Asterisk 1.6.2.15 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.6.2.15.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.15 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
- When using chan_skinny, don’t crash when parking a non-bridged call.
(Closes issue #17680. Reported, tested by jmhunter. Patched, tested by DEA) - Add ability for Asterisk to try both the encoded and unencoded subscription URI for a match in hints.
(Closes issue #17785. Reported, tested by ramonpeek. Patched by tilghman) - Set the caller id on CDRs when it is set on the parent channel.
(Closes issue #17569. Reported, patched by tbelder) - Ensure user portion of SIP URI matches dialplan when using encoded characters
(Closes issue #17892. Reported by wdoekes. Patched by jpeeler) - Resolve issue where Party A in an analog 3-way call would continue to hear ringback after party C answers.
(Patched by rmudgett) - Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time.
(Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by jpeeler) - Multiple fixes related to Local channels.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.15
Thank you for your continued support of Asterisk!
Asterisk 1.4.38 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.4.38. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.38 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
- Add ability for Asterisk to try both the encoded and unencoded subscription URI for a match in hints.
(Closes issue #17785. Reported, tested by ramonpeek. Patched by tilghman) - Set the caller id on CDRs when it is set on the parent channel.
(Closes issue #17569. Reported, patched by tbelder) - Ensure user portion of SIP URI matches dialplan when using encoded characters
(Closes issue #17892. Reported by wdoekes. Patched by jpeeler) - Fix a crash in res_jabber by ensuring that we don’t alter memory after it’s freed.
(Closes issue #17387. Reported, tested by jmls. Patched by tilghman) - Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time.
(Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by jpeeler) - Multiple fixes related to Local channels.
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.4.38
Thank you for your continued support of Asterisk!
AstLinux 0.7.4 Release
astlinux
The AstLinux Team is happy to announce the next release from the 0.7 branch. There are several security updates and fixes. All current AstLinux users should upgrade immediately.
This release is a dual release with both stable versions of Asterisk. You have your choice of using Asterisk 1.4.36 or Asterisk 1.8.0. While Asterisk 1.8.0 is considered stable by Digium, you will want to do thorough testing before using it in production.
Asterisk 1.8.1-rc1 Now Available
Posted by admin in asterisk, Release Candidates on November 23, 2010
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the first release candidate of Asterisk 1.8.1. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.1-rc1 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release:
- Fix issue when using directmedia. Asterisk needs to limit the codecs offered to just the ones that both sides recognize, otherwise they may end up sending audio that the other side doesn’t understand.
(Closes issue #17403. Reported, patched by one47. Tested by one47, falves11) - Resolve issue where Party A in an analog 3-way call would continue to hear ringback after party C answers.
(Patched by rmudgett) - Fix playback failure when using IAX with the timerfd module.
(Closes issue #18110. Reported, tested by tpanton. Patched by jpeeler) - Fix problem with qualify option packets for realtime peers never stopping. The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time.
(Closes issue #16382. Reported by lftsy. Tested by zerohalo. Patched by jpeeler) - Fix issue where it is possible to crash Asterisk by feeding the curl engine invalid data.
(Closes issue #18161. Reported by wdoekes. Patched by tilghman)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1-rc1
Thank you for your continued support of Asterisk!
libpri 1.4.11.5 Now Available
The Asterisk Development Team has announced the release of libpri 1.4.11.5.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/libpri/
The release of libpri 1.4.11.5 resolves several issues reported by the community and would not have been possible without your participation.
Thank you!
The following are some of the issues resolved in this release:
- Prevent a CONNECT message from sending a CONNECT ACKNOWLEDGE in the wrong state.
(issue #17360. Reported by: shawkris. Patched by rmudgett) - Made Q.921 delay events to Q.931 if the event could immediately generate response frames.
(closes issue #17360. Reported by: shawkris. Patched by rmudgett) - BRI PTMP: Active channels not cleared when the interface goes down.
(closes issue #17865. Reported by: wimpy. Patched by rmudgett) - Segfault in pri_schedule_del() – ctrl value is invalid.
(closes issue #17522)
(closes issue #18032. Reported by: schmoozecom. Patched by rmudgett) - Crash when receiving an unknown/unsupported message type.
(closes issue #17968. Reported by: gelo. Patched by rmudgett) - * B410P gets incoming call packets on ISDN but Asterisk doesn’t see the call.
(closes issue #18232. Reported by: lelio. Patched by rmudgett)
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.11.5
Thank you for your continued support of Asterisk!
Elastix and Kamailio Part 1 – Installing
WARNING: DO NOT INSTALL THIS ON A PRODUCTION MACHINE! Some of you may want be wanting to increase the capacity of your Elastix machine. Asterisk generally has issues when you start to put a lot of SIP peers on it. One of the best ways to get around this is to install a proper SIP server and join it with asterisk. This allows you to register all your SIP handsets to the SIP server and make calls through to your asterisk machine.
The Kamailio module for Elastix allows you to easily integrate Elastix with a enterprise SIP server. All SIP handsets register to Kamailio. All internal SIP calls stay within Kamailio however if a call is external then Kamailio will push the call out to asterisk to deal with it.
To install the Kamalio module you first need to create the directory for it to go in. It installs itself from the directory /var/kamailio
mkdir /var/kamailio
Now change to the directory
cd /var/kamailio
Download the kamailio integration file from the MBIT website
wget http://www.mbit.com.au/kamailio-v2.tgz
Untar the file into the directory
tar zxvf kamailio-integration.tgz
In the directory now will be a heap of files you want to execute the script that installs all the modules for you. To run the script execute install_kamailio.sh
./install_kamailio.sh
Soon as it is installed we will move to the next part for configuring Kamailio. The installation can take a while so please be patient. The installation can only be used with Elastix 2.0.
Asterisk 1.8.0 Now Available!
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team is proud to announce the release of Asterisk 1.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
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
The release of Asterisk 1.8.0 would not have been possible without the support and contributions of the community. Since Asterisk 1.6.2, we’ve had over 500 reporters, more than 300 testers and greater than 200 developers contributed to this release.
You can find a summary of the work involved with the 1.8.0 release in the sumary:
http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt
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!
A full list of new features can be found in the CHANGES file.
http://svn.digium.com/view/asterisk/branches/1.8/CHANGES?view=markup
For a full list of changes in the current release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0
Thank you for your continued support of Asterisk!
AstLinux 0.7.3 release
astlinux
The next release in the 0.7 branch is ready for use. There are several security updates and fixes included in this release. All AstLinux users are encouraged to update to this release.
The AstLinux Team is happy to announce the release of 0.7.3.
Please read the following pages before attempting to upgrade from a previous release.
Select which target most closely matches your system. The bootable ISO images are intended for testing. For a new installation, use one of the install images.
**Please note that we are aware of issues booting from SATA and USB devices and hope to correct those issues before the next release**
As with the 0.7.2 release, sound files are not included in the base image. You MUST install sound files using the instructions on the following page:
Read the rest of this entry »
Asterisk 1.6.2.14-rc1 Now Available
Posted by admin in asterisk, Release Candidates on September 21, 2010
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the first release candidate of Asterisk 1.6.2.14. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.14-rc1 resolves several issues reported by the community and would have not been possible without your participation.
Thank you!
The following is a sample of the issues resolved in this release candidate:
- Fix issue where session timers would be advertised as supported even when session-timers=refuse was set in sip.conf. Also fix interoperability problems with session timer behavior in Asterisk.
(Closes issue #17005. Reported by alexcarey. Patched by dvossel) - Fix issue with decoding ^-escaped characters in realtime (res_pgsql).
(Closes issue #17790. Reported by denzs. Patched by Qwell) - Parse all “Accept” headers for SIP SUBSCRIBE requests.
(Closes issue #17758. Reported by ibc. Patched by dvossel) - Fix issue where queue stats would be reset on reload.
(Closes issue #17535. Reported by raarts. Patched by tilghman) - Fix issue where MoH files were no longer rescanned on during a reload.
(Closes issue #16744. Reported by pj. Patched by Qwell) - Fix issue with dialplan pattern matching where the specificity for pattern ranges and pattern characters was inconsistent.
(Closes issue #16903. Reported, patched by Nick_Lewis)
For a full list of changes in the current release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.6.2.14-rc1
Thank you for your continued support of Asterisk!
AsteriskNOW 1.7.1 now available with add-on installer module for FreePBX
Asterisk NOW
AsteriskNOW 1.7.1 has been released with a new module for FreePBX that allows installation of Digium add-on software from within the web-based interface. Now there’s no command-line work to be done to get Digium’s G729 codec, Fax for Asterisk, HPEC, or Skype for Asterisk. We’ve also made some changes to make AsteriskNOW even friendlier for newcomers to the Asterisk community.
Together with the DAHDI configuration module that began shipping with 1.7.0, these modules make Asterisk administration even easier. Download AsteriskNOW today, and burn it to a CD or start it in a virtual machine. In minutes you can turn your computer into your next phone system!
