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Garmin Mobile PC

Windows XP is the OS that we used here.

Before start anything else, you need this tool which is called GPSCtrl.exe to activate or deactivate your built in GPS. This tool can be obtain freely from here

Garmin’s software needs some specific hardware device, which is, Garmin GPS, to used. To overcome this limitation, we will harness the power of Franson GPSGate. The problem is, Garmin Mobile PC (GMPC) is not able to recognize the ‘virtual com port’ created by GPSGate. Again we will try to solve this problem with the use of com0com free utility.

Impatient way

1. Prepare all the software mentioned above.
2. Extract GPSCtrl to C:\Tools folder
3. Install com0com. After installation, this utility will automatically add 2 virtual com ports, CNCA0 and CNCB0. Rename the port to any com number with setup (GUI interface). For me, I rename it into COM25 for CNCA0 and COM26 for CNCB0
4. Install GPSGate, cancelled the setup wizard if it appear
5. Activate the GPS with gpsOn.bat from the gpsCtrl utility
6. Run GpsGate.
7. At the input tab, choose COM2 (This should be the default port your internal GPS connected to). Click Run. If nothing wrong, you should see “Running OK!” and “GPS data is coming in.” Also at the system tray, you’ll see the icon’s color is yellow.
8. At the output tab add “COM Port” (NOT VIRTUAL or GARMIN emulated), and assigned to COM25 (was CNCA0).
9. Reboot the computer.
10. Activate GPS, run GpsGate then install and run GMPC
11. Stop the auto detection of GPS from Settings > System > Remote GPS > Disable
12. Choose serial port aka COM26 (was CNCB0) and click Enable
13. Your Garmin should be able to get connected to GPS for now.

So, the flow is like the diagram below

INTERNAL GPS -> GpsGate -> COM25 (CNCA0) -> COM26 (CNCB0) -> GMPC

GpsGate => to emulate Garmin GPS to run with Garmin software

com0com => NULL COM Port emulator

Connect to SIP VOIP using Asterisk Mobile Phone

This article is using the open source PBX Asterisk to setup a SIP account properly on the mobile phone. Use Nokia Series N81 to write this mobile configuration. Some of other mobile series will have different settings, therefore the configuration for different SIP servers may be similar.

Things you need to do is enable voip internet phone at your mobile phone
1. Your mobile phone must have WIFI connection that can connect to wireless access point .
2. Wireless account point connected to internet with open port UDP 5060
3. Asterisk / SIP Server username password

SIP Server /Asterisk Settings

We can use public VOIP or we can create our own SIP Server. In the next article, I will write more technical things about how to configure your own SIP Server / Asterisk. This article is written when the SIP Server is already installed

Create a new SIP account on the Asterisk / SIP Server

Add following setting to sip.conf (the username “agushalim” is just used for example!):
username=agushalim
type=friend
secret=SIP
qualify=no
port=5060
notransfer=yes
host=dynamic
context=from-internal
disallow=all
allow=alaw

SIP realm

The realm for digest authentication is set defaults to “asterisk”. Otherwise please verify your active realm in sip.conf.

Mobile Phone Settings

Navigate to Tools - Settings - Connection - SIP settings

Open the Options menu and select Add new - Use default profile :
Profile name : Asterisk
Service profile : IETF
Default access point : your WIFI / wireless access point
Public user name : sip:agushalim@SIP Server(SIP : username@domain or IP address)
Use compression : No
Registration : Always on
Use security : No

Proxy server
Proxy server address : sip: SIP Server (SIP : IP address of your asterisk / sip server)
Realm : asterisk (or realm of your asterisk)
Username : agushalim (Your SIP username)
Password : SIP (Your SIP password)
Allow loose routing : Yes
Transport type : UDP
Port : 5060

Registrar server
Registrar server address : sip: SIP Server (SIP : IP address of your asterisk / sip server)
Realm : -
Username : agushalim (Your SIP username)
Password : SIP (Your SIP password)
Transport type: UDP
Port: 5060

Now, navigate to Tools - Settings - Connection - SIP Settings / Internet telephone settings and create a new profile in the Options menu with the following settings:
Name: Default
SIP profiles: previously defined profile
Default call type

To switch between normal GSM calls or VOIP calls, navigate to Tools - Settings - Call - Default call type.

Select Cellular to make normal calls to the phone number or Internet to use VOIP to call the number or address.

Benefits of Asterisk

In my opinions, the only drawback of asterisk is that you need a really reliable network connection. Other than that, maybe the price of the IP Phone might be taken into consideration too, since the price is still higher if we compared to analog phone. But with that price, you can get additional feature that the analog phone doesn’t have, for example, video conference. Oh, and I forgot to mention that you will also need the skill to configure it.

Now, I’m here to talk about the benefit or advantageous as the pro’s of an Asterisk system compared to analog or even digital PBX system, so let’s get to it.

Asterisk is a VoIP (Voice over IP) telephony system based on IP (Internet Protocol). Therefore, for long distance call, you can have a very cheap communication channels. So the first advantage is the price you paid for communication will be reduced. For corporate which has distributed branches around the world, you can reduce voice communication cost significantly.

The next benefit of Asterisk compared to traditional PBX system (analog or digital) is you have unlimited dial plan. With traditional PBX, maybe you have to press certain key(s) to be able to make a call. With Asterisk, you can directly press the destination number without the need to press certain key(s) beforehand. This is the true advantageous over the traditional PBX system, especially when you have two or more PBX units involved.

From my own experience, in the company where I’m working at, we have 3 separated sites with 3 PBX units and all of them are different series. So, let’s say call from site ‘A’ to site ‘B’ started with pressing some keys to get a line (let’s say ‘88′). Call from site ‘A’ to site ‘C’ required pressing ‘30′. To simplify this for users, the call from site C to site B should also be started with ‘30′, but on the contrary, since the PBX units involved is from different series, so they have different kind of policy programmed inside and we can’t just do that. We have to dial ‘40′. Can you imagine, for an international call, you need to enter different country code if you are dialing from different country? With Asterisk, all I have to do is just programmed the extensions pattern in it, and then all users have to do to be able to make a call is to press the destination extensions directly without the need to know where they are.

Other pros of Asterisk compared to traditional PBX (from the point of view of a technical perspective). Brief of another benefits are: unlimited dial plan, answering machine, voice mail that can forwarded to your email, Interactive Menu, and still much more.

You should bear in mind that this is an open source software, so you have unlimited of choice to develop your own system.