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.