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1.4.27.1
1.4.28
1.4.28-rc1
1.4.29
1.4.29-rc1
1.4.29.1
1.4.3
1.4.30
1.4.30-rc1
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1.4.30-rc3
1.4.31
1.4.31-rc1
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1.4.32
1.4.32-rc1
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1.4.33
1.4.33-rc1
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1.4.34
1.4.34-rc1
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1.4.35
1.4.35-rc1
1.4.36
1.4.36-rc1
1.4.37
1.4.37-rc1
1.4.37.1
1.4.38
1.4.38-rc1
1.4.38.1
1.4.39
1.4.39-rc1
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1.4.39.2
1.4.4
1.4.40
1.4.40-rc1
1.4.40-rc2
1.4.40-rc3
1.4.40.1
1.4.40.2
1.4.41
1.4.41-rc1
1.4.41.1
1.4.41.2
1.4.42
1.4.42-rc1
1.4.42-rc2
1.4.43
1.4.44
1.4.5
1.4.6
1.4.7
1.4.7.1
1.4.8
1.4.9
1.6.0
1.6.0-beta1
1.6.0-beta2
1.6.0-beta3
1.6.0-beta4
1.6.0-beta5
1.6.0-beta6
1.6.0-beta7
1.6.0-beta7.1
1.6.0-beta8
1.6.0-beta9
1.6.0-rc1
1.6.0-rc2
1.6.0-rc3
1.6.0-rc4
1.6.0-rc5
1.6.0-rc6
1.6.0.1
1.6.0.10
1.6.0.11-rc1
1.6.0.11-rc2
1.6.0.12
1.6.0.13
1.6.0.13-rc1
1.6.0.14
1.6.0.14-rc1
1.6.0.15
1.6.0.16
1.6.0.16-rc1
1.6.0.16-rc2
1.6.0.17
1.6.0.18
1.6.0.18-rc1
1.6.0.18-rc2
1.6.0.18-rc3
1.6.0.19
1.6.0.2
1.6.0.20
1.6.0.20-rc1
1.6.0.21
1.6.0.21-rc1
1.6.0.22
1.6.0.23
1.6.0.23-rc1
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1.6.0.24
1.6.0.25
1.6.0.26
1.6.0.26-rc1
1.6.0.27
1.6.0.27-rc1
1.6.0.27-rc2
1.6.0.27-rc3
1.6.0.28
1.6.0.28-rc1
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1.6.0.3
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1.6.0.3.1
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1.6.0.5
1.6.0.6
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1.8.1
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1.8.7.1
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1.8.8.0-rc2
1.8.8.0-rc3
1.8.8.0-rc4
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1.8.8.1
1.8.8.2
1.8.9.0
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1.8.9.1
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10.0.0
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10.1.0
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10.1.1
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10.1.3
10.10.0
10.10.0-digiumphones
10.10.0-digiumphones-rc1
10.10.0-digiumphones-rc2
10.10.0-rc1
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10.10.1
10.10.1-digiumphones
10.11.0
10.11.0-digiumphones
10.11.0-digiumphones-rc1
10.11.0-digiumphones-rc2
10.11.0-digiumphones-rc3
10.11.0-rc1
10.11.0-rc2
10.11.0-rc3
10.11.1
10.11.1-digiumphones
10.12.0
10.12.0-digiumphones
10.12.0-digiumphones-rc1
10.12.0-digiumphones-rc2
10.12.0-rc1
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10.12.1
10.12.1-digiumphones
10.12.2
10.12.2-digiumphones
10.12.3
10.12.3-digiumphones
10.12.4
10.12.4-digiumphones
10.2.0
10.2.0-rc1
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10.2.0-rc3
10.2.0-rc4
10.2.1
10.3.0
10.3.0-rc1
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10.3.0-rc3
10.3.1
10.4.0
10.4.0-digiumphones-rc1
10.4.0-digiumphones-rc2
10.4.0-rc1
10.4.0-rc2
10.4.0-rc3
10.4.1
10.4.2
10.5.0
10.5.0-digiumphones
10.5.0-digiumphones-rc1
10.5.0-digiumphones-rc2
10.5.0-rc1
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10.5.1
10.5.1-digiumphones
10.5.2
10.5.2-digiumphones
10.6.0
10.6.0-digiumphones
10.6.0-digiumphones-rc1
10.6.0-digiumphones-rc2
10.6.0-rc1
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10.6.1
10.6.1-digiumphones
10.7.0
10.7.0-digiumphones
10.7.0-digiumphones-rc1
10.7.0-rc1
10.7.1
10.7.1-digiumphones
10.8.0
10.8.0-digiumphones
10.8.0-digiumphones-rc1
10.8.0-digiumphones-rc2
10.8.0-rc1
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10.9.0
10.9.0-digiumphones
10.9.0-digiumphones-rc1
10.9.0-digiumphones-rc2
10.9.0-digiumphones-rc3
10.9.0-rc1
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11.0.0
11.0.0-beta1
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11.0.1
11.0.2
11.1.0
11.1.0-rc1
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11.1.0-rc3
11.1.1
11.1.2
11.10.0
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11.11.0
11.11.0-rc1
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11.13.1
11.14.0
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11.14.1
11.14.2
11.15.0
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11.15.1
11.16.0
11.16.0-rc1
11.17.0
11.17.0-rc1
11.17.1
11.18.0-rc1
11.19.0-rc1
11.2.0
11.2.0-rc1
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11.2.1
11.2.2
11.20.0-rc1
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11.20.0-rc3
11.21.0
11.21.0-rc1
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11.21.1
11.22.0-rc1
11.3.0
11.3.0-rc1
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11.4.0
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11.4.0-rc3
11.5.0
11.5.0-rc1
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11.5.1
11.6-cert11
11.6.0
11.6.0-rc1
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11.6.1
11.7.0
11.7.0-rc1
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11.8.0
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11.8.0-rc3
11.8.1
11.9.0
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12.0.0
12.0.0-alpha1
12.0.0-alpha2
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12.1.0
12.1.0-rc1
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12.1.0-rc3
12.1.1
12.2.0
12.2.0-rc1
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12.2.0-rc3
12.3.0
12.3.0-rc1
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12.3.1
12.3.2
12.4.0
12.4.0-rc1
12.5.0
12.5.0-rc1
12.5.1
12.6.0
12.6.0-rc1
12.6.1
12.7.0
12.7.0-rc1
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12.7.1
12.7.2
12.8.0
12.8.0-rc1
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12.8.1
12.8.2
13.0.0
13.0.0-beta1
13.0.0-beta2
13.0.0-beta3
13.0.1
13.0.2
13.1-cert2
13.1.0
13.1.0-rc1
13.1.0-rc2
13.1.1
13.2.0
13.2.0-rc1
13.2.1
13.3.0
13.3.0-rc1
13.3.1
13.3.2
13.4.0-rc1
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13.7.0
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13.7.1
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certified/1.8.11-cert1
certified/1.8.11-cert10
certified/1.8.11-cert2
certified/1.8.11-cert3-rc1
certified/1.8.11-cert3-rc2
certified/1.8.11-cert4
certified/1.8.11-cert5
certified/1.8.11-cert5-rc1
certified/1.8.11-cert5-rc2
certified/1.8.11-cert6
certified/1.8.11-cert7
certified/1.8.11-cert8
certified/1.8.11-cert9
certified/1.8.11-cert9-rc1
certified/1.8.15-cert1
certified/1.8.15-cert1-rc1
certified/1.8.15-cert1-rc2
certified/1.8.15-cert1-rc3
certified/1.8.15-cert2
certified/1.8.15-cert3
certified/1.8.15-cert4
certified/1.8.15-cert5
certified/1.8.15-cert6
certified/1.8.15-cert7
certified/1.8.28-cert1
certified/1.8.28-cert1-rc1
certified/1.8.28-cert2
certified/1.8.28-cert3
certified/1.8.28-cert4
certified/1.8.28-cert5
certified/1.8.6-cert1
certified/11.2-cert1
certified/11.2-cert1-rc1
certified/11.2-cert1-rc2
certified/11.2-cert2
certified/11.2-cert3
certified/11.6-cert1
certified/11.6-cert1-rc1
certified/11.6-cert1-rc2
certified/11.6-cert10
certified/11.6-cert11
certified/11.6-cert12
certified/11.6-cert2
certified/11.6-cert3
certified/11.6-cert4
certified/11.6-cert5
certified/11.6-cert6
certified/11.6-cert7
certified/11.6-cert8
certified/11.6-cert9
certified/13.1-cert1
certified/13.1-cert1-rc1
certified/13.1-cert1-rc2
certified/13.1-cert1-rc3
certified/13.1-cert2
certified/13.1-cert3
${ noResults }
147 Commits (0933f0cf961333452cc317d6a1d6ec2db2c1e56b)
Author | SHA1 | Message | Date |
---|---|---|---|
|
cf79b62778 |
ast_format_cap_get_names: To display all formats, the buffer was increased.
ASTERISK-25533 #close Change-Id: Ie1a9d1a6511b3f1a56b93d04475fbf8a4e40010a |
10 years ago |
|
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 |
|
8a69aedd17 |
Fix building chan_phone on big endian systems
A left over from the formats conversion (Corey Farrell). ASTERISK-24458 #close Review: https://reviewboard.asterisk.org/r/4117/ ........ Merged revisions 426570 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426573 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
dcf1ad14da |
Add module support level to ast_module_info structure. Print it in CLI "module show" .
ASTERISK-23919 #close Reported by Malcolm Davenport Review: https://reviewboard.asterisk.org/r/3802 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
|
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 |
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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 |
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80ef9a21b9 |
uniqueid: channel linkedid, ami, ari object creation with id's
Much needed was a way to assign id to objects on creation, and much change was necessary to accomplish it. Channel uniqueids and linkedids are split into separate string and creation time components without breaking linkedid propgation. This allowed the uniqueid to be specified by the user interface - and those values are now carried through to channel creation, adding the assignedids value to every function in the chain including the channel drivers. For local channels, the second channel can be specified or left to default to a ;2 suffix of first. In ARI, bridge, playback, and snoop objects can also be created with a specified uniqueid. Along the way, the args order to allocating channels was fixed in chan_mgcp and chan_gtalk, and linkedid is no longer lost as masquerade occurs. (closes issue ASTERISK-23120) Review: https://reviewboard.asterisk.org/r/3191/ ........ Merged revisions 410157 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410158 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
11 years ago |
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28c0cb28d0 |
channel locking: Add locking for channel snapshot creation
Original commit message by mmichelson (asterisk 12 r403311): "This adds channel locks around calls to create channel snapshots as well as other functions which operate on a channel and then end up creating a channel snapshot. Functions that expect the channel to be locked prior to being called have had their documentation updated to indicate such." The above was initially committed and then reverted at r403398. The problem was found to be in core_local.c in the publish_local_bridge_message function. The ast_unreal_lock_all function locks and adds a reference to the returned channels and while they were being unlocked they were not being unreffed when no longer needed. Fixed by unreffing the channels. Also in bridge.c a lock was obtained on "other->chan", but then an attempt was made to unlock "other" and not the previously locked channel. Fixed by unlocking "other->chan" (closes issue ASTERISK-22709) Reported by: John Bigelow ........ Merged revisions 404237 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404260 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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e2630fcd51 |
channels: Return allocated channels locked.
This change makes ast_channel_alloc return allocated channels locked. By doing so no other thread can acquire, lock, and manipulate the channel before it is completely set up. (closes issue AST-1256) Review: https://reviewboard.asterisk.org/r/3067/ ........ Merged revisions 404204 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404210 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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1212906351 |
Reverting r403311. It's causing ARI tests to hang.
........ Merged revisions 403398 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403404 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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8e8b329e14 |
Add channel locking for channel snapshot creation.
This adds channel locks around calls to create channel snapshots as well as other functions which operate on a channel and then end up creating a channel snapshot. Functions that expect the channel to be locked prior to being called have had their documentation updated to indicate such. ........ Merged revisions 403311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403314 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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ee21eee7e0 |
Cache string values of formats on ast_format_cap() to save processing.
Channel snapshots have string representations of the channel's native formats. Prior to this change, the format strings were re-created on ever channel snapshot creation. Since channel native formats rarely change, this was very wasteful. Now, string representations of formats may optionally be stored on the ast_format_cap for cases where string representations may be requested frequently. When formats are altered, the string cache is marked as invalid. When strings are requested, the cache validity is checked. If the cache is valid, then the cached strings are copied. If the cache is invalid, then the string cache is rebuilt and copied, and the cache is marked as being valid again. Review: https://reviewboard.asterisk.org/r/2879 ........ Merged revisions 400356 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400363 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
12 years ago |
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3820f1586e |
Doxygen Updates - Title update
Update and extend the configuration_file group and enable linking. Update title that was left behind many years ago. (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375006 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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b5a6de76fc |
Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE)
This is the starting point for the Asterisk 11: Who Hung Up work and provides a framework which will allow channel drivers to report the types of hangup cause information available in SIP_CAUSE without incurring the overhead of the MASTER_CHANNEL dialplan function. The initial implementation only includes cause generation for chan_sip and does not include cause code translation utilities. This change deprecates SIP_CAUSE and replaces its method of reporting cause codes with the new framework. This change also deprecates the 'storesipcause' option in sip.conf. Review: https://reviewboard.asterisk.org/r/1822/ (Closes issue SWP-4221) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366408 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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786f5898d1 |
Finalize ast_channel opaquification
Review: https://reviewboard.asterisk.org/r/1786/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358907 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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99bd5b1e2e |
Eliminate a bunch of shadow warnings.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358647 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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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 |
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a9d607a357 |
Opaquify ast_channel structs and lists
Review: https://reviewboard.asterisk.org/r/1773/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357542 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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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 |
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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 |
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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 |
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23bc964e1c |
Constify some more channel driver technology callback parameters.
Review: https://reviewboard.asterisk.org/r/1707/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353685 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
13 years ago |
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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 |
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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 |
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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 |
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37aa52fd78 |
Merged revisions 316265 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316265 | russell | 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) | 5 lines Fix a bunch of compiler warnings generated by gcc 4.6.0. Most of these are -Wunused-but-set-variable, but there were a few others mixed in here, as well. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@316293 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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64ed1ba3e9 |
Fixes compile error in chan_phone for big endian
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@307927 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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c26c190711 |
Asterisk media architecture conversion - no more format bitfields
This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
14 years ago |
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5eae9f44f7 |
Merged revisions 284597 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284597 | tilghman | 2010-09-02 00:00:34 -0500 (Thu, 02 Sep 2010) | 29 lines Merged revisions 284593,284595 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r284593 | tilghman | 2010-09-01 17:59:50 -0500 (Wed, 01 Sep 2010) | 18 lines Merged revisions 284478 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284478 | tilghman | 2010-09-01 13:49:11 -0500 (Wed, 01 Sep 2010) | 11 lines Ensure that all areas that previously used select(2) now use poll(2), with implementations that need poll(2) implemented with select(2) safe against 1024-bit overflows. This is a followup to the fix for the pthread timer in 1.6.2 and beyond, fixing a potential crash bug in all supported releases. (closes issue #17678) Reported by: russell Branch: https://origsvn.digium.com/svn/asterisk/team/tilghman/ast_select Review: https://reviewboard.asterisk.org/r/824/ ........ ................ r284595 | tilghman | 2010-09-01 22:57:43 -0500 (Wed, 01 Sep 2010) | 2 lines Failed to rerun bootstrap.sh after last commit ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284598 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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cf7bbcc4c6 |
Expand the caller ANI field to an ast_party_id
Expand the ani field in ast_party_caller and ast_party_connected_line to an ast_party_id. This is an extension to the ast_callerid restructuring patch in review: https://reviewboard.asterisk.org/r/702/ Review: https://reviewboard.asterisk.org/r/744/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276393 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ec37ffbdaf |
ast_callerid restructuring
The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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f59fe83c56 |
More 32->64 bit codec conversions.
In the process of swapping ULAW to a place in the extended codec space, we found several unhandled cases, where a 32-bit integer was still being used to handle a codec field. Most of these have been fixed with this commit, although there is at least one case (codec_dahdi) which depends upon outside headers to be altered before a conversion can be made. (Fixes AST-278, SWP-459) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231850 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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d8e0c58437 |
Expand codec bitfield from 32 bits to 64 bits.
Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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adde72d7b1 |
gcc 4.4: Remove a nop memset size 0 that annoys gcc
This memset doesn't write beyond the end of the buffer. (tmpbuf has size of 4). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@218184 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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642bec4d6f |
AST-2009-005
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211539 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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0264eef115 |
Merge the new Channel Event Logging (CEL) subsystem.
CEL is the new system for logging channel events. This was inspired after facing many problems trying to represent what is possible to happen to a call in Asterisk using CDR records. For more information on CEL, see the built in HTML or PDF documentation generated from the files in doc/tex/. Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard work developing this code. Also, thanks to Matt Nicholson (mnicholson) and Sean Bright (seanbright) for their assistance in the final push to get this code ready for Asterisk trunk. Review: https://reviewboard.asterisk.org/r/239/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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6f53ed4c67 |
This commit introduces COLP/CONP and Redirecting party information into Asterisk.
