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February 28, 2006

Xorcom TS-1 solid state Asterisk server released

TS-1 is an out-of-the box Asterisk based iPBX. Its 100% solid state architecture provides maximum reliability for the long run. Configuration and set-up are possible from distant locations via a Web management portal and do not require connection of keyboard and monitor. For more advanced features, the TS-1 offers SSH or direct (keyboard and monitor) access to the system menus.

 

Features:

* Powered by Asterisk
* Includes AMP configuration and maintenance via Web interface
* Runs Flash Operator Panel
* Runs Zaptel auto configuration system
* Includes one PCI slot for telephony cards
* Compatible with Xorcom Astribank telephony interface

Hardware Specifications:

* VIA C3 1GHz processor
* 256 MB DDR memory
* 512 MB flash memory
* VIA PCI 10/100 Ethernet
* 4 x USB type 2
* VIA S3G Unicrome video with DVI and SVGA
* Very low power consumption 36 Watts
* 100% solid state embedded unit - no moving parts

Click Here for More Information

February 15, 2006

Configuring voicemail.conf for Asterisk

I know I said last week we would tackle IAX client configuration, but I was at a customer site today and they called me down to reset a password and change a few names for voicemail. So I promised her to write my weekly piece on the subject so she and others have a reference for resetting voicemail boxes.

 

When you are done with this tip of the week, you should be able to add voicemail extensions, change names, change passwords and enable Asterisk voicemail as an e-mail attachment. Sending the voicemail as an e-mail attachment is a great way to recieve and archive e-mails. The extensions.conf and sip.conf are located on the previous two post. They are still on the blog so you may reference them.

1) Open a terminal window. If you need to access the server remotely Download an SSH (Secure Shell) client to access the Asterisk server. You can use Secure Shell from a vareity of Microsoft Windows clients freely available on the world wide web. If you have Linux or Mac OS X just read the man files from a terminal.

example:

[matt@localhost ~]$ man ssh

NAME
ssh - OpenSSH SSH client (remote login program)

SYNOPSIS
ssh [-1246AaCfgkMNnqsTtVvXxY] [-b bind_address] [-c cipher_spec]
[-D port] [-e escape_char] [-F configfile] [-i identity_file] [-L
port:host:hostport] [-l login_name] [-m mac_spec] [-o option]
[-p port] [-R port:host:hostport] [-S ctl] [user@]hostname [command]

DESCRIPTION ssh (SSH client) is a program for logging into a remote machine and for executing commands on a remote machine. It is intended to replace rlogin and rsh, and provide secure encrypted communications between two untrusted hosts over an insecure network. X11 connections and arbitrary TCP/IP ports can also be forwarded over the secure channel. ssh connects and logs into the specified hostname (with optional user name). The user must prove his/her identity to the remote machine using one of several methods depending on the protocol version used.

If command is specified, command is executed on the remote host instead of a login shell.

2) After we log in succesfully, go to the /etc/asterisk directory. Oh by the way, if you are not logged in as root use the 'su' command to become the 'root' user. The 'root' user account is the account for administrators.

example:

[matt@localhost ~]$ cd /etc/asterisk

[matt@localhost asterisk]$ su
Password:

[root@localhost asterisk]#


Notice now that it says [root@localhost asterisk] instead of [matt@localhost asterisk]$.

3) Now it's time to edit the voicemail.conf which is located in the directory we just changed to '/etc/asterisk'. I'm using nano to edit the file, but pico, vi, emacs, or any text editor will do.

example:
[root@localhost asterisk]$ nano voicemail.conf

[general]

format=wav
attach=yes

9250 => 1000,matt,mattb@voiceipsolutions.com
9251 => 1000,some one,some1@voiceipsolutions.com

The voicemail.conf file is very easy to read. The first four digitits represent the extension number (as defined in extensions.conf). So my desk extension from inside the office is 9250. The second string of numbers is the password. The third is the name of the person whom owns this paticular box. In this case it's Matt, which is me. The fourth part is the e-mail address that asterisk will send a copy of the voicemail to.

So if for example user 1 forgot his password, I could follow these steps and change the password to 0000. User 1 would then dial intoo his password and change it from his phone to whatever he wanted.

example:
[general]

format=wav
attach=yes

9250 => 1000,matt,mattb@voiceipsolutions.com
9251 => 0000,some one,some1@voiceipsolutions.com

Or, if for example if I fired myself and VoiceIP Solutions hired an imaginary person called Lawson. The voicemail.conf would look like the example below. Keep in mind I'm changing the password to a default of '0000'. You can make the default whatever you like. I also want him to recieve copies of his voicemail as an attachment to his e-mail box.

example:
[general]

format=wav <------- This is value determines format for e-mail attachments I like
GSM or WAV
attach=yes <------- this value determines whether asterisk sends out voicemails as e-
mail atttachments

9250 => 0000,Law,law@voiceipsolutions.com
9251 => 0000,some one,some1else@voiceipsolutions.com

NOTE: you don't want my notes typed in the voicemail.conf.

Once you made your changes save voicemail.conf.

4) Now we have to reload Asterisks config files. I'm assuming asterisk has been running this whole time. There is no need to shut the server down for minor updates. When asterisk command line comes up type 'reload'.

example:

[root@localhost asterisk]# asterisk -r
Asterisk CVS-v1-0-10/13/05-13:41:20, Copyright (C) 1999-2004 Digium.
Written by Mark Spencer
-------
Connected to Asterisk CVS-v1-0-10/13/05-13:41:20 currently running on localhost (pid = 21007)
localhost*CLI>reload



After Asterisk reloads the configs type 'exit' and close your SSH terminal. Thats it. Not much too it to change basic voicemail options. Next week we'll get to trickier IAX tunnels or clients.

Note: My name is Matt Birkland, I work as a VoIP Engineer for VoiceIP Solutions an Asterisk Provider in Washington State. Every Monday I will be submitting a one page Asterisk/VoIP tip of the week on the blog. This week we will discuss voicemail.conf configuration and walk through the dial plan in this regard.

February 14, 2006

Nerd Vittles: Introducing TeleYapper 2.5: The Free Asterisk Message Broadcasting System

Excerpt: "TeleYapper 2.5 is an updated version of our Asterisk@Home-based Telephone Broadcasting System that actually works with Asterisk@Home 2.5 (Asterisk 1.2.4 for "purists"). And, just like the original, TeleYapper 2.5 can be used for announcements and reminders by neighborhood associations, schools, little leagues, fundraisers, municipal governments, and anyone else that just wants to pester folks with annoying, but free, prerecorded phone calls."

Click Here for the Full Nerd

February 07, 2006

Nerd Vittles: Installing Asterisk@Home on Your Windows PC for Free

Excerpt: If you've always wanted to try the Asterisk PBX but didn't have a spare machine for Linux or the expertise to wade into the bowels of Asterisk, your ship has arrived. Asterisk@Home now can be run as a virtual Windows application. Add the VMware Player, the Asterisk@Home 2.5 VMware app, mix, and serve. Presto! Your new Asterisk@Home server is now running in Microsoft Windows. And it's all still FREE!

Click Here for the Full Nerd

February 01, 2006

Skype-to-Asterisk(SIP): Progress - Part 2

Note: Here is the second part of John's SIP to Asterisk Progress. This is very exciting stuff.

So, we have liftoff.

I have received the updated code from the programmer, and it does work as requested. I can now make calls (via SIP) to a Windows system running Skype and the revised PSGW software, and that system then turns the calls around and uses the SIP destination as the Skype username, and the call is terminated on the Skype network to the appropriate user. I've tested to various users and the echo123 test, and it sounds great! So I have the first SIP-to-Skype end-user specifiable gateway running. To avoid being Slashdotted, I won't post my test SIP destinations publicly, and of course the system can only handle one Skype conversation at a time.

 

Plans for the system: it will become (among other things) a find-me, follow-me type extension in our Asterisk diaplan - not only will the system ring your desk phone and cell phone, but it'll also try your Skype account.

Downsides: there is no caller ID, at least not as part of the signalling. All the calls to the Skype destinations get whatever account information is associated with the Skype account that is running on the Windows system. Of course, other methods could be used here, such as text-to-speech pre-call completions, DTMF (?? not sure how that works with Skype), or semi-out-of-band instant messaging exchanges. None of these methods are implemented yet, but it appears that they are being developed.

I've asked the programmer if he plans to release the revised code as his next update to PSGW.

JT
http://www.loligo.com/asterisk/


Click Here to Read Skype to Asterisk Part 1

 

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