asterisk/main/asterisk.c

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/*
* Asterisk -- An open source telephony toolkit.
*
* Copyright (C) 1999 - 2018, Digium, Inc.
*
* Mark Spencer <markster@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists and IRC
* channels for your use.
*
* This program is free software, distributed under the terms of
* the GNU General Public License Version 2. See the LICENSE file
* at the top of the source tree.
*/
/* Doxygenified Copyright Header */
/*!
* \mainpage Asterisk -- The Open Source Telephony Project
*
* \par Welcome
*
* This documentation created by the Doxygen project clearly explains the
* internals of the Asterisk software. This documentation contains basic
* examples, developer documentation, support information, and information
* for upgrading.
Refactor RTCP events over to Stasis; associate with channels This patch does the following: * It merges Jaco Kroon's patch from ASTERISK-20754, which provides channel information in the RTCP events. Because Stasis provides a cache, Jaco's patch was modified to pass the channel uniqueid to the RTP layer as opposed to a pointer to the channel. This has the following benefits: (1) It keeps the RTP engine 'clean' of references back to channels (2) It prevents circular dependencies and other potential ref counting issues * The RTP engine now allows any RTP implementation to raise RTCP messages. Potentially, other implementations (such as res_rtp_multicast) could also raise RTCP information. The engine provides structs to represent RTCP headers and RTCP SR/RR reports. * Some general refactoring in res_rtp_asterisk was done to try and tame the RTCP code. It isn't perfect - that's *way* beyond the scope of this work - but it does feel marginally better. * A few random bugs were fixed in the RTCP statistics. (Example: performing an assignment of a = a is probably not correct) * We now raise RTCP events for each SR/RR sent/received. Previously we wouldn't raise an event when we sent a RR report. Note that this work will be of use to others who want to monitor call quality or build modules that report call quality statistics. Since the events are now moving across the Stasis message bus, this is far easier to accomplish. It is also a first step (though by no means the last step) towards getting Olle's pinefrog work incorporated. Again: note that the patch by Jaco Kroon was modified slightly for this work; however, he did all of the hard work in finding the right places to set the channel in the RTP engine across the channel drivers. Much thanks goes to Jaco for his hard work here. Review: https://reviewboard.asterisk.org/r/2603/ (closes issue ASTERISK-20574) Reported by: Jaco Kroon patches: asterisk-rtcp-channel.patch uploaded by jkroon (License 5671) (closes issue ASTERISK-21471) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-05 17:33:33 +00:00
*
* \section community Community
* Asterisk is a big project and has a busy community. Look at the
* resources for questions and stick around to help answer questions.
* \li \ref asterisk_community_resources
*
* \par Developer Documentation for Asterisk
*
* This is the main developer documentation for Asterisk. It is
* generated by running "make progdocs" from the Asterisk source tree.
*
* In addition to the information available on the Asterisk source code,
* please see the appendices for information on coding guidelines,
* release management, commit policies, and more.
*
* \arg \ref AsteriskArchitecture
*
* \par Additional documentation
* \arg \ref Licensing
* \arg \ref DevDoc
* \arg \ref configuration_file
* \arg \ref channel_drivers
* \arg \ref applications
*
* \section copyright Copyright and Author
*
* Copyright (C) 1999 - 2021, Sangoma Technologies Corporation.
* Asterisk is a <a href="https://cdn.sangoma.com/wp-content/uploads/Sangoma-Trademark-Policy.pdf">registered trademark</a>
* of <a rel="nofollow" href="http://www.sangoma.com">Sangoma Technologies Corporation</a>.
*
* \author Mark Spencer <markster@digium.com>
*
* See http://www.asterisk.org for more information about
* the Asterisk project. Please do not directly contact
* any of the maintainers of this project for assistance;
* the project provides a web site, mailing lists, and IRC
* channels for your use.
*
*/
/*!
* \page asterisk_community_resources Asterisk Community Resources
* \par Websites
* \li https://www.asterisk.org Asterisk Homepage
* \li https://docs.asterisk.org Asterisk documentation
*
* \par Mailing Lists
* \par
* All lists: http://lists.digium.com/mailman/listinfo
* \li aadk-commits SVN commits to the AADK repository
* \li asterisk-addons-commits SVN commits to the Asterisk addons project
* \li asterisk-announce [no description available]
* \li asterisk-biz Commercial and Business-Oriented Asterisk Discussion
* \li Asterisk-BSD Asterisk on BSD discussion
* \li asterisk-bugs [no description available]
* \li asterisk-commits SVN commits to the Asterisk project
* \li asterisk-dev Asterisk Developers Mailing List
* \li asterisk-doc Discussions regarding The Asterisk Documentation Project
* \li asterisk-embedded Asterisk Embedded Development
* \li asterisk-gui Asterisk GUI project discussion
* \li asterisk-gui-commits SVN commits to the Asterisk-GUI project
* \li asterisk-ha-clustering Asterisk High Availability and Clustering List - Non-Commercial Discussion
* \li Asterisk-i18n Discussion of Asterisk internationalization
* \li asterisk-r2 [no description available]
* \li asterisk-scf-commits Commits to the Asterisk SCF project code repositories
* \li asterisk-scf-committee Asterisk SCF Steering Committee discussions
* \li asterisk-scf-dev Asterisk SCF Developers Mailing List
* \li asterisk-scf-wiki-changes Changes to the Asterisk SCF space on wiki.asterisk.org
* \li asterisk-security Asterisk Security Discussion
* \li asterisk-speech-rec Use of speech recognition in Asterisk
* \li asterisk-ss7 [no description available]
* \li asterisk-users Asterisk Users Mailing List - Non-Commercial Discussion
* \li asterisk-video Development discussion of video media support in Asterisk
* \li asterisk-wiki-changes Changes to the Asterisk space on wiki.asterisk.org
* \li asterisknow AsteriskNOW Discussion
* \li dahdi-commits SVN commits to the DAHDI project
* \li digium-announce Digium Product Announcements
* \li Dundi Distributed Universal Number Discovery
* \li libiax2-commits SVN commits to the libiax2 project
* \li libpri-commits SVN commits to the libpri project
* \li libss7-commits SVN commits to the libss7 project
* \li svn-commits SVN commits to the Digium repositories
* \li Test-results Results from automated testing
* \li thirdparty-commits SVN commits to the Digium third-party software repository
* \li zaptel-commits SVN commits to the Zaptel project
*
* \par Forums
* \li Forums are located at http://forums.asterisk.org/
*
* \par IRC
* \par
* Use https://libera.chat IRC server to connect with Asterisk
* developers and users in realtime.
Refactor RTCP events over to Stasis; associate with channels This patch does the following: * It merges Jaco Kroon's patch from ASTERISK-20754, which provides channel information in the RTCP events. Because Stasis provides a cache, Jaco's patch was modified to pass the channel uniqueid to the RTP layer as opposed to a pointer to the channel. This has the following benefits: (1) It keeps the RTP engine 'clean' of references back to channels (2) It prevents circular dependencies and other potential ref counting issues * The RTP engine now allows any RTP implementation to raise RTCP messages. Potentially, other implementations (such as res_rtp_multicast) could also raise RTCP information. The engine provides structs to represent RTCP headers and RTCP SR/RR reports. * Some general refactoring in res_rtp_asterisk was done to try and tame the RTCP code. It isn't perfect - that's *way* beyond the scope of this work - but it does feel marginally better. * A few random bugs were fixed in the RTCP statistics. (Example: performing an assignment of a = a is probably not correct) * We now raise RTCP events for each SR/RR sent/received. Previously we wouldn't raise an event when we sent a RR report. Note that this work will be of use to others who want to monitor call quality or build modules that report call quality statistics. Since the events are now moving across the Stasis message bus, this is far easier to accomplish. It is also a first step (though by no means the last step) towards getting Olle's pinefrog work incorporated. Again: note that the patch by Jaco Kroon was modified slightly for this work; however, he did all of the hard work in finding the right places to set the channel in the RTP engine across the channel drivers. Much thanks goes to Jaco for his hard work here. Review: https://reviewboard.asterisk.org/r/2603/ (closes issue ASTERISK-20574) Reported by: Jaco Kroon patches: asterisk-rtcp-channel.patch uploaded by jkroon (License 5671) (closes issue ASTERISK-21471) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-05 17:33:33 +00:00
*
* \li \verbatim #asterisk \endverbatim Asterisk Users Room
* \li \verbatim #asterisk-dev \endverbatim Asterisk Developers Room
*
* \par More
* \par
* If you would like to add a resource to this list please create an issue
* on the issue tracker with a patch.
*/
/*! \file
* \brief Top level source file for Asterisk - the Open Source PBX.
* Implementation of PBX core functions and CLI interface.
*/
/*! \li \ref asterisk.c uses the configuration file \ref asterisk.conf
* \addtogroup configuration_file
*/
/*! \page asterisk.conf asterisk.conf
* \verbinclude asterisk.conf.sample
*/
/*** MODULEINFO
<support_level>core</support_level>
***/
#include "asterisk.h"
#include "asterisk/_private.h"
#undef sched_setscheduler
#undef setpriority
#include <sys/time.h>
#include <fcntl.h>
#include <signal.h>
#include <sched.h>
#include <sys/un.h>
#include <sys/wait.h>
#include <ctype.h>
#include <sys/resource.h>
#include <grp.h>
#include <pwd.h>
#include <sys/stat.h>
#if defined(HAVE_SYSINFO)
#include <sys/sysinfo.h>
#elif defined(HAVE_SYSCTL)
#include <sys/param.h>
#include <sys/sysctl.h>
#include <sys/vmmeter.h>
#if defined(__FreeBSD__) || defined(__DragonFly__)
#include <vm/vm_param.h>
#endif
#if defined(HAVE_SWAPCTL)
#include <sys/swap.h>
#endif
#endif
#include <regex.h>
#include <histedit.h>
#if defined(SOLARIS)
int daemon(int, int); /* defined in libresolv of all places */
#include <sys/loadavg.h>
#endif
#ifdef linux
#include <sys/prctl.h>
#ifdef HAVE_CAP
#include <sys/capability.h>
#endif /* HAVE_CAP */
#endif /* linux */
/* we define here the variables so to better agree on the prototype */
#include "asterisk/paths.h"
#include "asterisk/network.h"
#include "asterisk/cli.h"
#include "asterisk/channel.h"
#include "asterisk/translate.h"
#include "asterisk/pickup.h"
#include "asterisk/acl.h"
#include "asterisk/ulaw.h"
#include "asterisk/alaw.h"
#include "asterisk/callerid.h"
#include "asterisk/image.h"
#include "asterisk/tdd.h"
#include "asterisk/term.h"
#include "asterisk/manager.h"
#include "asterisk/cdr.h"
#include "asterisk/pbx.h"
#include "asterisk/app.h"
#include "asterisk/mwi.h"
#include "asterisk/lock.h"
#include "asterisk/utils.h"
#include "asterisk/file.h"
#include "asterisk/io.h"
#include "asterisk/config.h"
#include "asterisk/ast_version.h"
#include "asterisk/linkedlists.h"
#include "asterisk/devicestate.h"
Merge changes dealing with support for Digium phones. Presence support has been added. This is accomplished by allowing for presence hints in addition to device state hints. A dialplan function called PRESENCE_STATE has been added to allow for setting and reading presence. Presence can be transmitted to Digium phones using custom XML elements in a PIDF presence document. Voicemail has new APIs that allow for moving, removing, forwarding, and playing messages. Messages have had a new unique message ID added to them so that the APIs will work reliably. The state of a voicemail mailbox can be obtained using an API that allows one to get a snapshot of the mailbox. A voicemail Dialplan App called VoiceMailPlayMsg has been added to be able to play back a specific message. Configuration hooks have been added. Configuration hooks allow for a piece of code to be executed when a specific configuration file is loaded by a specific module. This is useful for modules that are dependent on the configuration of other modules. chan_sip now has a public method that allows for a custom SIP INFO request to be sent mid-dialog. Digium phones use this in order to display progress bars when files are played. Messaging support has been expanded a bit. The main visible difference is the addition of an AMI action MessageSend. Finally, a ParkingLots manager action has been added in order to get a list of parking lots. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-04 20:26:12 +00:00
#include "asterisk/presencestate.h"
#include "asterisk/module.h"
#include "asterisk/buildinfo.h"
#include "asterisk/xmldoc.h"
Merged revisions 182810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-18 02:28:55 +00:00
#include "asterisk/poll-compat.h"
#include "asterisk/test.h"
#include "asterisk/rtp_engine.h"
#include "asterisk/format.h"
#include "asterisk/aoc.h"
#include "asterisk/uuid.h"
#include "asterisk/sorcery.h"
#include "asterisk/bucket.h"
#include "asterisk/stasis.h"
#include "asterisk/json.h"
Initial support for endpoints. An endpoint is an external device/system that may offer/accept channels to/from Asterisk. While this is a very useful concept for end users, it is surprisingly not a core concept within Asterisk itself. This patch defines ast_endpoint as a separate object, which channel drivers may use to expose their concept of an endpoint. As the channel driver creates channels, it can use ast_endpoint_add_channel() to associate channels to the endpoint. This updated the endpoint appropriately, and forwards all of the channel's events to the endpoint's topic. In order to avoid excessive locking on the endpoint object itself, the mutable state is not accessible via getters. Instead, you can create a snapshot using ast_endpoint_snapshot_create() to get a consistent snapshot of the internal state. This patch also includes a set of topics and messages associated with endpoints, and implementations of the endpoint-related RESTful API. chan_sip was updated to create endpoints with SIP peers, but the state of the endpoints is not updated with the state of the peer. Along for the ride in this patch is a Stasis test API. This is a stasis_message_sink object, which can be subscribed to a Stasis topic. It has functions for blocking while waiting for conditions in the message sink to be fulfilled. (closes issue ASTERISK-21421) Review: https://reviewboard.asterisk.org/r/2492/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-08 13:39:08 +00:00
#include "asterisk/stasis_endpoints.h"
#include "asterisk/stasis_system.h"
#include "asterisk/security_events.h"
#include "asterisk/endpoints.h"
media formats: re-architect handling of media for performance improvements In the old times media formats were represented using a bit field. This was fast but had a few limitations. 1. Asterisk was limited in how many formats it could handle. 2. Formats, being a bit field, could not include any attribute information. A format was strictly its type, e.g., "this is ulaw". This was changed in Asterisk 10 (see https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for notes on that work) which led to the creation of the ast_format structure. This structure allowed Asterisk to handle attributes and bundle information with a format. Additionally, ast_format_cap was created to act as a container for multiple formats that, together, formed the capability of some entity. Another mechanism was added to allow logic to be registered which performed format attribute negotiation. Everywhere throughout the codebase Asterisk was changed to use this strategy. Unfortunately, in software, there is no free lunch. These new capabilities came at a cost. Performance analysis and profiling showed that we spend an inordinate amount of time comparing, copying, and generally manipulating formats and their related structures. Basic prototyping has shown that a reasonably large performance improvement could be made in this area. This patch is the result of that project, which overhauled the media format architecture and its usage in Asterisk to improve performance. Generally, the new philosophy for handling formats is as follows: * The ast_format structure is reference counted. This removed a large amount of the memory allocations and copying that was done in prior versions. * In order to prevent race conditions while keeping things performant, the ast_format structure is immutable by convention and lock-free. Violate this tenet at your peril! * Because formats are reference counted, codecs are also reference counted. The Asterisk core generally provides built-in codecs and caches the ast_format structures created to represent them. Generally, to prevent inordinate amounts of module reference bumping, codecs and formats can be added at run-time but cannot be removed. * All compatibility with the bit field representation of codecs/formats has been moved to a compatibility API. The primary user of this representation is chan_iax2, which must continue to maintain its bit-field usage of formats for interoperability concerns. * When a format is negotiated with attributes, or when a format cannot be represented by one of the cached formats, a new format object is created or cloned from an existing format. That format may have the same codec underlying it, but is a different format than a version of the format with different attributes or without attributes. * While formats are reference counted objects, the reference count maintained on the format should be manipulated with care. Formats are generally cached and will persist for the lifetime of Asterisk and do not explicitly need to have their lifetime modified. An exception to this is when the user of a format does not know where the format came from *and* the user may outlive the provider of the format. This occurs, for example, when a format is read from a channel: the channel may have a format with attributes (hence, non-cached) and the user of the format may last longer than the channel (if the reference to the channel is released prior to the format's reference). For more information on this work, see the API design notes: https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite Finally, this work was the culmination of a large number of developer's efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the work in the Asterisk core, chan_sip, and was an invaluable resource in peer reviews throughout this project. There were a substantial number of patches contributed during this work; the following issues/patch names simply reflect some of the work (and will cause the release scripts to give attribution to the individuals who work on them). Reviews: https://reviewboard.asterisk.org/r/3814 https://reviewboard.asterisk.org/r/3808 https://reviewboard.asterisk.org/r/3805 https://reviewboard.asterisk.org/r/3803 https://reviewboard.asterisk.org/r/3801 https://reviewboard.asterisk.org/r/3798 https://reviewboard.asterisk.org/r/3800 https://reviewboard.asterisk.org/r/3794 https://reviewboard.asterisk.org/r/3793 https://reviewboard.asterisk.org/r/3792 https://reviewboard.asterisk.org/r/3791 https://reviewboard.asterisk.org/r/3790 https://reviewboard.asterisk.org/r/3789 https://reviewboard.asterisk.org/r/3788 https://reviewboard.asterisk.org/r/3787 https://reviewboard.asterisk.org/r/3786 https://reviewboard.asterisk.org/r/3784 https://reviewboard.asterisk.org/r/3783 https://reviewboard.asterisk.org/r/3778 https://reviewboard.asterisk.org/r/3774 https://reviewboard.asterisk.org/r/3775 https://reviewboard.asterisk.org/r/3772 https://reviewboard.asterisk.org/r/3761 https://reviewboard.asterisk.org/r/3754 https://reviewboard.asterisk.org/r/3753 https://reviewboard.asterisk.org/r/3751 https://reviewboard.asterisk.org/r/3750 https://reviewboard.asterisk.org/r/3748 https://reviewboard.asterisk.org/r/3747 https://reviewboard.asterisk.org/r/3746 https://reviewboard.asterisk.org/r/3742 https://reviewboard.asterisk.org/r/3740 https://reviewboard.asterisk.org/r/3739 https://reviewboard.asterisk.org/r/3738 https://reviewboard.asterisk.org/r/3737 https://reviewboard.asterisk.org/r/3736 https://reviewboard.asterisk.org/r/3734 https://reviewboard.asterisk.org/r/3722 https://reviewboard.asterisk.org/r/3713 https://reviewboard.asterisk.org/r/3703 https://reviewboard.asterisk.org/r/3689 https://reviewboard.asterisk.org/r/3687 https://reviewboard.asterisk.org/r/3674 https://reviewboard.asterisk.org/r/3671 https://reviewboard.asterisk.org/r/3667 https://reviewboard.asterisk.org/r/3665 https://reviewboard.asterisk.org/r/3625 https://reviewboard.asterisk.org/r/3602 https://reviewboard.asterisk.org/r/3519 https://reviewboard.asterisk.org/r/3518 https://reviewboard.asterisk.org/r/3516 https://reviewboard.asterisk.org/r/3515 https://reviewboard.asterisk.org/r/3512 https://reviewboard.asterisk.org/r/3506 https://reviewboard.asterisk.org/r/3413 https://reviewboard.asterisk.org/r/3410 https://reviewboard.asterisk.org/r/3387 https://reviewboard.asterisk.org/r/3388 https://reviewboard.asterisk.org/r/3389 https://reviewboard.asterisk.org/r/3390 https://reviewboard.asterisk.org/r/3321 https://reviewboard.asterisk.org/r/3320 https://reviewboard.asterisk.org/r/3319 https://reviewboard.asterisk.org/r/3318 https://reviewboard.asterisk.org/r/3266 https://reviewboard.asterisk.org/r/3265 https://reviewboard.asterisk.org/r/3234 https://reviewboard.asterisk.org/r/3178 ASTERISK-23114 #close Reported by: mjordan media_formats_translation_core.diff uploaded by kharwell (License 6464) rb3506.diff uploaded by mjordan (License 6283) media_format_app_file.diff uploaded by kharwell (License 6464) misc-2.diff uploaded by file (License 5000) chan_mild-3.diff uploaded by file (License 5000) chan_obscure.diff uploaded by file (License 5000) jingle.diff uploaded by file (License 5000) funcs.diff uploaded by file (License 5000) formats.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) bridges.diff uploaded by file (License 5000) mf-codecs-2.diff uploaded by file (License 5000) mf-app_fax.diff uploaded by file (License 5000) mf-apps-3.diff uploaded by file (License 5000) media-formats-3.diff uploaded by file (License 5000) ASTERISK-23715 rb3713.patch uploaded by coreyfarrell (License 5909) rb3689.patch uploaded by mjordan (License 6283) ASTERISK-23957 rb3722.patch uploaded by mjordan (License 6283) mf-attributes-3.diff uploaded by file (License 5000) ASTERISK-23958 Tested by: jrose rb3822.patch uploaded by coreyfarrell (License 5909) rb3800.patch uploaded by jrose (License 6182) chan_sip.diff uploaded by mjordan (License 6283) rb3747.patch uploaded by jrose (License 6182) ASTERISK-23959 #close Tested by: sgriepentrog, mjordan, coreyfarrell sip_cleanup.diff uploaded by opticron (License 6273) chan_sip_caps.diff uploaded by mjordan (License 6283) rb3751.patch uploaded by coreyfarrell (License 5909) chan_sip-3.diff uploaded by file (License 5000) ASTERISK-23960 #close Tested by: opticron direct_media.diff uploaded by opticron (License 6273) pjsip-direct-media.diff uploaded by file (License 5000) format_cap_remove.diff uploaded by opticron (License 6273) media_format_fixes.diff uploaded by opticron (License 6273) chan_pjsip-2.diff uploaded by file (License 5000) ASTERISK-23966 #close Tested by: rmudgett rb3803.patch uploaded by rmudgetti (License 5621) chan_dahdi.diff uploaded by file (License 5000) ASTERISK-24064 #close Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose rb3814.patch uploaded by rmudgett (License 5621) moh_cleanup.diff uploaded by opticron (License 6273) bridge_leak.diff uploaded by opticron (License 6273) translate.diff uploaded by file (License 5000) rb3795.patch uploaded by rmudgett (License 5621) tls_fix.diff uploaded by mjordan (License 6283) fax-mf-fix-2.diff uploaded by file (License 5000) rtp_transfer_stuff uploaded by mjordan (License 6283) rb3787.patch uploaded by rmudgett (License 5621) media-formats-explicit-translate-format-3.diff uploaded by file (License 5000) format_cache_case_fix.diff uploaded by opticron (License 6273) rb3774.patch uploaded by rmudgett (License 5621) rb3775.patch uploaded by rmudgett (License 5621) rtp_engine_fix.diff uploaded by opticron (License 6273) rtp_crash_fix.diff uploaded by opticron (License 6273) rb3753.patch uploaded by mjordan (License 6283) rb3750.patch uploaded by mjordan (License 6283) rb3748.patch uploaded by rmudgett (License 5621) media_format_fixes.diff uploaded by opticron (License 6273) rb3740.patch uploaded by mjordan (License 6283) rb3739.patch uploaded by mjordan (License 6283) rb3734.patch uploaded by mjordan (License 6283) rb3689.patch uploaded by mjordan (License 6283) rb3674.patch uploaded by coreyfarrell (License 5909) rb3671.patch uploaded by coreyfarrell (License 5909) rb3667.patch uploaded by coreyfarrell (License 5909) rb3665.patch uploaded by mjordan (License 6283) rb3625.patch uploaded by coreyfarrell (License 5909) rb3602.patch uploaded by coreyfarrell (License 5909) format_compatibility-2.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-20 22:06:33 +00:00
#include "asterisk/codec.h"
#include "asterisk/format_cache.h"
media cache: Add a core API and facade for a backend agnostic media cache This patch adds a new API to the Asterisk core that acts as a media cache. The core API itself is mostly a thin wrapper around some bucket API provided implementation that itself acts as the mechanism of retrieval for media. The media cache API in the core provides the following: * A very thin in-memory cache of the active bucket_file items. Unlike a more traditional cache, it provides no expiration mechanisms. Most queries that hit the in-memory cache will also call into the bucket implementations as well. The bucket implementations are responsible for determining whether or not the active record is active and valid. This makes sense for the most likely implementation of a media cache backend, i.e., HTTP. The HTTP layer itself is the actual arbiter of whether or not a record is truly active; as such, the in-memory cache in the core has to defer to it. * The ability to create new items in the media cache from local resources. This allows for re-creation of items in the cache on restart. * Synchronization of items in the media cache to the AstDB. This also includes various pieces of important metadata. The API provides sufficient access that higher level APIs, such as the file or app APIs, do not have to worry about the semantics of the bucket APIs when needing to playback a resource. In addition, this patch provides unit tests for the media cache API. The unit tests use a fake bucket backend to verify correctness. Change-Id: I11227abbf14d8929eeb140ddd101dd5c3820391e
2015-01-29 14:38:23 +00:00
#include "asterisk/media_cache.h"
#include "asterisk/astdb.h"
#include "asterisk/options.h"
#include "asterisk/utf8.h"
#include "../defaults.h"
/*** DOCUMENTATION
<managerEvent language="en_US" name="FullyBooted">
<managerEventInstance class="EVENT_FLAG_SYSTEM">
<synopsis>Raised when all Asterisk initialization procedures have finished.</synopsis>
<syntax>
<parameter name="Status">
<para>Informational message</para>
</parameter>
<parameter name="Uptime">
<para>Seconds since start</para>
</parameter>
<parameter name="LastReload">
<para>Seconds since last reload</para>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
<managerEvent language="en_US" name="Shutdown">
<managerEventInstance class="EVENT_FLAG_SYSTEM">
<synopsis>Raised when Asterisk is shutdown or restarted.</synopsis>
<syntax>
<parameter name="Shutdown">
<para>Whether the shutdown is proceeding cleanly (all channels
were hungup successfully) or uncleanly (channels will be
terminated)</para>
<enumlist>
<enum name="Uncleanly"/>
<enum name="Cleanly"/>
</enumlist>
</parameter>
<parameter name="Restart">
<para>Whether or not a restart will occur.</para>
<enumlist>
<enum name="True"/>
<enum name="False"/>
</enumlist>
</parameter>
</syntax>
</managerEventInstance>
</managerEvent>
***/
#ifndef AF_LOCAL
#define AF_LOCAL AF_UNIX
#define PF_LOCAL PF_UNIX
#endif
#define AST_MAX_CONNECTS 128
#define NUM_MSGS 64
/*! Displayed copyright tag */
#define COPYRIGHT_TAG "Copyright (C) 1999 - 2022, Sangoma Technologies Corporation and others."
