VoIP - Do It Yourself...

Dell Voice: DIY BYOD Using PAP2T VoIP ATA

2012-02-20T15:07:00.002-07:00

In this article, I will explain how I configured the Cisco Linksys PAP2T-NA as a DIY BYOD VoIP ATA (Analog Telephone Adapter) with the new Dell Voice Fongo SIP based VoIP service.

This article assumes you already own a Linksys PAP2T-NA (unlocked). If not, there are Canadian mail order sites where you can purchase one. Some examples are: Dell Canada, VoIPdepot.ca, NewEgg.ca, Dytronix.com, TigerDirect.ca, DirectCanada.com, NCIX.com.

Once configured and working, you will essentially have your very own free VoIP telephone service that provides free long distance phone calling across Canada. And, the Dell Voice Fongo service provides you with a free Canadian phone number to receive free incoming phone calls (no recurring monthly fees), without having to have a PC running 24-7.

As explained in my previous blog article, you will first need to purchase the Dell Voice "VoIP Unlock Key" through your Dell Voice or Fongo account. Once your have your unlock key "password", you are set to proceed with configuring your DIY BYOD VoIP device. In the following tutorial, I'm illustrating configuration examples using my Cisco-Linksys PAP2T-NA.

I previously explained how to configure the PAP2T-NA ATA in my earlier article titled:

How to Configure Cisco Linksys PAP2T VoIP ATA with CallCentric

Everything you needed to know about configuring the PAP2T-NA with a BYOD VoIP provider like CallCentric was outlined in that article. Please read the article carefully before proceeding with your configuration with the Dell Voice Fongo service.

To save time and space in the following tutorial, I will describe only the specific settings and changes you need to make that will allow the PAP2T to register and connect directly with the Dell Voice - Fongo VoIP proxy servers. The settings I'm using here are working well for me, so far. I'll update this post in the future if I find any difficulties over time. Bearing in mind, different users may have different experiences due to their specific local network topology and network hardware configurations.

NAT Mapping Enable = No
NAT Keep Alive Enable = No
SIP Port = 5060 (Should not conflict with any other SIP ports on your PAP2T, or other VoIP devices on your local network. Each VoIP line requires a different SIP port number. Typically, any port between 5060 - 5070 will work).

Proxy: sip.fongo.com
Outbound Proxy: sip.fongo.com
Register: Yes
Register Expires: 1800
Use DNS SRV: No
Proxy Fallback Intvl: 1800

Use Outbound Proxy: Yes
Use OB Proxy in Dialog: Yes
Make Call Without Reg: No
Ans Call Without Reg: No
DNS SRV Auto Prefix: No

Display Name: (The name you want to show along with your Caller-ID)
User ID: 1XXXXXXXXXX (11-digit Dell Voice User Name; your phone number )
Use Auth ID: No
Password: **********

Password - this is your "Unlock VoIP Key" password provided via email after you purchase the key. It is crucial you enter your password explicitly correct. It will be a combination of Upper and lower case characters that must be entered in the explicit correct case.

Note - This is not the same password you use to login to your Dell Voice online account. It is a specific password for logging into the VoIP Proxy Servers, only.

I actually had trouble interpreting the password characters I received in the email. Sometimes, lower case letters can be incorrectly interpreted. For example, a lower case L (l) looks very much like a capital i (I), or the numeral one (1). So, be careful here.

Note - You will purchase your unlock key from the Dell Voice Online Store:

https://www.dellvoice.ca/redirectToPurchase

(you are required to LOGIN to access this option)

Currently, this is a one-time $50 unlock fee that will give you unlimited access to the Dell Voice Fongo servers with your own BYOD VoIP devices.

Preferred Codec: G.711u
G.729a Enable: Yes
Use Preferred Codec Only: No
DTMF Tx Method: AVT (AVT is the equivalent of RFC2833 - most reliable DTMF method)

Dial Plan:

You can probably get away with using the PAP2T-NA default dial plan. However, I have found the following simplified dial plan works fine with Dell Voice, for me:

(911 | [2-9]xxxxxxxxx | 1[2-9]xxxxxxxxx |011xxxxxxxxxxxx.)

Save all settings by clicking the "Save Settings" button at bottom of the configuration window.

RTP Ports
By default, the PAP2T uses the RTP port range 16384 - 16482.
Dell Voice only mentions configuring RTP ports 13000-13001. But based on my assumptions that Dell Voice Fongo service is a standard Asterisk based SIP VoIP service, then RTP ports are typically standardized using the RTP port range between 10001-20000. Thus, the default PAP2T RTP port range seems to work just fine with Dell Voice Fongo servers, from my testing so far.

Local SIP Ports
By default, the PAP2T is set to use SIP ports 5060-5061 (for Line1 - Line2). The standard SIP ports range generally fall between 5060 - 5070. The Dell Voice Fongo service recommends using port 5060 for their servers (standard recommendation). However, in my case, I have numerous VoIP devices behind my NAT-Router. One lesson I learned the hard way, years ago, was that each VoIP device needs to have its own assigned SIP port assigned to it..... or things will be unpredictable and unreliable. In reality, you can generally use any SIP ports in the 5060 - 5070 range.... just make sure multiple VoIP devices aren't trying to use the same SIP ports at the same time. For example, I may be using Line-2 with another VoIP provider (which I do), and I also have my 3CXPhone softphone configured with other VoIP providers, as well as Fongo. Each line used needs to be configured with a different SIP port... if you want to avoid local network conflicts.

I set my NAT-Router to Port Forward or Port Trigger on the SIP port range 5060-5070.
I also set my NAT-Router to Port Forward or Port Trigger on the range from 16384 - 16482.

Whatever SIP and RTP range you set in your VoIP devices, you must also be port forwarded or trigger those ranges through your NAT-Router.

When finished making setting changes, be sure to click the Save Settings button to save your changes.

TIP:
Note that in my last screen capture above, in the RTP Paramaters section of the SIP settings page, that I have changed the RTP Packet Size setting from 0.030 to 0.020. I have recently learned through research that reducing the RTP packet size to 0.020 will often "improve" your VoIP performance. If you are happy with your current VoIP performance, don't bother changing this value. But, if you notice you are experiencing excessive latency and broken voice, then this may be just the change you need to improve the situation.

Testing The PAP2T With Dell Voice
Besides calling friends and family in the Dell Voice free calling zones across Canada, I have tested with a few other numbers that I regularly use for testing my VoIP devices. Especially, the Environment Canada Weather Phones. These numbers are a good way to test the responsiveness of DTMF tone commands.

Weather Lines:

Calgary Weather: 403-299-7878
Regina Weather: 306-780-5744
Winnipeg Weather: 204-983-2050
Montreal Weather: 514-283-4006

Other Test Numbers:

TellMe: 800-555-8355
Microsoft Bing 411: 800-246-4411
Google411 (not-in-service message): 800GOOG411 (800-466-4411)

Also:

Dell Voice - Voice Mail: *98
Dell Voice - Customer Service: 611

Good luck using your PAP2T-NA ATA with Dell Voice and Fongo!

It works very well for me and hopefully it will work as well for you, too.

Dell Voice DIY BYOD on the Cisco SPA504G IP-Phone

2012-02-17T13:42:00.000-07:00

In this article, I will explain how I configured the Cisco SPA504G IP-Phone as a DIY BYOD VoIP user agent (UA) with the new Dell Voice Fongo SIP based VoIP service.

This article assumes you already own a Cisco SPA504G (unlocked). If not, there are Canadian mail order sites where you can purchase one. Some examples are: VoIPdepot.ca, NewEgg.ca, Dytronix.com, TigerDirect.ca, DirectCanada.com, NCIX.com.

Once configured and working, you will essentially have your very own free VoIP phone service that provides free long distance phone calling across Canada. And, Dell Voice provides you with a free Canadian phone number to receive free incoming phone calls, without having to have a PC running 24-7.

As explained in my previous blog article, you will first need to purchase the Dell Voice "VoIP Unlock Key" through your Dell Voice or Fongo account. Once your have your unlock key "password", you are set to proceed with configuring your DIY BYOD VoIP device. In this tutorial article, I'm illustrating configuration examples using the Cisco SPA504G 4-Line IP-Phone.

I previously explained how to configure the Cisco SPA504G in my earlier article titled:

- Configuring Cisco SPA504G IP-Phone From Scratch -

Everything you needed to know about configuring the SPA504G with a BYOD VoIP provider like CallCentric was outlined in that article. Please read the article carefully before proceeding with your configuration with the Dell Voice Fongo service.

To save time and space here, I will describe only the specific changes you need to make that will allow the SPA504G to register and connect directly with the Dell Voice - Fongo VoIP proxy servers.

While the Dell Voice - Fongo support only provides very basic configuration information, the settings I'm using here are working well for me, so far. I'll update this post in the future if I find any difficulties over time. Bearing in mind, different users may have different experiences due to their specific local network topology and network hardware configurations.

Of course, the following screen captures illustrate how I have configured the SPA504G for myself. Certain settings will need to be adjusted to your account with Fongo and need to be set accordingly. As you can see, I have configured my IP-Phone on Ext 3 (Line-3). What line you use, is your choice of course.

NAT Mapping Enable = No
NAT Keep Alive Enable = No
SIP Port = 5064 (Should not conflict with any other SIP ports on your SPA504G, or other VoIP devices your local network. Each VoIP line requires a different SIP port number.

Proxy: sip.fongo.com
Outbound Proxy: sip.fongo.com
Use Outbound Proxy: Yes
Register: Yes
Register Expires: 1800
Use DNS SRV: No
Proxy Fallback Intvl: 1800
Dual Registration: No

Use OB Proxy in Dialog: Yes
Make Call Without Reg: No
Ans Call Without Reg: No
DNS SRV Auto Prefix: No

Display Name: (The name you want to show along with your Caller-ID)
User ID: 1XXXXXXXXXX (11-digit Dell Voice User Name)
Use Auth ID: No
Password: **********

Password - this is your "Unlock VoIP Key" password provided via email after you purchase the key. It is crucial you enter your password explicitly correct. It will be a combination of Upper and lower case characters that must be entered in the explicit correct case.

Note - This is not the same password you use to login to your Dell Voice online account. It is a specific password for logging into the VoIP Proxy Servers, only.

I actually had trouble interpreting the password characters I received in the email. Sometimes, lower case letters can be incorrectly interpreted. For example, a lower case L (l) looks very much like a capital i (I), or the numeral one (1). So, be careful here.

Note - You will purchase your unlock key from the Dell Voice Online Store:

https://www.dellvoice.ca/redirectToPurchase

(you are required to LOGIN to access this option)

Currently, this is a one-time $50 unlock fee that will give you unlimited access to the Dell Voice Fongo servers with your own BYOD VoIP devices.

Preferred Codec = G.711u
Second Preferred Codec = G.729a
Use Preferred Codec Only = No
DTMF Tx Method: AVT (AVT is the equivalent of RFC2833 - most reliable DTMF method)

Dial Plan:

You can probably get away with using the SPA504G default dial plan. However, I have found the following dial plan works fine with Dell Voice, for me:

(911 | [2-9]xxxxxxxxx | 1[2-9]xxxxxxxxx |011xxxxxxxxxxxx.)

Save all settings by clicking the "Submit All Changes" button at bottom of the configuration window.

RTP Ports
By default, the SPA504G uses the RTP port range 16384 - 16538.
Dell Voice only mentions configuring RTP ports 13000-13001. But based on my assumptions that Dell Voice Fongo service is a standard Asterisk based SIP VoIP service, then RTP ports are typically standardized using the RTP port range between 10001-20000. Thus, the default SPA504G RTP port range seems to work just fine with Dell Voice Fongo servers, from my testing so far.

Local SIP Ports
By default, the SPA504G is set to use SIP ports 5060-5063 (for Ext's 1-4). The standard SIP ports range generally fall between 5060 - 5070. The Dell Voice Fongo service recommends using port 5060 for their servers (standard recommendation). However, in my case, I have numerous VoIP devices behind my NAT-Router. One lesson I learned the hard way, years ago, was that each VoIP device needs to have its own assigned SIP port assigned to it..... or things will be unpredictable and unreliable. This is why, in my example screen capture above, I'm using port 5064 on Ext Line 3... to avoid conflicts with my other local network VoIP adapters. In reality, you can use any SIP ports in the 5060 - 5070 range.... just make sure multiple VoIP devices aren't trying to use the same SIP ports at the same time.

I set my NAT-Router to Port Forward or Port Trigger on the SIP port range 5060-5070.
I also set my NAT-Router to Port Forward or Port Trigger on the range from 16384 - 16538.

Whatever, SIP and RTP range you set in your VoIP devices, must also be port forwarded or trigger those ranges through your NAT-Router.

When finished making setting changes, be sure to click the Submit All Changes button to save your settings.

TIP:
Note that in my last screen capture above, in the RTP Paramaters section of the SIP settings page, that I have changed the RTP Packet Size setting from 0.030 to 0.020. I have recently learned through research that reducing the RTP packet size to 0.020 will of often "improve" your VoIP performance. If you are happy with your current VoIP performance, don't bother changing this value. But, if you notice you are experiencing excessive latency and broken voice, then this may be just the change you need to improve the situation.

Testing The SPA504G With Dell Voice
Besides calling your friends and family in the Dell Voice free calling zones across Canada, I have tested with a few other numbers that I regularly use for testing my VoIP devices. Especially, the Environment Canada Weather Phones. These numbers are a good way to test the responsiveness of DTMF tone commands.

Note: When entering phone numbers into the SPA504G, do not use any spaces or dashes between numbers.

Weather Lines:

Calgary Weather: 403-299-7878
Regina Weather: 306-780-5744
Winnipeg Weather: 204-983-2050
Montreal Weather: 514-283-4006

Other Test Numbers:

TellMe: 800-555-8355
Microsoft Bing 411: 800-246-4411
Google411 (not-in-service message): 800GOOG411 (800-466-4411)

Also:

Dell Voice - Voice Mail: *98
Dell Voice - Customer Service: 611

Good luck using the SPA504G IP-Phone with Dell Voice and Fongo!

In future articles, I hope to discuss configuring the SPA2102-NA and PAP2T VoIP ATA's with Dell Voice.

Using Dell Voice BYOD on the 3CXPhone Softphone

2012-02-15T13:44:00.000-07:00

In this article, I will explain how I configured the free 3CXPhone softphone as a DIY BYOD VoIP user agent (UA) with the new Dell Voice Fongo SIP based VoIP service. From my experience thus far, the 3CXPhone performs superior to the desktop app provided by the Dell Voice service.

As explained in my previous blog article, you will first need to purchase the Dell Voice "VoIP Unlock Key" through your Dell Voice or Fongo account. Once your have your unlock key "password", you are set to proceed with configuring your BYOD DIY VoIP device. In this tutorial article, I'm illustrating configuration with the 3CXPhone softphone, provided free by 3CX Corp.

