How to connect Nokia e71 SIP to Asterisk?

Today I tried another wonderful feature that built in this Nokia e71, VoIP SIP client. After several testing, it's works! Now, my e71 not only as my mobile phone but also my "mobile office extension phone"! It become an Unified Communication Device with combinations of several features e.g. mobile phone, emails, web browser, extensions phone, instant messenger client etc.

In order to connect your Nokia e71 SIP ( Internet Call ), there are some configurations are needed in your Asterisk Server. You may refers to my sip.conf settings as below :

[1010]
type=friend
secret=1234
qualify=yes
port=5060
pickupgroup=
nat=yes
host=dynamic
dtmfmode=rfc2833
dial=SIP/1010
context=from-internal
canreinvite=no
callerid=device <1010>
call-limit=50

You may also like to configure the "realm" in the sip.conf under [General] context, else you may leave it as default which is "asterisk".

Remember to restart your SIP service to avoid getting Error 400 later in your asterisk debug.

The next step that you need to do is configure your phone ( Make sure there is Internet Connections ).

Step 1:

Go to "Menu"-->"Tools"-->"Settings"-->"Connections"-->"SIP Settings". Then, click on "Options"-->"Add New"

Profile name: <to be defined>
Service profile: IETF
Default access point: <your wifi access point>
Public user name: sip:<SIP user>@<domain or IP address>
Use compression: No
Registration: Always on
Use security: No

[Proxy server]
Proxy server address: sip:<IP address of your asterisk>
Realm: asterisk or <realm of your asterisk>
User name: <SIP user>
Password: <secret>
Allow loose routing: Yes
Transport type: Auto
Port: 5060

[Registrar server]
Registrar server address: sip:<IP address of your asterisk>
Realm: asterisk or <realm of your asterisk>
User name: <SIP user>
Password: <secret>
Transport type: Auto
Port: 5060

Now, you should able to see that your SIP is Registered.

Step 2:

Go to "Menu"-->"Tools"-->"Settings"-->"Connections"-->"internet Telephone Settings". Then, click on "Options"-->"Add New Profile"

Name: <to be defined>
SIP profiles: <previously defined profile>

Try to make an Internet Call by select "Options"-->"Call"-->"Internet Call" after key in the telephone number.

Internet Call for Nokia e71

MediaRing MobileVoIP is another Internet Mobile VoIP Call application Nokia E-series and N-series. Now, you can make VoIP call to mediaring subscriber FREE and lower rate to any non-subscriber.

It offers free mobile to mobile calls regardless of where you and your callers are around the world, and regardless of whether you are with the same mobile service provider or not. If your friend does not have a mobile phone or has not signed up for MediaRing MobileVoIP yet, you can benefit from very attractive low calling rates to anywhere in the world - fixed or mobile phones!

All you need is Nokia E or N series smartphone* and WiFi connectivity to the Internet. Get your family and friends around the world to sign up and you are off to making and receiving mobile calls amongst yourselves absolutely free!

Supported phones :

The below table shows the compatibility firmware for Nokia series :

More info on : http://www.mediaringmobile.com/

IM using your mobile

IM, so called Instant Messenger, i.e. MSN, Yahoo! Messenger, Skype, Google Talk etc. is a tool that transform real time communication on text-based form between two(2) or more individual over the IP data network. It has become an important communication tool in our daily life.

Todays mobile phone has been embedded with various form of data communication transport methods like GPRS, GSM 3G, HSPDA, and the most cost saving method --> 802.11b/g WiFi. And I believe the mobile phone which built in with the 802.16e WiMAX chipset will be on the market very soon.

Thus, it also means that you can have your Internet data communication ON with you wherever you are, and full mobility! The most important thing is that with the WiFi network meshed around the cities in the world and WiMAX network which start growing everywhere in the world! And it is FREE or should I call it unlimited usage on a fixed charges.

Today, Windows Live has ready with Messenger Mobile client - Windows Live Goes Mobile. You can chit chat via mobile, check your hotmail, even MSN "spacing". Check it out... http://www.messengergoesmobile.com/

I prefer another application called Fring. The greatest thing for Fring is it built in many messenger clients like MSN, Skype, Google Talk, ICQ etc and the greatest client is the SIP client. With this, now I can have my office extension number following me whenever I am, and it is a mobile extension phone!

