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742K views 3K replies 244 participants last post by  Satellite97 
#1 ·
Thanks to some of the other threads on this forum, I decided to try and set up my own voip, and signed up with voip.ms. I managed to set up a PAP2T in about 20 minutes, and, amazingly, I was calling across the country immediately. So far, voice quality is excellent. We'll see how it goes over a period of time and I will give an update in six months or so. Assuming all goes well, it begs the question --'Why doesn't everyone do it this way"?
 
#28 ·
I signed up with voip.ms after a trying out vbuzzer. I like both of them so far. Good to have many options. Voip.ms has nice features, and I like the quality of the LD phone calls so far as a paygo setup.
 
#29 ·
So based on what I've read here and at dslreports, I'm taking the plunge w/ voip.ms

I purchased a PAP2T this morning, so once that's delivered next week, I'll be setting up an account w/ voip.ms.

The only potential glitch I see ahead is that my wife wants to keep our number, so it may take a few weeks to get it ported, and since we're going from one VOIP provider to another, I assume that we'll have to setup everything on the very day the number is ported.

At least this ensures that we have service until transition day and it means I don't have to consider introducing the PAP2T to my Rogers VOIP box.

I purposely went with a Linksys ATA since they're inexpensive, ubiquitous, I don't need the FXO capability and finally, I figured it will smoothly integrate with my WRT54GL/ddwrt device.
 
#32 ·
99semaj said:
If I have a voip.ms account configured with one DID, can I have multiple IP phones that will behave like analog sets? (i.e. shared call capability which all ring at once, multiple sets can talk on the same line at once?)
You can set up ring groups, however they won't act like shared line appearances. Once you answer a call with an IP phone, the way another phone would join it is for the first phone to conference or transfer the call. If you pick up another IP phone without transferring or conferencing it in, you get a dial tone. If you want shared line appearances, you would want to use an Asterisk server.

99semaj said:
Do I need to use an asterix server to get other call features like call forwarding and conferencing? How about visual caller ID?
No, those features do not require an Asterisk server.

99semaj said:
Aastra 9112i
WANT!!!!!! [wipes drool off keyboard]

-Mango :D
 
#38 ·
Glad to hear the 9112i is working well for you. Did you end up setting up an Asterisk server? I've never actually set up shared call appearances and am curious as to how easy it was.

If it's not too much trouble, could you post some screenshots of the admin interface?

-m.
 
#40 ·
hi 99semaj,

Not sure why the IP phones were necessarily better than an ATA. If you bought 4 IP phones for $99 each, it would mean ~ $400 for making your whole house with voip.

I bought and setup a discounted ATA and connected a 3-handset cordless DECT 6.0 phone all over the house. Total cost under $90 including the cost of the ATA. All this with voip.ms and vbuzzer. If I want more handsets, I could buy additional DECT 6.0 handsets and expand my phone system to up to 6-handsets. But that is overkill.

I am keeping my Bell landline because I realized that I needed the landline as a backup, and for home security systems e.g. ADT, which requires a reliable phone line i.e. landline. A cellular based security system is more expensive if available (so this was ruled out), and some security systems do not work well with voip phone lines as reported by many users.

I believe IP hardware phones like yours are great if one has the money. The above works well for me and so far, I'm pretty happy with my low-cost setup. I like yours as well, but its costlier.
 
#41 ·
Hi Lawman:

You raise good points, and I did use an ATA for a full five-years before opting for this route.

For me, the main advantage is the number of channels. With an ATA, you are bound by the old telco paradigm that a line can be "busy" if any extension is in use.

With a pure VOIP solution, any free extension can recieve a call or make a call regardless of what the other sets' status is. A family of four could make four outgoing calls, recieve four incoming calls, etc, without every messing around with call waiting.

Also, there's some utility in having station-to-station calling, but that's pretty minor in a small home like mine.

WRT to the alarm system, I have an IP communicator on mine, so telephone doesn't play into the equation.
 
#43 ·
True if the line is busy then extension is in use and no one can use it. That's why I set up another ATA within the house. One ATA for lower floor and another ATA for upper floor. Both uses the same ports as each ATA is a different MAC addr. So, if someone upstairs uses the phone, another person downstairs can also call out.

I like your setup - down the road, if money is not a problem, I would not mind trying it out.
 
