Asterisk SCF: Make Open Source Cloud Telephony Happen

It’s been two months since Asterisk SCF was announced in AstriCon 2010. It really brings a lot of expectation to the open source community and it’s worth our waiting. Asterisk SCF is still in heavy development and will not be expected to be beta-level software until some time in 2011. But are you ready when it hits the ground? If you would like to start developing on Asterisk SCF, please Visit the Asterisk SCF wiki  to find information on downloading and building the Asterisk SCF source, documentation on the architecture, and see the Roadmap.

Asterisk SCF is a framework that allows developers to create real-time communications applications that include voice, video and text and that meet the demands of a full range of uses, from embedded applications to enterprise and carrier solutions. Asterisk SCF is architected to provide the highest levels of availability, scalability, extensibility, fault-tolerance and performance.

Asterisk SCF will be delivered as a system of distributed components that can be deployed in clusters on a single system or on many systems, transparently. The Asterisk SCF platform will support, as a part of its basic architecture, the full range of real-time IP communications, including video, multi-channel wideband and ultra-wideband audio, chat, desktop sharing and other media types that may arise in the future.

Asterisk SCF is not a replacement for Asterisk, the world’s most widely used open source voice communications platform. Digium and the Asterisk community are committed to the continued development and support of Asterisk, the telecommunications software.

asterisk · Asterisk SCF · AstriCon 2010 · digium · embedded application · open source

<< More Than Two Million Downloads of Asterisk in 2010

The First Open Source Telephony Conference in 2011 >>

More Than Two Million Downloads of Asterisk in 2010

Before turning off my laptop, an exciting new announcement came in my sight: Asterisk had more than 2 million downloads in 2010. That is to say about an average of 5500 downloads per day. And this is the growing power of the world’s most welcome open source PBX software! I would love to share the news with everyone of you.

Digium caps off a year marked by strong growth in the use of Asterisk and substantial technical advances in the product. Contributions from the open source community matched Digium’s investment in Asterisk over the past year. To date, more than 9,800 people have contributed code to Asterisk, including more than 200 who worked on Asterisk 1.8. After releasing version 1.8 in October, the momentum continued later that month when the company also announced a new open source project, Asterisk SCF™. Enthusiasm for Asterisk among users, including developers, resellers, integrators and systems administrators, also increased as they downloaded the software more than two million times in 2010.

Asterisk turns an ordinary computer into a communications server that can power IP PBX systems, VoIP gateways, conference servers and other communication applications. In over 170 countries today, small businesses, large enterprises, call centers, carriers and governments are using Asterisk to create standards-based, feature-rich communications systems at a fraction of the cost of proprietary systems. Digium estimates that more than one million servers around the world are currently running Asterisk to handle billions of minutes of phone calls.

“Asterisk has made an indelible impression on the voice communications industry in the 11 years since it was released,” said Bryan Johns, Digium’s community director. “Its appeal keeps growing as businesses look for the value, flexibility, standards compliance and the technical superiority that result from the contributions of thousands of talented and visionary software developers. Digium is proud to sponsor Asterisk and to be a part of its community, which now counts 73,000 registered members. The company continues to focus on Asterisk’s development, as we saw with the release of Asterisk 1.8 this fall and with the creation of a new open source project in Asterisk SCF.”

Well, enjoy it!

asterisk · asterisk downloads · Asterisk SCF · call centre · conference server · digium · IPPBX · open source · VoIP

<< Community News: Elastix Launches Security Module

Asterisk SCF: Make Open Source Cloud Telephony Happen >>

Asterisk consultants worldwide remote

The following consultants provide remote Asterisk services to customers worldwide. They may also provide onsite services; see the onsite page for details.

Add your entry here (Alphabetical order by company):

Evariste Systems Freevoice Telecom U.S.A. Fortuitous Technologies FluentStream Technologies Generation Software Solutions Godson Gera - Hyderabad - India GreenfieldTech CTS Consulting & Trade Service GmbH HomeUnix HireRussians @ sibers INTA Integrics Ltd Chennai, India People Tech systems private Limited INTELLECT Information Technology Pty Ltd Intuitive Voice Technology Jared Smith, Asterisk Consultant IP COMMUNICATION doo iPhonica LLC. GEDAM Europe SRL, Torino hagel IT-Services GmbH, Hamburg ITConnection iTFUTURE Tecnologia e Informação Ltda. (São Paulo-SP / Brasil) Mathias Kettner MikeOverIP Modulo Consulting SRL monoploid Linksoft Technologies Ltd (Hong Kong) Moki Systems Nasoni & Friends NDSoftware Nethawk (Pvt) Ltd NextGenData Next I.X., inc. Su socio tecnológico Oleg Titov, Asterisk Consultant Lintel Technologies Pvt Ltd Pbxnw.com PCDOCTOR - VoIP Division Progressive Voice Services Limited http://www.raynettech.com/ SineApps SIP:XX Communications - "Double-Xtra-Strength SIP!" SMB Consulting www.stuartelvish.com SviluppoIT The Sjobeck Company LLC Synaptic Pty Ltd The Uptime Group, Inc. Ulterius Pty Ltd VoIP Reach Communications Voiptech, Italy Voiscout, Germany, Hamburg Warrior Hill, LLC Italy

ASTNEXT SOLUTIONS PVT LTD
ASTERISK CONSULTANTS(GUJARAT,BANGALORE,INDIA)

AstNext was founded with intention to market Open source culture which is the creative

practice of appropriation
and free sharing of found and created content.

AstNext is able to provide expert guidance and technical support as we are focused on Open

Source IP PBX based telephony solutions.
Our affordable options can reduce monthly phone expenses,
offer you the flexibility to work from home or create a professional phone network in your office,
or even allow you to connect multiple offices with one system and a common set of phone

numbers and extensions.
Our team is dedicated to high-quality service.
We work to ensure that your applications have the greatest impact for your target audience.

We are Asterisk service provider, we are leaders in providing turnkey solutions in Asterisk

  * Asterisk: IVR/IP PBX/SOFT SWITCH 
  * Asterisk Gateway Interface (AGI) 
  * End to End VOIP Set-up 
  * VOIP Minutes 
  * Predictive Dialer 
  * Call Centre Solutions 
  * Hospitality Telecom solutions. 
  * Medical Telecom solutions. 
  * ACD for PRI/Analog/GSM Phones 
  * Softphone 
  * SER/billing soft switch 
  * LCR/CDR 
  * Web to call 
  * Click to call 
  * Inter-office/Intra-office Intercom 
  * IP Phones 
  * IP PBX,Installation,maintenance,configuration of linux systems,servers anddevices. 
  * Support for digium / sangoma / rhino E1 / PRI / FXO Digital / analog Telephony Cards 

/grandstream
Regards,
AstNext Team
Email: info@astnext.com
www.astnext.com
US: 209-349-9648
Cell: +91 8000770962

Having experience in asterisk and VoiP for more then 5 years
setup loadbalanced asterisk for more then 2000 users
setup many asterisk installation for Companies, call shops and contact centers
have done many Small to medium single and distributed load balanced Vicidial Installations
Of more than 300+, Seats plan done.
Providing support to 25 call centers at different locations in India
Provide remote and onsite Support with turnkey solutions
Provides installation and support for:
Asterisk Asterisk + vicidial Asterisk +video (GNU gk/Open MCU) gatekeeprs SER fax solutions fax broadcasting Voice Brooadcasting IVR solutions Auto Dialers sip gateway A2billing asterisk + trixbox gnu dialer Asterisk integration
Also provide support for existing vicidial, asterisk and trixbox setups any were in world
And do the configuration both onsite and remote configuration
Contact: aabhas bajpai
E-mail : aabhas.bajpai@gmail.com
Phone : +919160439711,+919936283432,
web : http://voip-info.org/users/view/abhasbajpai
India.
aabhas bajpai Having Experiance in Asterisk and Voip for more than 2 years
succesfully setup 200+ seats contact centre for various Activities(Sales, Collection etc)
Provides Remote and onsite support
Provides installation and support for:
   *  Asterisk Installation
   * Asterisk plus vicidial Installation
   * Asterisk integration
   * Elastix
   * Auto Dialers
   * Vicidial
   * Voice Broadcast
   * Voice/Video conferencing setup 

Also provide support for existing vicidial, asterisk setups anywhere in the world
And do the configuration both onsite and remote configuration
Contact: Ajay kumar
E-mail : aty2570@gmail.com
Phone : +919869559250
Web : http://www.voip-info.org/users/view/aty2570
India.

