Test asterisk without hardware


#1

hi

i’d like to test asterisk before buy hardware that it needs…can i do that? and how?
for example: i’ve tried bayonne, and first of all 'ive lanched it with the “dummy” driver, i tested it and when i saw that all works fine i’ve bought a D4-PCIU and it works!

now…i’m the sysadmin of a callcenter with 15 agents, now i am using a callcenter software developed by siemens, but i don’t like this software…i’d like to migrate to asterisk…but how can i test it before the migration?

can i start asterisk with a dummy driver?

thanks in advance Nicola


#2

Yes you can test Asterisk without buying the hardware first. These are brief instructions, more detailed instructions can be found on asteriskguru.com/tutorials/ ( ** ). The * and ** below indicate where more information can be found for each step

1.) Load your favorite Linux disto on to a computer ( faster is better )

  • 2.) Install Zaptel & Asterisk ( asterisk.org/index.php?menu=download ) you can use either the CSV or Stable versions. Follow the directions on the link. Make sure you uncomment the ztdummy in Makefile ( not built by default )
  • 3.) run ‘modprobe zaptel’ and ‘modprobe ztdummy’
  • 4.) edit the /etc/asterisk/sip.conf file as follows
    [xlite]
    type=friend
    regexten=1234
    username=xlite
    password=password
    callerid=“Jane Smith” <5678>
    host=dynamic
    nat=yes
    canreinvite=no
    disallow=all
    allow=gsm
    allow=ulaw
    allow=alaw
    ** 5.) Download the XTen X-Lite Sip phone from xten.com/index.php?menu=prod … u=download
    ** 6.) Run X-Lite, go to menu, click on system settings, SIP proxy, Default
    set username = xlite, Authorization User = xlite, your password, and IP address of machine
    7.) Make sure there is no firewall running on your asterisk machine or that the proper ports are open on the firewall.
    ** 8.) On the X-Lite Sip Phone, you should see Logged In - Phone: xlite. This means you have logged into the asterisk server and are ready to make a call. Dial extension 1000. This connects you to the demo setup in the sample extensions.conf file. You should hear the message congratulations and a bunch of info about asterisk.

That’s very brief but should get you started.


#3

use firefly , it is easier to configure for a sip phone.

virbiage.com/download.php


#4

Feel free to replace X-Lite with any softphone that is capable of using SIP. You can use a softphone that is capable of IAX, however the instructions above are for SIP. Here’s a giant list of Soft and Hard IP phones.

iptel.org/info/products/sipphones.php

Dan


#5

thanks!


#6

Hi,

I followed the above setup instructions but can’t seem to get it working.
When I dial 1000 from my xten, I just get a busy tone with “Call failed: 404 Not Found”.

Any suggestion where to look at?

Also, I was just wondering if there is a way to configure sip calls such that when an unknown extension is called, the sip caller will be directed to say the main menu.

Rgds


#7

'I followed the above setup instructions but can’t seem to get it working.
When I dial 1000 from my xten, I just get a busy tone with “Call failed: 404 Not Found”. ’

This means that you do not have extension 1000 setup in your dialplan. Please look at your extensions.conf file and make sure that you have an extension 1000 setup and that it is under the correct context. If you still cannont find the problem, then you will need to post the extensions.conf file.

Off hand, i’m guessing that you either aren’t using the sample configuration files, that you have removed the demo context from your default context, or that your incoming sip call is in the wrong context.

Also, I was just wondering if there is a way to configure sip calls such that when an unknown extension is called, the sip caller will be directed to say the main menu.

Asterisk has a special extension ‘i’ used for this. Below is a link to all of the special extensions that asterisk implements. It’s about half way down the page under ‘Predefined Extension Names’.

voip-info.org/wiki-Asterisk+ … sions.conf

Good Luck,
Dan


#8

Thanks dmikusa.

Its working now!!!