Download and Install the 3CXPhone Softphone
I have already explianed how to download and install the 3CXPhone in my previous article titled:

- 3CXPhone 6.0 SoftPhone - Installation - Configuration -

Everything you need to know about downloading and installing the 3CXPhone is outlined in that article. Please read the article before proceeding here with your configuration.

To save time and space here, I will describe only the specific changes you need to make that will allow the 3CXPhone to register and connect directly with the Dell Voice - Fongo VoIP proxy servers.

Of course, the following screen captures illustrate how I have configured the 3CXPhone for myself. Certain settings will be specific to your account with Fongo. For example:

Account Name:
This can be whatever you want to call your Dell Voice account on the 3CXPhone.

Caller ID:
You can put whatever you want your calls to display to the receiving party in this box. Typically, this will be your name or Dell Voice phone number. If you leave this blank, just your Dell Voice phone number will show at the receiving end of the call.

SIP Credentials:
Enter your 11-digit Dell Voice Fongo phone number issued to you at the time you opened your account with Dell Voice. i.e 1xxxxxxxxxx.

Extension: 1xxxxxxxxxx
ID: 1xxxxxxxxxx

Password: this is your "Unlock VoIP Key" password provided via email after you purchase the key.
It is crucial you enter your password explicitly correct. It will be a combination of Upper and lower case characters that must be entered in the explicit correct case.

Note - This is not the same password you use to login to your Dell Voice online account. It is a specific password for logging into the VoIP Proxy Servers, only.

I actually had trouble interpreting the password characters I received in the email. Sometimes, lower case letters can be incorrectly interpreted. For example, a lower case L (l) looks very much like a capital i (I), or the numeral one (1). So, be careful here.

NOTE
You will purchase your unlock key from the Dell Voice Online Store:
https://www.dellvoice.ca/redirectToPurchase (you are required to LOGIN to access this option)
Currently, this is a one-time $50 unlock fee that will give you unlimited access to the Dell Voice - Fongo servers with your own BYOD VoIP devices.

My Location:

Select "I am out of th office - external IP"; and enter: sip.fongo.com

sip.fongo.com is the VoIP proxy you will be registering the 3CXPhone with.

Use Outbound Proxy Server:
Enable this option and specify sip.fongo.com as the outbound proxy server.

Now, select the Advanced Settings button at the bottom of the Account Settings window.

Under Account Advanced Settings window, ensure the following values are set:

PBX Voicemail: *98
Stun Server: stun.3CX.com (default)

Registration Time: (Can be between 3 - 60 minutes) 10 minutes should be fine.

Support RFC2833 DTMF: check to ENABLE (only enable the RFC2833 DTMF option)

Audio Codecs: by default, PCMU is at top of list - this is preferred (equivalent to G.711u)

Save all settings by clicking the OK buttons at bottom of configuration windows.

Now, navigate to the PREFERENCES window. Set your Sound Devices preferences.
I myself use a Plantronics USB headset.

General Network Settings
These settings are very important:

RTP Ports
By default, the 3CXPhone uses the values 40000 - 40049. I don't think this will work with Dell Voice, or most any other open BYOD VoIP service.

The standard SIP VoIP RTP port range is between: 10001 - 20000.
However, for myself, I tend to use a reduced RTP range between 16384 - 16538. The reason is, because I also use this range on my Cisco SPA504G, PAP2T and SPA2102 VoIP phone adapters (their default range). It is interesting to note that Dell Voice and Fongo suggest RTP port 13000 only. But, I suspect that their servers will also accept anything in the "standard" SIP RTP range between 10001-20000. Anyway, my settings illustrated above work for me with Dell Voice and Fongo service.

I also set my NAT-Router to Port Forward or Port Trigger on the range from 16384 - 16538.
Whatever, RTP range you set in your VoIP devices, must also be port forwarded or triggered through your nat-router.

Local SIP Port
By default, the 3CXPhone is set to "Any" SIP port. The standard SIP ports range generally fall between 5060 - 5070. The Dell Voice Fongo service recommends using port 5060 for their servers (standard recommendation). However, in my case, I have numerous VoIP devices behind my NAT-Router. One lesson I learned the hard way, years ago, was that each VoIP device needs to have its own assigned SIP port assigned to it..... or things will be unpredictable and unreliable. This is why, in my example screen capture, I'm using port 5067... to avoid conflicts with my other local network VoIP adapters.

I also set my NAT-Router to Port Forward or Port Trigger on the SIP port range 5060-5070.

When finished, click the OK button to save your settings.

At this time, the 3CXPhone should restart, and if configured with the correct login credentials and port ranges, your 3CXPhone softphone will successfully register with Dell Voice Fongo servers and display "On Hook" to indicate that it is ready to make and receive phone calls.

Testing The 3CXPhone
From my experience thus far, the 3CXPhone works very good with the Dell Voice - Fongo service. In fact, I would go so far as to say it performs superior to the Dell Voice Desktop App downloaded from the Dell Voice web site. And, this is why purchasing the VoIP Unlock Key may be a wise choice for those who can afford the one-time fee and have some technical ability to configure their own BYOD DIY VoIP devices.

Besides calling your friends and family in the Dell Voice free calling zones across Canada, I have tested with a few other numbers that I regularly use for testing my VoIP devices. Especially, the Environment Canada Weather Phones. These numbers are a good way to test the responsiveness of DTMF tone commands.

Note: When entering phone numbers into the 3CXPhone, do not use any spaces or dashes between numbers.

Weather Lines:

Calgary Weather: 403-299-7878
Regina Weather: 306-780-5744
Winnipeg Weather: 204-983-2050
Montreal Weather: 514-283-4006

Other Test Numbers:

TellMe: 800-555-8355
Microsoft Bing 411: 800-246-4411
Google411 (not-in-service message): 800GOOG411 (800-466-4411)

Also:

Dell Voice - Voice Mail: *98
Dell Voice - Customer Service: 611

Good luck using the 3CXPhone with Dell Voice and Fongo!

In future articles, I hope to discuss configuring the Cisco SPA504G IP-Phone, and the SPA2102-NA and PAP2T VoIP ATA's with Dell Voice. I use all these devices myself with various BYOD VoIP providers.

Dell Voice and Fongo as a BYOD DIY VoIP Service

2012-02-14T14:10:00.000-07:00

Dell Voice in partnership with Fongo Inc. Canada is now providing one of the first free Canada-wide long distance VoIP phone calling services. While their marketing strategy is focused on targeting mobile VoIP users on the Android, IPhone, and Windows laptop PC's, it's not widely obvious that the service is also available to the BYOD DIY do-it-yourself crowd. This is where my interest is.

I suppose we could classify Dell Voice and Fongo as a "Freemium" service. The basic widely marketed "Free" features is what gets the most attention and is what gets the service off the ground and builds the initial subscriber base. And, if you want to expand beyond the basic free features, you then can take advantage of the advanced features for a "Premium". Thus, the term freemium.

If you are interested in BYOD (bring your own device) and DIY (do it yourself) to access these features of the Dell Voice - Fongo service, then you will have to pay a one-time premium of $50 to obtain a personalized password that enables you to log into the Fongo VoIP servers with your own BYOD Softphone, ATA (analog telephone adapter), or IP-Phone. They call this feature the "VoIP Unlock Key".

I have already done some preliminary testing of the Dell Voice PC Desktop App and was not very impressed with the initial results. But, this app is software pre-configured and locked to only work with Dell Voice, which most pre-branded softphones do (only work with their supplied service). However, just because my initial tests of their service with the Desktop App were not encouraging, I had a gut feeling that if I were to be able to directly connect by own VoIP UA's (User Agents) to the Fongo VoIP proxy servers, I would get much better results..... which, I did!

Gambling on my gut feeling meant that I had to give Dell Voice $50 just to see if my VoIP devices worked better than theirs. (I don't easily part with $50, especially on unknown VoIP services). I guess I thought it was worth the gamble for the following reasons:

I already had a Dell Voice free local phone number to receive incoming calls.
Most incoming and outgoing calls across Canada will be free from my own VoIP devices. (Prepaid minutes are required to call outside the Canada free zones - a premium service.)
Their service includes voice-mail, call forwarding, and 911 emergency calling for free.
I use Google Voice and Skype, but they don't provide free phone numbers in Canada. Google Voice and Skype also don't allow SIP BYOD DIY devices to connect to their services.
Dell Voice Fongo service is industry standard SIP based VoIP compatible and should work with most any SIP (Session Initiation Protocol) enabled user-configurable unlocked VoIP device (as far as I know).

I have already completed some initial testing of Dell Voice BYOD service after purchasing the VoIP Unlock Key. My initial tests are very encouraging.

I just hope the $50 unlock key is truly only a "one-time-fee", and that they don't turn around at some point in the future and start requiring "subscriptions" on an annual or monthly basis. If they do.... I'm out of here... as I do have plenty of other BYOD options that I already use.

My next blog article will cover how to configure my favorite, and top-pick, user-configurable free VoIP softphone for PC desktops and laptops. It's the 3CXPhone Softphone by 3CX Corp. In a past blog article, I explained how to configure the 3CXPhone with CallCentric. Next, I will explain how to configure it to work with the Dell Voice Fongo service.

I'm also testing Dell Voice on my Cisco SPA504G desktop IP-Phone with good results. I hope to cover this in another upcoming article, too.

I also plan to test Dell Voice on my Cisco Linksys SPA2102-NA and the PAP2T-NA. I'll keep you informed in future posts.

Dell Voice PC Desktop App - First Impressions

2012-02-13T22:15:00.001-07:00

Dell Voice and Fongo Canada have teamed up to provide free Canadian nation-wide VoIP calling from PC's, Android, and IPhone devices (and even ATA's and IP-Phones).

This is a welcome VoIP initiative for Canada. After all, we all know that Google Voice also provides USA/Canada free calling to PSTN land-lines and mobile phones from your PC. However, while the Dell Voice service provides free calling within Canada, it also gives you a free Canadian phone number, and 911 emergency access, which Google Voice and Skype do not do.

As a VoIP enthusiast, I was eager to give the PC desktop app a try. The following is my first impressions of using the Dell Voice PC Desktop App.

Download, Install, and Register your Dell Voice Account

The first step was to download and install the the Dell Voice App for PC Desktops. As part of this process, I was prompted to create a new Dell Voice account. This required my email address and that I create a password (always use strong passwords). They did require my current telephone number as part of the account registration. They currently do not allow registrations from outside of Canada.

The first time I launched the Dell Voice desktop app, I was prompted to login using my registered email and password. I was also required to acknowledge the limitations of 911 when accessed via VoIP devices. This is a standard VoIP 911 limitation acknowledgement in Canada.

Once launched, it became quite apparent that the Dell Voice desktop app (also known as a softphone) is quite a large app as far as softphones go. It also becomes quickly apparent the reason for the large window size of the app is so that Dell can display Dell marketing promo's. I guess marketing right in your softphone app is part of the price paid for using the free phone calling services.

Making Test Calls

For testing my VoIP services and devices, I often call one the Environment Canada Weather Lines. This turned out to be a quick way for me to assess how DTMF tone control functionality is working through the VoIP softphone or ATA.

(Calgary wx - 403-299-7878, Regina wx - 306-780-5744, Winnipeg wx - 204-983-2050)

Right away, I was disappointed with the DTMF touch-tone control of the Environment Canada weather options voice prompts when using the Dell Voice softphone. While the DTMF tone control worked, I consistently encountered broken and distorted incoming voice audio for the next 3-4 seconds following "every" DTMF command. While the Dell Voice softphone app does allow you to choose between "In-Band (tone)" or "Info (SIP)" DTMF control (found via the "Settings" button), only "In-Band (tone) worked for me. But, like I said, I would always experience 3-4 seconds of audio distortion following each DTMF commands.

Generally, the incoming audio quality seemed satisfactory. However, I suspect it is using a low-bandwidth Codec like G.729a, or lower, as the audio was slightly frequency limited and didn't sound quite as bright as if it were using a wider band Codec like G.711u.

I did make a test call to a real person at Dell Voice Customer Support by calling their hotline access number: 611. After navigating through a few voice prompts and using the DTMF commands, I was put on hold for about 5 minutes until a CS agent finally answered. Unfortunately, the first 15 or 20 seconds of the conversation was all broken-up and distorted. But, then the connection got better after I repeatedly said "hello, hello, I'm having difficulty hearing you". Then, the connection got better and I completed my questions with the CS agent.

At this time, I really haven't done any other testing of the VoiceMail or Follow Me features.

Initial Test Conclusion

Just from my brief tests with the Dell Voice desktop app softphone, I'm not impressed. They definitely need to fix the DTMF tone control functionality. If you don't ever use DTMF tone controls on your phone, then it may not be an issue. But for me, it means the phone has bugs that need to be fixed. I'm sure over time, they will fix it, perhaps through some future software updates. The audio quality, from my limited tests seems to be hit and miss. It will take many more calls to get a truer sense of the call quality using the desktop app.

(If you do experience call problems, make sure your router is port forwarding (or triggering) on SIP port 5060 UDP and RTP port 13000).

There is Hope for Dell Voice and Fongo

From my perspective, I don't actually care that the desktop app is a bit immature and needs to grow-up a bit. Why? Because, I found out that I don't really need Dell's desktop app to use Dell Voice and Fongo as a SIP-based VoIP home phone service.

Dell Voice and Fongo as a BYOD VoIP Provider

While Dell Voice desktop app was a bit of a bust for me, it's irrelevant because Dell Voice and Fongo service allows BYOD people like me to use my own user configurable SIP-based softphone, user-configurable ATA, or IP-Phones to connect with their service. What's the catch? You will have to pay a one-time "VoIP Unlock Fee" of $50. Well, I paid the price..... and I'll let you in on the results in upcoming post articles... (it'll be worth the read).

Dell Launches Free VoIP Service in Canada - Dell Voice

2012-02-13T14:33:00.004-07:00

Last week, Dell Canada in conjunction with Fongo, launched an innovative new SIP-based VoIP phone service that offers unlimited free incoming and outgoing Canada-wide phone calling.

The service branded as Dell Voice, is really a Dell branded marketing initiative in partnership with Fongo, a Canadian based VoIP service provider, focused at providing free mobile apps that enable free VoIP-based phone calling on the Android, I-Phone, and Windows desktop operating systems.

The Dell Voice service, powered by Fongo, provides Canadians with a free Canadian phone number that allows users to receive unlimited free incoming calls via their Android, IPhone, or PC Desktop softphone app. Other free features include Caller-ID, Voice-Mail, Call Forwarding, and 911 emergency calls.

The free calling features are free in the sense that VoIP calls travel over the Internet. Thus, if you have a prepaid data plan or Internet service through your ISP, then the calls you place ride over your existing data plan without using your phone calling minutes with your voice phone provider. This is really how all VoIP services work regardless of whether your Internet access is wired over landline or wireless via your wireless router, hotspot, or your mobile phone data carrier. The Dell Voice and Fongo portion of the service IS completely free, for their basic services.

The free calling feature covers most of the larger cities throughout Canada, but not all cities are covered. The service does provide prepaid world-wide calling outside the free-zones for a reasonable cost of 2-cents/min. However, the fine print does reveal a 4-cent "connect fee" for all calls connected outside the free zones. This is the equivalent to what Skype does. Skype charges connect fees too on their pay-per-call plan.

The feature that interests me the most is their service that allows you to connect directly to the Fongo VoIP servers using any unlocked BYOD softphone, VoIP ATA, or IP-Phone. This feature requires a one-time purchase of their "VoIP Unlock Key". This privilege does come with a one-time network "unlock fee" of $50. However, with this BYOD enabled feature, you can truly replace your home phone service with a Dell Voice and Fongo solution at an extremely low cost.

In my next blog post article, I will discuss my initial calling tests and review of using their free downloadable desktop softphone application with Dell Voice. I plan another blog article on my experience connecting a BYOD softphone, ATA, or IP-Phone to the Dell Voice service as a 24-7 connected phone service.

Differences Between Cisco SPA2102 and Linksys PAP2T

2012-01-27T13:28:00.000-07:00

I myself have been testing and using the Linksys-Cisco PAP2T-NA and SPA2102-NA ATA's (Analog Telephone Adapter) for a few years now.

When I bought my first PAP2T, I knew nothing about user configurable ATA's (unlocked and open), but I was determined to learn... the hard way, through Internet research and trial and error.

When I first skimmed through the web-based user interface of these ATA's, I became a bit dizzy from the maze of configuration options. Quite overwhelming and intimidating to the novice, which I was. Even now, over two years later, I'm still intimidated by some of the advanced options available in these units.

The good news is that the average user doesn't really need to understand anything about most of the options provided. What you do need to know though is how to navigate to the options that are essential. And, I will be providing future posts on just that - how to configure the PAP2T-NA and SPA2102-NA with the basic minimum configuration options possible. And, to keep it simple, I will be specifically illustrating how to configure them with my current preferred VoIP carrier, CallCentric.

What's the difference between PAP2T and SPA2102?
The primary differences of most importance is that the PAP2T does not have a built-in NAT-Router or QOS. So the PAP2T is really just a configurable Telephone Adapter. So, from this perspective, it is simpler to configure than the SPA2102 (but, only marginally simpler). However, because it does not have its own built-in NAT-Router, you will generally be using it behind your existing network router. And, this very fact does have some drawbacks of its own (VoIP firewall traversal issues). So, now you may have to fiddle with your local network NAT-Router gateway firewall to port-forward, or port-trigger, SIP and RTP ports required for VoIP services to work behind a firewall.

Whereas, the SPA2102 does have its own built-in NAT-Router and QOS. This very fact is a huge bonus for the SPA2102 because it automatically takes care of opening SIP and RTP ports and default QOS settings required for your ATA to function without any fiddling with the firewall and QOS options.

How do you decide which ATA is best for you?
If you already have a NAT-Router integral to your Internet Modem gateway, then you may have no choice but to pick the PAP2T-NA.

However, if your Modem does not have a built-in NAT-Router, then I highly recommend you pick the SPA2102-NA.

Even if you already have a separate 4-port NAT-Router, after the Modem, you don't need to stop using it.... just use it "after" the SPA2102.

For the SPA2102 to function at its best, you definitely want it connected "directly after" your Modem. This way you will benefit from the QOS packet management the SPA2102 provides. And, you won't have to worry about firewalls blocking the essential SIP and RTP VoIP ports required by the ATA.... the SPA2102 will open the necessary SIP and RTP ports it needs for your VoIP calls.

The SPA2102 does provide one Ethernet LAN port. So, you can connect your PC directly to this LAN port. Or, you could connect another 4-port router, or switch, to this port if you have multiple devices on your LAN side. This is the way I recommend most people use the SPA2102.

However, for myself, I have settled for the following configuration that works quite satisfactory for me:

I connect a 4-port Switch directly to my Modem LAN gateway.
I connect my SPA2102 to port-1 of the switch.
I connect my Linksys WRT54G to port-2 of the switch.
All other devices on my LAN are connected via the WRT54G LAN ports.

In the above configuration, the switch is acting as a kind of Internet access splitter between my Modem, SPA2102, and WRT54G. However, this configuration requires that each device connected to the switch be assigned a unique WAN IP address (public IP).

With my high-speed Internet service, I am allowed to be assigned two dynamic WAN IP addresses. Therefore, I get one IP address assigned to my SPA2102 and the other IP address assigned to the WRT54G. Most high-speed Internet services allow up to two dynamic IP addresses with high-speed service... but, you may have to contact their Customer Support to enable the 2nd IP address, which may not be enabled, by default (as it was in my case).

Then, I have my desktop PC connected to one of the WRT54G LAN ports and my wife's laptop through its Wireless-G link. I then set the WRT54G to limit its "uplink speed" to be 150 Kbps less than my ISP allowed uplink speed. This ensures that I leave enough uplink bandwidth left over for the SPA2102 VoIP packets to traverse the Modem uplink without being bottlenecked by simultaneous data from the WRT54G data through the Modem.

I do not have any other devices connected to my SPA2102 LAN port... but, I could if I wanted to. The QOS settings in the SPA2102 throttles "upload" LAN traffic through its LAN port to 128 Kbps (by default). This is to ensure that there is always enough upload bandwidth available for the VoIP packet traffic. You can adjust its "uplink speed" to suit your permissible upload bandwidth provided by your ISP. Actually, I do sometimes connect my PC to the SPA2102 LAN port when I need to view or change any of its web-based configuration settings.

So, if you are only going to be using your ATA behind an existing NAT-Router, the PAP2T-NA is probably your best choice for that scenario.

If you want much more flexibility, go for the SPA2102-NA. It is also small and compact enough to use as a travel NAT-Router for your laptop.... with the added bonus of being able to make cheap VoIP calls from your travel NAT-Router while on the road (assuming you have high-speed Internet access). However, this would also require you to have a small standard telephone to plug into the ATA to make those calls.

In previous articles, I explain and illustrate the basics of configuring the Linksys-Cisco PAP2T-NA and SPA2102-NA with CallCentric VoIP services.

How To Configure Cisco Linksys SPA2102 VoIP ATA With CallCentric

2012-01-27T13:22:00.000-07:00

The focus of this article is to illustrate how to configure the Cisco Linksys SPA2102-NA with a BYOD (bring your own device) VoIP carrier, such as CallCentric.

To keep things as simple as possible, I will start from the factory default settings and only change configuration settings that are most important to get started connecting to CallCentric on Line-1, which is my primary VoIP carrier I have been using for years.

NOTE: These instructions only work on UNLOCKED versions of SPA2102-NA.

If you purchased your ATA from an authorized Cisco-Linksys reseller, it will likely be an unlocked -NA version.

My examples are based on SPA2102-NA Firmware Version = 5.2.10 and Hardware Version 1.3.5(a).
My SPA2102-NA is directly connected to my ISP high-speed Modem. (Recommended)

I will be following these basic steps:

Reset SPA2102 to Factory Defaults using its IVR interface
Login to the SPA2102 web-based configuration interface
Configure primary/secondary DNS servers and NTP Time Servers
Set your local Time Zone clock offset
Configure Line-1: VoIP Proxy, User ID, Password, Codec, Dial-Plan

I recommend using the SPA2102 directly on the output of your Internet Modem. For best performance and reliability, there should not be any NAT-Routers before the SPA2102. This also means that your Modem should not have a NAT-Router built into it. The SPA2102 has its own built-in NAT-Router and Firewall. So, the rest of your network is protected when connected to the LAN port of the SPA2102. If you have multiple LAN devices that need to be connected, you can connect your router, or Ethernet Switch, "after" the SPA2102 through its LAN port.

Reset to Factory Defaults

The reason I am beginning with Reset to Factory Defaults is to ensure that we are on the same page with all the same default settings. If it works for me, I presume it will work for you too.

To reset to factory defaults requires that I access the SPA2102 IVR system (Interactive Voice Response) using a regular telephone plugged into the Line-1 phone jack (RJ-11 jack). Then, power up the adapter. It will only take about 15 seconds to power-up to the "ready" state.

To activate the SPA2102 IVR, lift the phone handset and press the Star key four (4) times (* * * *).
After the IVR voice prompt, enter the Factory Reset code: 73738, followed by the # key (73738#).
The IVR will prompt you to "press 1 to confirm".
As soon as you press "1", the unit will perform the factory reset.

Now that we are at a known default starting point, we can proceed to configure only those settings that are required, or otherwise recommended.

Connect and Login to the SPA2102 Web Based Configuration Interface

Connect your PC to the SPA2102 using its LAN-side gateway:

Connect your PC Ethernet LAN cable directly to the yellow Ethernet LAN port of the SPA2102.
Connect to the SPA2102 web interface by entering its LAN gateway IP address into your Web Browser using the default address of 192.168.0.1
By default, you will be viewing the SPA2102 "Status" page, in "Basic" view. By default, there is no password required to connect and login to Basic view.

The screen captures I'm currently posting here are based on firmware version 5.2.10.
In previous articles I posted the procedures for upgrading firmware in the SPA2102-NA and PAP2T-NA VoIP adapters. It is advisable to always use the latest firmware version available.

To begin configuration of the SPA2102, you need to switch to "Admin" and "Advanced" view.
By default, the SPA2102 does not require an admin password.

Now, navigate to the WAN Setup tab.

Configure Primary/Secondary DNS Servers and NTP Time Servers

DNS servers lookup the IP address of an Internet resource based on domain name to IP address cross referencing. Typically, we configure a VoIP proxy based on its domain name (i.e. callcentric.com ). DNS servers will redirect your Internet traffic to the appropriate IP address of the VoIP proxy. The same goes for the NTP (Network Time Protocol) time servers. Your ATA uses NTP time servers for tagging and maintaining the correct device time. NTP time servers are listed by domain names. The DNS servers will direct the NTP requests to the appropriate NTP time server IP addresses.

By default, the SPA2102-NA should retrieve the DNS server info from your Internet Modem DHCP server. However, I prefer to ensure that the DNS servers are hard coded, as well as automatically retrieved via DHCP, to ensure reliability, redundancy, and backup.

In my examples, I have configured my SPA2102-NA Primary and Secondary DNS servers to point to the new Google DNS servers at 8.8.8.8 and 8.8.4.4. I also use NTP time servers at time-nw.nist.gov and time.nist.gov. If you have different server preferences, feel free to use them instead of the ones I use. As well, you can set the DNS Server Order = "Manual, DHCP" to use DHCP DNS servers as backup to your manually set primary/secondary servers.

For now, the default QOS settings should be suitable to begin with.

Click the Submit All Changes button to SAVE your custom settings.

Next, navigate to the Voice and Regional tab page. Scroll to the bottom of the page.
Set your local Time Zone offset, depending on where you live.

Click the Submit All Changes button to SAVE your custom settings.

Configure Line-1: VoIP Proxy, User ID, Password, Codec, and Dial-Plan, etc...

Navigate to the Voice and Line-1 tab page.

The Line-1 page is where you will be configuring VoIP service with your primary line (Line-1) and carrier.

While you can configure any SIP based VoIP carrier here, that allows BYOD (bring your own device), I will be using CallCentric to illustrate the CallCentric "required settings". If you are configuring with another VoIP carrier, it is essential that you consult their support pages for their specific settings required by that specific service.

The following settings must be set as specified by CallCentric:

NAT Mapping Enable: No
Nat Keep Alive Enable: No
Proxy: callcentric.com
Use Outbound Proxy: Yes
Outbound Proxy: callcentric.com
Use OB Proxy in Dialog: Yes
Register: Yes
Make Call Without Reg: No
Ans Call Without Reg: No
Register Expires: 3600
Use DNS SRV: Yes
DNS SRV Auto Prefix: Yes
User ID: 1777XXXXXXX (use your CallCentric UserID)
Auth ID: 1777XXXXXXX (use same as UserID)
Use Auth ID: Yes
Password: ********** (use as set in your CallCentric user account - Preferences - Phone Password; SIP Password) (By default, it's the same as your web login password)

Remember to save all new settings when finished making changes.

Click the Submit All Changes button to SAVE your custom settings.

At this point, you have set all the basic requirements to connect to a VoIP service provider.

Following, are some additional suggestions and reminders:

Codecs

While the SPA2102 defaults to use G.711u Codec, CallCentric recommends using G.729a to save bandwidth. If you want the best voice fidelity, I would stick with G.711u. But, if you do have QOS or data bottlenecking problems, it may be prudent to switch to G.729a to reduce bandwidth requirements of your VoIP calls.

Dial Plans

While the SPA2102 default dial plan will work, you may want to replace the default with the dial plan recommended by CallCentric:

(*xx.|*xxx|*75xx|[3469]11|0|00|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.|**275*x.)

First, completely clear the default plan, then copy and paste this plan in its place.
Save your changes.

There are three reasons to use the CallCentric enhanced dial plan:

It accepts *xxx numbers, such as accessing your CallCentric Voice Mail using *123 command
It accepts CallCentric Speed Dial numbers such as *75xx (*7500 - *7599). Your CallCentric user control panel allows you to pre-program 100 predefined speed-dial numbers accessible from your VoIP ATA (i.e. SPA2102).
It accepts **275*x. number dialing to allow network peering calls through the Sip Broker network.

Disable Line-2

If you will only be using Line-1 on the SPA2102, I recommend disabling Line-2. You can always re-enable it in the future if you connect to another account or another VoIP service (like I do).

Disable Provisioning

As the SPA2102-NA is an "Open", "Unlocked", and "User Configurable" VoIP ATA, there is no reason to leave the default Provisioning service enabled. I recommend setting the "Provision Enable" = NO.

Remember to save any changes.

Before exiting the web administration pages, I recommend that you go back and review all settings and changes just to confirm that all values were entered correctly and saved.

I hope you enjoy using the SPA2102-NA with the CallCentric VoIP service as much as I have.

Happy VoIPing...

How To Configure Cisco Linksys PAP2T VoIP ATA With CallCentric

2012-01-27T13:14:00.000-07:00

The focus of this article is to illustrate how to configure the Cisco Linksys PAP2T-NA with a BYOD (bring your own device) VoIP carrier, such as CallCentric.

NOTE: These instructions only work on UNLOCKED versions of PAP2T-NA. If you purchased your ATA from an authorized Cisco-Linksys reseller, it will likely be an unlocked -NA version.
My examples are based on PAP2T-NA Firmware Version = 5.1.6 and Hardware Version 0.1.5

I'm also using the PAP2T behind my Linksys WRT54G NAT-Router.

I will be following these basic steps:

Reset PAP2T to Factory Defaults using its IVR interface
Login to the PAP2T web-based configuration interface
Configure Line-1: VoIP Proxy, User ID, Password, etc.

Reset to Factory Defaults

The reason I am beginning with Reset to Factory Defaults is to ensure that we are on the same page with all the same default settings. If it works for me, I presume it will work for you too.

To reset to factory defaults requires that I access the PAP2T IVR system (Interactive Voice Response) using a regular telephone plugged into the Line-1 phone jack (RJ-11 jack). Then, power up the adapter. It will only take about 15 seconds to power-up to ready state.

To activate the PAP2T IVR, lift the phone handset and press the Star key four (4) times (* * * *).
After the IVR voice prompt, enter the Factory Reset code: 73738, followed by the # key (73738#).
The IVR will prompt you to "press 1 to confirm".
As soon as you press "1", the unit will perform the factory reset.

Connect and Login to the PAP2T Web Based Configuration Interface

Connect to the PAP2T using its DHCP assigned IP address, assigned by your router. In my case, I'm using the WRT-54G as my LAN NAT-Router. My PC is connected to port-1 and the PAP2T is connected to port-2 of the router.

Typically, you will determine the IP address assigned to the PAP2T by using its IVR system:

To activate the PAP2T IVR, lift the connected phone handset and press the Star key four (4) times (* * * *).
After the IVR voice prompt, enter the Check IP Address code: 110, followed by the # key (110#).
Write down or remember this IP address.
Hang-up to end IVR.
Connect to the PAP2T by entering its assigned IP address, using your Web Browser
(In my case, I'm using Google Chrome browser)
By default, you will be viewing the PAP2T "Info" page, in "Basic" view. By default, there is no password required to connect and login.

Configure Line-1: VoIP Proxy, User ID, Password, etc...

To begin configuration of the PAP2T, you need to switch to "Admin" and "Advanced" view.

By default, the PAP2T does not require an admin password.

Click on Line 1 to go to Line 1 configuration page:

The Line-1 page is where you will be configuring VoIP service for your primary line and carrier. While you can configure any SIP based VoIP carrier here, that allows BYOD (bring your own device), I will be using CallCentric to illustrate the CallCentric "required settings". If you are configuring with another VoIP carrier, it is essential that you consult their support pages for their specific settings required by that specific service.

The following settings must be set as required by CallCentric:

NAT Mapping Enable: No
Nat Keep Alive Enable: No
Proxy: callcentric.com
Use Outbound Proxy: Yes
Outbound Proxy: callcentric.com
Use OB Proxy in Dialog: Yes
Register: Yes
Make Call Without Reg: No
Ans Call Without Reg: No
Register Expires: 3600
Use DNS SRV: Yes
DNS SRV Auto Prefix: Yes
User ID: 1777XXXXXXX (use your CallCentric assigned UserID)
Auth ID: 1777XXXXXXX (use same as UserID)
Use Auth ID: Yes
Password: ********** (use PW as set in your CallCentric user account Preferences page)
(by default, SIP PW is the same as your CallCentric web account login, but can be changed in Preferences)

Remember to "Save Settings" when finished making changes.

At this point, you have set all the basic requirements to connect to a VoIP service provider.

Following, are some additional suggestions and reminders:

Codecs

While the PAP2T defaults to use G.711u Codec, CallCentric recommends using G.729a to save bandwidth. If you want the best voice fidelity, I would stick with G.711u. But, if you do have QOS or data bottlenecking problems, it may be prudent to switch to G.729a to reduce bandwidth requirements of your VoIP calls.

Dial Plans

While the PAP2T default dial plan will work, you may want to replace the default with the dial plan recommended by CallCentric:

(*xx.|*xxx|*75xx|[3469]11|0|00|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.|**275*x.)

Just copy and paste this plan over the default plan.

There are three reasons to use the CallCentric enhanced dial plan:

It accepts *xxx numbers, such as accessing your CallCentric Voice Mail using *123 command
It accepts CallCentric Speed Dial numbers such as *75xx (*7500 - *7599). Your CallCentric user control panel allows you to pre-program 100 predefined speed-dial numbers accessible from your VoIP ATA (i.e. PAP2T).
It accepts **275*x number dialing to allow network peering calls through the Sip Broker network.

Disable Line-2

If you will only be using Line-1 on the PAP2T, I recommend disabling Line-2. You can always re-enable it in the future if you connect to another account or another VoIP service (like I do).

Disable Provisioning

As the PAP2T-NA is an "Open", "Unlocked", and "User Configurable" VoIP ATA, there is no reason to leave the default Provisioning service enabled. I recommend setting the "Provision Enable" = NO.

Remember to save any changes.

Allow VoIP to Pass Through Firewalls

Remember, the PAP2T is designed to function behind a router and firewall. For VoIP behind NAT-Routers to function "reliably", you need to Port Forward or Port Trigger the required VoIP SIP and RTP ports that carry the VoIP traffic.

Note:

If you use "port forwarding" (vs. port triggering) through your router, this usually requires forwarding to a fixed IP address. Thus, you would need to fix the IP address of the PAP2T to a static value, and then set port forwarding address in the router to match the static IP address of the PAP2T.

Initially, I recommend that you try using Port Triggering of SIP and RTP ports through your router and leave your PAP2T in DHCP mode. If triggering isn't working for you, then resort to Port Forwarding SIP/RTP ports to a fixed IP address on the PAP2T.

Enjoy your VoIP calls from your PAP2T-NA configurable ATA.

How to Update Firmware on the Linksys WRT54G Wireless Router

2012-01-27T12:55:00.000-07:00

(Article originally posted in August 2010)

In this article, I will cover how to update firmware on the WRT54G NAT-Router.

I'm using the WRT54G for this example because I use it on my home network, and the PC that I use VoIP Softphones with is also behind this router. In my previous articles on VoIP, Port Forwarding, and Port Triggering, I also used the WRT54G as my example router to illustrate configuration settings.

In the following examples I will be upgrading from firmware version 8.00.4 to 8.00.7

Login to the WRT54G using IP address 192.168.1.1
The default User Name is empty (no entry)
The default Password is "admin"
I myself have previously changed the UserName and PW to my own personal settings, so those are the values I used to login.

OR
After logging in, you will land on the "Setup" page:
Navigate to the "Administration" page.
Navigate to "Firmware Upgrade" page.
Before you proceed beyond this point, you need to visit the Cisco Linksys support website and download the latest version available for you WRT54G hardware version at:

http://homesupport.cisco.com/en-us/wireless/lbc/WRT54G
Select your hardware version from the drop-down list. Once selected, you will see the latest download version in the lower left corner of the Downloads section.
Click on the Download link to commence download of the update file.
Select the location to store the update file, and then proceed the download Save.
Once you have completed downloading the latest firmware file, go back to Administration and Firmware Upgrade page in the WRT54G.
Click on the "Choose File" button on the Upgrade Firmware page and navigate to the previously downloaded firmware file and select it.
This file name will now show up next to the "Choose File" button:
Click on the "Upgrade" button.
You will see the "progress" bar incrementing...

Be patient and do not interrupt the process until prompted that it has completed and rebooted:

When you see this screen "Upgrade is successful, Rebooting....", all is well !

Not only has Linksys upgraded the firmware (in my case versions from V8.00.4 to V8.00.7), but they have given the interface an obvious face lift in color, and a little more emphasis on the name Cisco:

Good luck with your WRT54G firmware upgrade.

(As of August 10, 2010 the latest firmware version for HW Version 8 was FW version V8.00.8)

Port Triggering with the Linksys WRT54G

2012-01-27T12:12:00.000-07:00

As I use the Linksys WRT54G wireless NAT-Router as my LAN gateway, I will use it as a reference guide for examples of Port Triggering as related to VoIP.

I login to the WRT54G using IP address 192.168.1.1 from the LAN side. By default I believe the username and password are both "admin". This may or may not be the same for your Router.

For greater security, I do not use the default password. I have reset it to use a password with greater security. Check your user documentation for the specific login information that came with your router.

Once logged in, navigate to the "Port Triggering" page. Typically, it will be found under the "Applications and Gaming" tab section.

Basically, what you want to do here is decide what application you wish to trigger ports open with. Assign the Application box the name of the application you will be triggering with.

If your application only requires a single port, enter that port as the Start Port and End Port in the "Triggered Range" section.

If the Triggered Range is also the Forwarded Range, enter that same port range in the Start Port and End Port under the "Forwarded Range" section. And, be sure to "Enable" the triggering by checking the Enable box.

For example, if you use Google Talk, you will only need to open port 5222 .
I give the Application section name "GTalk". Then, I assign the Trigger Range as Start Port = 5222 and End Port = 5222. Then, I set the triggered Forwarded Range Start Port = 5222 and End Port = 5222 and then check Enable.

Skype picks a random port to use. So if you are a Skype user, you will need to navigate to Skypes Tools Options Advanced Connection window to find out what random port it has selected for you. I myself pick my own personally specified port, which Skype allows you to do. Whatever port is selected, you will need to configure your NAT-Router to "trigger" on this port.

I use two separate settings for standard SIP based VoIP.
One for triggering SIP ports 5060-5065, and another trigger for the SIP RTP ports 16384 - 16482. In the first case, I use SIP triggers to open SIP ports. In the second case, I use the SIP triggers to open the RTP ports.

How to Port Forward with the Linksys WRT54G

2012-01-27T12:08:00.001-07:00

Port Forwarding will permanently open specified ports on your NAT-Router, where as Port Triggering only opens ports on an as needed basis, for a period of time.

I will use my WRT54G to illustrate examples of how I set Port Forwarding for my VoIP services.

First, use your web browser on its LAN side to access the router using its default LAN Gateway address: 192.168.1.1

By default, the User Name is blank, or empty. Password by default is "admin". Otherwise, use whatever values you may have changed it to for your personal preference.

Once you have logged in to the WRT54G, navigate to the "Applications and Gaming" page and then click on the "Port Range Forward" tab.

One primary difference I note for Port Forwarding (vs. Port Triggering) is that forwarding requires fixed IP addresses on the LAN side to be known. If your device your are forwarding to has a DHCP assigned dynamic address, the forwarding may not be reliable. Thus, it may be necessary to give your LAN devices a fixed IP address assignment for reliable port forwarding.

Another difference with Port Forwarding is that you also have to decide to forward UPD, TCP, or Both. It is considered safest to only use UDP protocol forwarding, unless you know that your services explicitly require TCP protocol. Skype is one service that seems to want to use TCP when possible, so you may want to set Skype's forwarding to use Both.

Skype picks a random port to use. So if you are a Skype user, you will need to navigate to Skypes Tools Options Advanced Connection window to find out what random port it has selected for you. I myself pick my own personally specified port, which Skype allows you to do. Whatever port is selected, you will need to configure your NAT-Router to Forward this port.

As far as I know, Google Talk currently only requires port 5222 UDP for voice calls.

SIP VoIP services like CallCentric can be triggered with SIP Ports 5060 - 5065 and RTP Ports 10,000 - 20,000. I tend to use RTP ports 16384 - 16482 only because that is the range my Linksys PAP2T and SPA2102 ATA's default to for RTP ports.

Port Forwarding can make all the difference when it comes to getting your VoIP services to work on your home network.

Skype Super Nodes - Are You One?

2011-12-29T17:23:00.000-07:00

I'd like to keep this article as simple as possible without getting into a long draw-out discussion on Skype Super Nodes, Relay Hosts, or the technical methods of detecting if your bandwidth is being eaten-up by your PC acting as a Skype Super Node, or Relay Host.

I'm certainly not an expert on this subject. But, I have done some homework and have come up with some basic understanding of who potentially becomes a Super Node or Relay Host for Skype.

More importantly, I'd like to convey just the two basic steps you can take to prevent your PC from becoming one of Skype's Super Node networks.

What Is A Skype Super Node?
"A Super Node is a Skype client that has a public IP address and enough spare CPU
cycles, RAM, and bandwidth to take on additional duties for the Skype P2P network. Super
Nodes hold a portion (up to several hundred users) of the distributed Skype directory."

Super Nodes essentially act as look-up directories allowing Skype users (Skype clients) to find and connect to other Skype users (other Skype clients).

What is a Skype Relay Host?
"A relay host is a Skype client that has a public IP address and enough spare CPU
cycles, RAM, and bandwidth to relay Skype content for other Skype users who are behind
restrictive firewalls or are otherwise unable to communicate with each other directly."

The Relay Host acts as an intermediary for relaying audio, video, etc. content between Skype clients who are unable to make direct connections between themselves. This is usually because the two Skype clients who want to connect, but can't due to being behind a NAT/Firewall network Router.

What's The Bottom Line Here?
The bottom line is that if your PC becomes a Super Node or Relay Host, your PC could potentially be using significantly more Internet bandwidth that you should be using (or, want to be using).

Two Basic Steps To Help Prevent Becoming A Super Node, or Relay Host:

Make sure your PC is behind a hardware NAT/Router/Firewall.
Disable Skype's use of Ports 80 and 443.

1) - Make Sure You Are Behind a hardware enabled NAT/Router/Firewall
Now days, everyone should be behind a hardware enabled NAT/Router/Firewall, for security reasons, if for no other reason. This step alone may be all it takes to ensure you don't become a Super Node.

2) - Disable Skype's use of Ports 80 and 443
This step is easily done by navigating to Skype's Options -->Advanced-->Connections settings.
In this dialog box, "uncheck" the option "Use Port 80 and 443 as alternate incoming connections". This step is crucial, if you are not behind a NAT/Router/Firewall (but, you really should be). If you are behind a NAT/Router/Firewall, this step may be optional. However, I always do this myself, just for the added peace of mind. (But, it does mean doing some extra work, like Port-Triggering in your router to pass on (trigger Open) Skype's randomly set port.)

2a) - While here, also note the port value randomly set by Skype to use instead of ports 80 and 443.

2b) - Be sure to Port-Forward, or Port Trigger on the port value used as the primary port for Skype to listen for incoming connections. To do this, you will need to know how to administer the settings in your NAT/Router/Firewall connected between your PC and the Internet modem.

In Conclusion
If you are able to perform the steps listed above, it appears to be unlikely that your PC will become one of Skype's notorious Super Nodes or Relay Hosts.

P.S.
It is also possible to restrict Skype's ability to act as a Super Node by use of Windows Policies using registry keys and Skype's config files. Skype's Network Admin Guide reveals what the settings are.

References:

Skype: A Practical Security Analysis - by Bert Hayes, SANS Institute
Guide For Network Administrators - by Skype Biz
Skype Relay Calls: Measurements and Experiments - by Dept. of Computer Science, Columbia University

3CXPhone 6.0 SoftPhone - Installation - Configuration

2011-12-29T17:07:00.001-07:00

When it comes to BYOD user-configurable softphones, the 3CXPhone VoIP Phone has been my top pick for the past few years, now.

3CX Corp. has recently upgraded their VoIP Phone from version 5 to version 6.0. This article outlines the process and procedures I used for downloading, installing, and configuring version 6.0 on my Windows 7 SP1 desktop.

Besides the fact that the 3CXPhone just works and doesn't crash like some of the other well known brand of softphones, the 3CXPhone VoIP Phone is FREE.

In this tutorial, I will focus on how to install and configure the 3CXPhone with CallCentric VoIP service, which I myself use.

I'll begin by downloading the Windows version directly from the 3CX website. It's a 12.3 MB installer file. The specific build version I will be using for this tutorial is Version: 6.0.20943.0 . The file name of the installer file is: 3CXPhone6.msi (download directly from this link).

Once downloaded, start and run the installer file to begin the installation:

Once the Installer runs, it will start the 3CXPhone Setup Wizard.

Click Next to start the Wizard. Accept the License Agreement. Click Next:

Accept, or move, the Installation Folder file path default.
Click Next Click Install

Installing... Click Finish

That's it. The whole process actually takes less than 1 minute.
Click FINISH and Launch the 3CXPhone.

Security Warning - Some users may have security software that may pop-up a warning the first time the 3CXPhone is run. This happens so you can validate that this is a new software application that needs your approval before running and allowing to connect to the Internet.

The first time the 3CXPhone is launched, it may present you with a "No SIP Profile found" message, or "No Active Accounts Configured" and "Not Set" status on the phone.

In my case, I upgraded from an earlier version. Even though I uninstalled the previous version, the un-installation did not remove the previous account configuration settings. This is very useful for people like me who just want to uninstall to upgrade without losing previous account registration settings and phone books.

However, if this is the very first time 3CXPhone has been installed on your PC, you will likely see the following pop-up message:

At this point, you would typically click on the "Create Profile" button.
This will Open the "Accounts - Manage SIP Accounts" window.

However, for the purpose of this tutorial, I'm going to suggest that you click on the "Cancel" button, for now. (We will configure the CallCentric account later.)

At this point, because there are no VoIP accounts configured, as yet, you will likely see the "No active accounts configured" message on the 3CXPhone screen:

Click on the "Options" button. This will open the MAIN MENU window on the phone.

Before setting up VoIP Accounts, I'm going to suggest some "essential" Preferences settings first. Without making the following suggested Preferences settings, your VoIP calls likely won't work properly.

Click on "Preferences" icon. - Default - Preferences -

Be sure to select your preferred Sound Devices for the Microphone input and Speaker output, as well as Ringing device.

It is always recommended to use a headset with built-in microphone, for best results.
In my case, I selected my Plantronics USB DSP headset:

If you will only be using pure VoIP like me (for now anyway), then you can disable Video functionality.

General Network Settings:

RTP Ports:
This is a very important setting to consider.

The default setting of 40000 - 40019 is apparently for the 3CX IP-PBX system. However, most WAN side SIP VoIP services (to my knowledge) use RTP ports in the 10000-20000 range. This is the case for CallCentric. Keeping in mind, whatever you set here may also need to be Port Forwarded or Port Triggered to pass through your NAT-Router Firewall. (I myself use Port Triggering on my routers.)

I'm using the RTP range of 16384 - 16538 which is what my NAT-Router is already set to trigger on, and is also the range my Cisco SPA504G IP-Phone and SPA2102-NA ATA use.

By default, the 3CXPhone is set to use Any Local Port. The screen capture above shows port 5070 (greyed out) as a default local port, I'm presuming this is the SIP ports setting. SIP ports commonly use the range between 5060-5070. You can leave the default setting set to Any, or try setting it to 5060. (Be sure your router is forwarding or triggering on this port, or range value.)

When finished, click the OK button to save your settings.

Once you have completed configuring the "Preferences" page, it is now time to configure the CallCentric Proxy and Log-in credentials into the "Accounts" page - Manage SIP Accounts. This is where you create your 1st SIP VoIP provider Profile.

When you click on "New", you will see the following default Account Settings window:

In this tutorial, I will be configuring the Account Settings Profile for CallCentric.

Account Settings Summary:
Set the Account Name as "CallCentric" (or, whatever you like).
Set the Caller-ID value as your CallCentric account number (i.e. 1777xxxxxxx).
In the Credentials section, make the following settings:

Extension: 1777xxxxxxx (set to your 11-digit CallCentric account number)
ID: 1777xxxxxxx (also, set to your 11-digit CallCentric account number)
Password: Set to match your CallCentric account login PW (or SIP login, if set different)

Set My Location to be: "I am out of the office - external IP"
Enable: "Use Outbound Proxy Server"

Enter the name of the Outbound Proxy Server as: callcentric.com

Click on the "Advanced Settings" button (lower left-hand corner of Account Settings window).
The following are the Default settings of the Advanced settings dialog box:

The only change (or addition) here is to set your Voice Mail access code. For CallCentric, the Voice Mail access code is "*123".

As far as codec settings go, PCMU is the equivalent of G.711u, which will give you the best audio quality. If you have bandwidth issues, you could move GSM up to the top of the list as the first choice codec. This would lower your bandwidth usage (and lower your voice quality).

Using RFC2833 DTMF Support is typically the best option for DTMF handling. However, you can change to "Support INBAND DTMF" if using PCMU (G.711u), but not while using GSM codec. As well, I do not recommend enabling more than one DTMF handling option at a time.

The 3CXPhone utilizes a STUN server to help with NAT-Router traversal. It defaults to stun.3cx.com, which should be fine. (Keep in mind STUN utilizes port 3478. You may want to Port Forward or Trigger your Firewall to include port 3478.... if you have connection/registration problems.

Click OK button to accept all changes to the Account Advanced Settings window.
This returns you back to the Manage SIP Accounts window.

Click OK button at bottom of Accounts - Manage SIP Accounts page.

At this point, the 3CXPhone will attempt to register with the CallCentric.com proxy servers.
If all settings were made correct, and if your firewall(s) aren't blocking any ports, the phone should successfully register and appear as follows:

If you see the "On Hook" status message just below the 3CX logo, you are all set and ready to make test calls!!!

Making Test Calls
Ok, now it's time for the big test..... making a call.
You can dial by clicking on the phones keypads with your mouse, or just type the numbers directly from your PC keyboard.

CallCentric does have a test number to call for call testing. It's the TellMe Information Services.
TellMe can be accessed by dialing the following CallCentric in-network number: 17771234567
This number is free to call, as all in-network calls are.

The numbers dialed are transmitted by clicking on the "Green" phone icon at the bottom center of the phone, or just by pressing the Enter key on your PC keyboard. (The phone icon is normally grey... until you begin entering numbers, and then the icon changes to Green. Once in a call, the icon changes to Red. Clicking the Red icon will disconnect the call.)

You can also test the phone by calling your CallCentric Voice Mail. A Voice Mail box is free with CallCentric accounts, but you do need to order it just the same as ordering any other add-on service (like a phone number), but for Voice Mail, there is no charge to order, or to use.

Once your Voice Mail is activated (through your online account dashboard when you order it), just Dial: *123 from the3CXPhone and you will be greeted by the Voice Mail IVR system. You can also Quick-Dial your voice mail by clicking on theVoice Mail icon located in the lower-right corner of the phone (envelope icon).

Of course, to make test calls to landlines and cell phones, you will need to have prepaid calling credits available in your account.

Free Calls to iNums:
CallCentric supports free calling to global iNum numbers (+883 country code).
I will provide a couple numbers you can dial for testing:
+883510000000093 (announces back your Caller-ID, provided by iNum.net)
+883510000000091 (Echo Test provided by iNum.net)
Note: the "+" can be replaced with 00 or 011 when dialing from North America.

To END you calls, just click on the RED phone button, during a call.

Free SIP Calling:
CallCentric supports free SIP URI dialing between VoIP peering networks.
To dial SIP numbers from the 3CXPhone, enter the full URI address of the SIP number using your PC keyboard. This latest version of 3CXPhone does not seem to require preceeding the number with "sip:".

I have provided a few SIP URI numbers for testing (all free calls):

200901@login.zipdx.com (VUC-VoIP Users Conference - Listen to live conference - Fridays 12 PM EST
883510012394882@inum.net (VUC - VoIP Users Conference via iNum)
301@proxy.ideasip.com (Echo Test)
8355@proxy.ideasip.com (TellMe Information)
8463@proxy.ideasip.com (Time Announcement EST)
music@iptel.org (music)

Notes:
You can easily test the above SIP URI numbers by selecting any number with your PC mouse, then copy with Right-Click-Copy, or pressing the PC keys CTL-C. Then, just Paste the selection onto the 3CXPhone screen by Left-Clicking on the 3CXPhone screen icon and then Paste the number by pressing your PC keys: CTL-V. Then click the green phone icon to dial the pasted number.

Troubleshooting
If you have set everything up correctly, this phone should work very well. The usual problems encountered are typically:

Incorrectly entering your CallCentric registration information (1777xxxxxxx 11-digit number, password, server, and proxy server)
Firewall issues. Port blocking by your Nat-Router and/or your PC Firewall (or anti-virus program) are most common issues.
Knowing how to Port Forward or Port Trigger your Nat-Router to pass all required SIP and RTP ports.
And, of course getting your Audio devices to be recognized and set for the phone.

I hope the 3CXPhone works as well for you with CallCentric as it does for me.

Google Extends Free Phone Calls Through 2012

2011-12-13T12:15:00.000-07:00

This morning, Google's Official Gmail Blog has just announced that they are extending their free domestic calling within the US and Canada throughout 2012.

What this means, is that if you are a Gmail user, all you need to do is install Google's Voice and Video Chat plug-in from Google's web site. Once installed in your web browser, you are set to make free voice calls to any US and Canadian telephone number, be it land-line or mobile numbers.

Google has been providing free calling through Gmail since mid 2010. Then, last year announced they would extend free calling through 2011. To be frank, I didn't expect they would extend free phone calling through 2012, as well. I figured Google would fall they way of Skype.

Skype used to provide free calls to land-lines too in US and Canada. But, once they built up their subscriber base, they inevitably started changing for all calls to PSTN phones and mobiles. I expect eventually Google will follow suit... just not in 2012.

But, for now, enjoy the free phone calls... I am!

Shaw WiFi Trials Begin in Western Canada

2011-12-06T15:27:00.000-07:00

Today, Shaw Communications announced that they have begun trials of their new anticipated WiFi service in Western Canada in the cities of Calgary, Edmonton, and Vancouver. They plan to aggressively expand the number of access points throughout 2012. Access will be free to all Shaw Cable Internet customers during the trial period.

If you are an IPad or Android device user in any of the trial cities, this may be a good opportunity to try out CallCentric's Android App or the IPad App.

CallCentric Launches New Android App

2011-12-01T21:35:00.000-07:00

Today, CallCentric announced the release of their new Android App.

"The Callcentric Android app allows you to use your Android phone to place calls worldwide from your Callcentric account, and save significant money over cellular carrier calling rates. This is particularly useful to customers who want to save money on international calls, or want the caller ID of their Callcentric account to show up on outbound calls they place instead of their Android phone number."

CallCentric has done an excellent job explaining the benefits of using their Android App to save money on long distance calls from your Android based phone.

Their web site gives full details of how to download and install the app on your Android device.
I'm looking forward to testing this on my Android tablet...

Configuring Cisco SPA504G IP-Phone From Scratch

2011-11-28T23:00:00.000-07:00

The Cisco SPA504G IP-Phone has a vast array of configuration settings that are user (admin) configurable. Fortunately for most of us, most of the Default settings will work right out of the box, as is. The point of this tutorial is to guide you through the most basic and essential setting changes required to get started making calls. And, hopefully with the least amount of confusion.

This tutorial is geared to the new and unfamiliar user who has just acquired their first Cisco SPA504G IP-Phone. While this tutorial is not comprehensive, it does get you connected to CallCentric and ready to make calls.

For the purposes of this tutorial, I will be configuring the SPA504G from scratch, on Line-1 (EXT1), and with one of my primary VoIP providers, CallCentric.

Connect and Login to the SPA504G Web Based Configuration Interface

Connect your PC to the SPA504G using its LAN side Ethernet port marked PC.
Login to the SPA504G web interface by entering its LAN gateway IP address into your Web Browser using its DHCP assigned address (In my case, it's 192.168.0.3).
By default, you will be landing on and viewing the SPA504G "Info" page, in "Basic" view. By default, there are no User or Admin passwords required to connect and login to the SPA504G.

The screen captures I'm currently posting here are based on firmware version 7.4.9c.

First Things First

Before configuring the SPA504G with a BYOD VoIP service provider, such as CallCentric, there are a few points that should be addressed first. While these settings, or procedures, are "not essential", they will certainly be helpful.

Confirm Firmware version
Perform Factory Reset, (if viable)
Turn Off Provisioning.
Disable Ext2, 3, and 4
Disable Line Keys 2, 3, and 4

If your current firmware version is older than the latest release, then I recommend updating the firmware to the latest version available from Cisco's web site. In a previous article, I posted the procedure for upgrading firmware on the SPA504G.
If viable, I highly recommend performing a Factory Reset. I have done this myself while in the process of writing this article. This way we are assured that if the configuration settings I make here work for me, I expect they will also work for you, too.
To turn off Provisioning, click on the Provisioning Tab. By default, Provision is Enabled.
Disable by changing Provision Enable to NO.
Save your changes by clicking on "Submit All Changes" button at bottom of page.
Disable Extensions 2, 3, and 4.
Set Line Extension 2, 3, and 4 Line Enable to No.
Disable Line Keys 2, 3, and 4.
In the Phone tab window, set Line keys 2, 3, and 4 to Disabled.
While you are here, you can set the Line Key 1 Short Name = "$CallCentric-1". This is the name you will see on the LCD display next to Line-1 (Ext1) key. If not set as above, the default $User will cause the line key label on the LCD display to show as your User account login name for that line (in this case it would be the CallCentric user account name = 1777xxxxxx)

Essential Settings before configuring your SIP VoIP provider.

We still have a couple system settings to check, and change, if necessary.

Click on the System tab.

Scroll down to the bottom of the window to the section called: Optional Network Configuration

Configure Primary/Secondary DNS Servers and NTP Time Servers

By default, the SPA504G should retrieve the DNS server info from your Internet Modem DHCP server. However, I prefer to ensure that the DNS servers are manually set, as well as automatically retrieved via DHCP, to ensure reliability, redundancy, and backup.

In my examples, I have configured my SPA504G Primary and Secondary DNS servers to point to the new Google DNS servers at 8.8.8.8 and 8.8.4.4.

I also use NTP time servers at time-a.nist.gov and time.nist.gov. If you have different server preferences, feel free to use them instead of the ones I use.

As well, you can set the DNS Server Order = "Manual, DHCP" to use DHCP DNS servers as backup to your manually set primary/secondary servers.

The NTP Time Servers, if correctly set, will update the SPA504G internal clock once every hour.

Click the Submit All Changes button to SAVE your custom settings.

The SPA504G will reset and reinitialize itself. This takes at least 35 seconds to complete.

If you now observe the time display on the SPA504G LCD display, you will see that the Date and Time are now set to the correct values (or should be).

However, it is worth noting that your Time Zone may likely be different from the default value set in the phone. To correct for your Time Zone, proceed as follows:

Click on the Regional tab. Scroll down to the bottom of Regional page.

Note that the Default Time Zone Offset is: GMT -8:00.

Because I live in Alberta, Canada, I'm in Mountain Time, which is GMT -7:00, Standard Time.

Obviously, what you set here depends on your area time zone. (Save any changes made.)

Now, if your phone's clock is off by one or two hours, it's probably because you need to compensate for daylight savings time. In which case, you need to implement the Daylight Savings Time Rule.

For North America, our daylight savings time begins 2nd Sunday in March, at 2 AM, and ends the 1st Sunday in November, at 2 AM. (As of 2007)

The new Daylight Savings Time Rule for North America, recommended by the Cisco SPA500 Admin guide (bottom of page 198), is as follows:

"start=3/8/7/02:0:0;end=11/1/7/02:0:0;save=1"

Click the Submit All Changes button to SAVE your custom settings.

The SPA504G will reset and reinitialize itself.

If your Daylight Savings Time Rule has been set correctly, your phone's clock should now be bang on. (Although, you may not know for sure until daylight savings time kicks in again next spring.)

Now, we are ready to get on to the business of actually configuring a VoIP provider on EXT1, so you can hopefully start making and testing VoIP calls.

Configuring Line-1 (EXT1) with a VoIP Provider

For the purposes of this tutorial, I will be using CallCentric as the VoIP service to be configured into EXT1 of the SPA504G. I have been using CallCentric myself for years now and know that they are an excellent and reliable BYOD VoIP service.

Click on the Ext1 tab .

The Ext1 page is where we will be configuring VoIP service with CallCentric.

While you can configure any SIP based VoIP carrier here, that allows BYOD (bring your own device), I will be using CallCentric to illustrate the CallCentric "required settings". If you are configuring with another VoIP carrier, it is essential that you consult their support pages for their specific settings required by that specific service. There are some subtle configuration differences between VSP's that can make the difference between success and failure.

The following settings must be set as specified by CallCentric:

NAT Mapping Enable: No (default is No)
Nat Keep Alive Enable: No (default is No)
Proxy: callcentric.com
Outbound Proxy: callcentric.com
Use Outbound Proxy: Yes (default is No - change to Yes)
Use OB Proxy in Dialog: Yes (default is Yes)
Register: Yes (default is Yes)
Make Call Without Reg: No (default is No)
Register Expires: 3600 (default is 3600)
Ans Call Without Reg: No (default is No)
Use DNS SRV: Yes (default is No - change to Yes)
DNS SRV Auto Prefix: Yes (default is No - change to Yes)
User ID: 1777XXXXXXX (use your CallCentric UserID)
Auth ID: 1777XXXXXXX (use same as UserID)
Use Auth ID: Yes (default is No - change to Yes)
Password: ********** (use as set in your CallCentric user account - Preferences - Phone Password; SIP Password) (By default, it's the same as your web login password)
(Be sure to always use Strong Passwords)

Click the Submit All Changes button to SAVE your custom settings.

At this point, you have set all the basic requirements to connect to CallCentric and make SIP calls.

Following, are some additional suggestions and reminders:

Codecs

While the SPA504G defaults to use G.711u Codec, CallCentric recommends using G.729a to save bandwidth. If you want the best voice fidelity, I would stick with G.711u. But, if you do have QOS or data bottlenecking problems, it may be prudent to switch to G.729a to reduce bandwidth requirements of your VoIP calls.

Dial Plans

While the SPA504G default dial plan will generally work, you may want to replace the default with the dial plan recommended by CallCentric:

(*xx.|*xxx|*75xx|[3469]11|0|00|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.|**275*x.)

First, completely clear the default plan, then copy and paste this plan in its place.

Save your changes.

There are three reasons to use the CallCentric enhanced dial plan:

It accepts *xxx numbers, such as accessing your CallCentric Voice Mail using *123 command.
It accepts CallCentric Speed Dial numbers such as *75xx (*7500 - *7599). Your CallCentric user control panel allows you to pre-program 100 predefined speed-dial numbers accessible from your VoIP phone (i.e. SPA504G).
It accepts **275*x. number dialing to allow network peering calls through the Sip Broker network.

Before exiting the web administration pages, I recommend that you go back and review all settings and changes just to confirm that all values were entered correctly and saved.

There is no doubt that the SPA504G IP-Phone has a vast array of customizable settings. This tutorial was by no means comprehensive. But, hopefully it was enough to get you started and familiar with the basics of configuring VoIP BYOD services, like CallCentric.

For more detailed information on the Cisco SPA504G, you can refer to the Cisco SPA500 Admin Guide on the Cisco website.

Good luck with configuring and using your Cisco SPA504G VoIP phone.

Performing a Factory Reset on the Cisco SPA504G IP-Phone

2011-11-26T22:52:00.000-07:00

This tutorial is a precursor to an upcoming tutorial on how to configure the Cisco SPA504G IP-Phone for VoIP service with a BYOD VoIP provider, such as CallCentric.

I'm recommending that you start from scratch following a Factory Reset. While, performing a factory reset isn't essential prior to configuring your VoIP provider, I'm suggesting it just to be sure we are starting from the same reference point. This is to ensure that no settings have been previously set that could cause problems when configuring the phone for service with CallCentric.

The following procedure will guide you step-by-step in performing the factory reset. As well, I will follow the reset with a procedure on how to check the IP-Address assigned to the SPA504G by your network DHCP server/router.

Press Setup button:

Navigate to Menu Item #14 - Factory Reset, and press Select:

Select OK at the Reset Warning...

Commencing Reset and Initialization... (lots of blinking lights...)

The SPA504G is now reset to its factory default values.

Following a Factory Reset, any passwords previously set have now been removed and will need to be re-set, if desired, later. (Not to mention that all previously set user configuration data is also lost)

Now, we have a clean slate from which to begin re-configuring the SPA504G IP-Phone.

This will be a good time to check your Firmware Software Version, and compare it against the latest available firmware on the Cisco support web site. I explain how to check and update firmware in a previous blog article.

Now, you should also check and confirm the IP address assigned to your SPA504G by your DHCP server/router. You will need to know the IP address in order to log into the SPA504G web interface via your LAN side PC's web browser for the configuration procedures.

Checking The SPA504G Assigned IP Address

Before proceeding here, you will want to be sure the SPA504G is connected to your network LAN with an Ethernet cable. Following a factory reset, the SPA504G is set, by default, to receive its IP address via DHCP server via its Ethernet port, marked SW (switch).

Navigate to the Setup Menu item #9 - Network, and press select.

Take note of the IP address shown here for reference.

Note:

If you will be Port Forwarding SIP and RTP ports to your SPA504G, you may want to turn off DHCP here and set a fixed IP address. However, if you will be using Port Triggering, then you should be fine with DHCP assigned IP addresses.

Now that we have successfully performed a factory reset, we are now set to configure the phone for service with a BYOD VoIP provider of your choice (starting with a clean slate).

Updating Firmware on Cisco SPA504G

2011-11-24T18:02:00.000-07:00

Periodically, it's advisable to check for and update your VoIP device firmware. This article is focused on updating the Cisco SPA504G IP-Phone.

The process of downloading and installing the latest firmware is a fairly straight forward and simple process. However, if your VoIP phone is tied and locked to your VoIP carrier or work company, who supplied the IP-Phone, this procedure may not work for you or may not be advisable.

NOTE: These instructions only work on UNLOCKED versions of SPA504G.

If you purchased your ATA from an authorized Cisco-Linksys reseller, and configured it yourself (like I did) it will likely be an unlocked version.

First, you need to determine if you have the latest firmware version installed, or not.

As of this post, the latest SPA504G firmware version was 7.4.9c.

Checking Your Current Firmware Version

While you can check the firmware version directly on the SPA504G LCD display, for the sake of the procedure I'm outlining here, I'll suggest that you connect directly to the SPA504G with the PC that you will be using to download and install the firmware update from. And, then check the version level from your web browser connection. This way you can confirm two essential elements of this process:

That you know the correct IP address of the SPA504G on your LAN.
That you can login to the SPA504G using administrator credentials (if set)

By default, the unlocked versions do not have User or Admin passwords set (blank PW).

However, if the Admin PW has been set, you need to know it or you will not be able to complete the firmware update process. (I always use my own set admin password.)

Enter the IP Address of your SPA504G into your PC web browser

In my case, I enter 192.168.0.3

By Default, you will land on the Basic - User page.

From this page, you will see the current Software Version number.

Log into the Administrator account by clicking on the Admin Login link in the upper right-hand corner of the window.

If the Admin password has been set, the "Authentication Required" window will pop-up.

Enter the Administrator User Name and PW here.

By default, the admin User is "Admin". And, the Password is blank, or empty. If it has been set otherwise, it's up to you to know what it is...

The whole point of logging in to the administrator account above is to confirm that the SPA504G will allow you to login with admin credentials. Once you have established that your PC can connect to the SPA504G and login as administrator, you can now proceed with downloading the firmware software download directly from Cisco.

Downloading Firmware From Cisco Website

The Cisco website hosts the support and download pages for the SPA504G VoIP phone. You will find the SPA504G firmware in the Voice and Unified Communications section under the link for IP Telephony, and then IP-Phones. More specifically, the SPA504G page is located here.

Once you have found the specific download page for the SPA504G, proceed to downloading it to your PC by clicking on the Download Now button.

Confirm that your are downloading the correct version.

Click Proceed With Download button.

Agree to License Agreement by clicking on Agree button.

Save the .Zip file to a folder on your PC that will be used for the upgrade procedure.

Uncompress the .Zip folder.

This completes the firmware software download procedure.

Begin the upgrade procedure.

Make sure your SPA504G is powered-up and that your PC Ethernet cable is connected to the PC port on the SPA504G.

Double-click on the upgrade .exe file to begin the installation.

Note the Warning message before proceeding. Click Continue button.

Enter the IP Address assigned to the SPA504G. Then, click OK button to proceed.

If your phone has an administrative password set, you will be prompted to enter it:

Confirm the information collected and presented for your review.

Click Upgrade, to proceed with the installation.

Security Alert Windows

Depending on the security software installed on your PC, you may see a pop-up Warning.

In my case, it is the Windows 7 Firewall Alert.

Whatever security alert pops-up, you need to ALLOW in order for the firmware upgrade to continue.

After allowing the software to proceed, you will see the following window:

This is where some people get stuck...... "Waiting For Firmware Download Request From SPA"...

If you get stuck here, CANCEL the procedure, and start the .exe program again. And, once again being sure your firewall and AntiVirus or other security software is allowing the program to proceed.

Once the program is actually able to connect from your PC to the SPA504G IP-Phone, you should see the following window:

At this point, be patient while the firmware is uploading to the SPA504G phone. This may take a few minutes...

You will see the LED lights on your IP-Phone blinking and a message on the LCD screen of the SPA504G indicating that the upgrade is in progress...

Once the process is completed, you will see all status lights on the phone lit as normal and the phone is ready for making calls (assuming it was properly configured before beginning the upgrade). When you finally see the "Upgrade Successful" pop-up, you are home free !

You can confirm that the SPA504G is now running on the latest firmware by logging back into the phone via the web browser and confirming the Software Version on the landing page.

If you were having some firmware bug issues prior to the upgrade, hopefully those issues may now be resolved. Otherwise..... perhaps the next upgrade version will fix those issues...

CallCentric iPhone App Now Available

2011-08-28T16:00:00.000-06:00

Callcentric has recently released their new iPhone app which is now available through the iTunes store. The Callcentric iPhone app allows you to use your iPhone to place calls worldwide from your Callcentric account.

In essence, the iPhone app uses Callcentric's Click 2 Dial feature, which is generally initiated through your PC's web browser, while logged into your Callcentric account. Now, you can initiate a Click 2 Dial call directly from your iPhone.

Click 2 Dial is essentially initiating a "callback" to your iPhone phone number, or to any phone number you stipulate for the callback (i.e. your work, home, or friends phone.... where ever you happen to want to receive the callback to).

Callcentric provides more detailed information on how to download, install, and use the iPhone app for making calls through your Callcentric VoIP account.

Expand CallCentric Calling Card Gateways from 74 to 535 PSTN Access Points

2011-03-08T12:00:00.000-07:00

CallCentric Calling Card PSTN (Public Switched Telephone Network) Gateways are an excellent way to save on long distance phone calling. Currently, CallCentric has 74 world-wide access numbers in 16 countries, primarily in North America, Europe, Australia.

CallCentric is a BYOD (Bring Your Own Device) VoIP provider located in NYC, USA with world-wide VoIP termination and calling card calling rates starting as low as 1.65¢/min. I've been using CallCentric myself for years and connect via my ATA, IP-Phone, softphones, as well as Calling Card PSTN access. They have many notable features including: free in-network calling, free world-wide VoIP peering using SIP URI dialing, SIP Broker peer access using codes, free +883 iNum's and routing, as well as Calling Card access at VoIP rates. Their list of free features goes on and on...

The Power of PSTN and VoIP Consolidation

The focus of this article is about how to leverage and expand CallCentric's PSTN gateways, from their existing 74 access points to as many as 535 global access points, for calling to PSTN/Mobile destinations through CallCentric's VoIP network, at their same low rates as Internet-based VoIP calls. There are no extra charges, contracts, or fees for using any of the 535 PSTN gateways presented in this article to access the CallCentric VoIP network, other than your normal cost of a typical local area phone call.

So, what's the secret to expanding 74 access points in 16 countries to 535 access points in more than 47 countries?

Three Networks Provide Greater Access Than Any Single One.... at No Extra Cost!

I've leveraged CallCentric's free peering services by collectively combining the PSTN VoIP gateways of the iNum network, the SIP Broker network, with the existing CallCentric network. Now, we have the power of three VoIP networks combined to create a much larger and accessible global calling card network... at no extra cost.

Here is the PSTN access number breakdown:

CallCentric - 74 numbers in 16 countries
iNum - 71 numbers in 47 countries
SipBroker - 390 numbers in 38 countries

The foundation of combining these three services, to create one larger calling card service, is based on CallCentric's ability to funnel all incoming calls, via an incoming DID (direct inbound dialing) phone number, through their calling card feature. In essence, you will be creating your own "private" calling card access number (an iNum) for funnelling all incoming calls to your CallCentric account and then re-directing it through their calling card system. This is the premise of making landline (or mobile) calls to landline and mobile numbers via the CallCentric VoIP network, at VoIP rates (no VoIP equipment required).

In the scenarios presented here, the DID we will be using is a CallCentric issued DID, which will be an iNum +883 global number (free). In my case, I use my CallCentric issued iNum DID as the bridge between SipBroker network, iNum network, and CallCentric's network. Routing of iNums between these networks is free. To obtain your CallCentric iNum, you need to order it via your CallCentric user account dashboard, the same as any other DID, but it is free to obtain and use.

In a previous article, I explained how to combine the power of iNum and SipBroker gateways with CallCentric for making free world-wide calls to global +883 iNum numbers. The only short-fall of that methodology is that the majority of iNum destinations terminate to VoIP services and VoIP-connected devices. So, how can we leverage this methodology to call to landlines and mobiles, via iNums?

SipBroker + iNum + CallCentric + PSTN Gateways = World Wide PSTN calling for as low as 1.65¢/min.

Using the information I've researched and assembled, I discovered that it is now possible to place calls via SipBroker and iNum PSTN access points, and then funnel those calls through CallCentric to landline and mobile phone numbers, for the same termination rates as placing a normal CallCentric VoIP to PSTN-terminated call.

The key to expanding CallCentric's 74 access numbers up to 535 numbers is to call your CallCentric iNum number from any of the globally available iNum or SipBroker PSTN gateways. Your incoming iNum call to your CallCentric number will be re-routed through the CallCentric calling card system (see step 4 below). Then, your call is re-routed to your desired destination PSTN number of choice, and charged as any other CallCentric call-card routed call.

Enabling Calling Card Access Through Your CallCentric Account

Here are the steps required to enable your CallCentric account for calling card access via PSTN gateways:

Create your CallCentric account. (free)
Order your iNum DID through your CallCentric Dashboard. (free)
Configure your user Preferences (Calling Card tab) to "activate" Global Access Numbers to allow incoming calls from any of CallCentric's global PSTN access numbers. (free)
Enable (set to "Yes") Personal Dedicated Access Numbers (in this case, your iNum DID number) into your account (Preferences -- Calling Card tab). (free)
Set your personal access calling card PIN (4 to 10 digit pin - not required, but recommended). (free)
Save all setting changes to the Preferences - Calling Card tab page.

Now, any incoming calls to your CallCentric account will be redirected through the CallCentric Calling Card System. There are no extra charges, or fees, for using the calling card feature.

Your call-card calls "to PSTN destinations" are charged at CallCentric's normal VoIP calling rates, just the same as if you were placing a call to a PSTN landline or mobile destination from your VoIP ATA, IP-Phone, or softphone. In this case, the only difference is that you originate your call "from" a global PSTN gateway landline number provided by CallCentric, iNum, or SipBroker, as a local call using one of the access numbers (hopefully, there's one in your local area code).

Top-up Your CallCentric Pre-Paid Calling Credits Balance

Before you can place calls "to" PSTN destinations, you need to top-up your CallCentric pre-paid calling balance. Top-up with little as $5.00 of credits using major credit cards, or PayPal. Your balance decrements based on your per/minute call usage to PSTN/mobile destinations. Calls cost as little as 1.65¢/min. (Calls to USA/Canada are 1.98¢/min.) There are no connect fees, or other charges.

NOTE: Your balance is displayed in the upper left corner of your account management DashBoard page and the upper right corner of all pages of the CallCentric website, when logged into your account. The Add Funds link is located just next to your visible account balance value. The Add Funds to Balance link is also located in the lower right corner of the DashBoard under "Important Links" section.

The CallCentric system will notify you via E-mail whenever your account balance drops below $5, or any value that you preset in your Preferences - Billing page. Here, you can also set your account to Auto-Top-Up with your credit card, if desired.

Routing CallCentric PSTN Gateway Calls Through Your CallCentric Account - (74 Access Points)

If you have "Activated" CallCentric Global Access numbers in your CallCentric account (step 3 above), you simply place your calls through any of the 74 Global numbers provided by CallCentric. You will be greeted by a CallCentric gateway voice prompt asking you to enter your CallCentric account number (11-digits plus #-key), followed by your PIN number plus #-key (as set in step 5 above). Then, you will be prompted to dial the destination number you wish to call (followed by #-key). Your call will be connected and your pre-paid balance decremented at the rate determined by the destination of your call (1.98¢/min to USA/Canada).

Routing Your iNum PSTN Gateway Calls Through Your CallCentric Account - (71 more Access Points)

If you've enabled calling card access through your account, any incoming calls (to your CallCentric issued iNum DID number) are automatically routed to the CallCentric calling card system.

When placing your call via an iNum PSTN gateway access number, you will be prompted to "enter the iNum number your wish to call". In this case, you will be calling your CallCentric issued iNum DID, which you have already set as your Personal Dedicated Access Number for calling in via your iNum DID.

iNum Gateway TIP:

When calling through iNum PSTN gateways, you can enter your iNum number as one of two ways:

As a full 15-digit iNum: "883-510-00xxxxxxx", or
Only enter the final 9-digits of your iNum: i.e. "00xxxxxxx" (this only works via iNum PSTN gateways)

Once dialed, the call to your iNum DID is now routed to your CallCentric calling card gateway via iNum. At this point, you will once again be prompted to enter your PIN (followed by #-key), followed by your destination number you wish to dial (standard long-distance format, followed by #-key).

Routing Your SipBroker PSTN Gateway Calls Through Your CallCentric Account - (390 access points)

Placing your CallCentric calling card calls via SipBroker PSTN gateways works exactly the same as placing your call through iNum PSTN gateways - with one exception: At the prompt, you must enter your full 15-digit "883-510-00xxxxxxx" iNum number (no 9-digit short-cut as with iNum gateways).

Clearly, you gain the greatest world-wide coverage by utilizing the additional 390 PSTN gateways provided by SipBroker. However, by combining all three services of CallCentric, iNum, and SipBroker we now end up with 535 access numbers through which to place your CallCentric calling card calls, world-wide.

It's nice to have options!

In Summary, here are the dialing sequence procedures for each of the three methods outlined above:

1. - CallCentric PSTN Access Gateway Dialing Sequence:

Dial your a local area CallCentric PSTN access number
At the Welcome message, enter your CallCentric account number (11-digits, followed by the #-key)
Enter your CallCentric calling card PIN (as set in your account Preferences), (followed by the #-key)
Enter the destination PSTN number you wish to call to (followed by the #-key)

Example:

When calling through my local CallCentric global numbers (Calgary, AB, Canada):

Dial: 403-775-1750 (My CallCentric local area PSTN gateway - in Calgary, AB)
Enter: 1777xxxxxxx # (my CallCentric account number)
Enter: xxxx... # (my CallCentric personal calling card PIN)
Enter: 1-xxx-xxx-xxxx (if dialing within North America, otherwise, regular int'l dial sequence)

2. - iNum PSTN Access Gateway Dialing Sequence:

Dial a local area iNum PSTN access number
At the iNum Welcome message, enter your full 15-digit CallCentric iNum DID (or, 9-digit short-cut)
At the CallCentric Welcome message, enter your CallCentric PIN, followed by #-key
Enter the destination PSTN number you wish to call to (followed by the #-key)

Example:

When calling through my local iNum global number:

Dial: 403-775-1446 (My local-area iNum PSTN gateway - in Calgary, AB, Canada)
Enter: 883-510-00xxxxxxx (or, just the last 9-digits) # (my CallCentric iNum number)
Enter: xxxx... # (my CallCentric personal calling card PIN)
Enter: 1-xxx-xxx-xxxx # (if dialing to North America numbers, otherwise, regular int'l dial sequence)

3. - SipBroker PSTN Access Gateway Dialing Sequence:

Dial a local area SipBroker PSTN access number
At the SipBroker Welcome message, enter your 15-digit CallCentric iNum DID
At the CallCentric Welcome message, enter your CallCentric PIN, followed by #-key
Enter the destination PSTN number you wish to call to (followed by the #-key)

Example:

When calling through my local SipBroker global numbers:

Dial: 403-775-2208 (My local area SipBroker PSTN gateway - in Calgary, AB, Canada)
Enter: 883-510-00xxxxxxx # (my CallCentric iNum number)
Enter: xxxxx # (my 5-digit CallCentric personal calling card PIN)
Enter: 1-xxx-xxx-xxxx # (if dialing to North America numbers, otherwise, regular int'l dial sequence)

iNum - More Ways to Call for Free From Landline Phones

2011-01-28T14:02:00.000-07:00

The goal of the iNum initiative is to enable iNum users to be reachable from any telecom network in the world. Currently, the majority of telecom services providing and/or routing iNum +883 phone numbers is ~ 50+ (at time of writing). These are all primarily VoIP service providers.

So, until the "Ma Bells" of the world realize that iNum call routing (at reasonably low cost) is a good thing for everyone, we will need to rely on the PSTN gateway access points put in place by the various SIP-based VoIP Service providers for making PSTN based calls to iNum users.

Initially, I thought the iNum initiative (by Voxbone.com) was the only provider of PSTN telephone gateways for calling iNums from landlines and mobiles. Then, I realized that CallCentric also enabled iNum calling through their Calling Card PSTN gateway access numbers.

Now, once again, I have discovered another SIP VoIP service that provides many more world-wide PSTN gateway access points - SIP Broker, by Voxalot.

SIP Broker is actually a kind of interconnecting gateway for VoIP peering between over 2000 VoIP networks. And, they provide this service free. By using appropriate network codes, any VoIP user can essentially call any other VoIP user via the SIP Broker network. iNum is among the many VoIP peers accessible via the of SIP Broker network.

The real focus of this article is to bring to light that the SIP Broker network, besides providing IP interconnect, also provides PSTN landline phone (and mobile) access to the network.

In terms of iNum interconnect, you can now call any iNum number via any of the ~ 390 world-wide PSTN gateway access phone numbers connected directly to the SIP Broker network, in ~ 38 countries, which in turn connects to the iNum network.... for free.

Here Is How To Call an iNum via SIP Broker PSTN Landlines:

Dial a SIP Broker PSTN access number in your area code.
At the "Welcome" IVR message, enter your iNum destination number you want to call (followed by #).

That's it! It's very easy and... free.

Example:

I dial my local SIP Broker access number in Calgary, Alberta, Canada at: 403-775-2208
At the IVR prompt, I enter iNum Test number: 883-5100-0000-0093 (# to connect)
Call connects; hang up when finished.

I was pleasantly surprised to find out that dialing iNums directly over the SIP Broker PSTN access number did not require me to enter any network access codes - which would normally be required when calling via SIP (Session Initiation Protocol) through a SIP gateway.

Now, we have a vastly expanded world-wide network of free PSTN gateway access numbers by which to call to iNums:

~ 390 access points in 38 countries via SIP Broker network of PSTN numbers.
~ 74 access points in 16 countries via CallCentric Calling Card PSTN numbers.
~ 71 access points in 47 countries via iNum (Voxbone) network of PSTN numbers.

I've now provided you with ~ 535 ways to call iNums from world-wide access points... for free.

Just think, if you were a world traveller who needed to keep your long distance calling costs down, you now have about 535 reasons to get a VoIP phone at home or the office so you can stay in touch for little or no cost in long distance fees...

Skype Charges for iNum +883 Calls

2011-01-27T14:36:00.000-07:00

Lately I've been blogging a lot about calling to +883 global iNum phone numbers - for free. I've talked about how "most" well known VoIP services now provide iNums to their customers and route world-wide calls to iNum phone numbers - for free.

The one exception to the "free" rule that stands out for me is the name.... Skype.
Skype charges 2.3¢/min. (U.S.), plus their standard connection fee, to connect to +883 iNum numbers.
http://www.skype.com/intl/en-us/prices/payg-rates#cc=VO
As far as I know, even their unlimited plans are not exempt from this charge to +883 numbers.

This really surprises me. Why?
Because iNums are routed over the Internet.... similar to email... for free. iNums do not route through PSTN telephone networks like landline phone calls and thus are not subject to the same mega-profit charges of the Telco's.

It is my understanding that the iNum initiative would like to invite all PSTN landline CLECs (competative local exchange carriers) and ILECs (incumbent local exchange carriers) to participate in the iNum initiative to at least route iNum calls for a low nominal fee. This is a reasonable expectation and would help in the interconnection of iNums through all telephone exchanges throughout the world at very low cost to all involved.

But, Skype is an entity of another color. Skype is an Internet based service. After all, how is it that Skype can provide calls between all Skype users... for free? Because those calls are routed over the..... Internet.

However, when Skype routes calls to a PSTN landline, it is reasonable for them to charge a small nominal fee for interconnecting to landlines. After all, they have to pay a fee to "Ma Bell" to terminate through the PSTN network. And, like everyone else, it's reasonable for Skype to make a modest profit in the process. But, not for calls routed over the Internet that also terminate over the Internet and related VoIP carriers.

I just don't get Skype's rationale for charging a fee for routing calls that are essentially free to route over the Internet. I hope they come to their ¢ and rethink this strategy...

iNums - Call via CallCentric PSTN Gateways - Free (Part-2)

2011-01-26T14:34:00.000-07:00

In my previous blog article, I revealed that you can expand your PSTN gateway access points for calling to iNum +883 global numbers by not only using the numerous iNum gateway numbers provided by Voxbone, but also by using the CallCentric global calling card access numbers. All the while, keeping in mind that I'm talking about free gateway access (when calling from a local area phone to a local area gateway, or access point).

First Things First...

In order to call global +883 iNums through CallCentric's network of PSTN access points, the user is first required to open a VoIP account with CallCentric. No, this is not a got-cha ploy. Creating VoIP accounts with CallCentric is free. Calling to other CallCentric accounts is free. And, calling to iNum +883 global numbers..... is also free. No money is required in these cases - prepaid, or post-paid.

Prerequisites Required to Make iNum Calls via CallCentric Access Numbers:

Create a free CallCentric IP Freedom account.
Activate your account Calling Card Global Access Numbers setting.
Set and save your calling card PIN authentication (prevents fraudulent access).

That's it! Just three easy steps. (and, no money or credit cards!)

Summary of steps to make a call through Callcentric gateways:

Dial your nearest CallCentric Access Number Gateway
At the IVR greeting voice prompt, enter your 11-digit CallCentric account number (1777xxxxxxx)
At the IVR prompt, enter your PIN (you previously set it up in your account - Preferences - Call Card - PIN section). Can be set as 4-10 digits.
At the IVR prompt, enter the 15-digit iNum +883 phone number you wish to reach (+883-5100-0xxx-xxxx).

Example:

As an example, I will dial an iNum test number from my local Callcentric access number in Calgary, AB, Canada.

I dial 403-775-1750
At the IVR voice prompt, I enter my 11-digit CallCentric account number (1777xxxxxxx)
At the IVR voice prompt, I enter my 4-digit PIN.
At the IVR prompt, I enter the iNum test number as 011-883-5100-0000-0093.
My call is connected.

Note:

When calling from within North America, I replace the + with 011.

SPEED DIALING TIPS

Most phones now days have a built-in contact list for speed dialing. You can save a lot of time by programming your CallCentric gateway access numbers into your telephones speed-dial feature.

Many phones also have the ability to enter a "P" between multiple dialing sequences (Pause for the IVR) for making calling card calls. This way you can enter a P after the local access number, then your PIN, another P for next IVR prompt, then your account number, another P and then your iNum number. Unfortunately, many phones don't store enough digits for this long a sequence in one dial string. So, you may only have enough space to enter your local access number and perhaps your PIN. Then, manually dial the rest.

Replace the 18-digit iNum Number with a 5-digit Code

One really great feature of CallCentric is their online Phone Book storage feature.

When dialing an iNum with the + or IDD code (011 in North America), the total iNum dial string becomes 18 digits. This can be problematic for fumble-fingers like me. The solution is to use CallCentric's Phone Book Feature.

With the phone book feature (accessed in your user online account), you can pre-store your 18-digit iNum phone numbers and assign each entry a 5-digit star code. Now, when the time comes to entering your destination number (in this case, iNum), just enter the *Code (*75xx). This will speed things up considerably and eliminate fumble-fingers problems.

Enjoy making free iNum calls through the world-wide CallCentric Calling Card Gateways.

Now, you have twice as many ways to call iNums from PSTN landline phones (and mobiles).

iNums - Call via CallCentric PSTN Gateways - Free (Part-1)

2011-01-25T16:23:00.000-07:00

The iNum initiative, created by Voxbone, provides PSTN telephone gateways for worldwide calling to any iNum phone number that uses the +883 country code - for free. They continue to expand their PSTN gateway availability ongoing.

The CallCentric VoIP service, which I myself use, provides iNum numbers to their customers for free. Once I set-up my iNum number with CallCentric, I was able to receive my incoming iNum calls via my VoIP ATA or IP-Phone connected to CallCentric.

The point I'm working my way up to here is that CallCentric also provides PSTN gateways to access their VoIP network. And, because CallCentric supports free calling to iNums, I realized that I could also use the CallCentric PSTN gateways as an alternate means of free landline calling to world-wide iNums - just like Voxbone provides PSTN access numbers for free iNum calling.

CallCentric refers to their PSTN Gateway numbers as Calling Card access numbers. This is because once you access the CallCentric network via a PSTN gateway number, you can then make long distance calls to any PSTN landline or mobile for a small fee (1.98¢/minute on average). But, keep in mind that once you're on the CC network, you can then call any other CC user for free, access the SIP Broker Network for free, call any SIP URI for free (using the CC online Phone Book speed dial feature), and......... call worldwide iNums... for free.

So, you may ask, why do you care about CallCentric PSTN gateways, if you can already call iNums via iNum PSTN gateways?

The answer is that CallCentric has PSTN gateway numbers in cities and area codes that iNum (Voxbone) does not have a presence in. So, the idea here is to "expand your options" by which you can make free iNum calls from PSTN landlines or mobile phones.

For example, iNum provides 71 access numbers in 47 countries (at time of writing).

And, CallCentric provides a total of 74 access numbers in 16 countries (at time of writing).

While iNum has access numbers in more countries, CallCentric has many more area codes covered in the USA and Canada. So, if you reside in North America, CallCentric offers more area code numbers for PSTN access.

What this all means is that you, your family, friends, and colleagues, will have many more ways to make free calls to iNums from PSTN landlines and mobiles.

iNum - Landlines Now Call VoIP Phones World-Wide - For Free

2011-01-11T13:36:00.000-07:00

We all know that making long distance Internet calls PC-PC or VoIP-VoIP is commonly free. But, what about landline PSTN-to-VoIP long distance calls? Well, in this case, the caller usually has to pay the LD toll charge.... which may be why you're not receiving as many long distance calls as you would like to.

Now, thanks to iNum and Voxbone, your friends, family, and colleagues can place calls to your VoIP device(s) from a PSTN landline (or mobile) for the cost of a local call (typically free from landlines). How? By dialing a Local iNum Gateway Access Number. Once a call originates through an iNum network gateway, it is routed via Internet to any world-wide VoIP device that has been issued an iNum number (many VSP's issue iNum's for free).

If family and friends are lucky enough to live in, or near, one of the world-wide cities where there is an iNum access number - then bingo! They can call you as often as they like.... usually for free.

The catch for you, the VoIP user, is to have an iNum from one of the world-wide VoIP providers that now issue free iNum numbers. I myself use CallCentric, who issues iNums to their customers for free.

Here is how it works:

The caller dials a "local" iNum gateway number as a local call.
At the IVR voice prompt, enter the destination iNum phone number you want to reach.
Your call will now be routed and connected to the iNum destination number you entered.

That's it! It's that simple to make free world-wide calls via iNum PSTN access gateways.

Example of a Test Call:

I dial my local iNum gateway in Calgary, AB, Canada: 403-775-1446
At the iNum IVR voice prompt, I dial an iNum Test Number at: 883-5100-0000-0093
The call is connected to the iNum Caller-ID TalkBack service.
I hear my phone number Caller-ID spoken back to me.
I hang-up when finished.

TIP:

Instead of dialing the full 15-digit iNum destination number (883510000000093), you can just enter the last 9-digits of the iNum.

For example, the Caller-ID TalkBack number can be entered as: "000000093" when prompted by the gateway IVR.

So, if you are a VoIP user (like me) who has service with one of the VSP's that issue iNums, pick-up your own iNum and pass it on to your world-wide contacts so they can all start calling you from world-wide destinations for free!

Remember: You can get your free iNum through CallCentirc. Ordering an iNum through CallCentric is free. And, making and receiving iNum calls through your CallCentric account... is also free.

Transitioning from VoIP Softphones to Hardphones

2010-12-02T17:28:00.000-07:00

When I first began learning about VoIP, it all started with the easiest ways to get started - softphones on a PC. Initially, it all started with the Skype softphone, (when they actually provided free calls from PC's to landlines). However, the Skype softphone is locked to ....Skype.

Then, along came Gizmo5. They were always just a few steps behind Skype. However, last year Google bought-out Gizmo5 and shut it down from any new registrations (I still have a working account - actually, three accounts for testing). It seems Gizmo5 was destined to be an integral part of the new Google Voice offerings. (I originally had a Google Voice account when it was still GrandCentral, in beta, until Google bought them out, too.)

Skype, Gizmo5 and a few others I tested were "locked" softphones, meaning they only worked with that one service. However, by this time I had progressed to the point where I wanted a more pure VoIP experience where I could use a generic user-configurable softphone not locked to any particular VoIP service.

After all, the whole point of configurable softphones is to allow you to use one VoIP device with any, or as many different VoIP service providers you choose. Part of the challenge for me was to seek out and test the lowest cost VoIP providers who also provided the best quality, openness, and features.

After doing extensive research (this began about 5 years ago) on configurable softphones, I experimented with what few available free softphones I could find. These originally included X-Lite 3.0, QuteCom, NCH Express Talk, and more recently the 3CXPhone.

The problem with softphones is that they are always bound to a PC. That's not a bad thing, especially when I consider I spend most of my time working on a PC. But, there are times when I just want to use a VoIP telephone not tied to the PC. This brought me to the next evolutionary step in the process of advancing my knowledge and experience with BYOD (bring your own device) VoIP devices: hardphones and IP-Phones.

After much research, I came to the conclusion that Linksys and Cisco phones were the path I wanted to pursue.

I began by purchasing a Linksys PAP2T-NA VoIP ATA (analog telephone adapter). After much experimentation and trial and error testing, I realized that this is a very good VoIP adapter. Now, with this ATA I can plug a regular telephone (corded or cordless) into it and talk just like on a regular PSTN landline phone. (Note: The PAP2T must be located behind a NAT-Router.)

My next progression was to migrate up to the Linksys-Cisco SPA2102-NA ATA. For the most part the SPA2102-NA is identical to the PAP2T except that the SPA2102 has its own built-in NAT-Router with QOS (quality of service) priority routing. No need for an external NAT-Router before the ATA like the PAP2T requires. From my experience, the SPA2102 is also an excellent ATA. (Actually, it's my favorite of the two).

Both the PAP2T-NA and thee SPA2102-NA are "unlocked" 2-Line ATA's. What this means is that I can configure two separate VoIP service providers in each box, and connect two separate analog telephones to each box. Of course, you can't connect to just any VoIP provider. They have to be BYOD "open" providers that allow you to use any softphone, hardphone, IP-Phone, or even an IP-PBX that your choose to configure for yourself.

To learn how to configure your own PAP2T-NA or SPA2102-NA ATA with a BYOD VoIP provider like CallCentric, whom I myself use, I have written the the following articles:

VoIP Calling Cards - Roll Your Own

2010-11-19T13:21:00.000-07:00

One of the outstanding features of CallCentric VoIP services is their Global and Personalized Calling Card access features. They provide local access gateway phone numbers in all major Canadian cities as well as major USA cities, and various locations throughout Europe and Australia. And, if you don't live near a local area VoIP gateway number, just create your own personalized number, all at very low rates, and no connect fees.

If you are traveling to any of these cities, the VoIP gateways offer an extremely low-cost alternative to making long distance calls back to home or office (from any PSTN landline).

If you live in or near any of the gateway access cities, you can use the gateway as your Long Distance Carrier alternative, with no commitments. You really can't go wrong at 2 cents/minute - just pay as you go.

iNum Dialing - Speed Dial Methods

2010-10-02T14:50:00.000-06:00

If you are familiar with the new international iNum country code (+883) numbers now being issued by numerous VoIP service providers (and eventually PSTN carriers), you may recognize the dialing format required to access iNum's (supplied by Voxbone) by the familiar +883-5100-xxxxxxxx dial sequence. And, if you have taken the time to count the digits required to dial an iNum number, you will recall that it requires a minimum of 15 digits to dial a number.

iNum's Are Great.... but, they are very, very long

15 digits is the best case scenario if you are dialing directly through your VoIP or PSTN carrier who supports iNum routing (not all do, as yet). In fact, the VoIP carriers I use require that the iNum be preceded with the IDD (International Direct Dial code). Remember, the + sign in front of a country code typically implies that it may also require addition of the IDD code. Because iNum (+883) by nature is an international country code, it will typically require dialing to be preceded by the IDD for your country (or VoIP carrier).

Here in North America, the international IDD direct dial code is 011. Thus, dialing an iNum from a North American carrier becomes: 011-883-5100-xxxxxxxx. Now, the dial count becomes 18 digits.

How to Reduce the iNum Digit Count

I've come across a few methods that can be used to reduce your digit entry count when calling iNums. How much you can shorten your iNum dial string depends on your calling method and which speed dial method you choose from the options discussed below.

Your Telephone's Contact List

Once you create a contact entry for an iNum number in your telephone's contact list, then there are no more fumble fingers trying to get the number right. Just look up the iNum contact of interest, and dial the contact at the press of a single button (once you find the contact in your directory). This is probably the first method most people will choose to use.... as long as your telephone has a built-in contact directory.

The iNum Gateway Digit Short Cut Method (9-digit dialing)

iNum.net themselves recognized that users were having fumble-finger problems (and digit tedium) while entering the 15 digit numbers when making calls through iNum world-wide gateway access numbers. Thus, iNum.net implemented the Digit Short Cut when calling from PSTN phones or mobiles through one of their iNum access gateway numbers. Now, when calling through an iNum PSTN gateway number, when prompted by the IVR, you only need to enter your last 9-digits of the destination iNum as: 0xxxxxxxx. That's a savings of 6 digits! (Good job iNum and Voxbone!)

The VoIP Provider Phone Book Method (5-digit dialing)

I obtained my iNum number through CallCentric, for free. One of the great features CallCentric provides on their user web-based dashboard is the Phone Book feature. It can store up to 100 phone number entries. Now, the bonus to this is that I can Speed-Dial any of my phone book entries just by directly entering a 5-digit speed-dial code matching a particular phone book entry hosted on their web site. The 5-digit code follows the *75xx format. Now, from any VoIP ATA, IP-Phone, or softphone registered to my CallCentric account, I just directly enter *75xx of the corresponding phone book entry on my CallCentric dashboard. That's a savings of 13 digits!

(You may need to edit your VoIP ATA dial-plan to accept *75xx dialing)

The Linksys-Cisco ATA Speed Dial Method (2-digit dialing)

I use the Linksys-Cisco SPA2102-NA and PAP2T-NA ATA's. These VoIP phone ATA adapters have the ability to store up to 8 Speed Dial entries. Once your numbers are programed and saved, it only requires 2 digits to dial an iNum. That's a savings of 16 digits!! I explain how to program the SPA2102 here, and the PAP2T here for 2-digit speed dialing.

The VoIP Phone Dial Plan Method (8-digit dialing)

If you use an IP-Phone or ATA and often manually dial your iNums, you will find that editing the VoIP device internal dial plan will enable you to enter only the last 8-digits of any iNum. This will provide a savings of 10-digits!

For example, the following entry into a Linksys Cisco IP-Phone or ATA will enable 8-digit dialing of iNums:

"|<:0118835100>xxxxxxxx|"

I explain about Cisco SPA2102, SPA504, and Linksys PAP2T Dial Plans here.

If you are an iNum user like I am, I hope you will find at least one of the above dialing methods useful for streamlining your calls.

Speed Dialing on the PAP2T

2010-09-28T13:28:00.000-06:00

The Cisco Linksys PAP2T-NA has the ability to store up to eight (8) phone numbers for each of its two VoIP lines to facilitate 2-digit speed dialing.

My PAP2T-NA is configured with CallCentric on VoIP Line-1.

For the purpose of this post, I have configured a few test numbers for illustration.

To program your speed dial numbers, log into your PAP2T (as admin or user) using your web browser and the PAP2T assigned IP address. By default, there is no password required. But, if a PW has been set, you will be prompted to enter the user or admin password before entering. Once logged in, click on the "Switch to Advanced View" link.

Navigate to the USER 1 tab.

Enter your desired numbers in the boxes labeled "Speed Dial 2: " through "Speed Dial 9: "

Be sure to SAVE your entries before testing.

Illustration of Speed Dial entries:

To initiate a speed dial, just press (x)# on your telephone dial pad. Substitute the (x) with the index number of your stored phone number (2-9) followed by the # sign. For example, only pressing 2# will speed dial your first indexed speed dial number stored. In my case, pressing 2# on my phone connected to Line-1 will speed dial the CallCentric TellMe test number at 17771234567 (a free in-network call).

Now, with speed dialing enabled on your PAP2T-NA, you can quick-dial just by entering 2-digits instead of the usual 10 or 11 digit dialing typically required. Even if your phone has its own stored phone book numbers, you don't need to scroll through the phone book before dialing your favorite numbers.... Just 2-digits and you are as good as connected!

Note:

You can enter any number into the speed dial boxes that you can dial through your handset which is allowed by the PAP2T dial-plan. You can even enter SIP URI's into the speed dial boxes as long as you have Enabled IP Dialing on the PAP2T under Line-1 (located just below Line-1 Dial-Plan box).