Web site) http://www.vontrade.com Phone) 416-884-2282, E-mail: ali@vontrade.com URL) sip:99999@it.vontrade.com Asterisk installation and configuration locally or remotely on Windows or Linux Pre-paid and Post-paid, Billing, Call rating, Least Cost Routing Voicemail, Fax to mail, push to talk, SMS, DID, IP phone provisioning ASR, ACD, PDD, Interactive call monitoring and online call control Database driven IVR and Configuration, web portal configuration SIP development specifically Soft phone. Script programming, AGI Development, Perl and PHP Turnkey VOIP solution for Retail and wholesale business
Based in Oman and located in more than 10 centres all over the middle east and possess more than 37 years of industry experience.
Services offered are:
Installation of Asterisk PBX with PSTN, GSM and FAX integration Installation of switches with trafffic management and Wifi Networks with wireless IP phones Turnkey solutions & Integration of asterisk with existing network Call center solutions Billing and CRM solutions Structured cabling and supply of all passive and active networking components including PC's and peripherals.
Phone:+968 24816303
Email: it@alkhalili.com
Web: www.alkhalili.com
http://www.asteriasgi.com sales@asterisksgi.com Phone Toll-free: 877-ASGi-4-ME (877-274-4462) ... or Local in Huntsville, AL: 256-705-0277 Digium | Asterisk Premier Solution Provider and Partner Design, Configuration, and Installation of Asterisk PBX Systems Deployment, Configuration, and 24-Hour support of our OmegaDial(tm) auto-dialer platform Integration Asterisk Consulting is our Specialty! Our products include: A Call Center Agent GUI, an Asterisk Queue Management Software, an Asterisk Autodialer, and a Graphical Asterisk Configurator and IVR Builder Flexible Payment Scheduling to meet your financial needs. We are based in Huntsville, AL, but service the entire World! Call today to find out what we can do for you!
http://www.AsteriskAgent.com/ Worldwide Support support@AsteriskAgent.com 800-763-2908 Specializing in Asterisk Based solutions. Asterisk Support & Staffing Live Website Support Available
Asterisk@Office is a leader in software applications and system used for IP-Telephony. Our consulting services help your organization connect its existing telephone lines with a state-of-the-art hybrid VOIP environment where inter-office, inter-branch and customer calls can take advantage of Web based voicemail, Conferencing, Automatic Call Distribution and many other advanced features.
Asterisk Design, Installation and Maintenance Vicidial,a2billing,Click2call nad many more Failover solutions , interconnecting multiple Asterisk servers Installation and Configuration support on asterisk,vicidial,a2billing,openser, We work on Hourly basis
Most affordable Price !! Visit us at http://www.aasteriskneeds.com
Highly professional support for home and office PBX systems based on asterisk. Expertise in Trixbox, Asterisk, FreePBX, Hylafax, ATA, OpenSER, FreeRadius Asterisk module development and source code customizations Custom application development based on asterisk IVR systems Billing Systems OpenSource Consulting Fax system
E-mail :info@asteriskware.com
web : http://www.asteriskware.com Contact us to discuss requirements We offer various services for Asterisk such as Asterisk installation, Asterisk configuration, Dialplan programming, IVR Designing, AGI scripting, manager api, Customized Asterisk Development, text to speech, speech to text, etc.

+1 888 505 VoIP | +1 212 937 8855 Asterisk Consulting Computer Telephony Integration Custom IVR's Advanced Asterisk Applications Trix Box Support Asterisk Support PBX Planning & Network Design VOIP Phones and Hardware Linux Administraton VOIP Server Hosting and Colocation
Website:http://www.astravoice.com
Phone: +1 212 937 8855 - +1 888 505 VoIP
E-Mail:info@astravoice.com We provide services for your all Asterisk projects, installation,
integration, custom development and online Asterisk support.

In English, En Français, Türkçe

Provides installation and support for:
Asterisk Asterisk + vicidial IPBX configuration 1 to 5000 extensions OpenSER sip gateway A2billing asterisk + trixbox ViciDial (Call Center) Asterisk integration Calling Card (open source) Call Shop (open source)
Contact: Ahmet Yilmaz
E-mail :info@atyconsulting.com
Phone : +33(0) 954 974134
web : http://www.atyconsulting.com
France
Ahmet Yilmaz We are specialized on Asterisk, and GNU/Linux, we offer consulting, remote assistance, monitoring and personalizations.
Feel free to contact us.
         o Location: Padua, Italy
         o Tel.: +39 049 796 72 72
         o E-Mail: info@beffect.net Asterisk Consultation Asterisk / Windows Network Intergration Turn Key Asterisk VOIP Systems VOIP Phones and Hardware Linux Administraton VOIP Telephone Service Provider National Internet Provider VOIP Server Hosting and Colocation Hosted VOIP Provder
Website:http://www.BIGNetworks.com
Phone: +1 225 214 5355
E-Mail:voip@bignetworks.com We not only install complete asterisk systems we also integrates customs solutions for each business with our vast experience
Working with asterisk and programming languages like pearl and .NET allow us to accomplish any need you may have
We go to a simple asterisk installation or web page until solutions like our own predictive dialer, CTI platforms, CRM integration
And Telco support.

We offer remote and on side support worldwide.

AgileCO.net offers Advanced Integration, Development and implementation for OpenSER/Asterisk/OpenWengo. Our professional services include the following:
SIP, Skype, GoogleTalk, 3GP-IMS, P2P-SIP, VoiceXML, Mobicents; Mobile-SMS, J2ME, ParlaX and XML WebServices SOAi.

Operate in Europe; Asia and Americas
   * Web site: http://agileco.net
   * E-mail: info at agileco.net
   * Phone: +16789357099
   * skype: edpimentl
   * Professional Services
         * OpenSER/Asterisk Installations
         * SIP/Skype/Mobile-SMS/J2ME Software Development
         * Project Management
         * Project Outsourcing 
   * IVR/VoiceXML Development
   * Custom AGI Development
   * Support Contracts 

Considering deployment of an Asterisk solution? We can provide design, installation, configuration, maintenance and support of single and multi-site configurations. We bring experience of consumer retail solutions such as calling card and low cost call solutions, call centre deployments with CRM integration, and integration with low cost VOIP suppliers.

Our most recent project is a low cost conference call service for the UK: http://www.ConferenceGenie.co.uk

Providing Experienced Asterisk deployment and customization. Small to large scale implementations including integration into enterprise Java, .NET, Databases and more. CallFire provides highly-scalable and affordable Cloud Telephony and Homesourcing solutions, including Voice API ,Voice Broadcast , Virtual Call Center , Hosted IVR , Call Tracking, Phone API and VoiceXML services.


Bitnetix Technology Consulting is a privately owned business dedicated to providing expert advice in all areas of technology, IT, computers, and telephony to credit unions, not-for-profit organizations, and small businesses. Located in Western New York, we offer a wide range of services to local clients as well as providing off-site assistance and travel to remote locations and other geographic areas.
VOIP Asterisk FreePBX Lots more Visit www.bitnetix.com for information on how to contact us, or give us a call at 1.877.33.VOICE (877-338-6423) Web site: http://callforsolutions.com/ E-mail: info AT callforsolutions dot COM Phone: +1 (514) 667 - 0716 Professional Services Asterisk Installations And others services With over a decade of experience in the computer field we can provide a variety of services such as installation, configuration and support of a big variety of products. provides VOIP consulting, installation, and other technology services.
Custom VOIP application and component development & deployments for commerical and residential needs.
Specialties include fault-tolerance, security, IVR, voice-recording, translation, and scalable clustering.
Economy and Premium services available for SIP origination and termination. Multilingual team for support in your language.


Asterisk for C/C++ programming. Very good at monkeying with the voicemail app.
20+ years of professional programming in telecom networking. Asterisk SIP, RADIUS, AAA, RtpProxy, MGCP, and voicemail changes. No project too small. Rates from $65 to $165 per hour (Depending on how good your specification is and how much of the project you want us to control). All work done on fixed price basis. Costs depend on detail in specifications. You get a free lecture with each poor specification.
Voicemail specailist. (if someone came up with the money I would rewrite the damn thing in C++)
All work is handled remotely; believe you me you could not afford to have me on-site...

