Gitex 2010:The Rise of Open Source Telephony in Middle East

You are expected to see a new face in Gitex 2010 Dubai this week: OpenVox has been presented in the Zabeel Hall in booth Z-P31. It’s the first time that OpenVox shows up in the Gitex in the Middle East Marketing.  And the some of OpenVox newly released products is being presented in this show including AE2410P, IPC100 for a better choice to build a best cost effective Open Source telephony solutions.

GITEX TECHNOLOGY WEEK is at the heart of the global technology network. Experience a spectacular celebration of interactive features, a showcase of leading technology brands and an unrivalled range of networking opportunities as GITEX celebrates 30 years of industry integration and leadership. Review and select from thousands of leading technology brands and industry experts.

Who else is to see in Gitex 2010? DVCOM is presenting Digium, Snom,Patton in its booth. You are not limited with your choice of asterisk products since OpenVox has already been here for you.

ElastixWorld 2010: Day One

Due to the time difference, it’s now the second day of Elastixworld 2010. But I am lucky that I just get some pictures from Jorge Gomez in VoPERo. VoPERo is the platimun sponsor for Elastixworld 2010 and they are presenting OpenVox during this great show. And I think Edgar must be very busy with the show because he is the man for Elastixworld! Well, let’s take a look at what’s going on in the Elastixworld.

ElastixWorld 2010:The Happy Hours & Days

The first edition of elastixWorld ended and left the community with great satisfaction.  But not every one has the time and chance to celebrate this great event onsite. Thanks to the Elastix community and Jorge Gomez, we have the chance to share something here with all of you. Some great moments to remember and to cherish.

I’ve got a lot of pictures from Jorge (It must take him a lot of time to send out the orginal pictures through email. Thank you so much Jorge! ), and Edgar as well. And I want to share some here to make sure we don’t miss what’s happening in the Elastix World 2010.

Jorge Gomez, Rodrigo Martin from 0351 Grupo, Luis Almeida and edgar Landivar from PaloSanto, Leonardo Hermidia from 0351 Grupo, Juan Pablo Romero from Audiocodes.

Osmar sepulveda, Edgar Landivar and Paul Estrella

Paul Estrella, Doug Vilim, Alfio Muñoz, Edgar landivar and Rafael Bonifaz

Enjoy the show and Cheers Elastix World 2010!

Elastix and OpenVox

Well, let me introduce one of my friends to you all,Edgar Landívar, the founder of Elastix. We’ve known each other for years but we never met before. And finally, this time, in astricon 2010, we met!!

Here we go:

By the way, the Elastixworld 2010 is just around the corner! Are you ready for that?

OpenVox Announces Free Support for Asterisk Configuration

OpenVox has announced today to provide free support for asterisk configuration to the open source community. It can surely lower the technical entrance to the open source telephony for the new users.

OpenVox is based on open source, and we are working with asterisk, Askozia, Elastix, FreePBX, FreeSWITCH, PBX in a flash, trixbox… You’re always not limited by your choice of application. But some of you may step back when you are new to open source or don’t know what to do with it. And now you can get the real free technical support to make your telephony system work. There will no longer be any worries about the complex system configurations and settings.

What Do We Have for Free Support?
If you are installing OpenVox hardware with the following software:
Asterisk, Askozia, Elastix, FreePBX, PBX in a flash, trixbox.
Then, you can have the following free support from OpenVox:
*Product Driver Installation & Configuration
*Setting for Inbound/Outbound Routing
*SIP/IAX Extension Configuration (A few extensions to ensure the system working properly)

More will be added to the list soon. If you have any suggestions for the list or something else you think will be good, please don’t hesitate to email us at james.zhu@openvox.cn.

What Kind of Info Needed for the Support?
To access the free support, we will require you to provide the following information to us:

*Name of the Product
*Serial Number for the Product
*The Name of the Software and its Version
*SSH Account and GUI Login Info
*What kind of free support do you need from us?
*When will be your best available time for us to get it done for you? Please kindly note that our working hours is from 9:00 -18:00 (GMT +8) even though we can make it as flexible as possible to fit yours.