How Fring work?

Check out more from Fring's website.

Let's see and now my Nokia e71 is equipped with Fring also!! haha...

Mobile Number Portability in Malaysia

After YEARS of discussing this topic, MNP, by the MCMC. Finally, it will be implemented on 29th August 2008 ( Friday ). It would started for Pre-paid user and followed up by Post-paid user in a couple of weeks. The nationwide launch for the MNP service is in October.

With the MNP implementation, the mobile user no longer 'tight' with their mobile service operator.  You will have the convenient freedom of switching to competing mobile service operators while retaining your respective phone numbers. It means that you can switch any operator from Digi to Maxis or Maxis to Celcom etc. with your phone number e.g. +6016-XXXYYYY remains the same!

There will be a RM25 charge everytime the MNP is used to switch to a new operators.

More info on MNP : http://en.wikipedia.org/wiki/Mobile_number_portability

Build you own SIP server : Asterisk

With these applications, you can configure your own IP PBX easily at office, or even at your own house! It is easier than you think~

Trixbox

It has FREEWARE! Now you no need to know and crack your head to figure out how to look into MANY MANY config files in linux...it is all GUI, for standard feature if you are looking for.

http://www.trixbox.org

AsteriskNOW

This is another open source freeware that just came across, an open source that built up with customized linux distributions that includes Asterisk! AsteriskNOW is much more easier than Trixbox, you can create your own voice sip server within a day! Easy to install, and offer flexibility, functionality...but it is not advanced type. If you are looking for some 'great' features, then it is not suitable for you.

http://www.asterisknow.org

If you are looking for Analog FXO/FXS Gateway ( What is FXO & FXS ? ), maybe you can try Digium or Planet for home user. Definately, you may go for Cisco as well...

What is FXO and FXS ?

FXO and FXS are the ports used by the analog phone or known as POTS ( Plain Old Telephone Service )

FXS, known as Foreign Exchange Subscriber interface, is the port which actually provide analog line to the subscriber. It is the "socket on the wall" which delivers a dialtone, battery current and ring voltage.

FXO, known as Foreign Exchange Office interface, that deliver the analog line. It is the port on your analog telephone/fax machine incoming port. It delivers an on-hook/off-hook indication (loop closure).

How does it works, technically ?

The FXO and FXS is always working on a pair. A FXO port is always connected to a FXS port or vice versa.

Technically, when you wish to make an outbound call :

1. You pick up the phone from FXO devices. Then FXS port will detect that it is OFF-HOOK.
2. When you dial a number, a DTMF ( Dual-Tone-Multi-Frequency ) digits will be passed to FXS port.

When there is an inbound call :

1. When FXS received an incoming call, it will generate a DC ring voltage to the FXO device.
2. The FXO device ( phone ) will rings.
3. You pick up the phone, and line is established.

Note: The analog phone line passes approximately 50 volts DC power to the FXS port. That’s why you get a faint ‘shock’ when you touch a connected phone line. This allows a call to be made in the event of a power cut.

Type of connections

Without PBX system ( i.e. home land line )

A phone is connected directly to the FXS port ( on the wall jack ) which provided by your landline company.

With PBX system

If you have a PBX, then you connect the lines provided by the landline company to the PBX and then the phones to the PBX. Therefore, the PBX must have both FXO ports (to connect to the FXS ports provided by the telephone company) and FXS ports (to connect the phone or fax devices to).

With FXO Gateway

To connect analog phone lines to an IP phone system, you need an FXO gateway. This allows you to connect the FXS port to the FXO port of the gateway, which then translates the analog phone line to a VOIP call.

With FXS Gateway

An FXS gateway is used to connect one or more lines of a traditional PBX to a VOIP phone system or provider. You need an FXS gateway because you want to connect the FXO ports (which normally are connected to the telephone company) to the Internet or a VOIP system.

With ATA adaptor ( FXS adaptor )

  An FXS adapter is used to connect an analog phone or fax machine to a VOIP phone system or to a VOIP provider. You need this because you need to connect the FXO port of the phone/fax machine to the adapter.

Source from 3CX