asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	c70d908db7	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
Richard Mudgett	c324e17d49	MESSAGE: Flush Message/ast_msg_queue channel alert pipe. ASTERISK-25083 Change-Id: Id54baa57a8dbca84e29f28bcd2ffc0a5ac12d8b2	9 years ago
George Joseph	90c0faa340	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
zuul	1de7e0eefa	Merge "tcptls: Use new certificate upon sip reload" into 14	9 years ago
Joshua Colp	7b1d990683	Merge "PJPROJECT logging: Made easier to get available logging levels." into 14	9 years ago
Joshua Colp	73f4df291c	Merge "res_rtp: Fix regression when IPv6 is not available." into 14	9 years ago
Joshua Colp	f1a7cf64e2	Merge "Frame deferral: Re-queue deferred frames one-at-a-time." into 14	9 years ago
Joshua Colp	02e3a69b07	Merge "OpenSSL 1.1.0 support" into 14	9 years ago
Guido Falsi	798179995b	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
Joshua Colp	12517ad1f3	Merge "chan_pjsip: fix switching sending codec when asymmetric_rtp_codec=no" into 14	9 years ago
Richard Mudgett	846c77b28c	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
Mark Michelson	2bc0b595a1	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
Tzafrir Cohen	0f33fa4ead	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
Alexei Gradinari	d49fbc5f4e	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
Michael Kuron	cbdf9cc948	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 `776a14386a`. The above wasn't enough to fix the segfault, which was now delayed to the point where connect() timed out. Therefore, it was necessary to also remove the SA_RESTART flag from the SIGURG sigaction so that pthread_kill() could be used to interruput the connect() syscall. This was a regression introduced by `5d313f51b9`. ASTERISK-26586 #close Change-Id: I76fd9d47d56e4264e2629bce8ec15fecba673e7b	9 years ago
gestoip2	f8f197a6ee	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
Michael Kuron	cc619d30bf	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
zuul	bba4204fa1	Merge "Add support for older name resolving version libraries like openBSD" into 14	9 years ago
George Joseph	34a763415f	pjproject_bundled: Use $(LIB_RT) for link of libasteriskpj libasteriskpj was hard coded to use -lrt but librt is linux specific so we now use the LIB_RT variable which gets set by configure. Change-Id: I41148884517e3031f7675a413d524c86e8614694	9 years ago
snuffy	729f1b63e2	Add support for older name resolving version libraries like openBSD Fix support of OS's like openBSD that use an older nameser.h, this change reverts the defines to the older style which on other systems is found in nameser_compat.h Tested on openBSD 6.0, Debian 8 ASTERISK-26608 #close Change-Id: Iffb36caab8c5aa9dece0ce2d009041f7b56cc86a	9 years ago
Joshua Colp	002a8e4f10	Merge "main/app.c: Transmit Silence on ControlPlayback pause" into 14	9 years ago
Joshua Colp	3a151b5205	Merge "build: Various OpenBSD issues" into 14	9 years ago
misha	4e64e4f35f	main/app.c: Transmit Silence on ControlPlayback pause ASTERISK-26562 #close Change-Id: Ie6cb0ffc2b8c775639ce7784fe96f4ea00cfa2f8	9 years ago
George Joseph	54d7e65014	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
Joshua Colp	0ee5b4dbe9	Merge "codec_opus: Fix warning when Opus negotiated but codec_opus not loaded." into 14	9 years ago
George Joseph	0825528322	Revert "Revert "AGI: Only defer frames when in an interception routine."" This reverts commit `7155020c96`. Change-Id: I4e1681d2ca8bf36eda849f129bc0dcbc39ef5bc6	9 years ago
George Joseph	6b2efc116f	Revert "Revert "autoservice: Use frame deferral API"" This reverts commit `8b9996a90c`. Change-Id: Ibb2fccfc37188dc54f3776c2d5b41cc425d5cd9c	9 years ago
George Joseph	f6acb765a8	Revert "Revert "channel: Use frame deferral API for safe sleep."" This reverts commit `97679ee846`. Change-Id: Ib68f76e75882c0230378c4fed97964988e690f3c	9 years ago
Joshua Colp	aa1482aed4	Merge "channel: Fix issues in hangup scenarios caused by frame deferral" into 14	9 years ago
Joshua Colp	caa207bc4f	Merge "Revert "Revert "Add API for channel frame deferral.""" into 14	9 years ago
zuul	fdea7fe9e4	Merge "res/ari/resource_bridges: Add the ability to manipulate the video source" into 14	9 years ago
George Joseph	42254406ad	Merge "cli: Fix ast_el_read_char to work with libedit >= 3.1" into 14	9 years ago
Richard Mudgett	d89a1645fc	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
George Joseph	d6f9e2b54a	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
Matt Jordan	62cbcb2e54	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	9 years ago
George Joseph	ac6051c302	channel: Fix issues in hangup scenarios caused by frame deferral ASTERISK-26343 Change-Id: I06dbf7366e26028251964143454a77d017bb61c8 (cherry picked from commit `0be46aaf6b`) (cherry picked from commit `7263a17ca0`) (cherry picked from commit `ebf8d94eb2`)	9 years ago
George Joseph	88111da235	Revert "Revert "Add API for channel frame deferral."" This reverts commit `b640f18a44`. Change-Id: I4ad8e7f66b3e6457ac07e7965822ae39ac08464d	9 years ago
Sebastien Duthil	eb5077fb26	res_ari: Add support for channel variables in ARI events. This works the same as for AMI manager variables. Set "channelvars=foo,bar" in your ari.conf general section, and then the channel variables "foo" and "bar" (along with their values), will appear in every Stasis websocket channel event. ASTERISK-26492 #close patches: ari_vars.diff submitted by Mark Michelson Change-Id: I5609ba239259577c0948645df776d7f3bc864229	9 years ago
George Joseph	e2046bfaf3	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
George Joseph	c7511029c3	Merge "Revert "AGI: Only defer frames when in an interception routine."" into 14	9 years ago
George Joseph	6384ac9f39	Merge "Revert "channel: Use frame deferral API for safe sleep."" into 14	9 years ago
George Joseph	b55bdb0740	Merge "Revert "Add API for channel frame deferral."" into 14	9 years ago
George Joseph	588336b738	Merge "Revert "autoservice: Use frame deferral API"" into 14	9 years ago
George Joseph	b640f18a44	Revert "Add API for channel frame deferral." This reverts commit `4a8b1940b8`. Multiple testsuite failures were detected after the fact. Change-Id: Ife9f86872d323adb9c068d2942339c7be9499257	9 years ago
George Joseph	7155020c96	Revert "AGI: Only defer frames when in an interception routine." This reverts commit `8d8323b142`. Multiple testsuite failures were detected after the fact. Change-Id: I322684ed82ba32cfce226a94fe19b87a4ec02567	9 years ago
George Joseph	8b9996a90c	Revert "autoservice: Use frame deferral API" This reverts commit `0288fba2f0`. Multiple testsuite failures were detected after the fact. Change-Id: I9951207019a0f528787f735ccfed7c4b76335505	9 years ago
George Joseph	97679ee846	Revert "channel: Use frame deferral API for safe sleep." This reverts commit `1db9e7886c`. Multiple testsuite failures were detected after the fact. Change-Id: Ifd5817e90371e05c4011d5cc97da145c196d60fe	9 years ago
Mark Michelson	1db9e7886c	channel: Use frame deferral API for safe sleep. This is another case where manual frame deferral can be replaced with centralized routines instead. Change-Id: I42cdf205f8f29a7977e599751a57efbaac07c30e (cherry picked from commit `d149c4b9e0`)	9 years ago
Mark Michelson	0288fba2f0	autoservice: Use frame deferral API Rather than use manual frame deferral, just let the channel API do it for us. ASTERISK-26343 Change-Id: I688386f36e765dbc07be863943a43f26bd5eac49 (cherry picked from commit `8ba3e2fc27`)	9 years ago
Mark Michelson	8d8323b142	AGI: Only defer frames when in an interception routine. AGI recently was modified to defer important frames. This was because when AGI was used in a connected line interception routine, the resulting connected line frame would end up getting discarded by the AGI. However, this caused bad behavior in other cases. Specifically, during a transfer, if someone attempted to manually set the Caller ID on a channel in an AGI, the deferred connected line frame would end up overwriting what had been manually set in the AGI. Since the initial issue was specific to interception routines, this change removes the manual frame deferral from AGI and instead uses the new frame deferral API in interception routines. ASTERISK-26343 #close Reported by Morton Tryfoss Change-Id: Iab7d39436d0ee99bfe32ad55ef91e9bd88db4208	9 years ago

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5514 Commits (8d4c188fda72de05be9424542dbb2a26110c9d32)