point your DID to the main SIP account?

[fuzzy]

Hey all, ok I signed up, and bought the PAP2 today.

I can pickup my cordless phone and dial out with the VOIP account but when I call my voip.ms 289 phone number, it just hangs up once I call it. My phone never rings like a regular house phone, it just straight up hangs up on me when I call it

What am I doing wrong?

Thanks all

Under "account setting"-->"inbound setting", you need to pick "ATA device" then apply.

[duckdown]

In the voip.ms control panel, does your account show as "registered" at the bottom?

You might have issues with NAT, or maybe you didn't point your DID to the main SIP account?

Says no registration found

I followed the Linksys example configuration instructions for the PAP2, still doesn't want to work. I enabled the 2 NAT options it said to enable. I think it is having trouble staying connected to the server, probably why it takes a long time when I try to dial out, theres like a 5-10 second delay before it starts ringing..

I wonder what the hell is wrong

[duckdown]

Under "account setting"-->"inbound setting", you need to pick "ATA device" then apply.

I've already done that in the very beginning :(

thanks

[Phils]

Says no registration found

I followed the Linksys example configuration instructions for the PAP2, still doesn't want to work. I enabled the 2 NAT options it said to enable. I think it is having trouble staying connected to the server, probably why it takes a long time when I try to dial out, theres like a 5-10 second delay before it starts ringing..

I wonder what the hell is wrong

I suggest you get into a tech support chat session with them. They should be able to help you.

It's their support among other things that sets them apart.

[RMSimard]

I think it is having trouble staying connected to the server, probably why it takes a long time when I try to dial out, theres like a 5-10 second delay before it starts ringing..

I wonder what the hell is wrong

For that, it's your ATA that is waiting on more numbers or a dial termination signal/pattern. Your ATA dial plan handles the dialing step, you'll need to read up on that and potentially modify it through your ATA's web interface. Or simply press # after dialing to tell the ATA to dial now, it's simpler.

[paulo500]

Just an fyi to those still thinking of switching to voip.ms.

I just checked my account and since I joined April 25th to now, i've used up just over $5 out of my account. (Not including the one-time porting fee and my ATA purchase obviously).
So before with Primus TBB voip, I was paying $30/month for their unlimited option. Now, with Voip.ms, im averaging about $4/month.

We don't use the phone a lot every day, as we are just a family of two. The wife also calls England maybe once a week. But compared to what we were paying..HUGE savings!

Thank You OP! :-0

[bahbuu]

Hey all, ok I signed up, and bought the PAP2 today.

I can pickup my cordless phone and dial out with the VOIP account but when I call my voip.ms 289 phone number, it just hangs up once I call it. My phone never rings like a regular house phone, it just straight up hangs up on me when I call it

What am I doing wrong?

Thanks all

I'm experiencing the exaaaact same problem as you.

- I can call out, but can't call in.

Let me know if you find a solution :)

[dorje]

i'm thinking of canceling rogers home phone, currently around $40 and get this. is it easy to set up, i'm thinking using pap2 ATA, do i need to buy SIP settings like from freephoneline.ca, will be glad if someone can help me out.

[bahbuu]

i'm thinking of canceling rogers home phone, currently around $40 and get this. is it easy to set up, i'm thinking using pap2 ATA, do i need to buy SIP settings like from freephoneline.ca, will be glad if someone can help me out.

you don't need to buy it from freephoneline.ca

I bought mine through craigslist

I'm sure you can buy it off ebay / canada computers / dell.ca / etc. etc.

And it's pretty easy to set up, although I haven't gotten it fully functional (I can call people, but people can't call me yet), but I only started setting it around 20 minutes ago (it took 5 - 10 minutes)

[Bskll]

I've already done that in the very beginning :(

thanks


Says no registration found

I followed the Linksys example configuration instructions for the PAP2, still doesn't want to work. I enabled the 2 NAT options it said to enable. I think it is having trouble staying connected to the server, probably why it takes a long time when I try to dial out, theres like a 5-10 second delay before it starts ringing..

I wonder what the hell is wrong


Under "account setting"-->"inbound setting", you need to pick "ATA device" then apply.


run through the recommendations here, and see if it helps:

http://www.toao.net/25-linksys-pap2t-voip-adapter-review

[bahbuu]

Update:

I didn't do any changes to my account settings or whatever, but everything now works (incoming + outgoing)

I realized that the problem was that the account state was not "registered" yet.

But now it reads as "registered" state, and now everything works properly :)

So I guess it was to due with voip.ms's side of things :)

[djino]

Just an fyi to those still thinking of switching to voip.ms.

I just checked my account and since I joined April 25th to now, i've used up just over $5 out of my account. (Not including the one-time porting fee and my ATA purchase obviously).
So before with Primus TBB voip, I was paying $30/month for their unlimited option. Now, with Voip.ms, im averaging about $4/month.

We don't use the phone a lot every day, as we are just a family of two. The wife also calls England maybe once a week. But compared to what we were paying..HUGE savings!

Thank You OP! :-0

Did you setup to allow e911? Or do you have an alternate way (cellphone) for emergencies? As this would cost $1.95/month alone.

djino

[djino]

I'm experiencing the exaaaact same problem as you.

- I can call out, but can't call in.

Let me know if you find a solution :)

Check your Portal Home settings within voip.ms, it should be showing as Registered (In Green text). If not, then thats the reason why.

djino

[jackie999]

Says no registration found

I realized that the problem was that the account state was not "registered" yet.

But now it reads as "registered" state, and now everything works properly

A few posts back (in this thread) around post #400 there is a suggestion that may help, i.e. using the IP instead of the name for the server. I'm using IP now and no problems with registration. HTH's

[jackie999]

Did you setup to allow e911? Or do you have an alternate way (cellphone) for emergencies? As this would cost $1.95/month alone.

djino

I don't have the e911 service enabled ..but I think the cost is $1.50 per month..plus taxes...

[lpsaad]

Where can I get a real inexpensive Linksys pap2t?? PM me. Thanks!

[jackie999]

Where can I get a real inexpensive Linksys pap2t?? PM me. Thanks!

I picked mine up at CC
http://www.canadacomputers.com/product_info.php?cPath=30_414&item_id=014833
Be warned there are clone/counterfeit ones out there that may or may not work...

[Rekognize]

Thanks OP, ditching Rogers Homephone for Voip.MS

:)

[paulo500]

Did you setup to allow e911? Or do you have an alternate way (cellphone) for emergencies? As this would cost $1.95/month alone.Nope, no e911 at this time.

[duckdown]

I'm really, really missing the lack of SMS functionality. More than I would have thought. I'm still racking up money on my Bell mobility account because many of my friends prefer to use SMS at times than chat on the phone (ie: if its late at night and hard to talk, or in a movie theatre, or so on)

Are there other VOIP providers that provide SMS functionality? I sent voip.ms a ticket regarding it but they say they have no plans and no ETA on when any SMS functions will work at all, which isn't that promising

THanks for the advice regarding the PAP2, I am going to try changing to an IP address rather than using 'toronto.voip.ms'

[FlintBlade]

So whats the difference between the $1/mo fee and the $5/mo fee they have for DID's?

[shadow_cruiser]

Can someone point me to a setup guide for voip.ms using the PAP2T?

EDIT: Nevermind, it's under the configuration samples on the voip.ms page. This thing is so simple to setup!. Thanks to the OP and the rest in the thread to introducing me to this, time to ditch Bell and Rogers.

[MXMike]

Can someone point me to a setup guide for voip.ms using the PAP2T?

These have probably been posted before, but they helped me out:
http://www.toao.net/25-linksys-pap2t-voip-adapter-review
http://www.consumedconsumer.org/2010/04/voip-from-scratch-iv-setup-and.html

[duckdown]

So whats the difference between the $1/mo fee and the $5/mo fee they have for DID's?

supposedly it's for "premium routing" among other things which should help with cutting in and out during conversations, a problem I have already had using the value package...

On that note, does anyone know why sometimes during conversations the voice will cut out for a second during talk? I've had to repeat myself sometimes because people say I'm cutting out..

[bahbuu]

Can someone point me to a setup guide for voip.ms using the PAP2T?

Also when you're registered, click on their support link --> Configuration Samples --> LinkSys / Sipura Adapters

I followed the instructions from there (it's very simple, clear, and straightforward)

And everything works perfectly now :) (setup only took 10 minutes)

[djino]

So whats the difference between the $1/mo fee and the $5/mo fee they have for DID's?

$0.99/month plus you pay $0.01/minute on incoming.. Or you can pay $5/month and have unlimited incoming (up to 3500 minutes).

djino

[jackie999]

So whats the difference between the $1/mo fee and the $5/mo fee they have for DID's?

This is your DID fee (calls you receive)
You have 2 choices
- .99 cents a month + .01cent per minute
OR
- $4.95 a month for 3,500 minutes (not sure what overage would be)

[duckdown]

Why are there conversation drop outs in mid-conversation??

[jackie999]

On that note, does anyone know why sometimes during conversations the voice will cut out for a second during talk? I've had to repeat myself sometimes because people say I'm cutting out..

I notice a 'choppyness' if I'm downloading something and I'm on the phone. My solution is to pause any downloads. You can try setting up QoS for your router to give the PAP2T priority...I haven't bothered myself..

[duckdown]

thats kind of corny, isn't there any kind of pre buffering or something ? why is it so tempremental for hogging all your modems bandwidth? I read something that says 1 minute of talk is only 110kb, so what the hell does it need complete modem data silence for (besides itself)?

Not really all that impressed with the service so far, so many pitfalls

[robinto]

Why are there conversation drop outs in mid-conversation??

Were you calling international ?
If so, you might want to switch to premium.
I had drop outs for international calls to Asia.
Quality seems to improve a lot after choosing "premium" for international.
It's kinda "hit & miss", sometimes it's ok on "value"

[shadow_cruiser]

I've made 3-4 10 minute conversations since setup and I'm really happy with the service. My friends say the quality is much better than the cell. So I guess it goes like this: Landline>VoiP>Cell. I haven't had any dropped or choppy conversations even though I'm a heavy downloader. On the other hand, I have QOS setup for this.

Duckdown is right though about SMS, I'm not sure how I can do without it as all friends text me min. 20 times on a daily basis. I'm currently looking into the cheapest method for texting and limited voice plan.

[duckdown]

Were you calling international ?
If so, you might want to switch to premium.
I had drop outs for international calls to Asia.
Quality seems to improve a lot after choosing "premium" for international.
It's kinda "hit & miss", sometimes it's ok on "value"

Nope, no international, not even long distance. All local calls within Brampton. People keep telling me that I'm cutting out while I'm talking and I keep having to repeat myself, it's getting rather annoying. No wonder it's only 1/2cent per minute

[duckdown]

I've made 3-4 10 minute conversations since setup and I'm really happy with the service. My friends say the quality is much better than the cell. So I guess it goes like this: Landline>VoiP>Cell. I haven't had any dropped or choppy conversations even though I'm a heavy downloader. On the other hand, I have QOS setup for this.


I don't think my craptastic Rogers extreme modem has QoS. It's a SMC3DGN and it's pretty much a piece of total crap.. Maybe I'll have to look into a third party router again



Duckdown is right though about SMS, I'm not sure how I can do without it as all friends text me min. 20 times on a daily basis. I'm currently looking into the cheapest method for texting and limited voice plan.

Yeah :(

I get way more than 20 texts a day, I kind of underestimated how important SMS was before I signed up for the service. Disappointing that their customer support says they have no plans at all for SMS anytime soon.. Not sure what to do at this point, might call them and tell them I am quite unhappy with their service so far, see if they can do anything for me

thanks for the responses

[jackie999]

Nope, no international, not even long distance. All local calls within Brampton. People keep telling me that I'm cutting out while I'm talking and I keep having to repeat myself, it's getting rather annoying. No wonder it's only 1/2cent per minute
Perhaps you should contact voip.ms live chat and ask them to look into it..
When I noticed choppyness it's when I'm listening to someone talk...people that I'm calling don't notice anything from their end..and as soon as I pause my download it clears up.
You mentioned 'many pitfalls' about the service...IMHO this is a small inconvienience..since I could take the time to look into QoS if I wasn't so lazy :)

[jackie999]

I get way more than 20 texts a day, I kind of underestimated how important SMS was before I signed up for the service. Disappointing that their customer support says they have no plans at all for SMS anytime soon.. Not sure what to do at this point, might call them and tell them I am quite unhappy with their service so far, see if they can do anything for me

I'm not a big cell phone user..in fact I'm quite happy with my speakout 7eleven phone service at $25 per year. I have done a little texting..is that SMS? I would be surprised if this voip.ms offered it since that, to me, is a feature of a cell phone. My understanding of this voip is it's a replacement of your land line and my bHell landline never had texting :)

[duckdown]

I'm not a big cell phone user..in fact I'm quite happy with my speakout 7eleven phone service at $25 per year. I have done a little texting..is that SMS? I would be surprised if this voip.ms offered it since that, to me, is a feature of a cell phone. My understanding of this voip is it's a replacement of your land line and my bHell landline never had texting :)

Yep, texting is SMS :)

I hear what you're saying, maybe I will have to look into a SpeakOut wireless phone again. I used them for a good part of last year and was generally happy. But I still think voip.ms should get with the times -- Bell has things like text to speech SMS and other ways to send SMS from a land line, so the fact that they don't have this feature in any form is disappointing.. Yes it does replace a landline but it also relies on the Internet to do so, which should have some kind of SMS gateway functionality IMO. Text is only getting more popular, not less popular, so hopefully they look into this sooner rather than later!

Just my 2 cents of course

I think I will try and lookup their tech support or customer care number and express my concerns, maybe something will come of it if their support is as good as people say!

cheers

[djino]

For SMS, use Google Voice in addition to voip.ms

djino

[duckdown]

For SMS, use Google Voice in addition to voip.ms

djino

Hi, I just tried google voice website and it says it's invite only :( Also says only available in the USA

Does anyone have an invite they can send me so I can try it? Having a working SMS alternative would make me ALOT happier !

thanks

[djino]

Hi, I just tried google voice website and it says it's invite only :( Also says only available in the USA

Does anyone have an invite they can send me so I can try it? Having a working SMS alternative would make me ALOT happier !

thanks

You can request your invite here -> https://services.google.com/fb/forms/googlevoiceinvite/

Yup, you can use it in Canada (but with a U.S number). When you finally get the invite, you will have to mask your IP with a U.S. IP and go through the sign up process to get your U.S. Telephone #.

djino

[jackie999]

You can request your invite here -> https://services.google.com/fb/forms/googlevoiceinvite/

Yup, you can use it in Canada (but with a U.S number). When you finally get the invite, you will have to mask your IP with a U.S. IP and go through the sign up process to get your U.S. Telephone #.

djino


WTF?!?!
Why isn't it available up here? What are we..chopped liver..?

[EPcjay]

I just registered...is it normal for them to ask for a scan of your DL?

[shadow_cruiser]

I don't think my craptastic Rogers extreme modem has QoS. It's a SMC3DGN and it's pretty much a piece of total crap.. Maybe I'll have to look into a third party router again




Yeah :(

I get way more than 20 texts a day, I kind of underestimated how important SMS was before I signed up for the service. Disappointing that their customer support says they have no plans at all for SMS anytime soon.. Not sure what to do at this point, might call them and tell them I am quite unhappy with their service so far, see if they can do anything for me

thanks for the responses

I bought the infamous WRTGL years ago and it has helped me in many ways throughout the years. Don't worry, it's a great investment. I've found 7/11 Speakout to be the cheapest for texting. At $10/month, you can have 2500 texts. Worth it I think.

http://speakout7eleven.ca/pages/rates/

[duckdown]

I just registered...is it normal for them to ask for a scan of your DL?

Where? at voip.ms or google voice?

[jackie999]

I just registered...is it normal for them to ask for a scan of your DL?

Yes it is..this was asked earlier in the thread as well. I assumed it's proof that the number you're asking to port is your own.

[EPcjay]

where? At voip.ms or google voice?

voip.ms

[EPcjay]

Yes it is..this was asked earlier in the thread as well. I assumed it's proof that the number you're asking to port is your own.

thanks

[duckdown]

voip.ms

I didn't need any proof and I also signed up with an abbreviated name..

IE: instead of Jim Lahey sign up as Jim L. and use a different address and so on

[jackie999]

I didn't need any proof and I also signed up with an abbreviated name..

IE: instead of Jim Lahey sign up as Jim L. and use a different address and so on

Did you port over your existing telephone number?

[Fatman]

how does this work?

Can I buy a DID and have it forwarded to my cell phone?

[djino]

how does this work?

Can I buy a DID and have it forwarded to my cell phone?

Yes.

But you would have to pay $0.99/month for the DID. Anytime someone calls your DID, you would pay $0.01/minute for the call into your DID as well as $0.0052/minute (1/2 cent) for sending the call out to your cell phone (termination forwarding).

djino

[shadow_cruiser]

What the??. Speakout is not available in Ottawa. Just moved here and that's another thing not available (no frills either lol). Looks like I'll have to wait until I visit Mississauga.

PS. I just did the changes from toao.net and the call quality is significantly better. We'll see how the next two weeks turns out

[jackie999]

What the??. Speakout is not available in Ottawa. Just moved here and that's another thing not available (no frills either lol). Looks like I'll have to wait until I visit Mississauga.

From what I read on the speakout unofficial forums you will soon be able to buy your top up online....having to find a 7-11 store is a PITA..but the service is great.

[shadow_cruiser]

Looks like QoS isn't working. I set my router up to the tee following this guide http://www.dslreports.com/forum/r21581951-HOWTO-QoS-and-Tomato-fixes-choppy-voice but once QoS is enabled, the voice is extremely choppy. When I disable it, it goes back to normal, nice and clear.

Anyone have another guide for Qos/Voip/Tomato?

[jackie999]

Looks like QoS isn't working. I set my router up to the tee following this guide http://www.dslreports.com/forum/r21581951-HOWTO-QoS-and-Tomato-fixes-choppy-voice but once QoS is enabled, the voice is extremely choppy. When I disable it, it goes back to normal, nice and clear.

Anyone have another guide for Qos/Voip/Tomato?
Heh..I was going to pm you and ask what your settings were :)
I was on linksys forums and there are a load of guides..but it's way over my head..best I could understand was that tomato Qos is okay for uploads..not downloads. What we need is something to give the PAP2T priority when it's needed...is Qos the right tool? If you figure it out..pass it on :)

[EPcjay]

Did you port over your existing telephone number?

I'm going to, but I haven't even advised them that yet.

[shadow_cruiser]

Heh..I was going to pm you and ask what your settings were :)
I was on linksys forums and there are a load of guides..but it's way over my head..best I could understand was that tomato Qos is okay for uploads..not downloads. What we need is something to give the PAP2T priority when it's needed...is Qos the right tool? If you figure it out..pass it on :)

Are you using Tomato with MLPPP?. QoS in Tomato is the best and have had no problems configuring it with Bittorrent in the past. One downside about QoS Tomato is it's advanced, a little too advanced in fact. I tried DD-WRT QoS and it's much more basic as well the understanding of it is there. I set it up and tested a few calls, all seems to be well. I don't need MLPPP anymore as I very rarely use bittorrent these days. Let me know if you want to switch to DD-WRT and I'll post my settings for you.

[jackie999]

Let me know if you want to switch to DD-WRT and I'll post my settings for you.

I quite like tomato, so will stick with it..but thanks for the offer!
I did use MLPPP for awhile, then I moved away from torrents so I updated my tomato to current version (non MLPPP). Now I'd like to find how to set up priority for voip over downloads...I download a lot of linux distros :) and I find pausing them when a call is coming in or going out works fine..but I can't help but wonder if there is something else out there. I did read (here maybe?..or dslreports) that the ISP is where the priority can be set up..hopefully something teksavvy will offer in the future.

[apn64]

I quite like tomato, so will stick with it...
Now I'd like to find how to set up priority for voip over downloads...I download a lot of linux distros :) and I find pausing them when a call is coming in or going out works fine..but I can't help but wonder if there is something else out there.

Using Tomato (I'm on v27), setup the IPaddr of your VoIP ATA to be the highest classification and have everything else in High <-> Low classifications i.e. only the ATA is in Highest. Also set the Highest classification to allow a minimum of 100Kbps uplink bandwidth (assuming you're using the G711 codec); double it if you're using both ports on an ATA e.g. PAP2T and expect to have concurrent calls.

It works extremely well, as I discovered when I downloaded >1.3TB in January; the connection was running hot for the entire month and I never once paused or stopped a download to take a call. Incidentally, my CDR stats for that month show; Calls: 355 | Duration: 32:09:00

[shadow_cruiser]

I've been using Tomato for years and I gotta stick with it. I installed DD-WRT and went back to Tomato right away. These are the settings I have working for me, so far. Tested downloading a file at full bandwidth and placed a call, no choppyness and noticed the graph's changed.

VOIP SIP signaling
- Any IP address
- UDP
- Dest port: 5060-5070
- Layer 7: sip
- Class: Highest.
- Desc: VOIP SIP signaling.

VOIP RTP voice

- Any IP address
- UDP
- Dest port: 10000-20000
- Layer 7: rtp-2
- Class: Highest.
- Desc: VOIP RTP voice

http://imgur.com/Aslth.jpg
http://imgur.com/5lniB.jpg

VOIP in use
http://imgur.com/XFaY3.jpg

VOIP not in use
http://imgur.com/5kLY3.jpg


Using Tomato (I'm on v27), setup the IPaddr of your VoIP ATA to be the highest classification and have everything else in High <-> Low classifications i.e. only the ATA is in Highest. Also set the Highest classification to allow a minimum of 100Kbps uplink bandwidth (assuming you're using the G711 codec); double it if you're using both ports on an ATA e.g. PAP2T and expect to have concurrent calls.

It works extremely well, as I discovered when I downloaded >1.3TB in January; the connection was running hot for the entire month and I never once paused or stopped a download to take a call. Incidentally, my CDR stats for that month show; Calls: 355 | Duration: 32:09:00
This sounds so much easier. Would you mind posting a screenshot of your classification settings?

[apn64]

This sounds so much easier.

That's because it is ;)


Would you mind posting a screenshot of your classification settings?

Sure, here's what I have;

http://img809.imageshack.us/img809/6226/qos1.jpg (http://img809.imageshack.us/i/qos1.jpg/)

So the ATA is Highest classification, and with a few exceptions, everything else is captured by the "Low" classification.

I added the Desktop/Laptop PC exceptions in January, but I later found out the machines couldn't connect because my Server was grabbing all the connections. I subsequently installed Tomato v1.27 w/ SpeedMod, increased the number of available connections. That solved the problem, so I should probably remove those two items from the High group.


http://img191.imageshack.us/img191/1394/qos2.jpg (http://img191.imageshack.us/i/qos2.jpg/)

Here you can see that no matter what happens, the Highest group (ATA) is guaranteed to get at least 200Kbps of my 1Mbps (Extreme) uplink i.e. enough for two simultaneous G711 calls @ 85Kbps a piece.

The High group also gets a minimum of 100Kbps, so DNS requests are quick etc., and the Low group gets whatever is left, but will never get more than 80% of the uplink.

Finally, note that I have no restrictions on the 10Mbps down link. This is my entire QoS strategy and I don't bother with protocol selection (slows things down), selective port forwarding or anything else. It works really well - even when I had a month-long download binge :D

[jackie999]

Very helpful information here..thanks!
Couple of things..Shadow Cruiser shows voip as "high" in match rule, from what I've read this doesn't work unless it's actually 'moved' to the top in the class listing?
Secondly, apn64 is using 1000 kbit/s on outbound. ATM my TSI connection is 5M/800k, everything I read says to take a percentage of that..since you're not configuring download, I'd use the 800 kbit/s as outbound?
Thanks for these screen shots..makes it so much easier!

[apn64]

Yes, you'd set 800Kbps in the Outbound Max Bandwidth box and base your percentages on that to ensure that you get a minimum of ~100Kbps for each of the simultaneous VoIP calls that you plan to use, so your % #'s will be higher.

I use both L1 and L2 on my PAP2T and constantly run parallel calls, hence the need to reserve ~200Kbps for the Highest group.

Finally to elaborate, I do not use protocol selection since it really hogs router CPU capacity i.e having to open the header of every frame to inspect the protocol and then decide if it meets a certain criteria. Port selection also loads the CPU, albeit to a lesser extent.

Every packet has a src and dest IPaddr. and must be routed accordingly. My QoS implementation uses this basic information to quickly prioritize all traffic to/from my ATA at the lowest possible overhead = maximum efficiency.

[FlintBlade]

I have a dumb question.

I purchased a 905 number from Dundas, Ontario. Does that mean when people call me it will be considered a Dundas, ON line? So it would be long distance for those that call me located in Hamilton and be registered as a Dundas number?

I wanted to keep a 905 number but Stoney Creek had nothing but 289s. I'll switch to a 289 from Stoney Creek if I have to. But I prefer to use a 905 number.

[jackie999]

I have a dumb question.

I purchased a 905 number from Dundas, Ontario. Does that mean when people call me it will be considered a Dundas, ON line? So it would be long distance for those that call me located in Hamilton and be registered as a Dundas number?

I wanted to keep a 905 number but Stoney Creek had nothing but 289s. I'll switch to a 289 from Stoney Creek if I have to. But I prefer to use a 905 number.
It's not a dumb question. Yes, people that call you will be calling Dundas. I had a relative (who didn't know any better) buy the magicjack device...he bought it with a Florida number. Everytime I called him, one town over, I was calling Florida :(
The 416's and the 905's are getting scarce..that's why I opted to pay the $25 and port my existing 905 number over. If you wait, I think the available number list is always updating..perhaps a more convenient one will pop up before long?

[RMSimard]

Anyone else having issues with outgoing voice randomly going "robotic" during incoming calls? Problem started this weekend.... driving me nuts :(
My upstream is only at 100kbps out of 400kbps during a phone call and nothing else uploading - it is definitely not a QoS/internal network issue.

Happened once to me this week. And another call apparently had unbearable echo/staccato effect. Both times it was the recipient who was experiencing the effects, everything sounds good on my end. And my network was quiet with no data transfer. Mmmhhh...

[shadow_cruiser]

Maybe some problems with the montreal server?. I also have problems too

EDIT: Noticed the montreal servers wasn't registering and changed to the Toronto servers. Bah, I guess that's voip, lol

[RMSimard]

Maybe some problems with the montreal server?. I also have problems too

EDIT: Noticed the montreal servers wasn't registering and changed to the Toronto servers. Bah, I guess that's voip, lol

Good point! I'll change over to Toronto for a few weeks, see if the problem is on my end or on the side of montreal.voip.ms.

[shadow_cruiser]

Good point! I'll change over to Toronto for a few weeks, see if the problem is on my end or on the side of montreal.voip.ms.

I never tried the Toronto server as I'm in Gatineau, so it didn't make sense. When I switched, the dial tone seemed much clearer. Before I could hear background noise, static, etc.

[apn64]

I'm the other side of the river and when I initially signed up (Feb09), I went w/ Montreal for L1 and NYC for L2.

I figured that since Montreal was closer and also home base for VoIP.ms, it would be a good choice. Sadly, I was mistaken. Whenever I had a problem, I'd switch over to Toronto and Voila! problem solved.

I've long since switched to using Toronto and NYC. Note that as well as diversity, this setup has the advantage of providing access to both CDN and USA toll-free services :D

[shadow_cruiser]

I submitted a ticket and this is the response I received, I guess it's generic as I already did this. The montreal server is very unstable. I've had issues since initial setup and I thought it was my configuration but it's actually the montreal server.

"Dear Customer,

We haven't received any report regarding Montreal issues, I have tested already the server by registering a softphone, try setting the server IP address instead of the DNS name (this information can be found from the customer portal >> main menu >> account information), and advise."

[EPcjay]

I just joined voip.ms, got the pap2 adapter. I can make calls and recieve calls without an issue.

The issue i am having is the phone doesn't ring like a regular phone when calls come in.. It randomly cuts out during the time it rings. I checked the NAT stuff, and it appears to be good. I even turned on dmz mode on my router to the voip adapter because i thought the ports weren't forwarding correctly.

Any help would be appreciated.

Thanks

[EPcjay]

Nevermind, I fixed it. The waveform for the ring need to be in sinusoid

[ToChicka]

hi guys I have a quick question...

Can I signup with voip.ms and get a toronto # and have it routed to a US# or even ring at the same time my TO#? More like a call forking?

I won't be using a PAP2 so was wondering if it;s possible in their admin site to accomplish this?

My cost would be monthly 416# fee + incoming call fee?

[destinet]

hi guys I have a quick question...

Can I signup with voip.ms and get a toronto # and have it routed to a US# or even ring at the same time my TO#? More like a call forking?

I won't be using a PAP2 so was wondering if it;s possible in their admin site to accomplish this?

My cost would be monthly 416# fee + incoming call fee?

Yes you can, you only need to setup the DID to a ring group. And after the ring group will ring the call forwarding number you created. So both will ring at the same time.

You will pay the did fees per month (0.99$) + the incoming call rate (0.01 / min) + the outgoing rate to your number (0.0052$ /min)

[ipopov]

Nevermind, I fixed it. The waveform for the ring need to be in sinusoid

Could you provide more details? Where can I set the waveform?

Thanks

[hostlead]

It's pretty obvious I couldn't be happier with the service. I just looked at my account..I've been with them 1 month yesterday. My costs for one month (excluding $25 set up) were
"Calls: 108 | Duration: 06:24:37 | Total: $2.11449415"
So usage was $2.11 + DID $1.49 (which is now .99cents a month) = $3.60
My $25 investment is sitting at $21..should last me a while :)
Take THAT bell :lol:

As for the APC backup..I have one on order. All of my voice mail is to voip.ms, so no worries there if power goes out here. I expect it will run off a 350 APC for quite awhile..there isn't much draw - maybe somebody else will let us know.

I'm with teksavvy and connection is nice and stable. Only problem I have found..when I'm on phone, and I'm downloading something large, the connection is a little choppy..so I pause my download. There are ways to go into your settings in router to give voip priority over downloads..but I've been too lazy to look into setting that up.

