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Asterisk 11.0.1 released

本帖最後由 ckleea 於 2012-11-6 07:52 編輯

The Asterisk Development Team is pleased to announce the first beta release of
Asterisk 11.0.0.  This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases

All interested users of Asterisk are encouraged to participate in the
Asterisk 11 testing process.  Please report any issues found to the issue
tracker, https://issues.asterisk.org/jira.  It is also very useful to see
successful test reports.  Please post those to the asterisk-dev mailing list.
All Asterisk users are invited to participate in the #asterisk-testing channel
on IRC to work together in testing the many parts of Asterisk.  

Asterisk 11 is the next major release series of Asterisk.  It will be a Long
Term Support (LTS) release, similar to Asterisk 1.8.  For more information about
support time lines for Asterisk releases, see the Asterisk versions page:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

For important information regarding upgrading to Asterisk 11, please see the
Asterisk wiki:

https://wiki.asterisk.org/wiki/d ... ding+to+Asterisk+11

A short list of new features includes:

* A new channel driver named chan_motif has been added which provides support
 for Google Talk and Jingle in a single channel driver.  This new channel
 driver includes support for both audio and video, RFC2833 DTMF, all codecs
 supported by Asterisk, hold, unhold, and ringing notification. It is also
 compliant with the current Jingle specification, current Google Jingle
 specification, and the original Google Talk protocol.

* Support for the WebSocket transport for chan_sip.

* SIP peers can now be configured to support negotiation of ICE candidates.

* The app_page application now no longer depends on DAHDI or app_meetme. It
 has been re-architected to use app_confbridge internally.

* Hangup handlers can be attached to channels using the CHANNEL() function.
 Hangup handlers will run when the channel is hung up similar to the h
 extension; however, unlike an h extension, a hangup handler is associated with
 the actual channel and will execute anytime that channel is hung up,
 regardless of where it is in the dialplan.

* Added pre-dial handlers for the Dial and Follow-Me applications.  Pre-dial
 allows you to execute a dialplan subroutine on a channel before a call is
 placed but after the application performing a dial action is invoked. This
 means that the handlers are executed after the creation of the caller/callee
 channels, but before any actions have been taken to actually dial the callee
 channels.

* Log messages can now be easily associated with a certain call by looking at
 a new unique identifier, "Call Id".  Call ids are attached to log messages for
 just about any case where it can be determined that the message is related
 to a particular call.

* Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in
 Asterisk. Unlike traditional ACLs defined in specific module configuration
 files, Named ACLs can be shared across multiple modules.

* The Hangup Cause family of functions and dialplan applications allow for
 inspection of the hangup cause codes for each channel involved in a call.
 This allows a dialplan writer to determine, for each channel, who hung up and
 for what reason(s).

* Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
 lets you set some of the configuration options from the general section
 of features.conf on a per-channel basis. FEATUREMAP() lets you customize
 the key sequence used to activate built-in features, such as blindxfer,
 and automon.

* Support for named pickupgroups/callgroups, allowing any number of pickupgroups
 and callgroups to be defined for several channel drivers.

* IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework.

More information about the new features can be found on the Asterisk wiki:

https://wiki.asterisk.org/wiki/d ... sk+11+Documentation

A full list of all new features can also be found in the CHANGES file.

http://svnview.digium.com/svn/asterisk/branches/11/CHANGES

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

http://downloads.asterisk.org/pu ... ngeLog-11.0.0-beta1

Thank you for your continued support of Asterisk!








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The Asterisk Development Team is pleased to announce the second beta release of
Asterisk 11.0.0.  This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases

All interested users of Asterisk are encouraged to participate in the
Asterisk 11 testing process.  Please report any issues found to the issue
tracker, https://issues.asterisk.org/jira.  It is also very useful to see
successful test reports.  Please post those to the asterisk-dev mailing list.
All Asterisk users are invited to participate in the #asterisk-testing channel
on IRC to work together in testing the many parts of Asterisk.