I am also doing Ruby on Rails with Video over IP for Security Applications now.
I have just completed changes to OpenSER & RTPproxy to record calls to wav files.

"Your Complete VoIP and IP PBX Consultants"
http://www.convertecsolutions.com
Convertec provides businesses of all sizes with expert, professional, and guaranteed Asterisk, PIAF, FreePBX, and Trixbox support and help.
We offer email, telephone, and remote session based assistance by the hour or with blocks available starting at ONLY $65.00/HOUR!
Why not select a VoIP consultant that not only provides support, but also provides SIP origination AND termination services? Convertec Has you covered!

We currently provide the following services
US DID coverage to over 90% of USA ratecenters VoIP and PSTN termination Toll-Free and Vanity Toll-Free numbers eFax services with desktop tools, online portals, or email receipt VoIP Network Support & Management 24x7x365 On-Call Engineer support services Emergency support services payable via credit card are available Asterisk, FreePBX, Trixbox, PIAF troubleshooting, diagnosis, and repair Complete installation support and coaching/training
Info@ConvertecSolutions.com OR 800-928-3109 Option 1 Createlsoft is a services company located in Delhi.We provide telecom solutions to Enterprises and Telcos.
We provide solutions based on popular Open source products (Asterisk, OpenSIPS...).
For any enquiry, please contact SURESH ARORA.
Email : sarora180673@yahoo.co.in
Mobile: +91-9810252984
Skype: suresh_arora73 A-5020 Salzburg, Fürstallergasse 36
D-83395 Freilassing, Sägewerkstrasse 3

Asterisk server consulting, solutions, integration with legacy PBX systems. Pre configured Servers, installation, maintainance. Open Source specialist. Company has a wide range of references thruout Europe. We do FreeBSD, Linux and Windows.

We have built networks for calling cards, conferencing, call centers, outbound dialers, custom IVR development, advanced database integration, embedded Linux product development, automated testing, and softswitch integration. Some of our installations have 1,000 lines or more in service.
Web site: http://www.directleap.com/ Located: Toronto, Canada E-mail: info @ directleap.com Phone: (416) 405-8073 We do high-volume and highly-available carrier-grade Asterisk / SER engineering. We deliver strategy-driven technology for marketing, operations, and product development. We will make your installation carrier-grade, with automated monitoring and testing. Let us give you advice.
Asterisk/VOIP/Telecom Consultant Services offered: Installation/maintenance/configuration of linux systems/servers VOIP Gatekeepers/Phones/devices. Small, Medium, Large Scale deployments of Asterisk, SER, Asterisk@home/Trixbox/FreePBX/Asterisk2Billing/Astbill Asterisk installation/configuration/Customization, Dialplan programming, AGI/perl scripting, Text-Speech/Speech-Text integration, Speech recognition Vicidial installation, customisation and support Inbound/outbound/predictive dialing solution for call centers. Onsite and remote assistance provided, can travel worldwide to implement large projects. Msn messenger: diyanat@hotmail.com Skype: diyanat Phone:+91 98850 39230 Voip: +18473801847 Email:diyanat@hotmail.com __ http://www.voip-info.org/users/view/diyanat
On the Web@ http://www.e4strategies.com
Traverse City IP PBX / VoIP, IP Telephony Consulting Company
Specialization- Traverse City PBX, IP Telephony, VoIP, (CRM) Client Relationship Management, and Carrier Services .e4 makes technology work for your business. We see technology as your engine for growth — helping businesses become more adaptive, competitive and profitable. We partner with industry leaders to deliver the hardware, applications and expertise to advance your business through the intelligent application of technology.
.e4's unique approach combines proven methods such as process management and known industry best practices together with forward-thinking concepts like running IT "as a business" not "within" a business.
By adopting e4's best practices companies are empowered with the ability to deliver quality services that satisfy business needs, are process-driven, meet cost targets, and achieve performance goals.

Contact Us
.e4 LLC
335W South Airport Road
Traverse City, MI 49686

Edwin is located in Manila, Philippines. Has extensive experience in setting up inbound and
outbound call centers. With experience as a Call Center IT Manager, Edwin knows what do you need in regards of call center technologies. Now currently as a consultant Edwin has installed and currently maintains various VICIDIAL implementation
ranging from 20 seats to 300 seats call centers. Edwin is also a member of VICIDIAL Vendor Association.

Services offered:
VICIDIAL Installation / Configuration
Custom VICIDIAL add-ons
Asterisk Installation / Configuration
Call Center Technology Setup
Asterisk billing solutions
Call Center Technology Consulting

Contact: Edwin A. Valbuena Jr.
Email: edvalbuena (at) gmail.com
Phone: +63921-773-1927
Mobile: +63915-789-4984
Website: Core Communications Inc.

Elisha Telecom offers both on-site and remote VoIP/Asterisk Consultancy worldwide. With staff located in the UK and the Philippines, our team is on hand 24/7 regardless of your time zone. No job is too small or too large.

Our expertise was obtained by implementing predominantly Asterisk based solutions in call centers, including the provision of sophisticated "call back" and conferencing solutions. As well as working with end users, we also work with other Telcos to offer support and advice with trouble shooting on an ongoing and ad hoc basis.

Our services include:
   * Asterisk PBX installation, setup and maintenance
   * Call Center setup
   * Asterisk AGI/AMI scripting
   * Custom Web based Asterisk GUI and CTI development
   * Legacy PBX integration
   * Open Source Convergence and Collaboration solutions 
   * White Label Hosted VoIP option
   * VoIP Origination and Termination (Wholesale and Retail)
   * VoIP Hardware (Digium TDM Cards, Polycom IP Phones, and more)
   * VoIP Consultancy and project management

Contact us for an informal chat and see what Elisha Telecom can do for your business:

The largest open source solutions provider in Pakistan. Our wide range of services and supported technologies are:

   * Remote Server administration & management
   * 24/7 Helpdesk support
   * Asterisk Integration
   * ViciDial integration 
   * CRM integration
   * Billing software
   * Trixbox expertise
   * OpenSER
   * FreeSwitch 
   * Voice/Video/Web Conference integration
   * IVR Integration

Contact us

http://www.evtmedia.com/ services "at " evtmedia.com (<-try this first - we travel a lot!) Phone: USA: +1 650 594 4630 , UK: +44 (0)207 043 3018 (NEW) Specialists in High-volume IVR, Police & Security Applications, Speech recognition, and Multi-lingual systems. We also provide micro-coding for specialized Bluetooth transmission software (Proximity Marketing apps) Offices in Silicon Valley, California and London, England Will Travel: Western Europe and USA
Asterisk India - Nurturing Asterisk Community in India - The Authentic Asterisk India Website
Enterux has been the pioneer in the field of Asterisk and Open Source Communications. It was instrumental in starting the asterisk revolution in the Indian market by starting the first ever Indian asterisk users community forum. Over a period of time, Enterux has been very much active in the Government and large enterprise sector and has also provided training to scientists from Depart of Defense - Center for Artificial Intelligence.
Enterux is the only company in India who has Digium Certified Asterisk Professional (dCAP) consultants on board.
Our consulting services help your organization connect its existing telephone lines with a state-of-the-art hybrid VOIP environment where inter-office, inter-branch and customer calls can take advantage of Web based voicemail, Conferencing, Automatic Call Distribution and many other advanced features.

Services Rendered:
Predictive Dialer Medium to Large Multi-Office Business Telephony Systems Call Centers - Local and International Asterisk Dialers and Bulk Calling Systems International Office Telephony Systems Specialty Calling Systems (Entertainment and Personals) Healthcare Application Integration Customer Relationship Management, CRM Application Integration Distributed Server Architecture and Asterisk Load Balancing SIP Express Router, SER Load Balancing Hospitality Telephony Systems (Hotel PBX Integration) Complete IVR Development Local or Datacenter PBX Customization Wireless Telephony Installations Database Integration and Customization Custom Application Development IP PBX/ Installation / maintenance / configuration of linux systems / servers VOIP Gatekeepers / Phones / devices. Distribution & Support for Digium range of hardware in India (E1 PRI / FXS / FXO / Echo cancellation / G729 Convertor Codec & Card) Distribution & Support for Xorcom range of hardware and software products in India (E1 PRI / FXS / FXS / AstriBank / AstriBank Server) Distribution & Support for Sangoma range of hardware and software products in India (E1 PRI/ FXO/ FXS / SS7)
Contact us to discuss requirements customizable pbx turnkey solution - e.com pbx VoIP consultancy and research pbx and soft-switch developement and integration callcentre desing and implementation
epbx Headquarters
Gotthelfstrasse 102, 4054 Basel, Switzerland
phone: +41 615115014
www: http://www.epbx.eu

epbx Czech office
V Haji 1214/13, 170 00 Prague 7, Czech republic
phone: +420 226517511
www: http://www.epbx.cz

Supporting asterisk based telephony systems since 2005, references available upon request. Also providers of SIP/IAX origination and termination services with over 90% of US ratecenters on-network. LNP available. NO contracts required We configure and troubleshoot Trixbox with VoIP, PRI, Digium, Rhino, ZapMicro, Sangoma , and many more hardware providers Guaranteed to fix your dropped calls, echo, "all channels are busy now", and all other common troubles. Assistance with phone configurations, features, and other asterisk/endpoint issues Hourly and contract pricing starting at ONLY $65.00/HOUR!!! We can service anyone anywhere in the world via ssh and web access. 24X7 emergency assistance is always available. 800-928-3109 | 208-664-2869 | www.convertecsolutions.com | info@convertecsolutions.com
Web site: http://www.evaristesys.com/ Knowledge base: http://www.evaristesys.com/workshop VoIP FAQ: http://www.evaristesys.com/workshop/index.php/VoIP_FAQ Contact: 678-954-0670 (or see contact form on web site) Location: Atlanta, Georgia, USA Specialisations: Kamailio (OpenSER) support & development (management advisory council member, official ASIPTO partner) Large-scale Asterisk deployments. SIP service delivery platform engineering. IP network engineering. VoIP contact centers project. QoS. VoIP application integration. Telecommunications operational support systems. Billing, rating, & least cost routing. Cisco voice gateways. SIP trunking. Private VoIP peering/settlement. Fax to email. Managed server colocation for VoIP.
http://freevoicetelecom.com/ E-mail: support@freevoicetelecom.com USA: 1-866-907-6158 Asterisk, Trixbox, & Freepbx $50 per hour Support, Maintenance, Upgrades, Custom Integration, & Consultation Digium Certified Asterisk Professional Onstaff Web Site Click to Call Integration CRM Integration Custom Asterisk Dialplan & AGI Scripting Several Outstanding references available Integrate Asterisk with existing PBX Systems
Performance Design, & Capacity Planning
http://fortuitous.com/ E-mail: support@fortuitous.com USA: 1-512-351-7783 System Performance Tuning and Debugging Capacity Planning for Enterprise Asterisk High Availability, VoIP Cluster, and Reliability Design Operating System and Kernel Tuning Network Stack Tuning & Optimization SAN, NAS, and Parallel Filesystem Design: GPFS, ZFS, GFS Full Network and Server Integrated Design Integrate and Tune Asterisk systems with SAN
Worldwide customers and partners! Small through Large Enterprise Asterisk designs and installations Highly Available solutions, Clustering, DUNDi Unbiased advice for SIP origination, termination, and network management Advanced monitoring / managed services offering Asterisk and FreeSwitch Virtualization and cloud / hosted design

A division of Oxford Consulting Group
Remote support: international via SSH / RD / PC Anywhere / VNC Local support: Ontario & Detroit Email: gss@ocg.ca Phone: London: (519) 672-8238 x270 Toronto: (416) 848-7506 x270 Detroit: (313) 483-1424 x270 FWD: 652811 WWW: http://www.generationd.com/ Asterisk and VOIP telephony consulting. Expierence with call centers, large office, small office, carriers Expertise integrating Asterisk into mixed Linux / Windows environments Custom AGI scripts, diaplan extensions, billing systems Linux & Windows admin, config, and security expertise (including ISA Server)
Asterisk/VOIP/Telecom Consultant Services offered: Installation/maintenance/configuration of linux systems/servers VOIP Phones/devices. Small, Medium, Large Scale deployments of Asterisk, OpenSIPS,Kamailio, Trixbox/FreePBX/Asterisk2Billing/Astbill Asterisk installation/configuration/Customization, Dialplan programming, AGI/AMI/python ,Text-Speech/Speech-Text integration, Speech recognition. IVR, Click2Call, Indian languages speech recognition, High Availability solutions. Vicidial installation, customisation and support Inbound/outbound/predictive dialing solution for call centers. Onsite and remote assistance provided, can travel worldwide to implement large projects. Msn messenger: godsong@hotmail.com Gtalk:godson.g@gmail.com Skype: godsongera Phone:+91 9246461828 Voip: +1253242384 Email:godson.g@gmail.com http://godson.in GreenfieldTech is Israel's premier solution provider for Asterisk and Open Source Telephony and VoIP infrastructures. With over 10 years experience in the Open Source IT industry and 5 years experience with Open Source Telephony projects (Asterisk, GnuGK, FreeRadius, OpenSER) - GreenfieldTech is your best choice for Open Source Telephony solutions in Israel.
GreenfieldTech also provides custom platform development services, serving as a one-stop-shop for all your analysis, design, proof-of-concept, development and deployment requirements - with proven track record.

Services offered via remote services:
Custom IVR and telephony application development, based upon Asterisk and Open Source structures. Custom application development for value added services. Integration of Open Source Telephony packages such as A2Billing, FreePBX and others. Integration of Open Source and Closed Source products, achieving best-of-breed architecture and solution. Training services for Asterisk and Linux (Certifications: dCAP, GNU-LCA, GNU-LCP, GNU-LCI).
A-5020 Salzburg, Fürstallergasse 36
D-83395 Freilassing, Sägewerkstrasse 3

Asterisk server consulting, solutions, integration with legacy PBX systems. Pre configured Servers, installation, maintainance. Open Source specialist. Company has a wide range of references thruout Europe. We do FreeBSD, Linux and Windows.

Asterisk PBX installation, configuration and support
provides different Asterisk based solutions with advanced features.
complex configuration, integration, high availability solution and clusters, DUNDI. Using Voice-over-IP (VoIP) technologies we offer you the most cost-effective and reliable way of voice communications via IP networks. We have wide experience in establishing and configuring VoIP systems integrated with such hardware as Cisco, Dlink, and Avaya. We can implement your VoIP systems in the following areas:
Security. If you require establishing security systems with remote access in your office and home, we can provide you these security systems by integrating VoIP with Smart Home system.
Business Automation. We can integrate your VoIP gateway with CRM/Call Centers or ERP systems. You immediately receive all the bells and whistles of digital voice stream such as recording, integration with the database, voice recognition and routing, etc. free.
Communication. We deployed and tuned in a new communication system based on Asterisk Open Source Project. This designed system can be tuned and integrated with your corporate software without the necessity to develop it from the scratch.
* Websites:  
Phones: 1-800-521-4091 (toll free), 1-650-681-4541 (CA) 7 383 3306626 (HireRussians/Sibers Russia) Email about:info@sibers.com
Asterisk integration.
Based in Vilnius, Lithuania. Web site: http://www.inta.lt/ Telephone: +37069811803, +37052120388 Email: Marat at inta d0t lt
Business software integration consultancy. Specialists in Asterisk, SER, Cisco Call Manager, IVRs, and PBX integration, including billing, redundancy, conferencing.
Web site: http://integrics.com/ Telephone: UK 0808 178 3170, USA 1-888-INT-E111, Rest of world +44 20 799 39 799, sip:sales@sip.integrics.com Email: sales (at) integrics (dot) com
Intelegent firmly believes in the power and ability of open source and open standards to deliver feature-rich, readily scalable, innovative and cost-effective solutions to technological challenges. We interact very closely with the open-source communities world-over to constantly innovate, optimize, contribute by sharing and learning to ensure that our clients get the best of what there is.
Asterisk based SOHO / Corporate PBX Converged Communication Solution - EMail + IM + VoIP Call Center (Predictive Dialer) Solutions Interactive voice response (IVR) Solutions
Asterisk Turnkey Solution providers Rate: Starts at $40 per hour Contact: Rajeev Natarajan Telephone: +91 9444072925 / +1-360-519-5969 Email:support AT intelegentnetworks DOT com
IVR,ASR,TTS,VXML,IMS applications, asterisk /Vyatta appliances , SS7 /SIP/TDM Gateways, Sangoma Hardware,Content Security
Contact Center Development , consulting
People Tech Systems Private Limited
F 105,3rd Street, No:4 ,Golden Brook,1st Floor
Anna Nagar East,Chennai -600102
Mob: +91 9789982453 | Dir: 044-26260810 | 044-26220019 Voice Over IP experts, Connecting businesses with VoIP since 2003
Asterisk, Switchvox, Call Manager and other VoIP PBX systems Customer-oriented IT integration and support company Expertise in communications and network systems, not only VoIP! Design, Implementation and Support of VoIP systems
With Intellect IT solution you can do telephony in new ways:
Connecting employees working from home to the office PBX over broadband connections. Enabling remote users to have calls directly transferred to them as if they were working from their office. Giving all employees voicemail, integrated with the Web and their E-mail. Allowing users to retrieve their voicemail messages from a web interface or have them directly delivered to their e-mail. Building interactive voice applications, that connect to your ordering system or other in-house applications Giving access to the company PBX for business travellers, directly from the laptop computers using a soft phone application connecting from airport or hotel wireless hotspots Free calls between interstate and International offices. Free calls to mobiles Virtually unlimited extensions and voicemail ...and much more Contact us to discover how our solutions will meet your business needs
Website: http://www.intellectit.com.au.
Phone: 1300 799 165 (+61 3 83171000) -=Easy to Use Full Featured Asterisk Solutions Available=-
http://www.intuitivevoice.com/ pbxsales@intuitivevoice.com Phone 480-422-1244 Over ten years of telephony experience with fortune 100 companies. Specialize in turnkey Small Business PBX solutions. Focus on ease of use and tightly integrated solutions. We have a full support offering and full Asterisk management suite of tools. Many happy customers and references available. Visit our site for a complete product offering.
http://www.jaredsmith.net/asterisk-consultant.php Jared Smith asterisk@jaredsmith.net Phone 801-231-7154 Respected industry expert Co-Author of Asterisk: The Future of Telephony published by O'Reilly Media, September 2005 Experienced in Call Center design and installation Experienced in Custom Asterisk Programming and Troubleshooting Available for lectures and training Based in Utah, United States Fluent in Spanish; Hablo español!
Belgrade Serbia based information technology firm providing computer networking, Asterisk PBX support, Call Center and IVR solutions and consulting.

