tag:blogger.com,1999:blog-34026308507309219752024-02-06T20:33:40.913-08:00Freephoneline.ca ATA SetupTelephone support - Resolved Solution to ATA SetupUnknownnoreply@blogger.comBlogger19125tag:blogger.com,1999:blog-3402630850730921975.post-12795150490483192282023-04-27T12:05:00.008-07:002023-04-27T12:07:38.210-07:00Server Outage Progress Fongo / FreephonelineServer Outage Progress: April 25, 26, 27 2023<br /><br />We have approx. 30% of Fongo Home Phone users back online. <br /><br />More Home Phone users will come back online as the day goes on. No action is required.<div><br /></div><span><a name='more'></a></span><div><br /></div><div><div>Outage Update: 7:30PM ET</div><div><br /></div><div>We are making progress on restoring all services, but currently have no ETA.</div><div><br /></div><div>What is currently working:</div><div>- Fongo Internet</div><div>- Fongo Wireless connectivity plans are operational but user portal is inaccessible</div><div><br /></div><div>Keep up to date:</div></div><div><br /></div><div>www.twitter.com/VoiPToronto</div><div><br /></div><div><br /></div><div><br /></div><div><br /></div>Unknownnoreply@blogger.com0Toronto, ON, Canada43.653226 -79.383184315.342992163821151 -114.5394343 71.963459836178842 -44.226934299999996tag:blogger.com,1999:blog-3402630850730921975.post-7346809717342842182018-01-04T00:11:00.001-08:002021-02-09T13:06:36.950-08:00Free VoIP Phone Service in CanadaVOIP is Voice over IP or Voice over Internet Protocol - using IP networks like the Internet to route phone calls vs using a regular land-line also known as POTS (Plain Old Telephone Service).<br /><br /> This setup can be totally free by using a "soft phone" on your PC or Mac and with a small initial investment it is possible to use a regular phone and still have a free VOIP phone line in Canada - or a Canadian phone number somewhere else in the world.<br /><br /><br /> Get a real phone number <br /> No monthly phone bill <br /> Call many cities in Canada free <br /><br /> This Instruction will cover getting a phone number, setting it up as a "soft phone" on your PC or Mac<br /><br /> But it gets better - with the correct VoIP/ATA device (an "Obihai") and configuration we can bridge VoIP calls over to POTS lines and vice-versa. What is the use of this? I have a VoIP device in Vancouver, Canada and a second device at a family members house in Auckland, New Zealand - this device in Auckland has a Vancouver number on it and is also connected to the POTS system and therefore has an Auckland phone number associated with it. From my device in Vancouver I can call the device in Auckland over the Internet, then call out on the Auckland land-line - so I can get Auckland dial-tone from Vancouver, for free! In Auckland they can pick up the phone and dial almost anywhere in Canada for free. There is also an app for both iPhone's and Android phones that allow you to call an Obihai device and use it to call out - for example my brother in New Zealand installs it on his iPhone and I add him to my trusted network. When he connects his iPhone to the Internet via WiFi he can call my Obihai device and use it to call me on my cell phone<br />
<a name='more'></a><br /><br />Note: I am NOT employed or have any affiliation with the companies or products mentioned in this Instructable, I do not get any financial or other compensation from them either - I am just a happy customer who can call family on the other side of the world for free and have them call me for free.<br /><br />
<br />
Step 1: Pro's and Con's<br /> <br /><br />As the saying goes there is no such thing as a free lunch. There are pro's and con's associated with using VoIP<br /><br />Pro's:<br />Real phone number from 1 of 4 provinces<br />Can use a "soft phone" or regular phone<br />Call most Canadian cities for free<br />Bridge between VoIP and POTS<br />Take your phone number with you when you travel<br /><br /><br />Con's:<br />No 911 service or service not as reliable as POTS<br />Small initial cost outlay for VoIP device and "config file" now called a "VoIP Unlock Key"<br />It can be technical to setup<br />Needs Internet<br />If Internet drops so does the phone.<br /><br />
<br />
Step 2: Sign Up for a Phone Number.<br /><br />Note: a "soft phone" is a software application you install onto a PC to give you phone calls via the PC microphone and sound card.<br /><br />I use a service called <a href="http://www.freephoneline.ca/">FreePhoneLine.ca</a><br />(Again I do not work for them or get any compensation financial or otherwise from them).<br /><br />To setup an account with them you will need to provide your name, address (including postal code, street address, province etc), DOB, gender, email address, a valid phone number like your cellular number or existing land line number and agree to their terms and conditions (a lot of info on 911, 1-900, 411 etc etc). If you aren't happy giving this information out then don't bother registering! If you are OK with this then lets proceed.<br /><br />Register on the site.<br /> Initially you will need to provide an email address and a valid phone number (like your cellular number).<br />You will get a confirmation email needed to activate your account.<br />Once activated login to the site and click on the "register" button - here you will have to provide the info listed above.<br />Continue - you will need to agree to their terms and conditions.<br />Once you agree to the terms and conditions the next step they require is to confirm your existing phone number (this is to probably confirm you are Canadian but I am not 100% sure). They will call the number you provided and give you a 3 digit confirmation code you need to enter into a box and confirm.<br />Almost there.....<br />Now you get to pick the area code and number. The Provinces you can get a number from are: BC, Alberta, Ontario and Quebec. Once you pick a province you get a choice of cities. You can pick a number from any of the listed provinces/cities. If for example you are in Vancouver and have family in Toronto you can pick a Toronto number and family in Toronto will be able to call you for free. If you don't like the first number picked for you you can get a couple more tries. <br /><br />Once you have got a phone number you can download either a Windows or Mac version of the software.<br />Install the software, launch it and login with the same login/password you used to register on the website.<br /><br />Congratulations you have a free working phone number - no credit card needed.<br /><br />To make long distance calls to places out of their free calling zone you can buy credits from your account on the freephoneline.ca website.<br /><br />Step 3: Using an ATA/VoIP Device<br /> <br />I used this setup for a couple of months but wanted to use a regular cordless phone so I didn't have to sit at my PC to make or receive calls. To do this I purchased an ATA/VoIP device and also a "config file" from freephoneline.ca - the "config file" was a one-time $50.00 fee. As of Nov 2017 this is called a "VoIP Unlock Key" and is $79.95The VoIP Unlock Key is a login & password for their SIP system along with information on what servers etc to use.<br /><br />Unknownnoreply@blogger.com0Toronto, ON, Canada43.653226 -79.383184315.342992163821151 -114.5394343 71.963459836178842 -44.226934299999996tag:blogger.com,1999:blog-3402630850730921975.post-49563390027003290592014-05-12T14:29:00.003-07:002021-02-09T13:06:30.354-08:00freephoneline.ca - How does it work? freephoneline.ca uses Voice-over-IP (VoIP) technology. You can sign up with their service for free online. Their website provides you a real Canadian phone number (your choice from random phone numbers generated by their system) upon signing up. You can make phone calls right away using a softphone (a software program for making phone calls from your computer) or Android and iPhone app. The phone calls are free as long as the cities (you are calling) are on their free calling city list. If you wish to use your regular telephone handsets with their service, you need to buy your own analog telephone adapter (ATA) and purchase a configuration file (VoIP Unlock Key) for your ATA. Once your ATA is setup with the configuration file and connected to Ethernet modem, you can hook up your regular phone to the ATA and start making and receiving phone calls. Please note, freephoneline.ca does not offer any support for a 3rd party ATA. <br />
<br />
Standard telephone features such as Caller ID, Voicemail and Follow Me Service (essentially a Call Forwarding service; however, you can forward up to 3 different phone numbers) are included. Call Waiting is not advertising but it is reported to be working (you have to set it up in your ATA). The Enhanced Voicemail feature allows your voicemails to be sent to your email account. In addition, porting of your existing phone number is possible (although not guaranteed and available in all cities) for a fee which makes their service attractive for those who have had their number for a long time and would like to keep using it.<br />
<br />
Depending on your current home phone bill, you can recover the initial setup cost within 3 - 6 months, (over 10 months if you are using a VoIP phone company such as iTalkBB which already offers very low monthly rate). It might sound too good to be true, but after the initial setup is paid off, you will never pay for your phone bills again! The only concern is that the sustainability of the company is unknown because freephoneline.ca does not collect any monthly fees for providing their service. As such, their revenue is probably dependant on long-distance charges and selling "VoIP Unlock Keys" to new customers.<br />
<br />
Please remember VoIP phone service is dependant on your Internet connection so you will lose your phone service if there is a network or power outage. Also, you must keep your address up-to-date in case of emergencies because unlike tra<br />
<a name='more'></a>ditional phone line, there is no easy way to confirm your location other than your last registered address with the company. Lastly, given the technology is not as reliable as traditional phone, it is probably not a good idea to use it as the primary phone line; especially if you do not have a cell phone as backup.<br />
<br />
<br />
Requirement:<br />
<br />
Computer PC or Mac (if using softphone only)<br />
High Speed Internet<br />
Headset with microphone or speakers and microphone<br />
Analog Telephone Adaptor a.k.a. ATA (required if you wish to use your regular telephone)<br />
Configuration file (VoIP Unlock Key) - if you already own an ATA (required if you wish to use your regular telephone)<br />
Ethernet router/DSL or cable modem<br />
Cordless phone with multiple handsets (optional)<br />
<br />
<br />
<br />
You might like it because: <br />
<br />
No monthly bills<br />
No contract<br />
Free unlimited calling to select cities in Canada<br />
Easy to sign-up online<br />
You can pick a real Canadian phone number (from random phone numbers until you are satisfied)<br />
Making and receiving phone calls immediately on your computer as soon as the freephoneline software is installed<br />
Porting of your current phone number is available in most cities (for a fee)<br />
Caller ID, Voicemail, Follow Me Service (same as Call Forwarding), Call Waiting.<br />
Voicemails could be sent to your email account<br />
Auto-renew for pre-paid World Credits for making long distance phone calls<br />
<br />
<br />
<br />
You might not fall in love with it because:<br />
<br />
No live-person customer support. All inquiries including billing are done via emails<br />
Using your computer (softphone) to make and receive phone calls may not be as convenient as regular home phone service<br />
You need to purchase a a configuration file (VoIP Unlock Key) from freephoneline.ca and a VoIP phone adapter (your own or from freephoneline.ca) in order for their service to function like a regular telephone<br />
There is no support for 3rd party ATA so you will need to set up the configuration file on your time. The set up time then depends on your computer skills. There is s no live-person support<br />
Porting of existing phone numbers are not available in all cities (also, a fee for porting your phone number is applicable)<br />
Number porting process is not as simple as some companies (you need to print out a form and email or fax it with a copy of your current telephone bill to freephoneline.ca). Some competitors just need you to fill out an online form.<br />
Receiving fax is possible but not guaranteed<br />
The need to purchase pre-paid World Credits for making long distance phone calls is not as good as paying for only the minutes used<br />
Only one phone jack is available from the VoIP phone adapter so you would need a cordless phone with multiple handsets (if you need phones in different rooms)<br />
Uncertainty about the company's future due to no monthly fees structure<br />
Usability is susceptible to power/network outage<br />
911 service is dependant on point above<br />
Not as reliable as traditional phone service<br />
Not recommended for use with home alarm monitoring system<br />
Voice quality and reliability may vary due to factors such as Internet speed, ATA, Ethernet router, phone frequencies/location, software compression, etc<br />
Security concerns, given the voice data travels through the Internet, the data is vulnerable to theft, viruses, spam, etc.<br />
<br />
<br />
Where to get it?<br />
<br />
Your own ATA (optional) - computer or electronics stores<br />
Configuration file (VoIP Unlock Key) (optional) - freephoneline.ca<br />
Computer - electronic stores, computer stores<br />
High Speed Internet - your local Internet Service Provider<br />
Headset with microphone or computer speakers and microphone - computer stores<br />
DSL or cable modem - usually supplied by your local Internet Service Provider<br />
Cordless phone with multiple handsets (optional) - electronic stores<br />
<br />
<br />
How to enhance this setup:<br />
<br />
freephoneline's parent company is Fongo Inc. If you feel there is too much hassle to purchase the ATA, the VoIP Unlock Key and need to configure the ATA on your own, Fongo does offer a home phone service for a small monthly fee (however, still no live-person support): http://www.freephoneline.ca/optionalServices#<br />
Always use your cell phone as a back up in case of a power/network outage<br />
Use a cheap Uninterrupted Power Supply (UPS) to power your modem and ATA in case of a power outage<br />
<br />
<br />
<br />
Summary:<br />
<br />
freephoneline.ca is a great option for eliminating your home phone bill as long as you have a High Speed Internet connection. You can sign up easily online for free and pick a working local Canadian phone number for making phone calls right away. You can make unlimited calls to most Canadian cities for free. You can also port your existing number to freephoneline for a fee. The only catch is you can only make and receive phone calls using your computer. If you wish to use a regular telephone with freephoneline.ca's service, you will need to buy/supply your own VoIP Internet Phone Adapter (ATA) (see "Product Info/Reviews section on the right for options) and the VoIP Unlock Key from freephoneline. The biggest hassle may be there is no live-person support in general and no support at all for any ATA and VoIP Unlock Key issues. Therefore, you are on your own to deal with configuring the ATA and any technical issues. If the lack of support doesn't bother you, then once the ATA is set up, you will never pay for another phone bill!Unknownnoreply@blogger.com0Toronto, ON, Canada43.653226 -79.383184315.342992163821151 -114.5394343 71.963459836178842 -44.226934299999996tag:blogger.com,1999:blog-3402630850730921975.post-27335920896337059322014-03-15T14:39:00.002-07:002021-02-09T13:06:14.388-08:00VOIP Landline: How to get a free home phone line for lifeAccording to the latest study by the US National Health Information Survey released in June 2012 (<a href="http://www.marketingcharts.com/direct/landline-phone-penetration-dwindles-as-cell-only-households-grow-22577/" rel="nofollow" target="_blank">link</a>),
from 2008 to 2012 the percentage of American households that only use
wireless phones went from 17.5% to 34%. All the while, the percentage of
households with a landline phone (with or without an accompanying
cellphone) decreased from 79.1% to 63.6%. The trend is clear. Consumers
are cutting their landlines in favor of cell phones at an ever
increasing rate. There are many reasons for this behavioral shift.
