Oldstone Place » VoIP

Testin Asterisk

dmortell — Wed, 23 Nov 2005 08:06:28 +0000

I downloaded Asterisk@Home from http://asteriskathome.sourceforge.net It took a couple of hours at 46kbps

I have VMware, but you could also use the free VMWare Player and QEMU.
Set up a new virtual machine in VMware with an IDE disk and set the CDROM to point to the Asterisk@home ISO file.
Start the virtual machine and the A@H install screen will be displayed. Pressing enter will repartition the virtual machines hard disk and install Red Hat Linux, Apache, MySQL, Asterisk and an Asterisk web-based administration tool called AMP.
After installation, login as root with password password.

Change the root password by typing passwd.
YOu can access advanced configuration settings by typing asterisk -r.
Run netconfig if you want to sassign a static IP address.

Note the URL displayed and paste it into you browser to open the AMP administration tool. Click Asterisc Management Portal and enter username maint and password passowrd.

First add an extension by clicking Setup > Extensions.
Select SIP
Assign an extension number (301 for example), a display name (Demo301).
Enter a password in secret (pass301 for example).
Set Voicemail & Directory to Enabled to access more options.
Enter a voicemail password and email address.
After clicking Submit, I got 2 errors. These didnt appear when I created another IAX2 extension, and they didnt seem to cause any problems.

Warning: Invalid argument supplied for foreach() in /var/www/html/admin/functions.php on line 2599
Warning: Invalid argument supplied for foreach() in /var/www/html/admin/functions.php on line 2460

To test your extensions, install a softphone such as X-Lite from www.xten.com or SJphone from www.sjlabs.com on your Windwos PC.

On installing Xlite, if it cannot connect the settings screen will open. Under System Settings, select the SIP proxy settings then double-click the first entry called Default and change the following:

Username: 301 (extenstion umber)
Authorization User: 301 (extension again)
Password: pass301 (secret password)
Domain/Realm: 192.168.11.6 (PBX IP address)
SIP Proxy: 192.168.11.6 (PBX IP Address)

X-Lite cannot use silence detection when it is used with Asterisk server. To solve this problem we must turn on the ‘Transmit Silence’ option. We can find this option in Menu>Advanced System settings>Audio Settings>Silence Settings>Transmit Silence. By doing this we are assured that X-lite will send always audio frames even after it detects a silence.

Next click Digital Receptionist in APM and enter your extension (301)
Try recording a message. I couldnt get anything out of mine, but maybe you’ll have better luck. But you need at least one message to set up Incoming Calls next. YOu can set up a complex IVR system here later if you want.

Next go to Incoming Calls and set the regular and after hours times to * and set it to go to Digital Receptionist Menu #1: Main Menu

Click Panel to watch the activity on your PBX

Now if you try calling your own extension (301) on your softphone, you’ll hear a pleasant womans voice telling you thet you are on the phone so please leave a message. After you leave a message and hang up you sould get an email notifying you of the voice mail. You can listen to the message by calling *98 on your softphone or using the web browser based interface linked in the email. The URL in my email was missing the server name so it didnt work, but with a but of editing I was able to open the webpage, enter my voicemail password and listen to my messages.

Now to set up another phone and get a connection to the outside world.

Running Asterisk on Windows

dmortell — Wed, 23 Nov 2005 04:45:25 +0000

This article shows how to run Asterisk on Windows by using the free VMware Player and QEMU emulator with the Asterisk@Home ISO (500Mb )

http://voipspeak.net/index.php?option=com_content&task=view&id=45&Itemid=28

Btw you can download OS images preinstalled with a couple of applications for QEMU from http://www.oszoo.org/

Two Softphones on LAN

dmortell — Wed, 23 Nov 2005 04:30:39 +0000

If you have two softphones that you want to receive calls from the internet, set the ports as below then set your router to forward to each phone.

X-Lite 1:
Listen SIP Port: 5060
Listen RTP Port: 8000
Forward port 5060 to this PC.

X-Lite 2:
Listen SIP Port: 5061
Listen RTP Port: 8006
Forward port 5061 to this second PC.

If you have problems try forwarding ports 8000-8005 to PC 1 and 8006-8011 to PC2. X-Lite has 3 LINES! and each uses 2 Listen RTP ports, that’s why the first uses 8000 and 8001 for line 1, 8002 and 8003 for line 2 and so on.

Wait, that cant be correct. 5060 is used as the inbound port as well that is the nature of the sip protocol unless you are using the fwdnat proxy. So dont change the listen SIP port, leave it at 5060.

Soft Phones

dmortell — Wed, 23 Nov 2005 03:26:28 +0000

If your PC has speakers and a microphone, you can make phone calls by downloading a soft phone. Or you could replace your microphone with a USB phone, a phone that plugs into your PCs USB port.

Download software phones from SJLabs at http://www.sjlabs.com/sjp.html
There are packages for Telephony providers such as Vonage, VoiceGlo, Free World Dialup, and Stanaphone.

Dante’s DIAX Software phone is small (300kb) and free for Windows from http://www.laser.com/dante/diax/diax.html

Versions of Microsoft Messenger 4.7

Idefisk is a good looking softphone client for IAX2
http://www.asteriskguru.com/tools/idefisk_beta.php

Useful Links

dmortell — Wed, 23 Nov 2005 02:12:54 +0000

The main Asterisk website, put together by the web developers at Digium

http://www.asterisk.org/

Astrisk documentation project
http://www.asteriskdocs.org

http://www.asteriskguru.com

A wiki about all things VOIP
http://www.voip-info.org

VOIPSpeak has news and articles all about VOIP, including useful articles on step-by-step configuration of Asterisk@Home and building your own PBX
http://voipspeak.net

Digium provides hardware for connecting phones and phone lines to your PC, and suppport for Asterisk PBX.
http://www.digium.com
They also host useful forums for the Asterisk community
http://forums.digium.com/index.php

Sangoma also provides hardware for connecting phones and lines to your PBX
http://www.sangoma.com/

Trying Asterisk

dmortell — Wed, 23 Nov 2005 01:57:59 +0000

There are a couple of ways to try Asterisk

The easiest is to download an ISO of a live Linux CD with Asterisk from AstLinux at http://www.astlinux.org
Burn the ISO to CD-R (or compact flash) and reboot and you’ll have a working Asterisk system. You’ll need a USB or flash card to save your configuration, or you can use the CD to install AstLinux on your harddisk. NOTE that this will erase your disk!
You could also downlaod the free VMware Player, set it to boot from the ISO using the virtual CDROM, then install AstLinux in the virtual PC.

Asterisk @ Home is another possibility. At 485MB, the ISO file is a hefty download compared to the 30MB of AstLinux, but it includes Asterisk, AMP, FOP, MySQL, Apache, etc

Asterisk

dmortell — Wed, 23 Nov 2005 01:47:39 +0000

Itoh-san was the first to tell me about the open-source PBX system called Asterisk. His company at the time was seriously thinking about changing from their existing Cisco IP phone network to Asterisk.

Asterisk 1.2 released on Nov 17 2005. Available from http://www.asterisk.org/

Calling PSTN

dmortell — Tue, 22 Nov 2005 22:10:56 +0000

fwdOut shows how to configure your Asterisk PBX to allow calls to PSTN phones via other Asterisk servers who have agreed to share their PSTN connections. InN return, you have to make your PSTN connection available for other users to call your local area. You get credit when someone uses your access point, this allows you to make calls through other peoples connections. If you make a lot of international calls, this could save money if your local phone bill doesnt increase too much.

http://www.fwdout.com/web/

Vonage is a fee-based PSTN replacement/enhancement service, SKYPE is an internet communications service with a fee-based PSTN connection, and FWD is an Internet-based calling service.

So, while you can’t usually call free, there are many services that allow calls for a few cents per minute. These are just a few. Most of the above allow you to provision an account with say $10 or so and use the service until the funds are depleted. Some have DID or dial in numbers so people can call you. Some offer 800 numbers in the USA, UK, Israel…

www.Libretel.com
www.IconnectHere.com
www.voicepulse.com
www.calgarytelecom.com
www.voiptalk.com
www.nufone.com [defunct]
www.voipjet.com
www.sipgate.com
http://www.voipfone.us/

sipphone.com offer 5 free 1 minute calls per day.

since 800 numbers are free and can be dialed from FWD, you can also use any calling card, which uses an 800 number for access, to dial normal phone numbers.

Make a new hotmail/yahoo email account for spam mail then sign up for a free calling card at www.phonehog.com

You can receive incoming calls from the normal PSTN network. You have to signup to make this happen in most cases.

Get a Seattle, WA, USA number through IPKall http://www.ipkall.com/
Get a Michigan, USA number through ClearPath http://www.clearpath1.com/fwd.htm
Get a UK number through Call UK http://www.calluk.com/
Use a LibreTel number http://www.libretel.com/ (does not support dialling FWD numbers higher than 80000)

(0845) 004 5566 xtraphone.com (UK Local only)
+31 20 3987567 XSALL