Posts Tagged ‘voip’

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


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:

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: , ,

KESC deploys asterisk across hundreds of sites

November 25, 2008 4 comments

KESC — Karachi Electric Power Supply Company (Company Engaged in Generation, Transmission and Supply of Electric energy to Karachi  deploys asterisk across hundreds of sites. KESC has selected Emergen‘s Asterisk‘s Business Edition Solution for their telephony infrastructure and will be supporting  around 1500 phones.

About Emergen (quoting from their website):

Emergen Consulting is Pakistan’s premier Open Source Solutions provider. Established in December 2004 we have provided front and back end IT solutions to several major corporations. Our solutions provide the most Reliable, Secure, Scalable and cost effective alternatives to the expensive proprietary, license based enterprise wide IT solutions.

previously Emergen has successfully deployed asterisk at NADRA


Voiceone – a better GUI for asterisk

November 24, 2008 1 comment


If you are bored of using freepbx or trixbox interface, try out this source forge based opensource webbased asterisk management interface, i just gave it a try and it looks promising. interface is much cooler than freePBX, you can create outbound and inbound rulesets,seems much like firewall rules 😉

They have an online demo as well, so you can try out the application and its functions.

Current version is 0.7.1. Voiceone requires Asterisk with mysql support and Php5+Apache. Following is a list of features from their website.

  • Client/Server architecture based on web services
  • Relies on Asterisk Realtime Architecture (ARA) for database storage
  • Two different panels, Personal for users and Configurator for administrators
  • Extensions management
  • Fully customizable users profile, including Voicemail, Call Forwarding (“Follow Me“) and Do Not Disturb
  • Highly configurable rulesets for outbound and inbound calls
  • Static LCR (Least Cost Routing)
  • Supports VoIP providers (SIP and IAX) and traditional Telco carriers
  • Links remote offices via IAX with RSA public key encryption
  • Powerful IVR creation system
  • Queues management
  • Conference rooms handling
  • Sounds and Music On Hold management
  • Applications and macros editor
  • System Macros and Functions preloaded (DID/DDI, Call Back and DISA included)
  • Plugins system to share ready-to-use macros and application with the VoiceOne community
  • Powerful configuration of mISDN and Zap drivers based hardware
  • Java SIP phone embedded
  • I/O interface and PBX CLI (Command Line Interface)
  • Static-like text editor for conf files
Categories: asterisk Tags: , , , ,