96 Commits (c4bf59b781d34b23138f5777bd4944918f206c98)

Author	SHA1	Message	Date
Naveen Albert	c4bf59b781	func_channel: Expose previously unsettable options. ... Certain channel options are not set anywhere or exposed in any way to users, making them unusable. This exposes some of these options which make sense for users to manipulate at runtime. Resolves: #442	1 year ago
Naveen Albert	0c70d497bc	documentation: Adds versioning information. ... Adds version information for applications, functions, and manager events/actions. This is not completely exhaustive by any means but covers most new things added that have release versioning information in the issue tracker. ASTERISK-29940 #close Change-Id: I506401e93c799715dbbe97c0a8ba18af2bf5e131	3 years ago
Naveen Albert	27fb4fd5bc	func_channel: Add lastcontext and lastexten. ... Adds the lastcontext and lastexten channel fields to allow users to access previous dialplan execution locations. ASTERISK-29840 #close Change-Id: Ib455fe300cc8e9a127686896ee2d0bd11e900307	3 years ago
Naveen Albert	5f59e0d36f	documentation: Document built-in system and channel vars ... Documentation for built-in special system and channel vars is currently outdated, and updating is a manual process since there is no XML documentation for these anywhere. This adds documentation for system vars to func_env and for channel vars to func_channel so that they appear along with the corresponding fields that would be accessed using a function. ASTERISK-29848 #close Change-Id: I6997f925c4a45fffe71321861f5898a8b7182fa9	3 years ago
Naveen Albert	de6ecd5e34	func_channel: Add CHANNEL_EXISTS function. ... Adds a function to check for the existence of a channel by name or by UNIQUEID. ASTERISK-29656 #close Change-Id: Ib464e9eb6e13dc683a846286798fecff4fd943cb	4 years ago
Sebastien Duthil	d40e343710	func_channel: allow reading 4 fields from dialplan ... The following fields return an error when read from dialplan: - exten - context - userfield - channame ASTERISK-28796 #close Change-Id: Ieacaac629490f8710fdacc9de80ed5916c5f6ee2	5 years ago
Ivan Poddubny	8bd5980e14	func_channel: Delete dead CHANNEL_TRACE code ... The functions behind the flag and the flag itself were removed from Asterisk 12 as incompatible with the new architecture. Change-Id: I058493ef7a53ee290fd225bbcbb07bf46b623ccf	7 years ago
Richard Mudgett	1d3dc9aea2	func_channel.c: Update MASTER_CHANNEL documentation ... Be more explicit in what is meant by the master channel to eliminate misunderstanding. ASTERISK-23133 Change-Id: I453bcaf4b99404a5a3e345dbf093ac6c1afcfc72	7 years ago
Richard Mudgett	3078b7adc2	CDR: Fix deadlock setting some CDR values. ... Setting channel variables with the AMI Originate action caused a deadlock when you set CDR(amaflags) or CDR(accountcode). This path has the channel locked when the CDR function is called. The CDR function then synchronously passes the job to a stasis thread. The stasis handling function then attempts to lock the channel. Deadlock results. * Avoid deadlock by making the CDR function handle setting amaflags and accountcode directly on the channel rather than passing it off to the CDR processing code under a stasis thread to do it. * Made the CHANNEL function and the CDR function process amaflags the same way. * Fixed referencing the wrong message type in cdr_prop_write(). ASTERISK-27460 Change-Id: I5eacb47586bc0b8f8ff76a19bd92d1dc38b75e8f	7 years ago
Richard Mudgett	8cb4f9cea1	CHANNEL(callid): Give dialplan access to the callid. ... * Added CHANNEL(callid) to retrieve the call identifier log tag associated with the channel. Dialplan now has access to the call log search key associated with the channel so it can be saved in case there is a problem with the call. ASTERISK-26878 Change-Id: I2c97ebd928b6f3c5bc80c5729e4d3c07f453049f	8 years ago
Corey Farrell	a6e5bae3ef	Remove ASTERISK_REGISTER_FILE. ... ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes all traces of it. Previously exported symbols removed: * __ast_register_file * __ast_unregister_file * ast_complete_source_filename This also removes the mtx_prof static variable that was declared when MTX_PROFILE was enabled. This variable was only used in lock.c so it is now initialized in that file only. ASTERISK-26480 #close Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966	9 years ago
Corey Farrell	824a4e84d1	Refactor usage pattern of xmldoc info tag. ... This updates func_channel.c and main/message.c to use a generic xpointer include instead of including info from each channel driver. Now the name attribute of info is CHANNEL or CHANNEL_EXAMPLES to be included in documentation for func_channel. Setting the name attribute of info to MessageToInfo or MessageFromInfo causes it to be included in the MessageSend application and AMI action. Change-Id: I89fd8276a3250824241a618009714267d3a8d1ea	9 years ago
Matt Jordan	ddab42e296	func_channel: Reorganize documentation ... * Following the example of the PJSIP channel driver, the channel technology specific documentation has been moved to the respective channel drivers that provide that functionality. This has the benefit of locating the documentation of items with those modules that provide it. * Examples of using the CHANNEL function for both standard items as well as for PJSIP have been added. * The 'max_forwards' standard item has been documented. Change-Id: Ifaa79a232c8ac99cf8da6ef6cc7815d398b1b79b	9 years ago
