Showing posts with label skype. Show all posts
Showing posts with label skype. Show all posts

Friday, March 26, 2010

VOIP Skype system replacment....?

For those of you who may find building your own VOIP/Skype system a bit overwhelming.... I may have just the thing for you.

Ooma! is a nice little VOIP box that the company claims has a one time fee and your set for life. (unless your doing international calling)



I have not had a chance myself to review the box, but would jump at the chance if it came up.

There have been a lot of rave reviews on line, and heck; even Cali Lewis gave it a good review!

From what I can tell the international prices are just as good as Skypes. For me the price of calling family in Russia is the same as Skype.

As much as I would love to jump on the Ooma bandwagon, I just cant justify the cost at the moment. It would take me about five to six years to make my money back using Ooma. Thats not saying Ooma is bad, its just I pay about "$110 a year" for Skype with international calling, and a backup service if Skype should fail. To top that I would loose all my cool siptosis/Skype features.

If anyone out there is using Ooma, please drop me a line and let me know what you think. Maybe one day I will jump on the Ooma bandwagon.

Saturday, May 2, 2009

SiptheeSkype Now SipToSis

Well my phone system had a little crash and I had to set my machine and ATA up once again. In the process I thought I'd take a look and see what updates SitptheeSkype had......

Well I found out the name changed again and is now called SipToSis. The name change is not that big of a deal, what is a big deal is how easy SipToSis is to set up.

NOTE: SipToSis now has a GNU GPL: $2.50 distribution fee. Believe me its worth it...and more.

Now just because it had a distribution fee does not say you couldn't distribute it yourself...after all it is under the GPL.

I actually wish I could have placed an additional $15 or so to the basket at check out to give a little gratitude for all the work that has gone into this!

So lets get with it!

Well you need the normal files....Java 1.5 JRE or higher, Skype, and SipToSis.

Basically; once you have java and Skype installed all you do is extract SipToSis files to a location and execute the SipToSis_win.bat file. From there allow jave to interact with skype by going to Tools-Options-Advance and click Manage other programs access to Skype (On the bottom). From there click allow.

The full process is well documented here, and gives you plenty of possibilities!

As for the dial plan....You may want to take a look over here.... and here... Once you have your dialplan in place take a look at the section over here that says "A word about the dialplan". Thats how I have mine set up, and it works nice!

If you have any questions...feel free to drop me a line.

Thursday, September 18, 2008

Skype/VOIP Gizmo Modded Phone Project!

So I was thinking that Skype has been working out pretty good and I was toying with the idea of making it my default phone service. The only problem...how do I get all my phones to use it without loosing any of my other services.

Hey....Wait a Minute What Happen To Gizmo?

I know, I know..... Gizmo has been working out good! Its just a bummer there is no subscription. Long story short....Skype is cheaper!

Come up with anything?

I thought I did.
I went out and bought a DPH-50U off newegg, for $19 with shipping and a $10 rebate.
The idea was I would build a PC and have this device hooked up to the PC. From there I would up-link it to my ATA and set my phones to default to Skype.


UPDATE:
The DPH-50U is absolute trash. I played around with it a bit and found you can't use Skype as the default call out service. If you want to use Skype you need to enter ## phone number * to call out.
The ## tells the device to use Skype. (you could in theory set all the phone number's in the address book on the phone with the ## and * to default.)
The other thing is the software to configure the DPH-50U doesn't seem to work. It looks like you should be able to set it to automatically set the ##, and *, but I couldn't get it to work no matter what I tried.

So What Are You Going To Do?

I did some more searching, and found this. Siptheeskype is a gateway for Skype and your ATA. You run it along side Skype on a PC, and you use your phone and ata like you normally would.


OK So What Can You Do With It?

What it gets me is:

* I will keep My PSTN
* One less device to deal with (not using the DPH-50U)
* I will Make Gizmo a secondary phone service
* Skype will be the Default Phone Service.


For a full list of options look here.

Ok So Tell US How And What You Did!!

Well if you remember my phone project; I will be using all the same hardware so there is no changes other than having to build an additional PC.
I decided on running Win2k on a PIC0-ITX with Skype and, SiptheeSkype. (I think the Fan less PICO case is ideal for this.)


Software needed:

java 1.6
Skype
SiptheeSkype

Obviously you will need to install java and Skype and have some credits.

Getting Things Ready.

