Digium has announced the latest member of the Switchvox appliance family, the Digium|Switchvox AA300 appliance to go along with the AA60 and AA350 models. The AA300 supports up to 150 users and is targeting mid-sized companies. The AA300 model size is 3U half-deph and it can be rackmounted. In addition to 150 users, it supports up to 45 simultaneous calls. It also supports up to 10 recorded calls and up to 15 simultaneous conference calls.

Switchvox AA300 highlighted features:

•  Full coverage by the Digium Exceptional Satisfaction Program, the strongest guarantee in open source telephony, which gives customers their money back if the product is defective or fails to perform as described.
•  VoIP ready—No extra hardware is necessary to connect to SIP or IAX voice over IP trunks.
•  A standard one-year warranty and optional three-year warranty.
•  Support for analog and T1/E1 interfaces gives customers flexibility in working with their existing network.
•  A choice of three subscription plans to suit every company’s support and service needs.

Pricing for the Digium Switchvox AA300 with SMB 3.5 software is $4,240.View the complete announcement here.

How would you like your voice or the voice of the person over the phone to sound like a chipmunk?

Lobstertech Technologies has developed Asterisk Voice Changer which allows you to change the pitch of your voice, or the voice of the called party in realtime, when making phone calls over Asterisk based systems.

The application, VoiceChangeDial, functions as a Dial() replacement except it allows you to change the pitch of your voice. Although the behavior of this application seeks to mimic Dial(), it is not perfect due to the fact that Asterisk does not properly abstract dial and bridging functionality.

Installation Procedure

On your functioning Linux machine running Asterisk with the header files installed to /usr/include/asterisk, run the following commands to install the voice changer as well as its two dependencies:

# install SoundTouch 1.3.1
# you can also install it through your package manager
cd /usr/src
wget http://www.surina.net/soundtouch/soundtouch-1.3.1.tar.gz
tar xvzf soundtouch-1.3.1.tar.gz
cd soundtouch-1.3.1/
./configure –prefix=/usr
make
make install

# install libsoundtouch4c
cd /usr/src
wget http://www.lobstertech.com/code/libsoundtouch4c/releases/libsoundtouch4c-0.4.tar.gz
tar -xzvf libsoundtouch4c-0.4.tar.gz
cd libsoundtouch4c-0.4
./configure –prefix=/usr
make
make install

# install the voice changer
cd /usr/src
wget http://www.lobstertech.com/code/voicechanger/releases/voicechanger-0.6.tar.gz
tar -xzvf voicechanger-0.6.tar.gz
cd voicechanger-0.6
make
make install

# load it in to asterisk
make start

Have fun!

I have been searching the web for an asterisk behind NAT configuration but couldn’t find a short but definite example so I decided to create a working example configuration of asterisk behind NAT. So if you guys are currently having problems configuring your asterisk behind NAT, please feel free to use my example below:

Put the following in your rc.local:
iptables -F
iptables -t nat -F
iptables -t mangle -F
iptables -t nat -A POSTROUTING -o eth0 -j SNAT –to your.public.ip.here
echo 1 > /proc/sys/net/ipv4/ip_forward
iptables -t nat -A PREROUTING -p udp –dport 10000:20000 -j DNAT –to-destination 192.168.30.1
iptables -t nat -A PREROUTING -p udp –dport 5060 -j DNAT –to-destination 192.168.30.1
iptables -A FORWARD -p udp -s 192.168.30.1 -j ACCEPT
iptables -A FORWARD -p udp –dport 10000:20000 -d 192.168.30.1 -j ACCEPT
iptables -A FORWARD -p udp –dport 5060 -d 192.168.30.1 -j ACCEPT

your sip.conf should be:
[general]
context=default
port = 5060
bindaddr = 0.0.0.0
context = default
externip = your.public.ip.here
nat=yes
localnet=192.168.30.0/255.255.255.0
canreinvite=no

Sample gateway-to-gateway SIP config for sip.conf:
[toyoursipprovider]
type=friend
host= sip.provider.gateway.ip
canreinvite=no
disallow=all
allow=g729
allow=ulaw
dtmfmode=rfc2833

ENJOY!