Rusty Newton	68cad96ffd	func_channel: Add help text for undocumented CHANNEL function arguments ... Adding help text documentation for: * hangupsource * appname * appdata * exten * context * channame * uniqueid * linkedid ASTERISK-24097 #close Reported by: Steven T. Wheeler Tested by: Rusty Newton Change-Id: Ib94b00568b0433987df87d5b67ea529b5905754d	9 years ago
Alexander Traud	cf79b62778	ast_format_cap_get_names: To display all formats, the buffer was increased. ... ASTERISK-25533 #close Change-Id: Ie1a9d1a6511b3f1a56b93d04475fbf8a4e40010a	10 years ago
Mark Michelson	aae45acbda	Detect potential forwarding loops based on count. ... A potential problem that can arise is the following: * Bob's phone is programmed to automatically forward to Carol. * Carol's phone is programmed to automatically forward to Bob. * Alice calls Bob. If left unchecked, this results in an endless loops of call forwards that would eventually result in some sort of fiery crash. Asterisk's method of solving this issue was to track which interfaces had been dialed. If a destination were dialed a second time, then the attempt to call that destination would fail since a loop was detected. The problem with this method is that call forwarding has evolved. Some SIP phones allow for a user to manually forward an incoming call to an ad-hoc destination. This can mean that: * There are legitimate use cases where a device may be dialed multiple times, or * There can be human error when forwarding calls. This change removes the old method of detecting forwarding loops in favor of keeping a count of the number of destinations a channel has dialed on a particular branch of a call. If the number exceeds the set number of max forwards, then the call fails. This approach has the following advantages over the old: * It is much simpler. * It can detect loops involving local channels. * It is user configurable. The only disadvantage it has is that in the case where there is a legitimate forwarding loop present, it takes longer to detect it. However, the forwarding loop is still properly detected and the call is cleaned up as it should be. Address review feedback on gerrit. * Correct "mfgium" to "Digium" * Decrement max forwards by one in the case where allocation of the max forwards datastore is required. * Remove irrelevant code change from pjsip_global_headers.c ASTERISK-24958 #close Change-Id: Ia7e4b7cd3bccfbd34d9a859838356931bba56c23	10 years ago
Matt Jordan	4a58261694	git migration: Refactor the ASTERISK_FILE_VERSION macro ... Git does not support the ability to replace a token with a version string during check-in. While it does have support for replacing a token on clone, this is somewhat sub-optimal: the token is replaced with the object hash, which is not particularly easy for human consumption. What's more, in practice, the source file version was often not terribly useful. Generally, when triaging bugs, the overall version of Asterisk is far more useful than an individual SVN version of a file. As a result, this patch removes Asterisk's support for showing source file versions. Specifically, it does the following: * Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and remove passing the version in with the macro. Other facilities than 'core show file version' make use of the file names, such as setting a debug level only on a specific file. As such, the act of registering source files with the Asterisk core still has use. The macro rename now reflects the new macro purpose. * main/asterisk: - Refactor the file_version structure to reflect that it no longer tracks a version field. - Remove the "core show file version" CLI command. Without the file version, it is no longer useful. - Remove the ast_file_version_find function. The file version is no longer tracked. - Rename ast_register_file_version/ast_unregister_file_version to ast_register_file/ast_unregister_file, respectively. * main/manager: Remove value from the Version key of the ModuleCheck Action. The actual key itself has not been removed, as doing so would absolutely constitute a backwards incompatible change. However, since the file version is no longer tracked, there is no need to attempt to include it in the Version key. * UPGRADE: Add notes for: - Modification to the ModuleCheck AMI Action - Removal of the "core show file version" CLI command Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e	10 years ago