/*! \brief Welcome message when starting a CLI interface */
#define WELCOME_MESSAGE \
ast_verbose("Asterisk %s, " COPYRIGHT_TAG "\n" \
"Created by Mark Spencer <markster@digium.com>\n" \
"Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.\n" \
"This is free software, with components licensed under the GNU General Public\n" \
"License version 2 and other licenses; you are welcome to redistribute it under\n" \
"certain conditions. Type 'core show license' for details.\n" \
"=========================================================================\n", ast_get_version()) \
static int ast_socket = -1; /*!< UNIX Socket for allowing remote control */
static int ast_socket_is_sd = 0; /*!< Is socket activation responsible for ast_socket? */
static int ast_consock = -1; /*!< UNIX Socket for controlling another asterisk */
pid_t ast_mainpid;
struct console {
int fd; /*!< File descriptor */
int p[2]; /*!< Pipe */
pthread_t t; /*!< Thread of handler */
int mute; /*!< Is the console muted for logs */
int uid; /*!< Remote user ID. */
int gid; /*!< Remote group ID. */
int levels[NUMLOGLEVELS]; /*!< Which log levels are enabled for the console */
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
/*! Verbosity level of this console. */
int option_verbose;
};
struct ast_atexit {
void (*func)(void);
Avoid unnecessary cleanups during immediate shutdown This patch addresses issues during immediate shutdowns, where modules are not unloaded, but Asterisk atexit handlers are run. In the typical case, this usually isn't a big deal. But the introduction of the Stasis message bus makes it much more likely for asynchronous activity to be happening off in some thread during shutdown. During an immediate shutdown, Asterisk skips unloading modules. But while it is processing the atexit handlers, there is a window of time where some of the core message types have been cleaned up, but the message bus is still running. Specifically, it's still running module subscriptions that might be using the core message types. If a message is received by that subscription in that window, it will attempt to use a message type that has been cleaned up. To solve this problem, this patch introduces ast_register_cleanup(). This function operates identically to ast_register_atexit(), except that cleanup calls are not invoked on an immediate shutdown. All of the core message type and topic cleanup was moved from atexit handlers to cleanup handlers. This ensures that core type and topic cleanup only happens if the modules that used them are first unloaded. This patch also changes the ast_assert() when accessing a cleaned up or uninitialized message type to an error log message. Message type functions are actually NULL safe across the board, so the assert was a bit heavy handed. Especially for anyone with DO_CRASH enabled. Review: https://reviewboard.asterisk.org/r/2562/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-30 17:05:53 +00:00
int is_cleanup;
AST_LIST_ENTRY(ast_atexit) list;
};
static AST_LIST_HEAD_STATIC(atexits, ast_atexit);
struct timeval ast_startuptime;
struct timeval ast_lastreloadtime;
static History *el_hist;
static EditLine *el;
static char *remotehostname;
struct console consoles[AST_MAX_CONNECTS];
static int ast_el_add_history(const char *);
static int ast_el_read_history(const char *);
static int ast_el_write_history(const char *);
static void ast_el_read_default_histfile(void);
static void ast_el_write_default_histfile(void);
static void asterisk_daemon(int isroot, const char *runuser, const char *rungroup);
static char *_argv[256];
HTTP: Stop accepting requests on final system shutdown. There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ ........ Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-11 17:39:13 +00:00
typedef enum {
HTTP: Stop accepting requests on final system shutdown. There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ ........ Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-11 17:39:13 +00:00
/*! Normal operation */
NOT_SHUTTING_DOWN,
/*! Committed to shutting down. Final phase */
SHUTTING_DOWN_FINAL,
/*! Committed to shutting down. Initial phase */
SHUTTING_DOWN,
/*!
* Valid values for quit_handler() niceness below.
* These shutdown/restart levels can be cancelled.
*
* Remote console exit right now
*/
SHUTDOWN_FAST,
HTTP: Stop accepting requests on final system shutdown. There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ ........ Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-11 17:39:13 +00:00
/*! core stop/restart now */
SHUTDOWN_NORMAL,
HTTP: Stop accepting requests on final system shutdown. There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ ........ Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-11 17:39:13 +00:00
/*! core stop/restart gracefully */
SHUTDOWN_NICE,
HTTP: Stop accepting requests on final system shutdown. There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ ........ Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-11 17:39:13 +00:00
/*! core stop/restart when convenient */
SHUTDOWN_REALLY_NICE
} shutdown_nice_t;
HTTP: Stop accepting requests on final system shutdown. There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ ........ Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-11 17:39:13 +00:00
static shutdown_nice_t shuttingdown = NOT_SHUTTING_DOWN;
HTTP: Stop accepting requests on final system shutdown. There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ ........ Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-11 17:39:13 +00:00
/*! Prevent new channel allocation for shutdown. */
static int shutdown_pending;
static int restartnow;
static pthread_t consolethread = AST_PTHREADT_NULL;
static pthread_t mon_sig_flags;
static int canary_pid = 0;
static char canary_filename[128];
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
static int multi_thread_safe;
static char randompool[256];
#ifdef HAVE_CAP
static cap_t child_cap;
#endif
static int sig_alert_pipe[2] = { -1, -1 };
static struct {
unsigned int need_reload:1;
unsigned int need_quit:1;
unsigned int need_quit_handler:1;
unsigned int need_el_end:1;
} sig_flags;
#if !defined(LOW_MEMORY)
struct thread_list_t {
AST_RWLIST_ENTRY(thread_list_t) list;
char *name;
pthread_t id;
int lwp;
};
static AST_RWLIST_HEAD_STATIC(thread_list, thread_list_t);
void ast_register_thread(char *name)
{
struct thread_list_t *new = ast_calloc(1, sizeof(*new));
if (!new)
return;
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
ast_assert(multi_thread_safe);
new->id = pthread_self();
new->lwp = ast_get_tid();
new->name = name; /* steal the allocated memory for the thread name */
AST_RWLIST_WRLOCK(&thread_list);
AST_RWLIST_INSERT_HEAD(&thread_list, new, list);
AST_RWLIST_UNLOCK(&thread_list);
}
void ast_unregister_thread(void *id)
{
struct thread_list_t *x;
AST_RWLIST_WRLOCK(&thread_list);
AST_RWLIST_TRAVERSE_SAFE_BEGIN(&thread_list, x, list) {
if ((void *) x->id == id) {
AST_RWLIST_REMOVE_CURRENT(list);
break;
}
}
AST_RWLIST_TRAVERSE_SAFE_END;
AST_RWLIST_UNLOCK(&thread_list);
if (x) {
ast_free(x->name);
ast_free(x);
}
}
/*! \brief Print the contents of a file */
static int print_file(int fd, char *desc, const char *filename)
{
FILE *f;
char c;
if (!(f = fopen(filename, "r"))) {
return -1;
}
ast_cli(fd, "%s", desc);
while ((c = fgetc(f)) != EOF) {
ast_cli(fd, "%c", c);
}
fclose(f);
/* no need for trailing new line, the file already has one */
return 0;
}
/*! \brief Give an overview of core settings */
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
static char *handle_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
char buf[BUFSIZ];
struct ast_tm tm;
char eid_str[128];
struct rlimit limits;
char pbx_uuid[AST_UUID_STR_LEN];
#if defined(HAVE_EACCESS) || defined(HAVE_EUIDACCESS)
char dir[PATH_MAX];
#endif
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
switch (cmd) {
case CLI_INIT:
e->command = "core show settings";
e->usage = "Usage: core show settings\n"
" Show core misc settings";
return NULL;
case CLI_GENERATE:
return NULL;
}
ast_eid_to_str(eid_str, sizeof(eid_str), &ast_eid_default);
ast_pbx_uuid_get(pbx_uuid, sizeof(pbx_uuid));
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
ast_cli(a->fd, "\nPBX Core settings\n");
ast_cli(a->fd, "-----------------\n");
ast_cli(a->fd, " Version: %s\n", ast_get_version());
make_buildopts_h, et. al. Allow adding all cflags to buildopts.h The previous behavior of make_buildopts_h was to not add the non-ABI-breaking MENUSELECT_CFLAGS like DETECT_DEADLOCKS, REF_DEBUG, etc. to the buildopts.h file because "it caused ccache to invalidate files and extended compile times". They're only defined by passing them on the gcc command line with '-D' options. In practice, including them in the include file rarely causes any impact because the only time ccache cares is if you actually change an option so the hit occurrs only once after you change it. OK so why would we want to include them? Many IDEs follow the include files to resolve defines and if the options aren't in an include file, it can cause the IDE to mark blocks of "ifdeffed" code as unused when they're really not. So... * Added a new menuselect compile option ADD_CFLAGS_TO_BUILDOPTS_H which tells make_buildopts_h to include the non-ABI-breaking flags in buildopts.h as well as the ABI-breaking ones. The default is disabled to preserve current behavior. As before though, only the ABI-breaking flags appear in AST_BUILDOPTS and only those are used to calculate AST_BUILDOPT_SUM. A new AST_BUILDOPT_ALL define was created to capture all of the flags. * make_version_c was streamlined to use buildopts.h and also to create asterisk_build_opts_all[] and ast_get_build_opts_all(void) * "core show settings" now shows both AST_BUILDOPTS and AST_BUILDOPTS_ALL. UserNote: The "Build Options" entry in the "core show settings" CLI command has been renamed to "ABI related Build Options" and a new entry named "All Build Options" has been added that shows both breaking and non-breaking options.
2023-09-13 20:08:02 +00:00
ast_cli(a->fd, " ABI related Build Options: %s\n", S_OR(ast_get_build_opts(), "(none)"));
ast_cli(a->fd, " All Build Options: %s\n", S_OR(ast_get_build_opts_all(), "(none)"));
if (ast_option_maxcalls)
ast_cli(a->fd, " Maximum calls: %d (Current %d)\n", ast_option_maxcalls, ast_active_channels());
else
ast_cli(a->fd, " Maximum calls: Not set\n");
if (getrlimit(RLIMIT_NOFILE, &limits)) {
ast_cli(a->fd, " Maximum open file handles: Error because of %s\n", strerror(errno));
} else if (limits.rlim_cur == RLIM_INFINITY) {
ast_cli(a->fd, " Maximum open file handles: Unlimited\n");
} else if (limits.rlim_cur < ast_option_maxfiles) {
ast_cli(a->fd, " Maximum open file handles: %d (is) %d (requested)\n", (int) limits.rlim_cur, ast_option_maxfiles);
} else {
ast_cli(a->fd, " Maximum open file handles: %d\n", (int) limits.rlim_cur);
}
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
ast_cli(a->fd, " Root console verbosity: %d\n", option_verbose);
ast_cli(a->fd, " Current console verbosity: %d\n", ast_verb_console_get());
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
ast_cli(a->fd, " Debug level: %d\n", option_debug);
Scope Tracing: A new facility for tracing scope enter/exit What's wrong with ast_debug? ast_debug is fine for general purpose debug output but it's not really geared for scope tracing since it doesn't present its output in a way that makes capturing and analyzing flow through Asterisk easy. How is scope tracing better? Scope tracing uses the same "cleanup" attribute that RAII_VAR uses to print messages to a separate "trace" log level. Even better, the messages are indented and unindented based on a thread-local call depth counter. When output to a separate log file, the output is uncluttered and easy to follow. Here's an example of the output. The leading timestamps and thread ids are removed and the output cut off at 68 columns for commit message restrictions but you get the idea. --> res_pjsip_session.c:3680 handle_incoming PJSIP/1173-00000001 --> res_pjsip_session.c:3661 handle_incoming_response PJSIP/1173 --> res_pjsip_session.c:3669 handle_incoming_response PJSIP/ --> chan_pjsip.c:3265 chan_pjsip_incoming_response_after --> chan_pjsip.c:3194 chan_pjsip_incoming_response P chan_pjsip.c:3245 chan_pjsip_incoming_respon <-- chan_pjsip.c:3194 chan_pjsip_incoming_response P <-- chan_pjsip.c:3265 chan_pjsip_incoming_response_after <-- res_pjsip_session.c:3669 handle_incoming_response PJSIP/ <-- res_pjsip_session.c:3661 handle_incoming_response PJSIP/1173 <-- res_pjsip_session.c:3680 handle_incoming PJSIP/1173-00000001 The messages with the "-->" or "<--" were produced by including the following at the top of each function: SCOPE_TRACE(1, "%s\n", ast_sip_session_get_name(session)); Scope isn't limited to functions any more than RAII_VAR is. You can also see entry and exit from "if", "for", "while", etc blocks. There is also an ast_trace() macro that doesn't track entry or exit but simply outputs a message to the trace log using the current indent level. The deepest message in the sample (chan_pjsip.c:3245) was used to indicate which "case" in a "select" was executed. How do you use it? More documentation is available in logger.h but here's an overview: * Configure with --enable-dev-mode. Like debug, scope tracing is #ifdef'd out if devmode isn't enabled. * Add a SCOPE_TRACE() call to the top of your function. * Set a logger channel in logger.conf to output the "trace" level. * Use the CLI (or cli.conf) to set a trace level similar to setting debug level... CLI> core set trace 2 res_pjsip.so Summary Of Changes: * Added LOG_TRACE logger level. Actually it occupies the slot formerly occupied by the now defunct "event" level. * Added core asterisk option "trace" similar to debug. Includes ability to specify global trace level in asterisk.conf and CLI commands to turn on/off and set levels. Levels can be set globally (probably not a good idea), or by module/source file. * Updated sample asterisk.conf and logger.conf. Tracing is disabled by default in both. * Added __ast_trace() to logger.c which keeps track of the indent level using TLS. It's #ifdef'd out if devmode isn't enabled. * Added ast_trace() and SCOPE_TRACE() macros to logger.h. These are all #ifdef'd out if devmode isn't enabled. Why not use gcc's -finstrument-functions capability? gcc's facility doesn't allow access to local data and doesn't operate on non-function scopes. Known Issues: The only know issue is that we currently don't know the line number where the scope exited. It's reported as the same place the scope was entered. There's probably a way to get around it but it might involve looking at the stack and doing an 'addr2line' to get the line number. Kind of like ast_backtrace() does. Not sure if it's worth it. Change-Id: Ic5ebb859883f9c10a08c5630802de33500cad027
2020-05-14 18:24:19 +00:00
ast_cli(a->fd, " Trace level: %d\n", option_trace);
ast_cli(a->fd, " Dump core on crash: %s\n", ast_opt_dump_core ? "Yes" : "No");
print_file(a->fd, " Core dump file: ", "/proc/sys/kernel/core_pattern");
ast_cli(a->fd, " Maximum load average: %lf\n", ast_option_maxload);
#if defined(HAVE_SYSINFO)
ast_cli(a->fd, " Minimum free memory: %ld MB\n", option_minmemfree);
#endif
if (ast_localtime(&ast_startuptime, &tm, NULL)) {
ast_strftime(buf, sizeof(buf), "%H:%M:%S", &tm);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
ast_cli(a->fd, " Startup time: %s\n", buf);
}
if (ast_localtime(&ast_lastreloadtime, &tm, NULL)) {
ast_strftime(buf, sizeof(buf), "%H:%M:%S", &tm);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
ast_cli(a->fd, " Last reload time: %s\n", buf);
}
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
ast_cli(a->fd, " System: %s/%s built by %s on %s %s\n", ast_build_os, ast_build_kernel, ast_build_user, ast_build_machine, ast_build_date);
ast_cli(a->fd, " System name: %s\n", ast_config_AST_SYSTEM_NAME);
ast_cli(a->fd, " Entity ID: %s\n", eid_str);
ast_cli(a->fd, " PBX UUID: %s\n", pbx_uuid);
ast_cli(a->fd, " Default language: %s\n", ast_defaultlanguage);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
ast_cli(a->fd, " Language prefix: %s\n", ast_language_is_prefix ? "Enabled" : "Disabled");
ast_cli(a->fd, " User name and group: %s/%s\n", ast_config_AST_RUN_USER, ast_config_AST_RUN_GROUP);
#if defined(HAVE_EACCESS) || defined(HAVE_EUIDACCESS)
#if defined(HAVE_EUIDACCESS) && !defined(HAVE_EACCESS)
#define eaccess euidaccess
#endif
if (!getcwd(dir, sizeof(dir))) {
if (eaccess(dir, R_OK | X_OK | F_OK)) {
ast_cli(a->fd, " Running directory: %s\n", "Unable to access");
} else {
ast_cli(a->fd, " Running directory: %s (%s)\n", dir, "Unable to access");
}
} else {
ast_cli(a->fd, " Running directory: %s\n", dir);
}
#endif /* defined(HAVE_EACCESS) || defined(HAVE_EUIDACCESS) */
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
ast_cli(a->fd, " Executable includes: %s\n", ast_test_flag(&ast_options, AST_OPT_FLAG_EXEC_INCLUDES) ? "Enabled" : "Disabled");
ast_cli(a->fd, " Transcode via SLIN: %s\n", ast_test_flag(&ast_options, AST_OPT_FLAG_TRANSCODE_VIA_SLIN) ? "Enabled" : "Disabled");
ast_cli(a->fd, " Transmit silence during rec: %s\n", ast_test_flag(&ast_options, AST_OPT_FLAG_TRANSMIT_SILENCE) ? "Enabled" : "Disabled");
Fix transcode_via_sln option with SIP calls and improve PLC usage. From reviewboard: The problem here is a bit complex, so try to bear with me... It was noticed by a Digium customer that generic PLC (as configured in codecs.conf) did not appear to actually be having any sort of benefit when packet loss was introduced on an RTP stream. I reproduced this issue myself by streaming a file across an RTP stream and dropping approx. 5% of the RTP packets. I saw no real difference between when PLC was enabled or disabled when using wireshark to analyze the RTP streams. After analyzing what was going on, it became clear that one of the problems faced was that when running my tests, the translation paths were being set up in such a way that PLC could not possibly work as expected. To illustrate, if packets are lost on channel A's read stream, then we expect that PLC will be applied to channel B's write stream. The problem is that generic PLC can only be done when there is a translation path that moves from some codec to SLINEAR. When I would run my tests, I found that every single time, read and write translation paths would be set up on channel A instead of channel B. There appeared to be no real way to predict which channel the translation paths would be set up on. This is where Kevin swooped in to let me know about the transcode_via_sln option in asterisk.conf. It is supposed to work by placing a read translation path on both channels from the channel's rawreadformat to SLINEAR. It also will place a write translation path on both channels from SLINEAR to the channel's rawwriteformat. Using this option allows one to predictably set up translation paths on all channels. There are two problems with this, though. First and foremost, the transcode_via_sln option did not appear to be working properly when I was placing a SIP call between two endpoints which did not share any common formats. Second, even if this option were to work, for PLC to be applied, there had to be a write translation path that would go from some format to SLINEAR. It would not work properly if the starting format of translation was SLINEAR. The one-line change presented in this review request in chan_sip.c fixed the first issue for me. The problem was that in sip_request_call, the jointcapability of the outbound channel was being set to the format passed to sip_request_call. This is nativeformats of the inbound channel. Because of this, when ast_channel_make_compatible was called by app_dial, both channels already had compatibly read and write formats. Thus, no translation path was set up at the time. My change is to set the jointcapability of the sip_pvt created during sip_request_call to the intersection of the inbound channel's nativeformats and the configured peer capability that we determined during the earlier call to create_addr. Doing this got the translation paths set up as expected when using transcode_via_sln. The changes presented in channel.c fixed the second issue for me. First and foremost, when Asterisk is started, we'll read codecs.conf to see the value of the genericplc option. If this option is set, and ast_write is called for a frame with no data, then we will attempt to fill in the missing samples for the frame. The implementation uses a channel datastore for maintaining the PLC state and for creating a buffer to store PLC samples in. Even when we receive a frame with data, we'll call plc_rx so that the PLC state will have knowledge of the previous voice frame, which it can use as a basis for when it comes time to actually do a PLC fill-in. So, reviewers, now I ask for your help. First off, there's the one line change in chan_sip that I have put in. Is it right? By my logic it seems correct, but I'm sure someone can tell me why it is not going to work. This is probably the change I'm least concerned about, though. What concerns me much more is the set of changes in channel.c. First off, am I even doing it right? When I run tests, I can clearly see that when PLC is activated, I see a significant increase in RTP traffic where I would expect it to be. However, in my humble opinion, the audio sounds kind of crappy whenever the PLC fill-in is done. It sounds worse to me than when no PLC is used at all. I need someone to review the logic I have used to be sure that I'm not misusing anything. As far as I can see my pointer arithmetic is correct, and my use of AST_FRIENDLY_OFFSET should be correct as well, but I'm sure someone can point out somewhere where I've done something incorrectly. As I was writing this review request up, I decided to give the code a test run under valgrind, and I find that for some reason, calls to plc_rx are causing some invalid reads. Apparently I'm reading past the end of a buffer somehow. I'll have to dig around a bit to see why that is the case. If it's obvious to someone reviewing, speak up! Finally, I have one other proposal that is not reflected in my code review. Since without transcode_via_sln set, one cannot predict or control where a translation path will be up, it seems to me that the current practice of using PLC only when transcoding to SLINEAR is not useful. I recommend that once it has been determined that the method used in this code review is correct and works as expected, then the code in translate.c that invokes PLC should be removed. Review: https://reviewboard.asterisk.org/r/622/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264452 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-05-19 21:29:08 +00:00
ast_cli(a->fd, " Generic PLC: %s\n", ast_test_flag(&ast_options, AST_OPT_FLAG_GENERIC_PLC) ? "Enabled" : "Disabled");
ast_cli(a->fd, " Generic PLC on equal codecs: %s\n", ast_test_flag(&ast_options, AST_OPT_FLAG_GENERIC_PLC_ON_EQUAL_CODECS) ? "Enabled" : "Disabled");
ast_cli(a->fd, " Hide Msg Chan AMI events: %s\n", ast_opt_hide_messaging_ami_events ? "Enabled" : "Disabled");
ast_cli(a->fd, " Sounds search custom dir: %s\n", ast_opt_sounds_search_custom ? "Enabled" : "Disabled");
ast_cli(a->fd, " Min DTMF duration:: %u\n", option_dtmfminduration);
#if !defined(LOW_MEMORY)
ast_cli(a->fd, " Cache media frames: %s\n", ast_opt_cache_media_frames ? "Enabled" : "Disabled");
#endif
ast_cli(a->fd, " RTP use dynamic payloads: %u\n", ast_option_rtpusedynamic);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
if (ast_option_rtpptdynamic == AST_RTP_PT_LAST_REASSIGN) {
ast_cli(a->fd, " RTP dynamic payload types: %u,%u-%u\n",
ast_option_rtpptdynamic,
AST_RTP_PT_FIRST_DYNAMIC, AST_RTP_MAX_PT - 1);
} else if (ast_option_rtpptdynamic < AST_RTP_PT_LAST_REASSIGN) {
ast_cli(a->fd, " RTP dynamic payload types: %u-%u,%u-%u\n",
ast_option_rtpptdynamic, AST_RTP_PT_LAST_REASSIGN,
AST_RTP_PT_FIRST_DYNAMIC, AST_RTP_MAX_PT - 1);
} else {
ast_cli(a->fd, " RTP dynamic payload types: %u-%u\n",
AST_RTP_PT_FIRST_DYNAMIC, AST_RTP_MAX_PT - 1);
}
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
ast_cli(a->fd, "\n* Subsystems\n");
ast_cli(a->fd, " -------------\n");
ast_cli(a->fd, " Manager (AMI): %s\n", ast_manager_check_enabled() ? "Enabled" : "Disabled");
ast_cli(a->fd, " Web Manager (AMI/HTTP): %s\n", ast_webmanager_check_enabled() ? "Enabled" : "Disabled");
Update Asterisk's CDRs for the new bridging framework This patch is the initial push to update Asterisk's CDR engine for the new bridging framework. This patch guts the existing CDR engine and builds the new on top of messages coming across Stasis. As changes in channel state and bridge state are detected, CDRs are built and dispatched accordingly. This fundamentally changes CDRs in a few ways. (1) CDRs are now *very* reflective of the actual state of channels and bridges. This means CDRs track well with what an actual channel is doing - which is useful in transfer scenarios (which were previously difficult to pin down). It does, however, mean that CDRs cannot be 'fooled'. Previous behavior in Asterisk allowed for CDR applications, channels, and other properties to be spoofed in parts of the code - this no longer works. (2) CDRs have defined behavior in multi-party scenarios. This behavior will not be what everyone wants, but it is a defined behavior and as such, it is predictable. (3) The CDR manipulation functions and applications have been overhauled. Major changes have been made to ResetCDR and ForkCDR in particular. Many of the options for these two applications no longer made any sense with the new framework and the (slightly) more immutable nature of CDRs. There are a plethora of other changes. For a full description of CDR behavior, see the CDR specification on the Asterisk wiki. (closes issue ASTERISK-21196) Review: https://reviewboard.asterisk.org/r/2486/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-06-17 03:00:38 +00:00
ast_cli(a->fd, " Call data records: %s\n", ast_cdr_is_enabled() ? "Enabled" : "Disabled");
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
ast_cli(a->fd, " Realtime Architecture (ARA): %s\n", ast_realtime_enabled() ? "Enabled" : "Disabled");
/*! \todo we could check musiconhold, voicemail, smdi, adsi, queues */
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
ast_cli(a->fd, "\n* Directories\n");
ast_cli(a->fd, " -------------\n");
ast_cli(a->fd, " Configuration file: %s\n", ast_config_AST_CONFIG_FILE);
ast_cli(a->fd, " Configuration directory: %s\n", ast_config_AST_CONFIG_DIR);
ast_cli(a->fd, " Module directory: %s\n", ast_config_AST_MODULE_DIR);
ast_cli(a->fd, " Spool directory: %s\n", ast_config_AST_SPOOL_DIR);
ast_cli(a->fd, " Log directory: %s\n", ast_config_AST_LOG_DIR);
ast_cli(a->fd, " Run/Sockets directory: %s\n", ast_config_AST_RUN_DIR);
ast_cli(a->fd, " PID file: %s\n", ast_config_AST_PID);
ast_cli(a->fd, " VarLib directory: %s\n", ast_config_AST_VAR_DIR);
ast_cli(a->fd, " Data directory: %s\n", ast_config_AST_DATA_DIR);
ast_cli(a->fd, " ASTDB: %s\n", ast_config_AST_DB);
ast_cli(a->fd, " IAX2 Keys directory: %s\n", ast_config_AST_KEY_DIR);
ast_cli(a->fd, " AGI Scripts directory: %s\n", ast_config_AST_AGI_DIR);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
ast_cli(a->fd, "\n\n");
return CLI_SUCCESS;
}
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
static char *handle_show_threads(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
int count = 0;
struct thread_list_t *cur;
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
switch (cmd) {
case CLI_INIT:
e->command = "core show threads";
e->usage =
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
"Usage: core show threads\n"
" List threads currently active in the system.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
AST_RWLIST_RDLOCK(&thread_list);
AST_RWLIST_TRAVERSE(&thread_list, cur, list) {
ast_cli(a->fd, "%p %d %s\n", (void *)cur->id, cur->lwp, cur->name);
count++;
}
AST_RWLIST_UNLOCK(&thread_list);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
ast_cli(a->fd, "%d threads listed.\n", count);
return CLI_SUCCESS;
}
#if defined (HAVE_SYSCTL) && defined(HAVE_SWAPCTL)
/*
* swapmode is rewritten by Tobias Weingartner <weingart@openbsd.org>
* to be based on the new swapctl(2) system call.