#45 ·
holymoly said:
If you are porting to voip.ms it takes longer for Canadian customers. Mine took eight weeks back in August.
My port hit eight weeks yesterday and I don't even have a FOC yet. I've also been waiting on a DID assignment that hit six weeks on Wednesday. My first DID was assigned in a few hours, and I suspect the length of time on this one is a casualty of the issues VoIP.ms was having with their ticket system. Anyway, they're telling me I should be able to use it by Tuesday of next week which will be one day short of seven weeks.

99semaj said:
Surprise! The Montreal server had two less hops (12 vs 14) and the ping time was HALF versus Toronto
Heh. Curious the way internet routing sometimes works. One thing I like about VoIP.ms is the fact that we have choice in the server we connect to. The packet loss issue I was having wasn't VoIP.ms' fault as it wasn't their server causing the packet loss, and I didn't have packet loss with another ISP with different routing. Not their fault, not my fault, and nothing either of us can do about it except wait. In this case it took about a week and a half for the problem to be sorted, but I didn't care because I just used a different server.

m.
 
#46 ·
LNP Wait Time

Glad I'm not the only one with LNP (porting) issues...

The service from voip.ms seems professional but their LNP process is killing me... I've been waiting since November 2008 to port my Toronto number. Unfortunately, I'll be ditching them if they cannot get it done before the end of this month. :(
 
#47 ·
Call ID / VoIP BYOD in Vancouver

Wow..had a chat with VoIP.ms support and found out that call display is charged extra PER USE...
I have never seen somthing that strange. They told me that if I receive a call more than 1 time a week I would not be charged for the additional call id showings, but the next week I would have to pay for it again.

Sounds like a strange concept. Too bad, because I wanted to go with them.

Anybody that can reommend a VoIP provder for Vancouver, BC that is around $10/month and includes local calls and call ID plus BYOD. Voicemail needs to be disabled.

Any suggestions, because netfone is not on my list after they dropped the ball on me before I even got started with them.

Thank you,
schag001
 
#48 ·
Wow..had a chat with VoIP.ms support and found out that call display is charged extra PER USE...
I have never seen somthing that strange. They told me that if I receive a call more than 1 time a week I would not be charged for the additional call id showings, but the next week I would have to pay for it again.
Yeah but it's only 1.2 cents per lookup in the LIBD/CNAM database .... like wow if that's too expensive for you then I don't know what to say.

CallerID Name Lookup
For Callers with a US/Canadian CallerID number, perform a lookup in the LIBD/CNAM database to find the name matching the number. When activated, the system will display the result of this query in the CallerID name portion of the CallerID, leading to a "CallerID name" <5551231234> format when people call your number.
 
#55 ·
Newbie here.. I'm fed up with paying too much for Phone/Internet/Basic cable.. thanks to DH Forum - I'm on my way to getting free (HD OTA) tv and cheaper phone service.. Too bad I haven't found a more economical internet yet! :)

Originally considering Vonage/Primus... but then stumbled across this thread which caught my interest... unfortunately, voip.ms website isn't newbie friendly... maybe it's just me but it's doesn't provide enough info on how to get started (what you need, etc).. anyway glad I'm getting bits and pieces here from this thread. :)

So from what I can understand (someone correct me please if I'm wrong..):
- if I want to continue using my existing regular telephones, I need to get one of those Voice Gateway/routers.. (any recommendation)?
- if I don't want to use/port my old tel #, sign up for voip.ms to get the DID # for the location I want, plus options like 911, call display...

Is it really that easy? If so, I guess the only advantage of going with Primus or Vonage (and paying more $/mo) is the support for the device and not having to purchase devices??

Few things I'm not clear on..
- is Visual Call Waiting feature... is this supported?
- has anyone successfully used this for Home Alarm systems??
- I understand there's different charges for different quality. I was looking at the order page for DID # which says 1.49Cent/min + $1.99/mo + 50Cent setup. But then looking at the termination rates for Toronto - it says $0.0049/min value or $0.0125/min premium... ????

Thanks.
 
#67 ·
- has anyone successfully used this for Home Alarm systems??
I can speak to this one with authority since I manufacture alarm systems in various markets globally.

