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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
Server Stats
  • Execution time: 0.94s
  • Memory usage: 2.23MB
  • Database queries: 32
  • GZIP: Disabled
  • Server load: 6.41

Asterisk Step-by-step Installation

Introduction


This how-to guide outlines the process for a fresh installation of Redhat Linux (Fedora Core 2 Final) and Asterisk. The purpose of this document is to get you up and running (making and receiving phone calls) in about an hour. Experimenting with Asterisk, enabling more features and unlocking its potential is left up to you!

This configuration was created and tested on:

   DELL PowerEdge 400SC ($350)
   ---------------------------
   Intel Pentium 4 2.40Ghz
   256MB RAM
   Standard configuration, no extra hardware


Installing Redhat Fedora Core 2 Final


- Download Redhat Fedora Core 2 FINAL from http://fedora.redhat.com/download and burn the CD ISO images (you only need CDs 1 and 2)
- Insert the CD and reboot into setup
- Hit enter to start graphical setup
- Skip the media test
- After the graphical setup starts, click Next to continue
- Select a language and click Next
- Select a keyboard configuratoin and click Next
- If asked, select "Install Fedora Core" and click Next
- Select an installation type of Custom and click Next
- Select Automatically Partition and click Next
- If asked, select "Remove all partitions on this system". WARNING: This will erase ALL data on your computer!
- Select Yes to confirm removing all data on your computer.
- Click Next to accept default Disk Setup
- Click Next to accept default Boot Loader Configuration
- Click Next to accept default Network Configuration
- Select No Firewall for the Firewall Configuration and click Next. WARNING: We do not recommend connecting this test server directly to the Internet! This server is configured without a firewall for simplicity. You can enable the firewall later and make the necessary changes to keep Asterisk working.
- Click Next to accept default Additional Language Support
- Select a time zone and click Next
- Enter a root password and click Next
- For Package Group Selection, select ONLY the following and de-select the other options:

   * X Window System
   * GNOME Desktop
   * Editors
   * Graphical Internet
   * Text-based Internet
   * Development Tools

- Click Next to accept Package Group Selection
- Click Next to begin installation
- Click Continue
- Insert Disc 2 when prompted and click OK
- When CD installation is complete, click Reboot
- After rebooting, the post-installation process will begin
- Click Next to continue
- Accept the License Agreement and click Next
- Set the date and time and click Next
- Click Next to accept the default Display settings
- Create a User Account with a different username and password than the "root" user created earlier and click Next
- If asked, confirm your sound card and click Next
- Click Next to skip the Additional CDs section
- Click Next to Finish setup

Installing Asterisk


- Login to your server as the user you created during install
- Right-click on the background and select Open Terminal
- Type "su -" on the command line, then enter the "root" user password when prompted
- Run the following commands to download Asterisk:

   cd /usr/src
   export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot
   cvs login <--- This command will prompt for a password, use anoncvs
   cvs checkout asterisk

- This will download the latest version of Asterisk to your server. WARNING: This will download the very latest DEVELOPMENT version of Asterisk. It is NOT suitable for production use, just for testing!
- Run the following commands to compile Asterisk:

   cd /usr/src/asterisk
   make clean
   make
   make install
   make samples



  • Note: You may need to "make install" several times before it really works.


Configuring Asterisk


- Login to your server as user "root"
- Right-click on the background and select Open Terminal


- Run the following commands to backup your current/sample configurations:

   cd /etc/asterisk
   mv iax.conf iax.backup
   mv extensions.conf extensions.backup

- Run the following commands to download VoicePulse sample configurations:

   cd /etc/asterisk
   wget http://connect.voicepulse.com/samples/iax.sample
   wget http://connect.voicepulse.com/samples/extensions.sample

- Run the following commands to rename the sample configurations:

   cd /etc/asterisk
   mv iax.sample iax.conf
   mv extensions.sample extensions.conf

- Run the following commands to read and edit the VoicePulse Asterisk configurations:

   cd /etc/asterisk
   gedit iax.conf &
   gedit extensions.conf &

- In iax.conf, make the changes outlined in the QUICKSTART section of the sample file, save the file and close it.
- In extensions.conf make the changes outlined in the QUICKSTART section of the sample file, save the file and close it.