*/
static int swapmode(int *used, int *total)
{
struct swapent *swdev;
int nswap, rnswap, i;
nswap = swapctl(SWAP_NSWAP, 0, 0);
if (nswap == 0)
return 0;
swdev = ast_calloc(nswap, sizeof(*swdev));
if (swdev == NULL)
return 0;
rnswap = swapctl(SWAP_STATS, swdev, nswap);
if (rnswap == -1) {
ast_free(swdev);
return 0;
}
/* if rnswap != nswap, then what? */
/* Total things up */
*total = *used = 0;
for (i = 0; i < nswap; i++) {
if (swdev[i].se_flags & SWF_ENABLE) {
*used += (swdev[i].se_inuse / (1024 / DEV_BSIZE));
*total += (swdev[i].se_nblks / (1024 / DEV_BSIZE));
}
}
ast_free(swdev);
return 1;
}
#endif
#if defined(HAVE_SYSINFO) || defined(HAVE_SYSCTL)
/*! \brief Give an overview of system statistics */
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
static char *handle_show_sysinfo(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
uint64_t physmem, freeram;
#if defined(HAVE_SYSINFO) || defined(HAVE_SWAPCTL)
int totalswap = 0;
uint64_t freeswap = 0;
#endif
int nprocs = 0;
long uptime = 0;
#if defined(HAVE_SYSINFO)
struct sysinfo sys_info;
#elif defined(HAVE_SYSCTL)
static int pageshift;
struct vmtotal vmtotal;
struct timeval boottime;
time_t now;
int mib[2], pagesize;
#if defined(HAVE_SWAPCTL)
int usedswap = 0;
#endif
size_t len;
#endif
switch (cmd) {
case CLI_INIT:
e->command = "core show sysinfo";
e->usage =
"Usage: core show sysinfo\n"
" List current system information.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
#if defined(HAVE_SYSINFO)
sysinfo(&sys_info);
uptime = sys_info.uptime / 3600;
physmem = sys_info.totalram * sys_info.mem_unit;
freeram = (sys_info.freeram * sys_info.mem_unit) / 1024;
totalswap = (sys_info.totalswap * sys_info.mem_unit) / 1024;
freeswap = (sys_info.freeswap * sys_info.mem_unit) / 1024;
nprocs = sys_info.procs;
#elif defined(HAVE_SYSCTL)
/* calculate the uptime by looking at boottime */
time(&now);
mib[0] = CTL_KERN;
mib[1] = KERN_BOOTTIME;
len = sizeof(boottime);
if (sysctl(mib, 2, &boottime, &len, NULL, 0) != -1) {
uptime = now - boottime.tv_sec;
}
uptime = uptime/3600;
/* grab total physical memory */
mib[0] = CTL_HW;
#if defined(HW_PHYSMEM64)
mib[1] = HW_PHYSMEM64;
#else
mib[1] = HW_PHYSMEM;
#endif
len = sizeof(physmem);
sysctl(mib, 2, &physmem, &len, NULL, 0);
pagesize = getpagesize();
pageshift = 0;
while (pagesize > 1) {
pageshift++;
pagesize >>= 1;
}
/* we only need the amount of log(2)1024 for our conversion */
pageshift -= 10;
/* grab vm totals */
mib[0] = CTL_VM;
mib[1] = VM_METER;
len = sizeof(vmtotal);
sysctl(mib, 2, &vmtotal, &len, NULL, 0);
freeram = (vmtotal.t_free << pageshift);
/* generate swap usage and totals */
#if defined(HAVE_SWAPCTL)
swapmode(&usedswap, &totalswap);
freeswap = (totalswap - usedswap);
#endif
/* grab number of processes */
#if defined(__OpenBSD__)
mib[0] = CTL_KERN;
mib[1] = KERN_NPROCS;
len = sizeof(nprocs);
sysctl(mib, 2, &nprocs, &len, NULL, 0);
#endif
#endif
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
ast_cli(a->fd, "\nSystem Statistics\n");
ast_cli(a->fd, "-----------------\n");
ast_cli(a->fd, " System Uptime: %ld hours\n", uptime);
ast_cli(a->fd, " Total RAM: %" PRIu64 " KiB\n", physmem / 1024);
ast_cli(a->fd, " Free RAM: %" PRIu64 " KiB\n", freeram);
#if defined(HAVE_SYSINFO)
ast_cli(a->fd, " Buffer RAM: %" PRIu64 " KiB\n", ((uint64_t) sys_info.bufferram * sys_info.mem_unit) / 1024);
#endif
#if defined(HAVE_SYSINFO) || defined(HAVE_SWAPCTL)
ast_cli(a->fd, " Total Swap Space: %d KiB\n", totalswap);
ast_cli(a->fd, " Free Swap Space: %" PRIu64 " KiB\n\n", freeswap);
#endif
ast_cli(a->fd, " Number of Processes: %d \n\n", nprocs);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return CLI_SUCCESS;
}
#endif
struct profile_entry {
const char *name;
uint64_t scale; /* if non-zero, values are scaled by this */
int64_t mark;
int64_t value;
int64_t events;
};
struct profile_data {
int entries;
int max_size;
struct profile_entry e[0];
};
static struct profile_data *prof_data;
#endif /* ! LOW_MEMORY */
/*! \brief allocates a counter with a given name and scale.
* \return Returns the identifier of the counter.
*/
int ast_add_profile(const char *name, uint64_t scale)
{
#if !defined(LOW_MEMORY)
int l = sizeof(struct profile_data);
int n = 10; /* default entries */
if (prof_data == NULL) {
prof_data = ast_calloc(1, l + n*sizeof(struct profile_entry));
if (prof_data == NULL)
return -1;
prof_data->entries = 0;
prof_data->max_size = n;
}
if (prof_data->entries >= prof_data->max_size) {
void *p;
n = prof_data->max_size + 20;
p = ast_realloc(prof_data, l + n*sizeof(struct profile_entry));
if (p == NULL)
return -1;
prof_data = p;
prof_data->max_size = n;
}
n = prof_data->entries++;
prof_data->e[n].name = ast_strdup(name);
prof_data->e[n].value = 0;
prof_data->e[n].events = 0;
prof_data->e[n].mark = 0;
prof_data->e[n].scale = scale;
return n;
#else /* if defined(LOW_MEMORY) */
return 0;
#endif
}
int64_t ast_profile(int i, int64_t delta)
{
#if !defined(LOW_MEMORY)
if (!prof_data || i < 0 || i > prof_data->entries) /* invalid index */
return 0;
if (prof_data->e[i].scale > 1)
delta /= prof_data->e[i].scale;
prof_data->e[i].value += delta;
prof_data->e[i].events++;
return prof_data->e[i].value;
#else /* if defined(LOW_MEMORY) */
return 0;
#endif
}
#if !defined(LOW_MEMORY)
/* The RDTSC instruction was introduced on the Pentium processor and is not
* implemented on certain clones, like the Cyrix 586. Hence, the previous
* expectation of __i386__ was in error. */
#if defined ( __i686__) && (defined(__FreeBSD__) || defined(__NetBSD__) || defined(linux))
#if defined(__FreeBSD__)
#include <machine/cpufunc.h>
#elif defined(__NetBSD__) || defined(linux)
static __inline uint64_t
rdtsc(void)
{
uint64_t rv;
__asm __volatile(".byte 0x0f, 0x31" : "=A" (rv));
return (rv);
}
#endif
#else /* supply a dummy function on other platforms */
static __inline uint64_t
rdtsc(void)
{
return 0;
}
#endif
#endif /* ! LOW_MEMORY */
int64_t ast_mark(int i, int startstop)
{
#if !defined(LOW_MEMORY)
if (!prof_data || i < 0 || i > prof_data->entries) /* invalid index */
return 0;
if (startstop == 1)
prof_data->e[i].mark = rdtsc();
else {
prof_data->e[i].mark = (rdtsc() - prof_data->e[i].mark);
if (prof_data->e[i].scale > 1)
prof_data->e[i].mark /= prof_data->e[i].scale;
prof_data->e[i].value += prof_data->e[i].mark;
prof_data->e[i].events++;
}
return prof_data->e[i].mark;
#else /* if defined(LOW_MEMORY) */
return 0;
#endif
}
#if !defined(LOW_MEMORY)
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
#define DEFINE_PROFILE_MIN_MAX_VALUES min = 0; \
max = prof_data->entries;\
if (a->argc > 3) { /* specific entries */ \
if (isdigit(a->argv[3][0])) { \
min = atoi(a->argv[3]); \
if (a->argc == 5 && strcmp(a->argv[4], "-")) \
max = atoi(a->argv[4]); \
} else \
search = a->argv[3]; \
} \
if (max > prof_data->entries) \
max = prof_data->entries;
static char *handle_show_profile(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
int i, min, max;
const char *search = NULL;
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
switch (cmd) {
case CLI_INIT:
e->command = "core show profile";
e->usage = "Usage: core show profile\n"
" show profile information";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (prof_data == NULL)
return 0;
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
DEFINE_PROFILE_MIN_MAX_VALUES;
ast_cli(a->fd, "profile values (%d, allocated %d)\n-------------------\n",
prof_data->entries, prof_data->max_size);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
ast_cli(a->fd, "%6s %8s %10s %12s %12s %s\n", "ID", "Scale", "Events",
"Value", "Average", "Name");
for (i = min; i < max; i++) {
struct profile_entry *entry = &prof_data->e[i];
if (!search || strstr(entry->name, search))
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
ast_cli(a->fd, "%6d: [%8ld] %10ld %12lld %12lld %s\n",
i,
(long)entry->scale,
(long)entry->events, (long long)entry->value,
(long long)(entry->events ? entry->value / entry->events : entry->value),
entry->name);
}
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return CLI_SUCCESS;
}
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
static char *handle_clear_profile(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
int i, min, max;
const char *search = NULL;
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
switch (cmd) {
case CLI_INIT:
e->command = "core clear profile";
e->usage = "Usage: core clear profile\n"
" clear profile information";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (prof_data == NULL)
return 0;
DEFINE_PROFILE_MIN_MAX_VALUES;
for (i= min; i < max; i++) {
if (!search || strstr(prof_data->e[i].name, search)) {
prof_data->e[i].value = 0;
prof_data->e[i].events = 0;
}
}
return CLI_SUCCESS;
}
#undef DEFINE_PROFILE_MIN_MAX_VALUES
#endif /* ! LOW_MEMORY */
int ast_pbx_uuid_get(char *pbx_uuid, int length)
{
return ast_db_get("pbx", "UUID", pbx_uuid, length);
}
static void publish_fully_booted(void)
{
struct ast_json *json_object;
int uptime = 0;
int lastreloaded = 0;
struct timeval tmp;
struct timeval curtime = ast_tvnow();
if (ast_startuptime.tv_sec) {
tmp = ast_tvsub(curtime, ast_startuptime);
uptime = (int) tmp.tv_sec;
}
if (ast_lastreloadtime.tv_sec) {
tmp = ast_tvsub(curtime, ast_lastreloadtime);
lastreloaded = (int) tmp.tv_sec;
}
json_object = ast_json_pack("{s: s, s: i, s: i}",
"Status", "Fully Booted",
"Uptime", uptime,
"LastReload", lastreloaded);
ast_manager_publish_event("FullyBooted", EVENT_FLAG_SYSTEM, json_object);
ast_json_unref(json_object);
}
Avoid unnecessary cleanups during immediate shutdown This patch addresses issues during immediate shutdowns, where modules are not unloaded, but Asterisk atexit handlers are run. In the typical case, this usually isn't a big deal. But the introduction of the Stasis message bus makes it much more likely for asynchronous activity to be happening off in some thread during shutdown. During an immediate shutdown, Asterisk skips unloading modules. But while it is processing the atexit handlers, there is a window of time where some of the core message types have been cleaned up, but the message bus is still running. Specifically, it's still running module subscriptions that might be using the core message types. If a message is received by that subscription in that window, it will attempt to use a message type that has been cleaned up. To solve this problem, this patch introduces ast_register_cleanup(). This function operates identically to ast_register_atexit(), except that cleanup calls are not invoked on an immediate shutdown. All of the core message type and topic cleanup was moved from atexit handlers to cleanup handlers. This ensures that core type and topic cleanup only happens if the modules that used them are first unloaded. This patch also changes the ast_assert() when accessing a cleaned up or uninitialized message type to an error log message. Message type functions are actually NULL safe across the board, so the assert was a bit heavy handed. Especially for anyone with DO_CRASH enabled. Review: https://reviewboard.asterisk.org/r/2562/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-30 17:05:53 +00:00
static void ast_run_atexits(int run_cleanups)
{
struct ast_atexit *ae;
AST_LIST_LOCK(&atexits);
while ((ae = AST_LIST_REMOVE_HEAD(&atexits, list))) {
Avoid unnecessary cleanups during immediate shutdown This patch addresses issues during immediate shutdowns, where modules are not unloaded, but Asterisk atexit handlers are run. In the typical case, this usually isn't a big deal. But the introduction of the Stasis message bus makes it much more likely for asynchronous activity to be happening off in some thread during shutdown. During an immediate shutdown, Asterisk skips unloading modules. But while it is processing the atexit handlers, there is a window of time where some of the core message types have been cleaned up, but the message bus is still running. Specifically, it's still running module subscriptions that might be using the core message types. If a message is received by that subscription in that window, it will attempt to use a message type that has been cleaned up. To solve this problem, this patch introduces ast_register_cleanup(). This function operates identically to ast_register_atexit(), except that cleanup calls are not invoked on an immediate shutdown. All of the core message type and topic cleanup was moved from atexit handlers to cleanup handlers. This ensures that core type and topic cleanup only happens if the modules that used them are first unloaded. This patch also changes the ast_assert() when accessing a cleaned up or uninitialized message type to an error log message. Message type functions are actually NULL safe across the board, so the assert was a bit heavy handed. Especially for anyone with DO_CRASH enabled. Review: https://reviewboard.asterisk.org/r/2562/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-30 17:05:53 +00:00
if (ae->func && (!ae->is_cleanup || run_cleanups)) {
ae->func();
}
ast_free(ae);
}
AST_LIST_UNLOCK(&atexits);
}
static void __ast_unregister_atexit(void (*func)(void))
{
struct ast_atexit *ae;
AST_LIST_TRAVERSE_SAFE_BEGIN(&atexits, ae, list) {
if (ae->func == func) {
AST_LIST_REMOVE_CURRENT(list);
ast_free(ae);
break;
}
}
AST_LIST_TRAVERSE_SAFE_END;
}
Avoid unnecessary cleanups during immediate shutdown This patch addresses issues during immediate shutdowns, where modules are not unloaded, but Asterisk atexit handlers are run. In the typical case, this usually isn't a big deal. But the introduction of the Stasis message bus makes it much more likely for asynchronous activity to be happening off in some thread during shutdown. During an immediate shutdown, Asterisk skips unloading modules. But while it is processing the atexit handlers, there is a window of time where some of the core message types have been cleaned up, but the message bus is still running. Specifically, it's still running module subscriptions that might be using the core message types. If a message is received by that subscription in that window, it will attempt to use a message type that has been cleaned up. To solve this problem, this patch introduces ast_register_cleanup(). This function operates identically to ast_register_atexit(), except that cleanup calls are not invoked on an immediate shutdown. All of the core message type and topic cleanup was moved from atexit handlers to cleanup handlers. This ensures that core type and topic cleanup only happens if the modules that used them are first unloaded. This patch also changes the ast_assert() when accessing a cleaned up or uninitialized message type to an error log message. Message type functions are actually NULL safe across the board, so the assert was a bit heavy handed. Especially for anyone with DO_CRASH enabled. Review: https://reviewboard.asterisk.org/r/2562/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-30 17:05:53 +00:00
static int register_atexit(void (*func)(void), int is_cleanup)
{
struct ast_atexit *ae;
ae = ast_calloc(1, sizeof(*ae));
if (!ae) {
return -1;
}
ae->func = func;
Avoid unnecessary cleanups during immediate shutdown This patch addresses issues during immediate shutdowns, where modules are not unloaded, but Asterisk atexit handlers are run. In the typical case, this usually isn't a big deal. But the introduction of the Stasis message bus makes it much more likely for asynchronous activity to be happening off in some thread during shutdown. During an immediate shutdown, Asterisk skips unloading modules. But while it is processing the atexit handlers, there is a window of time where some of the core message types have been cleaned up, but the message bus is still running. Specifically, it's still running module subscriptions that might be using the core message types. If a message is received by that subscription in that window, it will attempt to use a message type that has been cleaned up. To solve this problem, this patch introduces ast_register_cleanup(). This function operates identically to ast_register_atexit(), except that cleanup calls are not invoked on an immediate shutdown. All of the core message type and topic cleanup was moved from atexit handlers to cleanup handlers. This ensures that core type and topic cleanup only happens if the modules that used them are first unloaded. This patch also changes the ast_assert() when accessing a cleaned up or uninitialized message type to an error log message. Message type functions are actually NULL safe across the board, so the assert was a bit heavy handed. Especially for anyone with DO_CRASH enabled. Review: https://reviewboard.asterisk.org/r/2562/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-30 17:05:53 +00:00
ae->is_cleanup = is_cleanup;
AST_LIST_LOCK(&atexits);
__ast_unregister_atexit(func);
AST_LIST_INSERT_HEAD(&atexits, ae, list);
AST_LIST_UNLOCK(&atexits);
return 0;
}
Avoid unnecessary cleanups during immediate shutdown This patch addresses issues during immediate shutdowns, where modules are not unloaded, but Asterisk atexit handlers are run. In the typical case, this usually isn't a big deal. But the introduction of the Stasis message bus makes it much more likely for asynchronous activity to be happening off in some thread during shutdown. During an immediate shutdown, Asterisk skips unloading modules. But while it is processing the atexit handlers, there is a window of time where some of the core message types have been cleaned up, but the message bus is still running. Specifically, it's still running module subscriptions that might be using the core message types. If a message is received by that subscription in that window, it will attempt to use a message type that has been cleaned up. To solve this problem, this patch introduces ast_register_cleanup(). This function operates identically to ast_register_atexit(), except that cleanup calls are not invoked on an immediate shutdown. All of the core message type and topic cleanup was moved from atexit handlers to cleanup handlers. This ensures that core type and topic cleanup only happens if the modules that used them are first unloaded. This patch also changes the ast_assert() when accessing a cleaned up or uninitialized message type to an error log message. Message type functions are actually NULL safe across the board, so the assert was a bit heavy handed. Especially for anyone with DO_CRASH enabled. Review: https://reviewboard.asterisk.org/r/2562/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-30 17:05:53 +00:00
int ast_register_atexit(void (*func)(void))
{
return register_atexit(func, 0);
}
int ast_register_cleanup(void (*func)(void))
{
return register_atexit(func, 1);
}
void ast_unregister_atexit(void (*func)(void))
{
AST_LIST_LOCK(&atexits);
__ast_unregister_atexit(func);
AST_LIST_UNLOCK(&atexits);
}
/* Sending commands from consoles back to the daemon requires a terminating NULL */
static int fdsend(int fd, const char *s)
{
return write(fd, s, strlen(s) + 1);
}
/* Sending messages from the daemon back to the display requires _excluding_ the terminating NULL */
static int fdprint(int fd, const char *s)
{
return write(fd, s, strlen(s));
}
/*! \brief NULL handler so we can collect the child exit status */
static void _null_sig_handler(int sig)
{
}
static struct sigaction null_sig_handler = {
.sa_handler = _null_sig_handler,
.sa_flags = SA_RESTART,
};
static struct sigaction ignore_sig_handler = {
.sa_handler = SIG_IGN,
};
AST_MUTEX_DEFINE_STATIC(safe_system_lock);
/*! \brief Keep track of how many threads are currently trying to wait*() on
* a child process
*/
static unsigned int safe_system_level = 0;
static struct sigaction safe_system_prev_handler;
void ast_replace_sigchld(void)
{
unsigned int level;
ast_mutex_lock(&safe_system_lock);
level = safe_system_level++;
/* only replace the handler if it has not already been done */
if (level == 0) {
sigaction(SIGCHLD, &null_sig_handler, &safe_system_prev_handler);
}
ast_mutex_unlock(&safe_system_lock);
}
void ast_unreplace_sigchld(void)
{
unsigned int level;
ast_mutex_lock(&safe_system_lock);
level = --safe_system_level;
/* only restore the handler if we are the last one */
if (level == 0) {
sigaction(SIGCHLD, &safe_system_prev_handler, NULL);
}
ast_mutex_unlock(&safe_system_lock);
}
/*! \brief fork and perform other preparations for spawning applications */
static pid_t safe_exec_prep(int dualfork)
{
pid_t pid;
#if defined(HAVE_WORKING_FORK) || defined(HAVE_WORKING_VFORK)
ast_replace_sigchld();
#ifdef HAVE_WORKING_FORK
pid = fork();
#else
pid = vfork();
#endif
if (pid == 0) {
#ifdef HAVE_CAP
cap_set_proc(child_cap);
#endif
#ifdef HAVE_WORKING_FORK
if (ast_opt_high_priority) {
ast_set_priority(0);
}
/* Close file descriptors and launch system command */
ast_close_fds_above_n(STDERR_FILENO);
#endif
if (dualfork) {
#ifdef HAVE_WORKING_FORK
pid = fork();
#else
pid = vfork();
#endif
if (pid < 0) {
/* Second fork failed. */
/* No logger available. */
_exit(1);
}
if (pid > 0) {
/* This is the first fork, exit so the reaper finishes right away. */
_exit(0);
}
/* This is the second fork. The first fork will exit immediately so
* Asterisk doesn't have to wait for completion.
* ast_safe_system("cmd &") would run in the background, but the '&'
* cannot be added with ast_safe_execvp, so we have to double fork.
*/
}
}
if (pid < 0) {
ast_log(LOG_WARNING, "Fork failed: %s\n", strerror(errno));
}
#else
ast_log(LOG_WARNING, "Fork failed: %s\n", strerror(ENOTSUP));
pid = -1;
#endif
return pid;
}
/*! \brief wait for spawned application to complete and unreplace sigchld */
static int safe_exec_wait(pid_t pid)
{
int res = -1;
#if defined(HAVE_WORKING_FORK) || defined(HAVE_WORKING_VFORK)
if (pid > 0) {
for (;;) {
int status;
res = waitpid(pid, &status, 0);
if (res > -1) {
res = WIFEXITED(status) ? WEXITSTATUS(status) : -1;
break;
}
if (errno != EINTR) {
break;
}
}
}
ast_unreplace_sigchld();
#endif
return res;
}
int ast_safe_execvp(int dualfork, const char *file, char *const argv[])
{
pid_t pid = safe_exec_prep(dualfork);
if (pid == 0) {
execvp(file, argv);
_exit(1);
/* noreturn from _exit */
}
return safe_exec_wait(pid);
}
int ast_safe_system(const char *s)
{
pid_t pid = safe_exec_prep(0);
if (pid == 0) {
execl("/bin/sh", "/bin/sh", "-c", s, (char *) NULL);
_exit(1);
/* noreturn from _exit */
}
return safe_exec_wait(pid);
}
/*!
* \brief enable or disable a logging level to a specified console
*/
void ast_console_toggle_loglevel(int fd, int level, int state)
{
int x;
Fix a variety of potential buffer overflows * chan_mobile: Fixed an overrun where the cind_state buffer (an integer array of size 16) would be overrun due to improper bounds checking. At worst, the buffer can be overrun by a total of 48 bytes (assuming 4-byte integers), which would still leave it within the allocated memory of struct hfp. This would corrupt other elements in that struct but not necessarily cause any further issues. * app_sms: The array imsg is of size 250, while the array (ud) that the data is copied into is of size 160. If the size of the inbound message is greater then 160, up to 90 bytes could be overrun in ud. This would corrupt the user data header (array udh) adjacent to ud. * chan_unistim: A number of invalid memmoves are corrected. These would move data (which may or may not be valid) into the ends of these buffers. * asterisk: ast_console_toggle_loglevel does not check that the console log level being set is less then or equal to the allowed log levels of 32. * format_pref: In ast_codec_pref_prepend, if any occurrence of the specified codec is not found, the value used to index into the array pref->order would be one greater then the maximum size of the array. * jitterbuf: If the element being placed into the jitter buffer lands in the last available slot in the jitter history buffer, the insertion sort attempts to move the last entry in the buffer into one slot past the maximum length of the buffer. Note that this occurred for both the min and max jitter history buffers. * tdd: If a read from fsk_serial returns a character that is greater then 32, an attempt to read past one of the statically defined arrays containing the values that character maps to would occur. * localtime: struct ast_time and tm are not the same size - ast_time is larger, although it contains the elements of tm within it in the same layout. Hence, when using memcpy to copy the contents of tm into ast_time, the size of tm should be used, as opposed to the size of ast_time. * extconf: this treats ast_timing's minmask array as if it had a length of 48, when it has defined the size of the array as 24. pbx.h defines minmask as having a size of 48. (issue ASTERISK-19668) Reported by: Matt Jordan ........ Merged revisions 362485 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362496 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362497 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-19 02:40:55 +00:00
if (level >= NUMLOGLEVELS) {
level = NUMLOGLEVELS - 1;
}
for (x = 0;x < AST_MAX_CONNECTS; x++) {
if (fd == consoles[x].fd) {
/*
* Since the logging occurs when levels are false, set to
* flipped iinput because this function accepts 0 as off and 1 as on
*/
consoles[x].levels[level] = state ? 0 : 1;
return;
}
}
}
/*!
* \brief mute or unmute a console from logging
*/
void ast_console_toggle_mute(int fd, int silent)
{
int x;
for (x = 0;x < AST_MAX_CONNECTS; x++) {
if (fd == consoles[x].fd) {
if (consoles[x].mute) {
consoles[x].mute = 0;
if (!silent)
ast_cli(fd, "Console is not muted anymore.\n");
} else {
consoles[x].mute = 1;
if (!silent)
ast_cli(fd, "Console is muted.\n");
}
return;
}
}
ast_cli(fd, "Couldn't find remote console.\n");
}
/*!
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
* \brief log the string to all attached network console clients
*/
logger: Support JSON logging with Verbose messages When 2d7a4a3357 was merged, it missed the fact that Verbose log messages are formatted and handled by 'verbosers'. Verbosers are registered functions that handle verbose messages only; they exist as a separate class of callbacks. This was done to handle the 'magic' that must be inserted into Verbose messages sent to remote consoles, so that the consoles can format the messages correctly, i.e., the leading tabs/characters. In reality, verbosers are a weird appendage: they're a separate class of formatters/message handlers outside of what handles all other log messages in Asterisk. After some code inspection, it became clear that simply passing a Verbose message along with its 'sublevel' importance through the normal logging mechanisms removes the need for verbosers altogether. This patch removes the verbosers, and makes the default log formatter aware that, if the log channel is a console log, it should simply insert the 'verbose magic' into the log messages itself. This allows the console handlers to interpret and format the verbose message themselves. This simplifies the code quite a lot, and should improve the performance of printing verbose messages by a reasonable factor: (1) It removes a number of memory allocations that were done on each verobse message (2) It removes the need to strip the verbose magic out of the verbose log messages before passing them to non-console log channels (3) It now performs fewer iterations over lists when handling verbose messages Since verbose messages are now handled like other log messages (for the most part), the JSON formatting of the messages works as well. ASTERISK-25425 Change-Id: I21bf23f0a1e489b5102f8a035fe8871552ce4f96
2016-05-14 12:24:07 +00:00
static void ast_network_puts_mutable(const char *string, int level, int sublevel)
{
int x;
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
for (x = 0; x < AST_MAX_CONNECTS; ++x) {
if (consoles[x].fd < 0
|| consoles[x].mute
logger: Support JSON logging with Verbose messages When 2d7a4a3357 was merged, it missed the fact that Verbose log messages are formatted and handled by 'verbosers'. Verbosers are registered functions that handle verbose messages only; they exist as a separate class of callbacks. This was done to handle the 'magic' that must be inserted into Verbose messages sent to remote consoles, so that the consoles can format the messages correctly, i.e., the leading tabs/characters. In reality, verbosers are a weird appendage: they're a separate class of formatters/message handlers outside of what handles all other log messages in Asterisk. After some code inspection, it became clear that simply passing a Verbose message along with its 'sublevel' importance through the normal logging mechanisms removes the need for verbosers altogether. This patch removes the verbosers, and makes the default log formatter aware that, if the log channel is a console log, it should simply insert the 'verbose magic' into the log messages itself. This allows the console handlers to interpret and format the verbose message themselves. This simplifies the code quite a lot, and should improve the performance of printing verbose messages by a reasonable factor: (1) It removes a number of memory allocations that were done on each verobse message (2) It removes the need to strip the verbose magic out of the verbose log messages before passing them to non-console log channels (3) It now performs fewer iterations over lists when handling verbose messages Since verbose messages are now handled like other log messages (for the most part), the JSON formatting of the messages works as well. ASTERISK-25425 Change-Id: I21bf23f0a1e489b5102f8a035fe8871552ce4f96
2016-05-14 12:24:07 +00:00
|| consoles[x].levels[level]
|| (level == __LOG_VERBOSE && consoles[x].option_verbose < sublevel)) {
continue;
}
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
fdprint(consoles[x].p[1], string);
}
}
/*!
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
* \brief log the string to the root console, and all attached
* network console clients
*/
void ast_console_puts_mutable(const char *string, int level)
logger: Support JSON logging with Verbose messages When 2d7a4a3357 was merged, it missed the fact that Verbose log messages are formatted and handled by 'verbosers'. Verbosers are registered functions that handle verbose messages only; they exist as a separate class of callbacks. This was done to handle the 'magic' that must be inserted into Verbose messages sent to remote consoles, so that the consoles can format the messages correctly, i.e., the leading tabs/characters. In reality, verbosers are a weird appendage: they're a separate class of formatters/message handlers outside of what handles all other log messages in Asterisk. After some code inspection, it became clear that simply passing a Verbose message along with its 'sublevel' importance through the normal logging mechanisms removes the need for verbosers altogether. This patch removes the verbosers, and makes the default log formatter aware that, if the log channel is a console log, it should simply insert the 'verbose magic' into the log messages itself. This allows the console handlers to interpret and format the verbose message themselves. This simplifies the code quite a lot, and should improve the performance of printing verbose messages by a reasonable factor: (1) It removes a number of memory allocations that were done on each verobse message (2) It removes the need to strip the verbose magic out of the verbose log messages before passing them to non-console log channels (3) It now performs fewer iterations over lists when handling verbose messages Since verbose messages are now handled like other log messages (for the most part), the JSON formatting of the messages works as well. ASTERISK-25425 Change-Id: I21bf23f0a1e489b5102f8a035fe8871552ce4f96
2016-05-14 12:24:07 +00:00
{
ast_console_puts_mutable_full(string, level, 0);
}
static int console_print(const char *s);
void ast_console_puts_mutable_full(const char *message, int level, int sublevel)
{
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
/* Send to the root console */
logger: Support JSON logging with Verbose messages When 2d7a4a3357 was merged, it missed the fact that Verbose log messages are formatted and handled by 'verbosers'. Verbosers are registered functions that handle verbose messages only; they exist as a separate class of callbacks. This was done to handle the 'magic' that must be inserted into Verbose messages sent to remote consoles, so that the consoles can format the messages correctly, i.e., the leading tabs/characters. In reality, verbosers are a weird appendage: they're a separate class of formatters/message handlers outside of what handles all other log messages in Asterisk. After some code inspection, it became clear that simply passing a Verbose message along with its 'sublevel' importance through the normal logging mechanisms removes the need for verbosers altogether. This patch removes the verbosers, and makes the default log formatter aware that, if the log channel is a console log, it should simply insert the 'verbose magic' into the log messages itself. This allows the console handlers to interpret and format the verbose message themselves. This simplifies the code quite a lot, and should improve the performance of printing verbose messages by a reasonable factor: (1) It removes a number of memory allocations that were done on each verobse message (2) It removes the need to strip the verbose magic out of the verbose log messages before passing them to non-console log channels (3) It now performs fewer iterations over lists when handling verbose messages Since verbose messages are now handled like other log messages (for the most part), the JSON formatting of the messages works as well. ASTERISK-25425 Change-Id: I21bf23f0a1e489b5102f8a035fe8871552ce4f96
2016-05-14 12:24:07 +00:00
console_print(message);
/* Wake up a poll()ing console */
if (ast_opt_console && consolethread != AST_PTHREADT_NULL) {
pthread_kill(consolethread, SIGURG);
}
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
/* Send to any network console clients */
logger: Support JSON logging with Verbose messages When 2d7a4a3357 was merged, it missed the fact that Verbose log messages are formatted and handled by 'verbosers'. Verbosers are registered functions that handle verbose messages only; they exist as a separate class of callbacks. This was done to handle the 'magic' that must be inserted into Verbose messages sent to remote consoles, so that the consoles can format the messages correctly, i.e., the leading tabs/characters. In reality, verbosers are a weird appendage: they're a separate class of formatters/message handlers outside of what handles all other log messages in Asterisk. After some code inspection, it became clear that simply passing a Verbose message along with its 'sublevel' importance through the normal logging mechanisms removes the need for verbosers altogether. This patch removes the verbosers, and makes the default log formatter aware that, if the log channel is a console log, it should simply insert the 'verbose magic' into the log messages itself. This allows the console handlers to interpret and format the verbose message themselves. This simplifies the code quite a lot, and should improve the performance of printing verbose messages by a reasonable factor: (1) It removes a number of memory allocations that were done on each verobse message (2) It removes the need to strip the verbose magic out of the verbose log messages before passing them to non-console log channels (3) It now performs fewer iterations over lists when handling verbose messages Since verbose messages are now handled like other log messages (for the most part), the JSON formatting of the messages works as well. ASTERISK-25425 Change-Id: I21bf23f0a1e489b5102f8a035fe8871552ce4f96
2016-05-14 12:24:07 +00:00
ast_network_puts_mutable(message, level, sublevel);
}
/*!
* \brief write the string to all attached console clients
*/
static void ast_network_puts(const char *string)
{
int x;
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
for (x = 0; x < AST_MAX_CONNECTS; ++x) {
if (consoles[x].fd < 0) {
continue;
}
fdprint(consoles[x].p[1], string);
}
}
/*!
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
* \brief write the string to the root console, and all attached
* network console clients
*/
void ast_console_puts(const char *string)
{
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
/* Send to the root console */
fputs(string, stdout);
fflush(stdout);
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
/* Send to any network console clients */
ast_network_puts(string);
}
static pthread_t lthread;
/*!
* \brief read() function supporting the reception of user credentials.
*
* \param fd Socket file descriptor.
* \param buffer Receive buffer.
* \param size 'buffer' size.
* \param con Console structure to set received credentials
* \retval -1 on error
* \retval the number of bytes received on success.
*/
static int read_credentials(int fd, char *buffer, size_t size, struct console *con)
{
#if defined(SO_PEERCRED)
#ifdef HAVE_STRUCT_SOCKPEERCRED_UID
#define HAVE_STRUCT_UCRED_UID
struct sockpeercred cred;
#else
struct ucred cred;
#endif
socklen_t len = sizeof(cred);
#endif
#if defined(HAVE_GETPEEREID)
uid_t uid;
gid_t gid;
#else
int uid, gid;
#endif
int result;
result = read(fd, buffer, size);
if (result < 0) {
return result;
}
#if defined(SO_PEERCRED) && (defined(HAVE_STRUCT_UCRED_UID) || defined(HAVE_STRUCT_UCRED_CR_UID))
if (getsockopt(fd, SOL_SOCKET, SO_PEERCRED, &cred, &len)) {
return result;
}
#if defined(HAVE_STRUCT_UCRED_UID)
uid = cred.uid;
gid = cred.gid;
#else /* defined(HAVE_STRUCT_UCRED_CR_UID) */
uid = cred.cr_uid;
gid = cred.cr_gid;
#endif /* defined(HAVE_STRUCT_UCRED_UID) */
#elif defined(HAVE_GETPEEREID)
if (getpeereid(fd, &uid, &gid)) {
return result;
}
#else
return result;
#endif
con->uid = uid;
con->gid = gid;
return result;
}
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
/* This is the thread running the remote console on the main process. */
static void *netconsole(void *vconsole)
{
struct console *con = vconsole;
char hostname[MAXHOSTNAMELEN] = "";
char inbuf[512];
char outbuf[512];
const char * const end_buf = inbuf + sizeof(inbuf);
char *start_read = inbuf;
int res;
struct pollfd fds[2];
if (gethostname(hostname, sizeof(hostname)-1))
ast_copy_string(hostname, "<Unknown>", sizeof(hostname));
snprintf(outbuf, sizeof(outbuf), "%s/%ld/%s\n", hostname, (long)ast_mainpid, ast_get_version());
fdprint(con->fd, outbuf);
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
ast_verb_console_register(&con->option_verbose);
for (;;) {
fds[0].fd = con->fd;
fds[0].events = POLLIN;
fds[0].revents = 0;
fds[1].fd = con->p[0];
fds[1].events = POLLIN;
fds[1].revents = 0;
Merged revisions 182810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-18 02:28:55 +00:00
res = ast_poll(fds, 2, -1);
if (res < 0) {
if (errno != EINTR)
ast_log(LOG_WARNING, "poll returned < 0: %s\n", strerror(errno));
continue;
}
if (fds[0].revents) {
int cmds_read, bytes_read;
if ((bytes_read = read_credentials(con->fd, start_read, end_buf - start_read, con)) < 1) {
break;
}
/* XXX This will only work if it is the first command, and I'm not sure fixing it is worth the effort. */
if (strncmp(inbuf, "cli quit after ", 15) == 0) {
ast_cli_command_multiple_full(con->uid, con->gid, con->fd, bytes_read - 15, inbuf + 15);
break;
}
/* ast_cli_command_multiple_full will only process individual commands terminated by a
* NULL and not trailing partial commands. */
if (!(cmds_read = ast_cli_command_multiple_full(con->uid, con->gid, con->fd, bytes_read + start_read - inbuf, inbuf))) {
/* No commands were read. We either have a short read on the first command
* with space left, or a command that is too long */
if (start_read + bytes_read < end_buf) {
start_read += bytes_read;
} else {
ast_log(LOG_ERROR, "Command too long! Skipping\n");
start_read = inbuf;
}
continue;
}
if (start_read[bytes_read - 1] == '\0') {
/* The read ended on a command boundary, start reading again at the head of inbuf */
start_read = inbuf;
continue;
}
/* If we get this far, we have left over characters that have not been processed.
* Advance to the character after the last command read by ast_cli_command_multiple_full.
* We are guaranteed to have at least cmds_read NULLs */
while (cmds_read-- && (start_read = strchr(start_read, '\0'))) {
start_read++;
}
memmove(inbuf, start_read, end_buf - start_read);
start_read = end_buf - start_read + inbuf;
}
if (fds[1].revents) {
res = read_credentials(con->p[0], outbuf, sizeof(outbuf), con);
if (res < 1) {
ast_log(LOG_ERROR, "read returned %d\n", res);
break;
}
res = write(con->fd, outbuf, res);
if (res < 1)
break;
}
}
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
ast_verb_console_unregister();
if (!ast_opt_hide_connect) {
ast_verb(3, "Remote UNIX connection disconnected\n");
}
close(con->fd);
close(con->p[0]);
close(con->p[1]);
con->fd = -1;
return NULL;
}
static void *listener(void *unused)
{
struct sockaddr_un sunaddr;
int s;
socklen_t len;
int x;
int poll_result;
struct pollfd fds[1];
for (;;) {
if (ast_socket < 0) {
return NULL;
}
fds[0].fd = ast_socket;
fds[0].events = POLLIN;
poll_result = ast_poll(fds, 1, -1);
pthread_testcancel();
if (poll_result < 0) {
if (errno != EINTR) {
ast_log(LOG_WARNING, "poll returned error: %s\n", strerror(errno));
}
continue;
}
len = sizeof(sunaddr);
s = accept(ast_socket, (struct sockaddr *)&sunaddr, &len);
if (s < 0) {
if (errno != EINTR)
ast_log(LOG_WARNING, "Accept returned %d: %s\n", s, strerror(errno));
} else {
#if defined(SO_PASSCRED)
int sckopt = 1;
/* turn on socket credentials passing. */
if (setsockopt(s, SOL_SOCKET, SO_PASSCRED, &sckopt, sizeof(sckopt)) < 0) {
ast_log(LOG_WARNING, "Unable to turn on socket credentials passing\n");
close(s);
} else
#endif
{
for (x = 0; x < AST_MAX_CONNECTS; x++) {
if (consoles[x].fd >= 0) {
continue;
}
if (socketpair(AF_LOCAL, SOCK_STREAM, 0, consoles[x].p)) {
ast_log(LOG_ERROR, "Unable to create pipe: %s\n", strerror(errno));
fdprint(s, "Server failed to create pipe\n");
close(s);
break;
}
ast_fd_set_flags(consoles[x].p[1], O_NONBLOCK);
consoles[x].mute = 1; /* Default is muted, we will un-mute if necessary */
/* Default uid and gid to -2, so then in cli.c/cli_has_permissions() we will be able
to know if the user didn't send the credentials. */
consoles[x].uid = -2;
consoles[x].gid = -2;
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
/* Server default of remote console verbosity level is OFF. */
consoles[x].option_verbose = 0;
consoles[x].fd = s;
if (ast_pthread_create_detached_background(&consoles[x].t, NULL, netconsole, &consoles[x])) {
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
consoles[x].fd = -1;
ast_log(LOG_ERROR, "Unable to spawn thread to handle connection: %s\n", strerror(errno));
close(consoles[x].p[0]);
close(consoles[x].p[1]);
fdprint(s, "Server failed to spawn thread\n");
close(s);
}
break;
}
if (x >= AST_MAX_CONNECTS) {
fdprint(s, "No more connections allowed\n");
ast_log(LOG_WARNING, "No more connections allowed\n");
close(s);
} else if ((consoles[x].fd > -1) && (!ast_opt_hide_connect)) {
ast_verb(3, "Remote UNIX connection\n");
}
}
}
}
return NULL;
}
static int ast_makesocket(void)
{
struct sockaddr_un sunaddr;
int res;
int x;
uid_t uid = -1;
gid_t gid = -1;
for (x = 0; x < AST_MAX_CONNECTS; x++) {
consoles[x].fd = -1;
}
if (ast_socket_is_sd) {
ast_socket = ast_sd_get_fd_un(SOCK_STREAM, ast_config_AST_SOCKET);
goto start_lthread;
}
unlink(ast_config_AST_SOCKET);
ast_socket = socket(PF_LOCAL, SOCK_STREAM, 0);
if (ast_socket < 0) {
ast_log(LOG_WARNING, "Unable to create control socket: %s\n", strerror(errno));
return -1;
}
memset(&sunaddr, 0, sizeof(sunaddr));
sunaddr.sun_family = AF_LOCAL;
ast_copy_string(sunaddr.sun_path, ast_config_AST_SOCKET, sizeof(sunaddr.sun_path));
res = bind(ast_socket, (struct sockaddr *)&sunaddr, sizeof(sunaddr));
if (res) {
ast_log(LOG_WARNING, "Unable to bind socket to %s: %s\n", ast_config_AST_SOCKET, strerror(errno));
close(ast_socket);
ast_socket = -1;
return -1;
}
res = listen(ast_socket, 2);
if (res < 0) {
ast_log(LOG_WARNING, "Unable to listen on socket %s: %s\n", ast_config_AST_SOCKET, strerror(errno));
close(ast_socket);
ast_socket = -1;
return -1;
}
start_lthread:
if (ast_pthread_create_background(&lthread, NULL, listener, NULL)) {
ast_log(LOG_WARNING, "Unable to create listener thread.\n");
close(ast_socket);
return -1;
}
if (ast_socket_is_sd) {
/* owner/group/permissions are set by systemd, we might not even have access
* to socket file so leave it alone */
return 0;
}
if (!ast_strlen_zero(ast_config_AST_CTL_OWNER)) {
struct passwd *pw;
if ((pw = getpwnam(ast_config_AST_CTL_OWNER)) == NULL)
ast_log(LOG_WARNING, "Unable to find uid of user %s\n", ast_config_AST_CTL_OWNER);
else
uid = pw->pw_uid;
}
if (!ast_strlen_zero(ast_config_AST_CTL_GROUP)) {
struct group *grp;
if ((grp = getgrnam(ast_config_AST_CTL_GROUP)) == NULL)
ast_log(LOG_WARNING, "Unable to find gid of group %s\n", ast_config_AST_CTL_GROUP);
else
gid = grp->gr_gid;
}
if (chown(ast_config_AST_SOCKET, uid, gid) < 0)
ast_log(LOG_WARNING, "Unable to change ownership of %s: %s\n", ast_config_AST_SOCKET, strerror(errno));
if (!ast_strlen_zero(ast_config_AST_CTL_PERMISSIONS)) {
unsigned int p1;
mode_t p;
sscanf(ast_config_AST_CTL_PERMISSIONS, "%30o", &p1);
p = p1;
if ((chmod(ast_config_AST_SOCKET, p)) < 0)
ast_log(LOG_WARNING, "Unable to change file permissions of %s: %s\n", ast_config_AST_SOCKET, strerror(errno));
}
return 0;
}
static int ast_tryconnect(void)
{
struct sockaddr_un sunaddr;
int res;
ast_consock = socket(PF_LOCAL, SOCK_STREAM, 0);
if (ast_consock < 0) {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "Unable to create socket: %s\n", strerror(errno));
return 0;
}
memset(&sunaddr, 0, sizeof(sunaddr));
sunaddr.sun_family = AF_LOCAL;
ast_copy_string(sunaddr.sun_path, ast_config_AST_SOCKET, sizeof(sunaddr.sun_path));
res = connect(ast_consock, (struct sockaddr *)&sunaddr, sizeof(sunaddr));
if (res) {
close(ast_consock);
ast_consock = -1;
return 0;
} else
return 1;
}
/*! \brief Urgent handler
*
* Called by soft_hangup to interrupt the poll, read, or other
* system call. We don't actually need to do anything though.
* Remember: Cannot EVER ast_log from within a signal handler
*/
static void _urg_handler(int num)
{
return;
}
static struct sigaction urg_handler = {
.sa_handler = _urg_handler,
};
static void _hup_handler(int num)
{
int save_errno = errno;
if (restartnow) {
if (el) {
el_end(el);
}
execvp(_argv[0], _argv);
}
printf("Received HUP signal -- Reloading configs\n");
sig_flags.need_reload = 1;
if (ast_alertpipe_write(sig_alert_pipe)) {
fprintf(stderr, "hup_handler: write() failed: %s\n", strerror(errno));
}
errno = save_errno;
}
static struct sigaction hup_handler = {
.sa_handler = _hup_handler,
.sa_flags = SA_RESTART,
};
static void _child_handler(int sig)
{
/* Must not ever ast_log or ast_verbose within signal handler */
int n, status, save_errno = errno;
/*
* Reap all dead children -- not just one
*/
for (n = 0; waitpid(-1, &status, WNOHANG) > 0; n++)
;
if (n == 0 && option_debug)
printf("Huh? Child handler, but nobody there?\n");
errno = save_errno;
}
static struct sigaction child_handler = {
.sa_handler = _child_handler,
.sa_flags = SA_RESTART,
};
/*! \brief Set an X-term or screen title */
static void set_title(char *text)
{
if (getenv("TERM") && strstr(getenv("TERM"), "xterm"))
fprintf(stdout, "\033]2;%s\007", text);
}
static void set_icon(char *text)
{
if (getenv("TERM") && strstr(getenv("TERM"), "xterm"))
fprintf(stdout, "\033]1;%s\007", text);
}
/*! \brief Check whether we were set to high(er) priority. */
static int has_priority(void)
{
/* Neither of these calls should fail with these arguments. */
#ifdef __linux__
/* For SCHED_OTHER, SCHED_BATCH and SCHED_IDLE, this will return
* 0. For the realtime priorities SCHED_RR and SCHED_FIFO, it
* will return something >= 1. */
return sched_getscheduler(0);
#else
/* getpriority() can return a value in -20..19 (or even -INF..20)
* where negative numbers are high priority. We don't bother
* checking errno. If the query fails and it returns -1, we'll
* assume that we're running at high prio; a safe assumption
* that will enable the resource starvation monitor (canary)
* just in case. */
return (getpriority(PRIO_PROCESS, 0) < 0);
#endif
}
/*! \brief Set priority on all known threads. */
static int set_priority_all(int pri)
{
#if !defined(__linux__)
/* The non-linux version updates the entire process prio. */
return ast_set_priority(pri);
#elif defined(LOW_MEMORY)
ast_log(LOG_WARNING, "Unable to enumerate all threads to update priority\n");
return ast_set_priority(pri);
#else
struct thread_list_t *cur;
struct sched_param sched;
char const *policy_str;
int policy;
memset(&sched, 0, sizeof(sched));
if (pri) {
policy = SCHED_RR;
policy_str = "realtime";
sched.sched_priority = 10;
} else {
policy = SCHED_OTHER;
policy_str = "regular";
sched.sched_priority = 0;
}
if (sched_setscheduler(getpid(), policy, &sched)) {
ast_log(LOG_WARNING, "Unable to set %s thread priority on main thread\n", policy_str);
return -1;
}
ast_verb(1, "Setting %s thread priority on all threads\n", policy_str);
AST_RWLIST_RDLOCK(&thread_list);
AST_RWLIST_TRAVERSE(&thread_list, cur, list) {
/* Don't care about the return value. It should work. */
sched_setscheduler(cur->lwp, policy, &sched);
}
AST_RWLIST_UNLOCK(&thread_list);
return 0;
#endif
}
/*! \brief We set ourselves to a high priority, that we might pre-empt
* everything else. If your PBX has heavy activity on it, this is a
* good thing.
*/
int ast_set_priority(int pri)
{
struct sched_param sched;
memset(&sched, 0, sizeof(sched));
#ifdef __linux__
if (pri) {
sched.sched_priority = 10;
if (sched_setscheduler(0, SCHED_RR, &sched)) {
return -1;
}
} else {
sched.sched_priority = 0;
/* According to the manpage, these parameters can never fail. */
sched_setscheduler(0, SCHED_OTHER, &sched);
}
#else
if (pri) {
if (setpriority(PRIO_PROCESS, 0, -10) == -1) {
ast_log(LOG_WARNING, "Unable to set high priority\n");
return -1;
} else
ast_verb(1, "Set to high priority\n");
} else {
/* According to the manpage, these parameters can never fail. */
setpriority(PRIO_PROCESS, 0, 0);
}
#endif
return 0;
}
HTTP: Stop accepting requests on final system shutdown. There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ ........ Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-11 17:39:13 +00:00
int ast_shutdown_final(void)
{
return shuttingdown == SHUTTING_DOWN_FINAL;
}
int ast_shutting_down(void)
{
return shutdown_pending;
}
int ast_cancel_shutdown(void)
{
int shutdown_aborted = 0;
ast_mutex_lock(&safe_system_lock);
if (shuttingdown >= SHUTDOWN_FAST) {
shuttingdown = NOT_SHUTTING_DOWN;
shutdown_pending = 0;
shutdown_aborted = 1;
}
ast_mutex_unlock(&safe_system_lock);
return shutdown_aborted;
}
/*!
* \internal
* \brief Initiate system shutdown -- prevents new channels from being allocated.
*/
static void ast_begin_shutdown(void)
{
ast_mutex_lock(&safe_system_lock);
if (shuttingdown != NOT_SHUTTING_DOWN) {
shutdown_pending = 1;
}
ast_mutex_unlock(&safe_system_lock);
}
static int can_safely_quit(shutdown_nice_t niceness, int restart);
static void really_quit(int num, shutdown_nice_t niceness, int restart);
static void quit_handler(int num, shutdown_nice_t niceness, int restart)
{
if (can_safely_quit(niceness, restart)) {
really_quit(num, niceness, restart);
/* No one gets here. */
}
/* It wasn't our time. */
}
HTTP: Stop accepting requests on final system shutdown. There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ ........ Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-11 17:39:13 +00:00
#define SHUTDOWN_TIMEOUT 15 /* Seconds */
/*!
* \internal
* \brief Wait for all channels to die, a timeout, or shutdown cancelled.
* \since 13.3.0
*
* \param niceness Shutdown niceness in effect
* \param seconds Number of seconds to wait or less than zero if indefinitely.
*
* \retval zero if waiting wasn't necessary. We were idle.
* \retval non-zero if we had to wait.
*/
static int wait_for_channels_to_die(shutdown_nice_t niceness, int seconds)
{
time_t start;
time_t now;
int waited = 0;
time(&start);
for (;;) {
if (!ast_undestroyed_channels() || shuttingdown != niceness) {
break;
}
if (seconds < 0) {
/* No timeout so just poll every second */
sleep(1);
} else {
time(&now);
/* Wait up to the given seconds for all channels to go away */
if (seconds < (now - start)) {
break;
}
/* Sleep 1/10 of a second */
usleep(100000);
}
waited = 1;
}
return waited;
}
static int can_safely_quit(shutdown_nice_t niceness, int restart)
{
HTTP: Stop accepting requests on final system shutdown. There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ ........ Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-11 17:39:13 +00:00
int waited = 0;
/* Check if someone else isn't already doing this. */
ast_mutex_lock(&safe_system_lock);
if (shuttingdown != NOT_SHUTTING_DOWN && niceness >= shuttingdown) {
/* Already in progress and other request was less nice. */
ast_mutex_unlock(&safe_system_lock);
ast_verbose("Ignoring asterisk %s request, already in progress.\n", restart ? "restart" : "shutdown");
return 0;
}
shuttingdown = niceness;
ast_mutex_unlock(&safe_system_lock);
/* Try to get as many CDRs as possible submitted to the backend engines
* (if in batch mode). really_quit happens to call it again when running
* the atexit handlers, otherwise this would be a bit early. */
ast_cdr_engine_term();
HTTP: Stop accepting requests on final system shutdown. There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ ........ Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-11 17:39:13 +00:00
/*
* Shutdown the message queue for the technology agnostic message channel.
* This has to occur before we pause shutdown pending ast_undestroyed_channels.
*
* XXX This is not reversed on shutdown cancel.
*/
ast_msg_shutdown();
if (niceness == SHUTDOWN_NORMAL) {
/* Begin shutdown routine, hanging up active channels */
HTTP: Stop accepting requests on final system shutdown. There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ ........ Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-11 17:39:13 +00:00
ast_begin_shutdown();
if (ast_opt_console) {
ast_verb(0, "Beginning asterisk %s....\n", restart ? "restart" : "shutdown");
}
HTTP: Stop accepting requests on final system shutdown. There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ ........ Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-11 17:39:13 +00:00
ast_softhangup_all();
waited |= wait_for_channels_to_die(niceness, SHUTDOWN_TIMEOUT);
} else if (niceness >= SHUTDOWN_NICE) {
if (niceness != SHUTDOWN_REALLY_NICE) {
HTTP: Stop accepting requests on final system shutdown. There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ ........ Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-11 17:39:13 +00:00
ast_begin_shutdown();
}
if (ast_opt_console) {
ast_verb(0, "Waiting for inactivity to perform %s...\n", restart ? "restart" : "halt");
}
HTTP: Stop accepting requests on final system shutdown. There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ ........ Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-11 17:39:13 +00:00
waited |= wait_for_channels_to_die(niceness, -1);
}
/* Re-acquire lock and check if someone changed the niceness, in which
* case someone else has taken over the shutdown.
