5565 Commits (16fdb11bc3bb23a2fd645a24a7ba56d9708960ff)

Author	SHA1	Message	Date
Richard Mudgett	16fdb11bc3	core: Cleanup some channel snapshot staging anomalies. ... We shouldn't unlock the channel after starting a snapshot staging because another thread may interfere and do its own snapshot staging. * app_dial.c:dial_exec_full() made hold the channel lock while setting up the outgoing channel staging. Made hold the channel lock after the called party answers while updating the caller channel staging. * chan_sip.c:sip_new() completed the channel staging on off-nominal exit. Also we need to use ast_hangup() instead of ast_channel_unref() at that location. * channel.c:__ast_channel_alloc_ap() added a comment about not needing to complete the channel snapshot staging on off-nominal exit paths. * rtp_engine.c:ast_rtp_instance_set_stats_vars() made hold the channel locks while staging the channels for the stats channel variables. Change-Id: Iefb6336893163f6447bad65568722ad5d5d8212a	10 years ago
Joshua Colp	5422ec140c	srv: Fix crash when ast_srv_lookup is used and 0 records are returned. ... When performing an SRV lookup using the ast_srv_lookup function it did not properly handle the situation where 0 records are returned. If this happened it would wrongly assume that at least one record was present. This change fixes the code so it will exit early if an error occurs or if 0 records are returned. ASTERISK-26772 patches: srv_lookup.patch submitted by nappsoft (license 6822) Change-Id: I09b19081c74e0ad11c12bf54a257243b1bcb2351	10 years ago
Sebastien Duthil	7b280e7ccf	res_ari: fix memory leak for channelvars ... In ari.conf, when setting the option channelvars, every Stasis channel snapshot would create a list of variable/value that would not be freed when the snapshot is freed, resulting in a often-recurring memory leak. ASTERISK-26767 #close Change-Id: Ia37dd9d68063d7f879193df02ede293e5ded716d	10 years ago
Richard Mudgett	50029f585e	channel.c: Fix unbalanced read queue deadlocking local channels. ... Using the timerfd timing module can cause channel freezing, lingering, or deadlock issues. The problem is because this is the only timing module that uses an associated alert-pipe. When the alert-pipe becomes unbalanced with respect to the number of frames in the read queue bad things can happen. If the alert-pipe has fewer alerts queued than the read queue then nothing might wake up the thread to handle received frames from the channel driver. For local channels this is the only way to wake up the thread to handle received frames. Being unbalanced in the other direction is less of an issue as it will cause unnecessary reads into the channel driver. ASTERISK-26716 is an example of this deadlock which was indirectly fixed by the change that found the need for this patch. * In channel.c:__ast_queue_frame(): Adding frame lists to the read queue did not add the same number of alerts to the alert-pipe. Correspondingly, when there is an exceptionally long queue event, any removed frames did not also remove the corresponding number of alerts from the alert-pipe. ASTERISK-26632 #close Change-Id: Ia98137c5bf6e9d6d202ce0eb36441851875863f6	10 years ago
Richard Mudgett	97c308471d	res_agi: Prevent an AGI from eating frames it should not. (Re-do) ... A dialplan intercept routine is equivalent to an interrupt routine. As such, the routine must be done quickly and you do not have access to the media stream. These restrictions are necessary because the media stream is the responsibility of some other code and interfering with or delaying that processing is bad. A possible future dialplan processing architecture change may allow the interception routine to run in a different thread from the main thread handling the media and remove the execution time restriction. * Made res_agi.c:run_agi() running an AGI in an interception routine run in DeadAGI mode. No touchy channel frames. ASTERISK-25951 ASTERISK-26343 ASTERISK-26716 Change-Id: I638f147ca7a7f2590d7194a8ef4090eb191e4e43	10 years ago
Richard Mudgett	72e3fc5845	Frame deferral: Revert API refactoring. ... There are several issues with deferring frames that are caused by the refactoring. 1) The code deferring frames mishandles adding a deferred frame to the deferred queue. As a result the deferred queue can only be one frame long. 2) Deferrable frames can come directly from the channel driver as well as the read queue. These frames need to be added to the deferred queue. 3) Whoever is deferring frames is really only doing the __ast_read() to collect deferred frames and doesn't care about the returned frames except to detect a hangup event. When frame deferral is completed we must make the normal frame processing see the hangup as a frame anyway. As such, there is no need to have varying hangup frame deferral methods. We also need to be aware of the AST_SOFTHANGUP_ASYNCGOTO hangup that isn't real. That fake hangup is to cause the PBX thread to break out of loops to go execute a new dialplan location. 4) To properly deal with deferrable frames from the channel driver as pointed out by (2) above, means that it is possible to process a dialplan interception routine while frames are deferred because of the AST_CONTROL_READ_ACTION control frame. Deferring frames is not implemented as a re-entrant operation so you could have the unsupported case of two sections of code thinking they have control of the media stream. A worse problem is because of the bad implementation of the AMI PlayDTMF action. It can cause two threads to be deferring frames on the same channel at the same time. (ASTERISK_25940) * Rather than fix all these problems simply revert the API refactoring as there is going to be only autoservice and safe_sleep deferring frames anyway. ASTERISK-26343 ASTERISK-26716 #close Change-Id: I45069c779aa3a35b6c863f65245a6df2c7865496	10 years ago
Sean Bright	2849b726b6	audiohooks: Muting a hook can mute underlying frames ... If an audiohook is placed on a channel that does not require transcoding, muting that hook will cause the underlying frames to be muted as well. The original patch is from David Woolley but I have modified slightly. ASTERISK-21094 #close Reported by: David Woolley Patches: ASTERISK-21094-Patch-1.8-1.txt (license #5737) patch uploaded by David Woolley Change-Id: Ib2b68c6283e227cbeb5fa478b2d0f625dae338ed	10 years ago
zuul	6cbe894828	Merge "debug_utilities: Add ast_logescalator"	10 years ago
George Joseph	ef4deb8ecd	debug_utilities: Add ast_logescalator ... The escalator works by creating a set of startup commands in cli.conf that set up logger channels and issue the debug commands for the subsystems specified. If asterisk is running when it is executed, the same commands will be issued to the running instance. The original cli.conf is saved before any changes are made and can be restored by executing '$prog --reset'. The log output will be stored in... $astlogdir/message.$uniqueid $astlogdir/debug.$uniqueid $astlogdir/dtmf.$uniqueid $astlogdir/fax.$uniqueid $astlogdir/security.$uniqueid $astlogdir/pjsip_history.$uniqueid $astlogdir/sip_history.$uniqueid Some minor tweaks were made to chan_sip, and res_pjsip_history so their history output could be send to a log channel as packets are captured. A minor tweak was also made to manager so events are output to verbose when "manager set debug on" is issued. Change-Id: I799f8e5013b86dc5282961b27383d134bf09e543	10 years ago
Torrey Searle	178b90af02	libastssl/pj: libastssl/pj should have an so_version ... Issue introduced in b59956a87. In the non-darwin case libastssl/pj should be versioned. This causes the symbol file for this lib to not be generated. Change-Id: Ib07ae8c40252813c488e2c1ac6204fd42816dd4c (cherry picked from commit 54b027916a)	10 years ago
George Joseph	6f645a6d4e	Merge "media: Add experimental support for RTCP feedback."	10 years ago
zuul	10631bb209	Merge "PJPROJECT logging: Fix detection of max supported log level."	10 years ago
George Joseph	0ad6d2b3cf	Merge "ari: Implement 'debug all' and request/response logging"	10 years ago
George Joseph	2484c3ee39	Merge "frame.c: Fix off-nominal format ref leaks."	10 years ago
George Joseph	96dbf54e97	Merge "T.140: Fix format ref and memory leaks."	10 years ago
Joshua Colp	3abb17d172	Merge "main/app.c: Memory corruption from early format destruction."	10 years ago
Richard Mudgett	20aed30d9a	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	10 years ago
Richard Mudgett	930a24a730	astobj2.c: Add excessive ref count trap. ... Change-Id: I32e6a589cf9009450e4ff7cb85c07c9d9ef7fe4a	10 years ago
Richard Mudgett	de28c1b9f1	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	10 years ago
Richard Mudgett	2039eb8edf	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	10 years ago
Richard Mudgett	6f3e8c8e01	PJPROJECT logging: Fix detection of max supported log level. ... The mechanism used for detecting the maximum log level compiled into the linked pjproject did not work. The API call simply stores the requested level into an integer and does no range checking. Asterisk was assuming that there was range checking and limited the new value to the allowable range. To get the actual maximum log level compiled into the linked pjproject we need to get and save off the initial set log level from pjproject. This is the maximum log level supported. * Get and save off the initial log level setting before altering it to the desired level on startup. This has to be done by a macro rather than calling a core function to avoid incorrectly linking pjproject. * Split the initial log level warning messages to warn if the linked pjproject cannot support the requested startup level and if it is too low to get the pjproject buildopts for "pjproject show buildopts". * Adjust the CLI "pjproject set log level" to check the saved max log level and to generate normal output messages instead of a warning message. ASTERISK-26743 #close Change-Id: I40aa76653e2a1dece66c3f8734594b4f0471cfb4	10 years ago
George Joseph	6691606723	ari: Implement 'debug all' and request/response logging ... The 'ari set debug' command has been enhanced to accept 'all' as an application name. This allows dumping of all apps even if an app hasn't registered yet. To accomplish this, a new global_debug global variable was added to res/stasis/app.c and new APIs were added to set and query the value. 'ari set debug' now displays requests and responses as well as events. This required refactoring the existing debug code. * The implementation for 'ari set debug' was moved from stasis/cli.{c,h} to ari/cli.{c,h}, and stasis/cli.{c,h} were deleted. * In order to print the body of incoming requests even if a request failed, the consumption of the body was moved from the ari stubs to ast_ari_callback in res_ari.c and the moustache templates were then regenerated. The body is now passed to ast_ari_invoke and then on to the handlers. This results in code savings since that template was inserted multiple times into all the stubs. An additional change was made to the ao2_str_container implementation to add partial key searching and a sort function. The existing cli code assumed it was already there when it wasn't so the tab completion was never working. Change-Id: Ief936f747ce47f1fb14035fbe61152cf766406bf (cherry picked from commit 1d890874f3)	10 years ago
Lorenzo Miniero	1061539b75	media: Add experimental support for RTCP feedback. ... This change adds experimental support for providing RTCP feedback information to codec modules so they can dynamically change themselves based on conditions. ASTERISK-26584 Change-Id: Ifd6aa77fb4a7ff546c6025900fc2baf332c31857	10 years ago
Kevin Harwell	283c16c6b6	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	10 years ago
Richard Mudgett	f4e77a5678	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	10 years ago
zuul	6962a13466	Merge "core/pbx: dialplan show - display filename/line#"	10 years ago
Jonathan R. Rose	d96e350256	core/pbx: dialplan show - display filename/line# ... Adds the ability for extensions to be registered to include filename and line number so that dialplan show output can show the filename and line number of a config file responsible for generating a given extension. This only affects config modules that are written to use the new extension registering functions. In this patch, that only includes pbx_config, so extensions registered in extensions.conf and any included extension will be shown in this manner. Extensions registered in this manner will show the filename and line number *instead* of the registrar. ASTERISK-26658 #close Reported by: Jonathan R. Rose Change-Id: Ieccc6abccdff34ed5c7da3511fd24972b8f2dd30	10 years ago
Richard Mudgett	67cc8499a2	acl.c: Improve ast_ouraddrfor() diagnostic messages. ... * Made not generate strings unless they will actually be used. ASTERISK-26672 Change-Id: I155fbe7fdff5ce47dfe5326f3baf5446849702c3	10 years ago
Richard Mudgett	44e72c9d44	MESSAGE: Flush Message/ast_msg_queue channel alert pipe. ... ASTERISK-25083 Change-Id: Id54baa57a8dbca84e29f28bcd2ffc0a5ac12d8b2	10 years ago
Joshua Colp	18fe80e776	Merge "http: Send headers and body in one write."	10 years ago
Joshua Colp	119c41d001	Merge "Iostreams: Correct off-by-one error."	10 years ago
Mark Michelson	503006123a	http: Send headers and body in one write. ... This is a semi-regression caused by the iostreams change. Prior to iostreams, HTTP headers were written to a FILE handle using fprintf. Then the body was written using a call to fwrite(). Because of internal buffering, the result was that the HTTP headers and body would be sent out in a single write to the socket. With the change to iostreams, the HTTP headers are written using ast_iostream_printf(), which under the hood calls write(). The HTTP body calls ast_iostream_write(), which also calls write() under the hood. This results in two separate writes to the socket. Most HTTP client libraries out there will handle this change just fine. However, a few of our testsuite tests started failing because of the change. As a result, in order to reduce frustration for users, this change alters the HTTP code to write the headers and body in a single write operation. ASTERISK-26629 #close Reported by Joshua Colp Change-Id: Idc2d2fb3d9b3db14b8631a1e302244fa18b0e518	10 years ago
Mark Michelson	bf6423a336	Iostreams: Correct off-by-one error. ... ast_iostream_printf() attempts first to use a fixed-size buffer to perform its printf-like operation. If the fixed-size buffer is too small, then a heap allocation is used instead. The heap allocation in this case was exactly the length of the string to print. The issue here is that the ensuing call to vsnprintf() will print a NULL byte in the final space of the string. This meant that the final character was being chopped off the string and replaced with a NULL byte. For HTTP in particular, this caused problems because HTTP publishes the expected Contact-Length. This meant HTTP was publishing a length one character larger than what was actually present in the message. This patch corrects the issue by adding one to the allocation length. ASTERISK-26629 Reported by Joshua Colp Change-Id: Ib3c5f41e96833d0415cf000656ac368168add639	10 years ago
George Joseph	fe9f070885	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	10 years ago
Joshua Colp	faf2194fab	Merge "app_originate: Add option to execute gosub prior to dial"	10 years ago
Joshua Colp	cd5d9d1d69	Merge "tcptls: Use new certificate upon sip reload"	10 years ago
Joshua Colp	197e408395	Merge "PJPROJECT logging: Made easier to get available logging levels."	10 years ago
Joshua Colp	2679f80d3c	Merge "res_rtp: Fix regression when IPv6 is not available."	10 years ago
Joshua Colp	02588a1aab	Merge "Frame deferral: Re-queue deferred frames one-at-a-time."	10 years ago
zuul	68fc035795	Merge "OpenSSL 1.1.0 support"	10 years ago
Tzafrir Cohen	26c8552fff	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	10 years ago
Guido Falsi	75230f4c01	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	10 years ago
Richard Mudgett	1dfa11b65c	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	10 years ago
Mark Michelson	621d886ca7	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	10 years ago
zuul	a0c0b1c9cb	Merge "chan_pjsip: fix switching sending codec when asymmetric_rtp_codec=no"	10 years ago
Joshua Colp	bd20127e64	Merge "chan_sip: Fix segfault during module unload"	10 years ago
Alexei Gradinari	e5e887be53	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	10 years ago
David Kerr	ddc951060a	app_originate: Add option to execute gosub prior to dial ... Issue/patch ASTERISK-26587 was inspired by issue ASTERISK-22992 that requested ability to add callerid into app_originate. Comments in that issue suggested that it was better solved by adding an option to gosub prior to originating the call. The attached patch implements this much like app_dial with two options one to gosub on the originating channel and one to gosub on the newly created channel and behaves just like app_dial. I have tested this patch by adding callerid info to the new channel and also SIPAddHeader (to e.g. add header to force auto answer) and confirmed it works. Have also tested both 'exten' and 'app' versions of app_originate. Opened by: dkerr Patch by: dkerr Change-Id: I36abc39b58567ffcab4a636ea196ef48be234c57	10 years ago
Joshua Colp	c9cc64b911	Merge "ast_format: Adds an identifier for interleaved audio formats to the ast_format"	10 years ago
Joshua Colp	e3dae763ee	iostream: Move include of asterisk.h ... The asterisk.h header file needs to be included first or else some things go awry, such as: implicit declaration of function 'vasprintf' Change-Id: I981dc2a77a1ba791888e4f1726644d4656c0407c	10 years ago