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Lots of folks around here seem to be pretty excited about Voip.ms but it seems pretty intimidating for many of us including me.

Would anyone care to volunteer to put together a how-to of how to go about getting hook-up (okay a dummies guide)?

I would be happy to be the test guinea pig.

The goal would be to put together a step by step approach that we could then post on the site for future users.

Please contact me if you are interested.

Hugh
 

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I just did it last night, so I'll see how much of it I remember. I think you can probably change the order of these around a bit, but I think I did it like this...

1) Sign-up for a voip.ms account. Once you sign up you will be able to get into the members area. It will put you into the Account Information area. If you are not there, using the menus at the top of the screen go to Main Menu, Account Information.

2) Configure your ATA.
a) Connect you ATA to your router (before or after, depending on the model)
b) Under the Voip Servers section of the Account Information, click on the link for the configuration examples.
c) Pick your type of Adaptor. I use a Linksys PAP2T-NA, so I choose the Linksys example.
d)follow the steps in the configuration example
- At the top, select the closest server to you (I use Toronto)
- Follow the rest of the steps
- Note for the PAP2T-NA, you need to set 2 Nat fields in the adapter, and they are location in the ADVANCED settings. It took me a while to find them.

3) Configure your voip account
a) Go to Main Menu, Account Settings. Here you can adjust your account info. A couple things I changed:
- select the USA48/Canada Routing (premium or value, I selected value)
- set your CallerID Number
b) Scroll down to Default Routing Options, and select the server you will be using (I'm using Toronto)

4) Add some money to your account. You have to add money before you can order a DID or make any calls.
a) In the top menus, go to Finances, Add Funds
b) Follow the instructions to add $25 (the minimum amount)

5) Order a DID
a) In the top menus, go to DID Numbers, Order DID(s)
b) Select Canada (or whatever country you want a number for)
c) At the top you can select a province (for Canada, anyway), and below you can select the region (town) for which you would like to see a list of numbers
d) Pick a number that looks easy to remember or feels lucky. The number will be linked you your Voip account using the server you set is step 3b above.
e) Scroll down and select a payment plan (Per minute or flat rate)
f) Scroll a bit further down and confirm your server
g) Scroll down even further and select Click Here to Confirm You Order to buy the number

6) Connect a phone to your ATA and try and make a call. My outgoing calls worked right away. The only issue I had was getting my DID for incoming calls working, but it was due to me not being able to find the Nat settings in my ATA that I mentioned above.

Total time to set this all up was about 45 minutes, including calling a few people to test out the quality. I'm still playing around with settings, and this weekend will be a larger test than last night, so I'm sure I'll come across some tweaks.

Next step will be to do a bit of testing and then port my existing land line number over once I cancel from Bell.
 

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I used these instructions when I set up.

If you are new to VoIP.ms, please feel free to add any suggestions.
 

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I think a how-to for voip.ms would be a great idea Hugh. Voip can save most of us a fair amount of money and voip.ms probably has the best rates for Canadians. The setup however can be quite intimidating for most people, as you pointed out.

Voip.ms runs a discount rate service. They charge wholesale rates for people who bring their own hardware. They don't sell or provide plug and play equipment. They don't specialize in hand holding.

They do provide configuration files to copy and paste if you are using one of the common adapters. If you aren't using one of the common adapters, you have to work your way own way through the configuration, and this can add considerable complexity, especially with a multi-line sip phone.

To illustrate, I purchased an Aastra 6757i ct. It was recommended the best VoIP phone, period . It is a dream come true if you're a techno geek. More features than you can imagine. It even comes with a 1200 page administrators manual. :confused:

I think a users guide or a F.A.Q.s, sorted by make of adapter or phone, would be a great help. There are a only few commonly used adapters and a half dozen makes of popular sip phones.

Is that the sound of volunteers rushing the forum? :eek:
 

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If you'd like to add some helpful hints then please do.
 

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I can only offer some general tips at this point Hugh.

