Re: [asterisk-users] Asterisk 11.0.0-rc1 Now Available!
On 10/08/2012 05:15 PM, Asterisk Development Team wrote: 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/display/AST/Upgrading+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/display/AST/Asterisk+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/pub/telephony/asterisk/releases/ChangeLog-11.0.0-rc1 Thank you for your continued support of Asterisk! Thanks for all the great work. We've started using the silk codec a lot for phone app voip. We've found it the most effective low bit rate (16K) codec. Could we get a release 11 version of the silk codec in http://downloads.digium.com/pub/telephony/codec_silk/ ? That way we could start messing with RC 1. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory
[asterisk-users] Asterisk 11.0.0-rc1 Now Available!
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/display/AST/Upgrading+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/display/AST/Asterisk+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/pub/telephony/asterisk/releases/ChangeLog-11.0.0-rc1 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 11.0.0-rc1 Now Available!
One suggestion I have: Would it be helpful to know the revision number of rc1 in the release notes? I'm on a patched version of Asterisk from Doubango to deal with Chrome's non-standard ICE candidates, and unless this is included in rc1 (meaning rc1 is newer than what I have and also deals with Chrome's ICE issues) then I probably wouldn't upgrade. Also, I would prefer to check out from source but don't know the revision number to use. If I'm the only one that would benefit from this, then no worries, I'll deal with it. But if others would benefit from seeing a revision number/checking out from SVN, then maybe consider adding this to the release notes. :) Hope this helps! James On Mon, Oct 8, 2012 at 10:15 AM, Asterisk Development Team asteriskt...@digium.com wrote: 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/display/AST/Upgrading+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/display/AST/Asterisk+11+Documentation A full list of all new features can also be found in the CHANGES file.
Re: [asterisk-users] Asterisk 11.0.0-rc1 Now Available!
James Mortensen wrote: One suggestion I have: Would it be helpful to know the revision number of rc1 in the release notes? I'm on a patched version of Asterisk from Doubango to deal with Chrome's non-standard ICE candidates, and unless this is included in rc1 (meaning rc1 is newer than what I have and also deals with Chrome's ICE issues) then I probably wouldn't upgrade. Also, I would prefer to check out from source but don't know the revision number to use. Chrome Canary is actually using ICE according to the RFC these days and works fine with unpatched Asterisk. We don't presently include VP8 passthrough support though, so no video. If I'm the only one that would benefit from this, then no worries, I'll deal with it. But if others would benefit from seeing a revision number/checking out from SVN, then maybe consider adding this to the release notes. :) -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
