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Update to Asterisk 11.6.1: this is a security fix update to fix AST-2013-006 and AST-2013-007, and a minor bug fix update. pkgsrc change: disable SRTP on NetBSD as it doesn't link ---- 11.6.1 ---- The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4, 10.12.4-digiumphones, and 11.6.1. The release of these versions resolve the following issues: * A buffer overflow when receiving odd length 16 bit messages in app_sms. An infinite loop could occur which would overwrite memory when a message is received into the unpacksms16() function and the length of the message is an odd number of bytes. * Prevent permissions escalation in the Asterisk Manager Interface. Asterisk now marks certain individual dialplan functions as 'dangerous', which will inhibit their execution from external sources. A 'dangerous' function is one which results in a privilege escalation. For example, if one were to read the channel variable SHELL(rm -rf /) Bad Things(TM) could happen; even if the external source has only read permissions. Execution from external sources may be enabled by setting 'live_dangerously' to 'yes' in the [options] section of asterisk.conf. Although doing so is not recommended. These issues and their resolutions are described in the security advisories. For more information about the details of these vulnerabilities, please read security advisories AST-2013-006 and AST-2013-007, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.6.1 The security advisories are available at: * http://downloads.asterisk.org/pub/security/AST-2013-006.pdf * http://downloads.asterisk.org/pub/security/AST-2013-007.pdf Thank you for your continued support of Asterisk! ----- 11.6.0 ----- The Asterisk Development Team has announced the release of Asterisk 11.6.0. The release of Asterisk 11.6.0 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: * --- Confbridge: empty conference not being torn down (Closes issue ASTERISK-21859. Reported by Chris Gentle) * --- Let Queue wrap up time influence member availability (Closes issue ASTERISK-22189. Reported by Tony Lewis) * --- Fix a longstanding issue with MFC-R2 configuration that prevented users (Closes issue ASTERISK-21117. Reported by Rafael Angulo) * --- chan_iax2: Fix saving the wrong expiry time in astdb. (Closes issue ASTERISK-22504. Reported by Stefan Wachtler) * --- Fix segfault for certain invalid WebSocket input. (Closes issue ASTERISK-21825. Reported by Alfred Farrugia) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.6.0 Thank you for your continued support of Asterisk!
2013-12-23 02:34:03 +01:00
$NetBSD: patch-apps_app__queue.c,v 1.3 2013/12/23 01:34:03 jnemeth Exp $
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
Update to Asterisk 11.6.1: this is a security fix update to fix AST-2013-006 and AST-2013-007, and a minor bug fix update. pkgsrc change: disable SRTP on NetBSD as it doesn't link ---- 11.6.1 ---- The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4, 10.12.4-digiumphones, and 11.6.1. The release of these versions resolve the following issues: * A buffer overflow when receiving odd length 16 bit messages in app_sms. An infinite loop could occur which would overwrite memory when a message is received into the unpacksms16() function and the length of the message is an odd number of bytes. * Prevent permissions escalation in the Asterisk Manager Interface. Asterisk now marks certain individual dialplan functions as 'dangerous', which will inhibit their execution from external sources. A 'dangerous' function is one which results in a privilege escalation. For example, if one were to read the channel variable SHELL(rm -rf /) Bad Things(TM) could happen; even if the external source has only read permissions. Execution from external sources may be enabled by setting 'live_dangerously' to 'yes' in the [options] section of asterisk.conf. Although doing so is not recommended. These issues and their resolutions are described in the security advisories. For more information about the details of these vulnerabilities, please read security advisories AST-2013-006 and AST-2013-007, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.6.1 The security advisories are available at: * http://downloads.asterisk.org/pub/security/AST-2013-006.pdf * http://downloads.asterisk.org/pub/security/AST-2013-007.pdf Thank you for your continued support of Asterisk! ----- 11.6.0 ----- The Asterisk Development Team has announced the release of Asterisk 11.6.0. The release of Asterisk 11.6.0 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: * --- Confbridge: empty conference not being torn down (Closes issue ASTERISK-21859. Reported by Chris Gentle) * --- Let Queue wrap up time influence member availability (Closes issue ASTERISK-22189. Reported by Tony Lewis) * --- Fix a longstanding issue with MFC-R2 configuration that prevented users (Closes issue ASTERISK-21117. Reported by Rafael Angulo) * --- chan_iax2: Fix saving the wrong expiry time in astdb. (Closes issue ASTERISK-22504. Reported by Stefan Wachtler) * --- Fix segfault for certain invalid WebSocket input. (Closes issue ASTERISK-21825. Reported by Alfred Farrugia) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.6.0 Thank you for your continued support of Asterisk!
2013-12-23 02:34:03 +01:00
--- apps/app_queue.c.orig 2013-09-12 00:02:37.000000000 +0000
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
+++ apps/app_queue.c
Update to Asterisk 11.6.1: this is a security fix update to fix AST-2013-006 and AST-2013-007, and a minor bug fix update. pkgsrc change: disable SRTP on NetBSD as it doesn't link ---- 11.6.1 ---- The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4, 10.12.4-digiumphones, and 11.6.1. The release of these versions resolve the following issues: * A buffer overflow when receiving odd length 16 bit messages in app_sms. An infinite loop could occur which would overwrite memory when a message is received into the unpacksms16() function and the length of the message is an odd number of bytes. * Prevent permissions escalation in the Asterisk Manager Interface. Asterisk now marks certain individual dialplan functions as 'dangerous', which will inhibit their execution from external sources. A 'dangerous' function is one which results in a privilege escalation. For example, if one were to read the channel variable SHELL(rm -rf /) Bad Things(TM) could happen; even if the external source has only read permissions. Execution from external sources may be enabled by setting 'live_dangerously' to 'yes' in the [options] section of asterisk.conf. Although doing so is not recommended. These issues and their resolutions are described in the security advisories. For more information about the details of these vulnerabilities, please read security advisories AST-2013-006 and AST-2013-007, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.6.1 The security advisories are available at: * http://downloads.asterisk.org/pub/security/AST-2013-006.pdf * http://downloads.asterisk.org/pub/security/AST-2013-007.pdf Thank you for your continued support of Asterisk! ----- 11.6.0 ----- The Asterisk Development Team has announced the release of Asterisk 11.6.0. The release of Asterisk 11.6.0 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: * --- Confbridge: empty conference not being torn down (Closes issue ASTERISK-21859. Reported by Chris Gentle) * --- Let Queue wrap up time influence member availability (Closes issue ASTERISK-22189. Reported by Tony Lewis) * --- Fix a longstanding issue with MFC-R2 configuration that prevented users (Closes issue ASTERISK-21117. Reported by Rafael Angulo) * --- chan_iax2: Fix saving the wrong expiry time in astdb. (Closes issue ASTERISK-22504. Reported by Stefan Wachtler) * --- Fix segfault for certain invalid WebSocket input. (Closes issue ASTERISK-21825. Reported by Alfred Farrugia) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.6.0 Thank you for your continued support of Asterisk!
2013-12-23 02:34:03 +01:00
@@ -4735,7 +4735,7 @@ static int wait_our_turn(struct queue_en
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
if ((status = get_member_status(qe->parent, qe->max_penalty, qe->min_penalty, qe->parent->leavewhenempty))) {
*reason = QUEUE_LEAVEEMPTY;
- ast_queue_log(qe->parent->name, ast_channel_uniqueid(qe->chan), "NONE", "EXITEMPTY", "%d|%d|%ld", qe->pos, qe->opos, (long) time(NULL) - qe->start);
+ ast_queue_log(qe->parent->name, ast_channel_uniqueid(qe->chan), "NONE", "EXITEMPTY", "%d|%d|%jd", qe->pos, qe->opos, (intmax_t) time(NULL) - qe->start);
leave_queue(qe);
break;
}
Update to Asterisk 11.6.1: this is a security fix update to fix AST-2013-006 and AST-2013-007, and a minor bug fix update. pkgsrc change: disable SRTP on NetBSD as it doesn't link ---- 11.6.1 ---- The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4, 10.12.4-digiumphones, and 11.6.1. The release of these versions resolve the following issues: * A buffer overflow when receiving odd length 16 bit messages in app_sms. An infinite loop could occur which would overwrite memory when a message is received into the unpacksms16() function and the length of the message is an odd number of bytes. * Prevent permissions escalation in the Asterisk Manager Interface. Asterisk now marks certain individual dialplan functions as 'dangerous', which will inhibit their execution from external sources. A 'dangerous' function is one which results in a privilege escalation. For example, if one were to read the channel variable SHELL(rm -rf /) Bad Things(TM) could happen; even if the external source has only read permissions. Execution from external sources may be enabled by setting 'live_dangerously' to 'yes' in the [options] section of asterisk.conf. Although doing so is not recommended. These issues and their resolutions are described in the security advisories. For more information about the details of these vulnerabilities, please read security advisories AST-2013-006 and AST-2013-007, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.6.1 The security advisories are available at: * http://downloads.asterisk.org/pub/security/AST-2013-006.pdf * http://downloads.asterisk.org/pub/security/AST-2013-007.pdf Thank you for your continued support of Asterisk! ----- 11.6.0 ----- The Asterisk Development Team has announced the release of Asterisk 11.6.0. The release of Asterisk 11.6.0 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: * --- Confbridge: empty conference not being torn down (Closes issue ASTERISK-21859. Reported by Chris Gentle) * --- Let Queue wrap up time influence member availability (Closes issue ASTERISK-22189. Reported by Tony Lewis) * --- Fix a longstanding issue with MFC-R2 configuration that prevented users (Closes issue ASTERISK-21117. Reported by Rafael Angulo) * --- chan_iax2: Fix saving the wrong expiry time in astdb. (Closes issue ASTERISK-22504. Reported by Stefan Wachtler) * --- Fix segfault for certain invalid WebSocket input. (Closes issue ASTERISK-21825. Reported by Alfred Farrugia) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.6.0 Thank you for your continued support of Asterisk!
2013-12-23 02:34:03 +01:00
@@ -4975,12 +4975,12 @@ static void send_agent_complete(const st
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
"Channel: %s\r\n"
"Member: %s\r\n"
"MemberName: %s\r\n"
- "HoldTime: %ld\r\n"
- "TalkTime: %ld\r\n"
+ "HoldTime: %jd\r\n"
+ "TalkTime: %jd\r\n"
"Reason: %s\r\n"
"%s",
queuename, ast_channel_uniqueid(qe->chan), ast_channel_name(peer), member->interface, member->membername,
- (long)(callstart - qe->start), (long)(time(NULL) - callstart), reason,
+ (intmax_t)(callstart - qe->start), (intmax_t)(time(NULL) - callstart), reason,
qe->parent->eventwhencalled == QUEUE_EVENT_VARIABLES ? vars2manager(qe->chan, vars, vars_len) : "");
}
Update to Asterisk 11.6.1: this is a security fix update to fix AST-2013-006 and AST-2013-007, and a minor bug fix update. pkgsrc change: disable SRTP on NetBSD as it doesn't link ---- 11.6.1 ---- The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4, 10.12.4-digiumphones, and 11.6.1. The release of these versions resolve the following issues: * A buffer overflow when receiving odd length 16 bit messages in app_sms. An infinite loop could occur which would overwrite memory when a message is received into the unpacksms16() function and the length of the message is an odd number of bytes. * Prevent permissions escalation in the Asterisk Manager Interface. Asterisk now marks certain individual dialplan functions as 'dangerous', which will inhibit their execution from external sources. A 'dangerous' function is one which results in a privilege escalation. For example, if one were to read the channel variable SHELL(rm -rf /) Bad Things(TM) could happen; even if the external source has only read permissions. Execution from external sources may be enabled by setting 'live_dangerously' to 'yes' in the [options] section of asterisk.conf. Although doing so is not recommended. These issues and their resolutions are described in the security advisories. For more information about the details of these vulnerabilities, please read security advisories AST-2013-006 and AST-2013-007, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.6.1 The security advisories are available at: * http://downloads.asterisk.org/pub/security/AST-2013-006.pdf * http://downloads.asterisk.org/pub/security/AST-2013-007.pdf Thank you for your continued support of Asterisk! ----- 11.6.0 ----- The Asterisk Development Team has announced the release of Asterisk 11.6.0. The release of Asterisk 11.6.0 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: * --- Confbridge: empty conference not being torn down (Closes issue ASTERISK-21859. Reported by Chris Gentle) * --- Let Queue wrap up time influence member availability (Closes issue ASTERISK-22189. Reported by Tony Lewis) * --- Fix a longstanding issue with MFC-R2 configuration that prevented users (Closes issue ASTERISK-21117. Reported by Rafael Angulo) * --- chan_iax2: Fix saving the wrong expiry time in astdb. (Closes issue ASTERISK-22504. Reported by Stefan Wachtler) * --- Fix segfault for certain invalid WebSocket input. (Closes issue ASTERISK-21825. Reported by Alfred Farrugia) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.6.0 Thank you for your continued support of Asterisk!
2013-12-23 02:34:03 +01:00
@@ -5023,9 +5023,9 @@ static void queue_transfer_fixup(void *d
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
int callcompletedinsl = qtds->callcompletedinsl;
struct ast_datastore *datastore;
- ast_queue_log(qe->parent->name, ast_channel_uniqueid(qe->chan), member->membername, "TRANSFER", "%s|%s|%ld|%ld|%d",
- ast_channel_exten(new_chan), ast_channel_context(new_chan), (long) (callstart - qe->start),
- (long) (time(NULL) - callstart), qe->opos);
+ ast_queue_log(qe->parent->name, ast_channel_uniqueid(qe->chan), member->membername, "TRANSFER", "%s|%s|%jd|%jd|%d",
+ ast_channel_exten(new_chan), ast_channel_context(new_chan), (intmax_t) (callstart - qe->start),
+ (intmax_t) (time(NULL) - callstart), qe->opos);
update_queue(qe->parent, member, callcompletedinsl, (time(NULL) - callstart));
Update to Asterisk 11.6.1: this is a security fix update to fix AST-2013-006 and AST-2013-007, and a minor bug fix update. pkgsrc change: disable SRTP on NetBSD as it doesn't link ---- 11.6.1 ---- The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4, 10.12.4-digiumphones, and 11.6.1. The release of these versions resolve the following issues: * A buffer overflow when receiving odd length 16 bit messages in app_sms. An infinite loop could occur which would overwrite memory when a message is received into the unpacksms16() function and the length of the message is an odd number of bytes. * Prevent permissions escalation in the Asterisk Manager Interface. Asterisk now marks certain individual dialplan functions as 'dangerous', which will inhibit their execution from external sources. A 'dangerous' function is one which results in a privilege escalation. For example, if one were to read the channel variable SHELL(rm -rf /) Bad Things(TM) could happen; even if the external source has only read permissions. Execution from external sources may be enabled by setting 'live_dangerously' to 'yes' in the [options] section of asterisk.conf. Although doing so is not recommended. These issues and their resolutions are described in the security advisories. For more information about the details of these vulnerabilities, please read security advisories AST-2013-006 and AST-2013-007, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.6.1 The security advisories are available at: * http://downloads.asterisk.org/pub/security/AST-2013-006.pdf * http://downloads.asterisk.org/pub/security/AST-2013-007.pdf Thank you for your continued support of Asterisk! ----- 11.6.0 ----- The Asterisk Development Team has announced the release of Asterisk 11.6.0. The release of Asterisk 11.6.0 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: * --- Confbridge: empty conference not being torn down (Closes issue ASTERISK-21859. Reported by Chris Gentle) * --- Let Queue wrap up time influence member availability (Closes issue ASTERISK-22189. Reported by Tony Lewis) * --- Fix a longstanding issue with MFC-R2 configuration that prevented users (Closes issue ASTERISK-21117. Reported by Rafael Angulo) * --- chan_iax2: Fix saving the wrong expiry time in astdb. (Closes issue ASTERISK-22504. Reported by Stefan Wachtler) * --- Fix segfault for certain invalid WebSocket input. (Closes issue ASTERISK-21825. Reported by Alfred Farrugia) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.6.0 Thank you for your continued support of Asterisk!
2013-12-23 02:34:03 +01:00
@@ -5542,7 +5542,7 @@ static int try_calling(struct queue_ent
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
} else if (ast_check_hangup(qe->chan)) {
/* Caller must have hung up just before being connected */
ast_log(LOG_NOTICE, "Caller was about to talk to agent on %s but the caller hungup.\n", ast_channel_name(peer));
- ast_queue_log(queuename, ast_channel_uniqueid(qe->chan), member->membername, "ABANDON", "%d|%d|%ld", qe->pos, qe->opos, (long) time(NULL) - qe->start);
+ ast_queue_log(queuename, ast_channel_uniqueid(qe->chan), member->membername, "ABANDON", "%d|%d|%jd", qe->pos, qe->opos, (intmax_t) time(NULL) - qe->start);
record_abandoned(qe);
ast_autoservice_chan_hangup_peer(qe->chan, peer);
ao2_ref(member, -1);
Update to Asterisk 11.6.1: this is a security fix update to fix AST-2013-006 and AST-2013-007, and a minor bug fix update. pkgsrc change: disable SRTP on NetBSD as it doesn't link ---- 11.6.1 ---- The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4, 10.12.4-digiumphones, and 11.6.1. The release of these versions resolve the following issues: * A buffer overflow when receiving odd length 16 bit messages in app_sms. An infinite loop could occur which would overwrite memory when a message is received into the unpacksms16() function and the length of the message is an odd number of bytes. * Prevent permissions escalation in the Asterisk Manager Interface. Asterisk now marks certain individual dialplan functions as 'dangerous', which will inhibit their execution from external sources. A 'dangerous' function is one which results in a privilege escalation. For example, if one were to read the channel variable SHELL(rm -rf /) Bad Things(TM) could happen; even if the external source has only read permissions. Execution from external sources may be enabled by setting 'live_dangerously' to 'yes' in the [options] section of asterisk.conf. Although doing so is not recommended. These issues and their resolutions are described in the security advisories. For more information about the details of these vulnerabilities, please read security advisories AST-2013-006 and AST-2013-007, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.6.1 The security advisories are available at: * http://downloads.asterisk.org/pub/security/AST-2013-006.pdf * http://downloads.asterisk.org/pub/security/AST-2013-007.pdf Thank you for your continued support of Asterisk! ----- 11.6.0 ----- The Asterisk Development Team has announced the release of Asterisk 11.6.0. The release of Asterisk 11.6.0 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: * --- Confbridge: empty conference not being torn down (Closes issue ASTERISK-21859. Reported by Chris Gentle) * --- Let Queue wrap up time influence member availability (Closes issue ASTERISK-22189. Reported by Tony Lewis) * --- Fix a longstanding issue with MFC-R2 configuration that prevented users (Closes issue ASTERISK-21117. Reported by Rafael Angulo) * --- chan_iax2: Fix saving the wrong expiry time in astdb. (Closes issue ASTERISK-22504. Reported by Stefan Wachtler) * --- Fix segfault for certain invalid WebSocket input. (Closes issue ASTERISK-21825. Reported by Alfred Farrugia) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.6.0 Thank you for your continued support of Asterisk!
2013-12-23 02:34:03 +01:00
@@ -5591,8 +5591,8 @@ static int try_calling(struct queue_ent
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
/* if setqueueentryvar is defined, make queue entry (i.e. the caller) variables available to the channel */
/* use pbx_builtin_setvar to set a load of variables with one call */
if (qe->parent->setqueueentryvar) {
- snprintf(interfacevar, sizeof(interfacevar), "QEHOLDTIME=%ld,QEORIGINALPOS=%d",
- (long) time(NULL) - qe->start, qe->opos);
+ snprintf(interfacevar, sizeof(interfacevar), "QEHOLDTIME=%jd,QEORIGINALPOS=%d",
+ (intmax_t) time(NULL) - qe->start, qe->opos);
pbx_builtin_setvar_multiple(qe->chan, interfacevar);
pbx_builtin_setvar_multiple(peer, interfacevar);
}
Update to Asterisk 11.6.1: this is a security fix update to fix AST-2013-006 and AST-2013-007, and a minor bug fix update. pkgsrc change: disable SRTP on NetBSD as it doesn't link ---- 11.6.1 ---- The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4, 10.12.4-digiumphones, and 11.6.1. The release of these versions resolve the following issues: * A buffer overflow when receiving odd length 16 bit messages in app_sms. An infinite loop could occur which would overwrite memory when a message is received into the unpacksms16() function and the length of the message is an odd number of bytes. * Prevent permissions escalation in the Asterisk Manager Interface. Asterisk now marks certain individual dialplan functions as 'dangerous', which will inhibit their execution from external sources. A 'dangerous' function is one which results in a privilege escalation. For example, if one were to read the channel variable SHELL(rm -rf /) Bad Things(TM) could happen; even if the external source has only read permissions. Execution from external sources may be enabled by setting 'live_dangerously' to 'yes' in the [options] section of asterisk.conf. Although doing so is not recommended. These issues and their resolutions are described in the security advisories. For more information about the details of these vulnerabilities, please read security advisories AST-2013-006 and AST-2013-007, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.6.1 The security advisories are available at: * http://downloads.asterisk.org/pub/security/AST-2013-006.pdf * http://downloads.asterisk.org/pub/security/AST-2013-007.pdf Thank you for your continued support of Asterisk! ----- 11.6.0 ----- The Asterisk Development Team has announced the release of Asterisk 11.6.0. The release of Asterisk 11.6.0 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: * --- Confbridge: empty conference not being torn down (Closes issue ASTERISK-21859. Reported by Chris Gentle) * --- Let Queue wrap up time influence member availability (Closes issue ASTERISK-22189. Reported by Tony Lewis) * --- Fix a longstanding issue with MFC-R2 configuration that prevented users (Closes issue ASTERISK-21117. Reported by Rafael Angulo) * --- chan_iax2: Fix saving the wrong expiry time in astdb. (Closes issue ASTERISK-22504. Reported by Stefan Wachtler) * --- Fix segfault for certain invalid WebSocket input. (Closes issue ASTERISK-21825. Reported by Alfred Farrugia) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.6.0 Thank you for your continued support of Asterisk!
2013-12-23 02:34:03 +01:00
@@ -5809,8 +5809,8 @@ static int try_calling(struct queue_ent
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
}
}
qe->handled++;
- ast_queue_log(queuename, ast_channel_uniqueid(qe->chan), member->membername, "CONNECT", "%ld|%s|%ld", (long) time(NULL) - qe->start, ast_channel_uniqueid(peer),
- (long)(orig - to > 0 ? (orig - to) / 1000 : 0));
+ ast_queue_log(queuename, ast_channel_uniqueid(qe->chan), member->membername, "CONNECT", "%jd|%s|%jd", (intmax_t) time(NULL) - qe->start, ast_channel_uniqueid(peer),
+ (intmax_t)(orig - to > 0 ? (orig - to) / 1000 : 0));
if (ast_channel_cdr(qe->chan)) {
struct ast_cdr *cdr;
Update to Asterisk 11.6.1: this is a security fix update to fix AST-2013-006 and AST-2013-007, and a minor bug fix update. pkgsrc change: disable SRTP on NetBSD as it doesn't link ---- 11.6.1 ---- The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4, 10.12.4-digiumphones, and 11.6.1. The release of these versions resolve the following issues: * A buffer overflow when receiving odd length 16 bit messages in app_sms. An infinite loop could occur which would overwrite memory when a message is received into the unpacksms16() function and the length of the message is an odd number of bytes. * Prevent permissions escalation in the Asterisk Manager Interface. Asterisk now marks certain individual dialplan functions as 'dangerous', which will inhibit their execution from external sources. A 'dangerous' function is one which results in a privilege escalation. For example, if one were to read the channel variable SHELL(rm -rf /) Bad Things(TM) could happen; even if the external source has only read permissions. Execution from external sources may be enabled by setting 'live_dangerously' to 'yes' in the [options] section of asterisk.conf. Although doing so is not recommended. These issues and their resolutions are described in the security advisories. For more information about the details of these vulnerabilities, please read security advisories AST-2013-006 and AST-2013-007, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.6.1 The security advisories are available at: * http://downloads.asterisk.org/pub/security/AST-2013-006.pdf * http://downloads.asterisk.org/pub/security/AST-2013-007.pdf Thank you for your continued support of Asterisk! ----- 11.6.0 ----- The Asterisk Development Team has announced the release of Asterisk 11.6.0. The release of Asterisk 11.6.0 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: * --- Confbridge: empty conference not being torn down (Closes issue ASTERISK-21859. Reported by Chris Gentle) * --- Let Queue wrap up time influence member availability (Closes issue ASTERISK-22189. Reported by Tony Lewis) * --- Fix a longstanding issue with MFC-R2 configuration that prevented users (Closes issue ASTERISK-21117. Reported by Rafael Angulo) * --- chan_iax2: Fix saving the wrong expiry time in astdb. (Closes issue ASTERISK-22504. Reported by Stefan Wachtler) * --- Fix segfault for certain invalid WebSocket input. (Closes issue ASTERISK-21825. Reported by Alfred Farrugia) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.6.0 Thank you for your continued support of Asterisk!
2013-12-23 02:34:03 +01:00
@@ -5864,12 +5864,12 @@ static int try_calling(struct queue_ent
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
"Channel: %s\r\n"
"Member: %s\r\n"
"MemberName: %s\r\n"
- "HoldTime: %ld\r\n"
+ "HoldTime: %jd\r\n"
"BridgedChannel: %s\r\n"
- "RingTime: %ld\r\n"
+ "RingTime: %jd\r\n"
"%s",
queuename, ast_channel_uniqueid(qe->chan), ast_channel_name(peer), member->interface, member->membername,
- (long) time(NULL) - qe->start, ast_channel_uniqueid(peer), (long)(orig - to > 0 ? (orig - to) / 1000 : 0),
+ (intmax_t) time(NULL) - qe->start, ast_channel_uniqueid(peer), (intmax_t)(orig - to > 0 ? (orig - to) / 1000 : 0),
qe->parent->eventwhencalled == QUEUE_EVENT_VARIABLES ? vars2manager(qe->chan, vars, sizeof(vars)) : "");
ast_copy_string(oldcontext, ast_channel_context(qe->chan), sizeof(oldcontext));
ast_copy_string(oldexten, ast_channel_exten(qe->chan), sizeof(oldexten));
Update to Asterisk 11.6.1: this is a security fix update to fix AST-2013-006 and AST-2013-007, and a minor bug fix update. pkgsrc change: disable SRTP on NetBSD as it doesn't link ---- 11.6.1 ---- The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4, 10.12.4-digiumphones, and 11.6.1. The release of these versions resolve the following issues: * A buffer overflow when receiving odd length 16 bit messages in app_sms. An infinite loop could occur which would overwrite memory when a message is received into the unpacksms16() function and the length of the message is an odd number of bytes. * Prevent permissions escalation in the Asterisk Manager Interface. Asterisk now marks certain individual dialplan functions as 'dangerous', which will inhibit their execution from external sources. A 'dangerous' function is one which results in a privilege escalation. For example, if one were to read the channel variable SHELL(rm -rf /) Bad Things(TM) could happen; even if the external source has only read permissions. Execution from external sources may be enabled by setting 'live_dangerously' to 'yes' in the [options] section of asterisk.conf. Although doing so is not recommended. These issues and their resolutions are described in the security advisories. For more information about the details of these vulnerabilities, please read security advisories AST-2013-006 and AST-2013-007, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.6.1 The security advisories are available at: * http://downloads.asterisk.org/pub/security/AST-2013-006.pdf * http://downloads.asterisk.org/pub/security/AST-2013-007.pdf Thank you for your continued support of Asterisk! ----- 11.6.0 ----- The Asterisk Development Team has announced the release of Asterisk 11.6.0. The release of Asterisk 11.6.0 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: * --- Confbridge: empty conference not being torn down (Closes issue ASTERISK-21859. Reported by Chris Gentle) * --- Let Queue wrap up time influence member availability (Closes issue ASTERISK-22189. Reported by Tony Lewis) * --- Fix a longstanding issue with MFC-R2 configuration that prevented users (Closes issue ASTERISK-21117. Reported by Rafael Angulo) * --- chan_iax2: Fix saving the wrong expiry time in astdb. (Closes issue ASTERISK-22504. Reported by Stefan Wachtler) * --- Fix segfault for certain invalid WebSocket input. (Closes issue ASTERISK-21825. Reported by Alfred Farrugia) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.6.0 Thank you for your continued support of Asterisk!
2013-12-23 02:34:03 +01:00
@@ -5901,17 +5901,17 @@ static int try_calling(struct queue_ent
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
/* detect a blind transfer */
if (!(ast_channel_softhangup_internal_flag(qe->chan) | ast_channel_softhangup_internal_flag(peer)) && (strcasecmp(oldcontext, ast_channel_context(qe->chan)) || strcasecmp(oldexten, ast_channel_exten(qe->chan)))) {
- ast_queue_log(queuename, ast_channel_uniqueid(qe->chan), member->membername, "TRANSFER", "%s|%s|%ld|%ld|%d",
- ast_channel_exten(qe->chan), ast_channel_context(qe->chan), (long) (callstart - qe->start),
- (long) (time(NULL) - callstart), qe->opos);
+ ast_queue_log(queuename, ast_channel_uniqueid(qe->chan), member->membername, "TRANSFER", "%s|%s|%jd|%jd|%d",
+ ast_channel_exten(qe->chan), ast_channel_context(qe->chan), (intmax_t) (callstart - qe->start),
+ (intmax_t) (time(NULL) - callstart), qe->opos);
send_agent_complete(qe, queuename, peer, member, callstart, vars, sizeof(vars), TRANSFER);
Update to Asterisk 11.3.0: this is a bugfix release. The Asterisk Development Team has announced the release of Asterisk 11.3.0. The release of Asterisk 11.3.0 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 issue where chan_mobile fails to bind to first available port * --- Fix Queue Log Reporting Every Call COMPLETECALLER With "h" Extension Present * --- Retain XMPP filters across reconnections so external modules continue to function as expected. * --- Ensure that a declined media stream is terminated with a '\r\n' * --- Fix pjproject compilation in certain circumstances For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.3.0 Thank you for your continued support of Asterisk!
2013-05-05 03:32:34 +02:00
} else if (ast_check_hangup(qe->chan) && !ast_check_hangup(peer)) {
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
- ast_queue_log(queuename, ast_channel_uniqueid(qe->chan), member->membername, "COMPLETECALLER", "%ld|%ld|%d",
- (long) (callstart - qe->start), (long) (time(NULL) - callstart), qe->opos);
+ ast_queue_log(queuename, ast_channel_uniqueid(qe->chan), member->membername, "COMPLETECALLER", "%jd|%jd|%d",
+ (intmax_t) (callstart - qe->start), (intmax_t) (time(NULL) - callstart), qe->opos);
send_agent_complete(qe, queuename, peer, member, callstart, vars, sizeof(vars), CALLER);
} else {
- ast_queue_log(queuename, ast_channel_uniqueid(qe->chan), member->membername, "COMPLETEAGENT", "%ld|%ld|%d",
- (long) (callstart - qe->start), (long) (time(NULL) - callstart), qe->opos);
+ ast_queue_log(queuename, ast_channel_uniqueid(qe->chan), member->membername, "COMPLETEAGENT", "%jd|%jd|%d",
+ (intmax_t) (callstart - qe->start), (intmax_t) (time(NULL) - callstart), qe->opos);
send_agent_complete(qe, queuename, peer, member, callstart, vars, sizeof(vars), AGENT);
}
Update to Asterisk 11.6.1: this is a security fix update to fix AST-2013-006 and AST-2013-007, and a minor bug fix update. pkgsrc change: disable SRTP on NetBSD as it doesn't link ---- 11.6.1 ---- The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4, 10.12.4-digiumphones, and 11.6.1. The release of these versions resolve the following issues: * A buffer overflow when receiving odd length 16 bit messages in app_sms. An infinite loop could occur which would overwrite memory when a message is received into the unpacksms16() function and the length of the message is an odd number of bytes. * Prevent permissions escalation in the Asterisk Manager Interface. Asterisk now marks certain individual dialplan functions as 'dangerous', which will inhibit their execution from external sources. A 'dangerous' function is one which results in a privilege escalation. For example, if one were to read the channel variable SHELL(rm -rf /) Bad Things(TM) could happen; even if the external source has only read permissions. Execution from external sources may be enabled by setting 'live_dangerously' to 'yes' in the [options] section of asterisk.conf. Although doing so is not recommended. These issues and their resolutions are described in the security advisories. For more information about the details of these vulnerabilities, please read security advisories AST-2013-006 and AST-2013-007, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.6.1 The security advisories are available at: * http://downloads.asterisk.org/pub/security/AST-2013-006.pdf * http://downloads.asterisk.org/pub/security/AST-2013-007.pdf Thank you for your continued support of Asterisk! ----- 11.6.0 ----- The Asterisk Development Team has announced the release of Asterisk 11.6.0. The release of Asterisk 11.6.0 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: * --- Confbridge: empty conference not being torn down (Closes issue ASTERISK-21859. Reported by Chris Gentle) * --- Let Queue wrap up time influence member availability (Closes issue ASTERISK-22189. Reported by Tony Lewis) * --- Fix a longstanding issue with MFC-R2 configuration that prevented users (Closes issue ASTERISK-21117. Reported by Rafael Angulo) * --- chan_iax2: Fix saving the wrong expiry time in astdb. (Closes issue ASTERISK-22504. Reported by Stefan Wachtler) * --- Fix segfault for certain invalid WebSocket input. (Closes issue ASTERISK-21825. Reported by Alfred Farrugia) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.6.0 Thank you for your continued support of Asterisk!
2013-12-23 02:34:03 +01:00
if ((tds = ast_channel_datastore_find(qe->chan, &queue_transfer_info, NULL))) {
@@ -7088,8 +7088,8 @@ check_turns:
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
record_abandoned(&qe);
reason = QUEUE_TIMEOUT;
res = 0;
Update to Asterisk 11.6.1: this is a security fix update to fix AST-2013-006 and AST-2013-007, and a minor bug fix update. pkgsrc change: disable SRTP on NetBSD as it doesn't link ---- 11.6.1 ---- The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4, 10.12.4-digiumphones, and 11.6.1. The release of these versions resolve the following issues: * A buffer overflow when receiving odd length 16 bit messages in app_sms. An infinite loop could occur which would overwrite memory when a message is received into the unpacksms16() function and the length of the message is an odd number of bytes. * Prevent permissions escalation in the Asterisk Manager Interface. Asterisk now marks certain individual dialplan functions as 'dangerous', which will inhibit their execution from external sources. A 'dangerous' function is one which results in a privilege escalation. For example, if one were to read the channel variable SHELL(rm -rf /) Bad Things(TM) could happen; even if the external source has only read permissions. Execution from external sources may be enabled by setting 'live_dangerously' to 'yes' in the [options] section of asterisk.conf. Although doing so is not recommended. These issues and their resolutions are described in the security advisories. For more information about the details of these vulnerabilities, please read security advisories AST-2013-006 and AST-2013-007, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.6.1 The security advisories are available at: * http://downloads.asterisk.org/pub/security/AST-2013-006.pdf * http://downloads.asterisk.org/pub/security/AST-2013-007.pdf Thank you for your continued support of Asterisk! ----- 11.6.0 ----- The Asterisk Development Team has announced the release of Asterisk 11.6.0. The release of Asterisk 11.6.0 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: * --- Confbridge: empty conference not being torn down (Closes issue ASTERISK-21859. Reported by Chris Gentle) * --- Let Queue wrap up time influence member availability (Closes issue ASTERISK-22189. Reported by Tony Lewis) * --- Fix a longstanding issue with MFC-R2 configuration that prevented users (Closes issue ASTERISK-21117. Reported by Rafael Angulo) * --- chan_iax2: Fix saving the wrong expiry time in astdb. (Closes issue ASTERISK-22504. Reported by Stefan Wachtler) * --- Fix segfault for certain invalid WebSocket input. (Closes issue ASTERISK-21825. Reported by Alfred Farrugia) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.6.0 Thank you for your continued support of Asterisk!
2013-12-23 02:34:03 +01:00
- ast_queue_log(args.queuename, ast_channel_uniqueid(chan),"NONE", "EXITWITHTIMEOUT", "%d|%d|%ld",
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
- qe.pos, qe.opos, (long) time(NULL) - qe.start);
Update to Asterisk 11.6.1: this is a security fix update to fix AST-2013-006 and AST-2013-007, and a minor bug fix update. pkgsrc change: disable SRTP on NetBSD as it doesn't link ---- 11.6.1 ---- The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4, 10.12.4-digiumphones, and 11.6.1. The release of these versions resolve the following issues: * A buffer overflow when receiving odd length 16 bit messages in app_sms. An infinite loop could occur which would overwrite memory when a message is received into the unpacksms16() function and the length of the message is an odd number of bytes. * Prevent permissions escalation in the Asterisk Manager Interface. Asterisk now marks certain individual dialplan functions as 'dangerous', which will inhibit their execution from external sources. A 'dangerous' function is one which results in a privilege escalation. For example, if one were to read the channel variable SHELL(rm -rf /) Bad Things(TM) could happen; even if the external source has only read permissions. Execution from external sources may be enabled by setting 'live_dangerously' to 'yes' in the [options] section of asterisk.conf. Although doing so is not recommended. These issues and their resolutions are described in the security advisories. For more information about the details of these vulnerabilities, please read security advisories AST-2013-006 and AST-2013-007, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.6.1 The security advisories are available at: * http://downloads.asterisk.org/pub/security/AST-2013-006.pdf * http://downloads.asterisk.org/pub/security/AST-2013-007.pdf Thank you for your continued support of Asterisk! ----- 11.6.0 ----- The Asterisk Development Team has announced the release of Asterisk 11.6.0. The release of Asterisk 11.6.0 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: * --- Confbridge: empty conference not being torn down (Closes issue ASTERISK-21859. Reported by Chris Gentle) * --- Let Queue wrap up time influence member availability (Closes issue ASTERISK-22189. Reported by Tony Lewis) * --- Fix a longstanding issue with MFC-R2 configuration that prevented users (Closes issue ASTERISK-21117. Reported by Rafael Angulo) * --- chan_iax2: Fix saving the wrong expiry time in astdb. (Closes issue ASTERISK-22504. Reported by Stefan Wachtler) * --- Fix segfault for certain invalid WebSocket input. (Closes issue ASTERISK-21825. Reported by Alfred Farrugia) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.6.0 Thank you for your continued support of Asterisk!
2013-12-23 02:34:03 +01:00
+ ast_queue_log(args.queuename, ast_channel_uniqueid(chan),"NONE", "EXITWITHTIMEOUT", "%d|%d|%jd",
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
+ qe.pos, qe.opos, (intmax_t) time(NULL) - qe.start);
break;
}
Update to Asterisk 11.6.1: this is a security fix update to fix AST-2013-006 and AST-2013-007, and a minor bug fix update. pkgsrc change: disable SRTP on NetBSD as it doesn't link ---- 11.6.1 ---- The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4, 10.12.4-digiumphones, and 11.6.1. The release of these versions resolve the following issues: * A buffer overflow when receiving odd length 16 bit messages in app_sms. An infinite loop could occur which would overwrite memory when a message is received into the unpacksms16() function and the length of the message is an odd number of bytes. * Prevent permissions escalation in the Asterisk Manager Interface. Asterisk now marks certain individual dialplan functions as 'dangerous', which will inhibit their execution from external sources. A 'dangerous' function is one which results in a privilege escalation. For example, if one were to read the channel variable SHELL(rm -rf /) Bad Things(TM) could happen; even if the external source has only read permissions. Execution from external sources may be enabled by setting 'live_dangerously' to 'yes' in the [options] section of asterisk.conf. Although doing so is not recommended. These issues and their resolutions are described in the security advisories. For more information about the details of these vulnerabilities, please read security advisories AST-2013-006 and AST-2013-007, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.6.1 The security advisories are available at: * http://downloads.asterisk.org/pub/security/AST-2013-006.pdf * http://downloads.asterisk.org/pub/security/AST-2013-007.pdf Thank you for your continued support of Asterisk! ----- 11.6.0 ----- The Asterisk Development Team has announced the release of Asterisk 11.6.0. The release of Asterisk 11.6.0 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: * --- Confbridge: empty conference not being torn down (Closes issue ASTERISK-21859. Reported by Chris Gentle) * --- Let Queue wrap up time influence member availability (Closes issue ASTERISK-22189. Reported by Tony Lewis) * --- Fix a longstanding issue with MFC-R2 configuration that prevented users (Closes issue ASTERISK-21117. Reported by Rafael Angulo) * --- chan_iax2: Fix saving the wrong expiry time in astdb. (Closes issue ASTERISK-22504. Reported by Stefan Wachtler) * --- Fix segfault for certain invalid WebSocket input. (Closes issue ASTERISK-21825. Reported by Alfred Farrugia) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.6.0 Thank you for your continued support of Asterisk!
2013-12-23 02:34:03 +01:00
@@ -7133,7 +7133,7 @@ check_turns:
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
if ((status = get_member_status(qe.parent, qe.max_penalty, qe.min_penalty, qe.parent->leavewhenempty))) {
record_abandoned(&qe);
reason = QUEUE_LEAVEEMPTY;
- ast_queue_log(args.queuename, ast_channel_uniqueid(chan), "NONE", "EXITEMPTY", "%d|%d|%ld", qe.pos, qe.opos, (long)(time(NULL) - qe.start));
+ ast_queue_log(args.queuename, ast_channel_uniqueid(chan), "NONE", "EXITEMPTY", "%d|%d|%jd", qe.pos, qe.opos, (intmax_t)(time(NULL) - qe.start));
res = 0;
break;
}
Update to Asterisk 11.6.1: this is a security fix update to fix AST-2013-006 and AST-2013-007, and a minor bug fix update. pkgsrc change: disable SRTP on NetBSD as it doesn't link ---- 11.6.1 ---- The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4, 10.12.4-digiumphones, and 11.6.1. The release of these versions resolve the following issues: * A buffer overflow when receiving odd length 16 bit messages in app_sms. An infinite loop could occur which would overwrite memory when a message is received into the unpacksms16() function and the length of the message is an odd number of bytes. * Prevent permissions escalation in the Asterisk Manager Interface. Asterisk now marks certain individual dialplan functions as 'dangerous', which will inhibit their execution from external sources. A 'dangerous' function is one which results in a privilege escalation. For example, if one were to read the channel variable SHELL(rm -rf /) Bad Things(TM) could happen; even if the external source has only read permissions. Execution from external sources may be enabled by setting 'live_dangerously' to 'yes' in the [options] section of asterisk.conf. Although doing so is not recommended. These issues and their resolutions are described in the security advisories. For more information about the details of these vulnerabilities, please read security advisories AST-2013-006 and AST-2013-007, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.6.1 The security advisories are available at: * http://downloads.asterisk.org/pub/security/AST-2013-006.pdf * http://downloads.asterisk.org/pub/security/AST-2013-007.pdf Thank you for your continued support of Asterisk! ----- 11.6.0 ----- The Asterisk Development Team has announced the release of Asterisk 11.6.0. The release of Asterisk 11.6.0 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: * --- Confbridge: empty conference not being torn down (Closes issue ASTERISK-21859. Reported by Chris Gentle) * --- Let Queue wrap up time influence member availability (Closes issue ASTERISK-22189. Reported by Tony Lewis) * --- Fix a longstanding issue with MFC-R2 configuration that prevented users (Closes issue ASTERISK-21117. Reported by Rafael Angulo) * --- chan_iax2: Fix saving the wrong expiry time in astdb. (Closes issue ASTERISK-22504. Reported by Stefan Wachtler) * --- Fix segfault for certain invalid WebSocket input. (Closes issue ASTERISK-21825. Reported by Alfred Farrugia) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.6.0 Thank you for your continued support of Asterisk!
2013-12-23 02:34:03 +01:00
@@ -7155,7 +7155,7 @@ check_turns:
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
record_abandoned(&qe);
reason = QUEUE_TIMEOUT;
res = 0;
- ast_queue_log(qe.parent->name, ast_channel_uniqueid(qe.chan),"NONE", "EXITWITHTIMEOUT", "%d|%d|%ld", qe.pos, qe.opos, (long) time(NULL) - qe.start);
+ ast_queue_log(qe.parent->name, ast_channel_uniqueid(qe.chan),"NONE", "EXITWITHTIMEOUT", "%d|%d|%jd", qe.pos, qe.opos, (intmax_t) time(NULL) - qe.start);
break;
}
Update to Asterisk 11.6.1: this is a security fix update to fix AST-2013-006 and AST-2013-007, and a minor bug fix update. pkgsrc change: disable SRTP on NetBSD as it doesn't link ---- 11.6.1 ---- The Asterisk Development Team has announced security releases for Certified Asterisk 1.8.15, 11.2, and Asterisk 1.8, 10, and 11. The available security releases are released as versions 1.8.15-cert4, 11.2-cert3, 1.8.24.1, 10.12.4, 10.12.4-digiumphones, and 11.6.1. The release of these versions resolve the following issues: * A buffer overflow when receiving odd length 16 bit messages in app_sms. An infinite loop could occur which would overwrite memory when a message is received into the unpacksms16() function and the length of the message is an odd number of bytes. * Prevent permissions escalation in the Asterisk Manager Interface. Asterisk now marks certain individual dialplan functions as 'dangerous', which will inhibit their execution from external sources. A 'dangerous' function is one which results in a privilege escalation. For example, if one were to read the channel variable SHELL(rm -rf /) Bad Things(TM) could happen; even if the external source has only read permissions. Execution from external sources may be enabled by setting 'live_dangerously' to 'yes' in the [options] section of asterisk.conf. Although doing so is not recommended. These issues and their resolutions are described in the security advisories. For more information about the details of these vulnerabilities, please read security advisories AST-2013-006 and AST-2013-007, which were released at the same time as this announcement. For a full list of changes in the current releases, please see the ChangeLogs: http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.6.1 The security advisories are available at: * http://downloads.asterisk.org/pub/security/AST-2013-006.pdf * http://downloads.asterisk.org/pub/security/AST-2013-007.pdf Thank you for your continued support of Asterisk! ----- 11.6.0 ----- The Asterisk Development Team has announced the release of Asterisk 11.6.0. The release of Asterisk 11.6.0 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: * --- Confbridge: empty conference not being torn down (Closes issue ASTERISK-21859. Reported by Chris Gentle) * --- Let Queue wrap up time influence member availability (Closes issue ASTERISK-22189. Reported by Tony Lewis) * --- Fix a longstanding issue with MFC-R2 configuration that prevented users (Closes issue ASTERISK-21117. Reported by Rafael Angulo) * --- chan_iax2: Fix saving the wrong expiry time in astdb. (Closes issue ASTERISK-22504. Reported by Stefan Wachtler) * --- Fix segfault for certain invalid WebSocket input. (Closes issue ASTERISK-21825. Reported by Alfred Farrugia) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.6.0 Thank you for your continued support of Asterisk!
2013-12-23 02:34:03 +01:00
@@ -7183,8 +7183,8 @@ stop:
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
if (!qe.handled) {
record_abandoned(&qe);
ast_queue_log(args.queuename, ast_channel_uniqueid(chan), "NONE", "ABANDON",
- "%d|%d|%ld", qe.pos, qe.opos,
- (long) time(NULL) - qe.start);
+ "%d|%d|%jd", qe.pos, qe.opos,
+ (intmax_t) time(NULL) - qe.start);
res = -1;
} else if (qcontinue) {
reason = QUEUE_CONTINUE;
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