Impactos en la dinámica de las relaciones

There are a number of softphones available for use with Elastix. Some of them are free.

Following are 2 examples of the free softphones.

6.1

C

P

X-L

S

X-Lite Softphones can be downloaded here:

http://www.counterpath.com/xlitedownload.html

After installing the softphone (assuming that users can install the softphone), we need to configure the sofphone.

When you start X-Lite 3 for the first time, you will see the following screen.

Click the Add button.

You will then get the following screen.

This is where you will enter your credential

At the various fields, add the following:

• Display Name: Your Name - not Ben Sharif ☺

• User Name: Your extension number

• Password: The password of the extension when you created it in Elastix

• Authorization User name: The same as your User Name or extension

• Domain: Your Elastix IP address Put a check mark in the Register with domain and receive incoming calls

Make sure the Target Domain Radio Button is marked.

For this purpose, that’s all you need to do and Click apply, OK and close at the next screen. You are now ready to use the X-Lite softphone.

Sometime it is necessary to change the Honor setting in X-Lite for it to work correctly. To do this, you would need to go to the advance setting of X-lite.

Those using X-Lite having this problem may have to make the following correction in the advance setting of X-Lite.

How to get to the advance setting?

In X-lite:

Dial ***7469 (SEND)

This will bring up the advanced settings window Filter for honor

Double click on the honor entry and change the value to 1

6.2

BOL SIPP

I have found the BOL SIPPhone extremely simple to set up for use with Asterisk and it also has a call forward facility that I use from time to time.

You may obtain a copy of the BOL 2000 sipphone from the link below.

http://www.bol2000.com/website_c/download /sipphone/BOL%20SIPPhone_EN.msi After downloading and set up you will see the following when it is run:

To configure the softphone, click on the hammer icon and you will see the following.

Profile Tab

This is the only screen that is required to be filled in.

These are the only information required:

Account: <enter the extension number e.g.

3001>

Password: 3001 (I use the same number as password for simplicity. I use the same password when I set up the extension in Asterisk).

Domain/Realm: <leave it blank>

Proxy: Your Asterisk network address e.g. 192.168.1.100 Port: 5060

Check the Auto Login and Keep Password.

Then click OK.

Audio & Video Tab

Next Click on the Audio & Video Tab to ensure that audio properties set is consistent with the Audio card installed in your PC/Notebook.

The illustration above depicts the sound device installed in my Notebook.

• Click on the Tuning Wizard to tune your audio input and output.

• Check Auto Send Video (if you are using Video). I checked it anyway.

• Check Auto Receive Video (if you are using Video). I checked it anyway.

Click OK

Network Tab

Ensure that your Internet Connection Type is set to LAN.

You may ignore the Information of Network field.

Click OK

STUN Tab

You may or may not need to use a STUN server. If you need to use it, click on the STUN tab and enter the STUN server you want to use. If you do, a list of publicly available STUN server is listed in the section referring to STUN Servers later in this document.

In my case, I do not use STUN and therefore I left that section blank and Enable STUN check box un-ticked.

Click OK to close.

Call Forward

This is pretty simple to set-up. To forward an unanswered call to this extension, all you need to do is click on the Call Forward tab and enter the telephone number you want to forward your incoming calls to. You have 3 options of call forwarding – Always, On Busy or On No Answer. However this facility is only available if your PC is on and the softphone is active.

Click OK when done.

You might want to set-up a couple of PCs with the softphone after which you may start testing your brand new Phone System by dialling each extension in turn.

If you use one of the softphone and dial 7777, Asterisk will simulate an incoming call.

Make sure that in the general setting you have allowed for anonymous sip calls and you also have an inbound route for “any DID/any CID” otherwise the 7777 will not work.

Once done, you may test your softphone connection to Asterisk.

If everything has been done as explained above, we should be able to make and receive calls between our internal extensions. If not, it is time to re-inspect what we have done above and make the necessary correction before attempting to go any further.

Let’s take a break and test the soft phone extensions by making calls to each extension that we have created.