Richard Mudgett	e4738a59eb	CHANNEL(peer), chan_iax2, res_fax, SNMP agent: Fix deadlock from reaching across a bridge. ... Calling ast_channel_bridge_peer() cannot be done while holding any channel locks. The reported issue hit the deadlock in chan_iax2, but an audit of the ast_channel_bridge_peer() calls found three more locations where the same deadlock can occur. * Made CHANNEL(peer), res_fax, and the SNMP agent not call ast_channel_bridge_peer() with any channel locked. For CHANNEL(peer) I had to rework the logic to not hold the channel lock. * Made chan_iax2 no longer call ast_channel_bridge_peer(). It was done for legacy reasons that no longer apply. * Removed the iax.conf forcejitterbuffer option. It is now always enabled when the jitterbuffer option is enabled. If you put a jitter buffer on a channel it will be on the channel. ASTERISK-24600 #close Reported by: Jeff Collell Review: https://reviewboard.asterisk.org/r/4342/ ........ Merged revisions 430817 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430819 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	9183416fe2	func_channel: Add CHANNEL(onhold) item to get the current hold status of the channel. ... It would be useful to get the current hold status of a channel. Added CHANNEL(onhold) item that returns 1 (onhold) and 0 (not-onhold) for the hold status of a channel. ASTERISK-24038 Reported by: Matt Jordan AFS-113 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3983/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	025bd1bf3f	func_channel.c: Add missing locking to some CHANNEL() requests. ... * The CHANNEL() audionativeformat, videonativeformat, audioreadformat, and audiowriteformat now need locking since the media format rework when accessing the channel's format pointers. * Increased the buffer size for CHANNEL() audionativeformat and videonativeformat output strings since the allow=all can be a lengthy list. * Tweaked the CHANNEL() XML documentation for secure_bridge_signaling, secure_bridge_media, and state. * Ensured the output buffer is initialized for secure_bridge_signaling and secure_bridge_media. * Made use the locked_copy_string() macro instead of inlining it for trace and checkhangup. ........ Merged revisions 422700 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	8610daec3b	Add documentation to the ability to retrieve the source port of a SIP call. ... (belongs with r419970) ASTERISK-24040 #close Patches: func_channel.c.diff uploaded by dtryba Review: https://reviewboard.asterisk.org/r/3781/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420144 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	a2c912e997	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	11 years ago
Kinsey Moore	abd3e4040b	Allow Asterisk to compile under GCC 4.10 ... This resolves a large number of compiler warnings from GCC 4.10 which cause the build to fail under dev mode. The vast majority are signed/unsigned mismatches in printf-style format strings. ........ Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413587 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413588 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	fbe0dfaf44	Fix dialplan function NULL channel safety issues ... (closes issue ASTERISK-23391) Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3386/ ........ Merged revisions 411313 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411314 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411315 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	ce423d2ea4	func_channel, chan_pjsip: Add CHANNEL read function support for chan_pjsip ... This patch adds CHANNEL read support for chan_pjsip. This allows the dialplan to use the CHANNEL function on a chan_pjsip channel to obtain run-time information about the channel from the PJSIP channel driver and the PJSIP stack. This includes: * RTP information, including source/destination media addresses, whether or not the media is secure, held, and other properties. * RTCP information. This includes sets of parseable information, as well as individual statistic attriutes. * PJSIP information. This includes URIs, local/remote signalling addresses, whether or not the signalling is secure, and other properties. * The endpoint name. This can be used in conjunction with the PJSIP_ENDPOINT function to obtain more detailed endpoint information. Review: https://reviewboard.asterisk.org/r/3038/ ........ Merged revisions 403618 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403619 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	f8622e7c5c	Get rid of ast_bridged_channel() and the bridged_channel field on ast_channels. ... This commit is smaller than the initial review placed on review board. This is because a change to allow for channel drivers to access parking functionality externally was committed and invalidated quite a few of the changes initially made. (closes issue ASTERISK-22039) reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2717 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396103 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	cafc115896	A great big renaming patch ... This patch renames the bridging* files to bridge*. This may seem pedantic and silly, but it fits better in line with current Asterisk naming conventions: * channel is not "channeling" * monitor is not "monitoring" etc. A bridge is an object. It is a first class citizen in Asterisk. "Bridging" is the act of using a bridge on a set of channels - and the API that fulfills that role is more than just the action. (closes issue ASTERISK-22130) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	9d8a5ceb02	Move after bridge callbacks into their own file ... One more major refactoring to go. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395367 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	f3fcf0aa2e	func_channel: dtmf_features setting ... Allows reading andsetting dtmf features via a channel function CHANNEL(dtmf_features) (closes issue ASTERISK-21876) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2648/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395215 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	88e344d2df	Clean up documentation ... This patch cleans up documentation in func_channel for the following items: * rtpsource * secure_signaling * secure_media * various OOH323 parameters (closes issue ASTERISK-20969) Reported by: snuffy patches: func_chan-update.diff uploaded by snuffy (License 5024) ........ Merged revisions 394980 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 394981 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394982 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	d014ad2261	func_channel: Read/Write after_bridge_goto option ... Allows reading and setting of a channel's after_bridge_goto datastore (closes issue ASTERISK-21875) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2628/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393005 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	6258bbe7bd	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	12 years ago
Richard Mudgett	3d63833bd6	Merge in the bridge_construction branch to make the system use the Bridging API. ... Breaks many things until they can be reworked. A partial list: chan_agent chan_dahdi, chan_misdn, chan_iax2 native bridging app_queue COLP updates DTMF attended transfers Protocol attended transfers git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	1cafccdf87	Update documentation for CHANNEL function ... Document that you can read/write the 'accountcode' and 'amaflags' on a channel. ........ Merged revisions 384640 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 384641 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384642 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	da5944fc56	Named call pickup groups. Fixes, missing functionality, and improvements. ... * ASTERISK-20383 Missing named call pickup group features: CHANNEL(callgroup) - Need CHANNEL(namedcallgroup) CHANNEL(pickupgroup) - Need CHANNEL(namedpickupgroup) Pickup() - Needs to also select from named pickup groups. * ASTERISK-20384 Using the pickupexten, the pickup channel selection could fail even though there was a call it could have picked up. In a call pickup race when there are multiple calls to pickup and two extensions try to pickup a call, it is conceivable that the loser will not pick up any call even though it could have picked up the next oldest matching call. Regression because of the named call pickup group feature. * See ASTERISK-20386 for the implementation improvements. These are the changes in channel.c and channel.h. * Fixed some locking issues in CHANNEL(). (closes issue ASTERISK-20383) Reported by: rmudgett (closes issue ASTERISK-20384) Reported by: rmudgett (closes issue ASTERISK-20386) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/2112/ ........ Merged revisions 373220 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373221 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	9b16c8b0f6	Clean up and ensure proper usage of alloca() ... This replaces all calls to alloca() with ast_alloca() which calls gcc's __builtin_alloca() to avoid BSD semantics and removes all NULL checks on memory allocated via ast_alloca() and ast_strdupa(). (closes issue ASTERISK-20125) Review: https://reviewboard.asterisk.org/r/2032/ Patch-by: Walter Doekes (wdoekes) ........ Merged revisions 370642 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370643 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	ac35b92b62	Hangup handlers - Dialplan subroutines that run when the channel hangs up. ... Hangup handlers are an alternative to the h extension. They can be used in addition to the h extension. The idea is to attach a Gosub routine to a channel that will execute when the call hangs up. Whereas which h extension gets executed depends on the location of dialplan execution when the call hangs up, hangup handlers are attached to the call channel. You can attach multiple handlers that will execute in the order of most recently added first. (closes issue ASTERISK-19549) Reported by: Mark Murawski Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/2002/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369493 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	a7a3050de9	Add documentation to function CHANNEL for options echocan_mode and buffers ... The ability to set "echocan_mode" and "buffers" through the dialplan was added to chan_dahdi some time ago. This patch adds some documentation to func_channel. (Closes issue ASTERISK-19911) Reported by: Dale Noll Tested by: Michael L. Young Patches: asterisk-19911-branch18.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/1949/ ........ Merged revisions 368092 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 368093 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368094 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	a485f44022	Add missing newlines to CLI logging ... ........ Merged revisions 361471 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361472 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361476 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	0e5c761c28	Opaquify ast_channel typedefs, fd arrays, and softhangup flag ... Review: https://reviewboard.asterisk.org/r/1784/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	ebaf59a656	Opaquification for ast_format structs in struct ast_channel ... Review: https://reviewboard.asterisk.org/r/1770/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356573 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	57f42bd74f	ast_channel opaquification of pointers and integral types ... Review: https://reviewboard.asterisk.org/r/1753/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	34c55e8e7c	Opaquify char * and char[] in ast_channel ... Review: https://reviewboard.asterisk.org/r/1733/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354968 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	99cae5b750	Opaquify channel stringfields ... Continue channel opaque-ification by wrapping all of the stringfields. Eventually, we will restrict what can actually set these variables, but the purpose for now is to hide the implementation and keep people from adding code that directly accesses the channel structure. Semantic changes will follow afterward. Review: https://reviewboard.asterisk.org/r/1661/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Terry Wilson	04da92c379	Replace direct access to channel name with accessor functions ... There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Gregory Nietsky	fca9962766	Merged revisions 338995 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/10 ........ r338995 | irroot | 2011-10-03 16:21:40 +0200 (Mon, 03 Oct 2011) | 6 lines Remove the channel function OOH323() and place its options into CHANNEL() channel drivers should not have there own dialplan functions. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@338996 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	a525edea59	Merged revisions 328247 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	e789eb8b2d	CHANNEL(pickupgroup) ... Allow Setting / Reading the pickupgroup of a channel with func_channel.c (closes issue #19045) Reported by: irroot Review: https://reviewboard.asterisk.org/r/1148/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@320772 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	fd2f95b3c4	Merged revisions 314206 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314206 | lmadsen | 2011-04-19 09:28:15 -0500 (Tue, 19 Apr 2011) | 14 lines Merged revisions 314205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r314205 | lmadsen | 2011-04-19 09:27:50 -0500 (Tue, 19 Apr 2011) | 6 lines Remove duplicate documentation from func_channel.c (closes issue #18970) Reported by: IgorG Patches: func_channel.c.doc.diff uploaded by IgorG (license 20) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314207 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	928ec2b990	Merged revisions 309445 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r309445 | rmudgett | 2011-03-04 09:22:04 -0600 (Fri, 04 Mar 2011) | 46 lines Get real channel of a DAHDI call. Starting with Asterisk v1.8, the DAHDI channel name format was changed for ISDN calls to: DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number> There were several reasons that the channel name had to change. 1) Call completion requires a device state for ISDN phones. The generic device state uses the channel name. 2) Calls do not necessarily have B channels. Calls placed on hold by an ISDN phone do not have B channels. 3) The B channel a call initially requests may not be the B channel the call ultimately uses. Changes to the internal implementation of the Asterisk master channel list caused deadlock problems for chan_dahdi if it needed to change the channel name. Chan_dahdi no longer changes the channel name. 4) DTMF attended transfers now work with ISDN phones because the channel name is "dialable" like the chan_sip channel names. For various reasons, some people need to know which B channel a DAHDI call is using. * Added CHANNEL(dahdi_span), CHANNEL(dahdi_channel), and CHANNEL(dahdi_type) so the dialplan can determine the B channel currently in use by the channel. Use CHANNEL(no_media_path) to determine if the channel even has a B channel. * Added AMI event DAHDIChannel to associate a DAHDI channel with an Asterisk channel so AMI applications can passively determine the B channel currently in use. Calls with "no-media" as the DAHDIChannel do not have an associated B channel. No-media calls are either on hold or call-waiting. (closes issue #17683) Reported by: mrwho Tested by: rmudgett (closes issue #18603) Reported by: arjankroon Patches: issue17683_18603_v1.8_v2.patch uploaded by rmudgett (license 664) Tested by: stever28, rmudgett ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@309446 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago