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${ noResults }
33236 Commits (e26b57984f638bd965c681f99800bc4b15e1d8ce)
Author | SHA1 | Message | Date |
---|---|---|---|
|
e26b57984f |
asterisk: Add macro for curl user agent.
Currently, each module that uses libcurl duplicates the standard Asterisk curl user agent. This adds a global macro for the Asterisk user agent used for curl requests to eliminate this duplication. ASTERISK-29861 #close Change-Id: I9fc37935980384b4daf96ae54fa3c9adb962ed2d |
3 years ago |
|
1633410161 |
res_stir_shaken: refactor utility function
Refactors temp file utility function into file.c. ASTERISK-29809 #close Change-Id: Ife478708c8f2b127239cb73c1755ef18c0bf431b |
3 years ago |
|
39820e3561 |
app_voicemail: Emit warning if asking for nonexistent mailbox.
Currently, if VoiceMailMain is called with a mailbox, if that mailbox doesn't exist, then the application silently falls back to prompting the user for the mailbox, as if no arguments were provided. However, if a specific mailbox is requested and it doesn't exist, then no warning at all is emitted. This fixes this behavior to now warn if a specifically requested mailbox could not be accessed, before falling back to prompting the user for the correct mailbox. ASTERISK-29920 #close Change-Id: Ib4093b88cd661a2cabc5d685777d4e2f0ebd20a4 |
3 years ago |
|
a2aa881dcb |
res_pjsip_pubsub: fix Batched Notifications stop working
If Subscription refresh occurred between when the batched notification was scheduled and the serialized notification was to be sent, then new schedule notification task would never be added. There are 2 threads: thread #1. ast_sip_subscription_notify is called, if notification_batch_interval then call schedule_notification. 1.1. The schedule_notification checks notify_sched_id > -1 not true, then send_scheduled_notify = 1 notify_sched_id = ast_sched_add(sched, sub_tree->notification_batch_interval, sched_cb.... 1.2. The sched_cb pushes task serialized_send_notify to serializer and returns 0 which means no reschedule. 1.3. The serialized_send_notify checks send_scheduled_notify if it's false the just returns. BUT notify_sched_id is still set, so no more ast_sched_add. thread #2. pubsub_on_rx_refresh is called 2.1 it pushes serialized_pubsub_on_refresh_timeout to serializer 2.2. The serialized_pubsub_on_refresh_timeout calls pubsub_on_refresh_timeout which calls send_notify 2.3. The send_notify set send_scheduled_notify = 0; The serialized_send_notify should always unset notify_sched_id. ASTERISK-29904 #close Change-Id: Ifc50c00b213c396509e10326a1ed89d8cf8c7875 |
3 years ago |
|
c12cb899de |
res_pjsip_pubsub: provide a display name for RLS subscriptions
Whereas BLFs allow to show a display name for each RLS entry, the asterisk provides only the extension now. This is not end user friendly. This commit adds a new resource_list option, resource_display_name, to indicate whether display name of resource or the resource name being provided for RLS entries. If this option is enabled, the Display Name will be provided. This option is disabled by default to remain the previous behavior. If the 'event' set to 'presence' or 'dialog' the non-empty HINT name will be set as the Display Name. The 'message-summary' is not supported yet. ASTERISK-29891 #close Change-Id: Ic5306bd5a7c73d03f5477fe235e9b0f41c69c681 |
3 years ago |
|
b1765c93e4 |
func_db: Add validity check for key names when writing.
Adds a simple sanity check for key names when users are writing data to AstDB. This captures four cases indicating malformed keynames that generally result in bad data going into the DB that the user didn't intend: an empty key name, a key name beginning or ending with a slash, and a key name containing two slashes in a row. Generally, this is the result of a variable being used in the key name being empty. If a malformed key name is detected, a warning is emitted to indicate the bug in the dialplan. ASTERISK-29925 #close Change-Id: Ifc08a9fe532a519b1b80caca1aafed7611d573bf |
3 years ago |
|
4722c8b70a |
cli: Add core dump info to core show settings.