*/
ast_mutex_lock(&safe_system_lock);
if (shuttingdown != niceness) {
if (shuttingdown == NOT_SHUTTING_DOWN && ast_opt_console) {
ast_verb(0, "Asterisk %s cancelled.\n", restart ? "restart" : "shutdown");
}
ast_mutex_unlock(&safe_system_lock);
return 0;
}
HTTP: Stop accepting requests on final system shutdown. There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ ........ Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-11 17:39:13 +00:00
if (niceness >= SHUTDOWN_REALLY_NICE) {
shuttingdown = SHUTTING_DOWN;
ast_mutex_unlock(&safe_system_lock);
/* No more Mr. Nice guy. We are committed to shutting down now. */
ast_begin_shutdown();
ast_softhangup_all();
waited |= wait_for_channels_to_die(SHUTTING_DOWN, SHUTDOWN_TIMEOUT);
ast_mutex_lock(&safe_system_lock);
}
shuttingdown = SHUTTING_DOWN_FINAL;
ast_mutex_unlock(&safe_system_lock);
HTTP: Stop accepting requests on final system shutdown. There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ ........ Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-11 17:39:13 +00:00
if (niceness >= SHUTDOWN_NORMAL && waited) {
/*
* We were not idle. Give things in progress a chance to
* recognize the final shutdown phase.
*/
sleep(1);
}
return 1;
}
/*! Called when exiting is certain. */
static void really_quit(int num, shutdown_nice_t niceness, int restart)
{
int active_channels;
struct ast_json *json_object = NULL;
Avoid unnecessary cleanups during immediate shutdown This patch addresses issues during immediate shutdowns, where modules are not unloaded, but Asterisk atexit handlers are run. In the typical case, this usually isn't a big deal. But the introduction of the Stasis message bus makes it much more likely for asynchronous activity to be happening off in some thread during shutdown. During an immediate shutdown, Asterisk skips unloading modules. But while it is processing the atexit handlers, there is a window of time where some of the core message types have been cleaned up, but the message bus is still running. Specifically, it's still running module subscriptions that might be using the core message types. If a message is received by that subscription in that window, it will attempt to use a message type that has been cleaned up. To solve this problem, this patch introduces ast_register_cleanup(). This function operates identically to ast_register_atexit(), except that cleanup calls are not invoked on an immediate shutdown. All of the core message type and topic cleanup was moved from atexit handlers to cleanup handlers. This ensures that core type and topic cleanup only happens if the modules that used them are first unloaded. This patch also changes the ast_assert() when accessing a cleaned up or uninitialized message type to an error log message. Message type functions are actually NULL safe across the board, so the assert was a bit heavy handed. Especially for anyone with DO_CRASH enabled. Review: https://reviewboard.asterisk.org/r/2562/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-30 17:05:53 +00:00
int run_cleanups = niceness >= SHUTDOWN_NICE;
if (run_cleanups && modules_shutdown()) {
ast_verb(0, "Some modules could not be unloaded, switching to fast shutdown\n");
run_cleanups = 0;
}
if (!restart) {
ast_sd_notify("STOPPING=1");
}
if (ast_opt_console || (ast_opt_remote && !ast_opt_exec)) {
ast_el_write_default_histfile();
if (consolethread == AST_PTHREADT_NULL || consolethread == pthread_self()) {
/* Only end if we are the consolethread, otherwise there's a race with that thread. */
if (el != NULL) {
el_end(el);
}
if (el_hist != NULL) {
history_end(el_hist);
}
} else if (!restart) {
sig_flags.need_el_end = 1;
pthread_kill(consolethread, SIGURG);
}
}
active_channels = ast_active_channels();
/* Don't publish messages if we're a remote console - we won't have all of the Stasis
* topics or message types
*/
if (!ast_opt_remote) {
json_object = ast_json_pack("{s: s, s: s}",
"Shutdown", active_channels ? "Uncleanly" : "Cleanly",
"Restart", restart ? "True" : "False");
ast_manager_publish_event("Shutdown", EVENT_FLAG_SYSTEM, json_object);
ast_json_unref(json_object);
json_object = NULL;
}
ast_verb(0, "Asterisk %s ending (%d).\n",
active_channels ? "uncleanly" : "cleanly", num);
ast_verb(0, "Executing last minute cleanups\n");
Avoid unnecessary cleanups during immediate shutdown This patch addresses issues during immediate shutdowns, where modules are not unloaded, but Asterisk atexit handlers are run. In the typical case, this usually isn't a big deal. But the introduction of the Stasis message bus makes it much more likely for asynchronous activity to be happening off in some thread during shutdown. During an immediate shutdown, Asterisk skips unloading modules. But while it is processing the atexit handlers, there is a window of time where some of the core message types have been cleaned up, but the message bus is still running. Specifically, it's still running module subscriptions that might be using the core message types. If a message is received by that subscription in that window, it will attempt to use a message type that has been cleaned up. To solve this problem, this patch introduces ast_register_cleanup(). This function operates identically to ast_register_atexit(), except that cleanup calls are not invoked on an immediate shutdown. All of the core message type and topic cleanup was moved from atexit handlers to cleanup handlers. This ensures that core type and topic cleanup only happens if the modules that used them are first unloaded. This patch also changes the ast_assert() when accessing a cleaned up or uninitialized message type to an error log message. Message type functions are actually NULL safe across the board, so the assert was a bit heavy handed. Especially for anyone with DO_CRASH enabled. Review: https://reviewboard.asterisk.org/r/2562/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-30 17:05:53 +00:00
ast_run_atexits(run_cleanups);
ast_debug(1, "Asterisk ending (%d).\n", num);
if (ast_socket > -1) {
pthread_cancel(lthread);
close(ast_socket);
ast_socket = -1;
if (!ast_socket_is_sd) {
unlink(ast_config_AST_SOCKET);
}
pthread_kill(lthread, SIGURG);
pthread_join(lthread, NULL);
}
if (ast_consock > -1)
close(ast_consock);
if (!ast_opt_remote)
unlink(ast_config_AST_PID);
ast_alertpipe_close(sig_alert_pipe);
printf("%s", term_quit());
if (restart) {
int i;
ast_verb(0, "Preparing for Asterisk restart...\n");
/* Mark all FD's for closing on exec */
for (i = 3; i < 32768; i++) {
fcntl(i, F_SETFD, FD_CLOEXEC);
}
ast_verb(0, "Asterisk is now restarting...\n");
restartnow = 1;
/* close logger */
close_logger();
clean_time_zones();
/* If there is a consolethread running send it a SIGHUP
so it can execvp, otherwise we can do it ourselves */
if ((consolethread != AST_PTHREADT_NULL) && (consolethread != pthread_self())) {
pthread_kill(consolethread, SIGHUP);
/* Give the signal handler some time to complete */
sleep(2);
} else
execvp(_argv[0], _argv);
} else {
/* close logger */
close_logger();
clean_time_zones();
}
exit(0);
}
static void __quit_handler(int num)
{
sig_flags.need_quit = 1;
if (ast_alertpipe_write(sig_alert_pipe)) {
fprintf(stderr, "quit_handler: write() failed: %s\n", strerror(errno));
}
/* There is no need to restore the signal handler here, since the app
* is going to exit */
}
static void __remote_quit_handler(int num)
{
sig_flags.need_quit = 1;
}
Logger/CLI/etc.: Fix some aesthetic issues; reduce chatty verbose messages This patch addresses some aesthetic issues in Asterisk. These are all just minor tweaks to improve the look of the CLI when used in a variety of settings. Specifically: * A number of chatty verbose messages were removed or demoted to DEBUG messages. Verbose messages with a verbosity level of 5 or higher were - if kept as verbose messages - demoted to level 4. Several messages that were emitted at verbose level 3 were demoted to 4, as announcement of dialplan applications being executed occur at level 3 (and so the effects of those applications should generally be less). * Some verbose messages that only appear when their respective 'debug' options are enabled were bumped up to always be displayed. * Prefix/timestamping of verbose messages were moved to the verboser handlers. This was done to prevent duplication of prefixes when the timestamp option (-T) is used with the CLI. * Verbose magic is removed from messages before being emitted to non-verboser handlers. This prevents the magic in multi-line verbose messages (such as SIP debug traces or the output of DumpChan) from being written to files. * _Slightly_ better support for the "light background" option (-W) was added. This includes using ast_term_quit in the output of XML documentation help, as well as changing the "Asterisk Ready" prompt to bright green on the default background (which stands a better chance of being displayed properly than bright white). Review: https://reviewboard.asterisk.org/r/3547/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28 22:54:12 +00:00
static void set_header(char *outbuf, int maxout, char level)
{
const char *cmp;
Logger/CLI/etc.: Fix some aesthetic issues; reduce chatty verbose messages This patch addresses some aesthetic issues in Asterisk. These are all just minor tweaks to improve the look of the CLI when used in a variety of settings. Specifically: * A number of chatty verbose messages were removed or demoted to DEBUG messages. Verbose messages with a verbosity level of 5 or higher were - if kept as verbose messages - demoted to level 4. Several messages that were emitted at verbose level 3 were demoted to 4, as announcement of dialplan applications being executed occur at level 3 (and so the effects of those applications should generally be less). * Some verbose messages that only appear when their respective 'debug' options are enabled were bumped up to always be displayed. * Prefix/timestamping of verbose messages were moved to the verboser handlers. This was done to prevent duplication of prefixes when the timestamp option (-T) is used with the CLI. * Verbose magic is removed from messages before being emitted to non-verboser handlers. This prevents the magic in multi-line verbose messages (such as SIP debug traces or the output of DumpChan) from being written to files. * _Slightly_ better support for the "light background" option (-W) was added. This includes using ast_term_quit in the output of XML documentation help, as well as changing the "Asterisk Ready" prompt to bright green on the default background (which stands a better chance of being displayed properly than bright white). Review: https://reviewboard.asterisk.org/r/3547/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28 22:54:12 +00:00
char date[40];
switch (level) {
Logger/CLI/etc.: Fix some aesthetic issues; reduce chatty verbose messages This patch addresses some aesthetic issues in Asterisk. These are all just minor tweaks to improve the look of the CLI when used in a variety of settings. Specifically: * A number of chatty verbose messages were removed or demoted to DEBUG messages. Verbose messages with a verbosity level of 5 or higher were - if kept as verbose messages - demoted to level 4. Several messages that were emitted at verbose level 3 were demoted to 4, as announcement of dialplan applications being executed occur at level 3 (and so the effects of those applications should generally be less). * Some verbose messages that only appear when their respective 'debug' options are enabled were bumped up to always be displayed. * Prefix/timestamping of verbose messages were moved to the verboser handlers. This was done to prevent duplication of prefixes when the timestamp option (-T) is used with the CLI. * Verbose magic is removed from messages before being emitted to non-verboser handlers. This prevents the magic in multi-line verbose messages (such as SIP debug traces or the output of DumpChan) from being written to files. * _Slightly_ better support for the "light background" option (-W) was added. This includes using ast_term_quit in the output of XML documentation help, as well as changing the "Asterisk Ready" prompt to bright green on the default background (which stands a better chance of being displayed properly than bright white). Review: https://reviewboard.asterisk.org/r/3547/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28 22:54:12 +00:00
case 0: cmp = NULL;
break;
case 1: cmp = VERBOSE_PREFIX_1;
break;
case 2: cmp = VERBOSE_PREFIX_2;
break;
case 3: cmp = VERBOSE_PREFIX_3;
break;
default: cmp = VERBOSE_PREFIX_4;
break;
}
Logger/CLI/etc.: Fix some aesthetic issues; reduce chatty verbose messages This patch addresses some aesthetic issues in Asterisk. These are all just minor tweaks to improve the look of the CLI when used in a variety of settings. Specifically: * A number of chatty verbose messages were removed or demoted to DEBUG messages. Verbose messages with a verbosity level of 5 or higher were - if kept as verbose messages - demoted to level 4. Several messages that were emitted at verbose level 3 were demoted to 4, as announcement of dialplan applications being executed occur at level 3 (and so the effects of those applications should generally be less). * Some verbose messages that only appear when their respective 'debug' options are enabled were bumped up to always be displayed. * Prefix/timestamping of verbose messages were moved to the verboser handlers. This was done to prevent duplication of prefixes when the timestamp option (-T) is used with the CLI. * Verbose magic is removed from messages before being emitted to non-verboser handlers. This prevents the magic in multi-line verbose messages (such as SIP debug traces or the output of DumpChan) from being written to files. * _Slightly_ better support for the "light background" option (-W) was added. This includes using ast_term_quit in the output of XML documentation help, as well as changing the "Asterisk Ready" prompt to bright green on the default background (which stands a better chance of being displayed properly than bright white). Review: https://reviewboard.asterisk.org/r/3547/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28 22:54:12 +00:00
if (ast_opt_timestamp) {
struct ast_tm tm;
struct timeval now = ast_tvnow();
ast_localtime(&now, &tm, NULL);
ast_strftime(date, sizeof(date), ast_logger_get_dateformat(), &tm);
}
Logger/CLI/etc.: Fix some aesthetic issues; reduce chatty verbose messages This patch addresses some aesthetic issues in Asterisk. These are all just minor tweaks to improve the look of the CLI when used in a variety of settings. Specifically: * A number of chatty verbose messages were removed or demoted to DEBUG messages. Verbose messages with a verbosity level of 5 or higher were - if kept as verbose messages - demoted to level 4. Several messages that were emitted at verbose level 3 were demoted to 4, as announcement of dialplan applications being executed occur at level 3 (and so the effects of those applications should generally be less). * Some verbose messages that only appear when their respective 'debug' options are enabled were bumped up to always be displayed. * Prefix/timestamping of verbose messages were moved to the verboser handlers. This was done to prevent duplication of prefixes when the timestamp option (-T) is used with the CLI. * Verbose magic is removed from messages before being emitted to non-verboser handlers. This prevents the magic in multi-line verbose messages (such as SIP debug traces or the output of DumpChan) from being written to files. * _Slightly_ better support for the "light background" option (-W) was added. This includes using ast_term_quit in the output of XML documentation help, as well as changing the "Asterisk Ready" prompt to bright green on the default background (which stands a better chance of being displayed properly than bright white). Review: https://reviewboard.asterisk.org/r/3547/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28 22:54:12 +00:00
snprintf(outbuf, maxout, "%s%s%s%s%s%s",
ast_opt_timestamp ? "[" : "",
ast_opt_timestamp ? date : "",
ast_opt_timestamp ? "] " : "",
cmp ? ast_term_color(COLOR_GRAY, 0) : "",
cmp ? cmp : "",
cmp ? ast_term_reset() : "");
}
struct console_state_data {
char verbose_line_level;
};
static int console_state_init(void *ptr)
{
struct console_state_data *state = ptr;
state->verbose_line_level = 0;
return 0;
}
AST_THREADSTORAGE_CUSTOM(console_state, console_state_init, ast_free_ptr);
logger: Support JSON logging with Verbose messages When 2d7a4a3357 was merged, it missed the fact that Verbose log messages are formatted and handled by 'verbosers'. Verbosers are registered functions that handle verbose messages only; they exist as a separate class of callbacks. This was done to handle the 'magic' that must be inserted into Verbose messages sent to remote consoles, so that the consoles can format the messages correctly, i.e., the leading tabs/characters. In reality, verbosers are a weird appendage: they're a separate class of formatters/message handlers outside of what handles all other log messages in Asterisk. After some code inspection, it became clear that simply passing a Verbose message along with its 'sublevel' importance through the normal logging mechanisms removes the need for verbosers altogether. This patch removes the verbosers, and makes the default log formatter aware that, if the log channel is a console log, it should simply insert the 'verbose magic' into the log messages itself. This allows the console handlers to interpret and format the verbose message themselves. This simplifies the code quite a lot, and should improve the performance of printing verbose messages by a reasonable factor: (1) It removes a number of memory allocations that were done on each verobse message (2) It removes the need to strip the verbose magic out of the verbose log messages before passing them to non-console log channels (3) It now performs fewer iterations over lists when handling verbose messages Since verbose messages are now handled like other log messages (for the most part), the JSON formatting of the messages works as well. ASTERISK-25425 Change-Id: I21bf23f0a1e489b5102f8a035fe8871552ce4f96
2016-05-14 12:24:07 +00:00
static int console_print(const char *s)
{
struct console_state_data *state =
ast_threadstorage_get(&console_state, sizeof(*state));
char prefix[80];
const char *c;
int num, res = 0;
unsigned int newline;
do {
if (VERBOSE_HASMAGIC(s)) {
Logger/CLI/etc.: Fix some aesthetic issues; reduce chatty verbose messages This patch addresses some aesthetic issues in Asterisk. These are all just minor tweaks to improve the look of the CLI when used in a variety of settings. Specifically: * A number of chatty verbose messages were removed or demoted to DEBUG messages. Verbose messages with a verbosity level of 5 or higher were - if kept as verbose messages - demoted to level 4. Several messages that were emitted at verbose level 3 were demoted to 4, as announcement of dialplan applications being executed occur at level 3 (and so the effects of those applications should generally be less). * Some verbose messages that only appear when their respective 'debug' options are enabled were bumped up to always be displayed. * Prefix/timestamping of verbose messages were moved to the verboser handlers. This was done to prevent duplication of prefixes when the timestamp option (-T) is used with the CLI. * Verbose magic is removed from messages before being emitted to non-verboser handlers. This prevents the magic in multi-line verbose messages (such as SIP debug traces or the output of DumpChan) from being written to files. * _Slightly_ better support for the "light background" option (-W) was added. This includes using ast_term_quit in the output of XML documentation help, as well as changing the "Asterisk Ready" prompt to bright green on the default background (which stands a better chance of being displayed properly than bright white). Review: https://reviewboard.asterisk.org/r/3547/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28 22:54:12 +00:00
/* always use the given line's level, otherwise
we'll use the last line's level */
state->verbose_line_level = VERBOSE_MAGIC2LEVEL(s);
Logger/CLI/etc.: Fix some aesthetic issues; reduce chatty verbose messages This patch addresses some aesthetic issues in Asterisk. These are all just minor tweaks to improve the look of the CLI when used in a variety of settings. Specifically: * A number of chatty verbose messages were removed or demoted to DEBUG messages. Verbose messages with a verbosity level of 5 or higher were - if kept as verbose messages - demoted to level 4. Several messages that were emitted at verbose level 3 were demoted to 4, as announcement of dialplan applications being executed occur at level 3 (and so the effects of those applications should generally be less). * Some verbose messages that only appear when their respective 'debug' options are enabled were bumped up to always be displayed. * Prefix/timestamping of verbose messages were moved to the verboser handlers. This was done to prevent duplication of prefixes when the timestamp option (-T) is used with the CLI. * Verbose magic is removed from messages before being emitted to non-verboser handlers. This prevents the magic in multi-line verbose messages (such as SIP debug traces or the output of DumpChan) from being written to files. * _Slightly_ better support for the "light background" option (-W) was added. This includes using ast_term_quit in the output of XML documentation help, as well as changing the "Asterisk Ready" prompt to bright green on the default background (which stands a better chance of being displayed properly than bright white). Review: https://reviewboard.asterisk.org/r/3547/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28 22:54:12 +00:00
/* move past magic */
s++;
Logger/CLI/etc.: Fix some aesthetic issues; reduce chatty verbose messages This patch addresses some aesthetic issues in Asterisk. These are all just minor tweaks to improve the look of the CLI when used in a variety of settings. Specifically: * A number of chatty verbose messages were removed or demoted to DEBUG messages. Verbose messages with a verbosity level of 5 or higher were - if kept as verbose messages - demoted to level 4. Several messages that were emitted at verbose level 3 were demoted to 4, as announcement of dialplan applications being executed occur at level 3 (and so the effects of those applications should generally be less). * Some verbose messages that only appear when their respective 'debug' options are enabled were bumped up to always be displayed. * Prefix/timestamping of verbose messages were moved to the verboser handlers. This was done to prevent duplication of prefixes when the timestamp option (-T) is used with the CLI. * Verbose magic is removed from messages before being emitted to non-verboser handlers. This prevents the magic in multi-line verbose messages (such as SIP debug traces or the output of DumpChan) from being written to files. * _Slightly_ better support for the "light background" option (-W) was added. This includes using ast_term_quit in the output of XML documentation help, as well as changing the "Asterisk Ready" prompt to bright green on the default background (which stands a better chance of being displayed properly than bright white). Review: https://reviewboard.asterisk.org/r/3547/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28 22:54:12 +00:00
set_header(prefix, sizeof(prefix), state->verbose_line_level);
} else {
*prefix = '\0';
}
c = s;
/* for a given line separate on verbose magic, newline, and eol */
if ((s = strchr(c, '\n'))) {
++s;
newline = 1;
} else {
s = strchr(c, '\0');
newline = 0;
}
/* check if we should write this line after calculating begin/end
so we process the case of a higher level line embedded within
two lower level lines */
if (state->verbose_line_level > option_verbose) {
continue;
}
Logger/CLI/etc.: Fix some aesthetic issues; reduce chatty verbose messages This patch addresses some aesthetic issues in Asterisk. These are all just minor tweaks to improve the look of the CLI when used in a variety of settings. Specifically: * A number of chatty verbose messages were removed or demoted to DEBUG messages. Verbose messages with a verbosity level of 5 or higher were - if kept as verbose messages - demoted to level 4. Several messages that were emitted at verbose level 3 were demoted to 4, as announcement of dialplan applications being executed occur at level 3 (and so the effects of those applications should generally be less). * Some verbose messages that only appear when their respective 'debug' options are enabled were bumped up to always be displayed. * Prefix/timestamping of verbose messages were moved to the verboser handlers. This was done to prevent duplication of prefixes when the timestamp option (-T) is used with the CLI. * Verbose magic is removed from messages before being emitted to non-verboser handlers. This prevents the magic in multi-line verbose messages (such as SIP debug traces or the output of DumpChan) from being written to files. * _Slightly_ better support for the "light background" option (-W) was added. This includes using ast_term_quit in the output of XML documentation help, as well as changing the "Asterisk Ready" prompt to bright green on the default background (which stands a better chance of being displayed properly than bright white). Review: https://reviewboard.asterisk.org/r/3547/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28 22:54:12 +00:00
if (!ast_strlen_zero(prefix)) {
fputs(prefix, stdout);
}
num = s - c;
if (fwrite(c, sizeof(char), num, stdout) < num) {
break;
}
if (!res) {
/* if at least some info has been written
we'll want to return true */
res = 1;
}
} while (*s);
if (newline) {
/* if ending on a newline then reset last level to zero
since what follows may be not be logging output */
state->verbose_line_level = 0;
}
if (res) {
fflush(stdout);
}
Merge team/russell/ast_verbose_threadstorage - instead of defining a free() wrapper in a bunch of files, define it as ast_free() in utils.h and remove the copies from all the files. - centralize and abstract the code used for doing thread storage. The code lives in threadstorage.h, with one function being implemented in utils.c. This new API includes generic thread storage as well as special functions for handling thread local dynamic length string buffers. - update ast_inet_ntoa() to use the new threadstorage API - update ast_state2str() to use the new threadstorage API - update ast_cli() to use the new threadstorage API - Modify manager_event() to use thread storage. Instead of using a buffer of 4096 characters as the workspace for building the manager event, use a thread local dynamic string. Now there is no length limitation on the length of the body of a manager event. - Significantly simplify the handling of ast_verbose() ... - Instead of using a static char buffer and a lock to make sure only one thread can be using ast_verbose() at a time, use a thread local dynamic string as the workspace for preparing the verbose message. Instead of locking around the entire function, the only locking done now is when the message has been built and is being deliviered to the list of registered verbose message handlers. - This function was doing a strdup() on every message passed to it and keeping a queue of the last 200 messages in memory. This has been completely removed. The only place this was used was that if there were any messages in the verbose queue when a verbose handler was registered, all of the messages in the queue would be fed to it. So, I just made sure that the console verbose handler and the network verbose handler (for remote asterisk consoles) were registered before any verbose messages. pbx_gtkconsole and pbx_kdeconsole will now lose a few verbose messages at startup, but I didn't feel the performance hit of this message queue was worth saving the initial verbose output for these very rarely used modules. - I have removed the last three arguments to the verbose handlers, leaving only the string itself because they aren't needed anymore. For example, ast_verbose had some logic for telling the verbose handler to add a newline if the buffer was completely full. Now that the buffer can grow as needed, this doesn't matter anymore. - remove unused function, ast_verbose_dmesg() which was to dispatch the message queue - Convert the list of verbose handlers to use the linked list macros. - add missing newline characters to a few ast_verbose() calls - convert the list of log channels to use the linked list macros in logger.c - fix close_logger() to close all of the files it opened for logging - update ast_log() to use a thread local dynamic string for its workspace for preparing log messages instead of a buffer of size BUFSIZ (8kB on my system) allocated on the stack. The dynamic string in this case is limited to only growing to a maximum size of BUFSIZ. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@39272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-08-08 06:32:04 +00:00
return res;
}
static int ast_all_zeros(const char *s)
{
while (*s) {
if (*s > 32)
return 0;
s++;
}
return 1;
}
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
/* This is the main console CLI command handler. Run by the main() thread. */
static void consolehandler(const char *s)
{
printf("%s", term_end());
fflush(stdout);
/* Called when readline data is available */
if (!ast_all_zeros(s))
ast_el_add_history(s);
/* The real handler for bang */
if (s[0] == '!') {
if (s[1])
ast_safe_system(s+1);
else
ast_safe_system(getenv("SHELL") ? getenv("SHELL") : "/bin/sh");
} else
ast_cli_command(STDOUT_FILENO, s);
}
static int remoteconsolehandler(const char *s)
{
int ret = 0;
/* Called when readline data is available */
if (!ast_all_zeros(s))
ast_el_add_history(s);
while (isspace(*s)) {
s++;
}
/* The real handler for bang */
if (s[0] == '!') {
if (s[1])
ast_safe_system(s+1);
else
ast_safe_system(getenv("SHELL") ? getenv("SHELL") : "/bin/sh");
ret = 1;
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
} else if ((strncasecmp(s, "quit", 4) == 0 || strncasecmp(s, "exit", 4) == 0) &&
(s[4] == '\0' || isspace(s[4]))) {
quit_handler(0, SHUTDOWN_FAST, 0);
ret = 1;
}
return ret;
}
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
static char *handle_version(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
switch (cmd) {
case CLI_INIT:
e->command = "core show version";
e->usage =
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
"Usage: core show version\n"
" Shows Asterisk version information.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != 3)
return CLI_SHOWUSAGE;
ast_cli(a->fd, "Asterisk %s built by %s @ %s on a %s running %s on %s\n",
ast_get_version(), ast_build_user, ast_build_hostname,
ast_build_machine, ast_build_os, ast_build_date);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return CLI_SUCCESS;
}
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
static char *handle_stop_now(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
switch (cmd) {
case CLI_INIT:
e->command = "core stop now";
e->usage =
"Usage: core stop now\n"
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
" Shuts down a running Asterisk immediately, hanging up all active calls .\n";
ast_cli_allow_at_shutdown(e);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != e->args)
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return CLI_SHOWUSAGE;
quit_handler(0, SHUTDOWN_NORMAL, 0 /* not restart */);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return CLI_SUCCESS;
}
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
static char *handle_stop_gracefully(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
switch (cmd) {
case CLI_INIT:
e->command = "core stop gracefully";
e->usage =
"Usage: core stop gracefully\n"
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
" Causes Asterisk to not accept new calls, and exit when all\n"
" active calls have terminated normally.\n";
ast_cli_allow_at_shutdown(e);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != e->args)
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return CLI_SHOWUSAGE;
quit_handler(0, SHUTDOWN_NICE, 0 /* no restart */);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return CLI_SUCCESS;
}
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
static char *handle_stop_when_convenient(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
switch (cmd) {
case CLI_INIT:
e->command = "core stop when convenient";
e->usage =
"Usage: core stop when convenient\n"
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
" Causes Asterisk to perform a shutdown when all active calls have ended.\n";
ast_cli_allow_at_shutdown(e);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != e->args)
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return CLI_SHOWUSAGE;
ast_cli(a->fd, "Waiting for inactivity to perform halt\n");
quit_handler(0, SHUTDOWN_REALLY_NICE, 0 /* don't restart */);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return CLI_SUCCESS;
}
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
static char *handle_restart_now(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
switch (cmd) {
case CLI_INIT:
e->command = "core restart now";
e->usage =
"Usage: core restart now\n"
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
" Causes Asterisk to hangup all calls and exec() itself performing a cold\n"
" restart.\n";
ast_cli_allow_at_shutdown(e);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != e->args)
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return CLI_SHOWUSAGE;
quit_handler(0, SHUTDOWN_NORMAL, 1 /* restart */);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return CLI_SUCCESS;
}
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
static char *handle_restart_gracefully(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
switch (cmd) {
case CLI_INIT:
e->command = "core restart gracefully";
e->usage =
"Usage: core restart gracefully\n"
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
" Causes Asterisk to stop accepting new calls and exec() itself performing a cold\n"
" restart when all active calls have ended.\n";
ast_cli_allow_at_shutdown(e);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != e->args)
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return CLI_SHOWUSAGE;
quit_handler(0, SHUTDOWN_NICE, 1 /* restart */);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return CLI_SUCCESS;
}
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
static char *handle_restart_when_convenient(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
switch (cmd) {
case CLI_INIT:
e->command = "core restart when convenient";
e->usage =
"Usage: core restart when convenient\n"
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
" Causes Asterisk to perform a cold restart when all active calls have ended.\n";
ast_cli_allow_at_shutdown(e);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != e->args)
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return CLI_SHOWUSAGE;
ast_cli(a->fd, "Waiting for inactivity to perform restart\n");
quit_handler(0, SHUTDOWN_REALLY_NICE, 1 /* restart */);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return CLI_SUCCESS;
}
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
static char *handle_abort_shutdown(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
switch (cmd) {
case CLI_INIT:
e->command = "core abort shutdown";
e->usage =
"Usage: core abort shutdown\n"
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
" Causes Asterisk to abort an executing shutdown or restart, and resume normal\n"
" call operations.\n";
ast_cli_allow_at_shutdown(e);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return NULL;
case CLI_GENERATE:
return NULL;
}
if (a->argc != e->args)
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return CLI_SHOWUSAGE;
HTTP: Stop accepting requests on final system shutdown. There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ ........ Merged revisions 431692 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-02-11 17:39:13 +00:00
ast_cancel_shutdown();
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return CLI_SUCCESS;
}
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
static char *handle_bang(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
switch (cmd) {
case CLI_INIT:
e->command = "!";
e->usage =
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
"Usage: !<command>\n"
" Executes a given shell command\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
return CLI_SUCCESS;
}
static const char warranty_lines[] = {
"\n"
" NO WARRANTY\n"
"\n"
"BECAUSE THE PROGRAM IS LICENSED FREE OF CHARGE, THERE IS NO WARRANTY\n"
"FOR THE PROGRAM, TO THE EXTENT PERMITTED BY APPLICABLE LAW. EXCEPT WHEN\n"
"OTHERWISE STATED IN WRITING THE COPYRIGHT HOLDERS AND/OR OTHER PARTIES\n"
"PROVIDE THE PROGRAM \"AS IS\" WITHOUT WARRANTY OF ANY KIND, EITHER EXPRESSED\n"
"OR IMPLIED, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF\n"
"MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE. THE ENTIRE RISK AS\n"
"TO THE QUALITY AND PERFORMANCE OF THE PROGRAM IS WITH YOU. SHOULD THE\n"
"PROGRAM PROVE DEFECTIVE, YOU ASSUME THE COST OF ALL NECESSARY SERVICING,\n"
"REPAIR OR CORRECTION.\n"
"\n"
"IN NO EVENT UNLESS REQUIRED BY APPLICABLE LAW OR AGREED TO IN WRITING\n"
"WILL ANY COPYRIGHT HOLDER, OR ANY OTHER PARTY WHO MAY MODIFY AND/OR\n"
"REDISTRIBUTE THE PROGRAM AS PERMITTED ABOVE, BE LIABLE TO YOU FOR DAMAGES,\n"
"INCLUDING ANY GENERAL, SPECIAL, INCIDENTAL OR CONSEQUENTIAL DAMAGES ARISING\n"
"OUT OF THE USE OR INABILITY TO USE THE PROGRAM (INCLUDING BUT NOT LIMITED\n"
"TO LOSS OF DATA OR DATA BEING RENDERED INACCURATE OR LOSSES SUSTAINED BY\n"
"YOU OR THIRD PARTIES OR A FAILURE OF THE PROGRAM TO OPERATE WITH ANY OTHER\n"
"PROGRAMS), EVEN IF SUCH HOLDER OR OTHER PARTY HAS BEEN ADVISED OF THE\n"
"POSSIBILITY OF SUCH DAMAGES.\n"
};
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
static char *show_warranty(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
switch (cmd) {
case CLI_INIT:
e->command = "core show warranty";
e->usage =
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
"Usage: core show warranty\n"
" Shows the warranty (if any) for this copy of Asterisk.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
ast_cli(a->fd, "%s", warranty_lines);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return CLI_SUCCESS;
}
static const char license_lines[] = {
"\n"
"This program is free software; you can redistribute it and/or modify\n"
"it under the terms of the GNU General Public License version 2 as\n"
"published by the Free Software Foundation.\n"
"\n"
"This program also contains components licensed under other licenses.\n"
"They include:\n"
"\n"
"This program is distributed in the hope that it will be useful,\n"
"but WITHOUT ANY WARRANTY; without even the implied warranty of\n"
"MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the\n"
"GNU General Public License for more details.\n"
"\n"
"You should have received a copy of the GNU General Public License\n"
"along with this program; if not, write to the Free Software\n"
"Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA\n"
};
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
static char *show_license(struct ast_cli_entry *e, int cmd, struct ast_cli_args *a)
{
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
switch (cmd) {
case CLI_INIT:
e->command = "core show license";
e->usage =
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
"Usage: core show license\n"
" Shows the license(s) for this copy of Asterisk.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
ast_cli(a->fd, "%s", license_lines);
Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :) (closes issue #10724) Reported by: eliel Patches: chan_skinny.c.patch uploaded by eliel (license 64) chan_oss.c.patch uploaded by eliel (license 64) chan_mgcp.c.patch2 uploaded by eliel (license 64) pbx_config.c.patch uploaded by seanbright (license 71) iax2-provision.c.patch uploaded by eliel (license 64) chan_gtalk.c.patch uploaded by eliel (license 64) pbx_ael.c.patch uploaded by seanbright (license 71) file.c.patch uploaded by seanbright (license 71) image.c.patch uploaded by seanbright (license 71) cli.c.patch uploaded by moy (license 222) astobj2.c.patch uploaded by moy (license 222) asterisk.c.patch uploaded by moy (license 222) res_limit.c.patch uploaded by seanbright (license 71) res_convert.c.patch uploaded by seanbright (license 71) res_crypto.c.patch uploaded by seanbright (license 71) app_osplookup.c.patch uploaded by seanbright (license 71) app_rpt.c.patch uploaded by seanbright (license 71) app_mixmonitor.c.patch uploaded by seanbright (license 71) channel.c.patch uploaded by seanbright (license 71) translate.c.patch uploaded by seanbright (license 71) udptl.c.patch uploaded by seanbright (license 71) threadstorage.c.patch uploaded by seanbright (license 71) db.c.patch uploaded by seanbright (license 71) cdr.c.patch uploaded by moy (license 222) pbd_dundi.c.patch uploaded by moy (license 222) app_osplookup-rev83558.patch uploaded by moy (license 222) res_clioriginate.c.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
return CLI_SUCCESS;
}
#define ASTERISK_PROMPT "*CLI> "
/*!
* \brief Shutdown Asterisk CLI commands.
*
* \note These CLI commands cannot be unregistered at shutdown
* because one of them is likely the reason for the shutdown.
* The CLI generates a warning if a command is in-use when it is
* unregistered.
*/
static struct ast_cli_entry cli_asterisk_shutdown[] = {
AST_CLI_DEFINE(handle_stop_now, "Shut down Asterisk immediately"),
AST_CLI_DEFINE(handle_stop_gracefully, "Gracefully shut down Asterisk"),
AST_CLI_DEFINE(handle_stop_when_convenient, "Shut down Asterisk at empty call volume"),
AST_CLI_DEFINE(handle_restart_now, "Restart Asterisk immediately"),
AST_CLI_DEFINE(handle_restart_gracefully, "Restart Asterisk gracefully"),
AST_CLI_DEFINE(handle_restart_when_convenient, "Restart Asterisk at empty call volume"),
};
static struct ast_cli_entry cli_asterisk[] = {
AST_CLI_DEFINE(handle_abort_shutdown, "Cancel a running shutdown"),
AST_CLI_DEFINE(show_warranty, "Show the warranty (if any) for this copy of Asterisk"),
AST_CLI_DEFINE(show_license, "Show the license(s) for this copy of Asterisk"),
AST_CLI_DEFINE(handle_version, "Display version info"),
AST_CLI_DEFINE(handle_bang, "Execute a shell command"),
#if !defined(LOW_MEMORY)
AST_CLI_DEFINE(handle_show_threads, "Show running threads"),
#if defined(HAVE_SYSINFO) || defined(HAVE_SYSCTL)
AST_CLI_DEFINE(handle_show_sysinfo, "Show System Information"),
#endif
AST_CLI_DEFINE(handle_show_profile, "Display profiling info"),
AST_CLI_DEFINE(handle_show_settings, "Show some core settings"),
AST_CLI_DEFINE(handle_clear_profile, "Clear profiling info"),
#endif /* ! LOW_MEMORY */
};
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
static void send_rasterisk_connect_commands(void)
{
char buf[80];
/*
* Tell the server asterisk instance about the verbose level
* initially desired.
*/
if (option_verbose) {
snprintf(buf, sizeof(buf), "core set verbose atleast %d silent", option_verbose);
fdsend(ast_consock, buf);
}
if (option_debug) {
snprintf(buf, sizeof(buf), "core set debug atleast %d", option_debug);
fdsend(ast_consock, buf);
}
/* Leave verbose filtering to the server. */
option_verbose = INT_MAX;
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
if (!ast_opt_mute) {
fdsend(ast_consock, "logger mute silent");
} else {
printf("log and verbose output currently muted ('logger mute' to unmute)\n");
}
}
#ifdef HAVE_LIBEDIT_IS_UNICODE
#define CHAR_T_LIBEDIT wchar_t
#define CHAR_TO_LIBEDIT(c) btowc(c)
#else
#define CHAR_T_LIBEDIT char
#define CHAR_TO_LIBEDIT(c) c
#endif
static int ast_el_read_char(EditLine *editline, CHAR_T_LIBEDIT *cp)
{
int num_read = 0;
int lastpos = 0;
struct pollfd fds[2];
int res;
int max;
#define EL_BUF_SIZE 512
char buf[EL_BUF_SIZE];
for (;;) {
max = 1;
fds[0].fd = ast_consock;
fds[0].events = POLLIN;
if (!ast_opt_exec) {
fds[1].fd = STDIN_FILENO;
fds[1].events = POLLIN;
max++;
}
Merged revisions 182810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-18 02:28:55 +00:00
res = ast_poll(fds, max, -1);
if (res < 0) {
if (sig_flags.need_quit || sig_flags.need_quit_handler || sig_flags.need_el_end) {
break;
}
if (errno == EINTR) {
continue;
}
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "poll failed: %s\n", strerror(errno));
break;
}
if (!ast_opt_exec && fds[1].revents) {
char c = '\0';
num_read = read(STDIN_FILENO, &c, 1);
if (num_read < 1) {
break;
}
*cp = CHAR_TO_LIBEDIT(c);
return num_read;
}
if (fds[0].revents) {
res = read(ast_consock, buf, sizeof(buf) - 1);
/* if the remote side disappears exit */
if (res < 1) {
fprintf(stderr, "\nDisconnected from Asterisk server\n");
if (!ast_opt_reconnect) {
quit_handler(0, SHUTDOWN_FAST, 0);
} else {
int tries;
int reconnects_per_second = 20;
fprintf(stderr, "Attempting to reconnect for 30 seconds\n");
for (tries = 0; tries < 30 * reconnects_per_second; tries++) {
if (ast_tryconnect()) {
fprintf(stderr, "Reconnect succeeded after %.3f seconds\n", 1.0 / reconnects_per_second * tries);
printf("%s", term_quit());
WELCOME_MESSAGE;
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
send_rasterisk_connect_commands();
break;
}
usleep(1000000 / reconnects_per_second);
}
if (tries >= 30 * reconnects_per_second) {
fprintf(stderr, "Failed to reconnect for 30 seconds. Quitting.\n");
quit_handler(0, SHUTDOWN_FAST, 0);
}
}
continue;
}
buf[res] = '\0';
/* Write over the CLI prompt */
if (!ast_opt_exec && !lastpos) {
if (write(STDOUT_FILENO, "\r", 5) < 0) {
}
}
logger: Support JSON logging with Verbose messages When 2d7a4a3357 was merged, it missed the fact that Verbose log messages are formatted and handled by 'verbosers'. Verbosers are registered functions that handle verbose messages only; they exist as a separate class of callbacks. This was done to handle the 'magic' that must be inserted into Verbose messages sent to remote consoles, so that the consoles can format the messages correctly, i.e., the leading tabs/characters. In reality, verbosers are a weird appendage: they're a separate class of formatters/message handlers outside of what handles all other log messages in Asterisk. After some code inspection, it became clear that simply passing a Verbose message along with its 'sublevel' importance through the normal logging mechanisms removes the need for verbosers altogether. This patch removes the verbosers, and makes the default log formatter aware that, if the log channel is a console log, it should simply insert the 'verbose magic' into the log messages itself. This allows the console handlers to interpret and format the verbose message themselves. This simplifies the code quite a lot, and should improve the performance of printing verbose messages by a reasonable factor: (1) It removes a number of memory allocations that were done on each verobse message (2) It removes the need to strip the verbose magic out of the verbose log messages before passing them to non-console log channels (3) It now performs fewer iterations over lists when handling verbose messages Since verbose messages are now handled like other log messages (for the most part), the JSON formatting of the messages works as well. ASTERISK-25425 Change-Id: I21bf23f0a1e489b5102f8a035fe8871552ce4f96
2016-05-14 12:24:07 +00:00
console_print(buf);
if ((res < EL_BUF_SIZE - 1) && ((buf[res-1] == '\n') || (res >= 2 && buf[res-2] == '\n'))) {
*cp = CHAR_TO_LIBEDIT(CC_REFRESH);
return 1;
}
lastpos = 1;
}
}
*cp = CHAR_TO_LIBEDIT('\0');
return 0;
}
static struct ast_str *prompt = NULL;
static char *cli_prompt(EditLine *editline)
{
char tmp[100];
char *pfmt;
int color_used = 0;
static int cli_prompt_changes = 0;
struct passwd *pw;
struct group *gr;
if (prompt == NULL) {
prompt = ast_str_create(100);
} else if (!cli_prompt_changes) {
return ast_str_buffer(prompt);
} else {
ast_str_reset(prompt);
}
if ((pfmt = getenv("ASTERISK_PROMPT"))) {
char *t = pfmt;
struct timeval ts = ast_tvnow();
while (*t != '\0') {
if (*t == '%') {
char hostname[MAXHOSTNAMELEN] = "";
int i, which;
struct ast_tm tm = { 0, };
int fgcolor = COLOR_WHITE, bgcolor = COLOR_BLACK;
t++;
switch (*t) {
case 'C': /* color */
t++;
if (sscanf(t, "%30d;%30d%n", &fgcolor, &bgcolor, &i) == 2) {
ast_term_color_code(&prompt, fgcolor, bgcolor);
t += i - 1;
} else if (sscanf(t, "%30d%n", &fgcolor, &i) == 1) {
ast_term_color_code(&prompt, fgcolor, 0);
t += i - 1;
}
/* If the color has been reset correctly, then there's no need to reset it later */
color_used = ((fgcolor == COLOR_WHITE) && (bgcolor == COLOR_BLACK)) ? 0 : 1;
break;
case 'd': /* date */
if (ast_localtime(&ts, &tm, NULL)) {
ast_strftime(tmp, sizeof(tmp), "%Y-%m-%d", &tm);
ast_str_append(&prompt, 0, "%s", tmp);
cli_prompt_changes++;
}
break;
case 'g': /* group */
if ((gr = getgrgid(getgid()))) {
ast_str_append(&prompt, 0, "%s", gr->gr_name);
}
break;
case 'h': /* hostname */
if (!gethostname(hostname, sizeof(hostname) - 1)) {
ast_str_append(&prompt, 0, "%s", hostname);
} else {
ast_str_append(&prompt, 0, "%s", "localhost");
}
break;
case 'H': /* short hostname */
if (!gethostname(hostname, sizeof(hostname) - 1)) {
char *dotptr;
if ((dotptr = strchr(hostname, '.'))) {
*dotptr = '\0';
}
ast_str_append(&prompt, 0, "%s", hostname);
} else {
ast_str_append(&prompt, 0, "%s", "localhost");
}
break;
#ifdef HAVE_GETLOADAVG
case 'l': /* load avg */
t++;
if (sscanf(t, "%30d", &which) == 1 && which > 0 && which <= 3) {
double list[3];
getloadavg(list, 3);
ast_str_append(&prompt, 0, "%.2f", list[which - 1]);
cli_prompt_changes++;
}
break;
#endif
case 's': /* Asterisk system name (from asterisk.conf) */
ast_str_append(&prompt, 0, "%s", ast_config_AST_SYSTEM_NAME);
break;
case 't': /* time */
if (ast_localtime(&ts, &tm, NULL)) {
ast_strftime(tmp, sizeof(tmp), "%H:%M:%S", &tm);
ast_str_append(&prompt, 0, "%s", tmp);
cli_prompt_changes++;
}
break;
case 'u': /* username */
if ((pw = getpwuid(getuid()))) {
ast_str_append(&prompt, 0, "%s", pw->pw_name);
}
break;
case '#': /* process console or remote? */
ast_str_append(&prompt, 0, "%c", ast_opt_remote ? '>' : '#');
break;
case '%': /* literal % */
ast_str_append(&prompt, 0, "%c", '%');
break;
case '\0': /* % is last character - prevent bug */
t--;
break;
}
} else {
ast_str_append(&prompt, 0, "%c", *t);
}
t++;
}
if (color_used) {
/* Force colors back to normal at end */
ast_term_color_code(&prompt, 0, 0);
}
} else {
Logger/CLI/etc.: Fix some aesthetic issues; reduce chatty verbose messages This patch addresses some aesthetic issues in Asterisk. These are all just minor tweaks to improve the look of the CLI when used in a variety of settings. Specifically: * A number of chatty verbose messages were removed or demoted to DEBUG messages. Verbose messages with a verbosity level of 5 or higher were - if kept as verbose messages - demoted to level 4. Several messages that were emitted at verbose level 3 were demoted to 4, as announcement of dialplan applications being executed occur at level 3 (and so the effects of those applications should generally be less). * Some verbose messages that only appear when their respective 'debug' options are enabled were bumped up to always be displayed. * Prefix/timestamping of verbose messages were moved to the verboser handlers. This was done to prevent duplication of prefixes when the timestamp option (-T) is used with the CLI. * Verbose magic is removed from messages before being emitted to non-verboser handlers. This prevents the magic in multi-line verbose messages (such as SIP debug traces or the output of DumpChan) from being written to files. * _Slightly_ better support for the "light background" option (-W) was added. This includes using ast_term_quit in the output of XML documentation help, as well as changing the "Asterisk Ready" prompt to bright green on the default background (which stands a better chance of being displayed properly than bright white). Review: https://reviewboard.asterisk.org/r/3547/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28 22:54:12 +00:00
ast_str_set(&prompt, 0, "%s%s",
remotehostname ? remotehostname : "",
ASTERISK_PROMPT);
}
return ast_str_buffer(prompt);
}
static struct ast_vector_string *ast_el_strtoarr(char *buf)
{
char *retstr;
struct ast_vector_string *vec = ast_calloc(1, sizeof(*vec));
if (!vec) {
return NULL;
}
while ((retstr = strsep(&buf, " "))) {
if (!strcmp(retstr, AST_CLI_COMPLETE_EOF)) {
break;
}
retstr = ast_strdup(retstr);
if (!retstr || AST_VECTOR_APPEND(vec, retstr)) {
ast_free(retstr);
goto vector_cleanup;
}
}
if (!AST_VECTOR_SIZE(vec)) {
goto vector_cleanup;
}
return vec;
vector_cleanup:
AST_VECTOR_CALLBACK_VOID(vec, ast_free);
AST_VECTOR_PTR_FREE(vec);
return NULL;
}
static void ast_cli_display_match_list(struct ast_vector_string *matches, int max)
{
int idx = 1;
/* find out how many entries can be put on one line, with two spaces between strings */
int limit = ast_get_termcols(STDOUT_FILENO) / (max + 2);
if (limit == 0) {
limit = 1;
}
for (;;) {
int numoutputline;
for (numoutputline = 0; numoutputline < limit && idx < AST_VECTOR_SIZE(matches); idx++) {
numoutputline++;
fprintf(stdout, "%-*s ", max, AST_VECTOR_GET(matches, idx));
}
if (!numoutputline) {
break;
}
fprintf(stdout, "\n");
}
}
static char *cli_complete(EditLine *editline, int ch)
{
int len = 0;
char *ptr;
struct ast_vector_string *matches;
int retval = CC_ERROR;
char savechr;
int res;
LineInfo *lf = (LineInfo *)el_line(editline);
savechr = *(char *)lf->cursor;
*(char *)lf->cursor = '\0';
ptr = (char *)lf->cursor;
if (ptr) {
while (ptr > lf->buffer) {
if (isspace(*ptr)) {
ptr++;
break;
}
ptr--;
}
}
len = lf->cursor - ptr;
if (ast_opt_remote) {
#define CMD_MATCHESARRAY "_COMMAND MATCHESARRAY \"%s\" \"%s\""
char *mbuf;
char *new_mbuf;
int mlen = 0;
int maxmbuf = ast_asprintf(&mbuf, CMD_MATCHESARRAY, lf->buffer, ptr);
if (maxmbuf == -1) {
*((char *) lf->cursor) = savechr;
return (char *)(CC_ERROR);
}
fdsend(ast_consock, mbuf);
res = 0;
mlen = 0;
mbuf[0] = '\0';
while (!strstr(mbuf, AST_CLI_COMPLETE_EOF) && res != -1) {
if (mlen + 1024 > maxmbuf) {
/* Expand buffer to the next 1024 byte increment plus a NULL terminator. */
maxmbuf = mlen + 1024;
new_mbuf = ast_realloc(mbuf, maxmbuf + 1);
if (!new_mbuf) {
ast_free(mbuf);
*((char *) lf->cursor) = savechr;
return (char *)(CC_ERROR);
}
mbuf = new_mbuf;
}
/* Only read 1024 bytes at a time */
res = read(ast_consock, mbuf + mlen, 1024);
if (res > 0) {
if (!strncmp(mbuf, "Usage:", 6)) {
/*
* Abort on malformed tab completes
* If help (tab complete) follows certain
* special characters, the main Asterisk process
* provides usage for the internal tab complete
* helper command that the remote console processes
* use.
* If this happens, the AST_CLI_COMPLETE_EOF sentinel
* value never gets sent. As a result, we'll just block
* forever if we don't handle this case.
* If we get command usage on a tab complete, then
* we know this scenario just happened and we should
* just silently ignore and do nothing.
*/
break;
}
mlen += res;
mbuf[mlen] = '\0';
}
}
mbuf[mlen] = '\0';
matches = ast_el_strtoarr(mbuf);
ast_free(mbuf);
} else {
matches = ast_cli_completion_vector((char *)lf->buffer, ptr);
}
if (matches) {
int i;
int maxlen, match_len;
const char *best_match = AST_VECTOR_GET(matches, 0);
if (!ast_strlen_zero(best_match)) {
el_deletestr(editline, (int) len);
el_insertstr(editline, best_match);
retval = CC_REFRESH;
}
if (AST_VECTOR_SIZE(matches) == 2) {
/* Found an exact match */
el_insertstr(editline, " ");
retval = CC_REFRESH;
} else {
/* Must be more than one match */
for (i = 1, maxlen = 0; i < AST_VECTOR_SIZE(matches); i++) {
match_len = strlen(AST_VECTOR_GET(matches, i));
if (match_len > maxlen) {
maxlen = match_len;
}
}
fprintf(stdout, "\n");
ast_cli_display_match_list(matches, maxlen);
retval = CC_REDISPLAY;
}
AST_VECTOR_CALLBACK_VOID(matches, ast_free);
AST_VECTOR_PTR_FREE(matches);
}
*((char *) lf->cursor) = savechr;
return (char *)(long)retval;
}
static int ast_el_initialize(void)
{
HistEvent ev;
char *editor, *editrc = getenv("EDITRC");
if (!(editor = getenv("AST_EDITMODE"))) {
if (!(editor = getenv("AST_EDITOR"))) {
editor = "emacs";
}
}
if (el != NULL)
el_end(el);
if (el_hist != NULL)
history_end(el_hist);
el = el_init("asterisk", stdin, stdout, stderr);
el_set(el, EL_PROMPT, cli_prompt);
el_set(el, EL_EDITMODE, 1);
el_set(el, EL_EDITOR, editor);
el_hist = history_init();
if (!el || !el_hist)
return -1;
/* setup history with 100 entries */
history(el_hist, &ev, H_SETSIZE, 100);
el_set(el, EL_HIST, history, el_hist);
el_set(el, EL_ADDFN, "ed-complete", "Complete argument", cli_complete);
/* Bind <tab> to command completion */
el_set(el, EL_BIND, "^I", "ed-complete", NULL);
/* Bind ? to command completion */
el_set(el, EL_BIND, "?", "ed-complete", NULL);
/* Bind ^D to redisplay */
el_set(el, EL_BIND, "^D", "ed-redisplay", NULL);
/* Bind Delete to delete char left */
el_set(el, EL_BIND, "\\e[3~", "ed-delete-next-char", NULL);
/* Bind Home and End to move to line start and end */
el_set(el, EL_BIND, "\\e[1~", "ed-move-to-beg", NULL);
el_set(el, EL_BIND, "\\e[4~", "ed-move-to-end", NULL);
/* Bind C-left and C-right to move by word (not all terminals) */
el_set(el, EL_BIND, "\\eOC", "vi-next-word", NULL);
el_set(el, EL_BIND, "\\eOD", "vi-prev-word", NULL);
if (editrc) {
el_source(el, editrc);
}
return 0;
}
#define MAX_HISTORY_COMMAND_LENGTH 256
static int ast_el_add_history(const char *buf)
{
HistEvent ev;
char *stripped_buf;
if (el_hist == NULL || el == NULL) {
ast_el_initialize();
}
if (strlen(buf) > (MAX_HISTORY_COMMAND_LENGTH - 1)) {
return 0;
}
stripped_buf = ast_strip(ast_strdupa(buf));
/* HISTCONTROL=ignoredups */
if (!history(el_hist, &ev, H_FIRST) && strcmp(ev.str, stripped_buf) == 0) {
return 0;
}
return history(el_hist, &ev, H_ENTER, stripped_buf);
}
static int ast_el_write_history(const char *filename)
{
HistEvent ev;
if (el_hist == NULL || el == NULL)
ast_el_initialize();
return (history(el_hist, &ev, H_SAVE, filename));
}
static int ast_el_read_history(const char *filename)
{
HistEvent ev;
if (el_hist == NULL || el == NULL) {
ast_el_initialize();
}
return history(el_hist, &ev, H_LOAD, filename);
}
static void process_histfile(int (*readwrite)(const char *filename))
{
struct passwd *pw = getpwuid(geteuid());
int ret = 0;
char *name = NULL;
if (!pw || ast_strlen_zero(pw->pw_dir)) {
ast_log(LOG_ERROR, "Unable to determine home directory. History read/write disabled.\n");
return;
}
ret = ast_asprintf(&name, "%s/.asterisk_history", pw->pw_dir);
if (ret <= 0) {
ast_log(LOG_ERROR, "Unable to create history file name. History read/write disabled.\n");
return;
}
ret = readwrite(name);
if (ret < 0) {
ast_log(LOG_ERROR, "Unable to read or write history file '%s'\n", name);
}
ast_free(name);
return;
}
static void ast_el_read_default_histfile(void)
{
process_histfile(ast_el_read_history);
}
static void ast_el_write_default_histfile(void)
{
process_histfile(ast_el_write_history);
}
static void ast_remotecontrol(char *data)
{
char buf[256] = "";
int res;
char *hostname;
char *cpid;
char *version;
int pid;
char *stringp = NULL;
char *ebuf;
int num = 0;
ast_term_init();
printf("%s", term_end());
fflush(stdout);
memset(&sig_flags, 0, sizeof(sig_flags));
signal(SIGINT, __remote_quit_handler);
signal(SIGTERM, __remote_quit_handler);
signal(SIGHUP, __remote_quit_handler);
if (read(ast_consock, buf, sizeof(buf) - 1) < 0) {
ast_log(LOG_ERROR, "read() failed: %s\n", strerror(errno));
return;
}
if (data) {
char prefix[] = "cli quit after ";
char *tmp = ast_alloca(strlen(data) + strlen(prefix) + 1);
sprintf(tmp, "%s%s", prefix, data);
if (write(ast_consock, tmp, strlen(tmp) + 1) < 0) {
ast_log(LOG_ERROR, "write() failed: %s\n", strerror(errno));
if (sig_flags.need_quit || sig_flags.need_quit_handler || sig_flags.need_el_end) {
return;
}
}
}
stringp = buf;
hostname = strsep(&stringp, "/");
cpid = strsep(&stringp, "/");
version = strsep(&stringp, "\n");
if (!version)
version = "<Version Unknown>";
stringp = hostname;
strsep(&stringp, ".");
if (cpid)
pid = atoi(cpid);
else
pid = -1;
if (!data) {
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
send_rasterisk_connect_commands();
}
if (ast_opt_exec && data) { /* hack to print output then exit if asterisk -rx is used */
int linefull = 1, prev_linefull = 1, prev_line_verbose = 0;
struct pollfd fds;
fds.fd = ast_consock;
fds.events = POLLIN;
fds.revents = 0;
while (ast_poll(&fds, 1, 60000) > 0) {
char buffer[512] = "", *curline = buffer, *nextline;
int not_written = 1;
if (sig_flags.need_quit || sig_flags.need_quit_handler || sig_flags.need_el_end) {
break;
}
if (read(ast_consock, buffer, sizeof(buffer) - 1) <= 0) {
break;
}
do {
prev_linefull = linefull;
if ((nextline = strchr(curline, '\n'))) {
linefull = 1;
nextline++;
} else {
linefull = 0;
nextline = strchr(curline, '\0');
}
/* Skip verbose lines */
/* Prev line full? | Line is verbose | Last line verbose? | Print
* TRUE | TRUE* | TRUE | FALSE
* TRUE | TRUE* | FALSE | FALSE
* TRUE | FALSE* | TRUE | TRUE
* TRUE | FALSE* | FALSE | TRUE
* FALSE | TRUE | TRUE* | FALSE
* FALSE | TRUE | FALSE* | TRUE
* FALSE | FALSE | TRUE* | FALSE
* FALSE | FALSE | FALSE* | TRUE
*/
if ((!prev_linefull && !prev_line_verbose) || (prev_linefull && *curline > 0)) {
prev_line_verbose = 0;
not_written = 0;
if (write(STDOUT_FILENO, curline, nextline - curline) < 0) {
ast_log(LOG_WARNING, "write() failed: %s\n", strerror(errno));
}
} else {
prev_line_verbose = 1;
}
curline = nextline;
} while (!ast_strlen_zero(curline));
/* No non-verbose output in 60 seconds. */
if (not_written) {
break;
}
}
return;
}
ast_verbose("Connected to Asterisk %s currently running on %s (pid = %d)\n", version, hostname, pid);
ast_init_logger_for_socket_console();
remotehostname = hostname;
if (el_hist == NULL || el == NULL)
ast_el_initialize();
ast_el_read_default_histfile();
el_set(el, EL_GETCFN, ast_el_read_char);
for (;;) {
ebuf = (char *)el_gets(el, &num);
if (sig_flags.need_quit || sig_flags.need_quit_handler || sig_flags.need_el_end) {
break;
}
if (!ebuf && write(1, "", 1) < 0)
break;
if (!ast_strlen_zero(ebuf)) {
if (ebuf[strlen(ebuf)-1] == '\n')
ebuf[strlen(ebuf)-1] = '\0';
if (!remoteconsolehandler(ebuf)) {
res = write(ast_consock, ebuf, strlen(ebuf) + 1);
if (res < 1) {
ast_log(LOG_WARNING, "Unable to write: %s\n", strerror(errno));
break;
}
}
}
}
printf("\nDisconnected from Asterisk server\n");
}
static int show_version(void)
{
printf("Asterisk %s\n", ast_get_version());
return 0;
}
static int show_cli_help(void)
{
printf("Asterisk %s, " COPYRIGHT_TAG "\n", ast_get_version());
printf("Usage: asterisk [OPTIONS]\n");
printf("Valid Options:\n");
printf(" -V Display version number and exit\n");
printf(" -C <configfile> Use an alternate configuration file\n");
printf(" -G <group> Run as a group other than the caller\n");
printf(" -U <user> Run as a user other than the caller\n");
printf(" -c Provide console CLI\n");
printf(" -d Increase debugging (multiple d's = more debugging)\n");
#if HAVE_WORKING_FORK
printf(" -f Do not fork\n");
printf(" -F Always fork\n");
#endif
printf(" -g Dump core in case of a crash\n");
printf(" -h This help screen\n");
printf(" -i Initialize crypto keys at startup\n");
printf(" -L <load> Limit the maximum load average before rejecting new calls\n");
printf(" -M <value> Limit the maximum number of calls to the specified value\n");
printf(" -m Mute debugging and console output on the console\n");
printf(" -n Disable console colorization. Can be used only at startup.\n");
printf(" -p Run as pseudo-realtime thread\n");
printf(" -q Quiet mode (suppress output)\n");
printf(" -r Connect to Asterisk on this machine\n");
printf(" -R Same as -r, except attempt to reconnect if disconnected\n");
printf(" -s <socket> Connect to Asterisk via socket <socket> (only valid with -r)\n");
printf(" -t Record soundfiles in /var/tmp and move them where they\n");
printf(" belong after they are done\n");
printf(" -T Display the time in [Mmm dd hh:mm:ss] format for each line\n");
printf(" of output to the CLI. Cannot be used with remote console mode.\n\n");
printf(" -v Increase verbosity (multiple v's = more verbose)\n");
printf(" -x <cmd> Execute command <cmd> (implies -r)\n");
printf(" -X Enable use of #exec in asterisk.conf\n");
printf(" -W Adjust terminal colors to compensate for a light background\n");
printf("\n");
return 0;
}
PJPROJECT logging: Made easier to get available logging levels. Use of the new logging is as simple as issuing the new CLI command or setting the new pjproject.conf option. Other options that can affect the logging are how you have the pjproject log levels mapped to Asterisk log types in pjproject.conf and if you have configured Asterisk to log the DEBUG type messages. Altering the pjproject.conf level mapping shouldn't be necessary for most installations as the default mapping is sensible. Configuring Asterisk to log the DEBUG message type is standard practice for collecting debug information. * Added CLI "pjproject set log level" command to dynamically adjust the maximum pjproject log message level. * Added CLI "pjproject show log level" command to see the currently set maximum pjproject log message level. * Added pjproject.conf startup section "log_level" option to set the initial maximum pjproject log message level so all messages could be captured from initialization. * Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into bundled pjproject. Pjproject will use the currently set run time log level to determine if a log message is generated just like Asterisk verbose and debug logging levels. * In log_forwarder(), made always log enabled and mapped pjproject log messages. DEBUG mapped log messages are no longer gated by the current Asterisk debug logging level. * Removed RAII_VAR() from res_pjproject.c:get_log_level(). ASTERISK-26630 #close Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389
2016-11-24 00:27:54 +00:00
static void read_pjproject_startup_options(void)
{
struct ast_config *cfg;
struct ast_variable *v;
struct ast_flags config_flags = { CONFIG_FLAG_NOCACHE | CONFIG_FLAG_NOREALTIME };
ast_option_pjproject_log_level = DEFAULT_PJ_LOG_MAX_LEVEL;
ast_option_pjproject_cache_pools = DEFAULT_PJPROJECT_CACHE_POOLS;
PJPROJECT logging: Made easier to get available logging levels. Use of the new logging is as simple as issuing the new CLI command or setting the new pjproject.conf option. Other options that can affect the logging are how you have the pjproject log levels mapped to Asterisk log types in pjproject.conf and if you have configured Asterisk to log the DEBUG type messages. Altering the pjproject.conf level mapping shouldn't be necessary for most installations as the default mapping is sensible. Configuring Asterisk to log the DEBUG message type is standard practice for collecting debug information. * Added CLI "pjproject set log level" command to dynamically adjust the maximum pjproject log message level. * Added CLI "pjproject show log level" command to see the currently set maximum pjproject log message level. * Added pjproject.conf startup section "log_level" option to set the initial maximum pjproject log message level so all messages could be captured from initialization. * Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into bundled pjproject. Pjproject will use the currently set run time log level to determine if a log message is generated just like Asterisk verbose and debug logging levels. * In log_forwarder(), made always log enabled and mapped pjproject log messages. DEBUG mapped log messages are no longer gated by the current Asterisk debug logging level. * Removed RAII_VAR() from res_pjproject.c:get_log_level(). ASTERISK-26630 #close Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389
2016-11-24 00:27:54 +00:00
cfg = ast_config_load2("pjproject.conf", "" /* core, can't reload */, config_flags);
if (!cfg
|| cfg == CONFIG_STATUS_FILEUNCHANGED
|| cfg == CONFIG_STATUS_FILEINVALID) {
/* We'll have to use defaults */
return;
}
for (v = ast_variable_browse(cfg, "startup"); v; v = v->next) {
if (!strcasecmp(v->name, "log_level")) {
if (sscanf(v->value, "%30d", &ast_option_pjproject_log_level) != 1) {
ast_option_pjproject_log_level = DEFAULT_PJ_LOG_MAX_LEVEL;
} else if (ast_option_pjproject_log_level < 0) {
ast_option_pjproject_log_level = 0;
} else if (MAX_PJ_LOG_MAX_LEVEL < ast_option_pjproject_log_level) {
ast_option_pjproject_log_level = MAX_PJ_LOG_MAX_LEVEL;
}
} else if (!strcasecmp(v->name, "cache_pools")) {
ast_option_pjproject_cache_pools = !ast_false(v->value);
PJPROJECT logging: Made easier to get available logging levels. Use of the new logging is as simple as issuing the new CLI command or setting the new pjproject.conf option. Other options that can affect the logging are how you have the pjproject log levels mapped to Asterisk log types in pjproject.conf and if you have configured Asterisk to log the DEBUG type messages. Altering the pjproject.conf level mapping shouldn't be necessary for most installations as the default mapping is sensible. Configuring Asterisk to log the DEBUG message type is standard practice for collecting debug information. * Added CLI "pjproject set log level" command to dynamically adjust the maximum pjproject log message level. * Added CLI "pjproject show log level" command to see the currently set maximum pjproject log message level. * Added pjproject.conf startup section "log_level" option to set the initial maximum pjproject log message level so all messages could be captured from initialization. * Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into bundled pjproject. Pjproject will use the currently set run time log level to determine if a log message is generated just like Asterisk verbose and debug logging levels. * In log_forwarder(), made always log enabled and mapped pjproject log messages. DEBUG mapped log messages are no longer gated by the current Asterisk debug logging level. * Removed RAII_VAR() from res_pjproject.c:get_log_level(). ASTERISK-26630 #close Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389
2016-11-24 00:27:54 +00:00
}
}
ast_config_destroy(cfg);
}
static void *monitor_sig_flags(void *unused)
{
for (;;) {
struct pollfd p = { sig_alert_pipe[0], POLLIN, 0 };
Merged revisions 182810 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-18 02:28:55 +00:00
ast_poll(&p, 1, -1);
if (sig_flags.need_reload) {
sig_flags.need_reload = 0;
ast_module_reload(NULL);
}
if (sig_flags.need_quit) {
sig_flags.need_quit = 0;
if ((consolethread != AST_PTHREADT_NULL) && (consolethread != pthread_self())) {
sig_flags.need_quit_handler = 1;
pthread_kill(consolethread, SIGURG);
} else {
quit_handler(0, SHUTDOWN_NORMAL, 0);
}
}
ast_alertpipe_read(sig_alert_pipe);
}
return NULL;
}
static void *canary_thread(void *unused)
{
struct stat canary_stat;
struct timeval now;
/* Give the canary time to sing */
sleep(120);
for (;;) {
now = ast_tvnow();
if (stat(canary_filename, &canary_stat) || now.tv_sec > canary_stat.st_mtime + 60) {
ast_log(LOG_WARNING,
"The canary is no more. He has ceased to be! "
"He's expired and gone to meet his maker! "
"He's a stiff! Bereft of life, he rests in peace. "
"His metabolic processes are now history! He's off the twig! "
"He's kicked the bucket. He's shuffled off his mortal coil, "
"run down the curtain, and joined the bleeding choir invisible!! "
"THIS is an EX-CANARY. (Reducing priority)\n");
set_priority_all(0);
pthread_exit(NULL);
}
/* Check the canary once a minute */
sleep(60);
}
}
/* Used by libc's atexit(3) function */
static void canary_exit(void)
{
if (canary_pid > 0) {
int status;
kill(canary_pid, SIGKILL);
waitpid(canary_pid, &status, 0);
}
}
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
/* Execute CLI commands on startup. Run by main() thread. */
static void run_startup_commands(void)
{
int fd;
struct ast_config *cfg;
struct ast_flags cfg_flags = { 0 };
struct ast_variable *v;
if (!(cfg = ast_config_load2("cli.conf", "" /* core, can't reload */, cfg_flags)))
return;
if (cfg == CONFIG_STATUS_FILEMISSING || cfg == CONFIG_STATUS_FILEUNCHANGED || cfg == CONFIG_STATUS_FILEINVALID) {
return;
}
fd = open("/dev/null", O_RDWR);
if (fd < 0) {
ast_config_destroy(cfg);
return;
}
for (v = ast_variable_browse(cfg, "startup_commands"); v; v = v->next) {
if (ast_true(v->value))
ast_cli_command(fd, v->name);
}
close(fd);
ast_config_destroy(cfg);
}
static void env_init(void)
{
setenv("AST_SYSTEMNAME", ast_config_AST_SYSTEM_NAME, 1);
setenv("AST_BUILD_HOST", ast_build_hostname, 1);
setenv("AST_BUILD_DATE", ast_build_date, 1);
setenv("AST_BUILD_KERNEL", ast_build_kernel, 1);
setenv("AST_BUILD_MACHINE", ast_build_machine, 1);
setenv("AST_BUILD_OS", ast_build_os, 1);
setenv("AST_BUILD_USER", ast_build_user, 1);
setenv("AST_VERSION", ast_get_version(), 1);
}
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
static void print_intro_message(const char *runuser, const char *rungroup)
{
if (ast_opt_console || option_verbose || (ast_opt_remote && !ast_opt_exec)) {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
WELCOME_MESSAGE;
if (runuser) {
ast_verbose("Running as user '%s'\n", runuser);
}
if (rungroup) {
ast_verbose("Running under group '%s'\n", rungroup);
}
}
}
static void main_atexit(void)
{
ast_cli_unregister_multiple(cli_asterisk, ARRAY_LEN(cli_asterisk));
}
int main(int argc, char *argv[])
{
int c;
int x;
int isroot = 1, rundir_exists = 0;
RAII_VAR(char *, runuser, NULL, ast_free);
RAII_VAR(char *, rungroup, NULL, ast_free);
RAII_VAR(char *, xarg, NULL, ast_free);
struct rlimit l;
static const char *getopt_settings = "BC:cde:FfG:ghIiL:M:mnpqRrs:TtU:VvWXx:";
/* Remember original args for restart */
if (argc > ARRAY_LEN(_argv) - 1) {
fprintf(stderr, "Truncating argument size to %d\n", (int)ARRAY_LEN(_argv) - 1);
argc = ARRAY_LEN(_argv) - 1;
}
for (x = 0; x < argc; x++)
_argv[x] = argv[x];
_argv[x] = NULL;
if (geteuid() != 0)
isroot = 0;
/* if the progname is rasterisk consider it a remote console */
if (argv[0] && (strstr(argv[0], "rasterisk")) != NULL) {
ast_set_flag(&ast_options, AST_OPT_FLAG_NO_FORK | AST_OPT_FLAG_REMOTE);
}
ast_mainpid = getpid();
/* Process command-line options that affect asterisk.conf load. */
while ((c = getopt(argc, argv, getopt_settings)) != -1) {
switch (c) {
case 'X':
ast_set_flag(&ast_options, AST_OPT_FLAG_EXEC_INCLUDES);
break;
case 'C':
set_asterisk_conf_path(optarg);
break;
case 'd':
option_debug++;
break;
case 'h':
show_cli_help();
exit(0);
case 'R':
case 'r':
case 'x':
/* ast_opt_remote is checked during config load. This is only part of what
* these options do, see the second loop for the rest of the actions. */
ast_set_flag(&ast_options, AST_OPT_FLAG_REMOTE);
break;
case 'V':
show_version();
exit(0);
case 'v':
option_verbose++;
break;
case '?':
exit(1);
}
}
/* Initialize env so it is available if #exec is used in asterisk.conf. */
env_init();
load_asterisk_conf();
/* Update env to include any systemname that was set. */
env_init();
/*! \brief Check for options
*
* \todo Document these options
*/
optind = 1;
while ((c = getopt(argc, argv, getopt_settings)) != -1) {
/*!\note Please keep the ordering here to alphabetical, capital letters
* first. This will make it easier in the future to select unused
* option flags for new features. */
switch (c) {
case 'B': /* Force black background */
ast_set_flag(&ast_options, AST_OPT_FLAG_FORCE_BLACK_BACKGROUND);
ast_clear_flag(&ast_options, AST_OPT_FLAG_LIGHT_BACKGROUND);
break;
case 'X':
/* The command-line -X option enables #exec for asterisk.conf only. */
break;
case 'C':
/* already processed. */
break;
case 'c':
ast_set_flag(&ast_options, AST_OPT_FLAG_NO_FORK | AST_OPT_FLAG_CONSOLE);
break;
case 'd':
/* already processed. */
break;
#if defined(HAVE_SYSINFO)
case 'e':
if ((sscanf(&optarg[1], "%30ld", &option_minmemfree) != 1) || (option_minmemfree < 0)) {
option_minmemfree = 0;
}
break;
#endif
#if HAVE_WORKING_FORK
case 'F':
ast_set_flag(&ast_options, AST_OPT_FLAG_ALWAYS_FORK);
break;
case 'f':
ast_set_flag(&ast_options, AST_OPT_FLAG_NO_FORK);
break;
#endif
case 'G':
rungroup = ast_strdup(optarg);
break;
case 'g':
ast_set_flag(&ast_options, AST_OPT_FLAG_DUMP_CORE);
break;
case 'h':
/* already processed. */
break;
case 'I':
fprintf(stderr,
"NOTICE: The -I option is no longer needed.\n"
" It will always be enabled if you have a timing module loaded.\n");
break;
case 'i':
ast_set_flag(&ast_options, AST_OPT_FLAG_INIT_KEYS);
break;
case 'L':
if ((sscanf(optarg, "%30lf", &ast_option_maxload) != 1) || (ast_option_maxload < 0.0)) {
ast_option_maxload = 0.0;
}
break;
case 'M':
if ((sscanf(optarg, "%30d", &ast_option_maxcalls) != 1) || (ast_option_maxcalls < 0)) {
ast_option_maxcalls = 0;
}
break;
case 'm':
ast_set_flag(&ast_options, AST_OPT_FLAG_MUTE);
break;
case 'n':
ast_set_flag(&ast_options, AST_OPT_FLAG_NO_COLOR);
break;
case 'p':
ast_set_flag(&ast_options, AST_OPT_FLAG_HIGH_PRIORITY);
break;
case 'q':
ast_set_flag(&ast_options, AST_OPT_FLAG_QUIET);
break;
case 'R':
ast_set_flag(&ast_options, AST_OPT_FLAG_NO_FORK | AST_OPT_FLAG_REMOTE | AST_OPT_FLAG_RECONNECT);
break;
case 'r':
ast_set_flag(&ast_options, AST_OPT_FLAG_NO_FORK | AST_OPT_FLAG_REMOTE);
break;
case 's':
if (ast_opt_remote) {
set_socket_path(optarg);
}
break;
case 'T':
ast_set_flag(&ast_options, AST_OPT_FLAG_TIMESTAMP);
break;
case 't':
ast_set_flag(&ast_options, AST_OPT_FLAG_CACHE_RECORD_FILES);
break;
case 'U':
runuser = ast_strdup(optarg);
break;
case 'V':
case 'v':
/* already processed. */
break;
case 'W': /* White background */
ast_set_flag(&ast_options, AST_OPT_FLAG_LIGHT_BACKGROUND);
ast_clear_flag(&ast_options, AST_OPT_FLAG_FORCE_BLACK_BACKGROUND);
break;
case 'x':
/* -r is implied by -x so set the flags -r sets as well. */
ast_set_flag(&ast_options, AST_OPT_FLAG_NO_FORK | AST_OPT_FLAG_REMOTE);
ast_set_flag(&ast_options, AST_OPT_FLAG_EXEC | AST_OPT_FLAG_NO_COLOR);
xarg = ast_strdup(optarg);
break;
case '?':
/* already processed. */
break;
}
}
if (ast_opt_remote) {
int didwarn = 0;
optind = 1;
/* Not all options can be used with remote console. Warn if they're used. */
while ((c = getopt(argc, argv, getopt_settings)) != -1) {
switch (c) {
/* okay to run with remote console */
case 'B': /* force black background */
case 'C': /* set config path */
case 'd': /* debug */
case 'h': /* help */
case 'I': /* obsolete timing option: warning already thrown if used */
case 'L': /* max load */
case 'M': /* max calls */
case 'm': /* mute */
/*! \note The q option is never used anywhere, only defined */
case 'q': /* quiet */
case 'R': /* reconnect */
case 'r': /* remote */
/*! \note Can ONLY be used with remote console */
case 's': /* set socket path */
case 'T': /* timestamp */
case 'V': /* version */
case 'v': /* verbose */
case 'W': /* white background */
case 'x': /* remote execute */
case '?': /* ? */
break;
/* can only be run when Asterisk is starting */
case 'X': /* enables #exec for asterisk.conf only. */
case 'c': /* foreground console */
case 'e': /* minimum memory free */
case 'F': /* always fork */
case 'f': /* no fork */
case 'G': /* run group */
case 'g': /* dump core */
case 'i': /* init keys */
case 'n': /* no color */
case 'p': /* high priority */
case 't': /* cache record files */
case 'U': /* run user */
fprintf(stderr, "'%c' option is not compatible with remote console mode and has no effect.\n", c);
didwarn = 1;
}
}
if (didwarn) {
fprintf(stderr, "\n"); /* if any warnings print out, make them stand out */
}
}
/* For remote connections, change the name of the remote connection.
* We do this for the benefit of init scripts (which need to know if/when
* the main asterisk process has died yet). */
if (ast_opt_remote) {
strcpy(argv[0], "rasterisk");
for (x = 1; x < argc; x++) {
argv[x] = argv[0] + 10;
}
}
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
if (!ast_language_is_prefix && !ast_opt_remote) {
fprintf(stderr, "The 'languageprefix' option in asterisk.conf is deprecated; in a future release it will be removed, and your sound files will need to be organized in the 'new style' language layout.\n");
}
if (ast_opt_always_fork && (ast_opt_remote || ast_opt_console)) {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "'alwaysfork' is not compatible with console or remote console mode; ignored\n");
ast_clear_flag(&ast_options, AST_OPT_FLAG_ALWAYS_FORK);
}
if (ast_opt_dump_core) {
memset(&l, 0, sizeof(l));
l.rlim_cur = RLIM_INFINITY;
l.rlim_max = RLIM_INFINITY;
if (setrlimit(RLIMIT_CORE, &l)) {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "Unable to disable core size resource limit: %s\n", strerror(errno));
}
}
if (getrlimit(RLIMIT_NOFILE, &l)) {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "Unable to check file descriptor limit: %s\n", strerror(errno));
}
#if !defined(CONFIGURE_RAN_AS_ROOT)
/* Check if select(2) will run with more file descriptors */
do {
int fd, fd2;
ast_fdset readers;
struct timeval tv = { 0, };
if (l.rlim_cur <= FD_SETSIZE) {
/* The limit of select()able FDs is irrelevant, because we'll never
* open one that high. */
break;
}
if (!(fd = open("/dev/null", O_RDONLY))) {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "Cannot open a file descriptor at boot? %s\n", strerror(errno));
break; /* XXX Should we exit() here? XXX */
}
fd2 = ((l.rlim_cur > sizeof(readers) * 8) ? sizeof(readers) * 8 : l.rlim_cur) - 1;
if (dup2(fd, fd2) < 0) {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "Cannot open maximum file descriptor %d at boot? %s\n", fd2, strerror(errno));
close(fd);
break;
}
FD_ZERO(&readers);
FD_SET(fd2, &readers);
if (ast_select(fd2 + 1, &readers, NULL, NULL, &tv) < 0) {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "Maximum select()able file descriptor is %d\n", FD_SETSIZE);
}
ast_FD_SETSIZE = l.rlim_cur > ast_FDMAX ? ast_FDMAX : l.rlim_cur;
close(fd);
close(fd2);
} while (0);
#elif defined(HAVE_VARIABLE_FDSET)
ast_FD_SETSIZE = l.rlim_cur > ast_FDMAX ? ast_FDMAX : l.rlim_cur;
#endif /* !defined(CONFIGURE_RAN_AS_ROOT) */
if ((!rungroup) && !ast_strlen_zero(ast_config_AST_RUN_GROUP))
rungroup = ast_strdup(ast_config_AST_RUN_GROUP);
if ((!runuser) && !ast_strlen_zero(ast_config_AST_RUN_USER))
runuser = ast_strdup(ast_config_AST_RUN_USER);
/* Must install this signal handler up here to ensure that if the canary
* fails to execute that it doesn't kill the Asterisk process.
*/
sigaction(SIGCHLD, &child_handler, NULL);
/* It's common on some platforms to clear /var/run at boot. Create the
* socket file directory before we drop privileges. */
if (mkdir(ast_config_AST_RUN_DIR, 0755)) {
if (errno == EEXIST) {
rundir_exists = 1;
} else {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "Unable to create socket file directory. Remote consoles will not be able to connect! (%s)\n", strerror(x));
}
}
#ifndef __CYGWIN__
if (isroot) {
ast_set_priority(ast_opt_high_priority);
}
if (isroot && rungroup) {
struct group *gr;
gr = getgrnam(rungroup);
if (!gr) {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "No such group '%s'!\n", rungroup);
exit(1);
}
if (!rundir_exists && chown(ast_config_AST_RUN_DIR, -1, gr->gr_gid)) {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "Unable to chgrp run directory to %d (%s)\n", (int) gr->gr_gid, rungroup);
}
if (setgid(gr->gr_gid)) {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "Unable to setgid to %d (%s)\n", (int)gr->gr_gid, rungroup);
exit(1);
}
if (setgroups(0, NULL)) {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "Unable to drop unneeded groups\n");
exit(1);
}
}
if (runuser && !ast_test_flag(&ast_options, AST_OPT_FLAG_REMOTE)) {
#ifdef HAVE_CAP
int has_cap = 1;
#endif /* HAVE_CAP */
struct passwd *pw;
pw = getpwnam(runuser);
if (!pw) {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "No such user '%s'!\n", runuser);
exit(1);
}
if (chown(ast_config_AST_RUN_DIR, pw->pw_uid, -1)) {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "Unable to chown run directory to %d (%s)\n", (int) pw->pw_uid, runuser);
}
#ifdef HAVE_CAP
if (prctl(PR_SET_KEEPCAPS, 1, 0, 0, 0)) {
ast_log(LOG_WARNING, "Unable to keep capabilities.\n");
has_cap = 0;
}
#endif /* HAVE_CAP */
if (!isroot && pw->pw_uid != geteuid()) {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "Asterisk started as nonroot, but runuser '%s' requested.\n", runuser);
exit(1);
}
if (!rungroup) {
if (setgid(pw->pw_gid)) {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "Unable to setgid to %d!\n", (int)pw->pw_gid);
exit(1);
}
if (isroot && initgroups(pw->pw_name, pw->pw_gid)) {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "Unable to init groups for '%s'\n", runuser);
exit(1);
}
}
if (setuid(pw->pw_uid)) {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "Unable to setuid to %d (%s)\n", (int)pw->pw_uid, runuser);
exit(1);
}
#ifdef HAVE_CAP
if (has_cap) {
cap_t cap;
cap = cap_from_text("cap_net_admin=eip");
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
if (cap_set_proc(cap)) {
fprintf(stderr, "Unable to install capabilities.\n");
}
if (cap_free(cap)) {
fprintf(stderr, "Unable to drop capabilities.\n");
}
}
#endif /* HAVE_CAP */
}
#endif /* __CYGWIN__ */
#ifdef linux
if (geteuid() && ast_opt_dump_core) {
if (prctl(PR_SET_DUMPABLE, 1, 0, 0, 0) < 0) {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "Unable to set the process for core dumps after changing to a non-root user. %s\n", strerror(errno));
}
}
#endif
{
#if defined(HAVE_EACCESS) || defined(HAVE_EUIDACCESS)
char dir[PATH_MAX];
if (!getcwd(dir, sizeof(dir)) || eaccess(dir, R_OK | X_OK | F_OK)) {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "Unable to access the running directory (%s). Changing to '/' for compatibility.\n", strerror(errno));
/* If we cannot access the CWD, then we couldn't dump core anyway,
* so chdir("/") won't break anything. */
if (chdir("/")) {
/* chdir(/) should never fail, so this ends up being a no-op */
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "chdir(\"/\") failed?!! %s\n", strerror(errno));
}
} else
#endif /* defined(HAVE_EACCESS) || defined(HAVE_EUIDACCESS) */
if (!ast_opt_no_fork && !ast_opt_dump_core) {
/* Backgrounding, but no cores, so chdir won't break anything. */
if (chdir("/")) {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "Unable to chdir(\"/\") ?!! %s\n", strerror(errno));
}
}
}
verbosity: Fix performance of console verbose messages. The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14 18:14:02 +00:00
/* Initial value of the maximum active system verbosity level. */
ast_verb_sys_level = option_verbose;
if (ast_sd_get_fd_un(SOCK_STREAM, ast_config_AST_SOCKET) > 0) {
ast_socket_is_sd = 1;
}
/* DO NOT perform check for existing daemon if systemd has CLI socket activation */
if (!ast_socket_is_sd && ast_tryconnect()) {
/* One is already running */
if (ast_opt_remote) {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
multi_thread_safe = 1;
if (ast_opt_exec) {
ast_remotecontrol(xarg);
quit_handler(0, SHUTDOWN_FAST, 0);
exit(0);
}
ast_term_init();
printf("%s", term_end());
fflush(stdout);
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
print_intro_message(runuser, rungroup);
printf("%s", term_quit());
ast_remotecontrol(NULL);
quit_handler(0, SHUTDOWN_FAST, 0);
exit(0);
} else {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "Asterisk already running on %s. Use 'asterisk -r' to connect.\n", ast_config_AST_SOCKET);
printf("%s", term_quit());
exit(1);
}
} else if (ast_opt_remote || ast_opt_exec) {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "Unable to connect to remote asterisk (does %s exist?)\n", ast_config_AST_SOCKET);
printf("%s", term_quit());
exit(1);
}
#ifdef HAVE_CAP
child_cap = cap_from_text("cap_net_admin-eip");
#endif
/* Not a remote console? Start the daemon. */
asterisk_daemon(isroot, runuser, rungroup);
#ifdef HAS_CAP
cap_free(child_cap);
#endif
return 0;
}
static inline void check_init(int init_result, const char *name)
{
if (init_result) {
if (ast_is_logger_initialized()) {
ast_log(LOG_ERROR, "%s initialization failed. ASTERISK EXITING!\n%s", name, term_quit());
} else {
fprintf(stderr, "%s initialization failed. ASTERISK EXITING!\n%s", name, term_quit());
}
ast_run_atexits(0);
exit(init_result == -2 ? 2 : 1);
}
}
static void asterisk_daemon(int isroot, const char *runuser, const char *rungroup)
{
FILE *f;
sigset_t sigs;
int num;
char *buf;
char pbx_uuid[AST_UUID_STR_LEN];
/* Set time as soon as possible */
ast_lastreloadtime = ast_startuptime = ast_tvnow();
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
/* This needs to remain as high up in the initial start up as possible.
* daemon causes a fork to occur, which has all sorts of unintended
* consequences for things that interact with threads. This call *must*
* occur before anything in Asterisk spawns or manipulates thread related
* primitives. */
#if HAVE_WORKING_FORK
if (ast_opt_always_fork || !ast_opt_no_fork) {
#ifndef HAVE_SBIN_LAUNCHD
if (daemon(1, 0) < 0) {
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "daemon() failed: %s\n", strerror(errno));
} else {
ast_mainpid = getpid();
}
#else
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fprintf(stderr, "Mac OS X detected. Use 'launchctl load /Library/LaunchDaemon/org.asterisk.asterisk.plist'.\n");
#endif
}
#endif
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
/* At this point everything has been forked successfully,
* we have determined that we aren't attempting to connect to
* an Asterisk instance, and that there isn't one already running. */
multi_thread_safe = 1;
load_astmm_phase_1();
/* Check whether high prio was successfully set by us or some
* other incantation. */
if (has_priority()) {
ast_set_flag(&ast_options, AST_OPT_FLAG_HIGH_PRIORITY);
} else {
ast_clear_flag(&ast_options, AST_OPT_FLAG_HIGH_PRIORITY);
}
/* Spawning of astcanary must happen AFTER the call to daemon(3) */
if (ast_opt_high_priority) {
snprintf(canary_filename, sizeof(canary_filename), "%s/alt.asterisk.canary.tweet.tweet.tweet", ast_config_AST_RUN_DIR);
/* Don't let the canary child kill Asterisk, if it dies immediately */
sigaction(SIGPIPE, &ignore_sig_handler, NULL);
canary_pid = fork();
if (canary_pid == 0) {
char canary_binary[PATH_MAX], ppid[12];
/* Reset signal handler */
signal(SIGCHLD, SIG_DFL);
signal(SIGPIPE, SIG_DFL);
ast_close_fds_above_n(0);
ast_set_priority(0);
snprintf(ppid, sizeof(ppid), "%d", (int) ast_mainpid);
/* Use the astcanary binary that we installed */
snprintf(canary_binary, sizeof(canary_binary), "%s/astcanary", ast_config_AST_SBIN_DIR);
execl(canary_binary, "astcanary", canary_filename, ppid, (char *)NULL);
/* Should never happen */
_exit(1);
} else if (canary_pid > 0) {
pthread_t dont_care;
ast_pthread_create_detached(&dont_care, NULL, canary_thread, NULL);
}
/* Kill the canary when we exit */
ast_register_atexit(canary_exit);
}
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
/* Blindly write the PID file. */
unlink(ast_config_AST_PID);
f = fopen(ast_config_AST_PID, "w");
if (f) {
fprintf(f, "%ld\n", (long)ast_mainpid);
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
fclose(f);
} else {
fprintf(stderr, "Unable to open pid file '%s': %s\n", ast_config_AST_PID, strerror(errno));
}
/* Initialize the terminal. Since all processes have been forked,
* we can now start using the standard log messages.
*/
ast_term_init();
printf("%s", term_end());
fflush(stdout);
print_intro_message(runuser, rungroup);
register_config_cli();
check_init(astobj2_init(), "AO2");
check_init(ast_named_locks_init(), "Named Locks");
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
if (ast_opt_console) {
if (el_hist == NULL || el == NULL)
ast_el_initialize();
ast_el_read_default_histfile();
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
}
#ifdef AST_XML_DOCS
/* Load XML documentation. */
ast_xmldoc_load_documentation();
#endif
check_init(astdb_init(), "ASTdb");
ast_uuid_init();
if (ast_pbx_uuid_get(pbx_uuid, sizeof(pbx_uuid))) {
ast_uuid_generate_str(pbx_uuid, sizeof(pbx_uuid));
ast_db_put("pbx", "UUID", pbx_uuid);
}
ast_verb(0, "PBX UUID: %s\n", pbx_uuid);
check_init(ast_json_init(), "libjansson");
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
ast_ulaw_init();
ast_alaw_init();
ast_utf8_init();
Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will *greatly* improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 376441 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-11-18 20:27:45 +00:00
tdd_init();
callerid_init();
ast_builtins_init();
check_init(ast_utils_init(), "Utilities");
check_init(ast_tps_init(), "Task Processor Core");
check_init(ast_fd_init(), "File Descriptor Debugging");
check_init(ast_pbx_init(), "ast_pbx_init");
check_init(aco_init(), "Configuration Option Framework");
check_init(stasis_init(), "Stasis");
#ifdef TEST_FRAMEWORK
check_init(ast_test_init(), "Test Framework");
#endif
check_init(ast_translate_init(), "Translator Core");
ast_aoc_cli_init();
check_init(ast_sorcery_init(), "Sorcery");
check_init(ast_codec_init(), "Codecs");
check_init(ast_format_init(), "Formats");
check_init(ast_format_cache_init(), "Format Cache");
check_init(ast_codec_builtin_init(), "Built-in Codecs");
check_init(ast_bucket_init(), "Bucket API");
check_init(ast_stasis_system_init(), "Stasis system-level information");
check_init(ast_endpoint_stasis_init(), "Stasis Endpoint");
Initial support for endpoints. An endpoint is an external device/system that may offer/accept channels to/from Asterisk. While this is a very useful concept for end users, it is surprisingly not a core concept within Asterisk itself. This patch defines ast_endpoint as a separate object, which channel drivers may use to expose their concept of an endpoint. As the channel driver creates channels, it can use ast_endpoint_add_channel() to associate channels to the endpoint. This updated the endpoint appropriately, and forwards all of the channel's events to the endpoint's topic. In order to avoid excessive locking on the endpoint object itself, the mutable state is not accessible via getters. Instead, you can create a snapshot using ast_endpoint_snapshot_create() to get a consistent snapshot of the internal state. This patch also includes a set of topics and messages associated with endpoints, and implementations of the endpoint-related RESTful API. chan_sip was updated to create endpoints with SIP peers, but the state of the endpoints is not updated with the state of the peer. Along for the ride in this patch is a Stasis test API. This is a stasis_message_sink object, which can be subscribed to a Stasis topic. It has functions for blocking while waiting for conditions in the message sink to be fulfilled. (closes issue ASTERISK-21421) Review: https://reviewboard.asterisk.org/r/2492/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387932 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-05-08 13:39:08 +00:00
ast_makesocket();
/* GCC 4.9 gives a bogus "right-hand operand of comma expression has
* no effect" warning */
(void) sigemptyset(&sigs);
(void) sigaddset(&sigs, SIGHUP);
(void) sigaddset(&sigs, SIGTERM);
(void) sigaddset(&sigs, SIGINT);
(void) sigaddset(&sigs, SIGPIPE);
(void) sigaddset(&sigs, SIGWINCH);
pthread_sigmask(SIG_BLOCK, &sigs, NULL);
sigaction(SIGURG, &urg_handler, NULL);
signal(SIGINT, __quit_handler);
signal(SIGTERM, __quit_handler);
sigaction(SIGHUP, &hup_handler, NULL);
sigaction(SIGPIPE, &ignore_sig_handler, NULL);
/* ensure that the random number generators are seeded with a different value every time
Asterisk is started
*/
srand((unsigned int) getpid() + (unsigned int) time(NULL));
initstate((unsigned int) getpid() * 65536 + (unsigned int) time(NULL), randompool, sizeof(randompool));
threadstorage_init();
check_init(init_logger(), "Logger");
check_init(ast_rtp_engine_init(), "RTP Engine");
3) In addition to merging the changes below, change trunk back to a regular LIST instead of an RWLIST. The way this list works makes it such that a RWLIST provides no additional benefit. Also, a mutex is needed for use with the thread condition. Merged revisions 105563 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105563 | russell | 2008-03-03 09:50:43 -0600 (Mon, 03 Mar 2008) | 24 lines Merge in some changes from team/russell/autoservice-nochans-1.4 These changes fix up some dubious code that I came across while auditing what happens in the autoservice thread when there are no channels currently in autoservice. 1) Change it so that autoservice thread doesn't keep looping around calling ast_waitfor_n() on 0 channels twice a second. Instead, use a thread condition so that the thread properly goes to sleep and does not wake up until a channel is put into autoservice. This actually fixes an interesting bug, as well. If the autoservice thread is already running (almost always is the case), then when the thread goes from having 0 channels to have 1 channel to autoservice, that channel would have to wait for up to 1/2 of a second to have the first frame read from it. 2) Fix up the code in ast_waitfor_nandfds() for when it gets called with no channels and no fds to poll() on, such as was the case with the previous code for the autoservice thread. In this case, the code would call alloca(0), and pass the result as the first argument to poll(). In this case, the 2nd argument to poll() specified that there were no fds, so this invalid pointer shouldn't actually get dereferenced, but, this code makes it explicit and ensures the pointers are NULL unless we have valid data to put there. (related to issue #12116) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@105564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-03 15:59:50 +00:00
ast_autoservice_init();
check_init(ast_timing_init(), "Timing");
check_init(ast_ssl_init(), "SSL");
PJPROJECT logging: Made easier to get available logging levels. Use of the new logging is as simple as issuing the new CLI command or setting the new pjproject.conf option. Other options that can affect the logging are how you have the pjproject log levels mapped to Asterisk log types in pjproject.conf and if you have configured Asterisk to log the DEBUG type messages. Altering the pjproject.conf level mapping shouldn't be necessary for most installations as the default mapping is sensible. Configuring Asterisk to log the DEBUG message type is standard practice for collecting debug information. * Added CLI "pjproject set log level" command to dynamically adjust the maximum pjproject log message level. * Added CLI "pjproject show log level" command to see the currently set maximum pjproject log message level. * Added pjproject.conf startup section "log_level" option to set the initial maximum pjproject log message level so all messages could be captured from initialization. * Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into bundled pjproject. Pjproject will use the currently set run time log level to determine if a log message is generated just like Asterisk verbose and debug logging levels. * In log_forwarder(), made always log enabled and mapped pjproject log messages. DEBUG mapped log messages are no longer gated by the current Asterisk debug logging level. * Removed RAII_VAR() from res_pjproject.c:get_log_level(). ASTERISK-26630 #close Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389
2016-11-24 00:27:54 +00:00
read_pjproject_startup_options();
check_init(ast_pj_init(), "Embedded PJProject");
check_init(app_init(), "App Core");
check_init(mwi_init(), "MWI Core");
check_init(devstate_init(), "Device State Core");
check_init(ast_msg_init(), "Messaging API");
check_init(ast_channels_init(), "Channel");
check_init(ast_endpoint_init(), "Endpoints");
check_init(ast_pickup_init(), "Call Pickup");
check_init(ast_bridging_init(), "Bridging");
check_init(ast_parking_stasis_init(), "Parking Core");
check_init(ast_device_state_engine_init(), "Device State Engine");
check_init(ast_presence_state_engine_init(), "Presence State Engine");
check_init(dns_core_init(), "DNS Resolver Core");
check_init(ast_dns_system_resolver_init(), "Default DNS resolver");
check_init(ast_security_stasis_init(), "Security Stasis Topic and Events");
check_init(ast_image_init(), "Image");
check_init(ast_file_init(), "Generic File Format Support");
check_init(load_pbx(), "load_pbx");
check_init(load_pbx_builtins(), "Builtin PBX Applications");
check_init(load_pbx_functions_cli(), "PBX Functions Support");
check_init(load_pbx_variables(), "PBX Variables Support");
check_init(load_pbx_switch(), "PBX Switch Support");
check_init(load_pbx_app(), "PBX Application Support");
check_init(load_pbx_hangup_handler(), "PBX Hangup Handler Support");
check_init(ast_local_init(), "Local Proxy Channel Driver");
check_init(ast_refer_init(), "Refer API");
/* We should avoid most config loads before this point as they can't use realtime. */
check_init(load_modules(), "Module");
media cache: Add a core API and facade for a backend agnostic media cache This patch adds a new API to the Asterisk core that acts as a media cache. The core API itself is mostly a thin wrapper around some bucket API provided implementation that itself acts as the mechanism of retrieval for media. The media cache API in the core provides the following: * A very thin in-memory cache of the active bucket_file items. Unlike a more traditional cache, it provides no expiration mechanisms. Most queries that hit the in-memory cache will also call into the bucket implementations as well. The bucket implementations are responsible for determining whether or not the active record is active and valid. This makes sense for the most likely implementation of a media cache backend, i.e., HTTP. The HTTP layer itself is the actual arbiter of whether or not a record is truly active; as such, the in-memory cache in the core has to defer to it. * The ability to create new items in the media cache from local resources. This allows for re-creation of items in the cache on restart. * Synchronization of items in the media cache to the AstDB. This also includes various pieces of important metadata. The API provides sufficient access that higher level APIs, such as the file or app APIs, do not have to worry about the semantics of the bucket APIs when needing to playback a resource. In addition, this patch provides unit tests for the media cache API. The unit tests use a fake bucket backend to verify correctness. Change-Id: I11227abbf14d8929eeb140ddd101dd5c3820391e
2015-01-29 14:38:23 +00:00
/*
* This has to load after the dynamic modules load, as items in the media
* cache can't be constructed from items in the AstDB without their
* bucket backends.
*/
check_init(ast_media_cache_init(), "Media Cache");
media cache: Add a core API and facade for a backend agnostic media cache This patch adds a new API to the Asterisk core that acts as a media cache. The core API itself is mostly a thin wrapper around some bucket API provided implementation that itself acts as the mechanism of retrieval for media. The media cache API in the core provides the following: * A very thin in-memory cache of the active bucket_file items. Unlike a more traditional cache, it provides no expiration mechanisms. Most queries that hit the in-memory cache will also call into the bucket implementations as well. The bucket implementations are responsible for determining whether or not the active record is active and valid. This makes sense for the most likely implementation of a media cache backend, i.e., HTTP. The HTTP layer itself is the actual arbiter of whether or not a record is truly active; as such, the in-memory cache in the core has to defer to it. * The ability to create new items in the media cache from local resources. This allows for re-creation of items in the cache on restart. * Synchronization of items in the media cache to the AstDB. This also includes various pieces of important metadata. The API provides sufficient access that higher level APIs, such as the file or app APIs, do not have to worry about the semantics of the bucket APIs when needing to playback a resource. In addition, this patch provides unit tests for the media cache API. The unit tests use a fake bucket backend to verify correctness. Change-Id: I11227abbf14d8929eeb140ddd101dd5c3820391e
2015-01-29 14:38:23 +00:00
/* loads the cli_permissions.conf file needed to implement cli restrictions. */
ast_cli_perms_init(0);
ast_cli_channels_init(); /* Not always safe to access CLI commands until startup is complete. */
ast_stun_init();
dnsmgr_start_refresh();
if (ast_opt_no_fork) {
consolethread = pthread_self();
}
ast_alertpipe_init(sig_alert_pipe);
ast_process_pending_reloads();
ast_set_flag(&ast_options, AST_OPT_FLAG_FULLY_BOOTED);
publish_fully_booted();
Merged revisions 199022 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun 2009) | 40 lines Safely handle AMI connections/reload requests that occur during startup. During asterisk startup, a lock on the list of modules is obtained by the primary thread while each module is initialized. Issue 13778 pointed out a problem with this approach, however. Because the AMI is loaded before other modules, it is possible for a module reload to be issued by a connected client (via Action: Command), causing a deadlock. The resolution for 13778 was to move initialization of the manager to happen after the other modules had already been lodaded. While this fixed this particular issue, it caused a problem for users (like FreePBX) who call AMI scripts via an #exec in a configuration file (See issue 15189). The solution I have come up with is to defer any reload requests that come in until after the server is fully booted. When a call comes in to ast_module_reload (from wherever) before we are fully booted, the request is added to a queue of pending requests. Once we are done booting up, we then execute these deferred requests in turn. Note that I have tried to make this a bit more intelligent in that it will not queue up more than 1 request for the same module to be reloaded, and if a general reload request comes in ('module reload') the queue is flushed and we only issue a single deferred reload for the entire system. As for how this will impact existing installations - Before 13778, a reload issued before module initialization was completed would result in a deadlock. After 13778, you simply couldn't connect to the manager during startup (which causes problems with #exec-that-calls-AMI configuration files). I believe this is a good general purpose solution that won't negatively impact existing installations. (closes issue #15189) (closes issue #13778) Reported by: p_lindheimer Patches: 06032009_15189_deferred_reloads.diff uploaded by seanbright (license 71) Tested by: p_lindheimer, seanbright Review: https://reviewboard.asterisk.org/r/272/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-04 14:31:24 +00:00
pthread_sigmask(SIG_UNBLOCK, &sigs, NULL);
load_astmm_phase_2();
ast_cli_register_multiple(cli_asterisk_shutdown, ARRAY_LEN(cli_asterisk_shutdown));
ast_cli_register_multiple(cli_asterisk, ARRAY_LEN(cli_asterisk));
ast_register_cleanup(main_atexit);
run_startup_commands();
ast_sd_notify("READY=1");
Logger/CLI/etc.: Fix some aesthetic issues; reduce chatty verbose messages This patch addresses some aesthetic issues in Asterisk. These are all just minor tweaks to improve the look of the CLI when used in a variety of settings. Specifically: * A number of chatty verbose messages were removed or demoted to DEBUG messages. Verbose messages with a verbosity level of 5 or higher were - if kept as verbose messages - demoted to level 4. Several messages that were emitted at verbose level 3 were demoted to 4, as announcement of dialplan applications being executed occur at level 3 (and so the effects of those applications should generally be less). * Some verbose messages that only appear when their respective 'debug' options are enabled were bumped up to always be displayed. * Prefix/timestamping of verbose messages were moved to the verboser handlers. This was done to prevent duplication of prefixes when the timestamp option (-T) is used with the CLI. * Verbose magic is removed from messages before being emitted to non-verboser handlers. This prevents the magic in multi-line verbose messages (such as SIP debug traces or the output of DumpChan) from being written to files. * _Slightly_ better support for the "light background" option (-W) was added. This includes using ast_term_quit in the output of XML documentation help, as well as changing the "Asterisk Ready" prompt to bright green on the default background (which stands a better chance of being displayed properly than bright white). Review: https://reviewboard.asterisk.org/r/3547/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28 22:54:12 +00:00
ast_verb(0, COLORIZE_FMT "\n", COLORIZE(COLOR_BRGREEN, 0, "Asterisk Ready."));
logger_queue_start();
if (ast_opt_console) {
/* Console stuff now... */
/* Register our quit function */
char title[256];
char hostname[MAXHOSTNAMELEN] = "";
if (gethostname(hostname, sizeof(hostname) - 1)) {
ast_copy_string(hostname, "<Unknown>", sizeof(hostname));
}
ast_pthread_create_detached(&mon_sig_flags, NULL, monitor_sig_flags, NULL);
set_icon("Asterisk");
snprintf(title, sizeof(title), "Asterisk Console on '%s' (pid %ld)", hostname, (long)ast_mainpid);
set_title(title);
el_set(el, EL_GETCFN, ast_el_read_char);
for (;;) {
if (sig_flags.need_el_end) {
el_end(el);
return;
}
if (sig_flags.need_quit || sig_flags.need_quit_handler) {
quit_handler(0, SHUTDOWN_FAST, 0);
break;
}
buf = (char *) el_gets(el, &num);
if (!buf && write(1, "", 1) < 0)
return; /* quit */
if (buf) {
if (buf[strlen(buf)-1] == '\n')
buf[strlen(buf)-1] = '\0';
consolehandler(buf);
}
}
}
/* Stall until a quit signal is given */
monitor_sig_flags(NULL);
}