The channel drivers which have been most heavily tested with these enhancements are chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be introduced in a later commit. chan_skinny has code added to it here, but according to user pj, the support on chan_skinny is not working as of now. This will be fixed in a later commit. A special thanks goes out to bugtracker user gareth for getting the ball rolling and providing the initial support for this work. Without his initial work on this, this would not have been nearly as painless as it was. This functionality has been tested by Digium's product quality department, as well as a customer site running thousands of calls every day. In addition, many many many many bugtracker users have tested this, too. (closes issue #8824) Reported by: gareth Review: http://reviewboard.digium.com/r/201 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186525 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
16 years ago |
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12b7311782 |
Silence a build warning. (chan_phone.c:810: warning: value computed is not used)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160171 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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08af5bb312 |
Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiating
when a file is invalid from when a file is missing. This is most important when we have two configuration files. Consider the following example: Old system: sip.conf users.conf Old result New result ======== ========== ========== ========== Missing Missing SIP doesn't load SIP doesn't load Missing OK SIP doesn't load SIP doesn't load Missing Invalid SIP doesn't load SIP doesn't load OK Missing SIP loads SIP loads OK OK SIP loads SIP loads OK Invalid SIP loads incompletely SIP doesn't load Invalid Missing SIP doesn't load SIP doesn't load Invalid OK SIP doesn't load SIP doesn't load Invalid Invalid SIP doesn't load SIP doesn't load So in the case when users.conf doesn't load because there's a typo that disrupts the syntax, we may only partially load users, instead of failing with an error, which may cause some calls not to get processed. Worse yet, the old system would do this with no indication that anything was even wrong. (closes issue #10690) Reported by: dtyoo Patches: 20080716__bug10690.diff.txt uploaded by Corydon76 (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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bea1644dc2 |
Merge more changes from the resolve-shadow-warnings branch (henceforth known
as RSW since i am too lazy to keep typing it all out). This time a few of the channels. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136888 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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72de1ae386 |
rename DialTone.h to chan_phone.h because chan_phone.c is the only file using it
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118790 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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f1e9371da8 |
- revert change to ast_queue_hangup and create ast_queue_hangup_with_cause
- make data member of the ast_frame struct a named union instead of a void Recently the ast_queue_hangup function got a new parameter, the hangupcause Feedback came in that this is no good and that instead a new function should be created. This I did. The hangupcause was stored in the seqno member of the ast_frame struct. This is not very elegant, and since there's already a data member that one should be used. Problem is, this member was a void *. Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone wants to store another type in there in the future. This commit is so massive, because all ast_frame.data uses have to be altered to ast_frame.data.data Thanks russellb and kpfleming for the feedback. (closes issue #12674) Reported by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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496adc6fc0 |
Merged revisions 106235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4 lines Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things. (closes issue #12148) Reported by: jcomellas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106239 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7e8835e0d7 |
remove another set of redundant #include "asterisk/options.h"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89512 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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0595b5e2aa |
include "logger.h" and errno.h from asterisk.h - usage shows that they
were included almost everywhere. Remove some of the instances. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89424 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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fdb7f7ba3d |
Start untangling header inclusion in a way that does not affect
build times - tested, there is no measureable difference before and after this commit. In this change: use asterisk/compat.h to include a small set of system headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the inclusion is conditional on HAVE_FOO_H as determined by autoconf. Normally, source files should not include any of the above system headers, and instead use either "asterisk.h" or "asterisk/compat.h" which does it better. For the time being I have left alone second-level directories (main/db1-ast, etc.). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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0f6892419e |
One more typo in config.c; and missed conversions due to the constifying of ast_variable_new parameters
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89270 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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7c56918262 |
Commit some cleanups to the format type code.
- Remove the AST_FORMAT_MAX_* types, as these are consuming 3 out of our available 32 bits. - Add a native slin16 type, so that 16kHz codecs can translate without losing resolution. (This doesn't affect anything immediately, until another codec has wb support.) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89071 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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9a46a17c27 |
Merged revisions 81523 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10651) ........ r81523 | qwell | 2007-09-05 10:14:30 -0500 (Wed, 05 Sep 2007) | 5 lines Do not try to unregister a NULL channel tech. Also changed load_module function to use defines rather than numbers for return values. Issue 10651, patch by rbraun_proformatique, with additions by me. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81524 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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56b9568164 |
Don't reload a configuration file if nothing has changed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79747 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |