Posts Tagged 1.8
Asterisk 1.8.2 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.8.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.2 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:
- ‘sip notify clear-mwi’ needs terminating CRLF.
(Closes issue #18275. Reported, patched by klaus3000) - Patch for deadlock from ordering issue between channel/queue locks in app_queue (set_queue_variables).
(Closes issue #18031. Reported by rain. Patched by bbryant) - Fix cache of device state changes for multiple servers.
(Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested by russellb) - Resolve issue where channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.
(Closes issue #18171. Reported by: SantaFox)
(Closes issue #18185. Reported by: kwemheuer)
(Closes issue #18211. Reported by: zahir_koradia)
(Closes issue #18230. Reported by: vmarrone)
(Closes issue #18299. Reported by: mbrevda)
(Closes issue #18322. Reported by: nerbos) - Fix reloading of peer when a user is requested. Prevent peer reloading from causing multiple MWI subscriptions to be created when using realtime.
(Closes issue #18342. Reported, patched by nivek.) - Fix XMPP PubSub-based distributed device state. Initialize pubsubflags to 0 so res_jabber doesn’t think there is already an XMPP connection sending device state. Also clean up CLI commands a bit.
(Closes issue #18272. Reported by klaus3000. Patched by Marquis42) - Don’t crash after Set(CDR(userfield)=…) in ast_bridge_call. Instead of setting peer->cdr = NULL, set it to not post.
(Closes issue #18415. Reported by macbrody. Patched, tested by jsolares) - Fixes issue with outbound google voice calls not working. Thanks to az1234 and nevermind_quack for their input in helping debug the issue.
(Closes issue #18412. Reported by nevermind_quack. Patched by dvossel)
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.2
Thank you for your continued support of Asterisk!
Asterisk 1.8.1.1 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.8.1.1.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.1.1 resolves two issues reported by the community since the release of Asterisk 1.8.1.
- Don’t crash after Set(CDR(userfield)=…) in ast_bridge_call. Instead of setting peer->cdr = NULL, set it to not post.
(Closes issue #18415. Reported by macbrody. Patched, tested by jsolares) - Fixes issue with outbound google voice calls not working. Thanks to az1234 and nevermind_quack for their input in helping debug the issue.
(Closes issue #18412. Reported by nevermind_quack. Patched by dvossel)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1.1
Thank you for your continued support of Asterisk!
Asterisk 1.8.2-rc1 Now Available
Posted by admin in asterisk, Release Candidates on December 15, 2010
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the first release candidate of Asterisk 1.8.2. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.2-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:
- ‘sip notify clear-mwi’ needs terminating CRLF.
(Closes issue #18275. Reported, patched by klaus3000) - Patch for deadlock from ordering issue between channel/queue locks in app_queue (set_queue_variables).
(Closes issue #18031. Reported by rain. Patched by bbryant) - Fix cache of device state changes for multiple servers.
(Closes issue #18284, #18280. Reported, tested by klaus3000. Patched, tested by russellb) - Resolve issue where channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.
(Closes issue #18171. Reported by: SantaFox)
(Closes issue #18185. Reported by: kwemheuer)
(Closes issue #18211. Reported by: zahir_koradia)
(Closes issue #18230. Reported by: vmarrone)
(Closes issue #18299. Reported by: mbrevda)
(Closes issue #18322. Reported by: nerbos) - Fix reloading of peer when a user is requested. Prevent peer reloading from causing multiple MWI subscriptions to be created when using realtime.
(Closes issue #18342. Reported, patched by nivek.) - Fix XMPP PubSub-based distributed device state. Initialize pubsubflags to 0 so res_jabber doesn’t think there is already an XMPP connection sending device state. Also clean up CLI commands a bit.
(Closes issue #18272. Reported by klaus3000. Patched by Marquis42) - Don’t crash after Set(CDR(userfield)=…) in ast_bridge_call. Instead of setting peer->cdr = NULL, set it to not post.
(Closes issue #18415. Reported by macbrody. Patched, tested by jsolares) - Fixes issue with outbound google voice calls not working. Thanks to az1234 and nevermind_quack for their input in helping debug the issue.
(Closes issue #18412. Reported by nevermind_quack. Patched by dvossel)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.2-rc1
Thank you for your continued support of Asterisk!
Asterisk 1.8.1 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.8.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.8.1 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, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.1
Thank you for your continued support of Asterisk!
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!
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!
Asterisk 1.8.0-Beta3 Now Available
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta3.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, http://issues.asterisk.org/. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.
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
This release contains fixes since the last beta release as reported by the community. A sampling of the changes in this release include:
- Fix a regression where HTTP would always be enabled regardless of setting.
(Closes issue #17708. Reported, patched by pabelanger) - ACL errors displayed on screen when using dynamic_exclude_static in sip.conf
(Closes issue #17717. Reported by Dennis DeDonatis. Patched by mmichelson) - Support “channels” in addition to “channel” in chan_dahdi.conf.
