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Zaptel 1.2.4 Released

The Asterisk/Zaptel development team is pleased to announce the release of Zaptel 1.2.4.

This release contains a number of bug fixes, along some with new functionality:

* The driver for the Xorcom Astribank has been incorporated into this distribution. Xorcom will provide primary support and driver maintenance for customers using this product.

* The driver for the Digium Wildcard TDM2400P has been upgraded to support revision B of the VPM100M echo cancellation module.

* The special parameters required for the Digium Wildcard TDM400P when used on the Australian PSTN are now automatically set when the opermode is set to 'AUSTRALIA'.

The release is available on the Digium FTP Servers under the name zaptel-1.2.4.tar.gz, and also as a patch from version 1.2.3 (in file zaptel-1.2.4-patch.gz).

In addition, beginning with this release we have included an SHA-1 sum of the files (in files zaptel-1.2.4.tar.gz.sum and zaptel-1.2.4-patch.gz.sum) and GPG signatures (in files zaptel-1.2.4.tar.gz.sign and zaptel-1.2.4-patch.gz.sign) verifying that
this is an official Zaptel release.

You can retrieve the public keys for kpfleming@digium.com and russell@digium.com from the keyserver, pgp.mit.edu.

Next Montreal Asterisk Meeting - 02/21/2006 - Featuring a conference call with Mark Spencer

This is a reminder about our next meeting.



It will be held from 6pm to 8pm, February 21 at Modulis Offices which are at 360 Notre Dame ouest bureau 104, H2Y1T9, Old Montreal.



Thanks to Claude Patry, we will be having a 20 minute conference call with Mark Spencer.



If you'd like to ask Mark a question, please send it to me by email. We are limited to 5 questions, and will do our best to select those to be presented.



Please confirm your attendance at this meeting by replying to this email.



See you Next Week,





--

Adrien Laurent

adrien@modulis.ca

www.modulis.ca

(514) 284-2020 x 202

TeliaSonera First European Operator to Trial Wireless IP Telephony for Enterprise Customers with Nokia and Cisco

3GSM World Congress, BARCELONA, February 15, 2006 - In co-operation with Nokia and Cisco Systems Inc., TeliaSonera will be the first operator in Europe to trial wireless IP telephony via WLAN for enterprise customers. The trial is one of TeliaSonera’s numerous initiatives to take the lead in bringing new services to customers migrating from fixed to mobile and Internet-based services. Recent surveys on IT investment in Nordic enterprises shows that IP telephony, i e the convergence …

PIKA Technologies Announces Support for Asterisk PBX



PIKA Technologies today announced that they have integrated PIKA's high-density analog computer plug-in boards with the open source Asterisk PBX, with the introduction of PIKA Connect for Asterisk. PIKA Connect for Asterisk is a software
layer, available free of charge and distributed under the GNU Public License (GPL), which allows interoperability between PIKA high-density analog boards (Daytona MM) and Asterisk PBX software.

"The Asterisk development community can now benefit from advanced features for fax and echo cancellation in high density analog applications, made possible by PIKA's DSP processing power on the board," stated Wojciech Tryc, Enterprise VoIP architect at PIKA Technologies. "Because of the native bridging for TDM calls, latency is nearly eliminated in this implementation. The solution is very reliable, as we have witnessed not only in the lab, but in live customer environments."

Asterisk developers can be up and running quickly with PIKA Connect for Asterisk and PIKA hardware. "For those familiar with using the Asterisk platform, no additional training is required. They can take advantage of the PIKA solution with minimal effort or investment," said PIKA Technologies, Wojciech Tryc.

For more information on PIKA Connect for Asterisk go to:
Pika Technologies Asterisk Info

Help Article: 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.

Cisco Leads Enterprise & Small-Medium Business Voice Market for All of 2005

SAN JOSE, Calif., February 15, 2006 - Cisco Systems expanded its leadership position in the global voice market for large, small and medium-sized enterprises for the 2005 calendar year, based on the most recent report by the Synergy Research Group. The report names Cisco as the revenue leader in the worldwide enterprise voice market, including both Internet Protocol (IP) and traditional circuit-based systems, for both the fourth quarter of 2005 and for the entire calendar year. …

RSA: Network security is the key to keeping VoIP secure

Analysis of IP voice components is key to keeping VoIP networks secure.

Firmware version 1.3.1 released for Aastra IP Phones

Aastra Telecom has released SIP v1.3.1 firmware for the Aastra range of IP phones (480i, 480iCT, 9112i and 9133i).



The firmware and release notes (no updated admin and user guides yet) are available for download at:

http://www.aastra.com/support/enterpriseip



Contrary to what the version numbering would suggest, this is a significant update with many new features and bug fixes. See the release notes for full details, but here are some hightlights for Asterisk users:



- Context-sensitive softkeys. Softkeys can now be configured for each of the following call states: idle, incoming, outgoing and connected

- Speed dial using the BLF key

- Per-line outbound proxy

- Use the Icom key to make intercom calls

- Further XML enhancements

- Voice quality (transmit level) issues resolved

- Keypad now continues to work when a second incoming call appears



And much more.

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

Nortel ships Converged Office

Nortel Networks announced general availability of a Nortel IP voice switch integrated with Microsoft Office Live Communication Server.

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