mirror of https://github.com/asterisk/asterisk
master
20
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${ noResults }
5267 Commits (5b15ec966db9c686ca7fbc74be8d57b4c63253d7)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
bf957e1bf7 |
Merge "file: Ensure nativeformats remains valid for lifetime of use."
|
10 years ago |
|
|
516f49f316 |
stasis_endpoints: Add new Status and Headers to ContactStatus
ASTERISK-25903 added a new headers to AMI Event ContactStatusDetail. ASTERISK-25904 added a new Status to AMI Event ContactStatusDetail. These additions should be also in stasis_endpoints to include in command "manager show event ContactStatus" Change-Id: I7610ad02a998e1f26c20caa27aa50279d0164f6a |
10 years ago |
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17b6ba49ef |
file: Ensure nativeformats remains valid for lifetime of use.
It is possible for the nativeformats of a channel to change throughout its lifetime. As a result a user of it needs to either ensure the channel is locked when accessing the formats or keep a reference to the nativeformats themselves. This change fixes the file playback support so it keeps a reference to the nativeformats when accessing things. ASTERISK-25998 #close Change-Id: Ie45b65475e1481ddf05b874ee48f63e39fff8915 |
10 years ago |
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02f4ca1079 |
config_options.c: Expand #ifdef to contain whole if statement.
ASTERISK-25956 #close Change-Id: If6961ec54be276d5ab4f012ee7e7b420cb45de38 |
10 years ago |
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e61716b774 |
pjproject_bundled: Various fixes discovered during testing of OSes
For all OSes: * Disabled third-party codecs in pjproject and added '--disable-speex-codec --disable-speex-aec --disable-gsm-codec' to the configure options since we don't use the pjsip codec capability. FreeBSD: * Added FreeBSD support to install_prereq. * Changed pjproject/configure.m4 to use $GNU_MAKE instead of hardcoding "make". * Added __progname and environ to asterisk.exports.in. * Reverted the use of ldconfig to create shared library symlinks to ln. * Only enable epoll in pjproject if `uname -s` is Linux. * Added a patch to pjproject to take the name of the 'make' command from an environment variable if supplied. This is needed for the python bindings. (merged by Teluu into pjproject trunk 5/3/2016) FreeBSD support isn't complete. Still some general issues regarding make/gmake having nothing to do with pjproject. With some handholding it DOES build successfully. CentOS: Added 'patch' and 'bzip2' to install_prereq PACKAGES_RH. CentOS 6/7 32/64 build and run the pjsip testsuite successfully. Ubuntu: No changes required. Ubuntu 15/16 32/64 build and run the pjsip testsuite successfully. Debian: No changes required. Debian 6/7/8 32/64 build and run the pjsip testsuite successfully. There will utimately be a follow-up patch to create an install_prereq for the testsuite as I've discovered a few missing requirements. ASTERISK-25968 #close Change-Id: I5756a07facfc63798115a5e73a8709382fe9259c |
10 years ago |
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d53d494f0b |
Merge "app_chanspy: reduce audio loss on the spying channel."
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10 years ago |
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f324801763 |
Merge "config: Fix ast_config_text_file_save2 writability check for missing files"
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10 years ago |
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7281770710 |
app_chanspy: reduce audio loss on the spying channel.
ChanSpy was creating its audiohook with the flags AST_AUDIOHOOK_TRIGGER_SYNC and AST_AUDIOHOOK_SMALL_QUEUE, which caused audio frames to be lost when queues grow too large or when read and write queues go out of sync. Now these flags are set conditionally: - AST_AUDIOHOOK_TRIGGER_SYNC is not set if the option "o" is set - a new option "l" is created: if set, AST_AUDIOHOOK_SMALL_QUEUE will not be set on the audiohook ASTERISK-25866 Change-Id: I9c7652f41d9fa72c8691e4e70ec4fd16b047a4dd |
10 years ago |
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09d588dc2f |
Merge changes from topic 'system_stress_patches'
* changes: test_message.c: Wait longer in case dialplan also processes the test message. Manager: Short circuit AMI message processing. manager.c: Eliminate most RAII_VAR usage. |
10 years ago |
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0f29785101 |
Merge "manager_channels.c: Fix allocation failure crash."
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10 years ago |
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811e24f595 |
Merge "Bridge system: Fix memory leaks and double frees on impart failure."
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10 years ago |
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807a765cfb |
Merge "bridge_softmix.c: Fix crash if channel fails to join mixing tech."
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10 years ago |
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284bb814ac |
config: Fix ast_config_text_file_save2 writability check for missing files
A patch I did back in 2014 modified ast_config_text_file_save2 to check the writability of the main file and include files before truncating and re-writing them. An unintended side-effect of this was that if a file doesn't exist, the check fails and the write is aborted. This patch causes ast_config_text_file_save2 to check the writability of the parent directory of missing files instead of checking the file itself. This allows missing files to be created again. A unit test was also added to test_config to test saving of config files. The regression was discovered when app_voicemail's passwordlocation=spooldir feature stopped working. ASTERISK-25917 #close Reported-by: Jonathan Rose Change-Id: Ic4dbe58c277a47b674679e49daed5fc6de349f80 |
10 years ago |
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1df086f821 |
Merge "bridge: Hold off more than one imparting channel at a time."
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10 years ago |
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b3cc74fda9 |
manager_channels.c: Fix allocation failure crash.
An earlier allocation failure failed to create a channel snapshot for the AMI HangupRequest/SoftHangupRequest event which resulted in a crash in channel_hangup_request_cb(). Where the stasis message gets generated cannot tell if the NULL snapshot returned was because of an allocation failure or the channel was a dummy channel. * Made channel_hangup_request_cb() check if the channel blob has a snapshot and exit if it doesn't. * Eliminated the RAII_VAR usage in channel_hangup_request_cb(). Change-Id: I0b6a1c4e95cbb7d80b2a7054c6eadecc169dfd24 |
10 years ago |
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a63656b419 |
Bridge system: Fix memory leaks and double frees on impart failure.
You cannot reference the passed in features struct after calling ast_bridge_impart(). Even if the call fails. Change-Id: I902b88ba0d5d39520e670fb635078a367268ea21 |
10 years ago |
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71dfa35540 |
bridge_softmix.c: Fix crash if channel fails to join mixing tech.
softmix_bridge_join() failed because of an allocation failure. To address this, the softmix bridge technology now checks if the channel failed to join softmix successfully. In addition, the bridge now begins the process of kicking the channel out of the bridge so we don't have channels partially in the bridge for very long. * Fix the test_channel_feature_hooks.c unit tests. The test channel must have a valid codec to join the simple_bridge technology. This patch makes joining a bridge more strict by not allowing partially joined channels to remain in the bridge. Change-Id: I97e2ade6a2bcd1214f24fb839fda948825b61a2b |
10 years ago |
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06632a0d11 |
Manager: Short circuit AMI message processing.
Improve AMI message processing performance if there are no consumers listening for the messages. We now skip creating the AMI event message text strings. Change-Id: I7b22fc5ec4e500d00635c1a467aa8ea68a1bb2b3 |
10 years ago |
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6ddd856b86 |
manager.c: Eliminate most RAII_VAR usage.
* Made ast_manager_event_blob_create() not allocate the ao2 event object with a lock as it is not needed. Change-Id: I8e11bfedd22c21316012e0b9dd79f5918f644b7c |
10 years ago |
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24913e1540 |
Merge "lock.c: Check *lt before dereferencing it"
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10 years ago |
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b4dd374bc4 |
Merge "stringfields: Update extended string fields for master only."
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10 years ago |
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c991e5472e |
lock.c: Check *lt before dereferencing it
*lt is NULL if t->tracking == 0 ASTERISK-25948 #close Change-Id: I4a81af28f9c82a74aa82413d772a7dc8fa6f45ba |
10 years ago |
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1c5248c383 |
bridge: Hold off more than one imparting channel at a time.
An earlier patch blocked the ast_bridge_impart() call until the channel either entered the target bridge or it failed. Unfortuantely, if the target bridge is stasis and the imprted channel is not a stasis channel, stasis bounces the channel out of the bridge to come back into the bridge as a proper stasis channel. When the channel is bounced out, that released the block on ast_bridge_impart() to continue. If the impart was a result of a transfer, then it became a race to see if the swap channel would get hung up before the imparted channel could come back into the stasis bridge. If the imparted channel won then everything is fine. If the swap channel gets hung up first then the transfer will fail because the swap channel is leaving the bridge. * Allow a chain of ast_bridge_impart()'s to happen before any are unblocked to prevent the race condition described above. When the channel finally joins the bridge or completely fails to join the bridge then the ast_bridge_impart() instances are unblocked. ASTERISK-25947 Reported by: Richard Mudgett ASTERISK-24649 Reported by: John Bigelow ASTERISK-24782 Reported by: John Bigelow Change-Id: I8fef369171f295f580024ab4971e95c799d0dde1 |
10 years ago |
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cc9b72208f |
Merge "Dial: Combine frame handling functions."
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10 years ago |
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6365f0018f |
bridge_channel.c: Ignore role setup failure in channel push.
We have to setup the channel roles after the bridge class push is called because the bridge class push callback may have set roles on the incoming channel. Since we have already partially pushed the channel into the bridge and reversing what we have already done could be problematic, the only thing we can do is press on to complete pushing the channel into the bridge. * Ignore any channel role setup errors after pushing the channel into a bridge. The channel may behave incorrectly in the bridge but we can no longer abort the push at this time. Change-Id: I08a97082b729052ee65cdca6bb730cf1289ede00 |
10 years ago |
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5e64d7e7a3 |
Dial: Combine frame handling functions.
There is a good amount of repetition in the two frame handling routines in the Dial API. This commit combines the two functions into one. This is in preparation for an upcoming commit that adds the ability to handle frames for a channel in a bridge. ASTERISK-25925 Reported by Mark Michelson Change-Id: Iaae2f174e3058e774cb44e10659fcdfb85345c58 |
10 years ago |
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a6e2ba187a |
Codecs: strip codec name while parsing allow/disallow options
Failed registration using PJSIP/Realtime if one of the codec name in allow/disallow option is wrong or contains space. This patch strip codec name. ASTERISK-25914 Change-Id: Ifdf02de94e5ddbce305640f6f0666084a3b9283d |
10 years ago |
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caa416d5f3 |
stringfields: Update extended string fields for master only.
In 13, the new ast_string_field_header structure had to be dynamically allocated and assigned to a pointer in ast_string_field_mgr to preserve ABI compatability. In master, it can be converted to being a structure-in-place in ast_string_field_mgr to eliminate the extra alloc and free calls. Change-Id: Ia97c5345eec68717a15dc16fe2e6746ff2a926f4 |
10 years ago |
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2cc56573de |
core_unreal: Fix hangupcauses not getting set on Local channels
ASTERISK-25912 #close Change-Id: I8e72e6894feaf36c9450f2788d205d07baec23aa |
10 years ago |
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44fba00ca4 |
Merge "lock: Add named lock capability"
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10 years ago |
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8610f4344f |
Merge "pbx.h: Make ast_state_cb_type take more const."
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10 years ago |
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216abb0ae7 |
lock: Add named lock capability
Locking some objects like sorcery objects can be tricky because the underlying ao2 object may not be the same for all callers. For instance, two threads that call ast_sorcery_retrieve_by_id on the same aor name might actually get 2 different ao2 objects if the underlying wizard had to rehydrate the aor from a database. Locking one ao2 object doesn't have any effect on the other even if those objects had locks in the first place. Named locks allow access control by keyspace and key strings. Now an "aor" named "1000" can be locked and any other thread attempting to lock "aor" "1000" will wait regardless of whether the underlying ao2 object is the same or not. Mutex and rwlocks are supported. This capability will initially be used to lock an aor when multiple threads may be attempting to prune expired contacts from it. Change-Id: If258c0b7f92b02d07243ce70e535821a1ea7fb45 |
10 years ago |
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b47dfd1c6e |
Merge "pbx.c: Minor code rearangements."
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10 years ago |
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6138a75e8e |
pbx.h: Make ast_state_cb_type take more const.
This eliminates some casts that I made a note saying v10 and above would no longer need them. Better late than never :) Change-Id: I346cdb3032b6478ceb40eb6fe732978b54035572 |
10 years ago |
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72c19f7dc5 |
pbx.c: Minor code rearangements.
* Pull out a loop invariant. * Convert an else-if ladder to a switch statement. Change-Id: I0a95cfa9474a4600b9865f7b444534d275b37e95 |
10 years ago |
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28cefc3e88 |
pbx: Update doxygen for extension state watchers.
Change-Id: Id1403b12136de62a272c01bb355aef65fd2c2d1e |
10 years ago |
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724c16c543 |
Merge "pbx: Add support for autohints."
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10 years ago |
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8ed5f61152 |
frame.c: Copy the whole subclass in ast_frdup().
The problem is ast_frdup() does not copy whole frame.subclass for voice, video and image frames, only the format is copied. For video frames, the subclass structure contains the .frame_ending flag used to put the RTP marker where it needs to be. ASTERISK-25894 #close Change-Id: I812ca90e84ed5d4f473b997d0dd0d3c5a915fe33 |
10 years ago |
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4d06a4f366 |
Merge "Dial: Add function to append already-created channel."
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10 years ago |
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c49ee63db1 |
Merge "config: Allow filters when appending to a category"
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10 years ago |
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1dc5e28624 |
pbx: Add support for autohints.
This change introduces the concept of autohints. These are hints which are created as a result of device state changes occurring within the core. When this happens a hint will be created (if it does not exist already) using the device name as the extension. For example if a device state change is received for "PJSIP/bob" and autohints are enabled on a context then a hint will exist in that context for "bob" with a device of "PJSIP/bob". For virtual or custom device states the name after the type will be used. For example if the device state of "Custom:bob" changes then a hint will exist in that context for "bob" with a device of "Custom:bob". This functionality can be enabled in extensions.conf by placing "autohints=yes" in a context. ASTERISK-25881 #close Change-Id: I7e444c7da41b7b7d33374420fec658beeb18584e |
10 years ago |
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ef4d3f1328 |
Dial: Add function to append already-created channel.
The Dial API takes responsiblity for creating an outbound channel when calling ast_dial_append(). This commit adds a new function, ast_dial_append_channel(), which allows us to create the channel outside the Dial API and then to append the channel to the ast_dial structure. This is useful for situations where the channel's creation and dialing are distinct operations. Upcoming ARI early bridge work will illustrate its usage. ASTERISK-25889 Change-Id: Id8179f64f8f99132f80dead8d5db2030fd2c0509 |
10 years ago |
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245c9ca5ce |
Merge "stringfields: Refactor to allow fields to be added to the end of structures"
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10 years ago |
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984d6fd95c |
config: Allow filters when appending to a category
In sorcery based config files where there are multiple categories with the same name, you can't use the (+) operator to reliably append to a category because config.c stops looking when it finds the first one with the same name. Example: [1000] type = endpoint [1000] type = aor [1000](+) authenticate_qualify = yes This config will fail because config.c appends authenticate_qualify to the first category it finds, the endpoint, and that's not valid for endpoint. Solution: The capability to find a category that contains a certain variable already exists so the only real change was to parse anything after the '+' that's not a comma, as a filter string. [1000] type = endpoint [1000] type = aor [1000](+type=aor) authenticate_qualify = yes This now works as expected. Although the following example doesn't make any sense for pjsip, you can even specify multiple filters: [1000](+type=aor&qualify_frequency=10) ASTERISK-25868 #close Reported-by: Nick Repin Change-Id: I10773da4c79db36fbf1993961992af63d3441580 |
10 years ago |
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051da5c3af |
Merge "res_rtp_asterisk: Use separate SRTP session for RTCP with DTLS"
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10 years ago |
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4d40b161c3 |
stringfields: Refactor to allow fields to be added to the end of structures
String fields are great, except that you can't add new ones without breaking ABI compatibility because it shifts down everything else in the structure. The only alternative is to add your own char * field to the end of the structure and manage the memory yourself which isn't ideal, especially since you then can't use the OPT_STRINGFIELD_T type. Background: The reason string fields had to be declared inside the AST_DECLARE_STRING_FIELDS block was to facilitate iteration over all declared fields for initialization, compare and copy. Since AST_DECLARE_STRING_FIELDS declared the pool, then the fields, then the manager, you could use the offsets of the pool and manager and iterate over the sequential addresses in between to access the fields. The actual pool, field allocation and field set operations don't actually care where the field is. It's just iteration over the fields that was the problem. Solution: Extended String Fields An extended string field is one that is declared outside the AST_DECLARE_STRING_FIELDS block but still (anywhere) inside the parent structure. Other than using AST_STRING_FIELD_EXTENDED instead of AST_STRING_FIELD, it looks the same as other string fields. It's storage comes from the pool and it participates in string field compare and copy operations peformed on the parent structure. It's also a valid target for the OPT_STRINGFIELD_T aco option type. Implementation: To keep track of the extended fields and make sure that ABI isn't broken, the existing embedded_pool pointer in the manager structure was repurposed to be a pointer to a separate header structure that contains the embedded_pool pointer plus a vector of fields. The length of the manager structure didn't change and the embedded_pool pointer isn't used in the macros, only the stringfields C code. A side benefit of this is that changing the header structure in the future won't break ABI. ast_string_fields_init initializes the normal string fields and appends them to the vector, and subsequent calls to ast_string_field_init_extended initialize and append the extended fields. Cleanup, ast_string_fields_cmp, and ast_string_fields_copy can now work on the vector instead of sequentially traversing the addresses between the pool and manager. The total size of a structure using string fields didn't change, whether using extended fields or not, nor have the offsets of any structure members, either inside the original block or outside. Adding an extended field to the end of a structure is the same as adding a char *. Details: The stringfield C code was pulled out from utils.c and into stringfields.c. It just made sense. Additional work was done in ast_string_field_init and ast_calloc_with_stringfields to handle the allocation of the new header structure and the vector, and the associated cleanup. In the process some additional NULL pointer checking was added. A lot of work was done in stringfields.h since the logic for compare and copy is there. Documentation was added as well as somne additional NULL checking. The ability to call ast_calloc_with_stringfields with a number of structures greater than 1 never really worked. Well, the calloc worked but there was no way to access the additional structures or clean them up. It was agreed that there was no use case for requesting more than 1 structure so an ast_assert was added to prevent it and the iteration code removed. Testing: The stringfield unit tests were updated to test both normal and extended fields. Tests for ast_string_field_ptr_set_by_fields and ast_calloc_with_stringfields were also added. As an ABI test, 13 was compiled from git and the res_pjsip_* modules, except res_pjsip itself, saved off. The patch was then added and a full compile and install was performed. Then the older res_pjsip_* moduled were copied over the installed versions so res_pjsip was new and the rest were old. No issues. contact->aor, which is a char * at the end of contact, was then changed to an extended string field and a recompile and reinstall was performed, again leaving stock versions of the the res_pjsip_* modules. Again, no issues with the res_pjsip_* modules using the old stringfield implementation and with contact->aor as a char *, and res_pjsip itself using the new stringfield implementation and contact->aor being an extended string field. Finally, several existing string fields were converted to extended string fields to test OPT_STRINGFIELD_T. Again, no issues. Change-Id: I235db338c5b178f5a13b7946afbaa5d4a0f91d61 |
10 years ago |
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433d2c4bbf |
utils.c: Fix typo in handle_show_locks
ast_cli_allow_on_shutdown(e) should have been ast_cli_allow_at_shutdown(e). Change-Id: I4f092495c0b2bfd85c2651e0b5877bf4d05d9faf |
10 years ago |
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0ec3299c5f |
Merge "pjproject_bundled: Fix use of LDCONFIG for shared library link creation"
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10 years ago |
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304f81780d |
pjproject_bundled: Fix use of LDCONFIG for shared library link creation
LDCONFIG apparently isn't set to something sane on all systems so the creation of the shared library links fails. Instead of just testing for non-blank, main/Makefile now checks that LDCONFIG is actually executable and reverts to LN if it isn't. This applies to both libasteriskpj and libasteriskssl. Thanks to 'abelbeck' for pointing out that the issue was LDCONFIG. ASTERISK-25873 #close Reported-by: Hans van Eijsden Change-Id: I25b76379bc637726ec044b2c0e709b56b3701729 |
10 years ago |
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34457dd9db |
core_unreal.c: Add clarification comment about channel ref.
Change-Id: I0be0627260cd8d6b6c3cc345949dcfdf32eff1f3 |
10 years ago |