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回復 104# bubblestar


Login problem still exists for IE.

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回復 106# ckleea



As you said, there is still problem to access via IE.

I have been using APL to handle all Asterisk configuration and operation for long time.  It is no easy task to do the integration.  I have to examine line by line, word by word to ensure all settings are function as before.  As you know, the naming rules are quite different with GUI and I'm considering whether I should do it all from scratch.  But it takes time......

而且可能會牽一髮而動全身,要詳細考慮一下。

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問題係:免費的software 已經用咗 (asterisk),係時間學多D programming,整俾自已用,我相信很難再有新 GUI update

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我剛好是相反情況,用咗咁耐Asterisk Programming Language,現在要加入GUI,真是有 D 考起自己。或這會搵一些第三方軟件(當然要免費的),主要以GUI監察一下使用情況罷了,沒有太大期望。  如果找不到的話,APL也一定會是主力,不會放棄。因為已經做咗大量customized dialplan 俾自己用,好難一下子轉回用GUI 的。

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可以的話,請分享你的customized dialplan,學習學習。

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The Asterisk Development Team has announced the release of Asterisk 1.8.4.1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.4.1 resolves several issues reported by the
community. Without your help this release would not have been possible.
Thank you!

Below is a list of issues resolved in this release:

* Fix our compliance with RFC 3261 section 18.2.2. (aka Cisco phone fix)
  (Closes issue #18951. Reported by jmls. Patched by wdoekes)

* Resolve a change in IPv6 header parsing due to the Cisco phone fix issue.
  This issue was found and reported by the Asterisk test suite.
  (Closes issue #18951. Patched by mnicholson)

* Resolve potential crash when using SIP TLS support.
  (Closes issue #19192. Reported by stknob. Patched by Chainsaw. Tested by
   vois, Chainsaw)

* Improve reliability when using SIP TLS.
  (Closes issue #19182. Reported by st. Patched by mnicholson)


For a full list of changes in this release candidate, please see the ChangeLog:

http://downloads.asterisk.org/pu ... k/ChangeLog-1.8.4.1

Thank you for your continued support of Asterisk!

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最近update 都比較快啊

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回復 112# bubblestar

有空的時候,請看下面網頁的文章

http://nerdvittles.com/?p=749  

更新快未必是件好事!

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回復 113# ckleea


   
That is why I appreciate what Elastix is doing in its own ways, steps and schedule to integrate the latest Asterisk.  They won't upgrade too fast to follow Digium's release.  All the Elastix versions are a bit 慢半拍 so that they can have enough time to test by themselves first before rolling out.  As you may also aware, Elastix just releases its Asterisk 1.8 which is almost half-year slower than Digium's debut.

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The Asterisk Development Team has announced the release of Asterisk version 1.8.4.2, which is a security release for Asterisk 1.8.

This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases

The release of Asterisk 1.8.4.2 resolves an issue with SIP URI parsing which can lead to a remotely exploitable crash:

   Remote Crash Vulnerability in SIP channel driver (AST-2011-007)

The issue and resolution is described in the AST-2011-007 security
advisory.

For more information about the details of this vulnerability, please read the security advisory AST-2011-007, which was released at the same time as this announcement.

For a full list of changes in the current release, please see the ChangeLog:

http://downloads.asterisk.org/pu ... s/ChangeLog-1.8.4.2

Security advisory AST-2011-007 is available at:

http://downloads.asterisk.org/pub/security/AST-2011-007.pdf

Thank you for your continued support of Asterisk!

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Asterisk 1.4.41.1, 1.6.2.19-rc1, and 1.8.4.3 Now Available (Security Release).

Asterisk_OSR_ The Asterisk Development Team has announced the release of Asterisk versions 1.4.41.1, 1.6.2.19-rc1, and 1.8.4.3, which are security releases.

These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/releases

The release of Asterisk 1.4.41.1, 1.6.2.19-rc1, and 1.8.4.3 resolves several issues as outlined below:

* AST-2011-008: If a remote user sends a SIP packet containing a null,
   Asterisk assumes available data extends past the null to the
   end of the packet when the buffer is actually truncated when
   copied.  This causes SIP header parsing to modify data past
   the end of the buffer altering unrelated memory structures.
   This vulnerability does not affect TCP/TLS connections.
   -- Resolved in 1.6.2.18.1 and 1.8.4.3

* AST-2011-009: A remote user sending a SIP packet containing a Contact header
   with a missing left angle bracket (<) causes Asterisk to
   access a null pointer.
   -- Resolved in 1.8.4.3

* AST-2011-010: A memory address was inadvertently transmitted over the
   network via IAX2 via an option control frame and the remote party would try
   to access it.
   -- Resolved in 1.4.41.1, 1.6.2.18.1, and 1.8.4.3

The issues and resolutions are described in the AST-2011-008,
AST-2011-009, and
AST-2011-010 security advisories.

