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Update to Asterisk 11.1.0: this is a major new long term support release. As this is a major release, you should read the information about updating: https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 You can also find documentation in: /usr/pkg/share/doc/asterisk ----- 11.1.0: The Asterisk Development Team has announced the release of Asterisk 11.1.0. The release of Asterisk 11.1.0 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 execution of 'i' extension due to uninitialized variable. * --- Prevent resetting of NATted realtime peer address on reload. * --- Fix ConfBridge crash if no timing module loaded. * --- Fix the Park 'r' option when a channel parks itself. * --- Fix an issue where outgoing calls would fail to establish audio due to ICE negotiation failures. For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.1.0 ----- 11.0.1: The Asterisk Development Team has announced the release of Asterisk 11.0.1. 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 * --- confbridge: Fix a bug which made conferences not record with AMI/CLI commands * --- Fix an issue with res_http_websocket where the chan_sip WebSocket handler could not be registered. For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.1 Thank you for your continued support of Asterisk! ----- 11.0.0: The Asterisk Development Team is pleased to announce the release of Asterisk 11.0.0. 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/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 Thank you for your continued support of Asterisk!
2012-12-11 09:22:48 +01:00
Asterisk is a complete PBX in software. It provides all of the
features you would expect from a PBX and more. Asterisk does voice
over IP in three protocols, and can interoperate with almost all
standards-based telephony equipment using relatively inexpensive
hardware.
Initial import of asterisk-1.0.7, from pkgsrc-wip. Approved by jmcneill. There are still some features not enabled by default, but this is a solid foundation upon which to build - a fully-functional PBX can be built, including PSTN gatewaying using the comms/zaptel-netbsd package. From the DESCR: Asterisk is a complete PBX in software. It provides all of the features you would expect from a PBX and more. Asterisk does voice over IP in three protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. Asterisk provides Voicemail services with Directory, Call Conferencing, Interactive Voice Response, Call Queuing. It has support for three-way calling, caller ID services, ADSI, SIP and H.323 (as both client and gateway).
2005-04-08 05:10:52 +02:00
Asterisk provides Voicemail services with Directory, Call Conferencing,
Interactive Voice Response, Call Queuing. It has support for
three-way calling, caller ID services, ADSI, SIP and H.323 (as both
client and gateway).
Update to Asterisk 11.1.0: this is a major new long term support release. As this is a major release, you should read the information about updating: https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+11 You can also find documentation in: /usr/pkg/share/doc/asterisk ----- 11.1.0: The Asterisk Development Team has announced the release of Asterisk 11.1.0. The release of Asterisk 11.1.0 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 execution of 'i' extension due to uninitialized variable. * --- Prevent resetting of NATted realtime peer address on reload. * --- Fix ConfBridge crash if no timing module loaded. * --- Fix the Park 'r' option when a channel parks itself. * --- Fix an issue where outgoing calls would fail to establish audio due to ICE negotiation failures. For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.1.0 ----- 11.0.1: The Asterisk Development Team has announced the release of Asterisk 11.0.1. 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 * --- confbridge: Fix a bug which made conferences not record with AMI/CLI commands * --- Fix an issue with res_http_websocket where the chan_sip WebSocket handler could not be registered. For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.1 Thank you for your continued support of Asterisk! ----- 11.0.0: The Asterisk Development Team is pleased to announce the release of Asterisk 11.0.0. 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/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 Thank you for your continued support of Asterisk!
2012-12-11 09:22:48 +01:00
NOTE: This version does not work with the zaptel drivers. It
requires the newer DAHDI drivers which are still being ported.
So, there is no hardware support available at this moment.
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