Ironically, the older pulse formats work best with VOIP connection, and they work flawlessly. (These are similar to the old dial-style phones, and transmit more slowly than more modern alarms and cannot send as many messages)

FSK formats work equally as well on stable VOIP lines. FSK is best characterized as sounding like a fax or modem....that "white noise" sound. SIA is a common name for this method.

The most problematic has been DTMF (touchtone, etc) formats. Honeywell/ADEMCO Contact ID uses this. VOIP systems just seem to want to handle touchtones differently, and the speed and length of each tone causes problems.

Why do you say that? As far as I've been able to tell, it works just like any other hosted voicemail system I've ever used, with the useful addition of voicemail-to-email.
The access code to dial in is almost impossible to remember!
 
#56 ·
it just hit me... the 1.49Cents/min is the rate I'd be paying (plus extra charges like Call display) when receiving incoming calls..

And the Termination rate 0.49Cent/min or 1.25Cents/min is what I'd be paying to make outgoing calls? I'm assuming when you sign up you can select the quality you want to use when making outgoing calls?

If I'm understanding this correctly, why is there 2 options for outgoing quality and only 1 option for incoming? What is the quality of the incoming call?

sorry for the newbie Q's..
 
#57 ·
aooa,

I'm new to voip.ms too, and there's a lot of material to go through.

the 1.49Cents/min is the rate I'd be paying (plus extra charges like Call display) when receiving incoming calls
Yes, this is the per-minute plan pricing, or you can have unlimited incoming for $8.95/mo

Termination rate 0.49Cent/min or 1.25Cents/min is what I'd be paying to make outgoing calls? I'm assuming when you sign up you can select the quality you want to use when making outgoing calls?
Yes, these are the stated rates for Canada on the Value and Premium routing options respectively, however, the current value rate is actually 0.51 right now. Calls to the US are 1.05c/min (value) and 1.25c/min (premium). Note that you also pay for toll-free termination at a rate of 0.25c/min

There are two routing options to provide different levels of quality at different prices. I recall reading that CallerID is guaranteed on premium termination but best effort on value. You select your preferred routing option in your account preferences, but there are also dialling prefixes to override that setting and force a specific routing.

One thing I find odd is that it's cheaper to call the UK on premium (1.14c/min) than it is to call within North America (1.25c/min)

One more thing on the routing. Fearing the worst, I initially setup everything for premium routing. The quality was great, but I recently switched my NA calls to value and not really noticed any difference.

Finally, note that I've only been using voip.ms on termination; testing an alternative to my soon-to-be-gone Rogers VOIP service. I'm about to take the plunge into origination, and for that I have a bunch of questions myself.
 
#58 ·
voip.ms for Origination (DID)

I've been testing voip.ms termination for a couple of weeks (using PAP2T-NA) and I'm very happy with the results.

I was already thinking about DID porting, but Rogers' announcement to drop its VOIP offering (4/22/09) is going to accelerate my decision. However, before taking the plunge, I have some (noobie) questions about voip.ms DID services;

1. Voice Mail: Is voip.ms or my ATA providing this and where are the messages stored? Is there a mailbox storage (minute) limit etc?

2. VMWI: I recall reading something about voip.ms and/or my ATA providing stuttered dialtone. My (Nortel/Panasonic) phones also have the (CLASS) VMWI feature to flash the msg waiting light. Is this an option on voip.ms?

3. SCWID aka Visual Call Waiting: My phones can also display the CallerID of incoming call-waiting calls. Is this an option on voip.ms?
 
#64 ·
I've been testing voip.ms termination for a couple of weeks (using PAP2T-NA) and I'm very happy with the results.

I was already thinking about DID porting, but Rogers' announcement to drop its VOIP offering (4/22/09) is going to accelerate my decision. However, before taking the plunge, I have some (noobie) questions about voip.ms DID services;

1. Voice Mail: Is voip.ms or my ATA providing this and where are the messages stored? Is there a mailbox storage (minute) limit etc?

2. VMWI: I recall reading something about voip.ms and/or my ATA providing stuttered dialtone. My (Nortel/Panasonic) phones also have the (CLASS) VMWI feature to flash the msg waiting light. Is this an option on voip.ms?

3. SCWID aka Visual Call Waiting: My phones can also display the CallerID of incoming call-waiting calls. Is this an option on voip.ms?
Voicemail is done by voip.ms It ain't the prettiest setup I've seen though

VMWI is supported.

SCWID is also supported.
 
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