For more information about the details of these vulnerabilities, please read
the security advisories AST-2011-008, AST-2011-009, and AST-2011-010,
which were
released at the same time as this announcement.

For a full list of changes in the current releases, please see the
ChangeLog:

http://downloads.asterisk.org/pu ... /ChangeLog-1.4.41.1
http://downloads.asterisk.org/pu ... hangeLog-1.6.2.18.1
http://downloads.asterisk.org/pu ... s/ChangeLog-1.8.4.3

Security advisories AST-2011-008, AST-2011-009, and AST-2011-010 are
available
at:

http://downloads.asterisk.org/pub/security/AST-2011-008.pdf
http://downloads.asterisk.org/pub/security/AST-2011-009.pdf
http://downloads.asterisk.org/pub/security/AST-2011-010.pdf

Thank you for your continued support of Asterisk!

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The Asterisk Development Team has announced the release of Asterisk versions
1.4.41.2, 1.6.2.18.2, and 1.8.4.4, which are security releases.

These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases

The release of Asterisk 1.4.41.2, 1.6.2.18.2, and 1.8.4.4 resolves the
following issue:

AST-2011-011: Asterisk may respond differently to SIP requests from an
invalid SIP user than it does to a user configured on the system, even when the
alwaysauthreject option is set in the configuration. This can leak information
about what SIP users are valid on the Asterisk system.

For more information about the details of this vulnerability, please read
the security advisory AST-2011-011, which was released at the same time as this
announcement.

For a full list of changes in the current releases, please see the ChangeLog:

http://downloads.asterisk.org/pu ... /ChangeLog-1.4.41.2
http://downloads.asterisk.org/pu ... hangeLog-1.6.2.18.2
http://downloads.asterisk.org/pu ... s/ChangeLog-1.8.4.4

Security advisory AST-2011-011 is available at:

http://downloads.asterisk.org/pub/security/AST-2011-011.pdf

Thank you for your continued support of Asterisk!

--

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The Asterisk Development Team has announced the final maintenance release of
Asterisk, version 1.4.42
. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

Please note that Asterisk 1.4.42 is the final maintenance release from the
1.4 branch. Support for security related issues will continue until April 21,
2012.
For more information about support of the various Asterisk branches, see
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

The release of Asterisk 1.4.42 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:

* Resolve regression with ring groups in the Dial() application
(Closes issue ASTERISK-17874. Reported by mspuhler. Patched by elguero)
* Resolve deadlock when using tab completion on the 'meetme kick' CLI command
when an invalid (non-existent) conference room is specified.
(Closes issue ASTERISK-17771. Reported, patched by zvision)
* Resolve issue where voice frames could be dropped when checking for T.38
during early media.
(Closes issue ASTERISK-17705. Reported, patched by oej)
* Resolve issue where DYNAMIC_FEATURES would not activate after a recent
DTMF fix.
(Closes issue ASTERISK-17914. Reported by vrban. Patched by twilson)

Additionally security announcements AST-2011-010, and AST-2011-011 have been
resolved in this release.

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pu ... sk/ChangeLog-1.4.42

Thank you for your continued support of Asterisk!

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The Asterisk Development Team has announced the final maintenance release of
Asterisk, version 1.6.2.19. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

Please note that Asterisk 1.6.2.19 is the final maintenance release from the
1.6.2 branch. Support for security related issues will continue until April 21,
2012.
For more information about support of the various Asterisk branches, see
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

The release of Asterisk 1.6.2.19 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:

* Don't broadcast FullyBooted to every AMI connection
The FullyBooted event should not be sent to every AMI connection
every time someone connects via AMI. It should only be sent to
the user who just connected.
(Closes issue #18168. Reported, patched by FeyFre)
* Fix thread blocking issue in the sip TCP/TLS implementation.
(Closes issue #18497. Reported by vois. Tested by vois, rossbeer, kowalma,
Freddi_Fonet. Patched by dvossel)
* Don't delay DTMF in core bridge while listening for DTMF features.
(Closes issue #15642, #16625. Reported by jasonshugart, sharvanek. Tested by
globalnetinc, jde. Patched by oej, twilson)
* Fix chan_local crashs in local_fixup()
Thanks OEJ for tracking down the issue and submitting the patch.
(Closes issue #19053. Reported, patched by oej)
* Don't offer video to directmedia callee unless caller offered it as well
(Closes issue #19195. Reported, patched by one47)

Additionally security announcements AST-2011-008, AST-2011-010, and
AST-2011-011 have been resolved in this release.

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pu ... /ChangeLog-1.6.2.19

Thank you for your continued support of Asterisk!

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那麼,Asterisk 1.4 已差不多完成了它的歷史任務了。它的穩定性及使用上的便利性,都令人留下深刻印像,儘管後繼者的Asterisk 1.6 及 1.8 增加了不少獨有強勁的功能,但相信Asterisk 1.4 在未來的一段日子仍為人所津津樂道。

向Asterisk 1.4 致敬!!

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