mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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10.7.0-digiumphones
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${ noResults }
5371 Commits (f5b31cb2ae8f67d0d3e1ded8fc78efe39631b8db)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
f5b31cb2ae |
Merge "frame.c: Fix off-nominal format ref leaks." into 13
|
9 years ago |
|
|
616e98ce0a |
Merge "T.140: Fix format ref and memory leaks." into 13
|
9 years ago |
|
|
da098e2227 |
Merge "astobj2.c: Add excessive ref count trap." into 13
|
9 years ago |
|
|
c54f9d2bf0 |
T.140: Fix format ref and memory leaks.
* channel.c:ast_sendtext(): Fix T.140 SendText memory leak. * format_compatibility.c: T.140 RED and T.140 were swapped. * res_rtp_asterisk.c:rtp_red_init(): Fix ast_format_t140_red ref leak. * res_rtp_asterisk.c:rtp_red_init(): Fix data race after starting periodic scheduled red_write(). * res_rtp_asterisk.c: Some other minor misc tweaks. Change-Id: Ifa27a2e0f8a966b1cf628607c86fc4374b0b88cb |
9 years ago |
|
|
607b3ac736 |
astobj2.c: Add excessive ref count trap.
Change-Id: I32e6a589cf9009450e4ff7cb85c07c9d9ef7fe4a |
9 years ago |
|
|
ab8cb5a7ce |
main/app.c: Memory corruption from early format destruction.
* make_silence() created a malloced silence slin frame without adding a slin format ref. When the frame is destroyed it will unref the slin format that never had a ref added. Memory corruption is expected to follow. * Simplified and fixed counting the number of samples in a frame list for make_silence(). * Eliminated an unnecessary RAII_VAR associated with the make_silence() frame. Change-Id: I47de3f9b92635b7f8b4d72309444d6c0aee6f747 |
9 years ago |
|
|
dcd8e4b1a0 |
frame.c: Fix off-nominal format ref leaks.
* ast_frisolate() could leak frame format refs on allocation failures. * Similified code in ast_frisolate() and code used by ast_frisolate(). Change-Id: I79566d4d36b3d7801bf0c8294fcd3e9a86a2ed6d |
9 years ago |
|
|
883e7fde31 |
abstract/fixed/adpative jitter buffer: disallow frame re-inserts
It was possible for a frame to be re-inserted into a jitter buffer after it had been removed from it. A case when this happened was if a frame was read out of the jitterbuffer, passed to the translation core, and then multiple frames were returned from said translation core. Upon multiple frames being returned the first is passed on, but sebsequently "chained" frames are put back into the read queue. Thus it was possible for a frame to go back into the jitter buffer where this would cause problems. This patch adds a flag to frames that are inserted into the channel's read queue after translation. The abstract jitter buffer code then checks for this flag and ignores any frames marked as such. Change-Id: I276c44edc9dcff61e606242f71274265c7779587 |
9 years ago |
|
|
473330983b |
taskprocessor.c: Change when high water warning logged.
The task processor queue reached X scheduled tasks message was originally intended to get logged only once per task processor to prevent spamming the log. This is no longer necessary since high and low water thresholds can better control when the message is logged. It is beneficial to generate the warning each time a task processor reaches the high water level because PJSIP stops processing new requests while any high water alert is active. Without this change you would have to enable at least debug level 3 logging to know about a repeated alert trigger. * Made generate the warning message whenever a task is pushed into the task processor that triggers the high water alert. * Appended 'again' to the warning for a repeated high water alert trigger. Change-Id: Iabf75a004f7edaf1e5e8c323099418e667cac999 |
9 years ago |
|
|
ac31233dbe |
acl.c: Improve ast_ouraddrfor() diagnostic messages.
* Made not generate strings unless they will actually be used. ASTERISK-26672 Change-Id: I155fbe7fdff5ce47dfe5326f3baf5446849702c3 |
9 years ago |
|
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75a6afbec5 |
MESSAGE: Flush Message/ast_msg_queue channel alert pipe.
ASTERISK-25083 Change-Id: Id54baa57a8dbca84e29f28bcd2ffc0a5ac12d8b2 |
9 years ago |
|
|
4b233675d8 |
pjproject_bundled: Fix missing inclusion of symbols
Added back in a -g3, and an -O3 when DONT_OPTIMIZE is not set, to the CFLAGS. Not sure how they went missing. Also fixed an uninstall problem where we weren't removing the symlink from libasteriskpj.so.2 to libasteriskpj.so. While I was there, I fixed it for libasteriskssl as well. Change-Id: I9e00873b1e9082d05b5549d974534b48a2142556 |
9 years ago |
|
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8d56016ae4 |
Merge "tcptls: Use new certificate upon sip reload" into 13
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9 years ago |
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28b76ed667 |
Merge "PJPROJECT logging: Made easier to get available logging levels." into 13
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9 years ago |
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c3a509be7e |
Merge "res_rtp: Fix regression when IPv6 is not available." into 13
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9 years ago |
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bddc60df73 |
Merge "Frame deferral: Re-queue deferred frames one-at-a-time." into 13
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9 years ago |
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662a4741e2 |
Merge "OpenSSL 1.1.0 support" into 13
|
9 years ago |
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2ceb609edb |
res_rtp: Fix regression when IPv6 is not available.
The latest Release candidate fails to create RTP streams when IPv6 is not available. Due to the changes made in September the ast_sockaddr structure passed around to create these streams is always of AF_INET6 type, causing failure when used for IPv4. This patch adds a utility function to check for availability of IPv6 and applies such check at startup to determine how to create the ast_sockaddr structures. ASTERISK-26617 #close Change-Id: I627a4e91795e821111e1cda523f083a40d0e0c3e |
9 years ago |
|
|
44fe4a5769 |
PJPROJECT logging: Made easier to get available logging levels.
Use of the new logging is as simple as issuing the new CLI command or setting the new pjproject.conf option. Other options that can affect the logging are how you have the pjproject log levels mapped to Asterisk log types in pjproject.conf and if you have configured Asterisk to log the DEBUG type messages. Altering the pjproject.conf level mapping shouldn't be necessary for most installations as the default mapping is sensible. Configuring Asterisk to log the DEBUG message type is standard practice for collecting debug information. * Added CLI "pjproject set log level" command to dynamically adjust the maximum pjproject log message level. * Added CLI "pjproject show log level" command to see the currently set maximum pjproject log message level. * Added pjproject.conf startup section "log_level" option to set the initial maximum pjproject log message level so all messages could be captured from initialization. * Set PJ_LOG_MAX_LEVEL to 6 to compile in all defined logging levels into bundled pjproject. Pjproject will use the currently set run time log level to determine if a log message is generated just like Asterisk verbose and debug logging levels. * In log_forwarder(), made always log enabled and mapped pjproject log messages. DEBUG mapped log messages are no longer gated by the current Asterisk debug logging level. * Removed RAII_VAR() from res_pjproject.c:get_log_level(). ASTERISK-26630 #close Change-Id: I6dca12979f482ffb0450aaf58db0fe0f6d2e5389 |
9 years ago |
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17b0b91afa |
Frame deferral: Re-queue deferred frames one-at-a-time.
The recent change that made frame deferral into an API had a behavior change to it. When frame deferral was completed, we would take all of the deferred frames and queue them all onto the channel in one call to ast_queue_frame_head(). Before frame deferral was API-ized, places that performed manual frame deferral would actually take each deferred frame and queue them onto the channel. This change in behavior caused the confbridge_recording test to start failing consistently. Without going too crazily deep into the details, a channel was getting "stuck" in an ast_safe_sleep(). An AMI redirect was attempting to break it out of the sleep, but because there were more frames in the channel read queue than expected, the channel ended up being unable to break from its sleep loop. By restoring the behavior of individual frame queuing after deferral, the test starts passing again. Note, this points to a potential underlying issue pointing to an "unbalance" that can occur when queuing multiple frames at once, and so a follow-up issue is being created to investigate that possibility. Change-Id: Ied5dacacda06d343dea751ed5814a03364fe5a7d |
9 years ago |
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eec82c6221 |
Merge "chan_pjsip: fix switching sending codec when asymmetric_rtp_codec=no" into 13
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9 years ago |
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b0c9f07f04 |
OpenSSL 1.1.0 support
OpenSSL 1.1.0 includes some major changes in the interface. See https://wiki.openssl.org/index.php/1.1_API_Changes . Status: Right now there are still a few deprecation notes with OpenSSL 1.1.0. But it's a start. Changes: * CRYPTO_LOCK is no longer available. Replace it with its value for now. I don't completely understand what it is used for there. * Remove several functions from libasteriskssl that seem to no longer be needed. * Structures have become opaque and are accesses with accessors. * ERR_remove_thread_state() no longer needed. * SSLv2 code now could no longer be used in 1.1. ASTERISK-26109 #close Change-Id: I5e29d477d486ca29b6aae0dc2f5dff960c1cb82b |
9 years ago |
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0cc8351484 |
chan_sip: Fix segfault during module unload
If a TCP/TLS connection was pending (not accepted and not timed out) during unload of chan_sip, Asterisk would segfault when trying to send a signal to a thread whose thread ID hadn't been recorded yet. This commit fixes that by recording the thread ID before calling the blocking connect() syscall. This was a regression introduced by |
9 years ago |
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8756ce64b7 |
res_rtp_asterisk: RTT miscalculation in RTCP
When retrieving RTCP stats for PJSIP channels, RTT values are unreliable. RTT calculation is correct, but the data representation isn't. RTT is represented by a 32-bit fixed-point number with the integer part in the first 16 bits and the fractional part in the last 16 bits. In order to get the RTT value, the fractional part is miscalculated, there is an unnecessary 16 bit shift that causes overflow. Besides this there is another mistake, when transforming the integer value to the fixed point fractional part via bitwise operation, that loses precision. * RTT fractional part is no longer shifted, avoiding overflow. * RTT fractional part is transformed to its fixed-point value more precisely. * Fixed timeval2ntp() and ntp2timeval() second fraction conversions. * Fixed NTP timestamp report logging. The usec was inexplicably multiplied by 4096. ASTERISK-26566 #close Reported by Hector Royo Concepcion Change-Id: Ie09bdabfee75afb3f1b8ddfd963e5219ada3b96f |
9 years ago |
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8e77d6f520 |
tcptls: Use new certificate upon sip reload
Previously, a TLS server socket would only be restarted upon sip reload if the bind address had changed. This commit adds checking for changes to TLS parameters like certificate, ciphers, etc. so they get picked up without requiring a reload of the entire chan_sip module. This does not affect open connections in any way, but new connections will use the new TLS parameters. The changes also apply to HTTP and Manager. ASTERISK-26604 #close Change-Id: I169e86cefc6dcd627c915134015a6a1ab1aadbe6 |
9 years ago |
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425da14927 |
build: Backport addition of librt check to configure.ac
A while back, a master-only change was made to check for librt which should probably have been cherry-picked to 13 at that time. Sometime between then and now, part of that change did make it into 13 but it was incomplete and non-functional. This patch backports the rest of the librt check and allows the link of libasteriskpj to use the results. Change-Id: I1424008fd8c90f389dda53162ec4a340b253a3c1 |
9 years ago |
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5c9680da9d |
Merge "main/app.c: Transmit Silence on ControlPlayback pause" into 13
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9 years ago |
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12c4e664bc |
main/app.c: Transmit Silence on ControlPlayback pause
ASTERISK-26562 #close Change-Id: Ie6cb0ffc2b8c775639ce7784fe96f4ea00cfa2f8 |
9 years ago |
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b213045fe4 |
build: Various OpenBSD issues
OpenBSD's 'find' doesn't take the -delete argument so you have to pipe through 'xargs rm -rf'. 'echo -e' doesn't like \t starting a line. It just prints 't' which causes the libasteriskpj.exports file to be garbage. They were just cosmetic so they were removed. librt doesn't exist so the link of libasteriskpj.so fails. It's not actually needed for linux anyway so -lrt was removed from the link. res_rtp_asterisk was failing to load because of an undefined DTLS_method. '|| defined(LIBRESSL_VERSION_NUMBER)' was added to the #if so DTLSv1_method is used instead. ASTERISK-26608 Change-Id: I926ec95b0b69633231e3ad1d6e803b977272c49c |
9 years ago |
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c877801e5d |
Merge "channel: Fix issues in hangup scenarios caused by frame deferral" into 13
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9 years ago |
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404596b790 |
channel: Fix issues in hangup scenarios caused by frame deferral
ASTERISK-26343 Change-Id: I06dbf7366e26028251964143454a77d017bb61c8 |
9 years ago |
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19ca0e7fc8 |
Merge "codec_opus: Fix warning when Opus negotiated but codec_opus not loaded." into 13
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9 years ago |
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b745c326c2 |
Merge "res/ari/resource_bridges: Add the ability to manipulate the video source" into 13
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9 years ago |
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4c1b9fbefe |
Merge "Revert "Revert "channel: Use frame deferral API for safe sleep.""" into 13
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9 years ago |
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d0a72d3779 |
Merge "Revert "Revert "autoservice: Use frame deferral API""" into 13
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9 years ago |
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36b59ee9c1 |
Merge "Revert "Revert "AGI: Only defer frames when in an interception routine.""" into 13
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9 years ago |
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f68790d46a |
Merge "Revert "Revert "Add API for channel frame deferral.""" into 13
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9 years ago |
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ed0f1afc8c |
codec_opus: Fix warning when Opus negotiated but codec_opus not loaded.
When Opus is negotiated but not loaded, the log is spammed with messages because the system does not know how to calculate the number of samples in a frame. * Suppress the warning by supplying a function that assumes 20ms of samples in the frame. For pass through support it doesn't really seem to matter what number of samples is returned anyway. ASTERISK-26605 #close Change-Id: Icf2273692f040dc2c45b01e72a790d11092f9e0f |
9 years ago |
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4c3d25875f |
Merge "Add X.509 subject alternative name support to TLS certificate verification." into 13
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9 years ago |
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6672fd5faa |
Merge "cli: Fix ast_el_read_char to work with libedit >= 3.1" into 13
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9 years ago |
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cf6d13180e |
chan_pjsip: fix switching sending codec when asymmetric_rtp_codec=no
The sending codec is switched to the receiving codec and then is switched back to the best native codec on EVERY receiving RTP packets. This is because after call of ast_channel_set_rawwriteformat there is call of ast_set_write_format which calls set_format which sets rawwriteformat to the best native format. This patch adds a new function ast_set_write_format_path which set specific write path on channel and uses this function to switch the sending codec. ASTERISK-26603 #close Change-Id: I5b7d098f8b254ce8f45546e6c36e5d324737f71d |
9 years ago |
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c92dcc76da |
file.c/__ast_file_read_dirs: Fix issues on filesystems without d_type
One of the code paths in __ast_file_read_dirs will only get executed if the OS doesn't support dirent->d_type OR if the filesystem the particular file is on doesn't support it. So, while standard Linux systems support the field, some filesystems like XFS do not. In this case, we need to call stat() to determine whether the directory entry is a file or directory so we append the filename to the supplied directory path and call stat. We forgot to truncate path back to just the directory afterwards though so we were passing a complete file name to the callback in the dir_name parameter instead of just the directory name. The logic has been re-written to only create a full_path if we need to call stat() or if we need to descend into another directory. Change-Id: I54e4228bd8355fad65200c6df3ec4c9c8a98dfba |
9 years ago |
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7b96e8cc3d |
Add X.509 subject alternative name support to TLS certificate
verification. This way one X.509 certificate can be used for hosts that can be reached under multiple DNS names or for multiple hosts. Signed-off-by: Maciej Szmigiero <mail@maciej.szmigiero.name> ASTERISK-25063 #close Change-Id: I13302c80490a0b44c43f1b45376c9bd7b15a538f |
9 years ago |
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d23b4af477 |
res/ari/resource_bridges: Add the ability to manipulate the video source
In multi-party bridges, Asterisk currently supports two video modes:
* Follow the talker, in which the speaker with the most energy is shown
to all participants but the speaker, and the speaker sees the
previous video source
* Explicitly set video sources, in which all participants see a locked
video source
Prior to this patch, ARI had no ability to manipulate the video source.
This isn't important for two-party bridges, in which Asterisk merely
relays the video between the participants. However, in a multi-party
bridge, it can be advantageous to allow an external application to
manipulate the video source.
This patch provides two new routes to accomplish this:
(1) setVideoSource: POST /bridges/{bridgeId}/videoSource/{channelId}
Sets a video source to an explicit channel
(2) clearVideoSource: DELETE /bridges/{bridgeId}/videoSource
Removes any explicit video source, and sets the video mode to talk
detection
ASTERISK-26595 #close
Change-Id: I98e455d5bffc08ea5e8d6b84ccaf063c714e6621
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9 years ago |
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404a62eeee |
Revert "Revert "channel: Use frame deferral API for safe sleep.""
This reverts commit
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9 years ago |
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09d8febc91 |
Revert "Revert "autoservice: Use frame deferral API""
This reverts commit
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9 years ago |
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ffad2b44df |
Revert "Revert "AGI: Only defer frames when in an interception routine.""
This reverts commit
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9 years ago |
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2fefb6187f |
Revert "Revert "Add API for channel frame deferral.""
This reverts commit
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9 years ago |
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5e0c224043 |
cli: Fix ast_el_read_char to work with libedit >= 3.1
Libedit 3.1 is not build with unicode on as a default and so the prototype for the el_gets callback changed from expecting a char buffer to accepting a wchar buffer. If ast_el_read_char isn't changed, the cli reads garbage from teh terminal. Added a configure test for (*el_rfunc_t)(EditLine *, wchar_t *) and updated ast_el_read_char to use the HAVE_ define to detemrine whether to use char or wchar. ASTERISK-26592 #close Change-Id: I9099b46f68e06d0202ff80e53022a2b68b08871a |
9 years ago |
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ccc8a37584 |
Merge "Revert "autoservice: Use frame deferral API"" into 13
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9 years ago |