(https://reviewboard.asterisk.org/r/804) - Fix parsing error in sip_sipredirect(). The code was written in a way that did a bad job of parsing the port out of a URI. Specifically, it would do badly when dealing with an IPv6 address.
(Closes issue #17661. Reported by oej. Patched by mmichelson) - Fix inband DTMF detection on outgoing ISDN calls.
(Patched by russellb and rmudgett) - Fixes issue with translator frame not getting freed. This issue prevented g729 licenses from being freed up.
(Closes issue #17630. Reported by manvirr. Patched by dvossel) - Fixed IPv6-related SIP parsing bugs and updated documention.
(Reported by oej. Patched by sperreault) - Add new, self-contained feature FIELDNUM(). Returns a 1-based index into a list of a specified item. Matches up with FIELDQTY() and CUT().
(Closes #17713. Reported, patched by gareth. Tested by tilghman)
Asterisk 1.8 contains many new features over previous releases of Asterisk.
A short list of included features includes:
- Secure RTP
- IPv6 Support in the SIP Channel
- 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=checkout
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta3
Thank you for your continued support of Asterisk!
Asterisk 1.8.0-Beta2 Now Available
Posted by admin in asterisk, Release Candidates on July 27, 2010
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta2.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, http://issues.asterisk.org/. It is also very useful to see successful test reports. Please post those to the asterisk-dev mailing list.
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
This release contains fixes since the last beta release as reported by the community. Some of the changes include:
- Remove duplicate -c flag when using $(INSTALL)
(Closes issue #17695. Reported, patched by pabelanger) - Don’t re-register CDR module on reload.
(Closes issue #17304. Reported, tested by jnemeth. Patched by tilghman) - Don’t assume qlog is open.
(Closes issue #17704. Reported, tested by vrban. Patched by pabelanger) - Expand the correct value within AST_OPTION_ONLY.
(Closes issue #17703. Reported by stuarth. Patched by seanbright) - Allow for systems without locale support to be usable.
(Closes issue #17697. Reported, patched by pprindeville. Tested by mmichelson) - Fixes for sounds/Makefile to install on systems using older GNU make.
(Closes issue #17716. Reported by farisraouf. Patched by tilghman, qwell, seanbright) - Update logger.conf.sample to include documentation about new ‘fax’ logger level.
(Closes issue #17715. Reported, tested by vrban. Patched by pabelanger)
Asterisk 1.8 contains many new features over previous releases of Asterisk.
A short list of included features includes:
- Secure RTP
- IPv6 Support
- 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=checkout
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta2
Thank you for your continued support of Asterisk!
Asterisk 1.8, now in Beta
Howdy,
Digium is pleased to announce the availability of the first beta of Asterisk 1.8. Asterisk 1.8, the next major release of Asterisk, is a Long-Term Support (LTS) release and will be fully supported for 4 years, with an additional year of security-only fixes. For more information about support timelines for Asterisk releases, see the Asterisk versions page:
http://www.asterisk.org/asterisk-versions
A short list of new features that are a part of Asterisk 1.8 includes:
- Secure RTP (SRTP)
- IPv6 Support
- Connected Party Identification Support
- Calendaring Integration for CalDAV, iCal, Exchange or EWS calendars
- A new call logging system, Channel Event Logging (CEL)
- Distributed Device State, including Message Waiting Indicator using Jabber/XMPP PubSub
- Call Completion Supplementary Services (CCSS) Support, including Call Completion on Busy Subscriber (CCBS) and Call Completion on No Response (CCNR)
- Advice of Charge, including AOC-S, AOC-D, and AOC-E Support
- and much, much more!
A full list of new features can be found in the CHANGES file.
How can you help?
By testing, testing and more testing! We need your help finding bugs so that 1.8 is the best release possible. To help, download the beta:
http://www.asterisk.org/downloads/asterisk/releases/asterisk-1.8.0-beta1.tar.gz
Build and install it, and test, test, test!
Report all issues to our issue tracker:
Thank you for your assistance and for your continued support of Asterisk!
Asterisk 1.8.0-Beta1 Now Available
Posted by admin in asterisk, Release Candidates on July 23, 2010
Asterisk The Open Source PBX & Telephony Platform
The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1.
This release marks the beginning of the testing process for the eventual release of Asterisk 1.8.0.
This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/
All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any issues found to the issue tracker, http://issues.asterisk.org/. It is also very useful to hear successful test reports. Please post those to the asterisk-dev mailing list.
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
Asterisk 1.8 contains many new features over previous releases of Asterisk.
A short list of included features includes:
- Secure RTP
- IPv6 Support
- 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=checkout
For a full list of changes in the current release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.0-beta1
Thank you for your continued support of Asterisk!