iPhonica provides consultancy for Next Generation VoIP Solutions. Provides development, maintenance, support, consultancy for Asterisk, SER. Complete end to end solutions like call centers, IP-PBXs, calling card application for service providers, session border controllers can be provided.

GEDAM Europe The European Arm of GEDAM Advanced Communications in New Zealand and Partner of SineApps (The producer of the Daily Asterisk News has finally gone public. The main base of operations for Europe is in Torino in the north of Italy, although we provide worldwide support. We specialise in all types of Asterisk and SER projects. We also develop embedded hardware platforms for running voice applications. We also have extensive experience with video over IP and can custom develop solutions to meet your needs perfectly.
         o Home page:: http://www.gedameurope.com
         o Telephone: PSTN +39 011 5694900
         o Email: info@gedameurope.com  We provide you with Asterisk-installations, VoIP-Consulting and customized programming.

Authorized Digium reseller for Russian Federation. Great experience with IP-PBX remote deployment and maintenance. Great experience with various IP-phones and VoIP-gateways. Asterisk security solutions, including Ranch Networks MIDCOM firewalls. DUNDi networks.

Divisão iComm
   * Phone: +55 (11) 4063-4304
   * Corporate IP Telecom solutions
   * IP Telecom consulting and Distributed communications
   * Asterisk consulting and advanced implementations.
   * Pre-Paid VoIP
   * Contact :
   * web: http://www.itfuture.com.br/
   * e-mail: icomm@itfuture.com.br  Mathias Kettner Asterisk and Linux Consulting and Training.
We offer a wide range of services and support concerning Asterisk, VoIP, XEN, Nagios, RPM, Softwaredistribution, Automatic Installation, AutoYast, Kickstart etc.

Professional VoIP and Asterisk consultant since 2005
Professional Linux SysAdmin, coder and security manager since 2000


Free & Open Source ICT solutions

At monoploid, we provide a plethora of ICT services based on free software and open technologies. We perform:
Asterisk installation Callshop deployment Corporate VoIP setups Scalable load-balanced/high-availability VoIP setups
Other VoIP services include:
Custom Asterisk development VoIP peering Remote technical support
Linksoft Technologies Ltd is a telecom software solution provider specializing in VOIP and traditional telephony applications. We provides consultancy, software development, and system integration services. We specialize in:
turn-key Asterisk-based IP-PBX software solution with an initutive web-based configuration and management interface custom Asterisk modules (e.g. outbound dialer, enhanced voicemail, IVR, etc) custom software development (web- or Windows- based) design, configuration, and installation of Asterisk PBX system

Website: Moki Systems Located: St. George, UT Email: Sales Phone Number: (435)-674-3571 The Moki Systems Team has experience configuring, maintaining and installing Asterisk Phone Systems. Our rates are very competitive and we offer our services in English and Japanese. The Moki Systems Team is Polycom Certified allowing you access to the highest quality phone systems at a reasonable price. Nasoni & Friends Located in Montreal Email asterisk.qc at REMOVEME gmail dot com Italian Number +39-02-303123373
My Friends and I have decided to offer our knowledge that we had developed around asterisk to help others. We are able to offer our services in Italian, English and French. We have a very competitive rates. NDSoftware provides VoIP consulting services.


We are a R&D oriented company, we have strategic alliances with several hardware resellers, focus on reliability and security We provide Consulting, Planning, Deployment, support and training services for Asterisk based PBX and appliances. Installation of Asterisk PBX with PSTN, VoIP, GSM and FAX integration Installation of switches with traffic management and Wifi Networks with wireless IP phones Turnkey solutions & Integration of asterisk with existing network utilizing SIP and/or IAX2. Call center solutions. Custom AGI scripts, diaplan extensions, billing systems Asterisk / VoIP Network Support & Management. 12x5x365 On-Call Engineer support services.
Contact Person:
S.Khan. Rao
CEO
Cell: +92.344.5900.154
Email: info@nethawk.com.pk
Company Website: www.nethawk.com.pk
NextGenData is an IP services company providing businesses the ability to completely re-engineer IT infrastructure using the most cost effective and reliable technology available. We provide SIP voice trunks and feature servers for Asterisk users. We also provide consulting on the VoIP architectures for multiple platforms including Asterisk. They focus on technology solutions for Start-up and Rapidly Growing Call Centers. The company has done extensive testing, troubleshooting and installation of asterisk for real world production environment. Its integrated technology package is complemented by a dedicated technical support team that provides expert business process support services to boost the client's competitive advantage in the highly dynamic call center industry.
NextIX technical support and development team are Digium certified professionals.
NextIX also operates an offshore call center and medical transcription company in the Philippines.
NeuralHead Diseño de redes Voip (Voip Network design). Consultoria Asterisk (Asterisk consulting). Guias de resolución de problemas (Troubleshoting guides). Soporte remoto de Asterisk (Asterisk remote support). Desarrollos a medida para Asterisk (Ad-hoc Asterisk development). Integración de sistemas (System integration). Miembro del AAUG (Member of AAUG).
Website:http://www.neuralhead.com
E-Mail:info@neuralhead.com Web site: http://www.aleph-tec.com/ E-mail: oleg.titov at gmail.com FWD: 74758 fwdOUT: 52442 Languages: Spanish, Russian, English Asterisk PBX, VoIP, SIP, H.323, worldwide remote administration, Linux sysadm, consultancy and training courses.
Special pricing for personal projects, SOHO/SMB clients. Secure and easy payment with PayPal. Solutions Designed :
CALL Center : Predictive , Priview and Progressive Modes
Asterisk : IP PBX / Audio Conferencing / IVRS/Speech Recognition with all indian languages
A2Billing
OpenSer : Sip Express Router
Building API to integrate and interact with your Inhouse Softwares
Hosted pbx,Hosted Ivrs.
Asterisk Training,
Sugar CRM
Prepaid Calling Cards
Wholesale Billing
Call Shop
Custom IVR Development
Hosted IVR Solutions
IP IVR Solutions
Enterprise Voicemail Systems
Carrier Voicemail Systems
Automated Dialers
Economic IP Phones (SIP / H.323 ) especially designed for Asterisk and SIP based pbx
Authorised Distributor for Rhino Cards in India
Aserisk and Freeswtich Cards available @
" http://www.telephonycards.co.in
We do Asterisk

