Rattler: VAS developer (SMS, MMS, WAP, WEB, USSD, IVR, VoIP)

[posted by p_lindheimer on freepbx.org] When it comes to busy queues and FreePBX, it often results in requests for help in the forums and to our support department because of overloaded systems or certain functions that do not seem work right. FreePBX was designed to be an SMB PBX concentrating on feature rich capabilities at the expense of complex dialplan and AGI scripts, all of which take their toll when you setup a reasonable size call center and start throwing lots of simultaneous calls at a group of queue members. This Technical Corner post will discuss some of the advancements introduced in FreePBX over the years to help accommodate systems that make extensive use of queues. These same changes will have positive impact on low power appliances and any other system that uses queues.

Throughout the article we will refer to FreePBX Queue Agents as members to be consistent with the Asterisk naming convention. FreePBX commonly refers to any queue member as a Queue Agent which is someone who answers calls destined for the queue and does not specifically refer to the Asterisk agents.conf agents.

The Asterisk Development Team has announced the release of versions 1.6.0.6 and
1.6.1.4 of asterisk-addons. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The Asterisk-Addons releases for 1.6.0.6 and 1.6.1.4 are the last maintenance releases for Asterisk-Addons branches 1.6.0 and 1.6.1 and have now moved to security maintenance only.

The Asterisk Development Team has announced the release of versions 1.6.0.6 and
1.6.1.4 of asterisk-addons. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk

The Asterisk-Addons releases for 1.6.0.6 and 1.6.1.4 are the last maintenance releases for Asterisk-Addons branches 1.6.0 and 1.6.1 and have now moved to security maintenance only.

Customized solution clipped on the needs of the customer. All the functionality of the system VoiceOne on powerful hardware , with the possibility of integration with the systems and the software business, of unit switchboard. VoiceOne Entreprise is the version of the VoiceOne suitable for medium and big companies that need advanced solutions for communication.

Customized solution clipped on the needs of the customer. All the functionality of the system VoiceOne on powerful hardware , with the possibility of integration with the systems and the software business, of unit switchboard. VoiceOne Entreprise is the version of the VoiceOne suitable for medium and big companies that need advanced solutions for communication.

Customized solution clipped on the needs of the customer. All the functionality of the system VoiceOne on powerful hardware , with the possibility of integration with the systems and the software business, of unit switchboard. VoiceOne Entreprise is the version of the VoiceOne suitable for medium and big companies that need advanced solutions for communication.

Blink is simply the best SIP client . You can use it with any SIP provider or Blink provided fully-featured SIP service. A state of the art, easy to use SIP client. Now available for MacOSX, coming soon for Windows and Linux.

A2Billing continues to develop at a speed surpassed in its five year history, and should definitely be considered by any existing or potential telecoms operator wishing to provide any billable telecoms service.
We have just released A2Billing, version 1.7.1 This version, as well as fixing a number of bugs, adds some new features and is available free of charge under the AGPL license to install yourself at www.a2billing.net. Alternatively, Star2Billing S.L. - The commercial arm of A2Billing – can provide professional installation services, training, ongoing support and development.

The final VXI* VoiceXML browser 5.0  ref. 2010-05-21 32bit and 64bit is now released. This new release is suitable for production platforms running with all lastest Asterisk 1.4 and 1.6 kernels (both Asterisk’s packages are available for download too). Like our previous release, VXI* 5.0 has been built to run over Asterisk EC2 or Xen virtual servers. Stay tuned to the blog to keep up to date on our progress or check out our lasts builds.

You can download these new binary packages from this website for registered users.

Digium announce the addition of Digium’s Switchvox Mobile application - an app that seamlessly integrates your customers’ Switchvox phone systems with iPhone or Blackberry Bold, Curve, Tour and Storm.

Our new FREE app, though not yet announced publicly, is coming soon for Blackberry models and is available now in the iTunes App Store allowing you to stay on top of the competition and give your customers the best unified communication solution on the market today.

Tell your existing SOHO and SMB customers about this great feature that adds even more mobility, usability, and flexibility to their existing investment. With features like:

The Asterisk Development Team has announced the release of version 1.4.11.1 of libpri. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/

This release fixes a regression in multi component FACILITY messages and includes a minor bug fix for BRI spans:

* Multi component FACILITY messages only process the first component. The code
   was only processing the first ROSE component in the facility message.
   (Closes issue #17428. Reported, tested by: patrol-cz. Patched by rmudgett)

The Asterisk Development Team has announced the release of version 1.4.11.1 of libpri. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/

This release fixes a regression in multi component FACILITY messages and includes a minor bug fix for BRI spans:

* Multi component FACILITY messages only process the first component. The code
   was only processing the first ROSE component in the facility message.
   (Closes issue #17428. Reported, tested by: patrol-cz. Patched by rmudgett)

OpenVox Communication Co. Ltd, a global provider of open source asterisk® telephony hardware and software solutions, has announced today that the industry first design-for-asterisk industrial embedded motherboard-IPC100 is released to the Open Source community. The IPC100 series motherboards can work flawlessly with OpenVox Mini PCI cards-A400M/B100M/B200M/B400M as well as any combinations to build a complete embedded IPPBX.