Asterisk 11 is the next major release series of Asterisk.  It will be a Long
Term Support (LTS) release, similar to Asterisk 1.8.  For more information about
support time lines for Asterisk releases, see the Asterisk versions page:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

For important information regarding upgrading to Asterisk 11, please see the
Asterisk wiki:

https://wiki.asterisk.org/wiki/d ... ding+to+Asterisk+11

A short list of new features includes:

* A new channel driver named chan_motif has been added which provides support
  for Google Talk and Jingle in a single channel driver.  This new channel
  driver includes support for both audio and video, RFC2833 DTMF, all codecs
  supported by Asterisk, hold, unhold, and ringing notification. It is also
  compliant with the current Jingle specification, current Google Jingle
  specification, and the original Google Talk protocol.

* Support for the WebSocket transport for chan_sip.

* SIP peers can now be configured to support negotiation of ICE candidates.

* The app_page application now no longer depends on DAHDI or app_meetme. It
  has been re-architected to use app_confbridge internally.

* Hangup handlers can be attached to channels using the CHANNEL() function.
  Hangup handlers will run when the channel is hung up similar to the h
  extension; however, unlike an h extension, a hangup handler is associated with
  the actual channel and will execute anytime that channel is hung up,
  regardless of where it is in the dialplan.

* Added pre-dial handlers for the Dial and Follow-Me applications.  Pre-dial
  allows you to execute a dialplan subroutine on a channel before a call is
  placed but after the application performing a dial action is invoked. This
  means that the handlers are executed after the creation of the callee
  channels, but before any actions have been taken to actually dial the callee
  channels.

* Log messages can now be easily associated with a certain call by looking at
  a new unique identifier, "Call Id".  Call ids are attached to log messages for
  just about any case where it can be determined that the message is related
  to a particular call.

* Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in
  Asterisk. Unlike traditional ACLs defined in specific module configuration
  files, Named ACLs can be shared across multiple modules.

* The Hangup Cause family of functions and dialplan applications allow for
  inspection of the hangup cause codes for each channel involved in a call.
  This allows a dialplan writer to determine, for each channel, who hung up and
  for what reason(s).

* Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
  lets you set some of the configuration options from the general section
  of features.conf on a per-channel basis. FEATUREMAP() lets you customize
  the key sequence used to activate built-in features, such as blindxfer,
  and automon.

* Support for DTLS-SRTP in chan_sip.

* Support for named pickupgroups/callgroups, allowing any number of pickupgroups
  and callgroups to be defined for several channel drivers.

* IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework.

More information about the new features can be found on the Asterisk wiki:

https://wiki.asterisk.org/wiki/d ... sk+11+Documentation

A full list of all new features can also be found in the CHANGES file.

http://svnview.digium.com/svn/asterisk/branches/11/CHANGES

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

http://downloads.asterisk.org/pu ... ngeLog-11.0.0-beta2

Thank you for your continued support of Asterisk!

TOP

如果Asterisk 11 正式版本發佈,都想試試看,始終它跟Asterisk 1.4 及 1.8 都是 LTS 版本,但Asterisk 1.8 又好似沒有 1.4 的穩定可靠。

TOP

回復 3# bubblestar

Asterisk 1.8有什么不稳定呢?

TOP

回復 4# 角色


    應該說是部份功能不穩,例如: Google Voice 連線問題.

TOP

是吗?真的不知道。

TOP

The Asterisk Development Team is pleased to announce the first release candidate
of Asterisk 11.0.0.  This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases

All interested users of Asterisk are encouraged to participate in the
Asterisk 11 testing process.  Please report any issues found to the issue
tracker, https://issues.asterisk.org/jira.  It is also very useful to see
successful test reports.  Please post those to the asterisk-dev mailing list.
All Asterisk users are invited to participate in the #asterisk-testing channel
on IRC to work together in testing the many parts of Asterisk.

Asterisk 11 is the next major release series of Asterisk.  It will be a Long
Term Support (LTS) release, similar to Asterisk 1.8.  For more information about
support time lines for Asterisk releases, see the Asterisk versions page:
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

For important information regarding upgrading to Asterisk 11, please see the
Asterisk wiki:

https://wiki.asterisk.org/wiki/d ... ding+to+Asterisk+11

A short list of new features includes:

* A new channel driver named chan_motif has been added which provides support
  for Google Talk and Jingle in a single channel driver.  This new channel
  driver includes support for both audio and video, RFC2833 DTMF, all codecs
  supported by Asterisk, hold, unhold, and ringing notification. It is also
  compliant with the current Jingle specification, current Google Jingle
  specification, and the original Google Talk protocol.

* Support for the WebSocket transport for chan_sip.

* SIP peers can now be configured to support negotiation of ICE candidates.

* The app_page application now no longer depends on DAHDI or app_meetme. It
  has been re-architected to use app_confbridge internally.

* Hangup handlers can be attached to channels using the CHANNEL() function.
  Hangup handlers will run when the channel is hung up similar to the h
  extension; however, unlike an h extension, a hangup handler is associated with
  the actual channel and will execute anytime that channel is hung up,
  regardless of where it is in the dialplan.

* Added pre-dial handlers for the Dial and Follow-Me applications.  Pre-dial
  allows you to execute a dialplan subroutine on a channel before a call is
  placed but after the application performing a dial action is invoked. This
  means that the handlers are executed after the creation of the callee
  channels, but before any actions have been taken to actually dial the callee
  channels.

* Log messages can now be easily associated with a certain call by looking at
  a new unique identifier, "Call Id".  Call ids are attached to log messages for
  just about any case where it can be determined that the message is related
  to a particular call.

* Introduced Named ACLs as a new way to define Access Control Lists (ACLs) in
  Asterisk. Unlike traditional ACLs defined in specific module configuration
  files, Named ACLs can be shared across multiple modules.

* The Hangup Cause family of functions and dialplan applications allow for
  inspection of the hangup cause codes for each channel involved in a call.
  This allows a dialplan writer to determine, for each channel, who hung up and
  for what reason(s).

* Two new functions have been added: FEATURE() and FEATUREMAP(). FEATURE()
  lets you set some of the configuration options from the general section
  of features.conf on a per-channel basis. FEATUREMAP() lets you customize
  the key sequence used to activate built-in features, such as blindxfer,
  and automon.

* Support for DTLS-SRTP in chan_sip.

* Support for named pickupgroups/callgroups, allowing any number of pickupgroups
  and callgroups to be defined for several channel drivers.

* IPv6 Support for AMI, AGI, ExternalIVR, and the SIP Security Event Framework.

More information about the new features can be found on the Asterisk wiki:

https://wiki.asterisk.org/wiki/d ... sk+11+Documentation

A full list of all new features can also be found in the CHANGES file.

http://svnview.digium.com/svn/asterisk/branches/11/CHANGES

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

http://downloads.asterisk.org/pu ... hangeLog-11.0.0-rc1

Thank you for your continued support of Asterisk!

TOP

Google Voice 我用了sipgate 作bridge, 再用perl script 去控制google voice dial in
是不是有其他更直接容易方法?

TOP

回復 8# lamsoft

Should be very easy. But I have not yet tested out clearly

TOP

唔敢升上10或以上版本, 驚d conf file 同dial plan 寫法同我而家完全唔一樣...要重新寫過就免了

TOP

Not at all. the config files in asterisk 11 are basically copies of my asterisk 1.8.

I have deleted all those not required and non essential

優化版

TOP

Just upgrade remotely.

Working ok

TOP

心癢癢,稍後會試裝一下 Asterisk 11 RC1 on Atom。

至於Asterisk 1.8,我已成功地將之結合了最新版本的GUI 2.1.0 RC1。 該版本已經對Asterisk 1.8 有有較好的修正,不像以往 GUI 2.0 般問題多多。原本以為 GUI 2.0 之後,Digium 不再維護及放棄了自家的GUI 添,點知無意間發現已有GUI 2.1.0 RC1 的出現。

GUI210rc18.png

還有我是以 CentOS 6.3 64bit 作為 OS 的。
哈! 一於心口掛個勇字!! (當然,會先Backup 了現在使用中可運行的系統才繼續上試啦!)

TOP

回復 13# bubblestar

不见你上来,原来秘密练兵。

TOP

回復 13# bubblestar


    沒有需要用64 bits OS係 ATOM 機

TOP

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