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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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Already updated to Asterisk 11.0.1
  1. [/opt/source/asterisk/asterisk-11.0.1] # /opt/asterisk-11/sbin/asterisk  -rvvvvvvvvvvvvvv
  2. Asterisk 11.0.1, Copyright (C) 1999 - 2012 Digium, Inc. and others.
  3. Created by Mark Spencer <markster@digium.com>
  4. Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
  5. This is free software, with components licensed under the GNU General Public
  6. License version 2 and other licenses; you are welcome to redistribute it under
  7. certain conditions. Type 'core show license' for details.
  8. =========================================================================
  9.   == Parsing '/opt/asterisk-11/etc/asterisk/asterisk.conf': Found
  10.   == Parsing '/opt/asterisk-11/etc/asterisk/extconfig.conf': Found
  11. Connected to Asterisk 11.0.1 currently running on TWTS-269PRO (pid = 4004)
  12. TWTS-269PRO*CLI>
複製代碼

TOP

The Asterisk Development Team has announced the release of Asterisk 11.0.1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.0.1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

* --- chan_sip: Fix a bug causing SIP reloads to remove all entries
      from the registry
  (Closes issue ASTERISK-20611. Reported by Alisher)

* --- confbridge: Fix a bug which made conferences not record with
      AMI/CLI commands
  (Closes issue ASTERISK-20601. Reported by Vilius)

* --- Fix an issue with res_http_websocket where the chan_sip
      WebSocket handler could not be registered.
  (Closes issue ASTERISK-20631. Reported by danjenkins)

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

http://downloads.asterisk.org/pu ... sk/ChangeLog-11.0.1

Thank you for your continued support of Asterisk!

TOP

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

Asterisk 11 is the next major release series of Asterisk.  It is 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 ... es/ChangeLog-11.0.0

Thank you for your continued support of Asterisk!

TOP

The Asterisk Development Team has announced the second release candidate of
Asterisk 11.0.0. This release candidate is available for immediate
download at http://downloads.asterisk.org/pub/telephony/asterisk

The release of Asterisk 11.0.0-rc2 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release candidate:

* --- Fix an issue where outgoing calls would fail to establish audio
      due to ICE negotiation failures.
  (Closes issue ASTERISK-20554. Reported by mmichelson)

* --- Ensure Asterisk fails TCP/TLS SIP calls when certificate
      checking fails
  (Closes issue ASTERISK-20559. Reported by kmoore)

* --- Don't make chan_sip export global symbols.
  (Closes issue ASTERISK-20545. Reported by kmoore)

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

http://downloads.asterisk.org/pu ... hangeLog-11.0.0-rc2

Thank you for your continued support of Asterisk!

--

TOP

回復 18# ckleea


    It seems g719 is no longer support in Asterisk because of licensing problem.

TOP

According to Digium, there are issuse with motif drivers and fixes are not yet released!

TOP

I follow the same instruction. remember the RTP. You need to add to rtp.conf

TOP

本帖最後由 bubblestar 於 2012-10-15 22:04 編輯

https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google

We have to use dial motif instead of dial gtalk in our outgoing dialplan.

TOP

回復 27# lamsoft


    Same, I migrate my 1.8 config to it. No change

TOP

Config file 係咪一樣架? d dialplan syntax同1.8會唔會要寫過哂?

TOP

回復 25# ckleea

Have you tried the GV on Asterisk 11? If yes, what is the performance?

TOP

Look like I can find ways to enable google voice in asterisk 11  RC1

TOP

回復 23# 角色


   
到Asterisk 11 正式版推出時,應該更穩定可以作日常應用了。屇時我又可以跟你學吓安裝在NAS 的東西了。

TOP

回復 22# bubblestar

如果是这样的话,在我的ATOM NAS里也安装一个Asterisk 11 (等它又formal release开始)。

安装路径为/opt/asterisk11, 用SVN下载软件。

现在/opt/etc/asterisk是用自己compile的Asterisk,打算改为如下

/opt/etc/asterisk用ipkg安装的asterisk
/opt/asterisk14
/opt/asterisk18
/opt/asterisk11

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