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Author	SHA1	Message	Date
Joshua Colp	60f2e96d62	Merge "audiohooks: Muting a hook can mute underlying frames" into 13	9 years ago
Joshua Colp	4e18f9afac	Merge "res_pjsip: Handle invocation of callback on outgoing request when error occurs." into 13	9 years ago
Sean Bright	3aee199913	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	9 years ago
zuul	e4f742de15	Merge "res_rtp_asterisk: Swap byte-order when sending signed linear" into 13	9 years ago
Joshua Colp	e371e13b9e	res_pjsip: Handle invocation of callback on outgoing request when error occurs. ... There are some error cases in PJSIP when sending a request that will result in the callback for the request being invoked. The code did not handle this case and assumed on every error case that the callback was not invoked. The code has been changed to check whether the callback has been invoked and if so to absorb the error and treat it as a success. ASTERISK-26679 ASTERISK-26699 Change-Id: I563982ba204da5aa1428989a11c06dd9087fea91	9 years ago
Sean Bright	339c30f2b6	res_rtp_asterisk: Swap byte-order when sending signed linear ... Before Asterisk 13, signed linear was converted into network byte order by a smoother before being sent over the network. We restore this behavior by forcing the creation of a smoother when slinear is in use and setting the appropriate flags so that the byte order conversion is always done. ASTERISK-24858 #close Reported-by: Frankie Chin Change-Id: I868449617d1a7819578f218c8c6b2111ad84f5a9	9 years ago
George Joseph	7fd28cefdb	debug_utilities: Install ast_logescalator to /var/lib/asterisk/scripts ... Forgot to install it with the original patch Change-Id: I8bdb540a6694971ae5fe21f48d532332c6482e4c	9 years ago
zuul	ea54c0e1fe	Merge "make_build_h: handle backslashes in external strings" into 13	9 years ago
Joshua Colp	d2f30f0225	Merge "ast_careful_fwrite to support EPIPE gracefully" into 13	9 years ago
zuul	fd3f69cd05	Merge "app_queue: Fix queues randomly disappearing on reload" into 13	9 years ago
zuul	56e925f96a	Merge "libastssl/pj: libastssl/pj should have an so_version" into 13	9 years ago
zuul	484e8ed5e3	Merge "debug_utilities: Add ast_logescalator" into 13	9 years ago
George Joseph	456bc3c704	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	9 years ago
George Joseph	bcb2c1f62f	Merge "tests: use datadir for sound files" into 13	9 years ago
Torrey Searle	54b027916a	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	9 years ago
George Joseph	d6f67df8ce	Merge "test_voicemail_api: order of params to VERIFY macros" into 13	9 years ago
George Joseph	e28d3c4917	Merge "res_pjsip_endpoint_identifier_ip: Fix memory leak of hosts when resolving." into 13	9 years ago
George Joseph	cb504ad6ae	Merge "ari: Implement 'debug all' and request/response logging" into 13	9 years ago
kkm	3c8f84786e	make_build_h: handle backslashes in external strings ... LikewiseOpen creates user names with a backslash in them. A gentle massage with sed(1) allows such strings to be inserted into build.h properly quoted. I am also adding the same for host name and other strings used in the script that are more or less user-controlled. ASTERISK-26754 Change-Id: Iac5ef2b67a68ee58f35ddbf86bb818ba6eabecae	9 years ago
kkm	555e8cd2ba	ast_careful_fwrite to support EPIPE gracefully ... When a reading end of the network socket is closed by an AMI manager, the EPIPE is signaled when writing to our end, resulting in the spurious log error message ast_careful_fwrite: fwrite() returned error: Broken pipe Previously EPIPE was handled in ast_carefulwrite() a few lines above, but not in this function. ASTERISK-26753 Change-Id: I6a67335cd6526608bb9b78f796c626b1677664b8	9 years ago
