Archive for July, 2009
Skype for Asterisk: Public Beta available
Skype For Asterisk
I know many of you have been waiting for this for a while, so I’ll keep this short: The Skype for Asterisk Public Beta is now available on the Digium store.
We are pleased to announce the open beta of Skype For Asterisk is ready to begin and we look forward to you participation. To obtain your copy of the software, please visit Digium’s web store and purchase (for zero dollars) the Skype For Asterisk product. The web store does require a Digium.com account, which can be set up during the purchase process if you don’t already have one.
Once the web store process is complete, you will be e-mailed your license key and directions on where to download Skype For Asterisk beta software.
This is a “time-expiring” beta – the software will stop working on August 31. The download is also currently time-limited – it will be available until August 7 on our website. After the 31st, you would need to have purchased a license for the Skype For Asterisk software (sorry, no pricing that I can give you right now – that will be a separate announcement). I’m just the community guy – I have no idea about pricing or commercial contracts or the like, so please wait until that’s been announced as I will find out about the same time as you do.
Trial “purchase” page:
News in the list asterisk-dev:
News in the list asterisk-users:
Asterisk-Addons 1.4.9, 1.6.0.3, and 1.6.1.1 released
Posted by admin in asterisk addons, Releases on July 24, 2009
The Asterisk Development Team has announced several Asterisk-Addons releases, including Asterisk-Addons 1.4.9, 1.6.0.3, and 1.6.1.1. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
These releases are an incremental release after some community reported issues were resolved, primarily in the MySQL and chan_mobile realms.
- Using chan_local with chan_mobile (issue #15299, affects all 1.6.x versions)
- Don’t reset a reconnect time unless a reconnect really occurred (issue #15375, affects all versions)
For a full list of changes in these releases, please see the ChangeLogs:
http://svn.asterisk.org/svn/asterisk-addons/tags/1.4.9/ChangeLog
http://svn.asterisk.org/svn/asterisk-addons/tags/1.6.0.3/ChangeLog
http://svn.asterisk.org/svn/asterisk-addons/tags/1.6.1.1/ChangeLog
Thank you for your continued support of Asterisk!
Asterisk 1.4.26 released
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team is pleased to announce the release of Asterisk 1.4.26.
Asterisk 1.4.26 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
This release resolves a large assortment of issues reported by the community.
For a summary of the changes in this release, please see the release summary:
http://svn.asterisk.org/svn/asterisk/tags/1.4.26/asterisk-1.4.26-summary.txt
For a full list of changes in this release, please see the ChangeLog:
http://svn.asterisk.org/svn/asterisk/tags/1.4.26/ChangeLog
The following list of issues were resolved with the participation of the community, and this release would not have been possible without your help!
- Fix handling of the ‘state_interface’ option of the ‘queue add member’ CLI command.
(closes issue #15181. Reported and tested by loloski. Patch by seanbright) - Fix a possible crash in pbx_spool.
(closes issue #15072. Reported, and patched by garlew) - MWI NOTIFY contains a wrong URI if Asterisk listens to non-standard port and transport.
(closes issue #14659. Reported, patch, and testing by klaus3000) - Don’t fast forward past the end of a message.
(closes issue #14554. Reported and patched by lacoursj) - Prevent phantom calls to queue members.
(closes issue #14631. Reported and patched by latinsud) - No audio on calls from Asterisk to various ISDN devices until DTMF sent by caller.
(closes issues #15420, #15416, #15389, #15205. Reported by scottbmilne, avinoash, alecdavis, vinsik. Tested by scottbmilne, alecdavis. Patched by alecdavis)
Thank you for your continued support of Asterisk!
A Preview of Elastix 2.0
Elastix - The reliable PBX appliance software
It has been a while since the release of Elastix 1.0. There have been a lot of ideas since then that were scheduled for the next release. Meanwhile, we released the Call Center and Developer modules as GPL and a lot of other useful add-ons. These releases have brought satisfaction to all of us in Elastix. Our community has grown and our downloads are almost half a million in total.
Finally, the wait for Elastix 2.0 is over, and time for new ideas has come. We have started development of this new version and we will try to include as many ideas and suggestions from the community as possible.
For those of you that just can’t wait, here is a preview of some of the main features that have been scheduled for Elastix 2.0:
- Asterisk 1.6
- Elastix Operator Panel release
- Meeting / Presentation sharing
- Advanced reports
- Improvements to the Call Center Module
- Better integration between all components
Asterisk 1.4.26-rc6 released
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the sixth release candidate of Asterisk 1.4.26. Asterisk 1.4.26-rc6 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
This release resolves the following issues, in addition to some minor issues:
- Properly ACK 487 responses to canceled INVITEs. Reviewboard https://reviewboard.asterisk.org/r/308
- SIP registration authorization loop caused by stale nonce (issue #15102)
- Ensure outbound NOTIFY requests are properly routed through stateful proxies
(issue #14725) - Resolve no audio on chan_dahdi calls until DTMF is sent by caller (issues #15416, #15389, #15205)
- Removed confusing warning message in chan_misdn (issue #11974)
For a full list of changes in this release candidate, please see the ChangeLog: http://svn.asterisk.org/svn/asterisk/tags/1.4.26-rc6/ChangeLog
Issues found in this release candidate can be reported at https://issues.asterisk.org
Thank you for your continued support of Asterisk!
Asterisk-Addons 1.4.9-rc1, 1.6.0.3-rc2, 1.6.1.1-rc2, and 1.6.2.0-rc1 released
Posted by admin in asterisk addons on July 14, 2009
The Asterisk Development Team has announced several Asterisk-Addons release candidates, including Asterisk-Addons 1.4.9-rc1, 1.6.0.3-rc2, 1.6.1.1-rc2, and 1.6.2.0-rc1. The release candidates are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
These releases are an incremental release after some community reported issues were resolved, primarily in the MySQL and chan_mobile realms.
- Using chan_local with chan_mobile (issue #15299, affects all 1.6.x versions)
- Don’t reset a reconnect time unless a reconnect really occurred (issue #15375, affects all versions)
For a full list of changes in this release candidate, please see the ChangeLogs:
http://svn.asterisk.org/svn/asterisk-addons/tags/1.4.9-rc1/ChangeLog
http://svn.asterisk.org/svn/asterisk-addons/tags/1.6.1.1-rc2/ChangeLog
http://svn.asterisk.org/svn/asterisk-addons/tags/1.6.2.0-rc1/ChangeLog
Issues found in any release candidate can be reported at:
https://issues.asterisk.org
Thank you for your continued support of Asterisk!
Fax For Asterisk
T.38 fax for Asterisk
Digium‘s Fax For Asterisk is a commercial facsimile (Fax) termination and origination solution designed to enhance the capabilities of Open Source and commercial Asterisk as well as Switchvox. Fax For Asterisk bundles a suite of user-friendly Asterisk applications and a licensed version of the industry’s leading fax modem software from Commetrex. Fax For Asterisk provides low speed (14400bps) PSTN faxing via DAHDI-compatible telephony boards as well as VoIP faxing to T.38-compatible SIP endpoints and service providers. Licensed on a per-channel basis, Digium‘s Fax For Asterisk provides a complete, cost-effective, commercial fax solution for Asterisk users.
Fax For Asterisk provides two components: res_fax and res_fax_digium. Res_fax is an Asterisk resource module that adds fax termination and origination functionality in Asterisk. It provides Asterisk dialplan functions and dialplan applications to enable the user to build highly-customizable fax solutions. Res_fax_digium provides core fax processing functionality in the form of several supported fax modems — V.21, V.27ter, V.29, and V.17 — to achieve speeds up to 14400bps.
Asterisk 1.4.26-rc5 released
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the fifth release candidate of Asterisk 1.4.26. Asterisk 1.4.26-rc5 is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
This release resolves the following issues, plus some other minor issues:
* Improve the mapping of extension states from combined device states (issue #15413).
* Added an error message to make it clear why a SIP peer was not processed when a DNS lookup fails on a host or outbound proxy (issue #13432).
* Fix a problem where chan_sip would ignore “old” but valid responses (issue #11231).
* Resolve a crash in chan_iax2 (issue #15377).
* Resolve a crash related to a T.38 reinvite race condition.
For a full list of changes in this release candidate, please see the ChangeLog:
http://svn.asterisk.org/svn/asterisk/tags/1.4.26-rc5/ChangeLog
Issues found in this release candidate can be reported at https://issues.asterisk.org
Thank you for your continued support of Asterisk!
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