Smartphones are now ubiquitous. The majority of adolescents and adults
have or would like to buy a smartphone. You can carry it around with you
all day, others can reach you with one number, you can browse the web,
write/read email, play games, listen to music and some users even have
unlimited voice and data plans. It’s very convenient.<br />
<span id="more-1"></span><br />
<br />
We
believe that the primary reason why users are giving up landlines it
that they cost too much. Why pay for 2 phones when you have one with you
at all times that does so much more? That’s a very good question, but
what if we told you can you can setup a home phone for practically
nothing? Would you be interested then? No monthly fees. You can use
regular phones and it’s free for life. Now that we got your attention, a
lot of common responses that we receive are “this sounds like a scam”,
“that’s impossible”, “must be terrible service”. Well, we have been
using this service for over a year without any issues. We have even
helped setup such a line for a couple of friends, also without any
problems. We’ve read reports that others have been using such a service
for over 5 years and they’re still happy with it.<br />
<br />
Did we mention that you also get unlimited long distance calling in Canada? Oh ya, that’s free too.<br />
What
I’m describing is a VOIP (Voice Over Internet Protocol) telephone line.
Before we go any further, you should ensure that you have 1) a high
speed internet connection, 2) a computer/laptop, 3) a regular landline
phone (we recommend a DECT 6.0 cordless phone with a base station and
3-5 handsets), 4) a wireless router and 5) an analog telephone adapter
(commonly referred to as an ATA device). Below is a diagram of how
everything would be connected.<br />
<div class="separator" style="clear: both; text-align: center;">
<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEintAUKioFhvsxG9RjbuwyAFeBhJbXTAttLw7APRrJMtYyr2D1XrgjHR67mdYfRQmxZw3eTPNImmH28WJCLP9rGhkkcRPlB547ULwvLdux5WZOdCJJsTK_uBVCN76IpxMLKQpxcOlbPNUM/s1600/voip_Diagram.png" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEintAUKioFhvsxG9RjbuwyAFeBhJbXTAttLw7APRrJMtYyr2D1XrgjHR67mdYfRQmxZw3eTPNImmH28WJCLP9rGhkkcRPlB547ULwvLdux5WZOdCJJsTK_uBVCN76IpxMLKQpxcOlbPNUM/s1600/voip_Diagram.png" height="640" width="452" /></a></div>
<br />
<br />
<a name='more'></a><br />
Follow
these steps for a quick guide on how to setup a new VOIP line. You can
skip the rest of the article if you’re not interested in how VOIP works
or why we’re making these recommendations.<br />
1) Purchase a Linksys PAP2T-NA from eBay for $23-27<br />
2) Once you receive it, register an account at freephoneline.ca (<a href="https://www.freephoneline.ca/accountRegistrationStepOne" rel="nofollow" target="_blank">here</a>)<br />
3) Check your email and click on the link to activate your account<br />
4) You will be back on the freephoneline website, click “Register”<br />
5) Enter in your contact details for 911 emergency purposes (leave a space for the postal code), click “Continue”<br />
6) Review your information, if satisfactory, check the box and click “Accept”<br />
7) Click “Call me now” and it will call your contact number and a
machine will provide you with a 3 digit confirmation code. It may take
up to 15 minutes to receive a call. Please enter this code into the text
box and click “Submit”. If you do not receive a call, please contact
freephonline support desk and they will activate you manually.<br />
8)
You need to purchase an “VOIP Unlock Key” from freephoneline. The VoIP
Unlock Key allows you to access their network to make and receive calls
over the internet. You can choose to keep your current phone number or
you can pick a new home phone number. Login to freephoneline if you’re
not already logged in and purchase the VOIP Unlock Key for $50. Please
choose your new telephone number. If you would like to port your number
instead, please call freephoneline or send them the Letter of
Authorization as found <a href="http://support.freephoneline.ca/servlet/fileField?id=0BEa0000000TPLD" rel="nofollow" target="_blank">here</a>. Please follow their instructions.<br />
9) After purchasing the VOIP Unlock Key, you should receive an email
with your account information. Keep this information for step 12.<br />
For example:<br />
Username: <a href="tel:16473675814">4225927</a><br />
Password: f29l3gs<br />
Realm: voip.freephoneline.ca<br />
10) Connect your ATA to a power source and connect it to your Wireless Router with an ethernet cable (the cable looks like <a href="http://goo.gl/dYnxd" rel="nofollow" target="_blank">this</a>)<br />
11) Connect the base station of your cordless phone to your ATA using the supplied telephone cable (the cable looks like <a href="http://goo.gl/nd08t" rel="nofollow" target="_blank">this</a>)<br />
12) Follow the instructions <a href="http://voipfan.net/ATAs/PAP2_freephoneline.php" rel="nofollow" target="_blank">here</a> to
setup your ATA correctly. However, I would make a small amendment.
Instead of selecting G.729a as the guide suggests, please select G.711u
if you’re in America (I explain my reasoning below).<br />
13) Once the adapter reboots which may take a few minutes. Pick your receiver and you should hear a dial tone.<br />
14) If you ported your number, please call your telephone company to cancel your home phone service.<br />
15) Enjoy free Canadian long distance for life.<br />
If
you’re interested in how to port your existing number, how I came up
with these suggestions or how to setup and access your voice mail,
please read the following.<br />
<br />
<b>What is an ATA Device (Analog telephone adapter)</b><br />
This
is probably the only item that you need to purchase. It’s a small box
that sits between the cordless phone’s base station and to the wireless
router. In the simplest of terms, it acts as an interface between your
regular phone and the internet. The ATA device encodes/decodes voices
before/after it goes through the internet using voice compression
software. Another way of looking at is that your landline phone is an
analog device and the VOIP system is digital. Therefore, a device needs
to translate the digital signals to analog and analog signals to digital
in order for the telephony service to work. That’s what the little box
does.<br />
<br />
one of many examples: <br />
<br />
<div class="separator" style="clear: both; text-align: center;">
<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjauOzJ9E8hBte2SBbeI_NXqJ9gvEQ0zVPnFny2D3ZMgXpE2x23zL2rR0Jum20D1n7vLnTAl7UIVH66w2rhH1P-QtcWt3hN3CQIQPkAZoSia-u17DBZ7S_ree3kV5Q1r_KyNWgzj7qjmfU/s1600/PAP2t.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEjauOzJ9E8hBte2SBbeI_NXqJ9gvEQ0zVPnFny2D3ZMgXpE2x23zL2rR0Jum20D1n7vLnTAl7UIVH66w2rhH1P-QtcWt3hN3CQIQPkAZoSia-u17DBZ7S_ree3kV5Q1r_KyNWgzj7qjmfU/s1600/PAP2t.jpg" height="265" width="320" /></a></div>
<br />
<br />
<b>Which ATA do you recommend?</b><br />
We recommend and personally use the <b><a href="http://www.amazon.com/mn/search/?_encoding=UTF8&camp=1789&creative=390957&field-keywords=Cisco%2FLinksys%20PAP2T-NA&linkCode=ur2&tag=manysites04-20&url=search-alias%3Daps&linkId=DD4EPI5MBREUK6CB" rel="nofollow" target="_blank">Cisco/Linksys PAP2T-NA</a></b> (as pictured above) as it’s reliable and cost effective. The PAP2T-NA can be purchased from eBay for as little as $23-27 w/ free shipping worldwide. The <a href="http://www.amazon.com/mn/search/?_encoding=UTF8&camp=1789&creative=390957&field-keywords=Cisco%2FLinksys%20PAP2T-NA&linkCode=ur2&tag=manysites04-20&url=search-alias%3Daps&linkId=DD4EPI5MBREUK6CB" rel="nofollow" target="_blank">Amazon reviews</a> are
excellent as we expected. We also suggest that you purchase the
unlocked version. Linksys PAP2T models that have a suffix “NA” or “EU”
are unlocked. Without such suffices, it means that the ATA is locked,
therefore you can only use it with a specific VOIP service provider
(it’s the same concept as a locked cell phone). An unlocked ATA means
that you can use it with any compatible VOIP service provider. The
PAP2T-NA has been discontinued since May 2012, but you can still easily
find them online.<br />
The PAP2T-NA’s main feature is that it allows
you to hook up 2 different VOIP telephone lines to it. In other words,
you can connect 2 cordless base stations to the ATA (so, 2 different
phone numbers). That can be useful if you have a home office and would
like to setup an office line and a residential line using one ATA.<br />
<br />
<b>Which other ATA’s did you consider?</b><br />
We have also considered the <a href="http://www.amazon.com/mn/search/?_encoding=UTF8&camp=1789&creative=390957&field-keywords=Cisco%2FLinksys%20PAP2T-NA&linkCode=ur2&tag=manysites04-20&url=search-alias%3Daps&linkId=DD4EPI5MBREUK6CB" rel="nofollow" style="font-weight: bold;" target="_blank">Cisco SPA112</a> (the successor to the PAP2T), the more high end <b><a href="http://www.amazon.com/mn/search/?_encoding=UTF8&camp=1789&creative=390957&field-keywords=Cisco%2FLinksys%20PAP2T-NA&linkCode=ur2&tag=manysites04-20&url=search-alias%3Daps&linkId=DD4EPI5MBREUK6CB" rel="nofollow" target="_blank">Cisco SPA2012</a></b> and <b><a href="http://www.amazon.com/mn/search/?_encoding=UTF8&camp=1789&creative=390957&field-keywords=Cisco%2FLinksys%20PAP2T-NA&linkCode=ur2&tag=manysites04-20&url=search-alias%3Daps&linkId=DD4EPI5MBREUK6CB" rel="nofollow" target="_blank">Cisco SPA3102</a></b>,
but we do not recommend them to most users. These ATA’s have extra
features that most people will not use or benefit from. Therefore, why
pay for features that most users won’t use? These 3 ATA’s include a
router in the ATA, but we recommend that users purchase a dedicated
wireless router. We also find no compelling reason to purchase the
SPA112 instead of PAP2T-NA. It appears that many SPA112 users on Amazon
experience stability issues or have trouble getting caller ID to work
properly. The reviews are mixed at best. However, if you plan on setting
up a fax machine over VOIP, we recommend the SPA112 as many users have
experienced difficulties setting up fax correctly with the
PAP2T. The SPA3012 allows users to hook up a regular landline to the ATA
as well, but we don’t find such an exercise very useful.
The SPA2012 also allow the usage of voice coder G.729 (as explained
below) for 2 VOIP lines, but we recommend the usage of G.711 as the
voice quality is higher, therefore that factor becomes moot. The voice
call quality will about the same for all 3 ATA’s, therefore there is no
reason for most people to spend more money on the SPA112, SPA2012 or
SPA3012. Users would have to move up to enterprise level ATA’s for a
boost in voice quality, but they’re usually +$250. For all of these
reasons, we recommend the PAP2T-NA to most users.<br />
<br />
<b>What are codecs?</b><br />
A codec is short for <b>co</b>der-<b>dec</b>oder or <b>co</b>mpression-<b>dec</b>ompression
and it is usually a piece of software that encodes and decodes a
certain data file. A codec is commonly used to shrink video and music
files. For example, .MP3 files have been compressed with a codec (<a href="http://lame.sourceforge.net/" rel="nofollow" target="_blank">a popular MP3 encoder is called Lame Mp3 Encoder</a>).