Regarding the voip.ms portal: There is a lot of information there, most of which you don't see until you have an account and log in. Creating an account is free. Take some time, hover your mouse pointer over the drop down menus and click on the question marks wherever they are found. For most users with a basic phone setup the instructions provided by DdDave should get you up and running.

If uninterrupted phone service is important to you, remember to setup a call forwarding number to a land line or a cell phone. You will have more interruptions to your phone service on voip than you had on the phone service you are used to. Servers go down occasionally, so does the internet.

When choosing a device or ip phone, choose one of the commonly used models. If you run into any problems it will be much easier to fix.

Regarding DIDs (phone numbers). If you have a number you wish to keep, start the porting process to voip.ms before you advise your current carrier; you won't risk losing it that way.

Once I get all the bugs sorted out I will post screen shots of my configuration pages along with explanations where the default settings were changed.
 

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I started playing with VoIP last week, here's what I would recommend people do.

1) Create your voip.ms account
Be aware that the password you enter at this time will be your Customer Portal and your Main SIP/IAX Passwords. I recommend you change both to a complicated password (the webpage tells you if your password is complex or not) if you went with a simple one. For this go to Main Menu > Account Settings

2) Add some money by going to Finances > Add Funds

Install a Softphone on your computer. I installed X-Lite 4 for no particular reason other than this is what I found first.

3) Configure X-Lite 4 with voip.ms
Open Softphone > Account Settings and enter the following (my comments in brackets):
Account name: VOIPms (any name you want, I like naming things after the server I will connect to).
User Details:
User ID: <your SIP Username> (on voip.ms, go to Main Menu > Account Information)
Domain: seattle.voip.ms (select the server that's closest to you from the list of voip.ms Servers - I'm in Vancouver so Seattle it is)
Password: <SIP Password> (the one I asked you to change)
Display name: <Your name>

If all goes well, once you click OK the softphone should 'register' with voip.ms. If the green Call button letters are white, you're good.

Ensure you're using your headset on the client. Go to Softphone > Preferences > Devices and ensure that you are using the correct speaker, microphone amd HID device.

4) Switch to your web broswer and go to your Account Settings (Main Menu > Account Settings), look for the row that says: CallerID Number - hover on top of the (?) and see what it says - enter your current phone number (no spaces or dots or hyphens). Once you enter your phone number, press apply (just to the right of the box).

You should now be able to make phone calls from your softphone. And people who get them will see the number you entered in the CallerID field.

If you go to your Account Settings (voip.ms), you can play with the different options for your main SIP account - of interest here would be the routing (value or premium) options.

If you wish to be able to receive calls then you need to get a DID, but that’s another story.
 

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Great thread and posts. I am thinking about gradually easing my telecom needs to the voip.ms . Using the posts here, I will go with the softphone first. I would like to see more hardware, software guides/instructions. I will also contribute my own when I go . My goal is to move away from the expensive landline and to build an uninterrupted voip system (possibly with pbx as in mango's blog) for my home.

I currently use the regular POST landline, highspeed dsl internet, and basic cable modem. All members in the home have cell phones. I'm thinking about a digital home with all my tv/media programs, internet, and phone united in a box. I would also like to manage this system from my smart phone also.

Can't wait to see what others are doing!
 

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Calling Internationally

Other than enabling your account to allow international calls you will have to submit an ID (through you voip.ms account) to be cleared if you want to call to any one on the list "questionable" countries. High incidences of fraud linked to those countries probably have something to do with it.

These countries are

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, Cuba, Dem. Rep. of Congo, Egypt, Equatorial Guinea, Eritrea, Estonia, Falkland Islands, Faroe Islands, Fiji, French Guiana, French Polynesia, Gabon, Gambia, Georgia, Ghana, Gibraltar, Guinea, Guinea Bissau, Guyana, Honduras, Hungary, Iran, Iraq, Jordan, Kenya, 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, Tokelau, Tonga, Tunisia, Turkey, Turkmenistan, Tuvalu, Uganda, Ukraine, Uzbekistan, Vanuatu, Wallis & Futuna, Yemen (Arab Republic), Zambia and Zimbabwe.
 

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I am looking to cancel my home phone service and would like to port my current home number to a call forwarding service that would simply forward the calls to a cell phone. Can voip.ms do this? It seems like it, but I've read the above posts, and looked at the voip.ms site, and I can't seem to figure out if I can do it or not. Any thoughts on that? Or should I bee looking at one of the (literally hundreds) of providers that do that? I'd like to go with a Canadian company if possible.

Thanks
Tom
 

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Just signed up with voip.ms. I'm using Cisco SPA2102 ATA and one issue I had to sort out was the configuration where the I wanted to disable the built-in router functionality and put it behind the existing router. Following steps got it working:

  1. Connect the WAN port of SPA2102 to another computer (I used my laptop) and let that acquire an IP address from SPA2102 in router mode.
  2. Go in to the web access page of the ATA (default is 192.168.0.3) and change following settings. It is important to do all these things in one go and hit 'accept' once all the changes are done.
    • Disable DHCP server (critical). This turns off the DHCP server built on to ATA
    • Setup a static IP in the lan. It was easier as I didn't want to worry about the ATA getting an IP from the router via DHCP. I'm sure you could setup the ATAas a DHCP client and get the IP and DNS from router.
    • Manually setup gateway and DNS servers since we disabled DHCP client
    • Set the ATA to 'bridge/switch' mode (critical)
  3. Once all the above changes are done, click on the 'accept all' button. This will reset the ATA and you will not be able to access the setup page until it is connected to the router in the next step
  4. Remove the ATA from the computer and connect the WAN port of the ATA to one of the LAN ports of the router
  5. Test the setup by pointing a browser to the static IP of the ATA. If you see the configuration page, you are all set.
  6. If things don't work and you have lost all access to the ATA, do a hard reset: connect a phone to line-1 and dial '****'. You will hear a voice prompt. Enter '73738' (RESET) and it will change everything back to original settings. Now you can go back to step 1 and try again :)
Hope this helps

Sam
 

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I apologize in advanced if live links are not allowed. But, this site has setup guidelines for all the different devices for all the different services.



http://voipfan.net
 

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Thanks for contributing guys, I'll definitely be using it when I switch from the underlords at Rogers.

Which ATA would you recommend? I'm looking for something very user-friendly (I'm implying I'm an idiot)
 

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Thanks for contributing guys, I'll definitely be using it when I switch from the underlords at Rogers.

Which ATA would you recommend? I'm looking for something very user-friendly (I'm implying I'm an idiot)
I just started using voip.ms a little over a week ago (and initiated the port of my phone # from Rogers this past Saturday). For the ATA I bought a Cisco Linksys PAP2T-NA. It seems to be "the" popular choice out there, so there's a lot of help. voip.ms have a wiki page for the configuration (they do for other devices as well), and I also followed the instructions from Mango's page. Worked like a charm. The only thing I didn't do right away is change the settings to avoid the occasional "beep" (like a digit being pressed), and on my first long call I heard it once, so after that I went into the configuration pages and changed the settings. Didn't hear it after that.

I find the web interface for the PAP2T is a bit antiquated compared to some more recent devices (not talking about VoIP devices, more like routers and other such gizmos), but it's OK - it has the advantage of not requiring you to go through 20 tabs to configure stuff (unlike my router which is a bit of a maze, even though its interface is more "modern"). If you do end up getting that, first thing to do is check the firmware version. Mine was 3.xxxx (whatever) and the most recent is 5.xxxx. With taxes it cost me about $61 at Canada Computers (you can get it online for $6-7 less, but getting it at the store meant getting it "now").
 

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Just signed up with voip.ms. I'm using Cisco SPA2102 ATA and one issue I had to sort out was the configuration where the I wanted to disable the built-in router functionality and put it behind the existing router. Following steps got it working:

  1. Connect the WAN port of SPA2102 to another computer (I used my laptop) and let that acquire an IP address from SPA2102 in router mode.
  2. Go in to the web access page of the ATA (default is 192.168.0.3) and change following settings. It is important to do all these things in one go and hit 'accept' once all the changes are done.
    • Disable DHCP server (critical). This turns off the DHCP server built on to ATA
    • Setup a static IP in the lan. It was easier as I didn't want to worry about the ATA getting an IP from the router via DHCP. I'm sure you could setup the ATAas a DHCP client and get the IP and DNS from router.
    • Manually setup gateway and DNS servers since we disabled DHCP client
    • Set the ATA to 'bridge/switch' mode (critical)
  3. Once all the above changes are done, click on the 'accept all' button. This will reset the ATA and you will not be able to access the setup page until it is connected to the router in the next step
  4. Remove the ATA from the computer and connect the WAN port of the ATA to one of the LAN ports of the router
  5. Test the setup by pointing a browser to the static IP of the ATA. If you see the configuration page, you are all set.
  6. If things don't work and you have lost all access to the ATA, do a hard reset: connect a phone to line-1 and dial '****'. You will hear a voice prompt. Enter '73738' (RESET) and it will change everything back to original settings. Now you can go back to step 1 and try again :)
Hope this helps

Sam
I have an all in one modem+router. The problem I am getting is when I have all the wires connected, and I type in the IP address, it takes me to the set-up for my all-in-one modem/router, not the SPA2102. I've checked the IP address through the phone and it's all correct, so I don't know what to do to access the SPA2102 settings!
 

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SPA2102 set up on VOIP.ms

Here are the variables I used to set up my SPA 2102 ATA.

Set the DHCP on your home router to give your Linksys ATA the same IP address all the time (if you can).
Forward ports 5060 and 5061 from the internet to the IP address of the ATA.******** (5061 is for the second line if you every use it)
login to the ATA
Click the admin tab
Clink advanced tab
Click the voice tab
Under "System", set a admin password.


Click the SIP tab (you need to have clicked "advanced" to see this)
Change RTP packet size: 0.010 (This sends a packet every 10ms second vs the default of 30ms)
NAT Keep Alive Intvl: 15*

Click Regional Tab
Ring and Call Waiting Tone Spec, Ring Waveform : Sinusoid** (This is the north american standard)
Ring Voltage : 90** (lower voltage causes old mechanical ringer phones to sound like a dead cat)
Set time Zone: (GMT-05:00)
Daylight saving time rule: start=3/8/7/2:00;end=11/1/7/2:00;save=1

Under Line 1
NAT Mapping Enabled:NO
NAT Keep Alive Enabled: NO
NAT Keep Alive Msg:$PROXY
NAT Keep Alive Dest:$PROXY
Network Jitter Level: low
SIP port: 5060
Proxy: montreal2.voip.ms*** (This has better uptime the toronto2.voip.ms)* (needs to match your account setting on website)
Display Name:* <Enter the name for outgoing call name display>
User ID: <Account from voip.ms>
Password: <Voip.ms provided>
Register Expires:60************ (This means that the phones registers and updates your IP address to the VOIP server every 60 secs)


Preferred Coded: G711u
Dial Plan:(<: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 addition, he gave me the following information:

* For a voip connection, the amount of bandwidth is not the limiting factor but the jitter (change in latency). You can test out the following website, when you are running your applications to see if there is any problems on your line.* http://myspeed.visualware.com/indexvoip.php**

The standard codec for voip is G7.11 (64Kbps).* Its actually about 90Kbps when you factor in everything.
 

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"Set the DHCP on your home router to give your Linksys ATA the same IP address all the time (if you can)"

Can someone help with this. Still having some issues... Many thanks
 

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It varies with router. Look for the DHCP setup page and assign a static lease. The other way is to assign a static IP on the ATA itself. The first three numbers must be the same as the router's address. The last number must be unique and outside the router's DHCP range.
 
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