Please email the above information to James Zhu at james.zhu@openvox.cn.

When Will You Have the Support?
We will get back to you within one or two business days once your email is received. And the exact time will be confirmed with you afterwards. Well, enjoy the open source telephony!

Will Asterisk 1.8.0 Stable Version Be Far Away?

It’s been quite a while since asterisk team released  the first candidate  of asterisk 1.8.0 and just now the 3rd one is announced to the open source community. So, will the asterisk 1.8.0 stable one be far away? Let’s wait and see!

Asterisk 1.8 is the next major release series of Asterisk. It will be a Long Term Support (LTS) release, similar to Asterisk 1.4. For more information about support time lines for Asterisk releases, see the Asterisk versions page.

http://www.asterisk.org/asterisk-versions

With the availability of the Asterisk 1.8.0 release candidates, the binary add-on modules for Asterisk produced by Digium have been updated with new versions that are compatible with Asterisk 1.8. The availability of these modules will assist with the testing of Asterisk 1.8.0 in a wider variety of situations.

This release candidate contains fixes since the release candidate as reported by the community. A sampling of the changes in this release candidate include:

•Still build chan_sip even if res_crypto cannot be built (use, but not depend)
(Reported by a user on the mailing list. Patched by tilghman)
•Get notifications for call files only when a file is closed, not when created
(Closes issue #17924. Reported by mkeuter. Patched by abeldeck)
•Fixes to chan_gtalk to allow outbound DTMF support to work correctly. Gtalk expects the DTMF to arrive on the RTP stream and not via jingle DTMF signalling.
(Patched by dvossel. Tested by malcolmd)
•Fixes to allow chan_gtalk to communicate with the Gmail web client.
(Patched by phsultan and dvossel)
•Fix to GET DATA to allow audio to be streamed via an AGI.
(Closes issue #18001. Reported by jamicque. Patched by tilghman)
•Resolve dnsmgr memory corruption in chan_iax2.
(Closes issue #17902. Reported by afried. Patched by russell, dvossel)

A short list of available features includes:

•Secure RTP
•IPv6 Support in the SIP channel driver
•Connected Party Identification Support
•Calendaring Integration
•A new call logging system, Channel Event Logging (CEL)
•Distributed Device State using Jabber/XMPP PubSub
•Call Completion Supplementary Services support
•Advice of Charge support
•Much, much more!

Digium TDM410P Analog Interface Card Overview

The Digium® TDM410 is a half-length PCI 2.2-compliant modular gateway card for connecting analog telephone stations and analog POTS lines through a PC. It supports a combination of up to four station or trunk modules for a total of 4 lines.

Using Digium’s Asterisk® software and standard PC hardware, one can create a telephony environment that includes all of the sophisticated features of a high-end business telephone system.

Using an industry-standard bursting, bus-mastering interface chip that is found within millions of PC systems worldwide, and Digium VoiceBus™ technology, the TDM410 eliminates the requirement for external gateways, with industry-leading performance and price. The trunk and station modules are interchangeable, allowing the creation of any combination of interfaces. The optional hardware echo cancellation module provides 1024 taps (128 milliseconds) of echo cancellation for superior voice quality on both trunk and station interfaces. Scaling of this solution is accomplished by adding additional TDM410 or other Digium analog interface cards.

Digium Analog cards, with there interchangeable single and quad FXS and FXO modules can eliminate the requirement for seperate channel banks or access gateways. Digiums commercial, toll-free quality High Performance Echo Cancellation (HPEC) software is available to our analog customers at no additional cost. The optional VPMADT032 hardware echo cancellation module provides the same toll-quality HPEC but without the perfoamce inpact of a software based solution. Scaling of an analog card solution is accomplished by adding additional cards and FXO / FXS modules.

Core Features

4 Ports for connecting analog telephones or POTS linesHalf-length Analog CardUp to 4 FXS or FXO ModulesHigh Performance Echo Cancellation (HPEC) Software (Optional)TDM410 for use with a PCI 2.2 compliant slot