VoIP.ms: Basic How-to Setup - Page 2 - Canadian TV, Computing and Home Theatre Forums
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post #16 of 48 (permalink) Old 2012-05-13, 11:25 PM
Join Date: May 2012
Posts: 5
Hi All,

just created a video how I setup my phone numbers.
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post #17 of 48 (permalink) Old 2012-07-18, 02:11 PM
Join Date: Dec 2006
Location: Just outside the centre of Canada
Posts: 124
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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post #18 of 48 (permalink) Old 2012-09-28, 10:25 PM
Join Date: Oct 2005
Location: London, ON
Posts: 160
This helped me a ton and fixed my problems. Thanks
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post #19 of 48 (permalink) Old 2012-09-30, 08:08 PM
Join Date: Oct 2005
Location: London, ON
Posts: 160
"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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post #20 of 48 (permalink) Old 2012-10-01, 12:33 PM
Join Date: Feb 2009
Location: The Dandelion City
Posts: 7,131
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.

At 20 I had a good mind. At 40 I had money. At 60 I've lost my mind and my money. Oh, to be 20 again. --Scary
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post #21 of 48 (permalink) Old 2013-03-28, 08:28 AM
Join Date: Mar 2013
Location: Waterdown, ON
Posts: 46
The info in this thread is great.

The first thing is always to create your voip.ms account, transfer some money into it via paypal and then to order your DID's or initiate a DID transfer (normally $10 but they currently have a free promo on it). As it takes a few days.

For the hardware config, don't forget that voip.ms also has a number of configuration guides. I used their guide on the Obi100 and was up and running in minutes.


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post #22 of 48 (permalink) Old 2014-05-15, 03:20 PM
Join Date: Mar 2007
Location: Victoria, BC
Posts: 107
Hmm... not sure why but after setting everything up with my Cisco SPA112 and voip.ms when I call out my number is prefixed always with 011

Anyway to get rid of it? I only entered in my 10 digital number when doing the config.

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post #23 of 48 (permalink) Old 2014-05-15, 07:06 PM
Join Date: Mar 2006
Location: Calgary - Shaw phone/internet, OTA attic / Pigeon Lake - CCI Wireless, VoIP.ms, OTA, FTA, LTSS
Posts: 810
If you only have your 10 digit phone number in the CallerID Number field under Account Settings -> General then you will likely need to look through all of the possible SPA112 settings that could be adding a prefix.

I wonder what you would see if you leave the CallerID field blank in your voip.ms settings.
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post #24 of 48 (permalink) Old 2014-05-15, 07:09 PM
Join Date: Mar 2007
Location: Victoria, BC
Posts: 107
could it be my dialing plan?

I am using this (suggested on this thread)

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post #25 of 48 (permalink) Old 2014-08-07, 02:24 PM
Join Date: Jan 2010
Posts: 629
VOIP.ms - Obi110 setup

I've got this set up running fine for me at home. SP1 is voip.ms and SP2 is still google voice, even though it's supposed to be defunct. I set this up for my dad at his place with his own Obi110. Same exact setup. His number transferred over from Rogers today. I can call him and the call goes through. But when he calls out his call doesn't go through. The error response code he's getting is 488. voip.ms says this code means: 488 Not Acceptable Here. I copied my settings exactly, so I'm not sure what I missed. Oh and he can make calls using SP2 by dialing **2 before the number. Any suggestions?
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post #26 of 48 (permalink) Old 2014-08-07, 02:41 PM
Join Date: Nov 2008
Location: Alberta
Posts: 867
You've learned a lesson in why you should test your configuration to make sure it works, before you port your number.

It would be much easier for VoIP.ms to troubleshoot this, as they can look at the SIP Debug and see exactly what is wrong. I am just guessing, but here are some things to try. I'm assuming you use the main account and not a sub-account - if I'm wrong, adjust accordingly. On the VoIP.ms portal, navigate to Account Settings.

On the Account Restrictions tab, make sure all settings are appropriate, and that beside "Allow Calls to Countries" you see "All Countries Allowed".

On the Advanced tab, be sure NAT is set to yes, DTMF Mode is set to AUTO, and all the codecs are allowed. This error can be caused by codec mismatch, so I think this is a likely solution.

Is there any possibility your dad's router has a SIP ALG that is causing the problem? If you have a setting for SIP ALG on his router, disable it.

If none of that works, and you've compared your OBi configuration to his and are confident it matches, contact VoIP.ms again and ask for further assistance.

Let us know how things go.

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post #27 of 48 (permalink) Old 2014-08-07, 03:02 PM
Join Date: Jan 2010
Posts: 629
Thanks Mango. Yes it's the main account. Everything up to the codecs section is how I have it setup. The only codec not checked is gsm, which is how i have it on mine. I'll check that one as well and see if that makes a difference.
As for the router tip, I'll check that next time I'm there, which will probably have to be soon if this doesn't work.
I'll send voip.ms a message as well.
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post #28 of 48 (permalink) Old 2014-08-07, 05:22 PM
Join Date: Jan 2010
Posts: 629
Fixed. I set my dad's line up on my Obi as SP2 temporarily and it worked fine. They came home and tried calling me and calling out from their line was working. I didn't do anything. So maybe the transfer wasn't fully complete as of yet. Who knows. It works fine now.

One more Q. When I try to call the voip.ms echo test from my phone, I get an error: The number is rejected by the service provider. Reason is 484. The Obi echo test **9 222 222 222 works fine though.
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post #29 of 48 (permalink) Old 2014-08-25, 06:38 AM
Join Date: Aug 2014
Posts: 1
Thanks for share this information because recently I'masds use VoIP phone system so face this type of problem and not remember how to set up VoIP system. So now I'm easily install VoIP account.
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post #30 of 48 (permalink) Old 2014-09-29, 10:58 AM
Join Date: Apr 2003
Location: Oakville
Posts: 31
I know this may be a simple question, but I'm having trouble getting a definitive answer.
If i use VOIP MS and get, for example, a Cisco Linksys PAP2T, can i then just plug my existing home phones into the PAP2T?
I have a regular Panasonic Dect 6.0. It has a base that plugs into a pots wall jack, with 5 other wireless phones throughout the house.
Will the functionality be the same as it is with pots?
IE: If someone dieals the VOIP MS number it will ring on all devices?
Any of the 6 phones can dial out?

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