Would a cable connection keep running if the power went out and I had a UPS?

[hostlead]

also is it possible to have two seperate voip.ms accounts on one ATA?

[ToChicka]

Yes you can, you only need to setup the DID to a ring group. And after the ring group will ring the call forwarding number you created. So both will ring at the same time.

You will pay the did fees per month (0.99$) + the incoming call rate (0.01 / min) + the outgoing rate to your number (0.0052$ /min)

thanks Destinet, so I still pay outgoing rate even though I won't be calling out more likely wanting to use as a call forwarding option?

[Kirbyprime]

Question for anyone who's more familiar with fees and such. What's the total monthly fees for going with voip.ms? Currently I have Roger's Cable service and Bell Home Phone. If I cancel my POTS do I have to pay the $9 dry loop monthly fee?

[djino]

thanks Destinet, so I still pay outgoing rate even though I won't be calling out more likely wanting to use as a call forwarding option?

Yes. Forwarding consist of paying a combination of the 2 rates. $0.01/minute for incoming when someone calls your DID. Then if you are forwarding the call, the system needs to call out to the number you set for call forwarding - $0.0052/minute for it calling out to your Canadian number.

Total = $0.0152/minute = 1.52 cents/minute (all this on top of the $0.99/month for the DID).

djino

[djino]

Question for anyone who's more familiar with fees and such. What's the total monthly fees for going with voip.ms? Currently I have Roger's Cable service and Bell Home Phone. If I cancel my POTS do I have to pay the $9 dry loop monthly fee?

You only pay Dry loop fee if you have a DSL Internet connection with no POTS telephone line.

Since your Internet is with Rogers, this is cable internet (not DSL). You would continue to pay for Rogers Internet service with no Dry loop fee.

If you want to port your Bell Home Phone to voip.ms, its a one-time $25 charge.
You then choose whether you want to pay $5/month for unlimited incoming (up to 3500 minutes) or $1/month + $0.01/minute on incoming.

Termination (Outgoing calls) are priced at $0.0052 (0.52 cents)/minute to most places in Canada and $0.011 (1.1 cents)/minute to most places in the U.S.

djino

[Pete Jones]

Question for anyone who's more familiar with fees and such. What's the total monthly fees for going with voip.ms? Currently I have Roger's Cable service and Bell Home Phone. If I cancel my POTS do I have to pay the $9 dry loop monthly fee?

I'm assuming since you mention you have Rogers cable, that's how you're getting your internet. In that case, your POTS line is history and there's no dry loop fee.

If you were using DSL (in which case, I'm not sure why you mentioned Rogers) then yes, there would be a dry loop fee.

As far as monthly costs, Fixed costs are basically $1/month for the DID (the phone number) and then whatever you actually use. I haven't really used it long enough to do a proper assessment but I I'm about to pull the plug on my own POTS line and there's no way I will even come close to the $40/month that the POTS line costs me right now.

From other messages in this thread, there are folks here who talk up a storm and only pay $5/month. Ultimately, it will come down to usage but with the fixed cost of the POTS line out of the picture, you can pretty much expect some significant savings.

--Pete

[jackie999]

Would a cable connection keep running if the power went out and I had a UPS?

I don't have cable yet...teksavvy is only just setting it up in my area...
As far as I know, as long as you have the ATA, cable modem, router and a phone plugged into the backup you'll have a working phone during a power outage (which is more than people using cordless phones with a landline will have :) )

[Kirbyprime]

I'm assuming since you mention you have Rogers cable, that's how you're getting your internet. In that case, your POTS line is history and there's no dry loop fee.

If you were using DSL (in which case, I'm not sure why you mentioned Rogers) then yes, there would be a dry loop fee.

As far as monthly costs, Fixed costs are basically $1/month for the DID (the phone number) and then whatever you actually use. I haven't really used it long enough to do a proper assessment but I I'm about to pull the plug on my own POTS line and there's no way I will even come close to the $40/month that the POTS line costs me right now.

From other messages in this thread, there are folks here who talk up a storm and only pay $5/month. Ultimately, it will come down to usage but with the fixed cost of the POTS line out of the picture, you can pretty much expect some significant savings.

--Pete

Yep, I meant I had Rogers Internet. Sorry, too used to calling my internet "cable" since I don't really watch TV. Thanks for the quick reply!

[jackie999]

Could you provide more details? Where can I set the waveform?

Thanks

Try logging into your ATA as admin / advanced. On the Regional tab mine is set to trapezoid ....I'm not sure of the difference.

There is also hints on setting the ring on this site,
http://www.toao.net/25-linksys-pap2t-voip-adapter-review

[psyko514]

How does their service quality compare to Primus?

[Pete Jones]

I don't have cable yet...teksavvy is only just setting it up in my area...
As far as I know, as long as you have the ATA, cable modem, router and a phone plugged into the backup you'll have a working phone during a power outage (which is more than people using cordless phones with a landline will have :) )

True. But another thing to consider. If you set up voicemail, (which is free) then even if you are totally SOL because of a power failure, if someone calls you, the call will still go to voicemail.

Something else to consider is that unless you set it up and pay for it, you won't have 911 service. (it costs $1.50/month) However, this shouldn't be an issue if you have a cell phone.

--Pete

[ToChicka]

Yes. Forwarding consist of paying a combination of the 2 rates. $0.01/minute for incoming when someone calls your DID. Then if you are forwarding the call, the system needs to call out to the number you set for call forwarding - $0.0052/minute for it calling out to your Canadian number.

Total = $0.0152/minute = 1.52 cents/minute (all this on top of the $0.99/month for the DID).

djino

thanks again djino for the actual rates you posted.

[duckdown]

Can someone tell me how that feature works where your voip.ms account calls your cellphone with a dialtone ?

Thanks

[djino]

Can someone tell me how that feature works where your voip.ms account calls your cellphone with a dialtone ?

Thanks

Callback feature. When logged into voip.ms, click the DID Menu, and select callback.

djino

[duckdown]

Callback feature. When logged into voip.ms, click the DID Menu, and select callback.

djino

Thanks for the response, can't figure out how to get it working though.. I put my cell number in, then what am I supposed to do? I call my DID # ? When i do that, my house phone just rings, whats hooked up to the PAP2 and it doesn't call me back

Sorry for the noobish questions

[djino]

Thanks for the response, can't figure out how to get it working though.. I put my cell number in, then what am I supposed to do? I call my DID # ? When i do that, my house phone just rings, whats hooked up to the PAP2 and it doesn't call me back

Sorry for the noobish questions

Did you create a CallerID Number filter (Check the same DID Menu for this option)?

djino

[djino]

Basically, you create your Callback route.

Ie, the number that voip.ms will call you back on. (you do this through the callback feature).

Once created. Then go to the CallerID Filter Options (so that you can tell it to only call you back) when someone calls that matches your CallerID.

djino

EDIT: Remember you are billed for both legs of the call. $0.0052 (1/2 cent)/minute as the system needs to call you back on your Canadian number. Then another $0.0052 (1/2 cent)/minute to place your outgoing Canadian call (or $0.011/minute to place your outgoing US call).

[robinto]

I have an old telephone system in the house, i.e. no control panel for individual phone jacks. I am planning to use 2 DIDs and when the 1st line/number is being used, the rest of the phones can still be able to dial/receive from the 2nd line/number. Do I have to separate the phone system into 2 groups connect to the 2 phone jacks on the ATA? Is this the only way for my plan? Can anyone share your config with 2 lines ?
TIA

[apn64]

Older phone wiring consists of 2 pairs of wires; red/green and yellow/black.

It's safe to assume that if all phones currently ring at once, and you can answer by picking up any of the phones; that all your phones are connected in parallel.

If that's the case, then all your phone wires will be tied together i.e. all reds, all greens are twisted together and connected at the demarcation point in the basement, or perhaps in a box on the outside wall of the house.

The standard phone connection (Line1) uses the red/green pair, per above. It was customary to use the yellow/black pair if the customer ordered a 2nd line (Line2).

So you have a couple of options;

1. Split the phones into two groups for Line1 and Line2 using the current red/green pair.

2. If you have access, combine the yellow/black pairs at the demarc point AND in the wall jacks where you plan to install 2-line phones. You will need to buy 2-line phones if you don't have them today.

Option #1 is by far the least expensive, but you lose the flexibility of being able to pickup any phone if, for example, Line1 is ringing. Option #2 could cost a few $100's depending on the number and type of 2-line phones purchased.

Finally, perhaps some good news. Since we're talking about VoIP.ms, you don't need to order two DIDs and have all the attendant costs.

You can setup the main account to forward calls to another group of phones or to the 2nd line of a 2-line phone via a sub-account, when the main line is busy.

FWIW, I have a single DID and use both the L1 and L2 ports on my ATA. I have four 1-line phones and one 2-line phone. When the phone rings, all the phones in the house ring. However, when the main line is busy, I can still make calls on Line2. Per above, I could also direct incoming calls to Line2 when Line1 is busy, but I don't want that capability.

[robinto]

Very detail explanation.
Thank you so much.


Older phone wiring consists of 2 pairs of wires; red/green and yellow/black.

It's safe to assume that if all phones currently ring at once, and you can answer by picking up any of the phones; that all your phones are connected in parallel.

If that's the case, then all your phone wires will be tied together i.e. all reds, all greens are twisted together and connected at the demarcation point in the basement, or perhaps in a box on the outside wall of the house.

The standard phone connection (Line1) uses the red/green pair, per above. It was customary to use the yellow/black pair if the customer ordered a 2nd line (Line2).

So you have a couple of options;

1. Split the phones into two groups for Line1 and Line2 using the current red/green pair.

2. If you have access, combine the yellow/black pairs at the demarc point AND in the wall jacks where you plan to install 2-line phones. You will need to buy 2-line phones if you don't have them today.

Option #1 is by far the least expensive, but you lose the flexibility of being able to pickup any phone if, for example, Line1 is ringing. Option #2 could cost a few $100's depending on the number and type of 2-line phones purchased.

Finally, perhaps some good news. Since we're talking about VoIP.ms, you don't need to order two DIDs and have all the attendant costs.

You can setup the main account to forward calls to another group of phones or to the 2nd line of a 2-line phone via a sub-account, when the main line is busy.

FWIW, I have a single DID and use both the L1 and L2 ports on my ATA. I have four 1-line phones and one 2-line phone. When the phone rings, all the phones in the house ring. However, when the main line is busy, I can still make calls on Line2. Per above, I could also direct incoming calls to Line2 when Line1 is busy, but I don't want that capability.

[tdott]

If that's the case, then all your phone wires will be tied together i.e. all reds, all greens are twisted together and connected at the demarcation point in the basement, or perhaps in a box on the outside wall of the house.

APN, can you confirm that regular 1line phones don't need to be connected to the ATA directly?

Can I run port1 from the ATA to my Demarcation point to power all phones in parallel?

I'm just not clear on if the ata needs to be connected 1-to-1 or if it can be split like older landlines. I'm thinking if i can run 1 port to my demarcation point, it will power all 4 phones on that line.

In theory i think it should work.
My followup question is that if that works.....say the line is in use and someone else picks up another phone, can they hear the call like older landlines?

[apn64]

APN, can you confirm that regular 1line phones don't need to be connected to the ATA directly?

No they don't, assuming that they have a standard Ringer Equivalence Number (REN) rating. If your phones are tied together today, odds are they'll work just fine on the ATA.


Can I run port1 from the ATA to my Demarcation point to power all phones in parallel?

I'm just not clear on if the ata needs to be connected 1-to-1 or if it can be split like older landlines. I'm thinking if i can run 1 port to my demarcation point, it will power all 4 phones on that line.


YES, this is the standard configuration that I expect that most of us are using.


In theory i think it should work.
My followup question is that if that works.....say the line is in use and someone else picks up another phone, can they hear the call like older landlines?

YES again, since it's no different than what you probably have today.

The situation only changes if you do funky things like split the phones into groups for L1 and L2 respectively.

All five of my house phones ring when a call comes in, but only one of my phones has a 2-line capability. The 2nd line of that phone is connected to L2 on my ATA, thus allowing me to make calls when L1 is busy.

[Pete Jones]

Anyone from Ottawa using the service? I've been having issues since June 6.
During incoming phone calls only, there is a high probability of the outgoing voice going bad (sounds robotic). I am quite tech savvy and I've tried quite alot of different things.
Things were working great for a month but now it's garbage.
Reproducible with softphone, ata, multiple dids, you name it.... it could happen within a few seconds, minutes, more...

My only hope is to get this resolved through voip.ms support and despite many tickets, the issue is still unresolved.Very disappointed by the support I am receiving.

Am I the only one?

For someone who is tech savvy you're not giving us a lot to go on.

1. Are you using DSL or Cable?
2. What are your current speedtest.net numbers including ping latency
3. Are you using an ATA?
4. Is there any other data transfer going on during the call?
5. On your softphone, have you tried it on a different computer?

My service is rock solid. I've made some tweaks in my ATA to improve the overall experience and reduce echo but even with that it was more than adequate.

--Pete

[apn64]

Working fine here in the capital.

First instinct would be QoS related issues with such a small uplink. Are you absolutely sure that there's nothing else going upstream during these calls?

What server are you using and have you checked the recent ping time? BTW, the montreal server flat out sucks; I use TO and NYC after abandoning MTL a long time ago.

[jackboot]

g0blez: run this test and post the results:

http://myspeed.visualware.com/indexvoip.php

[nosnoop]

Does voip.ms work well if I just want a number to forward fax to OneSuite.com fax receive? I like OneSuite's unlimited fax receive, but you do not own the number and cannot port away even if their service deteriorates. So I am looking at getting a DID just for the fax forwarding. Any comments or suggestions?

[bridonca]

Anyone from Ottawa using the service? I've been having issues since June 6.
During incoming phone calls only, there is a high probability of the outgoing voice going bad (sounds robotic). I am quite tech savvy and I've tried quite alot of different things.
Things were working great for a month but now it's garbage.
Reproducible with softphone, ata, multiple dids, you name it.... it could happen within a few seconds, minutes, more...

My only hope is to get this resolved through voip.ms support and despite many tickets, the issue is still unresolved.Very disappointed by the support I am receiving.

Am I the only one?

Ok, I was hearing people say how great voip.ms is for calls, compared to some other provider. I see out voip.ms has the same problems! Sadly for you, it is extremely likely there is nothing wrong with voip.ms gear. It is your setup that is likely at fault.

I accuse your Internet provider, who is producing jitter, most likely because of the traffic shaping setup they deny using. Which would be true, because they turn it off when there are complaints. I would chew your Internet provider's ear out. You will be amazed how the problem will just go away.

What are you using for a codec? G.729a is very resistant to jitter.

[paulo500]

I don't know if this would make much of a difference, but maybe some folks are experiencing unusual jitter due to the world cup?
I could imagine a heavier than normal demand on the networks during this time because of live video streaming and downloading? Im not sure. Just puttin' it out there...

[Lulz]

Few questions in 1 post.
If something was answered, please let me know which page I can read the answer on. Typing this on slow laptop tethered from EDGE internet in Russia so can not search every page. Thank you!

Hello. I was wondering what International Routing and Toll-Free Routing is for and what difference would be in premium vs value?

2)If I am currently in Europe, should I change my DID POP to London or keep it on New York?

And finally 3rd question:

My CallerID Name Lookup is set to No Thanks, yet I can see who's calling me when they are not on my phonebook....and even sometimes I can see their registered name (similar to what Rogers has for NameDisplay when even their name comes up on top of number)
Thank you!

[lazychild]

Just wondering, are there anymore sites that like VOIP.ms (in terms of price and number selection) that you guys can recommend out there?

Thanks

[Pete Jones]

Just wondering, are there anymore sites that like VOIP.ms (in terms of price and number selection) that you guys can recommend out there?

Thanks

Yes.

www.callwithus.com

www.localphone.com

I have used both of these to make calls from my POTS and cell phones. They also have can be used with voip. (ATAs and soft phones) I have actually set up callwithus on my ATA and another provider that I can dial out with in case there are problems with voip.ms. Both these offer DIDs as well.

There are others, but these are the ones I have experience with.

--Pete

[death_hawk]

Does voip.ms work well if I just want a number to forward fax to OneSuite.com fax receive? I like OneSuite's unlimited fax receive, but you do not own the number and cannot port away even if their service deteriorates. So I am looking at getting a DID just for the fax forwarding. Any comments or suggestions?
FoIP is touchy. I'm pretty sure voip.ms doesn't have T.38 equipped yet (even as a passthrough) so I highly doubt any faxes coming in would make it over. There is limited success with ulaw and faxing, but it's not always consistent.
Rumor has it that voip.ms is getting t.38 but I have no idea what the timeline is. Best bet is to open a ticket to see if they support t.38 passthough.

I was always under the impression that anyone acting as a carrier HAD to allow port out (and probably in) of numbers. Granted, I guess this technically isn't a carrier but you get the idea.


~snip~

1) I have no idea what voip.ms carriers are, but I assume that Premium uses a better quality transit from point A to point B. What I'd do is test out both and see which is better for you. Anything anyone says here is moot (unless they're where you are and calling the same destination). I would also double check pricing as there's a few instances where premium is cheaper than value for some countries.

2) Again it depends on you. I'm not sure what the latency difference is from NY or London (or if you're in Russia, Cali) but I'd ping each server to see what's lowest. Then pick the POP from there.

3) I can't really answer that one.

[jackboot]

Has anyone here purchased a PAP2T from Canada Computers (LINK) (http://www.canadacomputers.com/product_info.php?cPath=30_414&item_id=014833&sid=j7ivpml4ica7gtpj3pl2a5bn25)?

I purchased one from CC but there is an admin password when trying to use the IVR (telephone interface). For example, when I try to enable/disable the web-interface I am asked to provide a password. This should not be the case on a factory-unlocked ATA. It appears as though Canada Computers sold me a locked ATA :mad:

Can anyone else verify if their Canada Computers purchased ATA is the same?

on your phone dial: ****
enter: 7932 #

at this point it may ask you for a password. If it does not, then your unit is factory unlocked. If it does, then you have the same problem as me.

[shadow_cruiser]

Has anyone here purchased a PAP2T from Canada Computers (LINK) (http://www.canadacomputers.com/product_info.php?cPath=30_414&item_id=014833&sid=j7ivpml4ica7gtpj3pl2a5bn25)?

I purchased one from CC but there is an admin password when trying to use the IVR (telephone interface). For example, when I try to enable/disable the web-interface I am asked to provide a password. This should not be the case on a factory-unlocked ATA. It appears as though Canada Computers sold me a locked ATA :mad:

Can anyone else verify if their Canada Computers purchased ATA is the same?

on your phone dial: ****
enter: 7932 #

at this point it may ask you for a password. If it does not, then your unit is factory unlocked. If it does, then you have the same problem as me.

I bought PAP2T from Canada Computers last week which seemed to be brand new in box but when I brought it home is when I realized it was open box. I just checked to make sure the serial number of the unit matched the box. Everything has been fine with my unit but could it be your unit was previously returned from a customer who applied a password to it?

[bridonca]

I've tried both G.711 and G.729a
Whenever the issue occurs, the outgoing voice is completely trashed with G.711a (robotic voice non stop).
With G.729a, a few seconds are trashed but the voice quality quickly resumes. I do not have the issue with outgoing phone calls which sort of rules out the isp throttling.

Sadly, it does not. Incoming and outgoing can and do use separate routes. As an example the outgoing server could be in Montreal, and the incoming DID actually from New York, and all US traffic is getting throttled because your ISP does not have a good peering agreement with the US. If anything is on Torix, you can pretty much abuse the bandwidth because it has the capability and is very low cost. US and World routes cost more and are the first to be throttled when there is too much volume.

[Lulz]

best Canadian company I have seen.
Really, when you think about the savings, it's just crazy comparing to what I am paying for Bell.

[apn64]

best Canadian company I have seen.
Really, when you think about the savings, it's just crazy comparing to what I am paying for Bell.

Very high on my list too, I ported to VoIP.ms ~18mo ago and have received numerous rate reduction notices.

Did anyone EVER see such a thing with the land/cable oligopoly? From my experience their prices only ratchet up.

[EPcjay]

Could you provide more details? Where can I set the waveform?

Thanks

It's an option in the PAPT2. It was in the regional settings, in the advanced view

[tdott]

No they don't, assuming that they have a standard Ringer Equivalence Number (REN) rating. If your phones are tied together today, odds are they'll work just fine on the ATA.

YES, this is the standard configuration that I expect that most of us are using.

YES again, since it's no different than what you probably have today.

The situation only changes if you do funky things like split the phones into groups for L1 and L2 respectively.


Awesome! Thank you for answering my questions.
Alot of ppl seem to think one phone needs to go directly into the ATA. I now know for sure that is not true.

[EPcjay]

for people who have successfully ported from Bell Landline.
What Service Provider Address did you specify in the request form?
The address that is on the payment stub of the bill (P.O. Box 9000 Stn. Don Mills) or some other one?

Thanks

That's exactly what I used on the bell bill
The address from the back of the bill.
I assume they do tons of these from bell, so they'll know where to send it anyways

[nosnoop]

FoIP is touchy. I'm pretty sure voip.ms doesn't have T.38 equipped yet (even as a passthrough) so I highly doubt any faxes coming in would make it over. There is limited success with ulaw and faxing, but it's not always consistent.
Rumor has it that voip.ms is getting t.38 but I have no idea what the timeline is. Best bet is to open a ticket to see if they support t.38 passthough.

Thanks for your info. In that case, maybe I'll look at CallCentric instead. Looks like they have fax to email too, and I don't even need OneSuite; though CallCentric has a disclaimer stating that fax reception is an option offered, but currently un-suppported.... whatever that means.

[Mykolas]

I am hoping to add a new physical line for both TekSavvy and Voip.ms. When I called Bell, they said the installation charge was $55 plus I am forced to sign up for Bell phone service for at least 1 month. Is there any way around this?

I don't mind paying the $55 at all, but you'd think that plus $8 a month dry loop for life would be sufficient tribute to the monopoly.

[jackboot]

I bought PAP2T from Canada Computers last week which seemed to be brand new in box but when I brought it home is when I realized it was open box. I just checked to make sure the serial number of the unit matched the box. Everything has been fine with my unit but could it be your unit was previously returned from a customer who applied a password to it?

I don't think it is possible for an end-user to apply an admin password. Does your unit have an admin password applied? To check, just pick up the phone and press **** and then 7932 # to see if it asks you for a password.

It looks to me like CC sold me a non-factory unlocked unit :mad:

[bridonca]

I am hoping to add a new physical line for both TekSavvy and Voip.ms. When I called Bell, they said the installation charge was $55 plus I am forced to sign up for Bell phone service for at least 1 month. Is there any way around this?

I don't mind paying the $55 at all, but you'd think that plus $8 a month dry loop for life would be sufficient tribute to the monopoly.

If you already have an internet connection, why do you need a land line? I am missing something here. You can have VOIP on the same line. You just need a good router to prioritize the traffic.

[robinto]

Ordered 2 DIDs but don't know where/how to assign them to Line 1 and Line 2 on the ATA.
TIA for any advice.

[jackie999]

I am hoping to add a new physical line for both TekSavvy and Voip.ms. When I called Bell, they said the installation charge was $55....

I'm missing something too...if you want teksavvy, why would you call bell?
Teksavvy is available via cable now in some areas...if you wanted to go that route then no there is no dry loop charge....I'll be switching my TSI DSL to TSI cable when it's in Aurora :)

[Pete Jones]

Ordered 2 DIDs but don't know where/how to assign them to Line 1 and Line 2 on the ATA.
TIA for any advice.

You associate the DID with an account or a sub-account at voip.ms

You then configure line1 or line2 on the ata to connect whatever account/sub account you want.

--Pete

[robinto]

Thank you for your quick response.

This is what I have done:

My voip.ms account:
123456

My 2 DIDs:
6471234567
6471235678

I created 2 sub-accounts:
123456_6471234567
123456_6471235678

On my LinksysPAP2:
Line 1 User ID:123456_6471234567
Line 2 User ID:123456_6471235678

No problem dialing out but getting busy signals when calling both 6471234567 & 6471235678.

[barrist]

Calling all voip experts..

I'm in the process of moving into a new house, and i signed up for voip.ms.

I'm waiting for our internet to be set up sometime next week.

I have a PAP2T that i wanted to connect to the existing wiring so i can still use all the jacks.

Here are pics of what's in my basement (its a townhouse, if that matters)

http://img693.imageshack.us/img693/8522/img1961iv.th.jpg (http://img693.imageshack.us/i/img1961iv.jpg/)
http://img14.imageshack.us/img14/2724/img1954z.th.jpg (http://img14.imageshack.us/i/img1954z.jpg/)
http://img188.imageshack.us/img188/6492/img1955h.th.jpg (http://img188.imageshack.us/i/img1955h.jpg/)
http://img535.imageshack.us/img535/1681/img1959c.th.jpg (http://img535.imageshack.us/i/img1959c.jpg/)
http://img8.imageshack.us/img8/2610/img1960zn.th.jpg (http://img8.imageshack.us/i/img1960zn.jpg/)
http://img695.imageshack.us/img695/6645/img1962a.th.jpg (http://img695.imageshack.us/i/img1962a.jpg/)
http://img249.imageshack.us/img249/8337/img1970g.th.jpg (http://img249.imageshack.us/i/img1970g.jpg/)
http://img42.imageshack.us/img42/6290/img1969nj.th.jpg (http://img42.imageshack.us/i/img1969nj.jpg/)


Any guidance would be helpful, thanks

[apn64]

If there's no POTS line, you could connect the ATA to the "Extension" port above the yellow sticker, but it would be better to split the Incoming vs. Outgoing ports on the telephone punch-down block.

White Box: right side = Bell line; left side = 5 telephone outlets in your house.

If it were my house, I'd cut the line from the grey Bell box to the white box and insert a pair of telephone terminal boxes, then you have flexibility to decide if you want POTS or the ATA to power your house phones.

http://img15.imageshack.us/img15/8461/phonesiv.jpg (http://img15.imageshack.us/i/phonesiv.jpg/)

[barrist]

If there's no POTS line, you could connect the ATA to the "Extension" port above the yellow sticker, but it would be better to split the Incoming vs. Outgoing ports on the telephone punch-down block.

White Box: right side = Bell line; left side = 5 telephone outlets in your house.

If it were my house, I'd cut the line from the grey Bell box to the white box and insert a pair of telephone terminal boxes, then you have flexibility to decide if you want POTS or the ATA to power your house phones.



Hi apn, thanks for the response. i also replied on another forum re: your suggestion of removing the cat5e connection:



so when you mean to remove the cat5e, its the punch down strip on the right side of the extension jack in the whtie box?


i think my other question is:

if i had my cable modem/router set up on another floor, how would i connect the pap2t to the internet? i do prefer the router near my desktop computer so that i have a wired connection. is there a way to connect the pap2t on another level through an existing jack?

Thanks in advance

[apn64]

Hopefully, there's a phone outlet in the room where the router is located.

If so, just connect the ATA to that outlet, and use a splitter (per "House Lines" above) if you also want a phone in that location.

If not, you need to find a location in your house where both phone and ethernet come together.

[jackie999]

I'm waiting for our internet to be set up sometime next week.


I'm no expert..but my question is...are you going cable or DSL? If you aren't using the POTS line at all wouldn't you just disconnect any bell lines coming in..then the internal wiring is yours. If, on the other hand you are going DSL, then you'll need to retain that copper feed....
Just my 2cents...

[barrist]

I'm no expert..but my question is...are you going cable or DSL? If you aren't using the POTS line at all wouldn't you just disconnect any bell lines coming in..then the internal wiring is yours. If, on the other hand you are going DSL, then you'll need to retain that copper feed....
Just my 2cents...

Cable

so no POTS line needed. i just wasn't sure WHAT to disconnect, as i'm pretty slow this this stuff :)

EDIT: And now i thought of another issue i may have.. the floor with the modem/router/ata would only have one cable outlet.. would a splitter for the modem and my digital tv boxes work? i tried this where i used to live, and my cable internet wouldn't work. but maybe it was just that particular line

[yatta]

Hey I just wanted to update that I have successfully ported my 416 Vbuzzer number to voip.ms It took less then 2 weeks. I could not be more satisfied so far. The Customer Support itself is worth the change. Easily changed the info on my pap2 and got me up and going in minutes. Man I can't believe Vbuzzer is still around with the lack of support, crap policy and shadiness they have been doing all this time. Again I was a long time customer 5-6 years with Vbuzzer but had enough, thanks to RFD for suggesting Voip.ms

any staff from Vbuzzer reading this, you should be a shamed of yourself!
Hope you get shut down, you do not deserve to be in business.

I got the dollar a month then premium per second calling. I will test this out and see if I need the unlimited incoming.

[apn64]

i thought of another issue i may have.. the floor with the modem/router/ata would only have one cable outlet.. would a splitter for the modem and my digital tv boxes work? i tried this where i used to live, and my cable internet wouldn't work. but maybe it was just that particular line

Passive splitter should be OK, but you can't use an active (e.g. amplifier) device, since they're typically one-way and kill the upstream link. You can combine passive/active as long as the latter is not in the path to the cable modem.

It's highly likely that like your phones, all your cable outlets are tied together in the basement. This is the most natural location for the cable modem, router and ATA. If you're accessing the router over a wireless link, you might save yourself some hassle by locating all the equipment in the basement.

Even if it's not a wireless link, the worst case might be to run a CAT5e from the basement to the location of your computer(s). Still, YMMV, based on access and topographical limitations.

[a2vr6]

Been with VOIp.MS for almost two months and have spent less then $8. lol...and I thought paying $22 a month with Vonage was a good deal. Great so far, knock on wood.

[ginko123]

hi all, I bought a PAP2NA from ebay back in 2008 and used it with vbuzzer and the pap2 died almost 3 months later :( That was from a seller in HK.

Is anyone in the same situation meaning most PAP2's bought outside of NA are like that?