Adds two pieces of information to the core show settings command which are useful in the context of getting backtraces. The first is to display whether or not Asterisk would generate a core dump if it were to crash. The second is to show the current running directory of Asterisk. ASTERISK-29866 #close Change-Id: Ic42c0a9ecc233381aad274d86c62808d1ebb4d83 |
3 years ago |
|
335c69ead4 |
documentation: Adds missing default attributes.
The configObject tag contains a default attribute which allows the default value to be specified, if applicable. This allows for the default value to show up specially on the wiki in a way that is clear to users. There are a couple places in the tree where default values are included in the description as opposed to as attributes, which means these can't be parsed specially for the wiki. These are changed to use the attribute instead of being included in the text description. ASTERISK-29898 #close Change-Id: I9d7ea08f50075f41459ea7b76654906b674ec755 |
3 years ago |
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c9ef2b3b86 |
app_mp3: Document and warn about HTTPS incompatibility.
mpg123 doesn't support HTTPS, but the MP3Player application doesn't document this or warn the user about this. HTTPS streams have become more common nowadays and users could reasonably try to play them without being aware they should use the HTTP stream instead. This adds documentation to note this limitation. It also throws a warning if users try to use the HTTPS stream to tell them to use the HTTP stream instead. ASTERISK-29900 #close Change-Id: Ie3b029be5258c5a701f71ed3b1a7a80d1e03b827 |
3 years ago |
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0da713168d |
app_mf: Add max digits option to ReceiveMF.
Adds an option to the ReceiveMF application to allow specifying a maximum number of digits. Originally, this capability was not added to ReceiveMF as it was with ReceiveSF because typically a ST digit is used to denote that sending of digits is complete. However, there are certain signaling protocols which simply transmit a digit (such as Expanded In-Band Signaling) and for these, it's necessary to be able to read a certain number of digits, as opposed to until receiving a ST digit. This capability is added as an option, as opposed to as a parameter, to remain compatible with existing usage (and not shift the parameters). ASTERISK-29877 #close Change-Id: I4229167c9aa69b87402c3c2a9065bd8dfa973a0b |
3 years ago |
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585c2d17bb |
ami: Allow events to be globally disabled.
The disabledevents setting has been added to the general section in manager.conf, which allows users to specify events that should be globally disabled and not sent to any AMI listeners. This allows for processing of these AMI events to end sooner and, for frequent AMI events such as Newexten which users may not have any need for, allows them to not be processed. Additionally, it also cleans up core debug as previously when debug was 3 or higher, the debug was constantly spammed by "Analyzing AMI event" messages along with a complete dump of the event contents (often for Newexten). ASTERISK-29853 #close Change-Id: Id42b9a3722a1f460d745cad1ebc47c537fd4f205 |
3 years ago |
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3b1debb28b |
taskprocessor.c: Prevent crash on graceful shutdown
When tps_shutdown is called as part of the cleanup process there is a chance that one of the taskprocessors that references the tps_singletons object is still running. The change is to allow for tps_shutdown to check tps_singleton's container count and give the running taskprocessors a chance to finish. If after AST_TASKPROCESSOR_SHUTDOWN_MAX_WAIT (10) seconds there are still container references we shutdown anyway as this is most likely a bug due to a taskprocessor not being unreferenced. ASTERISK-29365 Change-Id: Ia932fc003d316389b9c4fd15ad6594458c9727f1 |
3 years ago |
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b41440a179 |
app_queue: load queues and members from Realtime when needed
There are a lot of Queue AMI actions and Queue applications which do not load queue and queue members from Realtime. AMI actions QueuePause - if queue not in memory - response "Interface not found". QueueStatus/QueueSummary - if queue not in memory - empty response. Applications: PauseQueueMember - if queue not in memory Attempt to pause interface %s, not found UnpauseQueueMember - if queue not in memory Attempt to unpause interface xxxxx, not found This patch adds a new function load_realtime_queues which loads queue and queue members for desired queue or all queues and all members if param 'queuename' is NULL or empty. Calls the function load_realtime_queues when needed. Also this patch fixes leak of ast_config in function set_member_value. Also this patch fixes incorrect LOG_WARNING when pausing/unpausing already paused/unpaused member. The function ast_update_realtime returns 0 when no record modified. So 0 is not an error to warn about. ASTERISK-29873 #close ASTERISK-18416 #close ASTERISK-27597 #close Change-Id: I554ee0eebde93bd8f49df7f84b74acb21edcb99c |
3 years ago |
|
16fccf140d |
manager.c: Simplify AMI ModuleCheck handling
This code was needlessly complex and would fail to properly delimit the response message if LOW_MEMORY was defined. Change-Id: Iae50bf09ef4bc34f9dc4b49435daa76f8b2c5b6e |
3 years ago |
|
427bee9beb |
res_prometheus.c: missing module dependency
added res_pjsip_outbound_registration to .requires in AST_MODULE_INFO which fixes issue with module crashes on load "FRACK!, Failed assertion" ASTERISK-29871 Change-Id: Ia0f49d048427a40e1b763296b834a52a03610096 |
3 years ago |
|
e1b050d8a3 |
res_pjsip.c: Correct minor typos in 'realm' documentation.
Change-Id: I886936b808def5540d40071321e72f6bfa19063a |
3 years ago |
|
134cbebc1f |
manager.c: Generate valid XML if attribute names have leading digits.
The XML Manager Event Interface (amxml) now generates attribute names that are compliant with the XML 1.1 specification. Previously, an attribute name that started with a digit would be rendered as-is, even though attribute names must not begin with a digit. We now prefix attribute names that start with a digit with an underscore ('_') to prevent XML validation failures. This is not backwards compatible but my assumption is that compliant XML parsers would already have been complaining about this. ASTERISK-29886 #close Change-Id: Icfaa56a131a082d803e9b7db5093806d455a0523 |
3 years ago |
|
4126d703bf |
build_tools/make_version: Fix bashism in comparison.
In POSIX sh (which we indicate in the shebang), there is no == operator. Change-Id: Ic03d38214d14cdf329b0ba272279a815bb532965 |
3 years ago |
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38c3c7f498 |
bundled_pjproject: Add additional multipart search utils
Added the following APIs: pjsip_multipart_find_part_by_header() pjsip_multipart_find_part_by_header_str() pjsip_multipart_find_part_by_cid_str() pjsip_multipart_find_part_by_cid_uri() Change-Id: I6aee3dcf59eb171f93aae0f0564ff907262ef40d |
3 years ago |
|
e505337065 |
chan_sip.c Fix pickup on channel that are in AST_STATE_DOWN
resolve issue with pickup on device that uses "183" and not "180" ASTERISK-29832 Change-Id: I4c7d223870f8ce9a7354e0f73d4e4cb2e8b58841 |
3 years ago |
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bfc4d63d15 |
build: Add "basebranch" to .gitreview
If you have a development branch for a major project that will receive gerrit reviews it'll probably be named something like "development/16/newproject". That will necessitate setting "defaultbranch=development/16/newproject" in .gitreview. The make_version script uses that variable to construct the asterisk version however, which results in versions like "GIT-development/16/newproject-ee582a8c7b" which is probably not what you want. Worse, since the download_externals script uses make_version to construct the URL to download the binary codecs or DPMA. Since it's expecting a simple numeric version, the downloads will fail. To get this to work, a new variable "basebranch" has been added to .gitreview and make_version has been updated to use that instead of defaultversion: .gitreview: defaultbranch=development/16/myproject basebranch=16 Now git-review will send the reviews to the proper branch (development/16/myproject) but the version will still be constructed using the simple branch number (16). If "basebranch" is missing from .gitreview, make_version will fall back to using "defaultbranch". Change-Id: I2941a3b21e668febeb6cfbc1a7bb51a67726fcc4 |
3 years ago |
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8d571ea6b5 |
res_pjsip_outbound_authenticator_digest: Prevent ABRT on cleanup
In dev mode, if you call pjsip_auth_clt_deinit() with an auth_sess that hasn't been initialized, it'll assert and abort. If digest_create_request_with_auth() fails to find the proper auth object however, it jumps to its cleanup which does exactly that. So now we no longer attempt to call pjsip_auth_clt_deinit() if we never actually initialized it. ASTERISK-29888 Change-Id: Ib6171c25c9fe8e61cc8d11129e324c021bc30b62 |
3 years ago |
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386c5e495f |
cdr: allow disabling CDR by default on new channels
Adds a new option, defaultenabled, to the CDR core to control whether or not CDR is enabled on a newly created channel. This allows CDR to be disabled by default on new channels and require the user to explicitly enable CDR if desired. Existing behavior remains unchanged. ASTERISK-29808 #close Change-Id: Ibb78c11974bda229bbb7004b64761980e0b2c6d1 |
3 years ago |
|
70f8ea0d1a |
res_tonedetect: Fixes some logic issues and typos
Fixes some minor logic issues with the module: Previously, the OPT_END_FILTER flag was getting tested before options were parsed, so it could never evaluate to true (wrong ordering). Additionally, the initially parsed timeout (float) needs to be compared with 0, not the result int which is set afterwards (wrong variable). ASTERISK-29857 #close Change-Id: I0062bce3b391c15e5df7a714780eeaa96dd93d4c |
3 years ago |
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7ae8321925 |
func_frame_drop: Fix typo referencing wrong buffer
In order to get around the issue of certain frames having names that could overlap, func_frame_drop surrounds names with commas for the purposes of comparison. The buffer is allocated and printed to properly, but the original buffer is used for comparison. In most cases, this wouldn't have had any effect, but that was not the intention behind the buffer. This updates the code to reference the modified buffer instead. ASTERISK-29854 #close Change-Id: I430b52e14e712d0e62a23aa3b5644fe958b684a7 |
3 years ago |
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7b15ced930 |
res/res_rtp_asterisk: fix skip in rtp sequence numbers after dtmf
When generating dtmfs, asterisk can incorrectly think packet loss occured during the dtmf generation, resulting in a jump in sequence numbers when forwarding voice frames resumes. This patch forces asterisk to re-learn the expected sequence number after each DTMF to avoid this ASTERISK-29869 #close Change-Id: Icc7de3d947b207b82c99d3c327af8095884df853 |
3 years ago |
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851a759619 |
res_http_websocket: Add a client connection timeout
Previously there was no way to specify a connection timeout when attempting to connect a websocket client to a server. This patch makes it possible to now do such. Change-Id: I5812f6f28d3d13adbc246517f87af177fa20ee9d |
3 years ago |
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ce91a0fdbc |
build: Rebuild configure and autoconfig.h.in
autoconfigh.h.in was missed in the original review for this issue. Additionally it looks like I have newer pkg-config autoconf macros on my development machine. ASTERISK-29817 Change-Id: I3c85a4de82c5d7d6e0e23dad4c33bb650a86a57b |
3 years ago |
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b79a571279 |
sched: fix and test a double deref on delete of an executing call back
sched: Avoid a double deref when AST_SCHED_DEL_UNREF is called on an executing call-back. This is done by adding a new variable 'rescheduled' to the struct sched which is set in ast_sched_runq and checked in ast_sched_del_nonrunning. ast_sched_del_nonrunning is a replacement for now deprecated ast_sched_del which returns a new possible value -2 if called on an executing call-back with rescheduled set. ast_sched_del is modified to call ast_sched_del_nonrunning to maintain existing code. AST_SCHED_DEL_UNREF is also updated to look for the -2 in which case it will not throw a warning or invoke refcall. test_sched: Add a new unit test sched_test_freebird that will check the reference count in the resolved scenario. ASTERISK-29698 Change-Id: Icfb16b3acbc29cf5b4cef74183f7531caaefe21d |
3 years ago |
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93d090147f |
app_queue.c: Queue don't play "thank-you" when here is no hold time announcements
if holdtime is (0 min, 0 sec) there is no hold time announcements we should then also not playing queue-thankyou ASTERISK-29831 Change-Id: Ic7e51dcde526b23f1cd8d24e1d1e2d81e10f9d2c |
3 years ago |
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5875c7bb6c |
res_pjsip_sdp_rtp.c: Support keepalive for video streams.
ASTERISK-28890 #close Change-Id: Iad269a8dc36f892ede90fe8ceb3010560c0f70d1 |
3 years ago |
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23be22abf4 |
build_tools/make_version: Fix sed(1) syntax compatibility with NetBSD
Fix the sed(1) invocation used to process git-svn-id not to use "\s" that is a GNU-ism and is not supported by NetBSD sed. As a result, this call did not work properly and make_version did output the full git-svn-id line rather than the revision. ASTERISK-29852 Change-Id: Ie4b406e2748920643446851a0a252a4ca7245772 |
3 years ago |
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2b490787eb |
main/utils: Implement ast_get_tid() for NetBSD
Implement the ast_get_tid() function for NetBSD system. NetBSD supports getting the TID via _lwp_self(). ASTERISK-29850 Change-Id: If57fd3f9ea15ef5d010bfbdcbbbae9b379f72f8c |
3 years ago |
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dda02b8979 |
main: Enable rdtsc support on NetBSD
Enable the Linux rdtsc implementation on NetBSD as well. The assembly works correctly there. ASTERISK-29851 Change-Id: I460ad9b4d971913420ecb84186f5ba5ab03f6f37 |
3 years ago |
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6a879eea31 |
BuildSystem: Fix misdetection of gethostbyname_r() on NetBSD
Fix the configure script not to detect the presence of gethostbyname_r() on NetBSD incorrectly. NetBSD includes it as an internal libc symbol that is not exposed in system headers and that is incompatible with other implementations. In order to avoid misdetecting it, perform the symbol check only if the declaration is found in the public header first. ASTERISK-29817 Change-Id: Iafa359b09908251bcd299ff54be003ea129b9eda |
3 years ago |
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710c8f8b29 |
include: Remove unimplemented HMAC declarations
Remove the HMAC declarations from the includes. They are not implemented nor used anywhere, and their presence breaks the build on NetBSD that delivers an incompatible hmac() function in <stdlib.h>. ASTERISK-29818 Change-Id: I0c4b88645e30174b1b63846a6b328625b69c2ea7 |
3 years ago |
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27502b6dd2 |
frame.h: Fix spelling typo
Fixes CNG description from "noice" to "noise". ASTERISK-29855 #close Change-Id: Ie7cbbd7d72b426693df7447384ff8700318cd36d |
3 years ago |
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d35e292ae4 |
res_rtp_asterisk: Fix typo in flag test/set
The code currently checks to see if an RFC3389 warning flag is set, except if it is, it merely sets the flag again, the logic of which doesn't make any sense. This adjusts the if comparison to check if the flag has NOT been set, and if so, emit a notice log event and set the flag so that future frames do not cause an event to be logged. ASTERISK-29856 #close Change-Id: Ib7098c947c63537d087a03b4646199fbb963f8e1 |
3 years ago |
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97ace6b816 |
bundled_pjproject: Fix srtp detection
Reverted recent change that set '--with-external-srtp' instead of '--without-external-srtp'. Since Asterisk handles all SRTP, we don't need it enabled in pjproject at all. ASTERISK-29867 Change-Id: I2ce1bdd30abd21c062eac8f8fefe9b898787b801 |
3 years ago |
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b1dfc9c805 |
res_pjsip: Make message_filter and session multipart aware
Neither pjsip_message_filter's filter_on_tx_message() nor res_pjsip_session's session_outgoing_nat_hook() were multipart aware and just assumed that an SDP would be the only thing in a message body. Both were changed to use the new pjsip_get_sdp_info() function which searches for an sdp in both single- and multi- part message bodies. ASTERISK-29813 Change-Id: I8f5b8cfdc27f1d4bd3e7491ea9090951a4525c56 |
3 years ago |
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5d1407aa06 |
build: Fix issues building pjproject
The change to allow easier hacking on bundled pjproject created a few issues: * The new Makefile was trying to run the bundled make even if PJPROJECT_BUNDLED=no. third-party/Makefile now checks for PJPROJECT_BUNDLED and JANSSON_BUNDLED and skips them if they are "no". * When building with bundled, config_site.h was being copied only if a full make or a "make main" was done. A "make res" would fail all the pjsip modules because they couldn't find config_site.h. The Makefile now copies config_site.h and asterisk_malloc_debug.h into the pjproject source tree when it's "configure" is performed. This is how it used to be before the big change. ASTERISK-29858 Change-Id: I9427264fa3cb8b3f59a95e5f9693eac236a6f76d |
3 years ago |
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921ab52cf3 |
res_pjsip: Add utils for checking media types
Added two new functions to assist checking media types... * ast_sip_are_media_types_equal compares two pjsip_media_types. * ast_sip_is_media_type_in tests if one media type is in a list of others. Added static definitions for commonly used media types to res_pjsip.h. Changed several modules to use the new functions and static definitions. ASTERISK_29813 (not ready to close) Change-Id: Ief77675235bd3bf00a6b095d4673fd878d0801b9 |
3 years ago |
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0d1b9e6baf |
bundled_pjproject: Create generic pjsip_hdr_find functions
pjsip_msg_find_hdr(), pjsip_msg_find_hdr_by_name(), and pjsip_msg_find_hdr_by_names() require a pjsip_msg to be passed in so if you need to search a header list that's not in a pjsip_msg, you have to do it yourself. This commit adds generic versions of those 3 functions that take in the actual header list head instead of a pjsip_msg so if you need to search a list of headers in something like a pjsip_multipart_part, you can do so easily. Change-Id: I6f2c127170eafda48e5e0d5d4d187bcd52b4df07 |
3 years ago |
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65b2ddee26 |
say.c: Prevent erroneous failures with 'say' family of functions.
A regression was introduced in ASTERISK~29531 that caused 'say' functions to fail with file lists that would previously have succeeded. This caused affected channels to hang up where previously they would have continued. We now explicitly check for the empty string to restore the previous behavior. ASTERISK-29859 #close Change-Id: Ia2e5769868e2792313c2d7c07996efe009c6f8d5 |
3 years ago |
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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 |
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fbaf74bd3a |
pbx_variables: add missing ASTSBINDIR variable
Every config variable in the directories section of asterisk.conf currently has a counterpart built-in variable containing the value of the config option, except for the last one, astsbindir, which should have an ASTSBINDIR variable. However, the actual corresponding ASTSBINDIR variable is missing in pbx_variables.c. This adds the missing variable so that all the config options have their corresponding variable. ASTERISK-29847 #close Change-Id: I36006faf471825b36ebc8aa5e87a3bcb38d446fc |
3 years ago |
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bc59b66de3 |
bundled_pjproject: Make it easier to hack
There are times when you need to troubleshoot issues with bundled pjproject or add new features that need to be pushed upstream but... * The source directory created by extracting the pjproject tarball is not scanned for code changes so you have to keep forcing rebuilds. * The source directory isn't a git repo so you can't easily create patches, do git bisects, etc. * Accidentally doing a make distclean will ruin your day by wiping out the source directory, and your changes. * etc. This commit makes that easier. See third-party/pjproject/README-hacking.md for the details. ASTERISK-29824 Change-Id: Idb1251040affdab31d27cd272dda68676da9b268 |
3 years ago |
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0d62735f99 |
utils.c: Remove all usages of ast_gethostbyname()
gethostbyname() and gethostbyname_r() are deprecated in favor of getaddrinfo() which we use in the ast_sockaddr family of functions. ASTERISK-29819 #close Change-Id: Ie277c0ef768d753b169c121ef570a71665692ab7 |
3 years ago |
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262a4053ff |
say.conf: fix 12pm noon logic
Fixes 12pm noon incorrectly returning 0/a.m. Also fixes a misspelling typo in the config. ASTERISK-29695 #close Change-Id: Ie40f9618636eb4c483b449bd707a5dcffca5c406 |
3 years ago |
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3616dda066 |
pjproject: Fix incorrect unescaping of tokens during parsing
ASTERISK-29664 #close Change-Id: I29dcde52e9faeaf2609c604eada61c6a9e49d8f5 |
3 years ago |