I Unzip the archive into a folder called siptheeSkype.

Once everything was unzipped to the siptheeSkype directory I renamed the following.

siptheeSkype_sample.cfg to siptheeSkype.cfg.
SipToSkypeAuth_sample.props to SipToSkypeAuth.props.
SkypeToSipAuth_sample.props to SkypeToSipAuth.props.
SkypeOutDialingRules_sample.props to SkypeOutDialingRules.props.
SipOutDialingRules_sample.props to SipOutDialingRules.props

In the SipOutDialingRules.props I didnt change anything.

In the SkypeOutDialingRules I only changed the following.

# you could eliminate the 0 if you always use SkypeOut like this
#^([0-9]{7})$:+1561$1
#^([0-9]{10})$:+1$1
#^([0-9]{11})$:+$1

To

# you could eliminate the 0 if you always use SkypeOut like this
^([0-9]{7})$:+1myareacode$1
^([0-9]{10})$:+1$1
^([0-9]{11})$:+$1

Note: Where it says "myareacode", place your area code.
Example: if 365 is your area code it would look like this ^([0-9]{7})$:+1365$1

Next: Edit The SiptheeSkype.cfg

Note: I am using GizmoProject so these are the changes I made for my setup.

Find the following:

#Sample config with NO registration - change 192.168.0.4 to ip address of computer running siptheeSkype
# username and password not important in this mode
#Set to available port to transport SIP messages on siptheeSkype computer
host_port=5070
contact_url=sip:Skype@192.168.0.4:5070
from_url="Skype"
username=Skype
passwd=123456
realm=192.168.0.4
# --- end of NO registration example ---

Set the IP address to the IP address of the machine that will be running, siptheeSkype, and Skype. (you should make the address static)

Example: If your IP address is 10.8.80.2

#Sample config with NO registration - change 192.168.0.4 to ip address of computer running siptheeSkype
# username and password not important in this mode
#Set to available port to transport SIP messages on siptheeSkype computer
host_port=5070
contact_url=sip:Skype@10.8.80.2:5070
from_url="Skype"
username=Skype
passwd=123456
realm=10.8.80.2
# --- end of NO registration example ---

Enter your Skype username and password in the following fileds.

#Sample config with NO registration - change 192.168.0.4 to ip address of computer running siptheeSkype
# username and password not important in this mode
#Set to available port to transport SIP messages on siptheeSkype computer
host_port=5070
contact_url=sip:Skype@10.8.80.2:5070
from_url="Skype"
username=Skype
passwd=123456
realm=10.8.80.2
# --- end of NO registration example ---

Next: Find the following. (should be just below what you just edited.)

#Sample config WITH registration to GizmoProject - comment out NO registration info above first and uncomment the following
#contact_url=sip:1747???????@proxy01.sipphone.com:5060
#from_url="1747???????"
#username=1747???????
#passwd=?????
#realm=proxy01.sipphone.com
#expires=120
#minregrenewtime=60
#regfailretrytime=15
#do_register=yes
# --- end of WITH registration example ---

Un-comment/Remove # from the following.

#Sample config WITH registration to GizmoProject - comment out NO registration info above first and uncomment the following
contact_url=sip:1747???????@proxy01.sipphone.com:5060
from_url="1747???????"
username=1747???????
passwd=?????
realm=proxy01.sipphone.com
expires=120
minregrenewtime=60
regfailretrytime=15
do_register=yes
# --- end of WITH registration example ---

Add you Gizmo password to the line "passwd=?????" above.
Example: passwd=password

Add your Sip/Gizmo number to all lines that have 1747???????

Example: If your sip/gizmo number is 1-747-123-4567

contact_url=sip:17471234567@proxy01.sipphone.com:5060
from_url="17471234567"

Note: I was having audio problems so I had to set the following:

#If yes, will send RTP packets to address received from the otherside
# instead of what was received in the session descriptor.
# This may help with one way audio problems.
enableSendRTPtoReceivedAddress=no

To

#If yes, will send RTP packets to address received from the otherside
# instead of what was received in the session descriptor.
# This may help with one way audio problems.
enableSendRTPtoReceivedAddress=yes

If you still have audio problems....its a good chance it is because of a firewall.

Set the dial plan on your ATA.

To have Skype as the default phone service add the following to the begining of your ATA's dial plan.

Change 333 to your area code

(1333xxxxxxx<:@pcipaddress:5070>|remainder of dial plan)
Note: pcipaddress will be the address of the machine running Skype and SiptheeSkype.

To enable more then one area do the following.
Change 333 to your area code. and 444, and 555 to the area codes you will be calling to.

(1333xxxxxxx<:@pcipaddress:5070>|1444xxxxxxx<:@pcipaddress:5070>|1555xxxxxxx<:@pcipaddress:5070>remainder of dial plan)

To add 888 and 800 numbers

(1333xxxxxxx<:@pcipaddress:5070>|1444xxxxxxx<:@pcipaddress:5070>|1888xxxxxxx<:@pcipaddress:5070|1800xxxxxxx<:@pcipaddress:5070>remainder of dial plan)

Note: When dialing out you will need to enter the full phone number to use Skype.
Example: 1-333-123-1234

I have found with the current dial plan on my ata; If I dont enter the 1-area code it will use gizmo. (I think that is good! If Skype ever fails I can use the same phone to try calling from gizmo.)

I've tested this out making calls through the PICO (skype) from my house phones, and no problems as of yet. Sound is good, loud, and clear.

I will do a follow up as soon as I use SiptheSkype a bit more, and let you know what I think. So far I like it!!

This was written fairly quick so please excuse any spelling/wording.

Friday, August 22, 2008

UPDATE: To Possible Alternative to VOIP and Skype International Calling.

Well I was setting it up to test and I found that the price in the rates from, and the actual price differ for me. It looks like they quote you the cheapest price (Starting at) for the area you need to call.

For example, if you need to call Russia you are quoted $0.019 a minute. However, if you look to the right of the quote you will see a link that says "Click here for full price list" that link will give you the rates of the areas you would be calling. Examples Below.

$0.019 Landline Moscow, Saint Petersburg
$0.079 Landline All other landlines
$0.069 Mobile All mobile

Oh well....what doesnt work for one, may work for others!

Thursday, August 21, 2008

Possible Alternative to VOIP and Skype International Calling.

I was surfing the web and came across a company called Rebtel claiming cheap to free international calling.

Well this sounds good to me so I had to ask around.

What Did Ya Find?

In one of the forums I go to I asked if anyone has used them, and how was the experience.

I was told they seem to work well and sound quality is good.
So I decided to make an account.

Sign Up Process.

Very easy and straight froward.
The way it works is you sign up for the service (you are given a free ten minutes at sign up). When you enter your information in the form you enter your phone number and create a pin.

The next form/screen will ask you for your friends information:

Name
Country
Phone Number
Email address
Language (only choices are English, Spanish, Polish, and French) This will be for the prompt when your friend is calling you.

Hows the service work?

Once your account is set up Rebtel will give you a local number in your friends country. So in Rebtels own words:

"You give us the phone number of a friend in another
country, and we give you a local number for them."
"Save this number on your mobile so you can call
your friend whenever you want, for a fraction of your
normal international rate."


So How Much $$$?

Rates seem very good. For my situation its the best so far.


So What About Free International Calling?

Rebtel offers free international calling..Yes it true....But not for everyone.

The way it works is if your in one of the Rebtel-Countrys and your friend is in a Rebtel-Country, then you have free international calling.

Ok.... So How Do I Do It?

Once again, in Rebtels own Words.

Call your friend on their Rebtel number.
Ask your friend to call you back. Tell them to use the number shown on their phone screen.
Stay on the line. Don't hang up! In a few seconds your friend will join you back on the call. (this part sounds strange, but do as it says)

Or you can just watch the Video.

Thats it.


So What Do You Think Of It?

To be honest I haven't tried it yet..... but I'm going to!!
I love the Idea and think it should add more value to my Skype/VOIP phone system.

So When Are You Going To Tell Us How Well It Works?

I'm going to need my wifes assistance on that. I dont think it will take all that long....maybe a week or two.

I'll let you know the wifes verdict as soon as we give it a go!

Wednesday, August 13, 2008

Follow Up To The VOIP Phone Project

Things I've Learned:

Well....let me tell ya, I have learned a few things in the past day or two so...

Shall I go down the list?

Before I do let me explain something. I didn't learn all this because the system failed. I learned all this because I can't leave a working system alone and must always tweak things. So what I mean by that is at one point I had my network down! When I say down, I mean I unplugged everything. (Router, switch, and ATA no power, no network!)

Why Would You Do This....

