Archive for the ‘voip’ Category

Save 90% on international calls and sms with Vopium

March 15, 2010 3 comments

If you are living abroad or frequently call your family and friends abroad you can save upto 90% on international calls and sms with Vopium, a Copenhagen – Denmark based company. VopiumVopium provides it’s users variety of simple ways to make international calls and sms via it’s web portal and mobile application, Vopium is a mobile phone application supported on variety of platforms (symbian, iphone, android, java).

Apart from international calls and sms plans, Vopium enables it’s users to talk and chat using Skype, Google Talk, Msn and Yahoo.

Currently Vopium is available in 18 countries including Australia, Belgium, Denmark, France, Finland, Germany, Netherlands, Italy, Norway, Spain, Switzerland, Sweden, UK & USA.

To find out more and get started with Vopium, visit


asterisk 1.2 with realtime ldap driver

December 28, 2008 13 comments

I have created an auto install script based on to install Asterisk 1.2 with ldap realtime driver support (modified original script to enable realtime ldap support).

This script should work almost with any(Redhat, Centos, Mandrake, Debian, Slackware) Linux distro

Before running this script you must install following packages.

For Centos, Redhat:

yum install openldap-devel gcc kernel-devel bison openssl-devel libtermcap-devel ncurses-devel

For Debian, Ubuntu:

apt-get install libldap2-dev build-essential linux-headers-`uname -r` libncurses5-dev libssl-dev

You can download the script from here, simply run the script to start installation process.

download Asterisk schema for OpenLdap here

Installer and installation process is not tested, if you find any problem please let me know.

#Asterisk Download page
#Asterisk Package
#Asterisk Folder
if [ -f /etc/redhat-release ] ; then
elif [ -f /etc/debian_version ] ; then
elif [ -f /etc/SUSE-release ] ; then
elif [ -f /etc/mandrake-release ] ; then
elif [ -f /etc/slackware-release ] ; then
elif [ -f /etc/gentoo-release ] ; then

echo "Downloading and extracting zaptel and asterisk source"
cd /usr/local/src/
if [ ! -e $ZAPPACKAGE ]; then
  wget $ZAPTEL
if [ ! -e $ASTPACKAGE ]; then
  wget $ASTERISK
if [ ! -e $ASTADDONSPACKAGE ]; then
tar -zvxf $ZAPPACKAGE
tar -zvxf $ASTPACKAGE

echo "Installing zaptel"
make clean
make install
cd ..

echo "Installing asterisk"
echo "Downloading patches"
patch -p0<resmkfile.patch
patch -p0<config.c.patch
echo "Downloading Realtime Ldap driver"
cd res
wget $RTLDAP
patch -p0<res_config_ldap.c.patch
cd ../configs
cd ..
make clean
make install
make samples
cp contrib/init.d/rc.$DISTRO.asterisk /etc/init.d/asterisk
cd ..

echo "Installing asterisk-addons"
make clean
make install

echo "Loading ztdummy driver"
modprobe zaptel
modprobe ztdummy

echo "adding rules to /etc/rc.local"

echo "modprobe zaptel
modprobe ztdummy
" >> /etc/rc.local

echo "Downloading open source g729 codec"
cd /usr/lib/asterisk/modules/

echo "Running Asterisk"
/etc/init.d/asterisk start

echo "***********************************************************************"
echo "*             INSTALLATION SUCCESSFUL                                 *"
echo "***********************************************************************"
echo "* You can test if Asterisk installed successfully using               *"
echo "* asterisk -ncrvvv and start configuring your dial plan               *"
echo "***********************************************************************"

freeswitch knocks asterisk’s block off

December 4, 2008 10 comments

Freeswitch is gaining popularity among asterisk community these days and looks quite promising, we also migrated our office PBX from asterisk to freeswitch few months back and never faced any issue till now. freeswitch is built from the scratch to address scalability and deadlocks issue within channels,  it’s modular architecture makes it developer friendly.

Freeswitch‘s SIP stack is much better than Asterisk’s sip implementation, Freeswitch uses Sofia sip stack which is 100% RFC compliant(IETF RFC3261 specification). Other than SIP freeswitch supports IAX2, Jingle and Woomera.

Another great thing about freeswitch is that it keeps all its configuration (users/dialplan etc, etc) in XML files. Another more interesting this is unlike asterisk’s dial patterns (e.g. _NXX) freeswitch uses PCRE regular expressions.

Freeswitch comes with an enterprise grade eventing engine, features software based conferences (no hardware timing source required), detailed cdr in XML(btw, i really loved the way freeswitch formats call flow in CDR, specially transfer calls).