Dialogic announced that their Diva Media Processing Boards will support Asterisk. Because most of the Diva Media Boards have powerful Digital Signal Processors (DSPs) onboard and can move processor-intensive, low-level tasks to these DSPs, significant reductions in server CPU load are possible especially when used in an Asterisk based systems.

With the release of the chan_dialogicdiva channel driver, Dialogic is now offering active support for the Asterisk environment through its line of Diva Media Boards, along with an associated Starter Kit program.

Other functions handled by the Diva Media Boards include DTMF recognition/generation, pulse detection, tone detection/generation, fax detection, echo cancellation, and multi-party conferencing. The Diva Media Boards can make conference up to 120 active callers without causing load on the CPU. Some of the Diva Media boards’ signaling features include worldwide support of protocols and services, including most Q-SIG flavors and Signaling System 7.

SippySkype is a FREE Java software (Skype/SIP Bridge/Gateway/Proxy) which allows you to make and receive calls using your SIP/VoIP adapter.

Here are some features of the software:

• Call Skype Users using speed dial or use Skype out.
• SIP to Skype authentication/denial mappings via SIP caller ID and IP blocks - 1.1 or higher
• Skype to SIP authentication/denial mappings via incoming Skype User ID - 1.1 or higher
• Support RFC2833 touchtone decoding (DTMF) - 2.0beta or higher
• Could be used as an endpoint with Asterisk
• Auto play pre-recorded file(s) to SIP callers - 2.0beta or higher
• Incoming SIP Pin number authentication and dialing - 2.0beta or higher
• Open Source - You can modify/fix it if you like.

System requirements:

• Skype Client
• Working Java 1.6.0 or better runtime
• mjsip/mjua 1.6 http://www.mjsip.org/ - Use those included with SippySkype as some bugs have been fixed.
• Skype4Java 1.0 https://developer.skype.com/wiki/Java_API - Unmodified
• SIP/VOIP adapter such as a spa-3102 to make and receive Skype calls or register with a provider or Asterisk.
• Should work where Skype4Java works (windows/linux/osx). (I’m using it on Windows XP)

Download SippySkype 2.0 beta2

The Asterisk.org development team has released Asterisk 1.6.0-beta5. According to the announcement, the feature-set is frozen with beta5 of Asterisk version 1.6.0.

However, what’s missing in this version is the screening of “caller name” which allows you to screen first who is calling before you accept it.

In addition to a number of bug fixes, the following new features have been added since beta4:

• The SMDI interface in Asterisk has been reworked to fix a number of issues as well as add some new features. SMDI message information is now accessed in the dialplan using some new dialplan functions. New options have been added to map Asterisk voicemail boxes to SMDI station IDs. Also, MWI will now properly be sent for systems that have some external interface modifying voicemail boxes, such as a web interface, or with an email client in the case of IMAP storage.
• The Postgres CDR module now supports some of the features of cdr_adaptive_odbc. Specifically, you may add additional columns into the table and they will be set, if you set the corresponding CDR variable name. Also, if you omit columns in your database table, those fields will be silently skipped when inserting the record.
• The ResetCDR application now has an ‘e’ option that re-enables the CDR if it has been disabled using the NoCDR option.
• A new CLI command, “devstate change”, has been added which allows you to change the state of a Custom device. Custom device states were previously only settable by using the DEVICE_STATE() dialplan function.
• The Originate manager action now has its own permission level called originate. Also, if you want this action to be able to execute applications that call out to a subshell, it requires the system privilege, as well. These changes were made to enhance the security of the manager interface.

See the full list of features that have been introduced from Asterisk 1.4 to Asterisk 1.6.0.

For a full list of changes to Asterisk 1.6.0 from beta4 to beta5, see the ChangeLog.