Most video files uses codecs as well. Popular video codecs include
DivX, Mpeg-4, H.264, WMV, etc. to compress video files into a more
manageable size.<br />
<br />
<b>Which codec do you recommend?</b><br />
The
two most popular codecs for VOIP telephony are G.711 and G.729. We
recommend the usage of G.711 as it is the voice coder that is the most
similar to a landline phone. However, it requires a sustained upstream
and downstream data rate of 100Kbps (12KB/s). G.711 gives you “toll
quality” calling. Another popular alternative is codec G.729. The data
rate requirement for G.729 is only 8Kbps (or 1KB/s which is 12.5% of the
G.711). Therefore, it will use a lot less bandwidth, but voice quality
may be slightly inferior. You should know that dial tones cannot be
decoded with G.729, therefore you won’t be able to use a fax machine
with this coder. G.729 is very efficient at transmitting voice calls.
However, the human voice is synthesized by something called a vocoder.
If you understand how MIDI audio files work, vocoder uses a similar
principle. Instead of sending the speaker’s voice, G.729 analyzes the
speaker’s voice and compares it to a synthetic voice in its library. It
comes up with a certain formula that will best mimic the speaker’s voice
using the synthetic voice as a base. Therefore, it wouldn’t be
necessary to send the speaker’s voice across the internet. G.729 will
send the formula instead (which uses a lot less bandwidth). The other
person on the line hears your voice as it is being reproduced on their
side with the same formula. The end result of all this work is a voice
quality that sounds similar to G.711 but at almost a third of the
bandwidth usage. Vocoder uses a lot of computing power, therefore many
ATA’s only support G.729 for one phone line or channel (as in the
PAP2T). If you’d like to use G.729 for 2 phone lines, you need to buy
the SPA2102.<br />
We have conducted multiple non scientific tests
comparing G.711 and G.729 and we prefer G.711 much much more. The voices
sound a lot less robotic and sound crystal clear in comparison. We
don’t imagine ever going back to G.729 (which we used for a year). When
configuring the ATA, please note that G.711a is used for Australia and
Europe while G.711u is used in Canada and the United States. In
consideration of all of the above, the only voice coder that we can
recommend is G.711.<br />
<br />
<b>Which VOIP service provider do you recommend?</b><br />
There are many VOIP service providers out there, but we recommend www.freephoneline.ca
as it’s the only one that we know of that doesn’t require a monthly
fee. Please note that a VOIP service provider are also known as SIP
servers (Session Internet Protocol) or SIP proxies.<br />
<br />
<b>How much does freephoneline cost?</b><br />
There is a one time of $50 to activate the VOIP service.<br />
<br />
<b>Can I keep my phone number?</b><br />
<a href="http://support.freephoneline.ca/articles/FAQ/Keep-your-current-phone-number-porting/?l=en_US&fs=Search&pn=1" rel="nofollow" target="_blank">Yes</a>. You can port your current number to freephoneline by calling freephonline or sending them a Letter of Authorization found <a href="http://support.freephoneline.ca/servlet/fileField?id=0BEa0000000TPLD" rel="nofollow" target="_blank">here</a>. There is one time fee of $25 to port your number. Please check to see if your telephone number can be ported <a href="http://support.freephoneline.ca/articles/FAQ/Phone-numbers-that-can-be-ported-to-the-Fongo-network/?l=en_US&fs=Search&pn=1" rel="nofollow" target="_blank">here</a>.
Most area codes in Canada can be ported over to freephoneline. You
should contact freephoneline before you let your telephony company know
that you would like to port your number or terminate your service as
only “active” telephone numbers can be ported. If your number is flagged
as “disconnected” by your telephone company prior to porting, it may be
more difficult to port your number.<br />
<br />
<b>Does 911 still work?</b><br />
Yes.
You can still call 911, however you have to input your address and
contact details on freephonline’s website. Otherwise, emergency
personnel would not know where to go. You should update this information
when you change residence.<br />
<br />
<b>What happens if my internet isn’t working and I need to make a call?</b><br />
As it is a VOIP service, the telephony service is dependent on the internet. Therefore, no internet, no telephone.<br />
<br />
<b>Can I check my call logs online?</b><br />
Yes. Login
to freephoneline and click on “Call Logs”. Here you can see the call
logs for the current month and for the last 2 months.<br />
<br />
<b>How do I setup my voice mail password?</b><br />
Login
to freephoneline, click on “App Settings”, click the “Settings” button,
click “Voice Mail Settings” button, enter in your voice mail password.
You also have the option of entering in your email address if you want
your voice messages to be sent to your email address as an audio file.<br />
<br />
<b>How do I record my voice mail greeting?</b><br />
You
can set your voice mail greeting by dialing *98, pressing 3 for
Personal Options, then 3 again for Greetings where you will have the
option to record your greeting.<br />
<br />
<b>How do I access my voice mail?</b><br />
Please dial *98 on your home phone.<br />
<br />
<b>Does it support caller ID?</b><br />
Yes.
Caller ID is activated automatically. There is no extra fee. All
incoming phone numbers are displayed on the cordless phone. If the
number is saved in your contracts you can customize what name appears.<br />
<br />
<b>Can I forward my VOIP number?</b><br />
Yes.
You can forward your VOIP number to your cell phone or office line so
that people can always reach you through your VOIP number. To do this,
login to freephoneline, click on “App Settings”, click on the “Settings”
button, click on “Follow Me Settings” button, select “Enable” in
“Status” and for “Ring Mode”, either choose “Sequential” or
“Simultaneous”. Simultaneous means that your VOIP number and all of your
“Follow Me Numbers” will ring at the same time when someone calls the
VOIP number. The first person that picks up will be connected to the
caller. Sequential means that if you don’t answer the VOIP number, the
caller will be redirected to the 1st “Follow Me Number” you specified
after your specified number of rings. If no one answers after the
specified number of rings, it will redirect the caller to the 2nd
“Follow Me Number” and so on until all of the “Follow Me Numbers” have
been dialed. Once you’re satisfied, click “Submit”. A new button has
appeared, now click “Follow Me Numbers” to enter in your alternate
numbers and push submit.<br />
<br />
That’s it! If you require any
clarifications or have any comments on any of the above, please drop us a
line the comments below and we’ll respond to you shortly.<br />
<br />
<br />Unknownnoreply@blogger.com0Toronto, ON, Canada43.653226 -79.383184315.342992163821151 -114.5394343 71.963459836178842 -44.226934299999996tag:blogger.com,1999:blog-3402630850730921975.post-3369903520030448842013-09-11T13:41:00.003-07:002021-02-09T13:06:00.640-08:00Freephoneline.ca ATA Disconnect RESOLVEThe most comment support related issue at Freephoneline-Fongo is <b>ATA voip disconnects</b> and are related to Registration Interval.<br />
<br />
As the owner of 13 voip ATA's and have sent many Freephoneline-Fongo
support tickets. I also made many support phone calls to
Freephoneline-Fongo. With the ATA; <span style="font-size: small;"><b>OBi, VTA-CV, Granddstream, etc)</b><span style="font-size: x-small;"> all issues have been resolved with setup changes.</span></span><br />
<span style="font-size: small;"></span><br />
<br />
It seems recently as of September 2013 all the ATA with default setting
have had disconnect problems, my question then, is Freephoneline-Fongo
making changes, well I can not comment. But I have had to re-setup all
the ATA's that have been working for years until recent September
2013. <br />
<br />
<br />
<a name='more'></a><br />
<br />
<br />
<span style="font-size: small;"><b>Freephoneline.ca ATA Disconnect </b></span><span style="font-size: small;"><b>RESOLVE</b></span><br />
<span style="font-size: small;"><b>Reason is as follows: Registration Period</b></span><br />
The Registry expiry on the ATA needs to be setup for 3600 as per FPL guidelines: <br />
<br />
<div style="color: #2b2e2f; font-family: 'Lucida Sans Unicode','Lucida Grande','Tahoma',Verdana,sans-serif; font-size: 14px; line-height: 22px; margin-bottom: 15px; margin-top: 15px;">
<a href="http://support.freephoneline.ca/entries/23120323-VoIP-Unlock-Key-Credentials">http://support.freephoneline.ca/entries/23120323-VoIP-Unlock-Key-Credentials</a> </div>
<div style="color: #2b2e2f; font-family: 'Lucida Sans Unicode','Lucida Grande','Tahoma',Verdana,sans-serif; font-size: 14px; line-height: 22px; margin-bottom: 15px; margin-top: 15px;">
The values that need your attention are below: </div>
<div style="color: #2b2e2f; font-family: 'Lucida Sans Unicode','Lucida Grande','Tahoma',Verdana,sans-serif; font-size: 14px; line-height: 22px; margin-bottom: 15px; margin-top: 15px;">
Registration Interval:3600 seconds (1 hour) </div>
<div style="color: #2b2e2f; font-family: 'Lucida Sans Unicode','Lucida Grande','Tahoma',Verdana,sans-serif; font-size: 14px; line-height: 22px; margin-bottom: 15px; margin-top: 15px;">
Registration Expiry: 3600 seconds (1 hour) </div>
<div style="color: #2b2e2f; font-family: 'Lucida Sans Unicode','Lucida Grande','Tahoma',Verdana,sans-serif; font-size: 14px; line-height: 22px; margin-bottom: 15px; margin-top: 15px;">
Failed Registration Re-Try Interval:120 seconds </div>
<div style="color: #2b2e2f; font-family: 'Lucida Sans Unicode','Lucida Grande','Tahoma',Verdana,sans-serif; font-size: 14px; line-height: 22px; margin-bottom: 15px; margin-top: 15px;">
<br />
Your ATA is currently setup with 600 instead of 3600. </div>
Please try to make the necessary adjustments in order to avoid possible issues.Unknownnoreply@blogger.com2Toronto, ON, Canada43.653226 -79.383184315.342992163821151 -114.5394343 71.963459836178842 -44.226934299999996tag:blogger.com,1999:blog-3402630850730921975.post-89001857369554971782013-09-11T13:40:00.003-07:002021-02-09T13:05:53.972-08:00Freephoneline.ca ATA Registration Period RESOLVE<br />
<b>RegistrationPeriod</b> that need to be changed from 60 sec to 3600 sec<br />
<br />
the RegistrationPeriod is located in the Expert menu under <br />
<br />
<b>Service Providers > ITSP Profile X SIP > RegistrationPeriod</b><br />
<br />
Uncheck the Obitalk Settings and then the Device Default and change the value for 3600<br />
<br />
That should fixed the "486 Too many registers" Error Message<br />
<br />
<b>Default Value</b><br />
<b>RegistrationPeriod: 60</b><br />
<br />
<br />
<a name='more'></a><br />
<br />
<br />
<b>Freephoneline | Fongo Support</b><br />
1. Open the Service Providers menu, the ITSP Profile A, B, C or D (Obi202). Obi100 & 110 has only A & B<br />
2. Click on SIP<br />
3. Uncheck the default<br />
4. Change Registration period from “60” to “3600” (no double quote) <br />
Submit|Save - REBOOTUnknownnoreply@blogger.com1Toronto, ON, Canada43.653226 -79.383184315.342992163821151 -114.5394343 71.963459836178842 -44.226934299999996tag:blogger.com,1999:blog-3402630850730921975.post-61540753877661202382013-07-25T08:36:00.002-07:002021-02-09T13:05:39.855-08:00Freephoneline VoIP Unlock Key Credentials<b>The Freephoneline VoIP Unlock Key provides sip credentials
that can be used to configure any SIP client to work with the
Freephoneline service. Please review the following Freephoneline
guidelines to set up your SIP client. Failure to follow the required
guidelines will result in account suspension followed by a notification
email. Once your configuration adheres to the guidelines, service will
be restored.</b>
<br />
<i>These guidelines may change over time - if and when they are
changed freephoneline users will be notified at the email address used
for account login a minimum of 7 days prior to the changes being
required. Please ensure your contact information is accurate by visiting
your account profile on freephoneline.ca.</i><br />
<i></i><br />
<a name='more'></a><i> </i><br />
<hr />
<h3>
<b>Required Settings</b></h3>
<table>
<tbody>
<tr>
<td>SIP Server:</td>
<td><b>voip.freephoneline.ca </b></td>
</tr>
<tr>
<td>Alternative SIP Server: <b></b></td>
<td><b>voip2.freephoneline.ca</b></td>
</tr>
<tr>
<td>Transport: </td>
<td><b>UDP</b></td>
</tr>
<tr>
<td>Port: <b> </b></td>
<td><b>5060</b></td>
</tr>
</tbody>
</table>
<h4>
Notes:</h4>
<ul>
<li>
It is always best to use the DNS name for your SIP server as our
infrastructure is always expanding/changing/being maintained. The IP
addresses which you register to will change over time.<br />
</li>
<li>Use of Fongo SIP Servers that are not listed in this document will result in your account being suspended.</li>
</ul>
<table>
<tbody>
<tr>
<td>Registration Interval:</td>
<td><b>3600 seconds (1 hour)</b></td>
</tr>
<tr>
<td>Registration Expiry: </td>
<td><b>3600 seconds (1 hour)</b></td>
</tr>
<tr>
<td>Failed Registration Re-Try Interval:</td>
<td><b>120 seconds</b></td>
</tr>
</tbody>
</table>
<hr />
<h3>
<b>Recommended Settings</b></h3>
<table>
<tbody>
<tr>
<td>STUN/ICE:</td>
<td><b>Disable</b></td>
</tr>
<tr>
<td>NAT Mapping Enabled:</td>
<td><b>Yes</b></td>
</tr>
<tr>
<td>NAT Traversal:</td>
<td><b>Enable sending Keep-Alives only: </b><br />
<ul>
<li><b>on Grandstream HT-701 ATAs this setting is “no, but send keep-alive” </b></li>
</ul>
</td>
</tr>
<tr>
<td>Keep Alive Message:</td>
<td><b>NOTIFY or a UDP PING Packet</b><br />
<ul>
<li><b>For Linksys/Cisco devices, use ‘Nat Keep Alive Msg’ = $NOTIFY or $PING </b></li>
<li><b>Never use REGISTER as your Keep Alive message </b></li>
</ul>
</td>
</tr>
<tr>
<td>Keep Alive Interval: </td>
<td><b> 20 seconds*</b></td>
</tr>
</tbody>
</table>
*Audio may be affected if this value is adjusted<br />
<h4>
<b>Notes:</b></h4>
<ul>
<li>
The above settings can be used to configure your SIP client to
function in common home network configurations. Since there a thousands
of home network configurations, it is impossible for us to provide a
single set of parameters that will always work. As a VoIP Key purchaser,
it’s expected that you have knowledge of your network and how to
configure your SIP client properly.<br />
</li>
<li>
<span style="font-size: 1em; line-height: 1.45em;">Freephoneline does not offer STUN server. However, you may use a public one if your wish.</span><br />
</li>
</ul>
<hr />
<h3>
<b>RTP Settings</b></h3>
<table>
<tbody>
<tr>
<td>Supported Codecs:</td>
<td><b>g711-u (uLAW) and g729</b></td>
</tr>
<tr>
<td>Suggested RTP Packet size (psize):<b> </b></td>
<td><b>0.020 - This ensures audio packets every 20 milliseconds, achieving better quality (trade-off: bandwidth)</b></td>
</tr>
</tbody>
</table>
<h4>
<b>Notes:</b></h4>
<ul>
<li>
The above settings are used by your ATA to determine how the audio will be encoded/decoded across the Fongo network.<br />
</li>
</ul>
<hr />
<h3>
<b>Additional Information for users with multiple SIP clients on their network</b></h3>
If you use multiple VoIP providers or SIP clients, including Dell
Voice or Fongo Mobile on the same network you may encounter issues if
your router does not support UPNP.Unknownnoreply@blogger.com1Toronto, ON, Canada43.653226 -79.383184315.342992163821151 -114.5394343 71.963459836178842 -44.226934299999996tag:blogger.com,1999:blog-3402630850730921975.post-80460216626760249542013-05-28T09:04:00.001-07:002021-02-09T13:05:13.247-08:00freephoneline.ca - Free Local Soft Phone Line for lifetime VOIP I have found this pretty amazing, especially for their life time free phone line and long distance deal. If you need to attach a VOIP ATA device then you need to purchase their configuration file for a one time fee of $50.<br /><br />http://freephoneline.ca/<br /><br />Features:<br /><br />Free Phone Number<br />Free Caller ID<br />Free Voice mail<br />Free Call Forwarding<br />Free Long Distance to most Canadian Cities<br /><br />Have been using this for more than two years now, and personally speaking, works like a charm and found no difference between Bell voice quality and FPL as long as you use G711u codec. Unknownnoreply@blogger.com0Toronto, ON, Canada43.653226 -79.383184315.342992163821151 -114.5394343 71.963459836178842 -44.226934299999996tag:blogger.com,1999:blog-3402630850730921975.post-54975548733455969682013-05-08T11:22:00.001-07:002021-02-09T13:05:06.756-08:00Grandstream HT502 Setting for Freephoneline<div id="primary">
<div id="content" role="main">
<br />
<br />
<article class="page type-page status-publish hentry">
<header class="entry-header">Before you start configuring the adapter, make sure you have the following settings for your SIP account:</header><div class="entry-content">
- SIP Server address (sometimes called SIP Proxy)<br />
- SIP User ID (in most cases this is the phone number)<br />
- The password for the SIP account<br />
To obtain these settings, you must
contact Freephoneline and ask for your configuration file. There's a
one time charge for this, currently $50 CAD + tax. They will send you a
Word document with the settings.<br />
<br />
Connect all the cables: power cord, an ethernet cable from your router (or
modem) to the WAN port of the HT502, an ethernet cable from the LAN port to your
PC and a phone to
the Phone 1 port. Open a web browser and type in <a href="http://192.168.2.1/">
http://192.168.2.1</a>.
The login page will come up, enter <b>admin</b> to log in then click <b>Advanced Settings</b>:</div>
<div class="entry-content">
<br />
<a name='more'></a><br /></div>
<div class="entry-content">
<h1 class="entry-title">
Setting up the Grandstream HT502 for Freephoneline</h1>
<br />
<br />
Change the following settings:<br />
- <b>STUN server is</b>: enter the host name or IP address of a publicly available STUN server, such as stunserver.org<br />
- <b>Keep-alive interval</b>: change to 10<br />
Click Update at the bottom of the page, but don't reboot yet. Click <b>FXS Port1</b> to configure the settings for the Line 1.<br />
<br />
<br />
<br />
Enter the following settings:<br />
- <b>Primary SIP Server</b>: voip.freephoneline.ca<br />
- <b>Failover SIP Server</b>: leave blank<br />
- <b>Outbound proxy</b>: leave blank<br />
- <b>NAT Transversal (STUN)</b>: change to Yes<br />
- <b>SIP User ID</b>: enter your Freephoneline number, with "1" in front, for
example 14164771234<br />
- <b>Authenticate ID</b>: same as the SIP User ID<br />
- <b>Authenticate Password</b>: the SIP password for your Freephoneline account<br />
- <b>Name</b>: you can leave it blank or enter your name here (note that this
does not change the Caller ID)<br />
- <b>Register Expiration</b>: change to 2 minutes<br />
- (optional) <b>Local SIP port</b>: the default value is 5060. If you
have other VoIP adapters in your LAN, or computers running softphones,
you may have to change this to a value between 5060-5069 in order to
avoid conflicts<br />
- <b>Dial Plan</b>: copy/paste to the following string<br />
{*xx|[2-9]xxxxxxxxx|1[2-9]xxxxxxxxx|0xxx+}<br />
- <b>SUBSCRIBE for MWI</b>: set to Yes<br />
Scroll down to the bottom of the page and click <b>Update</b>. You will be prompted to reboot the adapter. Please allow it to reboot in order to start with the new settings.<br />
<br />
Once the adapter has come up, login back to the web settings and click the <b>
Status</b> page. A page similar to the one below will come up.<br />
<br />
<br />
<br />
Check the status for <b>Registration</b> under the port <b>FXS1</b>. If it shows <b>
Registered</b> then you're ready to place and receive calls. If it shows
<b>Not Registered</b>, wait 20-30 seconds then refresh the page. Do this a few times. If it still can't register after 3-4 retries, try changing the
<b>Local SIP port</b> as described earlier. Also check the value for <b>NAT</b>. If it shows
<b>Symmetric NAT</b>, go back to the FXS Port1 page and set the <b>NAT Transversal (STUN)</b> to <b>No, but send keep-alive</b>.<br />
<br />
<b>To factory reset the adapter</b><br />
<br />
Restoring the Factory Default settings:<br />
- find the MAC Address of the device. It is a 12 digits HEX number located on
the bottom of the unit (can also be found under the Status page).<br />
- encode the MAC address: keep the numbers as they are, and replace any of the
alpha characters like below<br />
A: 22<br />
B: 222<br />
C: 2222<br />
D: 33<br />
E: 333<br />
F: 3333<br />
For example, if the MAC address is 000b8200e395, it should be encoded as 0002228200333395<br />
- pick up the phone and dial *** (three times the * key). When you hear the voice prompts, dial 99. The adaptor will say
"reset" after which you can start entering the mac address encoded as above.
After the last digit of the MAC address has been entereed, the adapter will
factory reset and reboot. </div>
<div class="entry-content">
</div>
<div class="entry-content">
</div>
<div class="entry-content">
</div>
<div class="entry-content">
</div>
<div class="entry-content">
</div>
</article>
</div>
</div>
Unknownnoreply@blogger.com5Toronto, ON, Canada43.653226 -79.383184315.342992163821151 -114.5394343 71.963459836178842 -44.226934299999996tag:blogger.com,1999:blog-3402630850730921975.post-69181675333164463682013-04-09T07:18:00.002-07:002021-02-09T13:04:59.707-08:00Freephoneline.ca Config Info: Obihai OBi-110 ATAI read the on-line reviews of the Obihai OBi-110 and its
smaller sibling, the OBi-100, and decided to test an OBi-110 in
conjunction with FPL.<br />
<br />
The first challenge: if you purchase the
OBi-110 or OBi-100 from Amazon.com (.com not .ca) it won't get shipped
to Canada. Don't know why -- and don't really care except it involves
one more step. Have the device shipped to a friend / relative in the
good ol' US of A and then have it transshipped via UPS. It adds about
$28 to the price -- but I believe in the long run it'll be well worth
it.<br />
<br />
<span style="font-weight: bold;"><span style="text-decoration: underline;">STEP 1:</span></span><br />
I presume you can read the instructions which come with the OBi-110 / OBi-100 about what plugs in where.<br />
However,
to summarize (I purchased an OBi-110 so I'll use this going forward;
the OBi-100 is identical for purposes of these instructions):<br />
#1.
Plug in the supplied RJ-45 cable to the OBi-110 and plug the other end
of the cable into your router. I have a D-Link DIR-615 with 4 wired
ports of which 2 were free. The other two router ports have D-Link
Gigabit Ethernet DGS-1008D switches / hubs attached.<br />
#2. Plug the supplied telephone cable into the PHONE jack on the OBi-110 and the other end into your telephone handset<br />
#3.
If your router is set up for DHCP as soon as you connect the power
(standard 12 V adapter) the device should receive an IP address<br />
#4. Check that the POWER and PHONE LEDs are SOLID GREEN, and that the ACTIVITY LED flickers periodically<br />
#5. Perform an echo test of the Obihai unit (using the OBiTALK network) by dialing **9-222-222-222<br />
Alternatively you can make a test call by dialing **9-333-333-333 followed by the outgoing number<br />
<br />
<a name='more'></a><br /><br />
<span style="font-weight: bold;">IF THE ABOVE IS SUCCESSFUL, YOUR OBIHAI DEVICE IS WORKING AND YOU CAN MOVE ON TO THE NEXT STEP</span><br />
<br />
<span style="font-weight: bold;"><span style="text-decoration: underline;">STEP 2:</span></span><br />
#1.
I don't like random IP addresses, so I configured the OBi-110 to have a
static IP address. This can be done on the device itself by following
the voice prompts. To enter a "period" press the "star" key on your
phone handset. I gave my OBi-110 a fixed IP address of 192.168.0.188<br />
#2.
To make sure the router is on the same page, I logged-in to the D-Link
DIR-615 and mapped the OBi-110's MAC address to the IP address above<br />
<br />
<span style="font-weight: bold;">AND NOW THE FUN BEGINS</span><br />
<br />
<span style="font-weight: bold;"><span style="text-decoration: underline;">STEP 3:</span></span><br />
#1. If you haven't already, purchase the SIP "key" from FPL -- $50.00 plus HST ($56.50)<br />
#2. The e-mail will contain 3 items of info:<br />
1. The Server Name (FPL calls it a "Realm"):
voip.freephoneline.ca<br />
2. Your User Name (the FREE DID telephone number you signed
up for): 1AAAXXXNNNN (Your Number Goes Here)<br />
3. Your Password (unique to you; this is the SIP "key"):
12345678 (Your Password Goes Here)<br />
#3. Log in to the web configuration for the OBi-110: <a class="postlink" href="http://192.168.0.188/">http://192.168.0.188</a> using the default admin user name and password<br />
<br />
-------------------------------------------------------------------------------------------------------------------------------------------<br />
<span style="font-weight: bold;">Please note that in order to change the settings, you need to Uncheck the "Default" box on the right hand side.</span><br />
-------------------------------------------------------------------------------------------------------------------------------------------<br />
<br />
#4. Navigate to "System Management --> Network Settings"<br />
<br />
<div class="inline-attachment">
<div class="separator" style="clear: both; text-align: center;">
<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEh_cW9-y1K4eU5Mbd-mFPZ1Vf7HcOVx0S3BavT3HZhwe38d4aTT89mrhR_xsn-LOHi9WFd8qycXwKt_d58r44VBJQwG957kmnLp05J6jXBKg8qtxGRNlCVPF9XStBchfJ1y6588kCPazY8/s1600/1+System+Management+-+Network+Settings.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" height="305" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEh_cW9-y1K4eU5Mbd-mFPZ1Vf7HcOVx0S3BavT3HZhwe38d4aTT89mrhR_xsn-LOHi9WFd8qycXwKt_d58r44VBJQwG957kmnLp05J6jXBKg8qtxGRNlCVPF9XStBchfJ1y6588kCPazY8/s320/1+System+Management+-+Network+Settings.jpg" width="320" /></a></div>
<dl class="thumbnail"></dl>
</div>
<br />
#5. As you see from the screen capture, you need only to
enter the "Local Time Zone" -- and this isn't critical to the
functioning of the Obi-110 with FPL. It does help if you care about
accurate call logs.<br />
#6. Click on "Submit" and you will be prompted to reboot the OBi-110. Go ahead and do so as it takes less than 30 seconds.<br />
<br />
<br />
<span style="font-weight: bold;"><span style="text-decoration: underline;">STEP 4:</span></span><br />
#1. If prompted, log in to the web configuration for the OBi-110: <a class="postlink" href="http://192.168.0.188/">http://192.168.0.188</a> using the default admin user name and password<br />
#2. Navigate to "Service Providers --> ITSP Profile A --> General"<br />
<br />
<div class="inline-attachment">
<div class="separator" style="clear: both; text-align: center;">
<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgW3A18Tumf73iqKmm3Kf6pVVYjj6ynD-bgeRHzVf-h7_CFbQsEUdZP2OoEk6A6t3MW97cXIyzH1vI7Bz4CK9frkzleKnv13O8dTYZQb-DXlheRIh-IVq0YXRdo2Af3CvKUKuh8tUz8Duk/s1600/2+Service+Providers+-+ITSP+Profile+A+-+General.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" height="232" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEgW3A18Tumf73iqKmm3Kf6pVVYjj6ynD-bgeRHzVf-h7_CFbQsEUdZP2OoEk6A6t3MW97cXIyzH1vI7Bz4CK9frkzleKnv13O8dTYZQb-DXlheRIh-IVq0YXRdo2Af3CvKUKuh8tUz8Duk/s320/2+Service+Providers+-+ITSP+Profile+A+-+General.jpg" width="320" /></a></div>
<dl class="thumbnail"></dl>
</div>
<br />
#3. As you see from the screen capture, you need only to
enter the "ITSP Name" Please confirm that "SignalingProtocol" shows
"SIP" as the Default.<br />
#4. Click on "Submit" and you will be prompted to reboot the OBi-110. Go ahead and do so.<br />
<br />
<br />
<span style="font-weight: bold;"><span style="text-decoration: underline;">STEP 5:</span></span><br />
#1. If prompted, log in to the web configuration for the OBi-110: <a class="postlink" href="http://192.168.0.188/">http://192.168.0.188</a> using the default admin user name and password<br />
#2. Navigate to "Service Providers --> ITSP Profile A --> SIP"<br />
<br />
<div class="inline-attachment">
<div class="separator" style="clear: both; text-align: center;">
<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEg9uvluKDOD1pxn8UKBiLsCwn-WMqJD5v4Ar-IfpfinG-NOUYJ7YGJs9eAXiG98mY7BwYqMFSg-BIh0hAD8_TJjcFzEGZwGJjAzdHKhLTiQL_KQrGk0tHNi_Oy4KOSEqCFTzSJ2FEPG5qs/s1600/3+Service+Providers+-+ITSP+Profile+A.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" height="307" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEg9uvluKDOD1pxn8UKBiLsCwn-WMqJD5v4Ar-IfpfinG-NOUYJ7YGJs9eAXiG98mY7BwYqMFSg-BIh0hAD8_TJjcFzEGZwGJjAzdHKhLTiQL_KQrGk0tHNi_Oy4KOSEqCFTzSJ2FEPG5qs/s320/3+Service+Providers+-+ITSP+Profile+A.jpg" width="320" /></a></div>
<dl class="thumbnail"></dl>
</div>
<br />
#3. As you see from the screen capture, you need only to
enter the "ProxyServer" name. Enter "voip.freephoneline.ca" without
the quotes.<br />
#4. Click on "Submit" and you will be prompted to reboot the OBi-110. Go ahead and do so.<br />
<br />
<br />
<span style="font-weight: bold;"><span style="text-decoration: underline;">STEP 6:</span></span><br />
#1. If prompted, log in to the web configuration for the OBi-110: <a class="postlink" href="http://192.168.0.188/">http://192.168.0.188</a> using the default admin user name and password<br />
#2. Navigate to "Voice Services --> SP1 Service --> SIP Credentials"<br />
<br />
<div class="inline-attachment">
<div class="separator" style="clear: both; text-align: center;">
<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiG9D0DKpEhsNYmv__Hwyn5KSEdFhXghPXQqLJEbL2QXRb1MRYPAJ-wiQLBwwg7q3Wq3lMKmS-dsTTsH4mCg2xU5pp_dIEug-TIvBEknPdCXW89eUi7537hmkL3gRxR9vjxcesLRjeKkUI/s1600/4+Voice+Services+-+SP1+Service.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" height="320" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEiG9D0DKpEhsNYmv__Hwyn5KSEdFhXghPXQqLJEbL2QXRb1MRYPAJ-wiQLBwwg7q3Wq3lMKmS-dsTTsH4mCg2xU5pp_dIEug-TIvBEknPdCXW89eUi7537hmkL3gRxR9vjxcesLRjeKkUI/s320/4+Voice+Services+-+SP1+Service.jpg" width="308" /></a></div>
<dl class="thumbnail"></dl>
</div>
<br />
#3. As you see from the screen capture, you need to enter:<br />
1. The "AuthUserName" which should be your FPL Telephone Number<br />
2. The "AuthPassword" which is the SIP "key" you purchased for $56.50<br />
3. The "Caller ID Name" which is good phone etiquette.
The field allows -- and the OBi-110 transmits -- lowercase text.<br />
Note: If you want to block your ID you can do so on a one-off (*67) or persistent basis (*81)<br />
<br />
#4. Click on "Submit" and you will be prompted to reboot the OBi-110. Go ahead and do so.<br />
<br />
<br />
<span style="font-weight: bold;"><span style="text-decoration: underline;">STEP 7:</span></span><br />
#1. If prompted, log in to the web configuration for the OBi-110: <a class="postlink" href="http://192.168.0.188/">http://192.168.0.188</a> using the default admin user name and password<br />
#2. Navigate to "Physical Interfaces --> PHONE Port"<br />
<br />
<div class="inline-attachment">
<div class="separator" style="clear: both; text-align: center;">
<a href="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEg_8oiJRUEkP44rlRPydwBWqMBGdvhTy3SjfdtTP0dWQf6pl5vAEtWVgvJoD2MenAH0qQYSAlLHz5NlygFMiN1JNLhrp11R9jie7wkUWHOSCjSisVbgJ-mAxX1zHWT9A6xNVKDzMaqgoSw/s1600/5+Physical+Interfaces+-+PHONE+Port.jpg" imageanchor="1" style="margin-left: 1em; margin-right: 1em;"><img border="0" height="288" src="https://blogger.googleusercontent.com/img/b/R29vZ2xl/AVvXsEg_8oiJRUEkP44rlRPydwBWqMBGdvhTy3SjfdtTP0dWQf6pl5vAEtWVgvJoD2MenAH0qQYSAlLHz5NlygFMiN1JNLhrp11R9jie7wkUWHOSCjSisVbgJ-mAxX1zHWT9A6xNVKDzMaqgoSw/s320/5+Physical+Interfaces+-+PHONE+Port.jpg" width="320" /></a></div>
<dl class="thumbnail"></dl>
</div>
<br />
#3. As you see from the screen capture, you only need to
change the value for "PrimaryLine" to "SP1 Service" [which should be
FPL]<br />
#4. Click on "Submit" and you will be prompted to reboot the OBi-110. Go ahead and do so.<br />
<br />
<span style="font-weight: bold;">IF EVERYTHING WAS SET UP CORRECTLY PICK UP THE HANDSET AND MAKE A CALL</span><br />
<br />
<span style="font-weight: bold;">IF IT WORKS . . . G R E A T !!</span><br />
<br />
<span style="font-weight: bold;">IF IT DOESN'T . . . I'M NOT ACCEPTING ANY LIABILITY OR RESPONSIBILITY</span><br />
<br />
<span style="font-weight: bold;"><span style="text-decoration: underline;">REMARKS:</span></span><br />
I've
made a few calls local to Toronto and a LD call to Montreal and the
call quality is superb. A barely noticeable notch below Ma Bell's
POTS (or PSTN if you prefer) -- and at unbeatable value !!<br />
<br />
I'll
stress-test over the next 4 to 6 weeks . . . and if FPL and the
OBi-110 both hold up (now I've got to UPS the modem / router / ATA /
tel set combo for high availability) I'll make the move to port my Bell
land-line and say "Buh Bye" to Ma Bell. All for about $150.00 all-in.<br />
<br />
BTW,
there's much much more the OBi-110 can do: handle a second SIP / VoIP
provider; bridge PSTN and VoIP; call other OBi-110s and OBi-100s on
the OBiTALK network (every device has a unique 9-digit number) anywhere,
dialing plans, etc. etc. that it'll take me <span style="font-weight: bold;">months</span> to exploit even half of its capabilities.<br />
<br />
I
hope this post helps you if you're looking to purchase and ultimately
configure an Obihai OBi-110 or OBi-100 ATA with FPL. Small wonder
NerdVittles called the OBi-110 the "2011 VoIP Device of the Year" back
in January !!<br />
<br />
<br />
<br />Unknownnoreply@blogger.com5Toronto, ON, Canada43.653226 -79.383184315.342992163821151 -114.5394343 71.963459836178842 -44.226934299999996tag:blogger.com,1999:blog-3402630850730921975.post-43144297265295235832013-03-28T08:51:00.002-07:002021-02-09T13:04:42.713-08:00GrandStream HT-286 Configuration for freephoneline.ca <b>GrandStream HT-286 Configuration for freephoneline.ca </b><br />
<br />
<b>ATA Config - GrandStream HT-286</b><br />
<br />
http://support.fongo.com/attachments/token/l7ddqvurbehk2yf/?name=ATA+Config+-+GrandStream+HT-286+%2528Nov2010%2529+%281%29.pdf<br />
<br />
<div id="primary">
<div id="content" role="main">
<br />
<article class="page type-page status-publish hentry">
<header class="entry-header">
<h1 class="entry-title">
Setting up the Grandstream HT286 for Freephoneline</h1>
</header>
<div class="entry-content">
Before you start configuring the adapter, make sure you have the following settings for your SIP account:<br />
- SIP Server address (sometimes called SIP Proxy)<br />
- SIP User ID (in most cases this is the phone number)<br />
- The password for the SIP account<br />
To obtain these settings, you must
contact Freephoneline and ask for your configuration file. There's a
one time charge for this, currently $50 CAD + tax. They will send you a
Word document with the settings.<br />
<br />
Connect all the cables: power cord, an Ethernet cable to router and a phone to
the Phone port. Then pick up the phone and either push the white button on the
adapter or dial *** from the phone. The adapter should start playing some menu
options. Dial ** and the adapter will read back (with voice) the IP address it
has obtained from your router. Open a web browser and type in that IP address.
The login page will come up, enter <b>admin</b> to log in. You will be
automatically taken to the Advanced Settings page:<br />
<br />
<a name='more'></a><br />
<br />
<img border="0" height="1823" src="http://voipfan.net/ATAs/images/HT286_freephoneline_advanced.png" width="804" /><br />
<br />
Enter the following settings:<br />
- <b>SIP Server</b>: voip.freephoneline.ca<br />
- <b>Outbound proxy</b>: leave blank<br />
- <b>SIP User ID</b>: enter your Freephoneline number, with "1" in front, for
example 16475551234<br />
- <b>Phone number</b>: same as the SIP User ID<br />
- <b>Authenticate Password</b>: the SIP password for your Freephoneline account<br />
- <b>Name</b>: you can leave it blank or enter your name here (note that this
does not change the Caller ID)<br />
- <b>Register Expiration</b>: change to 120<br />
- (optional) <b>Local SIP port</b>: the default value is 5060. If you
have other VoIP adapters in your LAN, or computers running softphones,
you may have to change this to a value between 5060-5069 in order to
avoid conflicts<br />
- <b>NAT Transversal</b>: change to Yes, and set the STUN server to a publicly
available server, such as stunserver.org<br />
- <b>Keep-alive interval</b>: change to 10<br />
- <b>SUBSCRIBE for MWI</b>: set to Yes<br />
Scroll down to the bottom of the page and click <b>Update</b>. You will be prompted to reboot the adapter. Please allow it to reboot in order to start with the new settings.<br />
<br />
Once the adapter has come up, login back to the web settings and click the <b>
Status</b> page. A page similar to the one below will come up.<br />
<br />
<img border="0" height="569" src="http://voipfan.net/ATAs/images/HT286_status.png" width="804" /><br />
<br />
Check the status for <b>Registered</b>. If it shows <b>Yes</b> then you're ready to place and receive calls. If it shows
<b>No</b>, wait 20-30 seconds then refresh the page. Do this a few times. If it still can't register after 3-4 retries, try changing the
<b>Local SIP port</b> as described earlier. Also check the value for <b>NAT</b>. If it shows
<b>detected NAT type is symmetric NAT</b>, go back to the Advanced Settings page set the
<b>NAT Transversal</b> to No.<br />
<br />
<b>To factory reset the adapter</b><br />
<br />
Restoring the Factory Default settings:<br />
- find the MAC Address of the device. It is a 12 digits HEX number located on
the bottom of the unit (can also be found under the Status page).<br />
- encode the MAC address: keep the numbers as they are, and replace any of the
alpha characters like below<br />
A: 22<br />
B: 222<br />
C: 2222<br />
D: 33<br />
E: 333<br />
F: 3333<br />
For example, if the MAC address is 000b8200e395, it should be encoded as 0002228200333395<br />
- pick up the phone and either push the white button on the adapter or dial ***
from the phone. When you hear the voice prompts, dial 99. The adaptor will say
"reset" after which you can start entering the mac address encoded as above.
After the last digit of the MAC address has been entereed, the adapter will
factory reset and reboot. </div>
<div class="entry-content">
</div>
<div class="entry-content">
</div>
<div class="entry-content">
To access the web interface: dial ‘<br />
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***</div>
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’ then ‘</div>
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02</div>
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’ from the phone attached to the ATA</div>
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.</div>
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Next, enter the IP address provided in your web browser.</div>
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Login password is </div>
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admin</div>
<br />
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Under the ‘Advanced Settings 1’ page, you will find all the settings necessary to make the HT</div>
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-</div>
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286 work </div>
with the freephoneline.ca services<br />
<br />
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SIP Server</div>
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: voip.</div>
<div data-canvas-width="100.94976300853727" data-font-name="g_font_p0_7" dir="ltr" style="font-family: sans-serif; font-size: 14.72px; left: 193.467px; top: 800.773px; transform-origin: 0% 0% 0px; transform: scale(0.943456, 1);">
freephoneline.ca</div>
<div data-canvas-width="66.50496198200226" data-font-name="g_font_p0_1" dir="ltr" style="font-family: sans-serif; font-size: 14.72px; left: 96.032px; top: 821.413px; transform-origin: 0% 0% 0px; transform: scale(0.875065, 1);">
SIP User ID</div>
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: freephoneline number</div>
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Authenticate Password</div>
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: SIP password</div>
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Name</div>
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: Caller</div>
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-</div>
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ID Name (currently only supported on SIP to SIP calls)</div>
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Preferred Vocoder</div>
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: </div>
<div data-canvas-width="131.28768391267778" data-font-name="g_font_p0_7" dir="ltr" style="font-family: sans-serif; font-size: 14.72px; left: 215.867px; top: 896.613px; transform-origin: 0% 0% 0px; transform: scale(0.924561, 1);">
Follow diagram above</div>
<br />
<div data-canvas-width="115.34592343757627" data-font-name="g_font_p0_1" dir="ltr" style="font-family: sans-serif; font-size: 14.72px; left: 96.032px; top: 766.333px; transform-origin: 0% 0% 0px; transform: scale(0.937772, 1);">
Register Expiration</div>
<div data-canvas-width="136.54272406929016" data-font-name="g_font_p0_7" dir="ltr" style="font-family: sans-serif; font-size: 14.72px; left: 211.387px; top: 766.333px; transform-origin: 0% 0% 0px; transform: scale(0.935224, 1);">
: 3600 (usually default)</div>
<div data-canvas-width="17.738773861989973" data-font-name="g_font_p0_1" dir="ltr" style="font-family: sans-serif; font-size: 14.72px; left: 96.032px; top: 800.293px; transform-origin: 0% 0% 0px; transform: scale(0.985487, 1);">
No</div>
<div data-canvas-width="112.24000334501265" data-font-name="g_font_p0_1" dir="ltr" style="font-family: sans-serif; font-size: 14.72px; left: 116.992px; top: 800.293px; transform-origin: 0% 0% 0px; transform: scale(0.943193, 1);">
Key Entry Timeout</div>
<div data-canvas-width="427.86625275140756" data-font-name="g_font_p0_7" dir="ltr" style="font-family: sans-serif; font-size: 14.72px; left: 229.347px; top: 800.293px; transform-origin: 0% 0% 0px; transform: scale(0.932171, 1);">
: 2 (number of seconds to wait before dialing after a number is entered)</div>
<div data-canvas-width="103.6435230888176" data-font-name="g_font_p0_1" dir="ltr" style="font-family: sans-serif; font-size: 14.72px; left: 96.032px; top: 834.213px; transform-origin: 0% 0% 0px; transform: scale(0.950858, 1);">
Use random port</div>
<div data-canvas-width="28.468480848426818" data-font-name="g_font_p0_7" dir="ltr" style="font-family: sans-serif; font-size: 14.72px; left: 199.547px; top: 834.213px; transform-origin: 0% 0% 0px; transform: scale(0.769418, 1);">
: YES</div>
<div data-canvas-width="70.42048209869384" data-font-name="g_font_p0_1" dir="ltr" style="font-family: sans-serif; font-size: 14.72px; left: 96.032px; top: 868.133px; transform-origin: 0% 0% 0px; transform: scale(0.902827, 1);">
Disable Call</div>
<div data-canvas-width="4.5043201342391965" data-font-name="g_font_p0_1" dir="ltr" style="font-family: sans-serif; font-size: 14.72px; left: 166.587px; top: 868.133px; transform-origin: 0% 0% 0px; transform: scale(0.900864, 1);">
-</div>
<div data-canvas-width="85.61152255142213" data-font-name="g_font_p0_1" dir="ltr" style="font-family: sans-serif; font-size: 14.72px; left: 171.067px; top: 868.133px; transform-origin: 0% 0% 0px; transform: scale(0.951239, 1);">
Waiting Caller</div>
<div data-canvas-width="4.5043201342391965" data-font-name="g_font_p0_1" dir="ltr" style="font-family: sans-serif; font-size: 14.72px; left: 256.707px; top: 868.133px; transform-origin: 0% 0% 0px; transform: scale(0.900864, 1);">
-</div>
<div data-canvas-width="13.343307064328192" data-font-name="g_font_p0_1" dir="ltr" style="font-family: sans-serif; font-size: 14.72px; left: 261.187px; top: 868.133px; transform-origin: 0% 0% 0px; transform: scale(0.953093, 1);">
ID</div>
<div data-canvas-width="26.378240786132814" data-font-name="g_font_p0_7" dir="ltr" style="font-family: sans-serif; font-size: 14.72px; left: 274.467px; top: 868.133px; transform-origin: 0% 0% 0px; transform: scale(0.879275, 1);">
: NO</div>
<div data-canvas-width="69.47840207061768" data-font-name="g_font_p0_1" dir="ltr" style="font-family: sans-serif; font-size: 14.72px; left: 96.032px; top: 902.053px; transform-origin: 0% 0% 0px; transform: scale(0.890749, 1);">
Send DTMF</div>
<div data-canvas-width="46.33856138099671" data-font-name="g_font_p0_7" dir="ltr" style="font-family: sans-serif; font-size: 14.72px; left: 165.467px; top: 902.053px; transform-origin: 0% 0% 0px; transform: scale(0.858122, 1);">
: Check </div>
<div data-canvas-width="14.793600440883637" data-font-name="g_font_p0_2" dir="ltr" style="font-family: sans-serif; font-size: 14.72px; left: 211.867px; top: 902.053px; transform-origin: 0% 0% 0px; transform: scale(1.05669, 1);">
‘in</div>
<div data-canvas-width="4.5043201342391965" data-font-name="g_font_p0_7" dir="ltr" style="font-family: sans-serif; font-size: 14.72px; left: 226.627px; top: 902.053px; transform-origin: 0% 0% 0px; transform: scale(0.900864, 1);">
-</div>
audio’ and ‘via RTP (RFC2833)<br />
<div data-canvas-width="58.66896174847127" data-font-name="g_font_p0_1" dir="ltr" style="font-family: sans-serif; font-size: 21.28px; left: 202.107px; top: 935.808px; transform-origin: 0% 0% 0px; transform: scale(0.761935, 1);">
</div>
<div data-canvas-width="275.4270482083654" data-font-name="g_font_p0_1" dir="ltr" style="font-family: sans-serif; left: 338.307px; top: 935.808px; transform-origin: 0% 0% 0px; transform: scale(0.858028, 1);">
<span style="font-size: small;"><b>PRESS UPDATE OR SETTINGS WILL NOT APPLY!</b></span></div>
</div>
<div class="entry-content">
</div>
</article>
</div>
</div>
Unknownnoreply@blogger.com0Toronto, ON, Canada43.653226 -79.383184315.342992163821151 -114.5394343 71.963459836178842 -44.226934299999996tag:blogger.com,1999:blog-3402630850730921975.post-566053149260052382013-03-21T08:31:00.002-07:002021-02-09T13:04:21.633-08:00Factory reset your OBi110<b>There are 3 ways to do this:</b><br />
1. Power on the unit (no need to connect
phone or internet), and look for the distinctive hole on the back of
the OBi110 by the top-left corner of the product label. <b>Insert tip of
a paper-clip and press for 10 seconds</b>. The power LED will blink and
the unit will reboot.<br />
<br />
2. With the unit connected to the same
network as your PC, open the webpage of the device. Under <b>System
Management -> Device Update -> Reset Configuration.</b><br />
<br />
Option: <br />
3. Enter ***81 on the phone connected to the OBi's PHONE port. <br />
<b>Pressing ***81 reboots the device and it then shows in Obitalk as online-unmanaged.</b><br />
<br />
<blockquote class="tr_bq">
I'm not sure how I lost my OBi config settings.<br />
I did ***81 to
test it, and was poking around the OBi expert configuration in the OBi
portal. I now find that all my OBi settings are set to default, the
password was reset to admin. </blockquote>
<br />
<a name='more'></a><br />
<br />
Last time I had one show as unmanaged I couldn't figure out how to change the status so I just deleted it and re added it.<br />
<br />
after pressing ***81<br />
<br />
From the dashboard<br />
device shows as online-unmanaged<br />
device configuration still shows my selectred settings <br />
the service provider still shows my selected choice (google voice) but it shows "service not configured"<br />
clicking on "service provider", prior selected settings still showing<br />
<br />
From Expert Configuration<br />
the complete configuration is still showing, no changes from the supposed reset.<br />
<br />
From the web interface<br />
password now defaulted to "admin"<br />
everything is showing at factory settings<br />
<br />
From the phone<br />
attempted call results "there is no service"<br />
<br />Unknownnoreply@blogger.com0Toronto, ON, Canada43.653226 -79.383184315.342992163821151 -114.5394343 71.963459836178842 -44.226934299999996tag:blogger.com,1999:blog-3402630850730921975.post-63446919367191547262013-02-26T11:42:00.001-08:002021-02-09T13:04:02.294-08:00Linksys PAP2 | PAP2T for Freephoneline.ca<div id="primary">
<div id="content" role="main">
Before you start configuring the adapter, make sure you have the following settings for your SIP account:<br />
<article class="page type-page status-publish hentry"><div class="entry-content">
- SIP Server address (sometimes called SIP Proxy)<br />
- SIP User ID (the phone number, with 1 in front of the area code)<br />
- The password for the SIP account (a combination of letters and numbers)<br />
To obtain these settings, you must contact Freephoneline and ask for
your configuration file. There's a one time charge for this, currently
$50 CAD + tax. They will send you a Word document with the settings.<br />
<br />
Plug in the adapter (power, an Ethernet cable to your router and a phone into Line 1). Wait about
20 seconds then pick up the phone and dial <b>****110#.</b> The adapter will
read back (with voice) its IP address. Open a browser
and enter that IP address. The Info page of the adapter will show up.</div>
<div class="entry-content">
<h1 class="entry-title">
Setting up a Linksys PAP2 | PAP2T for Freephoneline.ca</h1>
<h1 class="entry-title">
<a name='more'></a></h1>
<img border="0" height="362" src="http://voipfan.net/ATAs/images/PAP2_home.png" width="855" /><br />
<br />
Click <b>Admin Login</b> on the right , then <b>Switch to advanced view</b> in
the middle, to get access to Advanced VoIP settings pages.<br />
<br />
<img border="0" height="362" src="http://voipfan.net/ATAs/images/PAP2_home_advanced.png" width="855" /><br />
<br />
Now you're ready to configure the VoIP settings. First, we'll adjust some of the SIP parameters, so click the <b>SIP</b> submenu.<br />
<br />
<img border="0" height="1260" src="http://voipfan.net/ATAs/images/PAP2_sip_freephoneline.png" width="854" /><br />
<br />
Change the following parameters:<br />
(in the middle of the page)<br />
- <b>RTP Packet Size</b>: 0.020<br />
(at the bottom of the page):<br />
- <b>STUN Server</b>: stunserver.org<br />
<br />
Now click the<b> Line 1</b> submenu<br />
<br />
<img border="0" height="1909" src="http://voipfan.net/ATAs/images/PAP2_line_freephoneline.png" width="853" /><br />
<br />
Enter the following settings:<br />
(at the top of the page)<br />
- <b>NAT Mapping Enable</b>: yes<br />
- <b>NAT Keep Alive Enable</b>: yes<br />
(about half way down on the page)<br />
- <b>Proxy</b>: voip.freephoneline.ca<br />
- <b>Register</b>: yes<br />
- <b>Register Expires</b>: 180<br />
- <b>Display Name</b>: enter your name here<br />
- <b>User ID</b>: your freephoneline phone number, with 1 in front<br />
- <b>Password</b>: the SIP password, from the configuration file received from freephonline<br />
- <b>Preferred Codec</b>: change this to G729a if you have issues with one-way audio<br />
- <b>Dial Plan</b>: use the following string (including parentheses)<br />
(*xx|911|1xxxxxxxxxx|[2-9]xxxxxxxxx|0xxxxx.)<br />
<br />
That is all, click Save Settings at the bottom to save all the changes. The
adapter will reboot and after 2-3 minutes you should get dial tone and should be
able to place and receive calls.
</div>
</article>
</div>
</div>
Unknownnoreply@blogger.com1tag:blogger.com,1999:blog-3402630850730921975.post-69152135189984432862013-02-13T07:58:00.002-08:002021-02-09T13:03:53.984-08:00Freephoneline.ca Setup Grandstream HandyTone HT-701Freephoneline.ca Configuration settings for the Grandstream HandyTone HT-701<br />
<br />
The instructions say 'Fongo' but will the same settings in PDF file work with freephoneline <br />
<blockquote class="tr_bq">
Yes they will. Freephoneline is simply a brand of Fongo </blockquote>
<a class="postlink" href="http://forum.fongo.com/download/file.php?id=900">ATA Config - Grandstream HT-701 (Fongo) FW 1.0.0.18 (Feb 2013).pdf</a><br />
<br />
<a class="postlink" href="http://forum.fongo.com/download/file.php?id=898">ATA Config - Grandstream HT-701 (Fongo) FW 1.0.4.3 (Feb 2013).pdf</a><br />
<br />
Fongo | Development Support Specialist.<br />
http://forum.fongo.com/viewtopic.php?f=15&t=536<br />
<br />
<br />Unknownnoreply@blogger.com0Toronto, ON, Canada43.653226 -79.383184315.342992163821151 -114.5394343 71.963459836178842 -44.226934299999996tag:blogger.com,1999:blog-3402630850730921975.post-80547851831970513202013-02-07T11:17:00.003-08:002021-02-09T13:03:34.221-08:00D-Link VTA Setting for Freephoneline<br />
This page discusses the VoIP settings for the D-Link VTA VoIP
router. For all the other settings, please consult the user manual available <a href="http://voipfan.net/ATAs/pdfs/VTA_UserGuide.pdf" target="_blank">here</a>. The guide applies to all the versions of the D-Link VTA (VTA-VR, VTA-CV, VTA-VD).<br />
<br />
Before you start configuring the adapter, make sure you
have the following settings for your SIP account:<br />
- SIP Server address (sometimes called SIP Proxy)<br />
- SIP User ID (in most cases this is the phone number)<br />
- The password for the SIP account<br />
To obtain these settings, you must contact
Freephoneline and ask for your configuration file. There's a one time charge for
this, currently $50 CAD + tax. They will send you a Word document with the
settings.<br />
<br />
Connect all the cables: power cord, an Ethernet cable to your router and a phone
into the green Phone 1 port. Wait about a minute for the router to power up,
then check your router's DHCP table to find the IP address assigned to the ATA (see some samples for different router brands <a href="http://voipfan.net/other/finddhcp.php">here</a>).
Put that IP address in a browser and login with username <b>Admin</b> and blank
password.<br />
<br />
<a name='more'></a><br /><br />
<img border="0" height="792" src="http://voipfan.net/ATAs/images/DLinkVTA_home.png" width="798" /><br />
<br />
Click <b>Admin</b> on the top menu, then <b>VoIP</b> on the left<br />
<br />
<img border="0" height="1065" src="http://voipfan.net/ATAs/images/DLinkVTA_SIP.png" width="800" /><br />
<br />
The first tab, <b>SIP</b> is very unlikely to need any changes. Please click on <b>Line 1</b><br />
<br />
<img border="0" height="903" src="http://voipfan.net/ATAs/images/DLinkVTA_line_freephoneline.png" width="800" /><br />
<br />
Enter the following settings:<br />
- <b>Line Enable</b>: yes<br />
- <b>Caller ID Number</b>: enter your freephoneline number, with "1" in front,
for example 14164771234<br />
- <b>Caller ID Name</b>: enter your name here<br />
- <b>User Name</b>: enter your freephoneline number, with 1 in front<br />
- <b>Password</b>: the SIP password for your freephoneline account<br />
- <b>Proxy Server</b>: enter voip.freephoneline.ca<br />
- <b>Proxy Port</b>: 5060<br />
- <b>Registrar Address</b>: voip.freephoneline.ca<br />
- <b>Registrar Port</b>: 5060 here<br />
- <b>Dial Plan</b>: the default value is OK<br />
- <b>Voice Coding Profile</b> (optional): the default value PCMU is for using
the g711 codec. If your VoIP provider advises you to use the g729 codec, enter
G729 in this field.<br />
Click Apply to save the settings.<br />
<br />
If you have issues dialing DTMF tones (such as dialing an extension after the
call is connected, or using calling cards), go to the <b>Common Coding</b>
option
and try different values for <b>Digit Relay Mode</b> (between 1 and 4)<br />
<br />
<img border="0" height="523" src="http://voipfan.net/ATAs/images/DLinkVTA_coding.png" width="801" /><br />
<br />
<b>To factory reset the adapter</b><br />
<br />
To restore the device to factory defaults, go to <b>Tech Support</b> (at the top) and
then <b>System</b> (on the left). Check the boxes that you desire (Provisioned
Parameters is for VoIP settings and Non-Provisioned Parameters is for router
settings<br />
<br />
<br />
<img border="0" height="562" src="http://voipfan.net/ATAs/images/DLinkVTA_reset.png" width="798" /><br />
<br />
<br />
<br />
<br />
<br />
<br />Unknownnoreply@blogger.com1Toronto, ON, Canada43.653226 -79.383184315.342992163821151 -114.5394343 71.963459836178842 -44.226934299999996tag:blogger.com,1999:blog-3402630850730921975.post-79549137250497608082013-01-17T08:01:00.001-08:002021-02-09T13:03:05.978-08:00D-LINK DIR-615 FREEPHONELINE SupportD-LINK DIR-615 has SIpAlg and (certainly older firmware) this makes SIP/Voip
very difficult (actually it is meant to help!). Try turning SipALg in
your router off, reboot both the router an the ATA. See if it works. If
not check the firmware version in the router and then read this<br />
<br />
<br />
<b>SPA2102 UNABLE IN/OUT CALL BEHIND THE D-LINK DIR-615, HARDWARE-C1; FRAME:3.11NA</b><br />
http://community.linksys.com/t5/VoIP-Adapters/SPA2102-UNABLE-IN-OUT-CALL-BEHIND-THE-D-LINK-DIR-615-HARDWARE-C1/td-p/278914<br />
<br />
<br />
From;<br />
<br />
http://forum.fongo.com/viewtopic.php?f=15&t=5366Unknownnoreply@blogger.com0tag:blogger.com,1999:blog-3402630850730921975.post-47902829211092775602012-05-28T09:06:00.001-07:002021-02-09T13:02:54.807-08:00FREE HOME PHONE LINE for the first 5000 Residents from Toronto or Montreal Not sure if this might be a hot deal but I came across this website offering a Free softphone service. <br />
<b><br />
"The first 5000 residents from either Toronto or Montreal receive FREE
calling to anywhere in Canada and your new phone number for life."</b><br />
<br />
Also not sure how long this offer has been around, but it's worth a try <img alt="" border="0" class="inlineimg" src="http://forums.redflagdeals.com/images/icons/new/smilies/thread-happy-16.png" title="Smile" /> <br />
<br />
<a href="http://www.freephoneline.ca/" rel="nofollow" target="_blank">www.freephoneline.ca</a><br />
<br />
These are some of the features included:<br />
<br />
<br />
<i>When you download and signup for our service you will receive free services and great choices!</i><br />
<br />
* <b>A FREE local phone number !</b>, if you live in the 416, 647 and most 905 area codes you can receive phone calls on your new <br />
FREEPHONELNE personal phone number. (Many new area codes will be added in 4th quarter ’07).<br />
* <b>Unlimited Free Local calling !</b><br />
* <b>Caller ID</b> - The incoming caller’s phone number is displayed on the freephoneline softphone.<br />
* <b>Unlimited FREE Canada wide long distance calling to these major cities :</b> Toronto, Vancouver, Montreal, Calgary, Edmonton, <br />
Winnipeg, London, Hamilton, Kitchener-Waterloo, Ottawa, Quebec City, and
Halifax, and, all the towns that are a local call to these cities!<br />
* <b>Incredibly low long distance rates</b> outside the coverage areas above. For rates go to <a href="http://www.1011295.com/" rel="nofollow" target="_blank">www.1011295.com</a> our sister company. <br />
With FREEPHONELINE you dial direct without dialing 1011295.<br />
* <b>Full Enhanced Voicemail Service</b> that is accessible from your phone no charge!<br />
* <b>Voicemail to Email</b> a copy of the voicemail is emailed to your private email account automatically , no charge !<br />
* <b>Follow me service</b> with enhanced unified messaging feature. When not at home, or if you use the PC phone and it is offline, <br />
you can set a number to call forward your local number.. You can also set the number of rings you would like the call forwarded <br />
for [example 3 rings to my cell phone] then if you are unavailable or you choose not to answer the call, freephoneline.ca <br />
voicemail will play your recorded greeting taking a message. <br />
This means you no longer need voicemail on your cell phone or other home phone lines, freephoneline.ca can be <br />
your central messaging service.
Unknownnoreply@blogger.com0Toronto, ON, Canada43.653226 -79.383184315.342992163821151 -114.5394343 71.963459836178842 -44.226934299999996tag:blogger.com,1999:blog-3402630850730921975.post-86653177806605763282011-09-11T20:58:00.002-07:002021-02-09T13:02:47.592-08:00VoIP DMZ - SIP war-dialers - SOHO routers<div>
<div style="-webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; color: black; font-family: Tahoma; font-size: medium; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px;">
<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;">Regarding SIP war-dialers (and I am getting dinged every frickin' night because I had made the mistake of NATing ports inbound for a single day):</span><br />
<br style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;" />
<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;">(1) I think we can all agree that the use of a "DMZ" with these SOHO routers (e.g. Linksys or D-Link, and these are not real firewalls) should be limited to testing purposes only, such as to validate whether or not certain ports need to be open, or if they appear to be not open. It's akin to allowing "any port / any source" from the Internet. Best practice is to prohibit ALL inbound traffic from the Internet, except that which is expressly required.</span><br />
<br style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;" />
<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; display: inline !important; font-family: Verdana; font-size: 13px; text-align: left;">(2) A firewall will generally NOT help you if you are allowing and NATing inbound traffic to your ATA. Forwarding purpose-specific ports inbound, will eliminate most firewalls from usefulness, because even application and protocol-aware firewalls will only deep dive the packets for RFC compliance and specifically-defined attack signatures. If the war-dialer's script is making use of the SIP protocol, a SOHO-class firewall offers little protection. You can impose little tricks by getting your listening service to use non-standard TCP/UDP ports - but this bites you in the ***** if legitimate traffic is trying to communicate with you on standard registered "well-known" ports associated with the listening service. For example, if you change your public-facing web server's listening ports to non-standard ports (e.g. ports other than TCP 80 and TCP 443), your site will likely not get the traffic it expects because the users' browsers are configured to hit web sites using their standard ports. If you somehow communicate to you your users that your web server is listening on non-standard ports (e.g. TCP 81 or TCP 444 for SSL), your users would have to manually suffix his URI with the port number to manually override the use of the standard ports (e.g. </span><a href="http://example.com:81/" rel="nofollow" style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; color: #000066; font-family: Verdana; font-size: 13px; text-align: left;">http://example.com:81</a><span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; display: inline !important; font-family: Verdana; font-size: 13px; text-align: left;"> or </span><a href="https://example.com:444/" rel="nofollow" style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; color: #000066; font-family: Verdana; font-size: 13px; text-align: left;">https://example.com:444</a><span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;">). As far as I can tell, there is no realistic way of communicating something like this to all users and all providers with VoIP.</span><br />
<br />
<a name='more'></a><br style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;" /><br />
<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;">All this to say, if the remote SIP/truck provider/VoIP system "needs" to generate inbound traffic to your ATA (and I contest this), it will likely do so on standard ports - changing the ports (e.g. UDP 5060 and UDP 5061) may not help at all.</span><br />
<br style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;" />
<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;">My personal experience is this - I use an Obi202 and NO inbound ports are required to be open/ NAT'd to my ATA for the following VoIP providers:</span><br />
<br style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;" />
<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;">(1) <a href="http://freephoneline.ca/">freephoneline.ca</a></span><br />
<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;">(2) Google Voice</span><br />
<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; display: inline !important; font-family: Verdana; font-size: 13px; text-align: left;">(3) CallCentric </span><br />
<br style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;" />
<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;">All my services work fine, other than some detailed configs. that I need to figure out to suit my particular purposes. But in general, all inbound and outbound calls work fine with no need to open any inbound ports.</span><br />
<br style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;" />
<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;">From my packet captures, the ATA makes periodic outbound socket calls to the respective provider, and when an inbound call is perceived, the session is managed via the OUTBOUND socket initiated by the ATA. It almost seems to work like UPnP, but I can't yet confirm this.</span><br />
<br style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;" />
<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;">So, my (unsolicited) advice would be to NOT "port-forward" to your ATA, or you run the risk of being added to the database of some dammed script kiddy.</span><br />
<br style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;" />
<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;">By contrast, I see some users here get no joy until they DO port-forward. Obviously, some more analysis is required to get the "right setup for all"</span><br />
<br style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;" />
<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;">I think that, if we had the time and ability, if we could compile some sort of comparison matrix, starting with:</span><br />
<br style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;" />
<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;">(1) The make/model ATA we use</span><br />
<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;">(2) The VoIP provider we use</span><br />
<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;">(3) Our ISP (and I doubt we really need this one)</span><br />
<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;">(4) Things don't work until I port forward (yes/no)</span><br />
<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;">Also </span></div>
<div style="-webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; color: black; font-family: Tahoma; font-size: medium; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px;">
<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;">(5) - the make/model of router we use and the firmware version.</span><br />
<br style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;" />
<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;">... we'd be closer to solving the port-forwarding mystery (do we need it or not?) once and for all.</span></div>
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<div style="-webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; color: black; font-family: Tahoma; font-size: medium; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px;">
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<div style="-webkit-text-size-adjust: auto; -webkit-text-stroke-width: 0px; color: black; font-family: Tahoma; font-size: medium; font-style: normal; font-variant: normal; font-weight: normal; letter-spacing: normal; line-height: normal; orphans: 2; text-align: -webkit-auto; text-indent: 0px; text-transform: none; white-space: normal; widows: 2; word-spacing: 0px;">
<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;">(1) Some folks are using their ATA's in "router mode" rather having them simply participate on the network as a node on the internal subnet. An easy way to know who is doing what - if you have network cable plugged only into you LAN port (or only into your WAN port, in the case of the OBI hardware), then you are NOT in "router mode". If you have a network cable plugged into both the LAN and WAN ports, you are likely in "router mode").</span><br />
<br style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;" />
<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;">What this could mean is that some people are using the regular router (i.e. their Linksys or D-Link, etc) as their demarc to the ISP, and then using their ATA also as a router, creating yet another layer of NAT and routing.... this could throw a wrench into the works as well.</span><br />
<br style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;" />
<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;">(2) If we know that our incoming calls are ONLY coming from the IP addresses our VoIP service providers, rather than the IP address of any VoIP user out there, then we should to isolate the inbound traffic to the SP's external IP address/subnet, and create an ACL (firewall rule) to allow SIP inbound traffic ONLY from those source IP addresses. This could function to thwart the War Dialers.... and then we only complain when the SP's change their IP's. I will take that over getting the SIP spam!</span></div>
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<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;"><br /></span></div>
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<span style="-webkit-tap-highlight-color: rgba(26, 26, 26, 0.292969); -webkit-text-size-adjust: auto; background-color: #fafafa; font-family: Verdana; font-size: 13px; text-align: left;"><br /></span></div>
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Unknownnoreply@blogger.com0Toronto, ON, Canada43.653226 -79.383184315.342992163821151 -114.5394343 71.963459836178842 -44.226934299999996tag:blogger.com,1999:blog-3402630850730921975.post-21659888267655651132011-09-11T08:39:00.001-07:002021-02-09T13:06:43.676-08:00Google Voice OBi202 | OBi100 | OBi110<h4>
Use Google Voice™ on your regular phone and make calls to the USA & Canada for FREE !! *</h4>
Make calls to other countries with Google Voice's incredibly <a href="https://www.google.com/voice/rates" target="_blank">low international rates</a> using an OBi202, OBi100 or OBi110 and your broadband connection to the Internet. <br />
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<b> <a href="http://www.obihai.com/sprint.html"> <span style="color: red;"> Sprint™ Customers: </span>Check Out the Amazing Benefits You'll Get with an OBi</a></b><br />
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<a name='more'></a><br /></div>
<b>Absolutely Convenient </b>use of Google Voice on a phone connected to the OBi device — <b>No PC required.</b><br />
<b>Amazingly Simple</b> is the best way to
describe how to get going with Google Voice on your OBi. Just add your
Gmail credential to the OBi and start making calls!<br />
<b>What You Need: </b><br />
<b>1. The OBi202, OBi100 or OBi110 device</b> – <a href="http://www.obihai.com/how-to-get.html">Get it here. </a> <br />
<b>2. Gmail username & password</b> – If you do not have this already, <a href="http://www.google.com/gmail" target="_blank">get it here.</a> <br />
<b>3. A Telephone</b> – Make A Google Voice call using your everyday phone – no need to be at your computer.<br />
<b>Set-Up:</b><br />
1. Sign-in to your <a href="http://www.google.com/gmail" target="_blank">Gmail</a>
account, locate the Gmail “Call Phone” feature and make a call from
your PC to any phone number. If you have already done this before, you
may skip this.<br />
2. Connect the OBi to a broadband connection via your Internet router.<br />
3. Register with the <a href="http://www.obitalk.com/" target="_blank">OBiTALK portal</a>, then add your OBi device and use the configuration wizard to set-up the OBi with your Gmail username + password.<br />
4. This step requires you have previously set-up a Google Voice phone number at <a href="http://www.google.com/voice" target="_blank">www.google.com/voice</a> <br />
- Sign-in to <a href="http://www.google.com/voice" target="_blank">Google Voice</a>.
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- Select the "Settings" link in the upper right corner of the window and then choose "<a href="https://www.google.com/voice#phones" target="_blank">Voice Settings</a>."
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- On the "Phones" tab, check the box next to the "Forwards to: Google Chat."
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- The selection will be saved automatically and you may sign-out of Google Voice.
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5. Now, start making and receiving calls on your OBi with Google Voice!<br />
<b>Helpful Resources:</b>
<br />
<a href="http://www.obihai.com/tutorial1.html" target="_blank"><b>Step-by-Step Tutorial: How to Use the OBi with Google Voice</b></a>
<br />
<a href="http://www.obihai.com/googlevoiceFAQ.html" target="_blank"><b>Getting Started & Troubleshooting with Google Voice on the OBi</b></a>
<br />
<a href="http://www.obihai.com/GoogleVoiceUG.htm" target="_blank"><b>Google Voice Set-Up & Troubleshooting Guide</b> </a><a href="http://www.obihai.com/docs/GoogleVoiceOBiTALK.pdf" target="_blank"><img alt="OBiTALK Device Management" class="" src="http://www.obihai.com/images/pdf.png" height="16" style="opacity: 1;" width="16" /></a>
<br />
<b>Great Things You Can Do with Google Voice on the OBi:</b><br />
- Direct dial calls to the USA and Canada for free! **<br />
- Direct dial calls to other countries at incredibly low international rates!<br />
- Use an Analog Telephone Adaptor-type VoIP device
(OBi202, OBi100 or OBi110) with Google Voice integration in conjunction
with the cool features available with Google Voice – e.g. voicemail
transcription, on-line voicemail, contact management, international
calling, etc.<br />
<b>More Cool Stuff You Can Do with OBi & Google Voice:</b><br />
- Receive calls to your Google Voice number on the phone attached to the OBi device. <br />
- Use the OBiON app for iPhone, iPad, iPod touch and
Android devices for true VoIP calls over Wi-Fi, 3G or 4G using the OBi
device to bridge to Google Voice and call traditional numbers (without
using your cell minutes).<br />
- Receive calls to your Google Voice number then use
the OBi device to bridge to your iPhone, iPad, iPod touch and Android
devices using Wi-Fi, 3G or 4G (without using your cell minutes).<br />
- Easily set-up Google Voice trunks on your IP PBX
phone system – no need for system upgrades or complex configs. The OBi
acts as a SIP to Google Voice gateway.<br />
<img alt="" class="" src="http://www.obihai.com/images/TechnologyGV.png" height="272" style="opacity: 1;" width="450" /> <br />
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<span style="color: #666666;"> * Google previously <a href="http://googlevoiceblog.blogspot.com/2011/12/free-calling-within-us-and-canada.html" target="_blank">announced</a>
that calls inside the USA and Canada to the USA and Canada will be free
through 2012. Calls to the USA and Canada from other countries are 1
cent per minute.<br />
Google Voice cannot be used to place or receive emergency services calls. <a href="http://www.google.com/googlevoice/legal-notices.html" target="_blank"> Google Voice Legal Notice (link)</a><br />
iPhone, iPad and iPod touch are trademarks of Apple Computer, Inc.<br />
Android, Gmail and Google Voice are trademarks of Google, Inc.</span><br />
<br />
<span style="color: #666666;">thanks to </span><br />
<span style="color: #666666;">http://www.obihai.com/googlevoice</span><br />
<br />Unknownnoreply@blogger.com0Toronto, ON, Canada43.653226 -79.383184315.342992163821151 -114.5394343 71.963459836178842 -44.226934299999996