Here is a list of Features and possible freeswitch uses:

Possible Uses

  • Rating & Routing Server
  • Transcoding B2BUA
  • IVR & Announcement Server
  • Conference Server
  • Voicemail Server
  • SBC (Session Border Controller)
  • Basic Topology Hiding Session Border Controller
  • Zaptel, Sangoma, Rhino, PIKA Hardware Support (Analog and PRI)


  • Centralized User/Domain Directory (directory.xml)
  • Nano Second CDR granularity
  • Call recording (In Stereo caller/callee left/right)
  • High Performance Multi-Threaded Core engine
  • Configuration via CURL to your http server (xml_curl).
  • XML Config files for easy parsing.
  • Protocol Agnostic
  • Configurable RFC2833 Payload type
  • Inband DTMF generation and detection.
  • Software based Conference (no hardware requirement)
  • Wideband Conferencing
  • Media / No Media modes
  • Proper ENUM/ISN dialing built in
  • Detailed CDR in XML
  • Radius CDR
  • Subscription server
    • Shared Line Appearances
    • Bridged Line Appearances
  • Enterprise/Carrier grade Eventing Engine. (XML Events, Name Value Events, Multicast Events)
  • Loadable File formats and streaming
  • Stream to Shoutcast
  • Multi-lingual Speech Phrase Interface
  • ASR/TTS support (native and via MRCP)
  • Basic IP/PBX features
  • Automated Attendant
  • Custom Ring Back Tones
  • XML RPC support
  • Multiple format CDR’s supported
  • SQL Engine provides session persistence
  • Thread Isolation
  • Parallel Hunting
  • Serial Hunting
  • Mozilla Public License
  • Support
    • Paid support available
    • Free support via IRC & e-mail
  • Many supported codecs
    • G.722 (wideband)
    • G.711
    • G.726 (16k,24k,32k,48k) AAL2 and RFC3551
    • G.723.1 (passthru)
    • G.729 (passthru)
    • AMR (passthru)
    • iLBC
    • speex (narrow and wideband)
    • lpc10
    • DVI4 (ADPCM) 8khz and 16khz

make calls from within gmail/gtalk

December 1, 2008 1 comment

Super-phone now enables you to make and recieve calls using your gmail/gtalk account, you can get a free trial US number upon sign-up.

details here:

asterisk realtime with ldap — got it working :)

November 28, 2008 4 comments

When we decided to write an asterisk module for Zivios, we went in several debates either we should keep asterisk’s configurations in plain config files or get benefit of asterisk realtime ldap driver.

if you don’t know much about Zivios’s architecture, check out Zivios Architecture page. In short Zivios keeps all of its configurations(Users/Groups/Config files) in OpenLdap as far as there is a possibility to use Ldap as config back-end.

Finally we decided to test asterisk realtime ldap driver, in first attempt everything worked fine, I was able to retrieve sip/iax peers from ldap tree,store registry updates in ldap, also created few dialplan extensions in ldap and it worked as expected. then all the hell broke loose when i added a pattern (say _5XX) in ldap and asterisk couldn’t find it in ldap, I tried to debug the realtime ldap driver, but nothing helped. lastly I decided to look into the realtime ldap driver’s C code(my C knowledge is worst), put several debug messages all over the file, recompiled and came out with the conclusion that there is something wrong with a function called *realtime_multi_ldap, posted bug report to digium with my logs (Faraz Khan posted a bounty as well, check it out), but all in vain, it felt like digium and nobody else doesn’t give a damn about realtime ldap driver, i decided to forget the realtime thing and keep configuration in plain text file, wrote some handy(helper) perl scripts to add/edit and delete configurations(users/routes/queues/ivrs). Zivios’s web interface communicates with these scripts using a special XML-RPC agent (called emsmagent) running on asterisk box. emsmagent executes these perl scripts and returns output to Zivios.

Last month browsing through digium’s bug tracker I found this *realtime_multi_ldap patch, I decided to give it a try, was not expecting it would work, but interestingly it worked and now I can use patterns in ldap exten entry and it really works :).

I have written a custom ldap schema for realtime ldap, You can also check original schema posted and contributed by suretec (guy who fixed the ldap driver) here. I will soon post a how-to on “setting up asterisk realtime ldap”. I am also going to port Zivios Asterisk module from plain text files to openldap backend soon (much busy in some other interesting projects these days :D)

let me know if you need any help in asterisk and realtime ldap driver setup.

one-half of the IT organizations worldwide are using VOIP

November 25, 2008 Leave a comment

Nearly half of the IT organisations surveyed in a global study by BT are using VoIP, the telecoms giant has announced.

More than 250 IT professionals took part in the study, which discovered that 48 per cent have a VoIP network at their work, an increase of 31 per cent from 2007.

A further 20 per cent are in the process of deploying the technology, with nearly three quarters (71 per cent) of these hopeful to have them working in two years.

In terms of criteria for choosing a specific solution, the most important were security, voice quality and network reliability.

‘Cost is a critical factor when building a VoIP business case, but other criteria must take higher precedence when evaluating various implementation strategies and solutions,’ the BT report said.

‘Make sure you understand what your top requirements are, be they network reliability, voice quality or security, before committing to a strategy.’

In terms of deployment, most are gradually replacing existing systems with VoIP.

The next most popular technique was to replace systems which were at the end of their lifespan with a new VoIP system.

Categories: voip Tags: , ,