OpenVox IPC100 motherboards are small form (15cm*15cm) factor system boards optimized for IPPBX and network security applications. The Fanless design, as low as 6W normal power, enables more reliability to the system and ability to operate in harsh industrial environments. Integrated up to 3 Ethernet ports, the IPC100 motherboards provide a flexible choice for different applications to the users. With OpenVox Mini PCI cards, SMB can easily have their phone systems built with IPC100. Installed two OpenVox B400M cards with IPC100, users can have up to 100 concurrent calls in one single system.

OpenVox Communication Co. Ltd, a global provider of open source asterisk® telephony hardware and software solutions, has announced today that the industry first design-for-asterisk industrial embedded motherboard-IPC100 is released to the Open Source community. The IPC100 series motherboards can work flawlessly with OpenVox Mini PCI cards-A400M/B100M/B200M/B400M as well as any combinations to build a complete embedded IPPBX.

OpenVox IPC100 motherboards are small form (15cm*15cm) factor system boards optimized for IPPBX and network security applications. The Fanless design, as low as 6W normal power, enables more reliability to the system and ability to operate in harsh industrial environments. Integrated up to 3 Ethernet ports, the IPC100 motherboards provide a flexible choice for different applications to the users. With OpenVox Mini PCI cards, SMB can easily have their phone systems built with IPC100. Installed two OpenVox B400M cards with IPC100, users can have up to 100 concurrent calls in one single system.

The Asterisk Development Team has announced the release of Asterisk 1.4.32. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.4.32 resolves several issues reported by the community, and would have not been possible without your participation.
Thank you!

The Asterisk Development Team has announced the release of Asterisk 1.4.32. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.4.32 resolves several issues reported by the community, and would have not been possible without your participation.
Thank you!

VOIP Today magazine (www.voiptoday.org) is privileged to conduct exclusive interview with Kamailio (OpenSER) co-founder and core developer of OpenSER and SER projects Daniel-Constantin Mierla.

Kamailio (former OpenSER) is an Open Source SIP Server released under GPL, able to handle thousands of call setups per second. Among features: asynchronous TCP, UDP and SCTP, secure communication via TLS for VoIP (voice, video), SIMPLE instant messaging and presence, ENUM, least cost routing, load balancing, routing fail-over, accounting, authentication and authorization against MySQL, Postgres, Oracle, Radius, LDAP, XMLRPC control interface, SNMP monitoring. It can be used to build large VoIP servicing platforms or to scale up SIP-to-PSTN gateways, PBX systems or media servers like Asterisk™, FreeSWITCH™ or SEMS.

VOIP Today magazine spoke with Daniel-Constantin Mierla about Business , technology and products .

[posted by Allison Smith on Digium blogs] I am not an engineer of IVR systems; I do not write code, and I certainly don’t program or implement IVR systems. I am a professional voice talent who specializes in voicing IVR and auto-attendant systems — I have voiced platforms for large telcos and independent companies and organizations internationally; and I am probably best known for my work voicing the prompts for Asterisk, practically since its inception.

The VBS Fax Server by VoIP Business Solutions is suitable for anyone who needs a solid, easy to use fax server system using the Asterisk PBX. From the smallest businesses and individual users on up, the VBS system is a full-featured industrial-strength fax server that does what you need at a fraction of the cost. It can be used as both a fax broadcasting system or can be used to send a single fax. With the VBS fax server, sending faxes to one recipient or one thousand has never been easier. This incredibly easy-to-use, yet powerful software package completely eliminates the tedious, time consuming, and error prone human labor of repetitive faxing while freeing up your staff to work on other tasks.

[posted by Amy Nutt - EzineArticles.com]  Asterisk phone systems are an excellent, fully integrated VOIP PBX solution for businesses of all sizes. The set up eliminates the tremendous cost of setting up a traditional PBX system. This high cost often eliminates the possibility of using PBX for small businesses, but with Asterisk, even the smallest business can have all of the benefits of PBX in their office without spending money they do not have.

What Is Asterisk?

Asterisk is an open source software PBX that can be used by businesses and individuals alike. It is designed to run on Linux, but it can run on other platforms as well, including Windows. Asterisk was developed by Digium, Inc., who also sells the hardware necessary to set up Asterisk phone systems. Compared to other PBX options, Asterisk is incredibly affordable and has many features that are hard to find elsewhere.