Testing, tweaking, and just messing around.

So How About That List?

So when the network was down and everything was unplugged (except one of my cordless phones)
The phone still worked! I had an incoming call!

Why You Ask?

1. The PSTN line is plugged directly into the ATA, from the ATA its plugged into the 5 port phone jack. The cordless phone is also plugged into the 5 port phone jack. So it looks like the PSTN carries the current from the main line through the ATA into the 5 port phone jack, even thought there is no power to the ATA. Cool eh? So in theory I can also make local out going calls on the PSTN line if both VOIP systems go down. (For $5 a month I get to keep my PSTN line...very basic)

The next thing I found was with the Skype phone.

2. I powered up the Skype phone, but kept the router, and ATA off. What I found was the same situation as above, but.....I could not use Skype because there was no network.

So time to move on and play with this a bit more!
I power up the router and now I can call out on Skype. No big deal. So I had an idea.....

Question: What happens If I press talk on the Skype phone and enter a phone number?
Answer: It dials out using sipphone/gizmo.....Cool

So with the Skype phone I can choose what service I want. For Example:
If I want to use Skypeout I just select the person in my contacts and press talk.
If I want to use Gizmo/Sip just dial a number and then press talk.
Or dial a number, select option, and select landline

Very cool!!!

Anyway, thats my big list of what I have learned.

Monday, August 4, 2008

Skyp, Gizmo, Grandcentral, VOIP Phone Project

Into the Crazyness:

I hate bills, and I hate giving money to Big corporations. I guess you can say for the most part I'm anti everything.

I've been tossing money out the window for many years now in the form of whats called a phone bill. So I decided instead of whining about it I should try to do something about it......and this is what I came up with.

Reasoning:

Before I started this project, I was paying about $130 a month for my phone bill. (In State, Local Long Distance, Nation Wide Long Distance, International, Caller ID, blah, blah, blah.)

After tossing money out the window for a few years, I said to my self “There must be a cheaper way.” So I did a little reading and this is what I came up with.

Before I jump into how I have everything hooked up and configured lets take a look at what I wanted to accomplish.

  1. No longer use the PSTN Line. (kinda true)

  2. Have very cheap local,long, and international calling.

  3. Keep my Land Line phone number.

  4. Keep all the perks. (Caller ID, Hold, voice mail, etc.)

  5. Have a backup service just in case one goes down. (2 Lines)

  6. Have everything transparent to the wife. (Very important)

  7. Have all my phones ring when someone calls me no matter what number is being called.


Now that we know what I want to accomplish, lets move on to the Pros and Cons of the system.

Pros:

    1. Cheap

    2. You have control....well for the most part.

    3. Better then paying the phone companies

    4. Save Lots, and Lots of $$$$

Cons:

    1. If you loose power you loose your phone.

    2. You are the Admin, so if the phone goes down the Wife gets mad at you!

    3. You must press 1 to answer all in coming calls from GrandCentral.


What I did to over come the power problem was buy a UPS and place the, Router, Switch, ATA, and two of my phone bases on it. All units are very low power therefor I will get a few hours of service if my power goes out.


Note: I have FIOS so I also had to put the Fiber modem/converter, or whatever the thing is also on a UPS.


Things I had research to accomplish all the above:

REN:

One very important thing to know is the REN value of your phones and ATA. If you plan on hooking up phones to your ATA you will need to know this information. The REN is usually found on the phone labels specs. Explanation to all this is below.

In order to connect all your home phones to your ATA it must have a FXO port (PSTN landline) and you will need to know the Maximum Ringer Load of the ATA. For example the Linksys SPA3102 has a Maximum Ringer Load of 3 REN

The REN value on the ATA tells us how may phones it is capable of supporting/Ringing by determining the REN values of the phones you plan on connecting to your ATA.


Example:

If you have an ATA with a Maximum Ringer Load of 3 REN, then it could support 2 phones with a REN vale of 0.5.

So if you have 2 phones with a REN of 0.5 that would be equal to 1 REN. Therefor the ATA could still support another 2 REN.


Lets move on....


Equipment used.

ATA: Linksys SPA3102

The SPA3102 data sheet says the SPA3102 has a Maximum Ringer Load of 3 REN.

Phones:

Philips VOIP841 (REN:0.1B) Skype/PSTN

Panasonic KX-TG5439 (Very good sound quality) (REN:0.1B)

Uniden DECT2080-3 (Lots's of $$ and sounds like crap) (REN:0.0B)


Services Needed:

Get a number

Note: You will need to reserve a number from one of these providers and getting one may take some time.


GrandCentral Or Ribbit

Configure GrandCentral Per these directions.


Get A Skype and Gizmo Account.

I bought a 1 year subscription to skype, and a $10 credit on Gizmo.

Skype

Gizmo5


Parts:

I already had some house phones, so I just hit ebay for the Philips VOIP841 and Linksys SPA3102.

I used CAT 5 to make all the connections to the phone lines. (Because I had it)

If you decide to make your own cables you will need some crimp's and RJ-11 plugs.

I went to HomeDepot and picked up:

GE 5-Jack Adaptor, White Model 26131004

GE Surface Mount Jack, White Model 26136004



Note: Before starting this be aware... Some people have not had a very pleasant experience with gizmo, However others have. I for one have enjoyed the service for as long as I have been using it.


So...Shall we start....?

First I cut the main phone line coming in the house. For some very good info on cutting and connecting your line see here under "Safely Cutting Your Phone Line".

Once the main line was cut and separated from the rest of the house, I ran a CAT5 cable with a RJ-11 plug on one end, and just the Blue and Blue/White wires of the CAT5 on the other end. I connect the Blue and Blue/white wires onto the main phone line and plugged the RJ-11 end into the "Line" port (PSTN landline) into the ATA.

(Click to open color chart below)

I then ran a new CAT5 cable with both ends cut only exposing the blue and blue/white wires on each side. I tied one side of the Blue Blue/White wires to the remaining phone lines and the other to the Surface Mount Jack (Model 26136004). Once that was done I plugged the 5-Jack Adaptor (Model 26131004) into the Surface Mount Jack. All thats left for the moment is to connect the Skype Phones base, (Philips VOIP841), the ATA (into the “Phone”), and whatever else you want to plug into the 5-Jack Adaptor.



Now for configuration.

On my main router I am using the Tomato firmware.

I enabled UPNP on the router for the skype phone and the ATA. (You could just open ports if you wanted.)

Once UPNP was enabled I plugged the skype phone into the router and set the Skype account up.

You can set up your Skype account multiple ways from:

  • Skype PC Software
  • VOIP841 Handset
  • Web Interface on the phone base


To set the account up from the web interface you will need to know your phones IP Address.

Default User namd and password:

User: philips

Password: voip841

Once that is complete make a test call out with the Skype phone.


Now comes the fun....

Note: Before configuring the ATA you may want to check for a firmware update. If one is available update per the manufacturing instructions.


Router Configuration:

Plug a RJ-45 cable into an open port from the router to the internet port on the ATA

Set the ATA up with a static IP on the Wan side.


ATA Configuration:


Wan Setup Tab:

Connection Type: DHCP

DNS Server Order:DHCP,Manual

Primary NTP Server: Put what you would use


Click the above Screen Shot to open.


Lan Setup Tab:

Disable DHCP

Click the above Screen Shot to open.


VOIP System Tab:

Enable Web Admin Access: Yes

Click the above Screen Shot to open.

I also added password.

Note: Even though Web Access is enabled it will only give access to my local Lan and not the internet. (This is how I wanted it!)

The ATA's VOIP settings were done per this post


Note: If the setting dont automatically load you will have to go to each link in the post above and enter the info manually.


Conclusion:

Once I had the system up and working I called my phone company and and took the most basic phone service that is offered.

What that gave me was the call in number, and a little credit for local...all for $5.

Over all I spent about $250 for this project. So every thing should pay for its self in about 2-3 months. If my math is correct I should be saving about $900-$1000 a year!

So the end result is if anyone calls my land line all the phones ring (including the Skype phone). If anyone calls my Gizmo or Grand Central account, all phones ring. I can pick up any phone in the house and call out using Gizmo (Even the Skype phone. It has an option to select default (Skype, or land line)) If someone calls my Skype account only the Skype phone will ring. No big deal to me. Think of it as a second line, so if someone is talking on your main line you can call out on Skype.....Kinda nice!

Other Cool Links:

http://thatsmith.com/2008/06/grandcentral-addon-for-firefox/

http://www.internettablettalk.com/forums/showthread.php?t=21394

http://www.linuxjournal.com/article/8592


Helpful Forums:

Voxilla

Gizmo5