Portland, Seattle and the Bay Area. Asterisk@Home solutions for small to mid-size business.
Asterisk based VoIP PBX and Call Center solutions with CRM integration; Video Phone, Video Conferencing and Video over IP to check, control and discuss face to face with your employees; Our consulting services will help your organization to covert its existing telephone system into cost saving, effective and flexible VoIP structure; On site and remote services from our consultants available everyday 8:30 - 20.00; We speak polish and english;

Specialising in the design, build and deployment of large scale telephony solutions. Progressive Voice Services provide several telephony solutions including operating one of the UKs leading conference call services.

sales@raynettech.com
Toll Free 877-693-2226
Local 860-693-2226
Full Implementation and Development of Asterisk and Cisco Phone Systems Custom Development Managed Cost Maintenance Contracts Full Service IT Support and Sales Call Centers VoIP over VSAT Emergency Communications Pre-packaged and Custom Systems Available
(Rodney Tamblyn) Dunedin, Otago, New Zealand Tel: (03) 4778606 x 1 Fax: (03) 4778606 Email: sales@sineapps.com Website: www.SineApps.com
US (NY, NJ): +1 800 824 6006, +1 212 537 6898
DE (Berlin): +49 30 868 70 9130
UK: +44 207 127 6310, +44 1510 601 2050
info@sipxx.com
sip:10888@fwd.pulver.com
Voice-over-IP, Asterisk PBX, SER, SIP Exchange, Snom4s, FreeSwitch SIP, IAX, MGCP, H323, SCCP, PSTN, IPv6 AudioCodes, Snom, Mediatrix, Sipura Design, Software, Integration, Installation, Support Custom SIP application development
Rafael A. Lama
Santo Domingo, Dominican Republic
Tel: (809) 620-2488
Email: info@smbconsulting.com.do
Website: www.smbconsulting.com.do Asterisk, Trixbox, Vicidial installation dCAP, FTOCC Certified
Melbourne, Victoria
Telephone +61 3 8888 5361 or +61 408 873 601
Email sales@stuartelvish.com
Specialised Asterisk systems for infrastructure projects including airports, roads, tunnels and carparks. Ad-hoc support or 24x7 remote support service level agreements available. Support for web, database and Asterisk integration.

Asterisk PBX installation, configuration and support. We also able to provide different solutions with advanced features. We offer remote support.
Melbourne, Australia. Integrated Open-Source Systems; since 1999. Installation and support of Asterisk and Linux server systems. Remote Support. Tel +61 3 9484 0013 email Robert Slater
IT and VoIP. Done right. Done now. (TM)
Worldwide remote consulting and support for Asterisk
General Asterisk/VoIP consulting Design, installation and maintenance of Asterisk-based PBX systems Call center infrastructure Custom application/reporting development Guaranteed satisfaction; many excellent references available Prompt service and response times Broad/in-depth team of telecom and data engineers
Contact: Dale Laushman
(303) 757-4611, Ext. 401
dale@theuptimegroup.com
www.theuptimegroup.com Sydney, NSW, Australia
Tel: +61 (2) 8090 0800

Product:
Mach 1 Asterisk-based PABX PABX Call Centre feature upgrades VoIP infrastructure suppliers
Consultancy:
CRM and CTI Integration services Design and Implementation consultancy VoIP Readiness Assesments
Managed Services:
Ongoing remote management and administration
http://www.ulterius.net/
sales@ulterius.net
Become your own Vonage with the VoIP Vendor solution. Everything you need in one asterisk box including a web site with customer interaface to view call history, billing, voicemail and custom settings. We are also a provider of te lowest rates on termination (outbound) and origination (inbound) for US and international calling. VoIP Reach also provides technical partnerships and professional project mamagement on VoIP Networks. Voiptech is an engineering company specialised in Enterprise Grade VoIP solutions for Providers, Telcos and Call Centers. Our portfolio included thousand of IP-PBX and Call Center installations. Voiptech is the developer of EasyPbx http://voip-info.org/wiki/view/EasyPbx an Hybrid Carrier-grade Software Platform to configure, control and monitor IP-PBX deployments in real-timebased on Asterisk and CallWeaver. Visit our web site, where you can find technical hints and further informations. Technical Contact : Giorgio Santini mailto:%20gsantini@voiptech.it Email : mailto:info@voiptech.it Telephone : +39 06 62 27 79 60 Home Page : http://www.easypbx.it
VoIP Consulting, Installation, Configuration and Programing for Aterisk, Yate, Web-Services and Apples MacOSX

Warrior Hill specializes in custom Asterisk solutions. Our team of developers can provide custom solutions for your Asterisk needs quickly and affordably. We provide a full range of remote Asterisk setup and management services.
Voices that bring asterisk to life
London, United Kingdom Tel: +44 (0)800 066 4864 (free phone) Email: sales(at)westany.com Website: www.westany.com
Westany provides international voice prompts for Asterisk. We currently have British English and French with German, Spanish and italian in production. All of the prompts can be extended by the same voice artists. Our pre-recorded packs start from just £149.99. Milano, Italy Tel: +39 02 40039010 Fax: +39 02 4151 203 Email: info(at)soluzionivocali.it Website: www.soluzionivocali.it
Provides you a wide range of Asterisk IVR solution for help desks, call centers, interactive voice applications and PBX integration.We can easily
interact and manage with database,TTS through the most modern tecnologies such SOAP,Corba,SIP,VoiceXML Located In Mumbai, India. Have implemented may Asterisk solutions for small and medium call centers and calling card shops. We have setup loadbalancing Vicidial Servers with high availability Mysql Cluster Database for high performance , stability and scalability.
 Phone +91-22-27581200
 Email dkumar at netsolutioninc.com
 Web : http://www.netsolutioninc.com
We provide a top tech services. Our main focus is in telecomunications and banking industry. Asterisk based solutions (IVR, Call center, billing, ...). Enterprise Application Integration via modern technologies. TintaDigital STI is a Portuguese company offering Information Technology services directed to all companies with technological needs.
Our team has solid skills in Asterisk systems, allowing the implementation of a set of services which goal is to support the planning and maintenance of the system's growth in a structured and consolidated manner.


We have extensive experience designing, installing and maintaining a different range of VOIP solutions. We can cluster and streamline your carrier solutions. We can help you design and implement you next big project. VoIP Experts’ team consists of qualified professionals with an extensive experience in VoIP area. The main spheres of our expertise are technical support, configuration, troubleshooting and system maintenancce of different VoIP solutions. We strive to provide highly qualified service along with fully satisfaction of our customers allowing them to achieve their key business strategies.
Phone: +15304051102 E-mail: sales@voip-experts.com
We offer various services for Asterisk such as SBC Configuration, custom live cd, wengophone customization, etc.
Contact us.

Dallas, Texas Sales@yellowboxsolutions.com Phone: 972-235-5800 Asterisk Consulting: Offer two different solution packages for SOHO and enterprise at economical prices. Our Consulting services also includes Customized Asterisk Development Asterisk configuration and installation Dialplan Designing AGI scripting IVR Designing Troubleshoot and other consulting services VOIP Hardware distribution include polycom and grandstream. see website for more details
Install and customize your VOIP PBX Solution.
Call Center Vicidial installation and integration.
Networking Linux Specialist

Radu Padure

www.voipexperts.com.ar / info@voipexperts.com.ar Location: Córdoba - Argentine Tel/Fax: +543515713406 Location: Buenos Aires - Argentine Tel/Fax: +541162612011 VoIP Consulting / Asterisk Specialists Design, Configuration,Integration, Installation and maintenance of Asterisk PBX Systems. We can provide installation and support of these systems worldwide. Custom asterisk PBXes for small, medium and enterprise level offices Custom business applications solutions and development based on * (Home Banking, Predictive Dialers, IVRs, CTI) Specializing in all facets of VoIP, Including provisioning, wholesale termination, high-volume Unified Messaging (Including Voice Mail, Fax Services, call conferencing, etc) Troubleshooting. Design. Analysis. Consulting
Guaranteed Results
Asterisk Consulting, integration, optimization and repair 24/7/365 Availablity Contact Center specialists with a proven track record Phone 602-505-4125 http://www.voipcitadel.com/ Creators of VicidialNOW distribution a call center suite automated installation pack in 1 CD!

Linux Server Installation (Web, Mail, etc...).
Asterisk Server Installation.

José Héctor Galimberti

Install and customize Asterisk PBX.
Using a powerful java swing GUI interface for User Management, CDR List, Call Recording, Call Spy, Server Health Monitor, Builtin IAX admin phone, Group Management, Remote Backup and Restore, Export every thing to excel. Call Center software and database integration. SMS & IVR Value added services provider.
Almontaha ICT Co.

Asterisk Telephony Platform for Call Center

Indosoft Inc., a premier contact center software provider offers Q-Suite, a robust, feature-rich and scalable call center ACD for people wanting to use Asterisk as their telephony platform. Q-Suite is enterprise grade, multi-tenant call center software that enables VoIP and TDM connectivity, as well as remote and distributed architectures. Q-Suite delivers sophisticated features like Automatic Call Distribution (ACD) with Skills Based Routing and Queue Prioritization, Predictive/Preview/Progressive Dialing, Interactive Voice Response (IVR), rich IP PBX functionality, call recording, agent scripting and a host of other features and reporting capabilities to meet the demands of modern day contact centers wanting to use Asterisk as their telephony platform. Workforce Management can be integrated into Q-Suite to deliver enterprise interaction through advanced scheduling, forecasting, agent tracking and seat planning capabilities. Q-Suite leverages the advantages of Asterisk, Linux, Apache and MySQL to provide an open architecture with unsurpassed functionality required for any next-generation contact center technology platform, out of the box.

Q-Suite is widely used around the world as premise-based software, as well as in remote and virtual call centers. Its Linux and Asterisk platform make it the most cost effective solution for call centers. Indosoft has been successfully deploying its technology in Canada, U.S.A., Europe, Australia, Asia and Africa for the past 10 years. Indosoft provides excellent remote production support from 9 a.m. EST to 12 a.m. EST, 7 days a week for all its products and services.

Q-Suite offers increased productivity within the contact center while dramatically improving the center’s ability to deliver a favorable customer experience. As such it delivers all the functionality and features required by next-generation contact centers, whether you operate inbound, outbound, or blended operations.

• ACD with Skills Based Routing• Real-time & Historical Reporting• API for CTI (.NET and Socket)• Hosted/Premise/Distributed Architectures• High Availability for Redundancy• Mid-call Recovery for Fail-over• Open Access and Full Knowledge Transfer• Web Agent Interface & Native Client
For more information, read our White Papers or visit our website at www.indosoft.com. Q-Suite’s full-featured Asterisk ACD is a unique product allowing enterprise-grade call centers to use Asterisk as their telephony platform. As a complete call center ACD software for Asterisk, Q-Suite provides advanced functionality like Skills Based Routing, Queue Prioritization, Virtual/Personal Queue and Agent Hot-Desking for the highly evolved hybrid telephony switch, Asterisk, which supports both VoIP and TDM connectivity.

Q-Suite ACD comes with a well developed set of APIs available in both .NET and socket (TCP/IP) library for Asterisk, enabling full ACD functionality to be embedded into your custom business application. With built-in redundancy and fail-safe call recovery, Q-Suite maintains ongoing calls and allows new calls to continue coming should there be any single point of failure, including software failure, within the components of the call center system. This unique feature enables Q-Suite ACD software to be utilized by mission critical applications.

For more information, read our White Papers or visit our website at www.indosoft.com.

Built on top of the open-standard open-architecture switch Asterisk, Q-Suite’s Predictive Dialer enables Predictive, Preview, and Progressive Dialing. Live Lead Queuing enables leads gathered from sources like websites to be posted directly to Q-Suite for immediate dialing. Built-in Concurrent Dialing capabilities, Drop Rate controls, as well as both Individual and Self-Pacing controls work together to maximize outbound call center productivity. A central administrative console provides powerful tools, including full Campaign and List Management as well as a dynamic, easy-to-use Agent Scripting tool, which enables on-page CRM integration, custom variables, text blocks with embedded variables, Post to URL functions and other popular dialer features.

For more information, read our White Papers or visit our website at www.indosoft.com.

Q-Suite’s unique Script Builder is a powerful tool used to generate scripts that guide effective agent-customer interactions for agents using Q-Suite through their web browser (as opposed to a custom application interface). Powered by an easy-to-use Graphical User Interface (GUI), Q-Suite delivers powerful script building capabilities.

A few of its popular functions include the ability to easily embed a web-enabled CRM or other Custom Application directly within the script. Other features include the abilities to insert text boxes capable of using data inline from a database, create custom fields for use within the script, and display script elements/pages based on conditional responses. Q-Suite’s Script Builder also enables data posting functions to be inserted within the script itself, which allows agents to input information received during the call to their agent screen, which is then automatically posted to the indicated URL once the call is finished.

For more information, read our White Papers or visit our website at www.indosoft.com.

The Dialplan Builder within Q-Suite is a Graphical User Interface (GUI) tool utilizing a What You See Is What You Get (WYSIWYG) editor, which allows easy-to-use drag and drop features to develop Dialplans for IVR and call routing in Asterisk. This sophisticated Dialplan Builder allows contact centers to accommodate new and diverse client requirements and increase productivity.

IVR or Dialplans are lists of instructions or steps that Asterisk will follow while handling incoming calls. Unlike traditional phone systems, Asterisk Dialplans are fully customizable and incredibly versatile, allowing full control over the handling and processing of calls. The flexibility and power of Dialplans come from its unique and powerful structure based on contexts, extensions, priorities, applications and a powerful syntax to construct instructions. Q-Suite Dialplan Builder enables the user to unleash the combined power of software and telecommunications to easily control and manage calls. Dialplans can go even further when you develop custom applications for call flow processing in well known and conventional programming languages like Java, C, PHP, PERL etc.

For more information, read our White Papers or visit our website at www.indosoft.com.

Many mature business verticals have a well developed CRM or Custom Application. Q-Suite allows full integration of these applications into the call center software.

A web-enabled CRM or Custom Applications that is capable of accepting command line parameters can be directly inserted into the agent script. The CRM or Custom Application will be displayed as a webpage and opened in the script once the appropriate customer data has been collected. This easy integration allows the agent to work efficiently from one screen.

If the Custom Application is driving the customer interaction, it is possible to embed the Computer Telephony Interface (CTI) into the Custom Application. A well developed set of APIs available in both .NET and socket (TCP/IP) library allows full call center functionality to be embedded into the Custom Application’s interface.

For more information, read our White Papers or visit our website at www.indosoft.com.

Q-Suite provides a uniform interface for CTI integration in the form of well-published API in both .NET and Socket library. This enables Q-Suite ACD to work under the hood, enabling vertical businesses to integrate CTI into any existing or new business application. This approach allows ease of integration and efficient application development, creating an opportunity for business systems with legacy telephony to incorporate CTI, embed Q-Suite ACD and move to IP telephony using Asterisk. Q-Suite API with its uniform interface for CTI integration presents a unique opportunity for deploying Asterisk as your IP telephony platform. Benefits of an Asterisk telephony platform include seamless VoIP and TDM interface, rich PBX functionality and a superior telephone switch at a fraction of the cost of comparable proprietary systems. The CTI library for Q-Suite ACD on Asterisk is available in two flavors (.NET and Socket) to make it easy for use in client applications under different platforms. The XML library allows client applications to log agent activity and tie it with telephony for ease of reporting.

Asterisk H323 channels

The Asterisk H.323 channel is included in the Asterisk source distribution in the channels/h323 directory in the source tree. The chan_h323 only acts like a H.323 Gateway not a gatekeeper, although it appears that the author is currently looking at adding basic gatekeeper functionality. See the channels/h323/README for installation instructions and software requirements and h323.conf for configuration.
There is another H.323 channel implementation (in fact the first one for asterisk that came into existence), named Asterisk-oh323 which is actively developed by InAccess Networks and can be found at http://www.inaccessnetworks.com/projects/asterisk-oh323.
asterisk-ooh323c is a part of asterisk-addons package. It is yet another, new (as of June '05) channel driver based on open source H.323 stack (ooh323c) from Objective systems. This stack is developed in C and contains only the code necessary to set up H.323 signaling channels. All media processing is handled by Asterisk itself. This provides scalability for H.323 calls that will depend primarily on the capability of Asterisk to handle media streams. Users should see call volume handling that is similar in magnitude to what can be handled by SIP. Currently (30-Jun-2005) the channel driver is available on asterisk-addons cvs and also from Objective Systems website at http://www.obj-sys.com/open. Note: You need CVS-HEAD version of asterisk.
The Woomera protocol makes it possible to put your voice over ip system in one server/process and your pbx in another and connect them with a simple raw-linear-over-udp protocol. chan_woomera is an asterisk channel_driver designed to interface the Asterisk PBX with woomera. Currently (June '05) this code is working but considered beta. Woomera currently only supports H323 but it should soon support the OPAL VOIP abstraction layer which will allow it to speak many other protocols. The number of protocols supported by the Woomera server is irrelevant to chan_woomera which will support anything Woomera supports because of it's thin-client-like design.
With woomera you can connect asterisk to a H.323 server (openh323 code) which will do H.323 over IPv6. Apparently openh323 also has some SIP code in their CVS. If added to chan_woomera, you'd get SIP over IPv6 also. h323 performs better, but has no jitter buffer. This implementation uses the Asterisk RTP stack. oh323 driver uses the RTP/RTCP stack and the adaptive jitter buffer implementation of OpenH323. oh323 does not use the codecs of OpenH323 but those of Asterisk. Using OH323 makes all these (stability) problems (with h323) go away, however at a cost of approximately 10-15 times the CPU usage in my situation: a G729 call coming from a Sipura and being rerouted with G.729 over H323 to a Quintum call proxy Jeremy McNamara on performance and why he started chan_h323 Compilation in channels/h323 subdirectory fails with a lot of syntax errors - you need to use exactly the version of PWLib and OpenH.323 mentioned in README. While connecting with H.323 client you get no audio or garbled audio and messages like this on Asterisk console: rtp.c:489 ast_rtp_read: Unknown RTP codec 107 received. Try to disable Speex or some other codec on Asterisk or/and client side. Compiling openh323 can require substantial memory resources, so make sure you have either enough RAM or sufficient SWAP (user report: 380 MB required) MS Neetmeeting is available on most Windows machines and thus a common tool to run the first h.323 test. Be sure to manually select a preferred audio codec in Netmeeting that is well supported by Asterisk, e.g. do not use g723.1 (which is the default) but A-Law or u-Law. Where bandwidth is a concern, use the installable GSM codec for Netmeeting which is available here: Netmeeting-GSM. Just download and start the instcodec.exe, select the GSM codecs and return to Netmeeting.

There is also a Netmeeting plugin for the free Open Source codec Speex, but it is a bit hard to set up: (Netmeeting Speex).

The next step is to edit h323.conf so that your netmeeting callers get dtmfmode=inband.

If you'd like to dial different extensions on Asterisk then you'll probably want to enter the Asterisk server's hostname (or IP address) as Gateway (not Gatekeeper!) in Netmeeting. After that just dial the extension. Without this Gateway setup you will probably want to dial the IP address of your Asterisk box.

chan_ooh323 with Siemens optiPoint 400 : if the RTP stream is closed after 30 seconds, it means chan_ooh323 didn't get a H.245 terminalCapabilitySetAck from the phone and timed out. This happens because the phone expects the dtmf item in the message and doesn't send the acknowledgment if the item isn't present. To work around the problem, use dtmfmode = h245signal in ooh323.conf.

allow=all in ooh323.conf won't work with asterisk-addons-1.4.0 or ealier (calls hang up just after being answered, see http://www.mail-archive.com/ooh323c-devel@lists.sourceforge.net/msg00385.html and the subsequent posts). You must have disallow=all, followed by allow= for each codec you want. Set that globally only. The codecs supported at the moment are ulaw, alaw, gsm, g723 and g729, and you must have the corresponding asterisk codecs installed.

Asterisk config h323.conf

Configuration for the Asterisk H323 channels
The NuFone Network's
Open H.323 driver configuration
;
[general]
port = 1720
bindaddr = 0.0.0.0
;tos=lowdelay
;

You may specify a global default AMA flag for iaxtel calls. It must be one of 'default', 'omit', 'billing', or 'documentation'. These flags are used in the generation of call detail records. (See Asterisk IAX channels for explanation of tos and amaflags)

;
;amaflags = default
;

You may specify a default account for Call Detail Records in addition to specifying on a per-user basis. See Asterisk billing

;
;accountcode=lss0101
;

You can fine tune codecs here using "allow" and "disallow" clauses with specific codecs. Use "all" to represent all formats.

;
;allow=all              ; turns on all installed codecs
;disallow=g723.1                ; Hm...  Proprietary, don't use it...
;allow=gsm              ; Always allow GSM, it's cool :)
;allow=ulaw
;
; User-Input Mode (  DTMF )
;
; valid entries are:   rfc2833, inband
; default is rfc2833
;dtmfmode=rfc2833
;
; Set the gatekeeper
; DISCOVER                      - Find the Gk address using multicast
; DISABLE                       - Disable the use of a GK
;  or    - The acutal IP address or hostname of your GK
;gatekeeper = DISABLE
;

Tell Asterisk whether or not to accept Gatekeeper routed calls or not. Normally this should always be set to yes, unless you want to have finer control over which users are allowed access to Asterisk. Default: YES

;
;AllowGKRouted = yes
;

Default context gets used in siutations where you are using the GK routed model or no type=user was found. This gives you the ability to either play an invalid message or to simply not use user authentication at all.

;
;context=default
;
; H.323 Alias definitions
;
; Type 'h323' will register aliases to the endpoint
; and Gatekeeper, if there is one.
;
; Example: if someone calls time@your.asterisk.box.com
; Asterisk will send the call to the extension 'time'
; in the context default
;
;   [default]
;   exten => time,1,Answer
;   exten => time,2,Playback,current-time
;

Keyword's 'prefix' and 'e164' are only make sense when used with a gatekeeper. You can specify either a prefix or E.164 this endpoint is responsible for terminating.

Example: The H.323 alias 'det-gw' will tell the gatekeeper to route any call with the prefix 1248 to this alias. Keyword e164 is used when you want to specifiy a full telephone number. So a call to the number 18102341212 would be routed to the H.323 alias 'time'.

;[time]
;type=h323
;e164=18102341212
;context=default
;
;[det-gw]
;type=h323
;prefix=1248,1313
;context=detroit
;
Reaources: Recover Raid 5

Inbound H.323 calls from BillyBob would land in the incoming context with a maximum of 4 concurrent incoming calls

Note: If keyword 'incominglimit' is omitted, Asterisk will not enforce a limit on concurrent calls.

;
;[BillyBob]
;type=user
;host=192.168.1.1
;context=incoming
;incominglimit=4

A2Billing – assigning an inbound number to a customer

Assigning an inbound number to a customer is fairly easy to achieve in A2Billing. Once assigned the customer can use the web interface to change the destination of that inbound number.

First you are going to need an inbound number from a DID/DDI provider. I’m going to use DIDX.net in this example.

We need to set the SIP URI destination for the inbound number with our DDI provider. In the DIDX control panel I’ve set the destination of a London DDI  (country code 44 / city code 20) to my demo server -

Next, in FreePBX on the livedemo server, we are going to create an inbound route that checks for the number specified in the SIP URI and passes that call through to A2Billing -

(The availble custom destinations need setting up like this first – http://sysadminman.net/blog/2009/integrating-freepbx-with-a2billing-621)

Now, in A2Billing, we are going to create the inbound number. There are various options for doing this such as importing from a CSV or linking to DIDX.net, obviously handy if you’re creating lots of DDI numbers.

There are also options for how the outbound leg of the call is charged, if at all. If the customer needs the number forwarded to a regular PSTN number and you select ‘dialout rate’ for the billing method then the customer will be charged based on their assigned call plan. It is also possible to charge a monthly amount for the DDI number.

Another option is to forward the call to a VOIP device, maybe a PBX if providing wholesale services or a SIP handset if providing end-user services. One thing it’s not easy to do in A2Billing is charge a per minute rate when forwarding the call as a VOIP call.

Create the number in A2Billing -

Now we’re going to assign that DDI to a customer and set the destination. In this exampe we are forwarding the inbound number to a UK mobile number. Note that VOIP_CALL is set to no. Also, the customer must be assigned to a Call Plan that allows them to call a UK mobile number as they will be billed for this leg of the call -

The customer is now able to see the inbound number in the A2Billing customer interface, and also change the destination -

Related posts: Part 6 – Create a customer in a2billing on ElastixGetting started with A2Billing – Part 4 Creating a customer and making a callUsing A2Billing to account for extension calls in Elastix