If I buy one from Dell or Canada Computers, I'll have atleast 1 year of warranty albeit little expensive?

[apn64]

The chinese knock-off's available on eBay and elsewhere are legendary for early mortality failures. Best to spend the $55 on a genuine device from a reputable source.

CC is fine. I got mine at SwiftGamers in Markham; still running strong after 18mo.

[a2vr6]

hi all, I bought a PAP2NA from ebay back in 2008 and used it with vbuzzer and the pap2 died almost 3 months later :( That was from a seller in HK.

Is anyone in the same situation meaning most PAP2's bought outside of NA are like that?

If I buy one from Dell or Canada Computers, I'll have atleast 1 year of warranty albeit little expensive?

The problem with buying from an unknown source is the risk of a fake. Many people end paying twice for stuff by going this route, I've learned this lesson the hard way.

[jackboot]

hi all, I bought a PAP2NA from ebay back in 2008 and used it with vbuzzer and the pap2 died almost 3 months later :( That was from a seller in HK.

Is anyone in the same situation meaning most PAP2's bought outside of NA are like that?

If I buy one from Dell or Canada Computers, I'll have atleast 1 year of warranty albeit little expensive?

I bought mine from Canada Computers, and best I can tell, it is NOT factory unlocked. I have an admin password on mine that doesn't allow me to turn on/off the web interface.

[shadow_cruiser]

I don't think it is possible for an end-user to apply an admin password. Does your unit have an admin password applied? To check, just pick up the phone and press **** and then 7932 # to see if it asks you for a password.

It looks to me like CC sold me a non-factory unlocked unit :mad:

Pressing **** brings me to the configuration menu but after pressing 7932#, it's saying invalid option:confused:

[jackie999]

I bought mine from Canada Computers, and best I can tell, it is NOT factory unlocked. I have an admin password on mine that doesn't allow me to turn on/off the web interface.

I bought mine at CC, with the advice of people here and elsewhere, and it's fine. If there is a problem with yours why don't you return it? That's the benefit of buying from a store, although they charge a restocking fee, you can get your money back. I would, and have, recommended CC as a source for the PAP2T and haven't heard back from anyone that there is a problem.

I've been searching to see how/what the IVR function is for..since I set mine up thru the web interface I never looked into programming it thru IVR. Can you post a link of where you got your information on your "7932#" ...I don't want to attempt inputting it in my PAP2T unless I know what it's for :)

[shadow_cruiser]

This service has been very unreliable for me so far. Some days it's nice and clear and then the past two days, the person who I'm talking to gets cut off or my voice is very garbled. I'm using the Toronto due to issues with the Montreal server previously.

I edited my PAP2T settings using the guide from toao.net. Are you guys using G711u as your preferred codec only?. I'm thinking of changing to only using G711u to see if that would make difference.

I decided to ping all the servers today for latency. Obviously the farther you are from a server the more latency but since I'm in Gatineau QC, Montreal showed up with the least latency. Very surprising though that New York has less latency than Toronto

Montreal 23ms
New York 30ms
Toronto 43ms
Dallas 69ms
Los Angeles 89ms
London 125ms
Houston 225ms

[jackboot]

Pressing **** brings me to the configuration menu but after pressing 7932#, it's saying invalid option:confused:

That happened to me the first time too. I just hung up and tried again and it worked :confused:


I bought mine at CC, with the advice of people here and elsewhere, and it's fine. If there is a problem with yours why don't you return it? That's the benefit of buying from a store, although they charge a restocking fee, you can get your money back. I would, and have, recommended CC as a source for the PAP2T and haven't heard back from anyone that there is a problem.

I've been searching to see how/what the IVR function is for..since I set mine up thru the web interface I never looked into programming it thru IVR. Can you post a link of where you got your information on your "7932#" ...I don't want to attempt inputting it in my PAP2T unless I know what it's for :)

I bought my adapter from CC something like 10 months ago so I don't think they will take it back or exchange it at this point.

I didn't have a problem until I turned off the web interface (from within the web interface) while trying to solve a remote access problem. I then tried using the IVR to turn the web interface back on but it then asks me for a password, which from everything that I've read, indicates that the unit is previously from some VOIP provider that sells their own locked adapters.

You can read about the IVR in chapter 4 of the product manual, downloadable here:

http://www.nyct.net/PAP2T_ug.pdf

[Mazda323]

Do you guys know how to change the tone of the ring?
Now I'm not satisfied with my phone's ring. It's shrill, short and weak. :mad:

[jackie999]

You can read about the IVR in chapter 4 of the product manual, downloadable here:

http://www.nyct.net/PAP2T_ug.pdf

I read a few links on the IVR, one was that you must use a phone with no battery plugged in directly to Line 1 or 2 and have nothing plugged into the ethernet ...have you tried that? I find the web interface easy to use and wouldn't suggest anyone shut it down lest you do lock yourself out...just my 2cents.

[jackie999]

Do you guys know how to change the tone of the ring?
Now I'm not satisfied with my phone's ring. It's shrill, short and weak. :mad:
Did you read through this (http://www.toao.net/25-linksys-pap2t-voip-adapter-review) link..there are suggestions on how to set up your ring. My panasonic phone also allows me to assign a few ringtones..if your phone is capable.

[shadow_cruiser]

That happened to me the first time too. I just hung up and tried again and it worked :confused:

I tried about 10 times again and it's the same thing. I enter **** and it says "You have reach Sephora's configuration menu, please enter". When I start keying in the number, after 9 it says invalid entry. It doesn't even let me know finish keying in everything. Even if I do key in everything, it still says invalid entry. I hung up 10 times.

Should I be able to enter this menu if it's unlocked?. Is 7932 the right code?

[jackboot]

I tried about 10 times again and it's the same thing. I enter **** and it says "You have reach Sephora's configuration menu, please enter". When I start keying in the number, after 9 it says invalid entry. It doesn't even let me know finish keying in everything. Even if I do key in everything, it still says invalid entry. I hung up 10 times.

Should I be able to enter this menu if it's unlocked?. Is 7932 the right code?

What adapter are you using exactly (PAP2T-NA?) and what firmware (5.1.6?)

When I press **** on mine, I get the following voice prompt:

"Sipura configuration menu. Please enter option followed by the pound key or hang up to exit."

You are getting a different voice prompt which tells you to "press enter." So I'm assuming we have different adapters or different firmware.

EDIT: I also just figured out why my ATA is asking me for an admin password. Apparently if you set a user/admin password in the web interface, this same password is then used in the IVR to permit certain functions. I guess I wasn't expecting this since the password in the web interface is alphanumeric and I incorrectly assumed that any password being entered in the IVR would be numeric only. But any alpha characters in the password can be entered as indicated on your phone's number pad. So, mystery solved: Canada Computers is NOT selling bunk ATAs.

[jackie999]

I also just figured out why my ATA is asking me for an admin password. Apparently if you set a user/admin password in the web interface, this same password is then used in the IVR to permit certain functions.

Good to hear jackboot..glad you figured it out..raises a question I had though..is there a reason to set a password for the PAP2T? I didn't with mine since it's only here in my house and remembering another password is always a bother...it is connected to my WRT54GL which is password encrypted...

[shadow_cruiser]

What adapter are you using exactly (PAP2T-NA?) and what firmware (5.1.6?)

When I press **** on mine, I get the following voice prompt:

"Sipura configuration menu. Please enter option followed by the pound key or hang up to exit."

You are getting a different voice prompt which tells you to "press enter." So I'm assuming we have different adapters or different firmware.

EDIT: I also just figured out why my ATA is asking me for an admin password. Apparently if you set a user/admin password in the web interface, this same password is then used in the IVR to permit certain functions. I guess I wasn't expecting this since the password in the web interface is alphanumeric and I incorrectly assumed that any password being entered in the IVR would be numeric only. But any alpha characters in the password can be entered as indicated on your phone's number pad. So, mystery solved: Canada Computers is NOT selling bunk ATAs.

I guess I heard it wrong, it is Sipura, lol. I'm using the PAP2T with 5.16 firmware.

[jackboot]

Good to hear jackboot..glad you figured it out..raises a question I had though..is there a reason to set a password for the PAP2T? I didn't with mine since it's only here in my house and remembering another password is always a bother...it is connected to my WRT54GL which is password encrypted...

The only reason I have user/admin passwords on my adapter is because I have a port forwarded to my ATA so that I can access the web interface remotely in case I need to troubleshoot when I'm not at home. If your web interface is on port 80 and you do not have any external ports forwarded to it then there might not be a need to have any passwords set.


I guess I heard it wrong, it is Sipura, lol. I'm using the PAP2T with 5.16 firmware.

Hmm weird, that's the same as me. I wonder why we have different IVRs?

[EPcjay]

For those who ported their home phone to do, I have a question.

My port is due today, and now when i call my phone, it says the phone is not in service.

voip still doesn't say anything other then it should be ported today. The VOIP.ms is setup and can call out already, but it's not taking any calls.

Should i be waiting a day longer?

[yatta]

For those who ported their home phone to do, I have a question.

My port is due today, and now when i call my phone, it says the phone is not in service.

voip still doesn't say anything other then it should be ported today. The VOIP.ms is setup and can call out already, but it's not taking any calls.

Should i be waiting a day longer?


give it a few more hours wait to you get your confirmation email that it is finalized

[jackie999]

For those who ported their home phone to do, I have a question.

My port is due today, and now when i call my phone, it says the phone is not in service.

voip still doesn't say anything other then it should be ported today. The VOIP.ms is setup and can call out already, but it's not taking any calls.

Should i be waiting a day longer?

Did you go into your voip.ms DID management area and set it all up..routing, dialout time, fallover, voicemail set up.... etc?

[Kirbyprime]

Has anyone used the USB-B2K with voip? CC has a variant of it.

http://www.canadacomputers.com/product_info.php?cPath=30_414&item_id=014408&sid=onsbfohb26k4itko8j675f4hg7

cheaper than the PAP2T but not sure how well it handles since reviews on it are scarce.

[Mykolas]

If you already have an internet connection, why do you need a land line? I am missing something here. You can have VOIP on the same line. You just need a good router to prioritize the traffic.

Just to clarify, my goal was to have two DSL connections. One is already set up, but the second is not. As the house is 50+ years old, I doubt it is pre-wired for two line capability, so I called Bell to ask about their $55 line install fee.

When I learned they also want to put a minimum of one month's phone service on the new line, I decided to ask here to see if there was a way to keep the price down.

[bridonca]

Has anyone used the USB-B2K with voip? CC has a variant of it.

http://www.canadacomputers.com/product_info.php?cPath=30_414&item_id=014408&sid=onsbfohb26k4itko8j675f4hg7

cheaper than the PAP2T but not sure how well it handles since reviews on it are scarce.

The B2K is awesome for Skype. Rock solid driver, a dream to use with Skype. For SIP, you need to use Counterpath Xlite . The latest driver is still buggy. Counterpath does NOT officially support the B2K

I would personally get the PAP2 instead. Unlike the B2K, you do not need a computer to be on to make or receive a phone call, which is nice. Only issue is when it comes to configuring it, but once that is done, it is pretty much trouble free, and acts like a phone!

[bridonca]

Just to clarify, my goal was to have two DSL connections. One is already set up, but the second is not. As the house is 50+ years old, I doubt it is pre-wired for two line capability, so I called Bell to ask about their $55 line install fee.

When I learned they also want to put a minimum of one month's phone service on the new line, I decided to ask here to see if there was a way to keep the price down.

I am sorry for sounding thick, but why would need 2 DSL connections? Other than a potential bandwidth increase, I see little use for another DSL connection.

You can get a SIP VOIP ATA device like a PAP2 which you can connect to your DSL modem, and that ATA can provide you with 2 extra telephone lines if you use a SIP VOIP provider like voip.ms When you use the PAP2, your phone calls will go through your DSL data connection. So there would be no need to call Bell to get more phone lines.

[death_hawk]

Speed and Bandwidth increase. Running dual lines (without bonding) doesn't do much for a single app or a single treaded (network wise) application like downloading via browser. However, for something like "large water" it can GREATLY benefit from multiple lines.

If your ISP doesn't suck, you could also bond it for double the speed on everything.

[tdott]

I just got everything setup and working! Incoming and outgoing calls work great.

I do have an issue with my ADT Home Alarm system.
They say they are not receiving the signal.

Currently i have the Linksys PAP2T hooked up to my demarcation point in the basement. (Exactly like how my Rogers Home Phone was setup)
Alarm system is connected exactly the same as it was before.

All the phones work in the house as they were before.

Any tips/suggestions on what i can do to troubleshoot n hopefully get the Home Alarm to work, otherwise i wont be able to cancel my rogers home phone.

When i look into my Call Detail Records on voip.ms I can see the alarm system is dailing out to it's 1800 number, and the durations range from 11s - 25s. But ADT is still saying they haven't received a signal.

Any help would be appreciated, hopefully i can get it working.

TIA

[luckydog2006]

I just got everything setup and working! Incoming and outgoing calls work great.

I do have an issue with my ADT Home Alarm system.
They say they are not receiving the signal.

Currently i have the Linksys PAP2T hooked up to my demarcation point in the basement. (Exactly like how my Rogers Home Phone was setup)
Alarm system is connected exactly the same as it was before.

All the phones work in the house as they were before.

Any tips/suggestions on what i can do to troubleshoot n hopefully get the Home Alarm to work, otherwise i wont be able to cancel my rogers home phone.

When i look into my Call Detail Records on voip.ms I can see the alarm system is dailing out to it's 1800 number, and the durations range from 11s - 25s. But ADT is still saying they haven't received a signal.

Any help would be appreciated, hopefully i can get it working.

TIA


I don't think you will be able to get your ADT working. You should research next time. Its been discussed many times that voip does not work well with alarm companies.

[AAA]

Do you port your phone # over to voip.ms right away or get a DID number first and port later? I read somewhere that we need to give bell 30 days notice before porting to avoid some $35 fee, is it true?

Another question, does you pap2t get hot?

[bridonca]

Do you port your phone # over to voip.ms right away or get a DID number first and port later? I read somewhere that we need to give bell 30 days notice before porting to avoid some $35 fee, is it true?

Another question, does you pap2t get hot?

You get a number assigned from voip.ms, and then you port your old number.

I cannot give you a good answer about getting charged $35 by Bell. My guess is that if you cancel, you have to give a month's notice, but if you port over, there is no penalty. I could be easily wrong about that though.

[yatta]

I didn't order any DID i just requested a port from my current provider (at the time) you have to pay a 25 dollar fee for porting the number to voip.ms so I can't see you getting dinged $25 and $35 from bell, I can see that you do need to give 30 days notice.. get on it now!

[Phils]

Do you port your phone # over to voip.ms right away or get a DID number first and port later? I read somewhere that we need to give bell 30 days notice before porting to avoid some $35 fee, is it true?

Another question, does you pap2t get hot?

The easy question first. My PAP2T gets slightly warm like any electronic device would. Apparently the cheap knockoffs from eBay do get pretty hot so buy it from a reputable company. A lot of people buy theirs from Canada Computers. I bought mine at Dell.ca.

I ported my phone line over right away. It took about 10 days from start to finish. Looking back, what I should have done is open the account; test it for a few days without a DID number; and after debugging then port the number over. Even without a DID, you can still make outgoing calls. That way you can back off if you have technical issues.

In my case, I had compatibility issues with my router that took me many hours to resolve. I was at risk for awhile of having ported over my DID to a system that wasn't going to work.

[tdott]

I don't think you will be able to get your ADT working. You should research next time. Its been discussed many times that voip does not work well with alarm companies.

I have, and there are some people using it.

[jackie999]

....I do have an issue with my ADT Home Alarm system.
They say they are not receiving the signal....

I can't answer your question..but have you looked over in the dslreport/voip forums... http://www.dslreports.com/forum/voip ...

[tdott]

I can't answer your question..but have you looked over in the dslreport/voip forums... http://www.dslreports.com/forum/voip ...

Thanks for the link, i'll continue searching but my initial results show old replies 2007 and older. Since VOIP has changed alot since then, i dont think any of it applies.

[mastermc]

Are locals calls free if I have a DID in the same area code?? I thought I read in one of the posts that it is, but I still get charged .005 a minute for local calls.

[plg_cp]

Are locals calls free if I have a DID in the same area code?? I thought I read in one of the posts that it is, but I still get charged .005 a minute for local calls.

No, no calls are free; your local area code is irrelevant. The termination fee applies regardless of where you are originating your call

[djino]

Are locals calls free if I have a DID in the same area code?? I thought I read in one of the posts that it is, but I still get charged .005 a minute for local calls.

Toll-free numbers are free (1-800, 1-866, 1-877, 1-888). Other than that you have to pay for every termination call. If its within Canada, then its $0.0052 / minute. $0.011/minute for U.S. termination

djino

[AAA]

The easy question first. My PAP2T gets slightly warm like any electronic device would. Apparently the cheap knockoffs from eBay do get pretty hot so buy it from a reputable company. A lot of people buy theirs from Canada Computers. I bought mine at Dell.ca.

I ported my phone line over right away. It took about 10 days from start to finish. Looking back, what I should have done is open the account; test it for a few days without a DID number; and after debugging then port the number over. Even without a DID, you can still make outgoing calls. That way you can back off if you have technical issues.

In my case, I had compatibility issues with my router that took me many hours to resolve. I was at risk for awhile of having ported over my DID to a system that wasn't going to work.

I got the pap2t from dell.ca as well, it gets very very warm after just a few hours while I was reading up some documents. Good suggestions about testing it without a DID first. I was thinking about getting a DID number to test for couple days before porting.


I didn't order any DID i just requested a port from my current provider (at the time) you have to pay a 25 dollar fee for porting the number to voip.ms so I can't see you getting dinged $25 and $35 from bell, I can see that you do need to give 30 days notice.. get on it now!

Do you think voip.ms collects the $25 porting fee on behalf of bell??


You get a number assigned from voip.ms, and then you port your old number.

I cannot give you a good answer about getting charged $35 by Bell. My guess is that if you cancel, you have to give a month's notice, but if you port over, there is no penalty. I could be easily wrong about that though.

We can't give bell 30 days notice to cancel because we can't time the porting time. Anyone porting a phone number over from bell without giving them a 30-day notice got dinged (or not) for $35 by bell?

[AAA]

Toll-free numbers are free (1-800, 1-866, 1-877, 1-888). Other than that you have to pay for every termination call. If its within Canada, then its $0.0052 / minute. $0.011/minute for U.S. termination

djino

Nice to know that toll free numbers are free!! Does it apply to USA toll free numbers too?

[djino]

Nice to know that toll free numbers are free!! Does it apply to USA toll free numbers too?

yes.

djino

[mastermc]

I keep on losing registration on my pap2, anything special settings I need to change? I noticed if I change the SIP port to either 5060 or 5061, it will work right away. I dont always want to be doing that.

[Sneaks]

We can't give bell 30 days notice to cancel because we can't time the porting time. Anyone porting a phone number over from bell without giving them a 30-day notice got dinged (or not) for $35 by bell?

I didn't notify bell, it occurred in less than 30 days, and didn't get any ding, but as per usual YMMV. That said I didn't move my DSL line for another 2 weeks after, so I don't know if that fell into/counts in the 30 days or not. Like most I suspect, I'm notoriously bad though for fully reading the changes every time companies update their terms, etc. so possible I was ok, where as others have different legaleze.

As I was moving, I didn't have any options with Bell that would keep my #, so I had to port somewhere, so figured, I'd pay whatever I had to. Fortunately my experience was straightforward and great.

[jackie999]

I keep on losing registration on my pap2, anything special settings I need to change? I noticed if I change the SIP port to either 5060 or 5061, it will work right away. I dont always want to be doing that.

Did you set the server to the IP? If no, look to post #400 in this thread for more info...

[mastermc]

Did you set the server to the IP? If no, look to post #400 in this thread for more info...

Yes it is set with the IP of the server.

[tdott]

I don't think you will be able to get your ADT working. You should research next time. Its been discussed many times that voip does not work well with alarm companies.


I got my PAP2T working with my ADT alarm system!!

I'm so happy, did about 5 tests today, just got off the phone with ADT after I set it off for real this time. All others were manual tests and they received all the signals.

I had to play around with it, changed a few things like the dialing plan, did a few other recommendations but i eventually got it working!

Screw the haters! and everyone telling me it couldn't be done without a reasonable explanation, because i knew technically there was nothing stopping it from working, except a few artificial barriers.

I know this is not the preferred approach and it will be less reliable, I do plan to put everything on a good UPS for power failures.

Now time to call up ROGERS to CANCEL my home phone.....
I'll probably wait and do a few more tests, but things are looking good.:-0

[apn64]

Now time to call up ROGERS to CANCEL my home phone.....


Best call I ever made ;)

Congrats on getting it working w/ your alarm.

[tdott]

Best call I ever made ;)

Congrats on getting it working w/ your alarm.

Thanks for your help!

[luckydog2006]

I got my PAP2T working with my ADT alarm system!!

I'm so happy, did about 5 tests today, just got off the phone with ADT after I set it off for real this time. All others were manual tests and they received all the signals.

I had to play around with it, changed a few things like the dialing plan, did a few other recommendations but i eventually got it working!

Screw the haters! and everyone telling me it couldn't be done without a reasonable explanation, because i knew technically there was nothing stopping it from working, except a few artificial barriers.

I know this is not the preferred approach and it will be less reliable, I do plan to put everything on a good UPS for power failures.

Now time to call up ROGERS to CANCEL my home phone.....
I'll probably wait and do a few more tests, but things are looking good.:-0


tdott,

As much as I'm happy you got this working, I really hope it works out for you. Your PAP2 device will occasionally de-register itself. Not sure what your alarm company will do then when your voip goes down.

When ADT does not get a phone line link, it should trigger an alarm. So, when your voip de-registers (its happen to many voip users), this will trigger your alarm to try to call you or the police if they cannot reach you.

If it does not for you, then great!

[tdott]

tdott,

When ADT does not get a phone line link, it should trigger an alarm. So, when your voip de-registers (its happen to many voip users), this will trigger your alarm to try to call you or the police if they cannot reach you.

If it does not for you, then great!

Not sure where you get your info from, but my rogers home phone was disconnected for a week and no such thing happened, nor did ADT say they monitor for this when i asked for a list of communications from my system. I think this is a myth, because when the line is cut no alarm system monitors that unless there is a cell backup, in which case they aren't monitoring for it they are just using the backup line.

To address your de-registering concern, there is a setting that forces the PAP2T to register more frequently, voip.ms advised me to change this to 200

Basically there would only be a few minutes before it re-registers.

Chat log below:


[Ray] Also, in your PAP2T, under SIP tab, RTP Parameters, RTP Port Min: 10000 and RTP Port Max: 20000. You can go to Line 1 settings and set register expire from 3600 to 200. These settings will help for not losing your registration.

Anyways thanks for your concern, I'm sure it may happen, I will monitor it over the next week and I will know exactly what numbers and when it dials out to the monitoring center.

[houstondallas]

I got my PAP2T working with my ADT alarm system!!

I'm so happy, did about 5 tests today, just got off the phone with ADT after I set it off for real this time. All others were manual tests and they received all the signals.

I had to play around with it, changed a few things like the dialing plan, did a few other recommendations but i eventually got it working!

Screw the haters! and everyone telling me it couldn't be done without a reasonable explanation, because i knew technically there was nothing stopping it from working, except a few artificial barriers.

I know this is not the preferred approach and it will be less reliable, I do plan to put everything on a good UPS for power failures.

Now time to call up ROGERS to CANCEL my home phone.....
I'll probably wait and do a few more tests, but things are looking good.:-0

What did you change? Can you post your dialing plan?

houston

[signup]

To address your de-registering concern, there is a setting that forces the PAP2T to register more frequently, voip.ms advised me to change this to 200



How long you've been using this value? I have same problem in my PAP2T. I would try this value if it really helps. I thougt my router or ISP has a problem.

[EPcjay]

I keep on losing registration on my pap2, anything special settings I need to change? I noticed if I change the SIP port to either 5060 or 5061, it will work right away. I dont always want to be doing that.

+1 for me

I keep losing registration, but my subaccount is god. It doesn't lose at all. My subaccount is point to newyork though. I wonder if that makes a difference.

I haven't tried the IP yet. Will try that when i get home.

[mastermc]

+1 for me

I keep losing registration, but my subaccount is god. It doesn't lose at all. My subaccount is point to newyork though. I wonder if that makes a difference.

I haven't tried the IP yet. Will try that when i get home.

I fixed my problem with lost registration by:

Changing Register Expires to 120 on PAP2, and
on my router(tomato) change UDP Timeout- Unreplied to 10 seconds

[jackboot]

New Server Locations

June 23, 2010, 4:56 pm

We're happy to announce that we've added 4 new server locations to our network. The new points of presence (aka POP) are located in the following cities: Atlanta, Chicago, Tampa and Seattle.

Please note that the new locations are currently in "BETA" status. They will remain in beta state until Wednesday, June 30th. If you're in a critical production environment, we do NOT recommend using one of the new locations until next week when the beta flag is lifted officially. If you're willing to test the new servers and find any anomalies, please send us an email to support and we'll correct it ASAP.

We hope the new locations help our customers obtain better connectivity with our network by reducing potential latency, jitter and packet loss. Our new locations also offer extra redundancy in case of a specific data center going down.



I did a ping comparison between the new Seattle server and the Toronto server from a computer located in Calgary:

Toronto: 53ms
Seattle: 45ms

The difference in latency is pretty minor in this case, but it will probably be very significant for some other users. I assume jitter and packet loss will be improved too, but I'm too dumb to know how to test for that. In any case, its safe to say that for those with ATAs in Alberta and BC, it looks like Seattle is going to be a better option now.

I'm curious if this will make my fax service a bit more reliable.

[4public]

I cannot connect to www.voip.ms this morning
Are they down?

[paulo500]

I cannot connect to www.voip.ms this morning
Are they down?Working fine for me. :)

[tdott]

What did you change? Can you post your dialing plan?


I changed a few things, not exactly sure what did it.

Dialing plan is:
( [23456789]11 | *xxx. | <:1>[2-9]xx[2-9]xxxxxxS0 | 1[2-9]xx[2-9]xxxxxxS0 | 011xxxxxxx. | [*x][*x][*x][*x]. )

I found with this plan it only takes about 4-5s to dail the number, wheraas before it was taking 10-12s, that could have been causing a diff.


How long you've been using this value? I have same problem in my PAP2T. I would try this value if it really helps. I thougt my router or ISP has a problem.

It's been a week and no issues with losing registration.
I am also using the ip for the proxy, i hear this helps keep the registration also.

[houstondallas]

I changed a few things, not exactly sure what did it.

Dialing plan is:
( [23456789]11 | *xxx. | <:1>[2-9]xx[2-9]xxxxxxS0 | 1[2-9]xx[2-9]xxxxxxS0 | 011xxxxxxx. | [*x][*x][*x][*x]. )

I found with this plan it only takes about 4-5s to dail the number, wheraas before it was taking 10-12s, that could have been causing a diff.



It's been a week and no issues with losing registration.
I am also using the ip for the proxy, i hear this helps keep the registration also.

Thanks for the info, I'll have to test it out with my ADT alarm.

houston

[duckdown]

How many simultaneous lines can I have and calls can I receive at once?

Can I buy one of these 4 line CISCO phones for my home business and make/receive 4 calls simultaneously? ie: 4 clients call at once and i can put them on hold while i take care of them in order, etc?

How many max simultaneous lines can I have?

Thanks again!

[djino]

How many simultaneous lines can I have and calls can I receive at once?

Can I buy one of these 4 line CISCO phones for my home business and make/receive 4 calls simultaneously? ie: 4 clients call at once and i can put them on hold while i take care of them in order, etc?

How many max simultaneous lines can I have?

Thanks again!

Unlimited Channels for Termination.
For Incoming DID, its Unlimited channels on the pay per use option and only 2 channels if you have the unlimited option for the DID.

djino

[Rekognize]

Just put in my request to port my number from Rogers homephone?

How long does it take and when can I call and cancel my Rogers homephone? or is that done automatically?

[djino]

Just put in my request to port my number from Rogers homephone?

How long does it take and when can I call and cancel my Rogers homephone? or is that done automatically?

It should take about 1-2 weeks. Once complete, it will automatically cancel your Homephone line. No need to call Rogers.

djino

[tdott]

I troubleshooted the issue for quite a while and finally, I realized one thing: the pap2t-na is 10mbps half-duplex only (at least my unit) and my wrt54gl is apparently 10/100 full-duplex only. I decided to plug a switch in between that would negotiate to all speeds and the problem went away.

Not sure how many peeps ran into a similar situation but I guess the lack of half-duplex and flow control caused the issue.

I have the exact same router but with Tomato firmware. I haven't had a single issue, the wife couldn't even notice the difference she kept asking if I still had the Rogers hooked up.

What firmware were you running?
I also have a very sable Internet connection and don't do alot of downloading.

[jackie999]

I have the exact same router but with Tomato firmware. I haven't had a single issue.....

I agree...quality is just great. Sometimes there is a faraway echo, but generally that's when I'm talking to someone that is using a cell phone - and I would expect less than perfect quality then, and I wouldn't blame that on voip.ms in any case.

[death_hawk]

How many simultaneous lines can I have and calls can I receive at once?

Can I buy one of these 4 line CISCO phones for my home business and make/receive 4 calls simultaneously? ie: 4 clients call at once and i can put them on hold while i take care of them in order, etc?

How many max simultaneous lines can I have?

Thanks again!

You're capped at 25 channels until you call (ticket) in and request more.
If you really have this many simultaneous calls, I'd consider putting up a PBX with an IVR and call queue. It may eliminate some call volume in a good way. (IE asking what time you're open etc)

But yeah, this will work fine.

[death_hawk]

Oh and for those having SIP deregistration problems, it's your local firewall that kills the connection.

Either set your SIP device to reregister faster, or tell your firewall to keep open connections longer.
If your SIP device doesn't support faster registrations, either change SIP devices, your firewall or stick a PBX with a custom reregister time in.

EDIT:
Protip: Look for a "NAT" setting in your SIP configs.

EDIT2:
After fixing my reregistration times, I haven't had a permanent deregister or a rejected incoming call.

[shadow_cruiser]

I changed a few things, not exactly sure what did it.

Dialing plan is:
( [23456789]11 | *xxx. | <:1>[2-9]xx[2-9]xxxxxxS0 | 1[2-9]xx[2-9]xxxxxxS0 | 011xxxxxxx. | [*x][*x][*x][*x]. )

I found with this plan it only takes about 4-5s to dail the number, wheraas before it was taking 10-12s, that could have been causing a diff.



It's been a week and no issues with losing registration.
I am also using the ip for the proxy, i hear this helps keep the registration also.

I tried this dialing plan but I'm getting a fast beep after I dial a number. Anyone know what's up?

I also live in a apartment and the maintenance number is 310-MAXX. When I call it, a message plays "This number cannot be reached from your dialing area". I called the building office and they tested it, everything is fine for them. The number cannot be dialed with an area code. If someone can help me figure this out, it would be appreciated. My area code is 819

[Chaseme]

Do you guys know if theres a way to call direct between PAP2Ts? In other words can I call a friend who also has VOIP.ms on his PAP2T directly without any level of billing?

Or is it just with the Virtual SIP method from VOIP.ms?

[shadow_cruiser]

Whenever my PAP2T loses power, whether from a power outage or or simply unplugging it, it has a hard time connecting to the voip server after start up. Does anyone else have this problem?

[jackie999]

New servers launched :)

We're glad to announce that we've added 4 new server locations to our network. The new points of presence (aka POP) are
located in the following cities: Atlanta, Chicago, Tampa and Seattle.

The hostname of the new locations are the following:

atlanta.voip.ms
chicago.voip.ms
seattle.voip.ms
tampa.voip.ms

If you decide to switch to a new server, don't forget to update the Point of Presence of your DID Numbers.

We hope the new locations help our customers obtain better connectivity with our network by reducing potential latency, jitter and packet loss. Our new locations also offer extra redundancy in case of a specific data center going down.

-------------------------------------------

New Feature: Toll-Free Termination Carrier routing

Additionally, we've added a new setting: Toll-Free Carrier. By default, when you dial a toll-free number using an American server, your call will be routed through an American Carrier, and when you dial a Toll-Free number via a Canadian server, your call is routed through a Canadian Carrier. You can override this setting for a carrier of your choice.

For example, if you are based in Vancouver and wish to use an American server closer to your location, such as the new Seattle POP, you can choose to have your toll-free calls routed through a Canadian Carrier.

The setting is available via the menu option "Main menu --> Settings" in the customer portal.

We welcome all your comments and suggestions regarding the new servers and the new toll-free routing feature.

Sincerely,

VoIP.ms Team

[jackie999]

Whenever my PAP2T loses power, whether from a power outage or or simply unplugging it, it has a hard time connecting to the voip server after start up. Does anyone else have this problem?

Not sure what you mean by "hard time" ...is it sweating and panting? I've noticed mine may take up to 2 minutes for the line light to come back on solid....

[shadow_cruiser]

Not sure what you mean by "hard time" ...is it sweating and panting? I've noticed mine may take up to 2 minutes for the line light to come back on solid....

Sorry, bad wording. It doesn't connect upon boot-up but as soon as I change to the Toronto server, everything is fine. I get better clarity with the Montreal server but better stability with the Toronto server. Why can't we have both? lol.

Can anyone help me with my previous post?.

"I also live in a apartment and the maintenance number is 310-MAXX. When I call it, a message plays "This number cannot be reached from your dialing area". I called the building office and they tested it, everything is fine for them. The number cannot be dialed with an area code. If someone can help me figure this out, it would be appreciated. My area code is 819". Basically I need to adjust the dial plan so I can dial out to numbers without area codes.

My dial plan is (<:1613>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|*xx.|[34689]11|822|4443|4747|0|00|[2-9]xxxxxx|1 [2-9]xx[2-9]xxxxxxS0|4XXX|xxxxxxxxxxxx.)

[WildChild]

Do you happen to have a WRT54G with Tomato WRT? If so, it's probably related UDP timeouts.

About 310-MAXX, your best guess would be to contact directly voip.ms. They probably would know more about the problem than people here.


Sorry, bad wording. It doesn't connect upon boot-up but as soon as I change to the Toronto server, everything is fine. I get better clarity with the Montreal server but better stability with the Toronto server. Why can't we have both? lol.

Can anyone help me with my previous post?.

"I also live in a apartment and the maintenance number is 310-MAXX. When I call it, a message plays "This number cannot be reached from your dialing area". I called the building office and they tested it, everything is fine for them. The number cannot be dialed with an area code. If someone can help me figure this out, it would be appreciated. My area code is 819". Basically I need to adjust the dial plan so I can dial out to numbers without area codes.

My dial plan is (<:1613>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|*xx.|[34689]11|822|4443|4747|0|00|[2-9]xxxxxx|1 [2-9]xx[2-9]xxxxxxS0|4XXX|xxxxxxxxxxxx.)

[Politikil]

If your area code is 819 and you have 10-digit dialing, then you will need to dial 819-310-MAXX. Your dialing plan is set to allow local 7-digit calling in the 613 area code (which is why the 613 is in there). So if you are dialing 310-MAXX from the 819 area code, then you are really calling 613-310-MAXX and getting the error as you would need to dial 1-613-310-MAXX. If you want to have 7-digit local dialing for the 819 area code, then change the '613' in your dial plan to '819' and I imagine you would be able to do what you need.


Sorry, bad wording. It doesn't connect upon boot-up but as soon as I change to the Toronto server, everything is fine. I get better clarity with the Montreal server but better stability with the Toronto server. Why can't we have both? lol.

Can anyone help me with my previous post?.

"I also live in a apartment and the maintenance number is 310-MAXX. When I call it, a message plays "This number cannot be reached from your dialing area". I called the building office and they tested it, everything is fine for them. The number cannot be dialed with an area code. If someone can help me figure this out, it would be appreciated. My area code is 819". Basically I need to adjust the dial plan so I can dial out to numbers without area codes.

My dial plan is (<:1613>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|*xx.|[34689]11|822|4443|4747|0|00|[2-9]xxxxxx|1 [2-9]xx[2-9]xxxxxxS0|4XXX|xxxxxxxxxxxx.)

[shadow_cruiser]

Do you happen to have a WRT54G with Tomato WRT? If so, it's probably related UDP timeouts.

About 310-MAXX, your best guess would be to contact directly voip.ms. They probably would know more about the problem than people here.

Thanks a bunch for the great info!!. I do have a WRT54GL with Tomato firmware. What a huge headache this has solved:). A simple google pointed me to this thread about this very same issue http://www.dslreports.com/forum/r23845317-Other-Loosing-Registration-Tomato-VOIPMS-PAP2T-NA. I haven't actually tested it yet but it seems this would do the trick.


If your area code is 819 and you have 10-digit dialing, then you will need to dial 819-310-MAXX. Your dialing plan is set to allow local 7-digit calling in the 613 area code (which is why the 613 is in there). So if you are dialing 310-MAXX from the 819 area code, then you are really calling 613-310-MAXX and getting the error as you would need to dial 1-613-310-MAXX. If you want to have 7-digit local dialing for the 819 area code, then change the '613' in your dial plan to '819' and I imagine you would be able to do what you need.

310-MAXX is a maintanance company outsourced by Oxford Properties. They are all over Canada and the number for all Canadian residents is 310-MAXX. You cannot dial this number with an area code otherwise it will not work. I called them using my cell number which is a 647 area code and it worked perfectly. Even changing the area code from 613 to 819 does not work. It definitely has to be something with the dialing plan. I probably will give voip.ms a call tomorrow and let you guys know the outcome, in case anyone else has this problem in the future.

[jackie999]

310-MAXX is a maintanance company outsourced by Oxford Properties. They are all over Canada and the number for all Canadian residents is 310-MAXX.

First of all, I don't have an answer ..but your post reminded me of Bells help number " 310-SURF " ...I did a quick google and found this old bit of info from a vontage forum

So far as 310 numbers goes, you cannot dial them with an area code in front. These numbers are actually shortcuts to the real phone number of the company. When you dial 310-xxxx, the phone company converts what you dialled to 999-888-7777 where 999 is the area code, 888 is the exchange, and 7777 is the number. This is why Vonage and other companies don't offer the service. This would mean they would have to know where you are located, and program all of the 310 numbers for your local area and their actual conversion. What a mess that would be.

It would be interesting to see what voip.ms support says..if this is still the case..if so, you can't dial it. One post said you can still call the number from a disconnected phone line...doesn't make sense to me..but give it a try...

[shadow_cruiser]

First of all, I don't have an answer ..but your post reminded me of Bells help number " 310-SURF " ...I did a quick google and found this old bit of info from a vontage forum


It would be interesting to see what voip.ms support says..if this is still the case..if so, you can't dial it. One post said you can still call the number from a disconnected phone line...doesn't make sense to me..but give it a try...

From chat support this morning


[Stephen] Unfortunately this number is not supported by our system, as it need to have an area code, otherwise our dial rule will not be able to perform the call, unfortunately there's nothing we can do about this

[jackie999]

Since these 310-xxxx numbers are short cuts, and with more and more of us using voip...perhaps you can get a real number to call. Companies must pay for this 310 service...I can think of a pizza joint uses the same 310 promotion...

[jackboot]

When I read your post about 310-MAXX I thought it was strange because I know that I have dialed 310 numbers before (like Panago :)). So I tried dialing Panago and it worked no problem. I then tried dialing 310-MAXX and it also worked no problem.

So I don't know what the difference is between my setup and yours, but I can dial 310 numbers without a problem it seems.

[shadow_cruiser]

Since these 310-xxxx numbers are short cuts, and with more and more of us using voip...perhaps you can get a real number to call. Companies must pay for this 310 service...I can think of a pizza joint uses the same 310 promotion...
I asked them today and they said this is the only number I can call. Just like with 1-877-ROGERS1, I don't think there's a 416 number for Rogers customer service.


When I read your post about 310-MAXX I thought it was strange because I know that I have dialed 310 numbers before (like Panago :)). So I tried dialing Panago and it worked no problem. I then tried dialing 310-MAXX and it also worked no problem.

So I don't know what the difference is between my setup and yours, but I can dial 310 numbers without a problem it seems.

Do you mind posting your dial plan?

[jackboot]

Do you mind posting your dial plan?

Sure, here it is:


(<:1647>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|*xx.|[34689]11|822|4443|4747|0|00|[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|4XXX|xxxxxxxxxxxx.)

[shadow_cruiser]

Sure, here it is:

Thanks. My dial plan is exactly the same. Still getting the "This number cannot be reached from your area". I'm not sure how you are getting through:confused::confused:

Maybe there's a setting in the PAP2T your using, that I'm not?

[jackboot]

Thanks. My dial plan is exactly the same. Still getting the "This number cannot be reached from your area". I'm not sure how you are getting through:confused::confused:

Maybe there's a setting in the PAP2T your using, that I'm not?

When you say "exactly", do you mean that you also have a default 647 area code?

[jackie999]

I tried 310-maxx, 310-bell, 310-surf and 310-1010 and got the same error that shadow_cruiser got "The number dialed cannot be reached from your area code..."

[shadow_cruiser]

When you say "exactly", do you mean that you also have a default 647 area code?

Despite the area code, everything else is the same. I did change my area code to 647 so I could see what would happen since I have 3 DID's with area codes 819,613 and 647. I got the same message.

[jackboot]

Despite the area code, everything else is the same. I did change my area code to 647 so I could see what would happen since I have 3 DID's with area codes 819,613 and 647. I got the same message.

All of those numbers work just fine for me!

What server are you using? I'm using the Toronto server.

[mastermc]

All of those numbers work just fine for me!

What server are you using? I'm using the Toronto server.


+1 for me too. using Toronto server and it works too.

[Rekognize]

My port was completed today but the number is out of service when I call it, any ideas?

I also enabled e911, callerid lookup, and voicemail. Using the Toronto server..

[djino]

My port was completed today but the number is out of service when I call it, any ideas?

I also enabled e911, callerid lookup, and voicemail. Using the Toronto server..

Did you setup the Ported DID to ring your voip.ms credentials?

djino

[Rekognize]

Did you setup the Ported DID to ring your voip.ms credentials?

djino

Thanks, fixed it by changing the Device type to ATA/IP Phone under Manage DID Numbers :D

It's pretty awesome, everything works and the voice is crystal clear.

[duckdown]

Can I hook up a faxmodem to my PC and use fax over the voip.ms service?

[shadow_cruiser]

All of those numbers work just fine for me!

What server are you using? I'm using the Toronto server.


+1 for me too. using Toronto server and it works too.

I was using the Toronto server but switched to Montreal. I think I have given up trying to figure this out, lol. Thanks everyone for their help.

It's nothing to do with the dialing plan, I guess maybe a setting in the PAP2T that's different.

[Rekognize]

I now have more control against telemarketers, I forward their calls to the FCC :lol:

[djino]

I now have more control against telemarketers, I forward their calls to the FCC :lol:

lol... how did you do that?

djino

[Mendes]

lol... how did you do that?

djino

under DID numbers choose caller id filtering

CallerID Filtering is a tool that let you create incoming routing rules according to incoming CallerID number. You can apply different rules to some or all of your DID(s) numbers. You can create as many filters as you want. Each filter you create can filter according to one the following criterias and change the routing if there's a match.

And rather then send them to the FCC just create a funny recording of you saying Hello then going off in a random tangent and send them to the recording

[djino]

under DID numbers choose call forwarding

meaning how does one know if an incoming call is a telemarketer?


Is it because they have unknown as caller ID or private as Caller ID and your forwarding all those calls to another number? if that's the case, well there is also friends/family on other carriers who purposely block their Caller ID who could be trying to contact you and not to sell you something.

djino

[Mendes]

meaning how does one know if an incoming call is a telemarketer?


Is it because they have unknown as caller ID or private as Caller ID and your forwarding all those calls to another number? if that's the case, well there is also friends/family on other carriers who purposely block their Caller ID who could be trying to contact you and not to sell you something.

djino

you're too quick. I thought it was Call Forwarding, then i logged in to check and editted my post

[djino]

you're too quick. I thought it was Call Forwarding, then i logged in to check and editted my post

I understood you regardless, but my previous question is still valid.

What do you base your filtering on? what Caller ID number is supposed to represent telemarketers?

djino

[duckdown]

Anyone know the answer to my fax question above? Will my fax machine work if I plug it into my PAP2 and try to send an outgoing fax?

[death_hawk]

Can I hook up a faxmodem to my PC and use fax over the voip.ms service?

You can do fax over IP on G.711u, but you have to have a dead stable line. Any jitter at all and it'll screw up the fax.
Apparently voip.ms is getting T.38 (Fax over IP) at some point, but I have no idea when.



Is it because they have unknown as caller ID or private as Caller ID and your forwarding all those calls to another number? if that's the case, well there is also friends/family on other carriers who purposely block their Caller ID who could be trying to contact you and not to sell you something.

djino

I just forward calls that I know are telemarketers (usually by googling their number). Anyone who's blocked I route to a recorded message to tell them to call back using a non blocked number.

[Mendes]

I understood you regardless, but my previous question is still valid.

What do you base your filtering on? what Caller ID number is supposed to represent telemarketers?

djino

Correct. you CAN filter all 18xx's but sometimes creditcard companies have legit calls that come through on these DIDs

Ive got filtering by specific DID instead of the whole 18xx spectrum

It has options for blocked etc numbers also

[Rekognize]

I have a list of telemarketer phone #s that call me frequently and now with voip.ms's detailed call logging, it's easier to see who called and just throw those numbers on google and you'd get a site with a bunch of people complaining about the same number.

Now under CallerID Filtering, you can set an action for a specific caller; send to voicemail, hangup, play out of service recording, call forward, etc.

For known telemarketer DIDs I send them straight to the FCC helpdesk :lol:

[duckdown]

You can do fax over IP on G.711u, but you have to have a dead stable line. Any jitter at all and it'll screw up the fax.
Apparently voip.ms is getting T.38 (Fax over IP) at some point, but I have no idea when.


Ah, thanks for clearing that up! I'm not sure how "stable" my line is or if theres any jitter, maybe I'll just keep hounding them for Fax over IP :)

cheers

[jackie999]

Anyone know the answer to my fax question above? Will my fax machine work if I plug it into my PAP2 and try to send an outgoing fax?

As far as I know, no, you can't fax - but the feature is under development..anyone correct me?

Little more here,
http://www.dslreports.com/forum/r24431700-regular-fax-machine-or-alike-using-VoIP

[djino]

I just forward calls that I know are telemarketers (usually by googling their number). Anyone who's blocked I route to a recorded message to tell them to call back using a non blocked number.

Exactly, you can only do this from experience on the telemarketer calls you've received and even then its not 100% as their numbers could change depending on agent location within the company.

Also, some of these "telemarketers" could be people with companies that you already do business with (banks, credit card companies etc) and yes sometimes they will call to sell you other products and you could respond with by blocking further communication from them. But they could later be trying to get in contact with you to report a fraud incident on your account, but you having filtered there number would mean you wouldn't receive their calls.

djino

[Rekognize]

Anyone who's blocked I route to a recorded message to tell them to call back using a non blocked number.

Do you have a link to the .wav file??

[duckdown]

As far as I know, no, you can't fax - but the feature is under development..anyone correct me?

Little more here,
http://www.dslreports.com/forum/r24431700-regular-fax-machine-or-alike-using-VoIP

Cool, I will check into that! thanks for the response :)

[Ruri]

I tried to create an account to see if to move my voip services to this company

My account was immediately set to 'manual verification' and they requested me a copy of my passport or a driver's license

I wasn't even taking the service, I just created an account to browse the options available, as suggested earlier in this tread.

Most unfriendly company I've dealt up to now. If this is for learning about their service, I wonder what they will request when an actual issue happens.

My first born? :evil:

[djino]

I tried to create an account to see if to move my voip services to this company

My account was immediately set to 'manual verification' and they requested me a copy of my passport or a driver's license

I wasn't even taking the service, I just created an account to browse the options available, as suggested earlier in this tread.

Most unfriendly company I've dealt up to now. If this is for learning about their service, I wonder what they will request when an actual issue happens.

My first born? :evil:

The reason why they asked for that is because you choose a very specific option when you were browsing the options.

Allowed International Destination


Please note that for security reasons, some destinations are locked by default for new accounts. In order to provide you a full access to all international destinations, our security department requires verification of your ID. This can be provided as a scan of an ID card (full color and high resolution image) matching the address you have provided on file. Feel free to black out any confidential information that does not belong to the verification process.

This is a one time verification only. We apologize for any inconvenience this may cause.

Destinations that are locked:

Afghanistan, Albania, Algeria, Andorra, Angola, Anguilla, Armenia, Australia, Australia Territories, Azerbaijan, Bangladesh, Belarus, Benin, Bhutan, Bosnia & Herzegovina, Botswana, Brunei, Bulgaria, Burkina Faso, Burundi, Cameroon, Cape Verde, Central African Republic, Chad, Comoros, Comoros, Congo, Cook Islands, Croatia, Dem. Rep. of Congo, Equatorial Guinea, Eritrea, Estonia, Falkland Islands, Faroe Islands, Fiji, French Guiana, French Polynesia, Gabon, Gambia, Georgia, Ghana, Gibraltar, Guinea, Guinea Bissau, Guyana, Hungary, Iran, Iraq, Jordan, Kyrgyzstan, Laos, Latvia, Lebanon, Lesotho, Liberia, Libya, Lithuania, Macau, Macedonia, Madagascar, Malawi, Malaysia, Maldives, Mali, Malta, Marshall Islands, Mauritania, Mauritius Island, Micronesia, Moldova, Mongolia, Montserrat, Morocco, Mozambique, Namibia, Nauru, New Caledonia, Niger, Nigeria, Niue Island, North Korea, Oman, Papua New Guinea, Reunion Island, Romania, Russia, Rwanda, Sao Tome & Principe, Senegal, Serbia, Sierra Leone, Slovakia, Slovenia, Solomon Islands, South Africa, Sri Lanka, St. Helena, Sudan, Suriname, Swaziland, Syria, Tajikistan, Tanzania, Togo, Tonga, Tunisia, Turkey, Turkmenistan, Tuvalu, Uganda, Ukraine, Uzbekistan, Vanuatu, Wallis & Futuna, Yemen (Arab Republic), Zambia and Zimbabwe.

djino

[jackie999]

I tried to create an account to see if to move my voip services to this company

My account was immediately set to 'manual verification' and they requested me a copy of my passport or a driver's license

I wasn't even taking the service, I just created an account to browse the options available, as suggested earlier in this tread.

Most unfriendly company I've dealt up to now. If this is for learning about their service, I wonder what they will request when an actual issue happens.

My first born? :evil:

Wow..that's odd. I set up my account and browsed through the configurations before I decided to try it. In fact, it was within the voip.ms portal that I did my paypal transfer - once I'd decided to give it a try.
Are you saying you can't even sign up? Once I applied to port my number over, then I had to give some information i.e. to verify that the number I was porting was really mine. Very odd..

At NO TIME did I have to supply drivers license or passport. DO NOT DO THAT!!

[Pete Jones]

I tried to create an account to see if to move my voip services to this company

My account was immediately set to 'manual verification' and they requested me a copy of my passport or a driver's license

I wasn't even taking the service, I just created an account to browse the options available, as suggested earlier in this tread.

Most unfriendly company I've dealt up to now. If this is for learning about their service, I wonder what they will request when an actual issue happens.

My first born? :evil:

That's strange. When I set my own account up back in May, this wasn't an issue.

Could it be that you want to make calls to one of the "locked" destinations? They do require you provide ID if you want to call countries from what seems to be a rather lengthy list. To be honest, I'm not sure why they do this since other voip companies don't require it. I've configured my device to use callwithus and localphone as alternate services and I don't recall there being any need for verification with them. Then again, I don't call any of those countries.

--Pete

[Ruri]

thank you for your replies.

I was just browsing the site, I did not want to do any calls, just check the options they have.

All this happened when I was trying to sign up, to create an account in their site.

I had not requested service, choose any plans or anything else.

If anything, I wanted this account for Canada and US calls I don't know anything about any of the locked destinations.

But to request these documents without any explanations, very hostile IMHO.

[jackboot]

thank you for your replies.

I was just browsing the site, I did not want to do any calls, just check the options they have.

All this happened when I was trying to sign up, to create an account in their site.

I had not requested service, choose any plans or anything else.

If anything, I wanted this account for Canada and US calls I don't know anything about any of the locked destinations.

But to request these documents without any explanations, very hostile IMHO.

I hardly think simply requesting documents is hostile...a misunderstanding sounds more accurate IMO. In any case, my experience with voip.ms has been 100% positive and I've dealt with their support maybe a dozen times now.

Just contact them and ask for clarification. That's a lot more productive than complaining about it here.

[Ruri]

I already did.

If you consider that those documents are not being requested to anybody here and they were requested to me with no further explanation perhaps you would understand why I qualify that action as hostile.

This forum is for sharing tips and experiences, that is the reason of my posting here.

[rubin303]

I just have a quick question for DISA setup. If I call this number from my cell phone do I get charged with incoming minute only? I know someone probably asked this already but seeing as it lets u instantly dial the number right away feels like an outgoing call.

[barrist]

I already did.

If you consider that those documents are not being requested to anybody here and they were requested to me with no further explanation perhaps you would understand why I qualify that action as hostile.

This forum is for sharing tips and experiences, that is the reason of my posting here.

Sounds like an aberration, because EVERYONE I've ever heard talk about it has raved about its service.

Maybe you chose some option by mistake which requires this ID check.

[trini_2_d_bone]

I signed up about 1 mth now and I'm totally satisfied with their service.
My bill was about $10.00 because we use a lot of LD mins.
The only documents they requested was a copy of my Bell bill to facilitate the number porting.

[djino]

I just have a quick question for DISA setup. If I call this number from my cell phone do I get charged with incoming minute only? I know someone probably asked this already but seeing as it lets u instantly dial the number right away feels like an outgoing call.

You have a DID assigned to the DISA which I assume you chose the DID with the option of paying 99 cents/month + $0.01/minute on incoming.

So if you are calling your DID thats assigned to the DISA, you will be charged $0.01/minute.

Once you've punched in your PIN (as per DISA), you are presented with a Dial-tone so you can then make an outbound call. This outbound call will be charged on top of the $0.01/minute depending on the location of the outbound call ($0.0052/minute if calling a number within Canada and $0.011/minute if calling a number within the USA).

So if you are calling your DID to make a call to someone in Canada for 10 minutes, you would be charged: (10 * 0.01) + (10 * 0.0052) OR (10 * 0.0152) = $0.15

djino

[rubin303]

You have a DID assigned to the DISA which I assume you chose the DID with the option of paying 99 cents/month + $0.01/minute on incoming.

So if you are calling your DID thats assigned to the DISA, you will be charged $0.01/minute.

Once you've punched in your PIN (as per DISA), you are presented with a Dial-tone so you can then make an outbound call. This outbound call will be charged on top of the $0.01/minute depending on the location of the outbound call ($0.0052/minute if calling a number within Canada and $0.011/minute if calling a number within the USA).

So if you are calling your DID to make a call to someone in Canada for 10 minutes, you would be charged: (10 * 0.01) + (10 * 0.0052) OR (10 * 0.0152) = $0.15

djino

Ah..okay. Didn't know it was charged both way.

Thanks for the reply.

But what about my cell phone charge if it was let's say $0.05/min incoming and $0.10/min outgoing.

[djino]

Ah..okay. Didn't know it was charged both way.

Thanks for the reply.

But what about my cell phone charge if it was let's say $0.05/min incoming and $0.10/min outgoing.

I have no clue what you mean by cell phone charge? There is no cell phone charge.

djino

[kcorscadden]

So I downloaded the latest firmware upgrade of 5.16, I extracted the .exe file and it asks for the ATA's IP address. I entered in 192.168.001.116 and it comes back with an error of "Upload failed: No response from SPA"

I rebooted the ATA, router and modem and still get the same error

Any ideas why?

[Pete Jones]

Ah..okay. Didn't know it was charged both way.

Thanks for the reply.

But what about my cell phone charge if it was let's say $0.05/min incoming and $0.10/min outgoing.

Your cell phone charges will be whatever you would normally pay for a call. But that's your cell phone bill. This is completely exclusive of the voip.ms situation.

If you're paying less for incoming, (or you unlimited incoming) you may want to consider setting up a callback on your DID at voip.ms

--Pete

[rubin303]

I have no clue what you mean by cell phone charge? There is no cell phone charge.

djino

Ops. Sorry for making it unclear.
My question was if I called my DID number(which has DISA set up) by my cell phone then would the cell phone be charged for the incoming minutes or outgoing minute? (Aside the DID number charges)
I think Pete answered that the cell phone is charged by the outgoing minutes.
For some reason I thought since DISA is similar to call back I thought somehow it magically charges your cell phone by the incoming minutes.


Your cell phone charges will be whatever you would normally pay for a call. But that's your cell phone bill. This is completely exclusive of the voip.ms situation.

If you're paying less for incoming, (or you unlimited incoming) you may want to consider setting up a callback on your DID at voip.ms

--Pete

Ah okay. Thank you for clearing that up.

[djino]

Ops. Sorry for making it unclear.
My question was if I called my DID number(which has DISA set up) by my cell phone then would the cell phone be charged for the incoming minutes or outgoing minute? (Aside the DID number charges)
I think Pete answered that the cell phone is charged by the outgoing minutes.
For some reason I thought since DISA is similar to call back I thought somehow it magically charges your cell phone by the incoming minutes.



Ah okay. Thank you for clearing that up.

If you want it to just your cell phone incoming minutes, then you are NOT wanting to look at DISA, but Voip.ms Callback service (Click Callback under the DID Menu).

Basically, with your cell phone, you call your DID, it will return a busy signal inwhich you would hang up. Seconds later, Voip.ms will call your cell phone back and give you a dial-tone to make the outbound call. This comes though as an incoming call to your cell phone only.

The charges here from Voip.ms will be slighly lower than DISA.

You calling the DID that returns a busy signal = no charge
Voip.ms uses its termination rate to contact your cell phone = $0.0052/minute
Voip.ms uses its termination rate to make the outgoing call to the party you want to call ($0.0052/minute to Canada and $0.011/minute to USA).

Making a 10 Minute call to someone in Canada (like the example in my previous post) would cost you 2 * 10 * $0.0052/minute = $0.10. This is 5 cents less than if you would have used DISA.

djino

[Lulz]

Oh man, day after day I find more and more useful features from VOIP.ms
I am glad you guys started talking about Callback because its going to be really awesome for me right now.


I am currently located in Russia and spend a lot of money on long distance cards to call back to Canada. Not to mention I am also using cell phone minutes as well.

But my questions are:

1) what number sequence do I have to call after I get a "callback" to call someone say...in Canada with a 416 number? standard 011-1-416-..... or how?

and

2) I currently have 1 number with VOIP.ms, does it mean that if I want to use Callback service, that number will be "used" and nobody can reach me anymore by dialing that number?

Let me explain that further:

I have 416-111-1111 and it's being currently forwarded to my Russian cell phone with free incoming minutes....everytime somebody calls me on 416-111-1111, I pick up with my Russian cell phone.

But if I use that 416-111-1111 as a Callback, then wouldn't it mean everybody who tries to call me at that number, will receive a callback from VOIP.ms with a dialtone and they won't know what the heck is going on and why they are receiving a call with dialtone, everytime they try to contact me.

Thank you.

special thanks to djino for most awesome help!!!

[ganiba]

Oh man, day after day I find more and more useful features from VOIP.ms
I am glad you guys started talking about Callback because its going to be really awesome for me right now.


I am currently located in Russia and spend a lot of money on long distance cards to call back to Canada. Not to mention I am also using cell phone minutes as well.

But my questions are:

1) what number sequence do I have to call after I get a "callback" to call someone say...in Canada with a 416 number? standard 011-1-416-..... or how?

and

2) I currently have 1 number with VOIP.ms, does it mean that if I want to use Callback service, that number will be "used" and nobody can reach me anymore by dialing that number?

Let me explain that further:

I have 416-111-1111 and it's being currently forwarded to my Russian cell phone with free incoming minutes....everytime somebody calls me on 416-111-1111, I pick up with my Russian cell phone.

But if I use that 416-111-1111 as a Callback, then wouldn't it mean everybody who tries to call me at that number, will receive a callback from VOIP.ms with a dialtone and they won't know what the heck is going on and why they are receiving a call with dialtone, everytime they try to contact me.

Thank you.

special thanks to djino for most awesome help!!!

Hi Lulz,

you can use "CallerID Filtering"
As a default rule, you can keep that for your DID number, received calls will be Forwarded to your cell.
Then, you go to the "CalledID Filtering" and add the exception that "for calls received from inside Russia", you will select as a Routing, the "Call Back" option.

Good luck.

[ganiba]

If you want it to just your cell phone incoming minutes, then you are NOT wanting to look at DISA, but Voip.ms Callback service (Click Callback under the DID Menu).

Basically, with your cell phone, you call your DID, it will return a busy signal inwhich you would hang up. Seconds later, Voip.ms will call your cell phone back and give you a dial-tone to make the outbound call. This comes though as an incoming call to your cell phone only.

The charges here from Voip.ms will be slighly lower than DISA.

You calling the DID that returns a busy signal = no charge
Voip.ms uses its termination rate to contact your cell phone = $0.0052/minute
Voip.ms uses its termination rate to make the outgoing call to the party you want to call ($0.0052/minute to Canada and $0.011/minute to USA).

Making a 10 Minute call to someone in Canada (like the example in my previous post) would cost you 2 * 10 * $0.0052/minute = $0.10. This is 5 cents less than if you would have used DISA.

djino

5 cents less, plus cell fees for using outgoing calls on his cell with DISA instead of free (if it is the case) incoming calls with callback service.

[Lulz]

Anybody got their free iNum number yet from VOIP?

http://www.inum.net/what-is-inum/what-is-inum/

can i delete my number if i decide not to use it? or if i dont like the number at all?

and about DISA: whats the point of it if you could just use callback? what are benefits of DISA?

[rubin303]

If you want it to just your cell phone incoming minutes, then you are NOT wanting to look at DISA, but Voip.ms Callback service (Click Callback under the DID Menu).

Basically, with your cell phone, you call your DID, it will return a busy signal inwhich you would hang up. Seconds later, Voip.ms will call your cell phone back and give you a dial-tone to make the outbound call. This comes though as an incoming call to your cell phone only.

The charges here from Voip.ms will be slighly lower than DISA.

You calling the DID that returns a busy signal = no charge
Voip.ms uses its termination rate to contact your cell phone = $0.0052/minute
Voip.ms uses its termination rate to make the outgoing call to the party you want to call ($0.0052/minute to Canada and $0.011/minute to USA).

Making a 10 Minute call to someone in Canada (like the example in my previous post) would cost you 2 * 10 * $0.0052/minute = $0.10. This is 5 cents less than if you would have used DISA.

djino

lol..I always thought callback and DISA would be charged one way.

How wrong of me.

I appreciate your response, thanks agian.

[shadow_cruiser]

Going back to the 310 number subject, I've that Acanac and www.atlasvoice.com can dial out 310 numbers no problem. Those that can dial out 310 numbers, are you using premium routing?. I'm not sure if that even matters.

[plg_cp]

Going back to the 310 number subject, I've that Acanac and www.atlasvoice.com can dial out 310 numbers no problem. Those that can dial out 310 numbers, are you using premium routing?. I'm not sure if that even matters.

Regarding your issue...I know others have gotten it working using a similar dial plan, but isn't your plan (posted earlier) causing the unit to automatically add the area code when you dial only 7 digits? This would mean it's "impossible" for the call to go out with only 7 digits as needed for 310-MAXX since the dial plan adds the area code.
I'm not sure how to edit it, but I think you need to change it so that it will only add the area code if the first 3 of the 7 digits dialed are not "310".

[shadow_cruiser]

Regarding your issue...I know others have gotten it working using a similar dial plan, but isn't your plan (posted earlier) causing the unit to automatically add the area code when you dial only 7 digits? This would mean it's "impossible" for the call to go out with only 7 digits as needed for 310-MAXX since the dial plan adds the area code.
I'm not sure how to edit it, but I think you need to change it so that it will only add the area code if the first 3 of the 7 digits dialed are not "310".

I never thought of that but looking at this tutorial page http://www.netphonedirectory.com/pap2_dialplan2.htm , I can't figure out how to configure my dial plan without the area code. I always dial the area code before the number so is the 613 really needed?. When I remove the 613, I get a fast beeping after I dial out any number. I want to configure my dial plan so the person MUST dial the area code first.

[duckdown]

I'm having a problem, I just got a Presidents Choice MasterCard and I'm having a problem on my account, so I'm trying to call their toll free number and I get an error saying the phone number isn't available from my area at this time.

What gives? I can call any other numbers fine, just not this toll free one.

1-866-246-PCMC (7262)

Can anyone try?

Thanks!

[jackie999]

Can anyone try?

Works okay here..
My dialing plan is pretty much default...
(<:1905>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|*xx.|[34689]11|822|4443|4747|0|00|[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|4XXX|xxxxxxxxxxxx.)

In voip.ms account settings I have,
toll free routing - value (free)
toll free termination carrier - server default

not sure if they make a difference...

[jackie999]

Also, I'm using the Toronto server...are you using a US one?

[Jonavin]

I'm having a problem, I just got a Presidents Choice MasterCard and I'm having a problem on my account, so I'm trying to call their toll free number and I get an error saying the phone number isn't available from my area at this time.

What gives? I can call any other numbers fine, just not this toll free one.

1-866-246-PCMC (7262)

Can anyone try?

Thanks!

I just tried it on my voip.ms account without problems. Did you check to see whether your account was setup for US 1-800 numbers or Canadian ones?

[shadow_cruiser]

I'm having a problem, I just got a Presidents Choice MasterCard and I'm having a problem on my account, so I'm trying to call their toll free number and I get an error saying the phone number isn't available from my area at this time.

What gives? I can call any other numbers fine, just not this toll free one.

1-866-246-PCMC (7262)

Can anyone try?

Thanks!

Works fine for me too. I'm using the default dialing plan given by voip.ms

[Rekognize]

Since cancelling our Rogers homephone and internet, they signed us up for spam and been calling our voip phone non stop!

http://i.imgur.com/nnsi6.jpg

[pipolchap]

:arrowu:

When you play those "no service" messages, you're still charged for their incoming call, aren't you?

I don't want to spend a cent on theses guys, so I treat my telemarketers with a Hangup. No charge.

[Rekognize]

:arrowu:

When you play those "no service" messages, you're still charged for their incoming call, aren't you?

I don't want to spend a cent on theses guys, so I treat my telemarketers with a Hangup. No charge.

Yeah, $0.002

[death_hawk]

Do you have a link to the .wav file??
http://cepstral.com
Bonus points if you license a neat voice.
Hell, I'm tempted to license Damien for Telemarketers.
"Don't call back or you will lose your soul!"



and about DISA: whats the point of it if you could just use callback? what are benefits of DISA?

It depends on your usage.
For me, I have unlimited incoming calls, but I pay for outgoing. Some people have a Favorite X, but Fido doesn't offer it for some stupid reason.

I like DISA better IMO. It's a step less than Callback. You call in and get a dialtone instead of calling in, waiting for busy, waiting for a callback then dialing out.

[Madwand]

I'm having a problem, I just got a Presidents Choice MasterCard and I'm having a problem on my account, so I'm trying to call their toll free number and I get an error saying the phone number isn't available from my area at this time.

What gives? I can call any other numbers fine, just not this toll free one.

1-866-246-PCMC (7262)

Can anyone try?

Thanks!

Did you try without the 1 in front?

[EPcjay]

I'm having a problem, I just got a Presidents Choice MasterCard and I'm having a problem on my account, so I'm trying to call their toll free number and I get an error saying the phone number isn't available from my area at this time.

What gives? I can call any other numbers fine, just not this toll free one.

1-866-246-PCMC (7262)

Can anyone try?

Thanks!

Sounds like your using a US server. My dad had trouble calling CRA the other day, and it indicated "You cannot call from your area". I had to change the toll-free termination carrier to "Canadian Carrier" from "server default".

http://img717.imageshack.us/img717/6420/52051458.th.jpg (http://img717.imageshack.us/i/52051458.jpg/)

Uploaded with ImageShack.us (http://imageshack.us)

[Chaseme]

Since cancelling our Rogers homephone and internet, they signed us up for spam and been calling our voip phone non stop!

http://i.imgur.com/nnsi6.jpg

Nice, I always though it would be the best bet though to call forward the telemarketers to the Corp reception desk of their own company though. Should cause enough internal hassle for them to drop your number from the list.

[Rekognize]

Nice, I always though it would be the best bet though to call forward the telemarketers to the Corp reception desk of their own company though. Should cause enough internal hassle for them to drop your number from the list.

What's the number for Rogers' corp reception desk?

Nevermind, found it. Tried it and someone picked up, no auto-answer :D

[imran909]

Some people I spoke personally have also noted issue of increased tele-marketting when they switched their phone numbers to either freephoneline.ca or even to the VOIP.MS

In my case I have service from both for more than 6 months but did not ported the numbers and so far no single issue of telemarketters.

[apn64]

I've noticed that most telemarketers use toll-free numbers, hence it might be easier to simply forward all 800/888/866/877 numbers to;

- voice-mail
- one of the predefined system messages
- one of the many pre-recorded "crazy" messages on SIP services
- a personal "honey-pot" recording to keep them on the line for 10+ minutes
- the FCC
- the CRTC
- many other choices.

I also use the "Anonymous" rule and a specific Montreal number for a scammer. However, the former sometimes catches unintended recipients e.g. friends calling from a mobile that doesn't provide CID.

Using these techniques, I've had zero telemarketing calls in more than a year.

[Chaseme]

What's the number for Rogers' corp reception desk?

Nevermind, found it. Tried it and someone picked up, no auto-answer :D

Boom! I'm sure they'll love deal with that BS.

[Rekognize]

Boom! I'm sure they'll love deal with that BS.

Awesome, thanks for the tip.

[kcorscadden]

Alright I can't figure this out:

Every time I or someone else calls our phone it goes directly to VM. As far as I can tell DND is not on as I pressed the deactivate code and it still goes to VM.

Voip.ms support is offline.

ATA is a PAP2T

Any help?

[GT!!]

Alright I can't figure this out:

Every time I or someone else calls our phone it goes directly to VM. As far as I can tell DND is not on as I pressed the deactivate code and it still goes to VM.

Voip.ms support is offline.

ATA is a PAP2T

Any help?

Likely a firewall or router (NAT) issue. I had weird problems until I replaced the firmware on my Linksys WRTG router with DD-WRT.

-GT

[jackie999]

Alright I can't figure this out:

Every time I or someone else calls our phone it goes directly to VM. As far as I can tell DND is not on as I pressed the deactivate code and it still goes to VM.

Voip.ms support is offline.

ATA is a PAP2T

Any help?

How is your DID set up on voip.ms ....routed to ???

[Dave98]

(<:1416>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|*xx.|[34689]11|822|4443|4747|0|00|[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|4XXX|xxxxxxxxxxxx.)


Does anyone see a problem with this dialing plan for international calls? My father needed to place a call to China but had problems with the line disconnecting once the call recipient picked up the phone.

Placed the call with our RHP line and it went through fine.

[kcorscadden]

Likely a firewall or router (NAT) issue. I had weird problems until I replaced the firmware on my Linksys WRTG router with DD-WRT.

-GT

Just disabled my firewall and still goes directly to VM.


How is your DID set up on voip.ms ....routed to ???

Modem > Router > ATA

I assume this is what you mean. I have a Linksys WRT54GL with Tomato

[jackie999]

JI assume this is what you mean. I have a Linksys WRT54GL with Tomato


No, log into voip.ms with your username etc..under manage DID..where is the did routing to?

[Mendes]

Alright I can't figure this out:

Every time I or someone else calls our phone it goes directly to VM. As far as I can tell DND is not on as I pressed the deactivate code and it still goes to VM.

Voip.ms support is offline.

ATA is a PAP2T

Any help?

Someone asked where it is routed to which is a good starting point.

This is the default when there are no registrations(ie phones connected). so keep the voip.ms page open that shows if you're connected or not. also check the timeout before going to voicemail to make sure its not like 1s

[Rekognize]

Alright I can't figure this out:

Every time I or someone else calls our phone it goes directly to VM. As far as I can tell DND is not on as I pressed the deactivate code and it still goes to VM.

Voip.ms support is offline.

ATA is a PAP2T

Any help?

DID numbers > Manage DIDs > Main Account inbound Settings > Device type: ATA device, IP Phone, or Softphone

[apn64]

(<:1416>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|*xx|*xx.|[34689]11|822|4443|4747|0|00|[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxxS0|4XXX|xxxxxxxxxxxx.)


Does anyone see a problem with this dialing plan for international calls? My father needed to place a call to China but had problems with the line disconnecting once the call recipient picked up the phone.

Placed the call with our RHP line and it went through fine.

Your dial plan looks like hell, but if the call connected, then the dial plan CANNOT be the source of the problem.

I have a *lot* of calls with China, and frankly, their telephony services are 19th century in terms of quality.

One thing that I can tell you leads to random disconnects is the DTMF setting if you're using a PAP2T. Basically, the fax detection mode kicks in and you will find that certain sounds especially female voices trigger it, leading to a disconnect.

[apn64]

Alright I can't figure this out:

Every time I or someone else calls our phone it goes directly to VM. As far as I can tell DND is not on as I pressed the deactivate code and it still goes to VM.

Voip.ms support is offline.

ATA is a PAP2T

Any help?

Might want to check your call routing on failover (Manage DID menu), since it might be sending all calls to VM if your ATA can't be reached.

The above might point to an ATA registration problem e.g NAT settings or something else preventing registration to a server.

Another possibility is that if you're using the Anonymous CID filtering and sending those calls to VM. I use that setting, and find that certain cellphone callers always go to VM because they have their phone setup not to provide CID.

BTW, I find VoIP.ms' Anonymous option flawed. I often see "Anonymous" in the CID name field, with a valid # in number field. IMHO, the Anonymous setting should only pickup cases where both CID name AND number are absent, or name is absent and number is all 0's.

[pipolchap]

BTW, I find VoIP.ms' Anonymous option flawed. I often see "Anonymous" in the CID name field, with a valid # in number field. IMHO, the Anonymous setting should only pickup cases where both CID name AND number are absent, or name is absent and number is all 0's.

If there is a valid #, I find that the CID name field is just the number (i.e. "123-456-7891" <123-456-7891>) or "PRIVATE NAME" (i.e. "PRIVATE NAME" <123-456-7891>).

I've yet to encounter: "Anonymous" <123-456-7891>

[Pete Jones]

Your dial plan looks like hell, but if the call connected, then the dial plan CANNOT be the source of the problem.

That leads me to ask, why the need for complicated dial plans?

Here's mine:

(*xx.|xx.)

Basically, whatever I dial out, goes out. Never fails. The only thing I find is it might take a few seconds before the ATA makes the call.

I have recently added a couple of special double-star codes to reroute my dialing through a couple different gateways, but basically, that's it.

--Pete

[kcorscadden]

No, log into voip.ms with your username etc..under manage DID..where is the did routing to?


Someone asked where it is routed to which is a good starting point.

This is the default when there are no registrations(ie phones connected). so keep the voip.ms page open that shows if you're connected or not. also check the timeout before going to voicemail to make sure its not like 1s


DID numbers > Manage DIDs > Main Account inbound Settings > Device type: ATA device, IP Phone, or Softphone


Might want to check your call routing on failover (Manage DID menu), since it might be sending all calls to VM if your ATA can't be reached.

The above might point to an ATA registration problem e.g NAT settings or something else preventing registration to a server.

Another possibility is that if you're using the Anonymous CID filtering and sending those calls to VM. I use that setting, and find that certain cellphone callers always go to VM because they have their phone setup not to provide CID.

BTW, I find VoIP.ms' Anonymous option flawed. I often see "Anonymous" in the CID name field, with a valid # in number field. IMHO, the Anonymous setting should only pickup cases where both CID name AND number are absent, or name is absent and number is all 0's.

It is all fixed now and basically it is what you guys said. Where were you guys last night? LOL

Anyways, I was going to wrong section to change servers. I was doing Main Menu > Account Settings > Change Server, when I should have been doing DID numbers > Manage DIDs > Edit DID > Server > Save. I didn't know there was a difference and figured once you change the server on the portal and ATA that all was fine.

[Traian2003]

That leads me to ask, why the need for complicated dial plans?

Here's mine:

(*xx.|xx.)

Basically, whatever I dial out, goes out. Never fails. The only thing I find is it might take a few seconds before the ATA makes the call.

I have recently added a couple of special double-star codes to reroute my dialing through a couple different gateways, but basically, that's it.

--Pete

Change the interdigit timer from 10 to something like 3 or 4. It's under regional or sip (advanced settings tho). That should make it much faster. ;)

[apn64]

Change the interdigit timer from 10 to something like 3 or 4. It's under regional or sip (advanced settings tho). That should make it much faster. ;)

You could also specify the inter-digit timer IN the dial-plan. Something like;

(*xx.S5|xx.S5)

..which gives you 5 seconds between digits. Changing 5 to another integer gives you more/less time as desired.

Another use for dial-plan is to redirect calls e.g. if you don't want to pay for 411 calls, you can redirect to something like 800-GOOG-411 (Google's FREE service) with;

<411:18004664411>

I also use the redirect option to send all calls to relatives (also on VoIP) to SipBroker, so they're FREE :D If they reciprocate, then the incoming calls (to me) are FREE too :D :D

[jackie999]

It is all fixed now and basically it is what you guys said. Where were you guys last night? LOL

Anyways, I was going to wrong section to change servers. I was doing Main Menu > Account Settings > Change Server, when I should have been doing DID numbers > Manage DIDs > Edit DID > Server > Save. I didn't know there was a difference and figured once you change the server on the portal and ATA that all was fine.

It's all a little confusing for me too..but when you said there were problems with calls coming in..that's your DID (direct inward dial) settings..so that's first place to look.
Your server setting is where your service registers to. There is a setting in the voip.ms portal and also the PAP2T. To further confuse you I'll throw in a quote from mango from the DSLR forums:

Actually, registration isn't required to make outgoing calls. Also, if you register to a different POP than your DID is routed to, the portal should still state that you are registered. (Though you're correct, incoming calls won't work.)

m.

I'm just happy the voip thing works and I've saved buckets of money using it :) and if I can get the netflix working in the fall...I'll be finally rid of bell completely !!

[shadow_cruiser]

Your dial plan looks like hell, but if the call connected, then the dial plan CANNOT be the source of the problem.

Can you post your current dial plan please?

[apn64]

Can you post your current dial plan please?

Certainly. I'm sure it has flaws too, but it appears a lot more coherent than what's posted above.


(*xxS0 | [89]11S0 | <411:18004664411>S0 | <(telnum11):*9780(telnum10)@sipbroker.com> | [2-9]xx[2-9]xxxxxxS0 | 1[2-9]xx[2-9]xxxxxxS0 | 011[2-9]x.S5 | 1900! | 900! | 1976! | 976! )

...and explanations;

*xx Feature codes

[89]11S0 811 echo test and 911; S0 = no interdigit delay after match, sometimes called "straight out"

<411:18004664411>S0 Redirect 411 calls to Google's free service

<(telnum11):*9780(telnum10)@sipbroker.com>S0 Redirect calls to (telnum) via SIP. Where telnum11 is the full 1+10-digit number of my Inlaws, telnum10 is the 10-digit version of the same number.

[2-9]xx[2-9]xxxxxxS0 10-digit local dialing

1[2-9]xx[2-9]xxxxxxS0 Long-distance dialing; not actually needed (10 digits OK on VoIP, so this is for legacy users and so I don't have to reprogram all my phones)

011[2-9]x.S5 International calls

1900! | 900! | 1976! | 976! Block access to the pr0n services

[Rekognize]

BTW, where do your [FWD] go to? Are you aware that redirecting people to system messages actually cost you money (aka no service message)?

FWD to FCC/CRTC, and the Rogers Telemarketers are FWD'ed to the Rogers Corp Reception. Seems like it worked, no telemarketing calls today.

[apn64]

Anyone else here get the emails (yesterday) advising of a partial-outage, affecting ~20 customers on a certain carrier?

I'm one of those impacted (also found 4 others) and discovered that outbound calling is fine, but all incoming calls get fast-busy.

VoIP.ms is handling the situation well, offering temp and permanent replacement DIDs on a new, more reliable carrier, but some of us don't want to change a number we've had for a while.

Edit: No worries. I was conversing w/ VoIP.ms on another site and my original DID is back in service. Life's good :)

[Pete Jones]

Change the interdigit timer from 10 to something like 3 or 4. It's under regional or sip (advanced settings tho). That should make it much faster. ;)

I tweaked the value on my ATA. Thanks. Funny thing is, we got used to the default of 10 seconds. :)


You could also specify the inter-digit timer IN the dial-plan. Something like;

(*xx.S5|xx.S5)

..which gives you 5 seconds between digits. Changing 5 to another integer gives you more/less time as desired.

Another use for dial-plan is to redirect calls e.g. if you don't want to pay for 411 calls, you can redirect to something like 800-GOOG-411 (Google's FREE service) with;

<411:18004664411>

I also use the redirect option to send all calls to relatives (also on VoIP) to SipBroker, so they're FREE :D If they reciprocate, then the incoming calls (to me) are FREE too :D :D

I use redirecting to use my other providers for certain calls.

Here's my complete dial plan:

(<**1 :> xx. <:@gw1>|<**2 :> xx. <:@gw2>|<*98SO:*97>|*xx.|xx.)

Any number preceded by **1 routes through gateway 1 (callwithus.com)

Any number preceded by **2 routes through gateway 2 (localphone.com)

*98 gets rerouted to *97 (we're just used to dialing *98 for voicemail so I wanted to keep things consistent)

Anything else goes through as dialed.

--Pete

[runeash]

I got free 416 phone number from freephoneline when they first started I think it was a promotion. How do you use it ie what is ATA, any steps would be appreciated. Any way to port my existing number onto freephoneline


Why pay for it if I can get freephoneline for free?

or for only 60 CAD (only once forever) if you want use your ATA

[kcorscadden]

It's all a little confusing for me too..but when you said there were problems with calls coming in..that's your DID (direct inward dial) settings..so that's first place to look.
Your server setting is where your service registers to. There is a setting in the voip.ms portal and also the PAP2T. To further confuse you I'll throw in a quote from mango from the DSLR forums:


I'm just happy the voip thing works and I've saved buckets of money using it :) and if I can get the netflix working in the fall...I'll be finally rid of bell completely !!

Ya it was a tad confusing at first. I guess what happened was that I was logged into the Toronto server, and then I would go Main Menu > Account Setting > Server > Save and go and select the same server I chose for the ATA. Problem is that even after changing the server, I was still connected to the Toronto server, but my ATA was saying NY or CHI, etc. It wasn't until I went to DID Numbers > Manage DID > Edit DID > Server > Save and then do the same on the ATA that it worked.

As for Netflix, I am not really all that excited about it as I have usenet. $100 a yr unlimited downloading of tv shows/movies/etc, Netflix won't beat that....

[Rekognize]

As for Netflix, I am not really all that excited about it as I have usenet. $100 a yr unlimited downloading of tv shows/movies/etc, Netflix won't beat that....

When you say unlimited, is your ISP unlimited as well?

[Lulz]

Are you aware that redirecting people to system messages actually cost you money (aka no service message)?

Oh really? Thanks for heads up, didn't know.

How much does it cost though? Lets say a telemarketer calls me and reaches LINE DISCONNECTED system message. How much will I pay for that call? And does the system disconnect you after a while, or will it keep saying LINE DISCONNECTED forever?

Thanks

[kcorscadden]

When you say unlimited, is your ISP unlimited as well?

No my ISP is not unlimited, but that doesn't really matter with Usenet or Netflix as anything you download goes towards your cap.

I am with Teksavvy and have a 200GB cap, so I guess my usenet cap is 200GB unless I download stuff from another ISP (aka: Brother, Sister, Parents, etc).

Personally, there is way more value with usenet then there is with Netflix. Netflix will have either a monthly fee of say $10 or they will have an itunes type setup where you pay like $3.00 per movie/tv show to download.

[jackie999]

Agreed...netflix and usenet is like comparing apples and oranges...
I also have the $96 a year unlimited astraweb and think it's great. The netflix I was hoping would be for hubby...he likes to lay around at night and surf ...so I keep the bell sub, although it's overpriced and bulked up with channels that we'd never watch..so I was hoping netflix will be an alternative but I won't know until it's here...have my wdlive plus on order so I'll be ready :)

[Dave98]

Your dial plan looks like hell, but if the call connected, then the dial plan CANNOT be the source of the problem.

I have a *lot* of calls with China, and frankly, their telephony services are 19th century in terms of quality.

One thing that I can tell you leads to random disconnects is the DTMF setting if you're using a PAP2T. Basically, the fax detection mode kicks in and you will find that certain sounds especially female voices trigger it, leading to a disconnect.

:lol: Well the dialing plan was copy/pasted from somewhere. I just forget where. I remember all I did to modify it was add in the 416 area code. I'll make some adjustments there.

I do have a PAP2T so I'll look into this DTMF setting. Thanks.

[mastermc]

If you goto this website: http://www.payphone-directory.org/sounds.html

You can download and then upload the wav file to VOIP.ms for a disconnected recording. Pick a good disconnected message for telemarketers that call you. The Best thing with some of these recordings is that it includes the SIT tones. With some of the dialers out there, your number will automatically be removed when it dials a number and gets a SIT tone.

[dorje]

has anyone had problems with the phone line not registering with voip.ms, this is the second time it has happened to me, when i tried to make a call, the line was dead, on checking my voip.ms it showed my did was not registered.

i checked my PAP2T ATA, and there was only two lights; power and internet and no phone 1 or phone 2,

so i had to unplug the ATA power for few secs and plug it back in.

is there anyway to set it up so that it doesn't happen often, there's no way to know if the line is there or not unless i make calls and don't want to lose calls.

[jackie999]

has anyone had problems with the phone line not registering with voip.ms, this is the second time it has happened to me, when i tried to make a call, the line was dead, on checking my voip.ms it showed my did was not registered.

i checked my PAP2T ATA, and there was only two lights; power and internet and no phone 1 or phone 2,

so i had to unplug the ATA power for few secs and plug it back in.

is there anyway to set it up so that it doesn't happen often, there's no way to know if the line is there or not unless i make calls and don't want to lose calls.

I was advise, by Mango on the voip DSL forums - "If you have Tomato firmware, try changing the Unreplied UDP Timeout in Conntrack/Netfilter settings to 10."
It's something to try...
Here (http://www.dslreports.com/forum/r24544129-torontovoipms-Registration) is the thread if you are interested...

[neo1973]

Has anyone ever successfully put a Linksys SPA2102 behind a router and an access point? Tried everything I could but I could only call out. When people try to dial the number they heard nothing. Any help would be much appreciated!

[shadow_cruiser]

Certainly. I'm sure it has flaws too, but it appears a lot more coherent than what's posted above.


(*xxS0 | [89]11S0 | <411:18004664411>S0 | <(telnum11):*9780(telnum10)@sipbroker.com> | [2-9]xx[2-9]xxxxxxS0 | 1[2-9]xx[2-9]xxxxxxS0 | 011[2-9]x.S5 | 1900! | 900! | 1976! | 976! )

...and explanations;

*xx Feature codes

[89]11S0 811 echo test and 911; S0 = no interdigit delay after match, sometimes called "straight out"

<411:18004664411>S0 Redirect 411 calls to Google's free service

<(telnum11):*9780(telnum10)@sipbroker.com>S0 Redirect calls to (telnum) via SIP. Where telnum11 is the full 1+10-digit number of my Inlaws, telnum10 is the 10-digit version of the same number.

[2-9]xx[2-9]xxxxxxS0 10-digit local dialing

1[2-9]xx[2-9]xxxxxxS0 Long-distance dialing; not actually needed (10 digits OK on VoIP, so this is for legacy users and so I don't have to reprogram all my phones)

011[2-9]x.S5 International calls

1900! | 900! | 1976! | 976! Block access to the pr0n services

Thanks so much for posting that, it's a lot more clearer than the defaulted linksys or voip.ms dial plan. I edited yours to suit my needs. I removed telnum, pRon services and added 911 so whenever someone dials, it will go directly to the police station. I wanted to edit the long distance dialing so I don't have to dial 1 before the area code, do I just remove the 1?. Tell me what you think. I just tried this dial plan and it's a huge difference over the longer defaulted one. Calls connect so much quicker now. There's no longer that delay between after you dial and before the first ring. Thanks dude!

(*xxS0 | [8]11S0 | <411:18004664411>S0 | <911:8195957600>S0 | [2-9]xx[2-9]xxxxxxS0 | 1[2-9]xx[2-9]xxxxxxS0 | 011[2-9]x.S5 )

[jackboot]

I was advise, by Mango on the voip DSL forums - "If you have Tomato firmware, try changing the Unreplied UDP Timeout in Conntrack/Netfilter settings to 10."
It's something to try...
Here (http://www.dslreports.com/forum/r24544129-torontovoipms-Registration) is the thread if you are interested...

Thanks for the tip - sounds like it makes sense. I use Tomato firmware and have been having the same problem where occasionally I cannot register on the Toronto server (but can register on others) and I haven't been able to figure out why.

[apn64]

I wanted to edit the long distance dialing so I don't have to dial 1 before the area code, do I just remove the 1?. Tell me what you think. I just tried this dial plan and it's a huge difference over the longer defaulted one. Calls connect so much quicker now. There's no longer that delay between after you dial and before the first ring. Thanks dude!

(*xxS0 | [8]11S0 | <411:18004664411>S0 | <911:8195957600>S0 | [2-9]xx[2-9]xxxxxxS0 | 1[2-9]xx[2-9]xxxxxxS0 | 011[2-9]x.S5 )

It looks good!

If you (and more importantly, the other users of you phones) don't want to dial 1+long-distance, you can simply remove the whole 1[2-9]xx[2-9]xxxxxxS0 string from the dial-plan.

In that case all North-American calls will have to be dialed as (NPA)-NXX-YYYY i.e. AreaCode-Exchange-Subscriber. You may also need to reprogram any speed-dial or directories in your phones, as well as inform visitors that they can't dial 1+ for long-distance calls.

I'm glad this helped you :)

BTW, there was a typo in my original post; 811 is the VoIP.ms CID test, not the echo test. If [8]11 is the only string, you can change it to 811S0, but also remove the string if you don't expect to use the CID test.

[dorje]

I was advise, by Mango on the voip DSL forums - "If you have Tomato firmware, try changing the Unreplied UDP Timeout in Conntrack/Netfilter settings to 10."
It's something to try...
Here (http://www.dslreports.com/forum/r24544129-torontovoipms-Registration) is the thread if you are interested...

thanks, will give that a try.

[jackie999]

I don't think this was posted already..I only just noticed it in my voip.ms portal:

July 16, 2010, 10:02 pm

Calls to other VoIP.ms numbers dialed from our network are now free for both caller and receiver.

If a customer dials a DID on the VoIP.ms network, the call will be free for both the caller and the receiver of the call.

We would like to take the opportunity to also remind you that there are other methods to reach VoIP.ms numbers for free:

- Each of your DID's can be reached via SIP URI. The format is NXXNXXXXXX@sip.voip.ms (example: 5552221234@sip.voip.ms)

- Each of your DID's can also be reached via SIP Broker. The format is: *9780 + DID number. (Example: If you DID is 5552221234, the format to reach via SIP Broker is *97805552221234

Please note that those features are currently not enabled for international DID numbers.

[newkid]

Has anyone ever successfully put a Linksys SPA2102 behind a router and an access point? Tried everything I could but I could only call out. When people try to dial the number they heard nothing. Any help would be much appreciated!

+1
I am having problems with setting up mine with a DID as well. I had a solid light, made 1 call, then the connection never came back.

[Rekognize]

+1
I am having problems with setting up mine with a DID as well. I had a solid light, made 1 call, then the connection never came back.

Put it under DMZ?

[Lulz]

Am I understanding this correctly?

If you have "Favourite" number and also Unlimited Incoming with your provider, it's cheaper to use DISA as opposed to Callback?

Because with Callback, it charges you for Callback, and also the location you are dialing.

But with DISA it just charges you the location you are dialing.
Is this correct?

[Pete Jones]

Am I understanding this correctly?

If you have "Favourite" number and also Unlimited Incoming with your provider, it's cheaper to use DISA as opposed to Callback?

Because with Callback, it charges you for Callback, and also the location you are dialing.

But with DISA it just charges you the location you are dialing.
Is this correct?

It depends on whether or not you're using premium routing.

The DISA will incur whatever charge (1 cent/min) for dialing into the DID, then the termination charges for the call.

The callback will incur termination charges for calling you, plus termination charges for the call you're making.

--Pete

[Lulz]

It depends on whether or not you're using premium routing.

Thank you for answering, but I am just curious what premium/value has to do with it. Let's assume that all my routing is Premium and nothing is value.



The DISA will incur whatever charge (1 cent/min) for dialing into the DID, then the termination charges for the call.

Ahhh I see. I had no idea dialing my own DID would get charged 1cent/minute. I thought only termination rate to whatever country I am calling would be charged.



The callback will incur termination charges for calling you, plus termination charges for the call you're making.

--Pete

So essentially both DISA and Callback cost the same don't they?
Assuming my cellphone has Unlimited Outgoing and Incoming, it costs the same regarding VoIP.ms?

[flyboy320]

How does the 911e service work? If you don't subscribe to the 911e service, can you still dial 911 and just not have the caller id when you make the call, or can you not dial 911 at all unless you pay for the 911e service?

[mondriano]

Since it seems to be the norm that companies state one offer on the phone, bill a completely different amount, and then claim that they never offered you the first offer in the first place, I am looking for a solution. Is it possible to record a conversation, using the voip.ms service? I see a lot of "exotic" uses of voip.ms in this thread, and am hoping someone can offer advice...

[Pete Jones]

It Seems I wasn't entirely accurate:


Ahhh I see. I had no idea dialing my own DID would get charged 1cent/minute. I thought only termination rate to whatever country I am calling would be charged.

Turns out you're right.

I have a Callback set up on one of my DIDs. I'm using caller ID filtering so any calls from my cell phone to that number will trigger a callback.

So I made a call this way and checked out my call detail records. Here's what I found:

- The initial call to my DID did not register at all.
- The call detail records showed the call coming back to my cell
- The call detail records showed the call I then placed when my callback came in.

So, basically, you only pay termination for the callback, and the call going out.

Now, if I call that number from a phone that isn't set up, then it will just ring through as would my primary number and I get charged appropriately for an incoming call.


So essentially both DISA and Callback cost the same don't they?
Assuming my cellphone has Unlimited Outgoing and Incoming, it costs the same regarding VoIP.ms?

That's a good question. I'll have to set up a DISA and see how the billing comes out to be sure. But it would appear that in the worst case, they would be the same.

--Pete

Edit: So, I set up a DISA and yes, you do get charged for calling your DID and then whatever the termination is for the call you're making. However, the billing for the call to your DID only starts at the point where you have successfully entered the PIN.

[MXMike]

Since it seems to be the norm that companies state one offer on the phone, bill a completely different amount, and then claim that they never offered you the first offer in the first place, I am looking for a solution. Is it possible to record a conversation, using the voip.ms service? I see a lot of "exotic" uses of voip.ms in this thread, and am hoping someone can offer advice...

If you set up X-Lite (http://www.counterpath.com/x-lite.html)to work with your voip.ms account, you can use that software to record a call to your PC, but you have to answer or make the call using X-Lite (not your phone adapter). That's the simplest method I've found so far.

[Poco2]

First off, this thread is amazing, there's a wealth of information that's made it so easy to setup my voip.ms service. Many thanks to everyone who's contributed.

I have it up and running and everything is working perfectly (almost). One small thing... if I call from my voip.ms line to my Bell line and hang up after a ring or two, the Bell line keeps ringing until voicemail catches it. I can't figure out why the call doesn't terminate as soon as I hang up the voip.ms line. Not a big deal, but puzzling. Any ideas?

[flyboy320]

First off, this thread is amazing, there's a wealth of information that's made it so easy to setup my voip.ms service. Many thanks to everyone who's contributed.

I have it up and running and everything is working perfectly (almost). One small thing... if I call from my voip.ms line to my Bell line and hang up after a ring or two, the Bell line keeps ringing until voicemail catches it. I can't figure out why the call doesn't terminate as soon as I hang up the voip.ms line. Not a big deal, but puzzling. Any ideas?

According to this thread here;

http://www.toao.net/25-linksys-pap2t-voip-adapter-review

This tip may help (just a guess, not sure if it will);


You may want to set the CPC delay to 10 and the CPC duration to 0.5. With the default settings, our phones had to be on the hook for an inordinate amount of time before it would actually end the call.

[Barry]

Now that FPL offers long distance at per minute rates does voip.ms offer anything of substance that FPL does not?

[Madwand]

Now that FPL offers long distance at per minute rates does voip.ms offer anything of substance that FPL does not?

well voip.ms has inum numbers (at no charge unless you forward the incoming call to a phone number other than SIP, plus there is also SipBroker.

IMHO, Voip.ms is great and has a lot of features, and even more on the way. FPL is only good for major cities in Canada and local. I have both. :)

[Barry]

Please explain inum and sipbroker (these things are foreign to me) and I would like to understand them.

Also you state that
FPL is only good for major cities in Canada and local.
Please explain why this is the case. FPL allows free calling to major cities in Canada and 2 cents per minute for the rest of Canada and the US. It also allows per minute calling worldwide at listed rates.

From a cost compare point, it would seem to me they are close, but over the long term FPL will win out.

[Mendes]

Please explain inum and sipbroker (these things are foreign to me) and I would like to understand them.

Also you state that
Please explain why this is the case. FPL allows free calling to major cities in Canada and 2 cents per minute for the rest of Canada and the US. It also allows per minute calling worldwide at listed rates.

From a cost compare point, it would seem to me they are close, but over the long term FPL will win out.

I honestly didnt know that FPL did half of what voip.ms did so i really wont comment. But i will now sign up with them to have a look.

I believe what he is referring to up above is the fact that Voip.ms has DID's for a HUGE amount of small cities in which if you are calling other locals it would be cheaper as you wouldnt pay the 2c/s. also 2c/s isn't a great rate

EDIT: so unless the admin panel changes after ordering the $50 SIP config, yes there is lots it doesnt do. there is no even inbound call termination option? ring groups, ivrs, direct to voicemail, filtering etc. Voip.ms has much more technical options. I have seperate subaccounts for all my phones (1 linksys voip phone, 3 softphones, 1 blackberry with blackvoib and 1 nokia e73) and different calls go to different places and different voicemails.

[Shimso]

I was about to signup with freephoneline, but I cancelled after I got the popup saying that their 911 service is forwarded to a dispatch centre where you confirm your address and details, and they you get routed to the closest dispatch centre. From what I'm reading, that's not the case with voip.ms, right? I'm hoping you just dial 911 and it'll connect to the nearest dispatch centre...

And will the number be unlisted?

Finally, my phone doesn't have voicemail/answering machine or caller ID features on it, I understand that I won't get call display, but can I still get voicemails (delivered to my email?)?

Thanks!

So far I have the Cisco ATA and I've created an account.

[jackie999]

....but can I still get voicemails (delivered to my email?)?

Yes, with voip.ms, you set this up in the voip.ms portal. If you get a voice mail it is instantly emailed to you in .wav format.

[Barry]

Oops double post.

[Barry]

Voice mail on Freephoneline allows three options 1, 2 or 3.

1) none: your voicemails won't be transferred to your email
2) forward: your voicemails will be forwarded to your correspondence email and will be deleted from your voicemail box
3) copy: your voicemails will be copied to your correspondence email, but will stay in your voicemail box too, and you will be able to check/delete them using the desktop application

As for the phone book, My number is no longer published with canada411 even though it used to be (I ported Bell to Rogers, then Rogers to FPL).

For 911 I suspect they are the same (offering E911). FPL provides this info http://www.freephoneline.ca/911 I can tell you I called when I first got the phone and I did have a dispatcher. It should be noted that on freephoneline you end up with this info under the change detail tab of your profile "You should ensure your address/location information, updated below, is kept current at all times. In case you are not able to speak during the 911 call, the call taker would dispatch emergency response vehicles to your last registered address. You need to update your 911 Dialing information whenever you move your device to a new location. Note: please allow up to 3 business days for the 9-1-1 data base to be updated."

[Shimso]

Yes, with voip.ms, you set this up in the voip.ms portal. If you get a voice mail it is instantly emailed to you in .wav format.


Voice mail on Freephoneline allows three options 1, 2 or 3.

1) none: your voicemails won't be transferred to your email
2) forward: your voicemails will be forwarded to your correspondence email and will be deleted from your voicemail box
3) copy: your voicemails will be copied to your correspondence email, but will stay in your voicemail box too, and you will be able to check/delete them using the desktop application

As for the phone book, My number is no longer published with canada411 even though it used to be (I ported Bell to Rogers, then Rogers to FPL).

For 911 I suspect they are the same (offering E911). FPL provides this info http://www.freephoneline.ca/911 I can tell you I called when I first got the phone and I did have a dispatcher. It should be noted that on freephoneline you end up with this info under the change detail tab of your profile "You should ensure your address/location information, updated below, is kept current at all times. In case you are not able to speak during the 911 call, the call taker would dispatch emergency response vehicles to your last registered address. You need to update your 911 Dialing information whenever you move your device to a new location. Note: please allow up to 3 business days for the 9-1-1 data base to be updated."

Thanks!


So far I've added the $25 to my account, haven't picked a DID yet, configured my Cisco ATA and plugged in my phone, ethernet and power cables, but I can't get a dial tone and when I try to call my cell nothing happens. I'm guessing its something wrong in how I've set it up but I can't seem to figure out what I did wrong....

[Barry]

http://www.isptelecom.net/index.htm

It would appear this is the company behind voip.ms

[apn64]

http://www.isptelecom.net/index.htm

It would appear this is the company behind voip.ms

Actually, that's not VoIP.ms, but one of the many carriers they use to provide service. They recently informed me that my number is registered to isptelecom.

[EPcjay]

Thanks!


So far I've added the $25 to my account, haven't picked a DID yet, configured my Cisco ATA and plugged in my phone, ethernet and power cables, but I can't get a dial tone and when I try to call my cell nothing happens. I'm guessing its something wrong in how I've set it up but I can't seem to figure out what I did wrong....


Login to your account on VOIP.MS, and see if it shows "Registered" under registration status on the bottom of the main page. If it doesn't, then your box isn't connecting to voip.ms. You'll need to check your settings at that point.

Your suppose to get a dialtone like a regular phone line if its set up correctly.

Also, noticed your signature says looking for a LCD screen for a E71. I saw one on dealextreme for 24.xxUSD.

[fuzzy]

Just found out, you actually can receive phone call over 3G using Acrobits on iphone. Acrobits is using "push notification". Tested it, it works pretty well.


Same here; I can place 3G calls over Rogers, just don't expect to be able to receive calls over 3G since you can't forward the needed ports to receive calls over the 3G internet.

[tdott]

Just found out, you actually can receive phone call over 3G using Acrobits on iphone. Acrobits is using "push notification". Tested it, it works pretty well.

+1 I also set mine up to get push notifications when my home phone rings.

It's a great feature for me, since i never really use my home phone.

[EPcjay]

Just found out, you actually can receive phone call over 3G using Acrobits on iphone. Acrobits is using "push notification". Tested it, it works pretty well.

i've been using nimbuzz for this. It works great on my E71 and my gf's iphone when we are out of the city

[kamran]

I am having occasional problems when someone calls my voip.ms and I don't hear anyone on the call. Does not occur all of the time however quite annoying.

have voip.ms with the linksys pap2t configured behind my dd-wrt router.

followed the settings on the following page
http://www.toao.net/25-linksys-pap2t-voip-adapter-review

only have one line no secondary line

[Lulz]

Hello folks. I just bought SPA2102 from Canada Computers and I have been told it's what I need to get up and running a VOIP line. I have not ever used a router in my life (only had one computer so no need for router). But I am thinking I made a bad choice by not getting PAP2 and a separate router from Linksys or DLink. I am just afraid that a 2-in-1 solution might not be the best choice if I am after reliability.

[duckdown]

Hi folks, just curious, any news on the SMS front? Has voip.ms made any announcements regarding SMS messaging yet? Supposedly the feature is on their horizon but I haven't heard anything since then

Thanks!

[RMSimard]

Hi folks, just curious, any news on the SMS front? Has voip.ms made any announcements regarding SMS messaging yet? Supposedly the feature is on their horizon but I haven't heard anything since then

Thanks!

Contact them and suggest it to them! I asked for SMS service a while back, but it's not yet listed on their wishlist (https://www.voip.ms/m/features.php). The more customers ask for the feature, the more likely they will implement it.

[Lulz]

Finally set up my DID with sisco SPA2102-NA but have an issue still:

My outgoing number is my account number. I tried putting in my phone number in the Display Name field, but it just comes up as this on people's CallerID:

416-444-2222
123123 where 123123 is lower field of CallerID and is my account number

[neo1973]

Finally set up my DID with sisco SPA2102-NA but have an issue still:

My outgoing number is my account number. I tried putting in my phone number in the Display Name field, but it just comes up as this on people's CallerID:

416-444-2222
123123 where 123123 is lower field of CallerID and is my account number
I am having same kind of problem. My phone number didn't even show up. All people can see is 1-200-00.

[junglee]

Login to your account

Go to Main Menu ---> Account Settings and put in your number for 'CallerID Number'

[Lulz]

Login to your account

Go to Main Menu ---> Account Settings and put in your number for 'CallerID Number'

Appreciate it, buddy.
Feel dumb, after checking VoIP interface nearly every single hour for the past month (i like to fiddle around and see what setting does what) I have never noticed that there.

The CallerID will only be for home phone right? Because I have other DID's not connected to any adapter (they are being used for Call Forwarding purposes only) and don't want my home number to be associated with them in any way.

<3 VOIP.ms so much!!!

[Coolin]

My number's going to be ported in two days. :)

Just a question to those who ported from Rogers Digital Home Phone. I couldn't call in ahead of time and cancel, because that would have prevented the porting process. Did you guys get charged an extra month for the cancellation fee? Did you have to return the home phone hardware to them?

[duckdown]

I put my number as 666-666-6666 and it works 90% of the time.. the other 10% of the time it shows my 6 digit account number or comes up restricted

I've been getting a kick out of spoofing the number to appear as someone else too, like say when a bunch of us are sitting around at the bar, I called friend #1 with friend #2's phone number a few times and see what kind of funny argument ensues.. I just login to the voip.ms console via my phones web browser, and then dial with SipDroid

[Aske001]

Hello folks. I just bought SPA2102 from Canada Computers and I have been told it's what I need to get up and running a VOIP line. I have not ever used a router in my life (only had one computer so no need for router). But I am thinking I made a bad choice by not getting PAP2 and a separate router from Linksys or DLink. I am just afraid that a 2-in-1 solution might not be the best choice if I am after reliability.

If you didn't have a router before, the 2102 is a good solution. It acts as a PAP2T-like ATA combined with a simple router to connect your computer. An ethernet switch or even a hub would be simpler in networking terms, but you'd have an extra box to deal with and extra expense for no good reason.

[Lulz]

how do I block CallerID again when dialing from DID itself?

[Lulz]

Hello folks. Does anybody else had an issue today of funds being widthrawn for ridicolously long calls they did not make?

My funds reached $1, so I topped it with $25 again. Today at 5PM, My funds were $26 and now it's 9PM and somehow I used up $5 worth of funds without making any LD calls.

The CDR usage shows that my doctor called my home phone DID which is forwarded to my cell to set up an appointment. I could not hear him for some reason due to voip.ms's fault. It's okay...no biggie. He calls me immediately again and sets up an appointment for me, total call lasts 48 seconds. The CDR shows that correctly.

Then an hour later my mother calls home phone thinking I am home, and it gets forwarded to my cell. Again she can not hear me at all, so she calls me back again and can hear me. I asked if she heard numerous "hello's" I said and she said yes. Again...voip.ms screwed something up. We talk for about a minute and CDR also shows that correctly. The issue is that when my doctor and my mother called first time (when I didnt hear them), the call seemed to continue for 1 hour with doctor and 59 minutes with my mother. There was also 2 other people who CDR shows that I talked to them for 1 hour and 2 minutes.

~$5 was taken from my funds in the last 4 hours.
Did they hire Bell CEO or something?:|:evil:

http://i38.tinypic.com/2prix3t.jpg

[Derka]

Does anyone here have Trixbox running ?

[Derka]

I have Trixbox installed on a dedicated machine.
Trying to figure out what I need next to get it up and running.
I need a DID and what else?

Is it better to run your own PBX or just use the Linksys SPA device?

[djino]

The CDR usage shows that my doctor called my home phone DID which is forwarded to my cell to set up an appointment. I could not hear him for some reason due to voip.ms's fault. It's okay...no biggie. He calls me immediately again and sets up an appointment for me, total call lasts 48 seconds. The CDR shows that correctly.

I'm thinking that when your doctor first called and couldn't hear you properly, that he didn't actually hang up the phone to call you back but actually hit the flash button (like if you had a call on the other line and switching between lines, etc). This would mean the first call is kept on hold while he continued the 2nd call that lasted 48 seconds.


Then an hour later my mother calls home phone thinking I am home, and it gets forwarded to my cell. Again she can not hear me at all, so she calls me back again and can hear me. I asked if she heard numerous "hello's" I said and she said yes. Again...voip.ms screwed something up. We talk for about a minute and CDR also shows that correctly.

I assume the same thing with your mother happened here as well.

djino

[djino]

Does anyone here have Trixbox running ?

I use pbxes.org.

I purchased the SPA2102 from cia.com and have a DID from them as well that is $4/month. From CIA, I get unlimited incoming, unlimited local outgoing, caller ID, voicemail (not used), 911 access, and pay an extra $2/month for Call Waiting (so $6/month total).

Since CIA charges 4 cents/minute for long distance calls Canada/US, I've decided to use voip.ms for termination (as they charge 0.52 cents/minute Canada and 1.1 cents/minute US and charge in 6 second increments).

Using pbxes.org, I setup 2 trunks (one for CIA, and another for Voip.ms). Created a SIP extension with pbxes.org that I insert into the SPA2102. Created another SIP extension to use with X-Lite on my desktop home computer, and a 3rd SIP extension to use on my fring app on my cell phone.

I also created outbound routes on pbxes.org so that if I make a long distance call (a call outside my home Ottawa area) then pbxes uses voip.ms to complete the call, otherwise it uses CIA.

pbxes.org is free as long as you do not require more than 5 extensions, but I recently upgraded to their Soho account that allows for 10 extensions and a bunch of other useful features (at the cost of 3 euros = $4.20 CAD/month).

--

I have not used/heard of tribox to know what you'd need to know to setup. But I recommend using pbxes.org as I have. As it allows me to use the best of both services (CIA and voip.ms) for a cheap multi-featured home phone service.

djino

[Chaseme]

I've run into an issue with people hearing me all broken up. I figured this was bandwidth related and not having the priorities right etc, but the more I look at it the more it seems to be a VOIP.ms issue.

Basically if my connection is under heavy use or not 1 in say 8 calls the caller complains I'm breaking up. If we immediately hang up and call each other back the call is then fine. Anyone experience this? I'm connected to the Montreal server, haven't tried changing it yet to see if it helps.

[Mendes]

Does anyone here have Trixbox running ?

you dont need to unless you are using a lot of the higher functioning trixbox add ons

You can setup subaccounts for all your SIP devices and have most of the trixbox functionality directly from voip.ms admin panel.

did routing, call groups, ivrs etc. I used to run a trixbox rig but dumped it as the voip.ms panel did everything trixbox used to.

[RMSimard]

I've run into an issue with people hearing me all broken up. I figured this was bandwidth related and not having the priorities right etc, but the more I look at it the more it seems to be a VOIP.ms issue.

Basically if my connection is under heavy use or not 1 in say 8 calls the caller complains I'm breaking up. If we immediately hang up and call each other back the call is then fine. Anyone experience this? I'm connected to the Montreal server, haven't tried changing it yet to see if it helps.

Same here, tried on both Montreal and Toronto servers and either connections turn silent within 10 seconds or suddenly come up all garbled like a cooked MP3. Only for the other party, I hear fine on my end. Definitely server-related as I could hear it lately while I was picking up a voip.ms call forwarded to my cell phone. I used to be very happy with voip.ms ...now not so sure. :mad:

[barrist]

Mine's been perfect since getting QOS set up correctly

[Lulz]

I'm thinking that when your doctor first called and couldn't hear you properly, that he didn't actually hang up the phone to call you back but actually hit the flash button (like if you had a call on the other line and switching between lines, etc). This would mean the first call is kept on hold while he continued the 2nd call that lasted 48 seconds.



I assume the same thing with your mother happened here as well.

djino

What you said makes sense...but it I actually hanged up my cell phone after talking to doctor. And of course I am positive he hanged his phone as well. How many people leave their phones on tables un-hanged after they finish talking.

Also regarding my mother, she called from cell phone (as opposed to doctor who called from land line) and she hanged up as well, as did I.

And additionally, someone else called and I did not speak to them (the one that says I dont know who this is and ti says I spoke to them for 58 minutes)

Definitely error here, djino =( $5 lost just like that, I am scared what else is to come. I will try chatting with voip about this now.

[Fire]

you dont need to unless you are using a lot of the higher functioning trixbox add ons

You can setup subaccounts for all your SIP devices and have most of the trixbox functionality directly from voip.ms admin panel.

did routing, call groups, ivrs etc. I used to run a trixbox rig but dumped it as the voip.ms panel did everything trixbox used to.

does voip.ms can do the google voice call? I basically use my trixbox to call NA free by google voice.

[Mendes]

does voip.ms can do the google voice call? I basically use my trixbox to call NA free by google voice.

no it wont let you use other trunks or at least i havnt seen the option

[jmlebeau]

I use pbxes.org.

I purchased the SPA2102 from cia.com and have a DID from them as well that is $4/month. From CIA, I get unlimited incoming, unlimited local outgoing, caller ID, voicemail (not used), 911 access, and pay an extra $2/month for Call Waiting (so $6/month total).

Since CIA charges 4 cents/minute for long distance calls Canada/US, I've decided to use voip.ms for termination (as they charge 0.52 cents/minute Canada and 1.1 cents/minute US and charge in 6 second increments).

Using pbxes.org, I setup 2 trunks (one for CIA, and another for Voip.ms). Created a SIP extension with pbxes.org that I insert into the SPA2102. Created another SIP extension to use with X-Lite on my desktop home computer, and a 3rd SIP extension to use on my fring app on my cell phone.

I also created outbound routes on pbxes.org so that if I make a long distance call (a call outside my home Ottawa area) then pbxes uses voip.ms to complete the call, otherwise it uses CIA.

pbxes.org is free as long as you do not require more than 5 extensions, but I recently upgraded to their Soho account that allows for 10 extensions and a bunch of other useful features (at the cost of 3 euros = $4.20 CAD/month).

--

I have not used/heard of tribox to know what you'd need to know to setup. But I recommend using pbxes.org as I have. As it allows me to use the best of both services (CIA and voip.ms) for a cheap multi-featured home phone service.

djino

Hey, I've been following your advice but I think I'm having problems to configure pbxes.. When I dial from a soft phone, I doesn't hear any ringing nor any voices (if I dial 4443), tried with x-lite and Zoiper.

Here are a few screenshots of my configurations :

pbxes:

Extension: http://skitch.com/jml01/dwkgc/p-b-x-es-your-virtual-pbx-and-crm-on-demand
Trunk: http://skitch.com/jml01/dwk8y/p-b-x-es-your-virtual-pbx-and-crm-on-demand
Route: http://skitch.com/jml01/dwk8j/p-b-x-es-your-virtual-pbx-and-crm-on-demand + http://skitch.com/jml01/dwk8q/p-b-x-es-your-virtual-pbx-and-crm-on-demand

x-lite: http://skitch.com/jml01/dwk82/properties-of-account-1
http://skitch.com/jml01/dwk86/options

voip.ms : http://skitch.com/jml01/dwke9/voip.ms-customer-portal-view-did

Let me know if there is any missing info! Any help appreciated!

Thanks!

JM

[djino]

Hey, I've been following your advice but I think I'm having problems to configure pbxes.. When I dial from a soft phone, I doesn't hear any ringing nor any voices (if I dial 4443), tried with x-lite and Zoiper.

Here are a few screenshots of my configurations :

pbxes:

Extension: http://skitch.com/jml01/dwkgc/p-b-x-es-your-virtual-pbx-and-crm-on-demand
Trunk: http://skitch.com/jml01/dwk8y/p-b-x-es-your-virtual-pbx-and-crm-on-demand
Route: http://skitch.com/jml01/dwk8j/p-b-x-es-your-virtual-pbx-and-crm-on-demand + http://skitch.com/jml01/dwk8q/p-b-x-es-your-virtual-pbx-and-crm-on-demand

x-lite: http://skitch.com/jml01/dwk82/properties-of-account-1
http://skitch.com/jml01/dwk86/options

voip.ms : http://skitch.com/jml01/dwke9/voip.ms-customer-portal-view-did

Let me know if there is any missing info! Any help appreciated!

Thanks!

JM

So far with your extension / Trunk setup, I have "YES" selected for audio bypass, try changing this to see if it makes a difference?

When you check the voip.ms online portal, does it show it being registered?

What happens when you set your Outbound route to Valid for all numbers?

Try removing the Display name info from your X-Lite configuration.

djino
"Let me know what happens"

[jmlebeau]

So far with your extension / Trunk setup, I have "YES" selected for audio bypass, try changing this to see if it makes a difference?

When you check the voip.ms online portal, does it show it being registered?

What happens when you set your Outbound route to Valid for all numbers?

Try removing the Display name info from your X-Lite configuration.

djino
"Let me know what happens"

All done and nothing better :(

[djino]

All done and nothing better :(

Just sent you a PM, might be best to help you this way as per my request.

djino

[eurekabru]

Anybody using Nokia E75?

Everything works fine except that if I leave the phone alone for a few minutes (which is 99,9% of the time) incoming calls give a busy signal.

Voip.ms worked fine using a Linksys PAP2 box.
I switched from unlimitel to voip.ms and Unlimitel was working fine on the Nokia.

Thanks

[djino]

Anybody using Nokia E75?

Everything works fine except that if I leave the phone alone for a few minutes (which is 99,9% of the time) incoming calls give a busy signal.

Voip.ms worked fine using a Linksys PAP2 box.
I switched from unlimitel to voip.ms and Unlimitel was working fine on the Nokia.

Thanks

Which mobile voip app are you using on the E75?

djino

[eurekabru]

Which mobile voip app are you using on the E75?

djino

I'm not even sure what the name is... That's how knowledgeable I am!
I remember downloading and installing an app that installed sip capability to the nokia e75. I set it up and now it's accessed from the home menu. I select the wifi I want to use and then Enable my voip.

Does that make sense ?

[Shimso]

Can anyone help me out a bit with my Cisco PAP2T-NA & Voip.ms?

I don't have a dial tone but the settings and everything seem to be fine. I'm running through the list of things that Cisco says needs to be checked to see if there's a dial tone, and the one thing I'm stuck on is forwarding the ports.

To do so I have to:
-disable DHCP on my ATA (did that)
-assign a static IP to the ATA (I did that, although I'm not sure if I assigned the right number: the ATA's original IP was 192.168.1.8, so I assigned a static IP of 192.168.1.9)
-forward a list of 4 ports in my ASUS router's Virtual Server page (which I did, by putting the ports to be forwarded as the port-range, static IP was inputted as the local IP, local port was inputted, UDP protocol, no protocol number, and I added a brief description)


Supposedly that's all I should need to do, yet when I apply the settings and the router automatically restarts, my internet stops working (even when I unplug the router and modem and restart them).


Can anyone help?

Thanks!

[djino]

Can anyone help me out a bit with my Cisco PAP2T-NA & Voip.ms?

I don't have a dial tone but the settings and everything seem to be fine. I'm running through the list of things that Cisco says needs to be checked to see if there's a dial tone, and the one thing I'm stuck on is forwarding the ports.

To do so I have to:
-disable DHCP on my ATA (did that)
-assign a static IP to the ATA (I did that, although I'm not sure if I assigned the right number: the ATA's original IP was 192.168.1.8, so I assigned a static IP of 192.168.1.9)
-forward a list of 4 ports in my ASUS router's Virtual Server page (which I did, by putting the ports to be forwarded as the port-range, static IP was inputted as the local IP, local port was inputted, UDP protocol, no protocol number, and I added a brief description)


Supposedly that's all I should need to do, yet when I apply the settings and the router automatically restarts, my internet stops working (even when I unplug the router and modem and restart them).


Can anyone help?

Thanks!

from the ASUS router's settings, I would set the Cisco PAP2T to be DMZ (ie, all ports are forwarded to the PAP2T except whatever ports you assign elsewhere).
I'd leave the Cisco to be assigned DHCP (forget static setting)

What happens when this is done?

[Shimso]

from the ASUS router's settings, I would set the Cisco PAP2T to be DMZ (ie, all ports are forwarded to the PAP2T except whatever ports you assign elsewhere).
I'd leave the Cisco to be assigned DHCP (forget static setting)

What happens when this is done?

I'm not near that router right now, but for when I am there, I basically:

-Re-enable DHCP
-Plug in the original IP address into that field (since, from my limited understanding, that'll overwrite the static IP)

And try it out?

Or should I still proceed to the ASUS Virtual Server (port forward) page and forward the ports mentioned in CISCO's troubleshooting file ( http://docs.google.com/viewer?a=v&q=cache:ck_exf_dRAoJ:www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/pap2t/administration/guide/pap2t_user.pdf+http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/pap2t/administration/guide/pap2t_user.pdf&hl=en&gl=ca&pid=bl&srcid=ADGEESjrHtJkSwnU8Km6BcRWJXXoZYe4M0MChI9c3Utl 9O7dnV_WHwhB-LKCVN0jRrf80zQABjf3AfwpMKilNdk8Oyk68IiE25QmX2QqYKd 1A8fMAO8cVAuQsHDe65I7vWYWoVu_hj6l&sig=AHIEtbRhJfpI9HeNWSn4q4FAq08cPabj-A )?

thanks a lot for the help!

[djino]

All done and nothing better :(

For some reason, RDF is not letting me send PMs. I figured my mailbox was full, but I've already gone through deleting a bunch and I still cannot send.

Anyways, I was requesting that you create another "temprorary" pbxes extension and send me the credentials (username/password) so that I may test it out here.


If you can send me an email for now and we'll communicate there -> kdouglas@hotmail.com

djino

[djino]

I'm not near that router right now, but for when I am there, I basically:

-Re-enable DHCP
-Plug in the original IP address into that field (since, from my limited understanding, that'll overwrite the static IP)

And try it out?

Or should I still proceed to the ASUS Virtual Server (port forward) page and forward the ports mentioned in CISCO's troubleshooting file ( http://docs.google.com/viewer?a=v&q=cache:ck_exf_dRAoJ:www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/pap2t/administration/guide/pap2t_user.pdf+http://www.cisco.com/en/US/docs/voice_ip_comm/csbpvga/pap2t/administration/guide/pap2t_user.pdf&hl=en&gl=ca&pid=bl&srcid=ADGEESjrHtJkSwnU8Km6BcRWJXXoZYe4M0MChI9c3Utl 9O7dnV_WHwhB-LKCVN0jRrf80zQABjf3AfwpMKilNdk8Oyk68IiE25QmX2QqYKd 1A8fMAO8cVAuQsHDe65I7vWYWoVu_hj6l&sig=AHIEtbRhJfpI9HeNWSn4q4FAq08cPabj-A )?

thanks a lot for the help!

First of all I assume your setup is as follows. DSL/CABLE MODEM -> ASUS ROUTER -> CISCO PAP2T -> TELEPHONE??

- Log into the CISCO PAP2T settings and Enable DHCP on its WAN and LAN settings
- Log into the ASUS Virtual Server settings and enable DMZ (under the port forwarding section) for the CISCO PAP2T
- Reboot Modem/Asus router/pap2t

djino
"Let me know how that works out|"

[Mendes]

This is a known issue with the Nokia phones.. the WLAN powers off and goes into a sleep mode after ~10 minutes. You can disable this but then your battery will last ~5 hours. If you expect to use it as a home phone, leave the phone plugged in to the power and the wlan wont go into sleep mode.

[Derka]

Alright i need some help

I signed up and have TrixBox running this is for testing purposes and if it goes good i will port over.

Once i registered i added funds and went to configuration and got my configuration settings for Trixbox
So i went to Trunks in Trixbox and copied and pasted my configuration and my dialing rule is 1+NXXNXXXXXX.
What's next do i setup a Route? Any help would be greatly appreciated.


canreinvite=nonat
; nat=yes ; uncomment if behind nat
context=from-trunk
host=toronto.voip.ms
secret=*****
type=friend
username=*****
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you purchased g.729 from Digium
fromuser=******
trustrpid=yes
sendrpid=yes
insecure=port,invite

Register String:
*******:********@toronto.voip.ms:5060

[GT!!]

Can anyone help me out a bit with my Cisco PAP2T-NA & Voip.ms?

I don't have a dial tone but the settings and everything seem to be fine. I'm running through the list of things that Cisco says needs to be checked to see if there's a dial tone, and the one thing I'm stuck on is forwarding the ports.

To do so I have to:
-disable DHCP on my ATA (did that)
-assign a static IP to the ATA (I did that, although I'm not sure if I assigned the right number: the ATA's original IP was 192.168.1.8, so I assigned a static IP of 192.168.1.9)
-forward a list of 4 ports in my ASUS router's Virtual Server page (which I did, by putting the ports to be forwarded as the port-range, static IP was inputted as the local IP, local port was inputted, UDP protocol, no protocol number, and I added a brief description)


Supposedly that's all I should need to do, yet when I apply the settings and the router automatically restarts, my internet stops working (even when I unplug the router and modem and restart them).


Can anyone help?

Thanks!

What model ASUS router? I had trouble with my router (LinkSys) until I upgraded the firmware to dd-wrt. Not only did it fix VoIP, but it also gave me way more features than I had before. You can go here to see if your router i supported: http://www.dd-wrt.com/site/support/router-database. With dd-wrt it just worked, no fiddling with ports and DMZ.

-GT

[Derka]

Alright i need some help

I signed up and have TrixBox running this is for testing purposes and if it goes good i will port over.

Once i registered i added funds and went to configuration and got my configuration settings for Trixbox
So i went to Trunks in Trixbox and copied and pasted my configuration and my dialing rule is 1+NXXNXXXXXX.
What's next do i setup a Route? Any help would be greatly appreciated.


canreinvite=nonat
; nat=yes ; uncomment if behind nat
context=from-trunk
host=toronto.voip.ms
secret=*****
type=friend
username=*****
disallow=all
allow=ulaw
; allow=g729 ; uncomment if you purchased g.729 from Digium
fromuser=******
trustrpid=yes
sendrpid=yes
insecure=port,invite

Register String:
*******:********@toronto.voip.ms:5060


On my main homepage
Main Account Registration Status:
Server: ---
State: No registration found

I just want to be able to make outbound calls to test.

[eurekabru]

This is a known issue with the Nokia phones.. the WLAN powers off and goes into a sleep mode after ~10 minutes. You can disable this but then your battery will last ~5 hours. If you expect to use it as a home phone, leave the phone plugged in to the power and the wlan wont go into sleep mode.

Even with the phone plugged in the WLAN goes into sleep mode. I went in the options to disable it and it's still not working.

Thanks for your helps guys. I give up for now...

[kamran]

issue: bittorrent impacting call quality
This probably does not belong in this thread however I wanted to get peoples input on voip.ms quality while using bittorrent. My experience is call quality is bad while using torrents. The odd thing is that the call quality remains excellent while using newsgroups.
I have DD-WRT installed with QOS enabled, the linksys pap is set to premium via mac accress, and newsgroups/torrents are set to bulk.
My temporary solution is to schedule my torrents between midnight and 6am, and disable during the day - utorrent makes this easy.

The connection detail below was obtained while downloading via newsgroups - not bad
http://myspeedmtl.netcelerate.net/myspeed/db/report?id=102752

[pipolchap]

I've got a PAP2T hooked up to all my landlines. I notice when I use one phone, the conversation volume is loud. However, when 2 people in my house use the same line at the same time using 2 phones, the conversation volume becomes softer. Is this normal?

[jackie999]

issue: bittorrent impacting call quality
This probably does not belong in this thread however I wanted to get peoples input on voip.ms quality while using bittorrent. My experience is call quality is bad while using torrents. The odd thing is that the call quality remains excellent while using newsgroups.


While I can't say what call quality is like with torrents, since I don't use them anymore, I can say that while downloading via newsgroups calls are choppy. QoS never worked for me (I'm using tomato) ..that's not to say it wouldn't work for you..I could just never get the right mix and I read on some forums that QoS doesn't work on voip because it only monitors outgoing..not incoming.
My solution is to pause downloads during a call :)

[jackboot]

I've got a PAP2T hooked up to all my landlines. I notice when I use one phone, the conversation volume is loud. However, when 2 people in my house use the same line at the same time using 2 phones, the conversation volume becomes softer. Is this normal?

I can't say if that is normal or not, but I'm pretty sure you can fix it in the PAP2T settings. I'm NOT an expert, but I think the setting you are after is the "FXS Port Output Gain" under Voice -> Regional -> Miscellaneous. I think you might want to try googling about this setting and see what you come up with. If not that setting, I'm sure that there is one of the other multitudinous settings that will fix the problem for you.

[tchai]

issue: bittorrent impacting call quality
This probably does not belong in this thread however I wanted to get peoples input on voip.ms quality while using bittorrent. My experience is call quality is bad while using torrents. The odd thing is that the call quality remains excellent while using newsgroups.
I have DD-WRT installed with QOS enabled, the linksys pap is set to premium via mac accress, and newsgroups/torrents are set to bulk.
My temporary solution is to schedule my torrents between midnight and 6am, and disable during the day - utorrent makes this easy.

The connection detail below was obtained while downloading via newsgroups - not bad
http://myspeedmtl.netcelerate.net/myspeed/db/report?id=102752

My workaround is slightly different dealing with BT. I set the VOIP encoding to G729a (which lowers the quality of the sound) and set my BT to have an upper limit of 100K. I know it is not ideal. But this is what I am living with right now.
I am with Acanac for DSL.

[Drew_W]

Is there any guide anywhere to setting up a 1 DID, 3 extension w/individual voicemail and IVR office phone system? I'm pretty sure I can figure it out myself, but if there's a guide, it'll make it faster!

[jackboot]

I need to buy another ATA and am planning on a PAP2T - any leads on the cheapest source these days?

[jackie999]

I need to buy another ATA and am planning on a PAP2T - any leads on the cheapest source these days?

It's $53.99 at CanadaComputers...I would stay away from ebay for this since there are 'clone' devices out there.

[jackboot]

It's $53.99 at CanadaComputers...I would stay away from ebay for this since there are 'clone' devices out there.

That's the cheapest I can find too since I can do local pickup and save on shipping...seems like these devices should cost less by now and have more competition from other manufacturers.

Hasn't the PAP2T been discontinued for a number of years now?

[ganiba]

I need to buy another ATA and am planning on a PAP2T - any leads on the cheapest source these days?

Buy this one from Ebay (AG-188N), it can register to two SIP servers simultaneously, and to register to an IAX server, highly needed if the SIP port is blocked by your service provider, such as in hotels or universities:
http://cgi.ebay.com/VOIP-ATA-Router-PSTN-2X-SIP-line-1-IAX2-AG188N-/200398658251?pt=LH_DefaultDomain_0&hash=item2ea8b0dacb

[champlinD]

It's $53.99 at CanadaComputers...I would stay away from ebay for this since there are 'clone' devices out there.
correct. I have two at half the price. Works very well. But I will buy authorized one from CC next time as I didn't know this crap.

[flyboy320]

Spend the money and get it from CC....peace of mind. This way you can also run a separate router (like the Asus WL-520GU) and have tomato firmware on it so you can have a good QOS setup...can make a big difference on call quality.

[typer100]

http://www.swiftgamers.com/ProductDetails.asp?ProductCode=PAP2T-NA

Just over 48$... PM NCIX for cheaper shipping and HST.

[pipolchap]

I've got voip.ms and a PAP2T-NA at home, that is my total knowledge on VOIP.

I'm going to be traveling with a netbook. There are lots of access places in Europe.
Can I make/receive calls with my netbook? If so, what do I need to do?

[Derka]

I've got voip.ms and a PAP2T-NA at home, that is my total knowledge on VOIP.

I'm going to be traveling with a netbook. There are lots of access places in Europe.
Can I make/receive calls with my netbook? If so, what do I need to do?

You need a softphone to make calls on your netbook.
Either X-Lite or Zoipher

[flyboy320]

Anyone hear why the Toronto server was down on Sept. 11th? Had to switch to Montreal for a while until it was back up....

[jackie999]

Anyone hear why the Toronto server was down on Sept. 11th? Had to switch to Montreal for a while until it was back up....


I did same thing..switched to Montreal for a bit...

From voip.ms portal

September 11, 2010, 2:19 pm

Our Toronto equipment is currently unreachable, we're working with the data center to restore service as quick as we can. We'll provide updates as they come.

Update:
Toronto is already UP and Running. We are awaiting for an official statement from the data center with the cause of the outage.

We apologize for the service interruption.

Little more info here...
http://www.dslreports.com/forum/r24785438-voipms-toronto-server-is-DOWN

[Lulz]

Well, I am getting really really angry and frustrated with voip.ms

I am currently in Russia, so I amusing Callback feature to call my russian cell which has unlimited incoming. The problem is that it works only 10% of the time. So out of 10 calls to my Callback number, only 1 or 2 will call me back. Tried both value and premium, tried different handsets, tried different servers, tried tech support. Nothing helps.

But there's another problem, when I do get a callback with dial tone, and start punching in my number, the digits do not get entered (they are called DMTF tones). First I thought it was problem with my new touch screen phone, but after trying another old Nokia, same problem. The numbers just do not get entered and after 30 of so seconds, the call hangs up thinking I did not enter anything. A chat with tech support did not help. He just mentioned that they do not guarantee DMTF tones to be working.

I spent so much money because of voip.ms, had to call Canada directly from Russia paying roughly $1/minute because I did not have calling card handy because I was relying on voip.ms

Definetely not cool.

[ganiba]

Well, I am getting really really angry and frustrated with voip.ms

I am currently in Russia, so I amusing Callback feature to call my russian cell which has unlimited incoming. The problem is that it works only 10% of the time. So out of 10 calls to my Callback number, only 1 or 2 will call me back. Tried both value and premium, tried different handsets, tried different servers, tried tech support. Nothing helps.

But there's another problem, when I do get a callback with dial tone, and start punching in my number, the digits do not get entered (they are called DMTF tones). First I thought it was problem with my new touch screen phone, but after trying another old Nokia, same problem. The numbers just do not get entered and after 30 of so seconds, the call hangs up thinking I did not enter anything. A chat with tech support did not help. He just mentioned that they do not guarantee DMTF tones to be working.

I spent so much money because of voip.ms, had to call Canada directly from Russia paying roughly $1/minute because I did not have calling card handy because I was relying on voip.ms

Definitely not cool.

Try the following options in the Callback setting:
- Delay before callback : 15s
- Response timeout: 30s
- Digit Timeout: 20s or more

Let us know if you have success, or at least an improvement in Callback quality.

[Pete Jones]

The problem could also be with the DTMF tones not being transmitted properly by your cell network.

I've had situations locally trying to log into my Fido voice mail on my cell where the DTMF tones for my password didn't make it through when the signal was marginal. You might want to call someone locally and find out how the tones are coming through.

Likewise, it could just be a matter that the connection voip.ms uses for Russian calls isn't the greatest. The thing is, you would think that calls to Russia and Eastern/Central Europe would be pretty decent because their infrastructure would be somewhat newer what with all the new building and develop going on in the last 10 or so years.

Does kind of suck, though. I suspect you'll probably need to use a cheap calling card to make calls to Canada.

--Pete

[smihaila]

Indeed, it may be a problem with the way the DTMF tones get transmitted. If the audio quality is bad, you should change the settings from "in-band" to "SIP Info", or a third option which I can't remember exactly (RFCxxx).

[blindfury]

Help Wanted! I'm having trouble setting up Cisco PAP2T-NA with an analogue phone to a working condition.

there is no dial tone coming out from the phone, consequently i cannot make any outbound or receive any inbound calls.

the old school phone however, is able to connect to PAP2T, i was able to do the **** command and get the device ip by 110#.

I have also gone into the configuration page via browser, entered the basic information required, and PAP2T showed "online" and registered.

my voip.ms account is active. I am able to call out with softphone on computer.

so, i think the problem has been narrowed down to:

1) PAP2T configurations.
2) problem of the oldschool phone itself (kind of unlikely).
3) problem is generated by the interaction between PAP2T, oldschoolphone, and maybe router.

I have tried everything I can over the period of 5 days but still no luck :( So, $50 bucks to whoever that can make this work, paid on completion.

of course, you will be welcomed and observed in the house with friendliness :D

location: Yonge & Steeles.

[Kwirky]

Help Wanted! I'm having trouble setting up Cisco PAP2T-NA with an analogue phone to a working condition.

there is no dial tone coming out from the phone, consequently i cannot make any outbound or receive any inbound calls.

the old school phone however, is able to connect to PAP2T, i was able to do the **** command and get the device ip by 110#.

I have also gone into the configuration page via browser, entered the basic information required, and PAP2T showed "online" and registered.

my voip.ms account is active. I am able to call out with softphone on computer.

so, i think the problem has been narrowed down to:

1) PAP2T configurations.
2) problem of the oldschool phone itself (kind of unlikely).
3) problem is generated by the interaction between PAP2T, oldschoolphone, and maybe router.

It's something up with your PAP2 configuration. If you'd like to take some screenshots someone here should be able to diagnose - PM me if you don't feel comfortable sharing your config pages for any reason.

[blindfury]

I was going to give up... was going to disconnect RJ45 and home line from the adapter.... then I realized line was plugged into Phone 2 this whole time....... lol :razz:

such a simple and stupid mistake :mad: to think that was the reason why it didnt work this whole time....

guys, i just assumed phone 1 will be in the same orientation as the lights.... so, ALWAYS CHECK WHAT IT SAYS IN THE BACK.

mucho thanks for Kwirky and his intention to help tho!

[Lulz]

realy getting tired of trying to make Callback work.

sometimes it doesn't call back
and most of the time when it calls back, I can not get DMTF tones to send through my Russian cell phone.
it used to work normally without any hitch.
tried literally every single thing there is to try to fix.
thinking of abandoning voip.ms in favour of traditional calling cards.

[ipxxx]

first wanted to say thanks to all those contributing to this thread
signed up yesterday and so far things are looking good.

now for a few questions =)
back on post #634 (http://forums.redflagdeals.com/voip-ms-did-sale-ont-que-878432/16/#post11009809) barrist posted some pictures of his setup.
in my house i have a similar setup - in particular this picture:

http://img14.imageshack.us/img14/2724/img1954z.th.jpg (http://img14.imageshack.us/i/img1954z.jpg/)

the difference is, we have DSL coming in so we need to somehow connect the line coming in to a modem. the DSL is running on a dry-loop. In reply to barrist's post, apn64 posted a picture which i believe will help me out but i wanted to confirm here first.

Essentially, my thought is that I'll cut the line coming in on the punch down block (the right side) leaving it punched down and leaving enough wire to wire up a telephone jack (we'll call this jack 2). The other half of the cut wire coming in from the outside would then go to another jack (we'll call this jack 1). Jack 1 would connect to the DSL modem, which connects to a router, the router then is connected to the PAP2T, and from the PAP2T line 1 connects to Jack 2. This essentially allows me to utilize the phone ports in all the rooms that are currently wired in the punch down block as well as easily switch to plain telephone service by connecting jack 1 to jack 2 directly should i decide to in the future.

Is this correct? =) hope this makes sense. If not...here's a diagram =P

http://img203.imageshack.us/img203/488/setupf.jpg

Thanks for the help!

[Aske001]

Is this correct? =) hope this makes sense. If not...here's a diagram =P

I would do this:
1. Buy a proper punch-down tool if you don't have one.
2. Pull the 2 active wires on the incoming line (Red + Green) from the punch-down block and reconnect them to the secondary line (Yellow + Black).
3. Install a two-line wall jack at your modem location, wiring Line 1 to Red + Green and Line 2 to Yellow + Black.
4. Connect your modem to Line 2
5. Plug your ATA into the Line 1 wall jack wherever it happens to be.

Now you have your VoIP telephone line available at every wall jack, and your DSL signal is also available anywhere in the house you feel line installing a two-line wall jack.

If you ever go back to POTS, reconnect the incoming wires to Red + Green at the punch-down block and reconnect the DSL modem to the Line 1 jack.

[flyboy320]

I tried the new Toronto server (toronto2.voip.ms) and seem to get better ping times. I was wondering why my voice mail greeting wasn't working correct and then I realized that you have to transfer any greetings and messages over if you change servers. I just sent an email to them and they had everything swapped over within half and hour!

[thedonald]

first wanted to say thanks to all those contributing to this thread
signed up yesterday and so far things are looking good.

now for a few questions =)
back on post #634 (http://forums.redflagdeals.com/voip-ms-did-sale-ont-que-878432/16/#post11009809) barrist posted some pictures of his setup.
in my house i have a similar setup - in particular this picture:

http://img14.imageshack.us/img14/2724/img1954z.th.jpg (http://img14.imageshack.us/i/img1954z.jpg/)

the difference is, we have DSL coming in so we need to somehow connect the line coming in to a modem. the DSL is running on a dry-loop. In reply to barrist's post, apn64 posted a picture which i believe will help me out but i wanted to confirm here first.

Essentially, my thought is that I'll cut the line coming in on the punch down block (the right side) leaving it punched down and leaving enough wire to wire up a telephone jack (we'll call this jack 2). The other half of the cut wire coming in from the outside would then go to another jack (we'll call this jack 1). Jack 1 would connect to the DSL modem, which connects to a router, the router then is connected to the PAP2T, and from the PAP2T line 1 connects to Jack 2. This essentially allows me to utilize the phone ports in all the rooms that are currently wired in the punch down block as well as easily switch to plain telephone service by connecting jack 1 to jack 2 directly should i decide to in the future.

Is this correct? =) hope this makes sense. If not...here's a diagram =P

http://img203.imageshack.us/img203/488/setupf.jpg

Thanks for the help!


Where did you buy the MediaSync equipment? That looks really easy to setup...and I want to have something like that I my house. This fall I am going to wire for RG6, CAT6 and telephone.

I was going to install a 110 block...but I just can't figure out how to install it. I will only have 1 VOIP line to start...but i want the capacity to add 4-5 lines if need be, and run them to different parts of the house.

--TheDonald

[duckdown]

Sucks that they still have nothing for SMS

[dgege]

+1

That is exactly how it is setup at my place and it works fine. Took me a couple of hours to replace 5 (6?) phone jacks, but it's worth it.



I would do this:
1. Buy a proper punch-down tool if you don't have one.
2. Pull the 2 active wires on the incoming line (Red + Green) from the punch-down block and reconnect them to the secondary line (Yellow + Black).
3. Install a two-line wall jack at your modem location, wiring Line 1 to Red + Green and Line 2 to Yellow + Black.
4. Connect your modem to Line 2
5. Plug your ATA into the Line 1 wall jack wherever it happens to be.

Now you have your VoIP telephone line available at every wall jack, and your DSL signal is also available anywhere in the house you feel line installing a two-line wall jack.

If you ever go back to POTS, reconnect the incoming wires to Red + Green at the punch-down block and reconnect the DSL modem to the Line 1 jack.

[ipxxx]

I would do this:
1. Buy a proper punch-down tool if you don't have one.
2. Pull the 2 active wires on the incoming line (Red + Green) from the punch-down block and reconnect them to the secondary line (Yellow + Black).
3. Install a two-line wall jack at your modem location, wiring Line 1 to Red + Green and Line 2 to Yellow + Black.
4. Connect your modem to Line 2
5. Plug your ATA into the Line 1 wall jack wherever it happens to be.

Now you have your VoIP telephone line available at every wall jack, and your DSL signal is also available anywhere in the house you feel line installing a two-line wall jack.

If you ever go back to POTS, reconnect the incoming wires to Red + Green at the punch-down block and reconnect the DSL modem to the Line 1 jack.

I'm going to have to re-read this a few times before I can even begin to understand what you're suggesting =P I'm new to all this so I thought cutting the wiring and then just screwing them to two new jacks would be the best/easiest approach. Thanks for the info!


Where did you buy the MediaSync equipment? That looks really easy to setup...and I want to have something like that I my house. This fall I am going to wire for RG6, CAT6 and telephone.

I was going to install a 110 block...but I just can't figure out how to install it. I will only have 1 VOIP line to start...but i want the capacity to add 4-5 lines if need be, and run them to different parts of the house.

--TheDonald

I did not buy it =P It came with the house when we had it built. barrist (the person who originally posted that picture) might know where...

[Phils]

Is there a way to change the greeting on VOIP.MS voice mail? I uploaded a wav file but that was just for people to hear if they dialed in and I didn't want them to leave a message.

[pipolchap]

Is there a way to change the greeting on VOIP.MS voice mail? I uploaded a wav file but that was just for people to hear if they dialed in and I didn't want them to leave a message.

call your number, mailbox#, password#, 0, 1.

[jasuklee]

just started using the service last night.
so far so good!

[redmileage99]

realy getting tired of trying to make Callback work.

sometimes it doesn't call back
and most of the time when it calls back, I can not get DMTF tones to send through my Russian cell phone.
it used to work normally without any hitch.
tried literally every single thing there is to try to fix.
thinking of abandoning voip.ms in favour of traditional calling cards.

I have to use premium route for international calls to make call back work. Value route won't work for call back any more. I really hate this bait and switch.

[vipuls]

Hi,

I received SPA2102 recently. I connected to Airport Extreme router and used 110# on phone to get IP address and it returned 10.0.1.8
Typing 10.0.1.8 doesn't bring menu on the browser.

On firefox it returns message "Failed to Connect
Firefox can't establish a connection to the server at 10.0.1.8"

What could be problem?

Tried with all the browsers and 2 different computers on same routers.

If someone has already answered this problem in past, please point me to that message.

Thanks

Vipul

[typer100]

Hi,

I received SPA2102 recently. I connected to Airport Extreme router and used 110# on phone to get IP address and it returned 10.0.1.8
Typing 10.0.1.8 doesn't bring menu on the browser.

On firefox it returns message "Failed to Connect
Firefox can't establish a connection to the server at 10.0.1.8"

What could be problem?

Vipul

Log in to your router and you'll see all your DHCP client and their IP.

[vipuls]

Thanks. I resolved it. Cisco FAQ guided me on how to enable access to SPA2102 Web access.

Use the analog phone and press ****.
From the IVR (voice menu), press 7932#.
In order to enable this feature, press 1 and 1 again.


Log in to your router and you'll see all your DHCP client and their IP.

[romsan04]

voip.ms does not allow calls to be made internationally, unless you send them ID....very pissed of about it.

[djino]

voip.ms does not allow calls to be made internationally, unless you send them ID....very pissed of about it.

Not completely accurate. You can still place international calls to most countries, but there is a list of countries you cannot call until you provide them with the ID information as requested.

djino

[romsan04]

Not completely accurate. You can still place international calls to most countries, but there is a list of countries you cannot call until you provide them with the ID information as requested.

djino

How safe my ID with them??? With the goverment ID + my address + phone... you can pretty much do anything...... I dont get why VISA cannot be considered as an ID..... visa does have my billing information......

[rivet]

So Kingston, ON DID will still be $2, right, that sucks though. Don't know why some numbers are more expensive than others.

[djino]

So Kingston, ON DID will still be $2, right, that sucks though. Don't know why some numbers are more expensive than others.

Yup. $1.99/month for Kingston.

djino

[Aske001]

I had to scan a copy of my DL and send it to them. I just took a pic and attached it....

I've had an existing voip.ms account for several months, and I've previously paid for it by credit card. But when I went to add money recently, the credit card and Paypal payment options were missing. When I submitted a Support ticket, they came back with a demand for personal photo ID (driver's license or passport) before they will accept new payments. WTF? They didn't send any previous messages or give any explanation about why they need it, and I don't feel comfortable providing copies of my DL or passport information to VoIP company just because they asked for it. You never know where those things end up. How secure are their files?

So this isn't just affecting new sign-ups. And it's nothing to do with requesting access to countries on their blocked list (I haven't). (And BTW, their blocked list includes Australia - what's up with that?).

Perhaps we should avoid voip.ms until they start providing a satisfactory explanation for this sudden need for personal ID.

[Badder]

I just paid without problem. Perhaps your account has been tampered with?

[smihaila]

I can't believe that voip.ms would begin asking for such complicated procedures in order to do a simple payment. If that's true, we should all begin boycotting them, it's not reasonable.

[flyboy320]

I just added $25 today with my CC, and no problem...

[smihaila]

Checked it myself, too. No issue whatsoever. False alarm/isolated case (fraud etc)

[Aske001]

Read the previous postings in this thread - several other people have posted recently that they have been asked for this extra ID information. Mostly it seems to have been new sign-ups, but now it seems they're asking for existing accounts that have been paid by credit card in the past. I can't think of any legitimate reason for voip.ms to be asking for such sensitive personal information.

[jackboot]

Why not just ask them why? We can make guesses here but only they can give you a legit explanation.

My guess is that this isn't their requirement. Why would they go through the hassle if they didn't need to?

[Aske001]

Why not just ask them why? We can make guesses here but only they can give you a legit explanation.

My guess is that this isn't their requirement. Why would they go through the hassle if they didn't need to?

I asked them, and they have responded that they need to verify my personal identity because I'm paying for a company account with my personal credit card in a different name, even though they have been charging the card successfully. And then they enabled it anyway since I complained.

To me that doesn't make a lot of sense. They don't have any option to invoice companies, so how else can you pay for a company account? If you have requirements for additional information on sign-up, why not post them on the site and ask for the information on sign-up? And if you're going to ask for sensitive personal information from existing customers, give a reason, and state what your policies are on protecting personal information.

[jackboot]

So I guess the issue is that if you are paying for a business line with a non-business credit card, they may (or may not?) ask for ID?

I agree that it would be best to request that information during sign-up, but to be totally honest, I don't think the request is unreasonable nor do I think it is something to get worked up about. Just my opinion. Customer service at this company has always been top-notch for me, and I've had to contact them probably much more than the average customer.

As an aside, I pay for a business account with that business' credit card and haven't ever had an issue.

[Aske001]

How often are you asked for copies of your passport or driver's license why you buy something for your business online? Unusual requests require unusual justification, and I don't see that voip.ms has any unusual justification. Nevertheless, I would have responded more graciously if there had been a request with an explanation rather than a demand with no explanation.

[jackboot]

You are right - I've never been asked for dl/passport info when making business purchases online. Maybe this is because my method of payment (credit card) is a business credit card?

[Lulz]

Does anybody know how to add voip.ms to Fring?

Or is there a better SIP client for iPhone and Android? (The reason I ask for both platforms, is because I will be on vacation, so I will use Desire Z along side iPhone 4)

[RMSimard]

Does anybody know how to add voip.ms to Fring?

Or is there a better SIP client for iPhone and Android? (The reason I ask for both platforms, is because I will be on vacation, so I will use Desire Z along side iPhone 4)

Fring -> Add-ons -> SIP -> Other -> Enter voip.ms account details.

However unless they've changed things lately Fring cannot send out DTMF tones after the call has started, rendering voicemail useless and company IVRs even more frustrating (now that's quite a feat!). On my iOS device I've switched to Counterpath Bria (http://www.counterpath.com/bria-iphone-edition.html) (~8$) and everything works smoothly. :)

[EPcjay]

How safe my ID with them??? With the goverment ID + my address + phone... you can pretty much do anything...... I dont get why VISA cannot be considered as an ID..... visa does have my billing information......

I blocked out my # on the drivers license. and birth date. All the other info is public info (can be located from a phone book)

They activated IDD without any questions


Does anybody know how to add voip.ms to Fring?

Or is there a better SIP client for iPhone and Android? (The reason I ask for both platforms, is because I will be on vacation, so I will use Desire Z along side iPhone 4)


I use nimbuzz. Works well on the iphone and S60 phone. Able to send out DTMF tones to access voicemails and such ..

[romsan04]

Recently the quality of service of VOIP.ms went down for me. People calling me and they cannot reach me. When I call someone one, they get 00000000 number.....

Any other good VOIP providers out there???

[Jonavin]

Does anybody know how to add voip.ms to Fring?

Or is there a better SIP client for iPhone and Android? (The reason I ask for both platforms, is because I will be on vacation, so I will use Desire Z along side iPhone 4)

Bria is the best iOS sip client that I've found. It's not free through.

[yu130960]

Recently the quality of service of VOIP.ms went down for me. People calling me and they cannot reach me. When I call someone one, they get 00000000 number.....

Any other good VOIP providers out there???

I agree. I get the Number not in service and dropped calls. After 3 years of VOIP after the vbuzzer debacle on crap customer service, I am thinking of just going back to POTS with Bell.

[Dave98]

Recently the quality of service of VOIP.ms went down for me. People calling me and they cannot reach me. When I call someone one, they get 00000000 number.....

Any other good VOIP providers out there???

I've only been fully ported to voip.ms for about 2 weeks but it's been fairly stable here so far. I think it depends on choosing the right server. I currently use the newyork server
because I get better latency on it compared to the toronto servers.

000000 number is normal sometimes if you use value routing.

[houstondallas]

Recently the quality of service of VOIP.ms went down for me. People calling me and they cannot reach me. When I call someone one, they get 00000000 number.....

Any other good VOIP providers out there???

I'm having the same issue on a premium routing. Inbound calls from Bell landlines, Rogers cell phones sometimes get the "number not in service" message. VOIP.ms has offered no assistance other than to suggest I call Bell/Rogers and tell them that their routing isn't always getting calls to my line. As you can imagine, Bell and Rogers were less than helpful when I inquired.

Anyone have any suggestions for a stable voip provider?

houston

[yu130960]

I'm having the same issue on a premium routing. Inbound calls from Bell landlines, Rogers cell phones sometimes get the "number not in service" message. VOIP.ms has offered no assistance other than to suggest I call Bell/Rogers and tell them that their routing isn't always getting calls to my line. As you can imagine, Bell and Rogers were less than helpful when I inquired.

Anyone have any suggestions for a stable voip provider?

houston

I get the same errors and all voip.ms does is do a test and wait until it goes through. Obviously there is a problem as I never had problems with others, but the fact that it is infrequent is brutal, especially when you get a message like that, essentially saying the number is out of service.

I like everything about their service other then the reliability and would hate to once again move somewhere else....I thought they were supposed to be the most reliable. Who else is there with an established track record.....

[hagbard]

Don't even know what a DID is. That's the problem with their site too...if you don't have any knowledge of VOIP you're just going to be confused by it and walk away to freephone or Magic Jack.

[Mcpie]

Bria is the best iOS sip client that I've found. It's not free through.

Agreed, I run Bria and forward my cell # to it, works quite well.


Don't even know what a DID is. That's the problem with their site too...if you don't have any knowledge of VOIP you're just going to be confused by it and walk away to freephone or Magic Jack.

Without meaning to sound like a ***** , if you don't know what a DID is/have no inclination to learn about it, voip.ms is probably not the best choice for you. http://www.voip-info.org/wiki/view/DID

As for quality issues, I personally have had very few issues with voip.ms, and man are their rates cheap. Note I run a PBX at home for fun, no business purposes, so volumes/experiences may vary. The only issue I've had lately is that on particularly long calls, sometimes I get dropped without warning. Whether that is a hardware issue with my Aastra phones or PBX setup, or voip.ms, remains unclear.

[flyboy320]

I'm finding more and more when I answer an incoming call, it sounds like a fax machine when I pick up. The person on the other end leaves a voice mail because I was unavailable. Might have to write a support ticket for this one.

[hagbard]

Without meaning to sound like a ***** , if you don't know what a DID is/have no inclination to learn about it, voip.ms is probably not the best choice for you. http://www.voip-info.org/wiki/view/DID



Which is pretty much what I'm saying. It closes their market to a few narrow range of geeks, but maybe that's what they want.

[jackie999]

Which is pretty much what I'm saying. It closes their market to a few narrow range of geeks, but maybe that's what they want.

I agree there is a learning curve...BUT the money you save is worth the research. I would think that anyone that sits down and takes the time to read this thread will learn everything they need to know. When I set mine up this thread wasn't available - I read the thread over on digitalhome.
I disagree that magicjack is better - it doesn't work very well and chances are you'll have to format your computer to get rid of it - to me that's more work than setting up a PAP2T.

[hammer]

As a veteran member, I urge any of you thinking about voip to get it and say goodbye to telemarketers and Bell forever! After switching Bell keeps calling me to switch back...now I just direct their call to "this number is not in service"...LOVE IT LOVE IT!!! Never had any problems whatsoever. My wife didn't even know that I had switched from Bell and she mentioned that it sounded better.:) ....all for $8/month average so far and that includes long distance to Europe now and then and lots of long distance to US.

[romsan04]

I'm having the same issue on a premium routing. Inbound calls from Bell landlines, Rogers cell phones sometimes get the "number not in service" message. VOIP.ms has offered no assistance other than to suggest I call Bell/Rogers and tell them that their routing isn't always getting calls to my line. As you can imagine, Bell and Rogers were less than helpful when I inquired.

Anyone have any suggestions for a stable voip provider?

houston

I am actively looking for the different provider right now it looks like Voip.ms has reached its capacity and it is going to get only worse.....
I guess what you pay is what you get..... too bad I like it in the beginning.....

UPDATE:
After 30 min of chatting with voip.ms customer supprot... I am switching to http://www.callcentric.com/, for $6 u get unlimited us/canada... feature list is very compatible.....

Update #2. for $6 is incoming only.... not that good.....
Good bye voip.ms.

[hagbard]

I agree there is a learning curve...BUT the money you save is worth the research. I would think that anyone that sits down and takes the time to read this thread will learn everything they need to know. When I set mine up this thread wasn't available - I read the thread over on digitalhome.
I disagree that magicjack is better - it doesn't work very well and chances are you'll have to format your computer to get rid of it - to me that's more work than setting up a PAP2T.

Not saying MJ is better, its just easer for the layman to understand.

[Aske001]

After 30 min of chatting with voip.ms customer supprot... I am switching to http://www.callcentric.com/, for $6 u get unlimited us/canada... feature list is very compatible.....
Good bye voip.ms.

What was their response? Just the usual? (i.e. "not seeing any problem at present, have you tried a, b, c etc..."). Unfortunately I think that's pretty typical of all the voip companies, and even the regular phone companies. It's not great, but I've seen others ignore trouble tickets completely (Glentel), or screw it up and make things worse (Telus).

[kamran]

I could not find their $6/month plan


I am actively looking for the different provider right now it looks like Voip.ms has reached its capacity and it is going to get only worse.....
I guess what you pay is what you get..... too bad I like it in the beginning.....

UPDATE:
After 30 min of chatting with voip.ms customer supprot... I am switching to http://www.callcentric.com/, for $6 u get unlimited us/canada... feature list is very compatible.....
Good bye voip.ms.

[EPcjay]

I could not find their $6/month plan

cloest thing i can see form their website is
http://www.callcentric.com/dids/personal_unlimited

but that's incoming only.

I've have VOIP.ms, using the value routing since August, and I have no problems. Switched my parents and my sisters line, and its working great.

[EPcjay]

Recently the quality of service of VOIP.ms went down for me. People calling me and they cannot reach me. When I call someone one, they get 00000000 number.....

Any other good VOIP providers out there???

I had this problem, and I thought it was voip.ms, since others have said it's normal. Every outbound call would show 00000, so I was not convinced it's because of value routing. Switched to premium, same problem.

I decided to factory reset the PAPT2, and config everything again, and the outbound callerID worked,

[romsan04]

I updated my post... $6 from www.callcentric.com was for incoming only.
What about vbuzzer.com? They have $4.5 for unlimited local calling.

I am still having issues with Voip.ms, and they are not really helping.
Today my VM stopped working... and after talking to CS they fixed it but I wasn't given any reason why it was not working.....

[paulo500]

Recently the quality of service of VOIP.ms went down for me. People calling me and they cannot reach me. When I call someone one, they get 00000000 number.....

Any other good VOIP providers out there???I too have had the same issues as you. The CID number the other party would see is that 00000000 number or some weird random long distance number. Some issues were probably related to my Acanac DSL internet.
I searched around for any sort of tweaks for my Linksys PAP2T box and I came across this site which helped alot...Helped me understand the ATA more too:
http://www.toao.net/25-linksys-ata-configuration

Since tweaking the thing, i haven't had any Call display issues, and im pretty confident the quality of the calls is much better. Im trying anything i can to keep my voip and bills down. When i was getting all these issues before, the wife was on my case about switch to a more reliable land-line...*shudder* :facepalm: I keep telling her how much $$ we're saving!

Some other useful links for our Linksys PAP2T ATA:
Accessing ATA menu options: http://www.phone4internet.com/linksys_IVR.html
Dial plan options: http://www.toao.net/108-linksys-dial-plan-tips

Hope this helps.

[romsan04]

I too have had the same issues as you. The CID number the other party would see is that 00000000 number or some weird random long distance number. Some issues were probably related to my Acanac DSL internet.
I searched around for any sort of tweaks for my Linksys PAP2T box and I came across this site which helped alot...Helped me understand the ATA more too:
http://www.toao.net/25-linksys-ata-configuration

Since tweaking the thing, i haven't had any Call display issues, and im pretty confident the quality of the calls is much better. Im trying anything i can to keep my voip and bills down. When i was getting all these issues before, the wife was on my case about switch to a more reliable land-line...*shudder* :facepalm: I keep telling her how much $$ we're saving!

Some other useful links for our Linksys PAP2T ATA:
Accessing ATA menu options: http://www.phone4internet.com/linksys_IVR.html
Dial plan options: http://www.toao.net/108-linksys-dial-plan-tips

Hope this helps.

The funny thing was that call display was working fine before that then it stopped working all the sudden. Then I talked to them and apparently I turned it of by dial *67 (*68 turns it on)?? I am 100% sure I did not do it....
Thank you for the links, I'll take a look at it at home.

[romsan04]

What was their response? Just the usual? (i.e. "not seeing any problem at present, have you tried a, b, c etc..."). Unfortunately I think that's pretty typical of all the voip companies, and even the regular phone companies. It's not great, but I've seen others ignore trouble tickets completely (Glentel), or screw it up and make things worse (Telus).

No they actually fixed the problem, but I still dont understand why all of the sudden some features are being disabled/not working when they worked 6 months in a row?? Are they doing this on purpose?? To save some more $$$$???

[barrist]

if you want relatively-worry free phone service you can just plug in.. go with a traditional phone company.

if you take the time to research, set up and be able to interpret your voip.ms account site, you can get a relatively worry-free phone line for (in my case) basically 3-4 bucks a month. i was 6 months in before even needing to replenish the initial $25 deposit.

[romsan04]

if you want relatively-worry free phone service you can just plug in.. go with a traditional phone company.

if you take the time to research, set up and be able to interpret your voip.ms account site, you can get a relatively worry-free phone line for (in my case) basically 3-4 bucks a month. i was 6 months in before even needing to replenish the initial $25 deposit.

It was working fine for the first 6 months.... and now I am getting all this wired issues. I did take time initially to set up everything so it should work as expected, but it is not. Other people having the same issues as me, and they also not a new customers... I didn't change my provider... so nothing basically changed in the last 6 months....

[Aske001]

Bria is the best iOS sip client that I've found. It's not free through.

I've been comparing all the available iOS sip clients I can get with voip.ms, but Bria is currently unavailable from the app store, supposedly due to an update for unspecified reasons which has been going on for some time and has no announced availability date (personally I suspect they ran into trouble either with Apple for supporting voip calls over 3G, or something to do with codec licensing).

Did you have a chance to compare Bria to Acrobits? That seems to be the most recommended of the other voip apps.

NetDial (free) worked fine for me initially with voip.ms, but just quit working at one point for unknown reasons and I haven't been able to get it to work again. It still connects to voip.ms, but it just exits silently after dialing out dial now.

[hagbard]

I'm only up to post #242 and my brain will explode if I can't find a quick answer to this question (that popped in my mind):

Lets say I want a phone number (DID) for my wife, and one for me. I have one set of wireless phones. Would I be able to set them up to give me a different ring or something so that I know if the call is for me or my wife? Or should I be getting another phone?

I'm assuming here that I can run both DID through one ATA device if it has two phone ports?

[Aske001]

I'm only up to post #242 and my brain will explode if I can't find a quick answer to this question (that popped in my mind):

Lets say I want a phone number (DID) for my wife, and one for me. I have one set of wireless phones. Would I be able to set them up to give me a different ring or something so that I know if the call is for me or my wife? Or should I be getting another phone?

I'm assuming here that I can run both DID through one ATA device if it has two phone ports?

If you use the two separate phone ports, then you would have to hook up to two separate phones.

I think it's possible to use different rings the way you are thinking, if you current wireless phone set supports different ring tones by caller ID. You can only register one ATA phone port to one voip.ms account/sub-account. I think it's possible to have more than one DID for an account/sub-account, but I'm not sure that you can get different specific caller IDs for them. However if you can't, you could call-forward one DID to the other. and set the caller ID override in the Call Forward settings so that it uses a specific caller ID. You'll lose the ability to see the original Caller ID on your wireless phone handset of course.

[Hugh]

I'm only up to post #242 and my brain will explode if I can't find a quick answer to this question (that popped in my mind):

Lets say I want a phone number (DID) for my wife, and one for me. I have one set of wireless phones. Would I be able to set them up to give me a different ring or something so that I know if the call is for me or my wife? Or should I be getting another phone?

I'm assuming here that I can run both DID through one ATA device if it has two phone ports?

I understand your head exploding. This telephone stuff is full of odd characteristics due to the 100+ year historical baggage supplemented by computer networking oddities.

Any ring tone generation depends on your device, not on what your ITSP (voip.ms) provides you. So it depends on what you can hack together.

I don't know if any ordinary device you buy can do what you want. As Aske001 says, something like a PAP2 has two phone lines and you could have each DID activate a different phone. Or even hook up a two-line handset.

In the limit case, you could set up an Asterisk box to do almost anything. Asterisk is open-source PBX software. It can run on something as small as a wireless router but it normally runs on a PC (as a phone server, you'd want it on all the time). Do you need a new hobby? :-)

[hagbard]

In the limit case, you could set up an Asterisk box to do almost anything. Asterisk is open-source PBX software. It can run on something as small as a wireless router but it normally runs on a PC (as a phone server, you'd want it on all the time).

Really? Can I get it on my Linksys WRT54G?


Do you need a new hobby? :-)

I do. I'm really bored. :razz:

[Hugh]

Really? Can I get it on my Linksys WRT54G?


Asterisk can work on WRT54GS (twice the memory of the WRT54GL) and other routers. Not that I've done it. I've seen it at a couple of trade shows at the Toronto Asterisk Users Group booth. Not all features were enabled. I think that it was running the phone menu version of Adventure. One limitation: it didn't have the CPU power to transcode more than one stream.

I think that it was running under OpenWRT http://openwrt.org/. You could look to see if they have current versions of Asterisk in their repository.

OpenWRT is portable. There are more powerful routers available now that will run it. I would guess (don't know) that the Netgear WNDR3700 on sale at Future Shop would be a good platform. Another good platform might be one of the Plug computers (like the $30 Seagate DockStar). But it is probably better to figure out Asterisk on a PC first.

I've never set up an Asterisk system but I was interested enough to attend a few TAUG meetings http://taug.ca/

[jackie999]

I'm only up to post #242 and my brain will explode if I can't find a quick answer to this question (that popped in my mind):

Lets say I want a phone number (DID) for my wife, and one for me. I have one set of wireless phones. Would I be able to set them up to give me a different ring or something so that I know if the call is for me or my wife? Or should I be getting another phone?

I'm assuming here that I can run both DID through one ATA device if it has two phone ports?


I have the 6 handset panasonic phones - the money I've saved with voip covered the cost of the phones in the first few months. I have mine set to a different ringtone dependant on who is calling..close family has one tone, others in my phone book another..and a generic ring for calls not in my phonebook.
There is also this feature (described in post #66)

Here's another option for those that want to use the 2nd port on a PAP2T;

Setup a FREE sub-account and (w/o buying a 2nd DID) use it for outgoing calls, or to simply manage/route incoming calls e.g. basement, home office, kids room (you can setup a rule that says "send all calls from these numbers to that phone" etc).

Hope this helps...

[hagbard]

I have the 6 handset panasonic phones - the money I've saved with voip covered the cost of the phones in the first few months. I have mine set to a different ringtone dependant on who is calling..close family has one tone, others in my phone book another..and a generic ring for calls not in my phonebook.
There is also this feature (described in post #66)



I have my Panasonic phones doing this now, using Cogeco phone. But I'd like to have separate numbers. Guess I'll go with an additional phone. And about this 800 service, is it more than regular DID?

One more thing, who has be best deal on a two line ATA device.

[jackie999]

I have my Panasonic phones doing this now, using Cogeco phone. But I'd like to have separate numbers. Guess I'll go with an additional phone. And about this 800 service, is it more than regular DID?

One more thing, who has be best deal on a two line ATA device.


Not sure of the 'best deal' on the ATA - but CC still sells for $54
http://www.canadacomputers.com/product_info.php?cPath=30_414&item_id=014833

I would avoid ebay for this device if I were you - you don't want one of the overheating clones that are around.

[krazykoncepts]

For those of you without a current phone (hardware) do you suggest getting a regular phone + adapter or buy a voip phone?

Looking to have multiple voip lines for office use.

[Mcpie]

I've been comparing all the available iOS sip clients I can get with voip.ms, but Bria is currently unavailable from the app store, supposedly due to an update for unspecified reasons which has been going on for some time and has no announced availability date (personally I suspect they ran into trouble either with Apple for supporting voip calls over 3G, or something to do with codec licensing).

Did you have a chance to compare Bria to Acrobits? That seems to be the most recommended of the other voip apps.

NetDial (free) worked fine for me initially with voip.ms, but just quit working at one point for unknown reasons and I haven't been able to get it to work again. It still connects to voip.ms, but it just exits silently after dialing out dial now.

I compared with Acrobits. I went with Bria ultimately but Acrobits was a great client. Can't really go wrong with either. Audio quality was great with both-- even better with G729 (I only shelled out for this with Bria).

I have a jailbroken phone so I tried both for free before deciding on Bria and buying that solution. Not technically legal, but I don't consider it unethical if I ultimately end up purchasing the one I like. YMMV.

[hagbard]

Not sure of the 'best deal' on the ATA - but CC still sells for $54
http://www.canadacomputers.com/product_info.php?cPath=30_414&item_id=014833

I would avoid ebay for this device if I were you - you don't want one of the overheating clones that are around.

For me it would be more expensive than buying at Dell.

CC = $54+$7 shipping = $61

Dell = $60 and free shipping.

Anything better out there?

[Mcpie]

For those of you without a current phone (hardware) do you suggest getting a regular phone + adapter or buy a voip phone?

Looking to have multiple voip lines for office use.

I really, really like the Aastra 6757i CT

http://www.aastra.com/aastra-6757i-CT.htm

I bought mine in the States (telephonydepot) http://www.telephonydepot.com/Catalog/Aastra-Phones/Aastra-57i-CT

But I work for a company that has tons of people traveling across the border all the time, so it was easy to order to an office and have a friend bring it back. It's a good bit more expensive in Canada, at least when I checked.

[typer100]

I have my Panasonic phones doing this now, using Cogeco phone. But I'd like to have separate numbers. Guess I'll go with an additional phone. And about this 800 service, is it more than regular DID?

One more thing, who has be best deal on a two line ATA device.

50$ http://www.onhop.ca/product/10182894/cisco-small-business-pro-pap2t-internet-phone-adapter-with-2-voip-ports-voip-phone-adapter-ethernet-external?f=Shopbot

PM NCIX

[krazykoncepts]

voip.ms doesnt offer 416 or 905 numbers...has anyone used another provider just to obtain their desired number/area code if so who? and then port it to voip.ms

[lanslord]

I compared with Acrobits. I went with Bria ultimately but Acrobits was a great client. Can't really go wrong with either. Audio quality was great with both-- even better with G729 (I only shelled out for this with Bria).

I have a jailbroken phone so I tried both for free before deciding on Bria and buying that solution. Not technically legal, but I don't consider it unethical if I ultimately end up purchasing the one I like. YMMV.
I have been using Nimbuzz app on iphone. It works well and it is free. The only issue is that it does not support DTMF.

[Aske001]

I have been using Nimbuzz app on iphone. It works well and it is free. The only issue is that it does not support DTMF.

Thanks to both you and Mcpie for your comments.

It looks like Nimbuzz is a separate voip service, not general SIP client that I could use with voip.ms.

The lack of DTMF would also be an issue, as one of the main uses for me is when I'm traveling to dial back into home and office voice mail. I already have enough trouble with DTMF on voip.ms, trying to find a happy medium between over-strict settings in my SPA2102 and random tones generated accidentally by false DTMF recognition during conversation.

What advantages did Bria have over Acrobits? The main one I see offhand is the ability to have multiple accounts (which the free NetDial app has).

[Intricated]

voip.ms doesnt offer 416 or 905 numbers...has anyone used another provider just to obtain their desired number/area code if so who? and then port it to voip.msI bought a 416 number that was on Fido PAYG, and ported it easily to voip.ms by following their instructions. Whole process took about 10 days if I recall.

[Mcpie]

Thanks to both you and Mcpie for your comments.

It looks like Nimbuzz is a separate voip service, not general SIP client that I could use with voip.ms.

The lack of DTMF would also be an issue, as one of the main uses for me is when I'm traveling to dial back into home and office voice mail. I already have enough trouble with DTMF on voip.ms, trying to find a happy medium between over-strict settings in my SPA2102 and random tones generated accidentally by false DTMF recognition during conversation.

What advantages did Bria have over Acrobits? The main one I see offhand is the ability to have multiple accounts (which the free NetDial app has).

Honestly? I liked the UI better. They are both excellent apps-- to the point where if someone declared one the winner over the other, I would argue you're talking to a fanboy.

The other reason I went with Bria is that they have a rock solid company behind them, I'm familiar with/use their PC softphones, and they are very well known in this space and really put this softphone together in a very short period of time/made it rock. Again though, Acrobits was/is a great client, there really is no going wrong with either.

For what it's worth, I've had 0 issues with DTMF with my home PBX + Bria, or with my Aastra 6757i CT. The only problem I've had is that, during long calls ~45+ mins, sometimes I'll get inexplicably dropped. Then again, that happens with my real cell carrier too :) DTMF is a must have for me, as 98% of my meetings require conference bridges and they are, by far, the biggest suck of my telephone minutes.

Here is a decent comparison of all the softphones, though a bit dated: http://nerdvittles.com/?p=686

One thing I never did, which I'm kicking myself for, is compare battery drain. Bria running 24/7 on my iPhone drains the battery pretty harshly.

[hagbard]

Hey guys, so I got the PAP2T from Dell, now what? Someone said something about configuration files or help files on voip.ms but I can't find them.

[djino]

Hey guys, so I got the PAP2T from Dell, now what? Someone said something about configuration files or help files on voip.ms but I can't find them.

now login to your voip.ms account and gather your SIP account credentials, then log into your PAP2T and insert them. DONE!

djino

[flyboy320]

Hey guys, so I got the PAP2T from Dell, now what? Someone said something about configuration files or help files on voip.ms but I can't find them.

Couple of links that may help...

http://www.toao.net/108-linksys-dial-plan-tips

http://blog.2talk.co.nz/pap2t.html

http://www.toao.net/25-linksys-ata-configuration

[marcR]

Couple of links that may help...

http://www.toao.net/108-linksys-dial-plan-tips

http://www.toao.net/25-linksys-ata-configuration

I was having a lot of dropped calls and a lot of issues with callers not getting through until I followed the instructions on the 2nd link, it doesn't take much time and is well worth it. The author posts a lot on another forum of a digital home nature and is very helpful. Probably the biggest thing was updating the firmware on my ATA.

[Phils]

Hey guys, so I got the PAP2T from Dell, now what? Someone said something about configuration files or help files on voip.ms but I can't find them.

They call it a "sample configuration" file. Once you log in it can be downloaded from the support menu (I think). It shouldn't be too hard to find. Also, it's just a few settings - nothing dramatic.

[hagbard]

They call it a "sample configuration" file. Once you log in it can be downloaded from the support menu (I think). It shouldn't be too hard to find. Also, it's just a few settings - nothing dramatic.

I did that, at first it didn't work but now I have a dial tone...so I guess I'm off to the races. ;)

Oh hey, how do you update the firmware? Wonder why they sent me a device with old firmware when the newest is already four years old? And if I do update it, will it keep my current configuration?

[Phils]

Dialing out is one thing. Getting your DID to work is quite another. Just make sure that your phone rings when people call you.

[hagbard]

I was having a lot of dropped calls and a lot of issues with callers not getting through until I followed the instructions on the 2nd link, it doesn't take much time and is well worth it. The author posts a lot on another forum of a digital home nature and is very helpful. Probably the biggest thing was updating the firmware on my ATA.

"Important note to users of Tomato firmware: This isn't strictly related to Linksys ATAs, but since many users of Linksys ATAs have a router with Tomato firmware, we feel it prudent to mention a bug that prevents VoIP devices from working properly upon reboot of the router. To work around the bug, simply enter Conntrack/Netfilter and set Unreplied UDP Timeout to 10."

Huh, where? In the router or the ATA?? And where?

[hagbard]

Dialing out is one thing. Getting your DID to work is quite another. Just make sure that your phone rings when people call you.

And how do u do that? They should do some Youtube videos.

[jackie999]

"Important note to users of Tomato firmware: This isn't strictly related to Linksys ATAs, but since many users of Linksys ATAs have a router with Tomato firmware, we feel it prudent to mention a bug that prevents VoIP devices from working properly upon reboot of the router. To work around the bug, simply enter Conntrack/Netfilter and set Unreplied UDP Timeout to 10."

Huh, where? In the router or the ATA?? And where?


The setting is in router running tomato firmware..under advanced.

As for making sure your DID is working - use a cell phone and call yourself. Make sure you've set up your DID info in voip.ms portal.

[hagbard]

The setting is in router running tomato firmware..under advanced.

As for making sure your DID is working - use a cell phone and call yourself. Make sure you've set up your DID info in voip.ms portal.

Thanks. Fixed.