Test incoming & outgoing calls


- Start Asterisk on your test server by running:


   /usr/sbin/asterisk -vvvgc

- Run the following command to get the IP address of your Asterisk server:

   ifconfig

- Look for the value after "inet addr:" to determine the IP address
- Download "Dante's DIAX Software Phone" to your Windows PC
- Start DIAX
- Click on Config > Registration
- Enter the following information (this "user" is already created in the sample iax.conf you downloaded from VoicePulse):

   Alias: VoicePulse
   Server: the IP address of your server that you determined above
   Username: diax
   Password: diaxpassword
   Password: diaxpassword
   Register: checked

- Click Save
- Click OK
- Dial a non-VoicePulse phone number to test outgoing calls like 1-888-225-5322
- You should see something similar to the following scroll across your Asterisk terminal window:

    — Accepting AUTHENTICATED call from 192.168.1.100, requested format = 2, actual format = 2
    — Executing Dial("IAX2/diax@diax/3", "IAX2/MY_DEVICE_LOGIN:MY_DEVICE_PASSWORD@gwiaxt01.voicepulse.com/18882255322") in new stack
    — Call accepted by 66.234.228.160 (format GSM)

- Add a phone number to your VoicePulse Connect! account from the Phone Numbers menu in your Account Center.
- Dial the incoming VoicePulse Connect! phone number on your account from a non-VoicePulse phone.
- You should see something similar to the following scroll across your Asterisk terminal window:

    — Accepting AUTHENTICATED call from 66.234.228.170, requested format = 4, actual format = 4
    — Executing Playback("IAX2/voicepulse-in-01@66.234.228.170:4569/4", "beep") in new stack

- Dialing into your Asterisk server should read back your phone number to you and then read back any digits you dial.
- If incoming and outgoing calls work, your Asterisk setup is complete! See www.asterisk.org or www.voip-info.org for more details on customizing your Asterisk setup.


See Also



Created by ketanp, Last modification by Jay Phillips on Thu 26 of Oct, 2006 [03:02 UTC]

Comments Filter

Instructions need updating.

by Eric-Sebastien Lachance on Wednesday 04 of April, 2007 [03:29:24 UTC]
Please note that these exact insctructions will not work anymore - digium has removed their CVS server permanently.

Furthermore, ./configure on the 1.4.2 version of asterisk fails on this default install with "termcap support not found".
Edit

Re: Similar script required for local operation (ideally Mandrake distribution)

by Anonymous on Saturday 29 of January, 2005 [22:00:27 UTC]
I totally agree. A very "ground up" approach detailed configuration process would be awesome.

Asterisk on Gentoo

by Alexandru Thomae on Saturday 22 of January, 2005 [20:11:56 UTC]
Please note that on a Gentoo install (I don't know about other distro's), it will work only if the user is added in /etc/asterisk/sip.conf
Edit

The above sounds insecure to me...

by Anonymous on Friday 07 of January, 2005 [13:19:29 UTC]
Looking at the instructions above I notice the software is fetched and built as root, whch is using root without a good excude. This sounds like a bad move to me---instead do everything until make install as a normal user instead. I plan to go further and not run asterisk as special, unprviledged, user instead of root.

I would also note that a box doing SIP, etc in production is probably and exposed server, so maximum parnoia and minimum software are in order. This rules out at least X11 and compilers (and ideally anything else only required by nomal users, which should not exist).

Having said that I a system admin, so might be paranoid :-)
Edit

Similar script required for local operation (ideally Mandrake distribution)

by Anonymous on Tuesday 28 of December, 2004 [10:07:38 UTC]
This kind of page is extremely useful for people initially investigating the possibilities. I think many people would investigate further if a very simple "out of the box" configuration worked locally using Mandrake distribution. For example, sample files that allow KPhone to call a demo number (10000), and from there can be extended to make/relay Voip calls. I note that Mandrake 10.1 official does include asterisk rpm as an optional installation package, but it crashed on my machine and it isn't (totally) obvious how to setup KPhone as a local SIP client. I know GUI based systems are not appropriate for (high capacity) Asterisk systems, but surely for some initial dabbling this would be very effective and help promote the solution.javascript:insertAt('editpost','(:idea:)');

Adding user?

by andyj on Thursday 11 of November, 2004 [03:56:51 UTC]
Am I missing something? How are users added to asterisk?
In one of the config files I see where to define the extension and such but where does the voicemail store - I would assume under linux user but in the instructions I don't see any references to creating users?
Also how do you do this more dynamically like fwd or a service would?

I know too many questions - but right now now enough answers.

Thank you,
 Andy

Voicepulse IAX2 Changes

by qagwaai on Tuesday 19 of October, 2004 [19:38:17 UTC]
Looks like some of the changes affect the configuration of voicepulse for Asterisk. See this for a good example. Remember that the in-(devicelogin):(devicepassword) is not longer used. Use just the (devicelogin):(devicepassword)@gwiax-in-01.voicepulse.com for the registry. Also make sure to grab the rsa public key from here and put in /var/lib/asterisk/keys.
Edit

Thank you

by Anonymous on Saturday 09 of October, 2004 [05:35:23 UTC]
This worked perfect. Thank you for the easy guide.

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