kkm	be92f10a16	app_queue: Fix queues randomly disappearing on reload ... With 500+ queues and a reload every minute, a random queue disappears upon reload. The cause is mususe of the 'dead' flag. Namely, all queues were marked dead up front, and then "resurrected" by dropping this flag for those found in the configuration. But a queue marked dead can be removed also when control leaves the app entry point on a PBX thread. With this change, the queue is marked only not found, and at the end of reload only the queues that are still not found are actually marked as dead, so the dead flag is never reset, and set only on positively dead queues. ASTERISK-26755 Change-Id: I3a4537aec9eb8d8aeeaa0193407e3523feb004bf	9 years ago
George Joseph	b39c04fc02	Merge "PJPROJECT logging: Fix detection of max supported log level." into 13	9 years ago
George Joseph	f5b31cb2ae	Merge "frame.c: Fix off-nominal format ref leaks." into 13	9 years ago
zuul	1a81ba335b	Merge "res_musiconhold.c: Fix format ref leak when parsing MOH config class." into 13	9 years ago
Joshua Colp	616e98ce0a	Merge "T.140: Fix format ref and memory leaks." into 13	9 years ago
Joshua Colp	aae9df0643	res_pjsip_endpoint_identifier_ip: Fix memory leak of hosts when resolving. ... This change adds a missing unreference of the hostname when resolving and also cleans up the iterator. ASTERISK-26735 Change-Id: Ic012ebaf3d89e714eec340b7b0c5e63c66af857a	9 years ago
zuul	a7e6f0dee5	Merge "Add notes about embedded ast_frame structs holding a format ref." into 13	9 years ago
George Joseph	da098e2227	Merge "astobj2.c: Add excessive ref count trap." into 13	9 years ago
zuul	3410000948	Merge "main/app.c: Memory corruption from early format destruction." into 13	9 years ago
zuul	89e9d2cfb8	Merge "stasis_bridge.c: Fix off-nominal stasis control ref leak." into 13	9 years ago
Richard Mudgett	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
zuul	e51193bf72	Merge "chan_oss.c: Fix format ref leak in oss_read()." into 13	9 years ago
George Joseph	f93861b204	Merge "res_pjsip_endpoint_identifier_ip: Ensure error defaults to 0." into 13	9 years ago
George Joseph	f2c1ee6d71	Merge "pjproject_bundled: Fix setting max log level" into 13	9 years ago
George Joseph	284b8b981e	Merge "Free endpoint ACLs when destroying PJSIP endpoints." into 13	9 years ago
George Joseph	7b06dae2d8	Merge "res_pjsip_pubsub.c: Implement "pjsip show subscriptions" commands." into 13	9 years ago
Joshua Colp	a2f0adccbd	res_pjsip_endpoint_identifier_ip: Ensure error defaults to 0. ... When configuring a match using a netmask the error variable was not defaulting to 0. For some people this would cause the code to think an error occurred when adding the match when in reality it added perfectly fine. ASTERISK-26693 Change-Id: I850c250813742bddde65c84e739093c9e01dfe56	9 years ago
Richard Mudgett	607b3ac736	astobj2.c: Add excessive ref count trap. ... Change-Id: I32e6a589cf9009450e4ff7cb85c07c9d9ef7fe4a	9 years ago
Richard Mudgett	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
Richard Mudgett	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
Richard Mudgett	00a227e93d	stasis_bridge.c: Fix off-nominal stasis control ref leak. ... Change-Id: Ib17218343a6596832060180e19386da9df150ac8	9 years ago
Richard Mudgett	38a2021c68	res_musiconhold.c: Fix format ref leak when parsing MOH config class. ... Change-Id: Ica8e8e2ce7604c2c61ec55bef07dc675361d2ea5	9 years ago
Richard Mudgett	ab7a9fc5b2	chan_oss.c: Fix format ref leak in oss_read(). ... Change-Id: I0a5d56c7dcf327d60f86a4c25a23571733709fd0	9 years ago
Richard Mudgett	1484a991e1	Add notes about embedded ast_frame structs holding a format ref. ... mod_format.h: Note ast_filestream.fr holds a format ref. translate.h: Note ast_trans_pvt.f holds a format ref. Change-Id: I86bda354d725207b41e08920355d7c31b2d7f749	9 years ago
George Joseph	17f4989d49	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	9 years ago
Richard Mudgett	30cb4eb57f	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	9 years ago
Tzafrir Cohen	cd2677f966	tests: use datadir for sound files ... Some (voicemail-related) tests API symlinks beep.gsm and other files from ast_config_AST_VAR_DIR. It should use ast_config_AST_DATA_DIR. ASTERISK-26740 #close Change-Id: Id49c56fb9e16df64b1a2b829693ca7601252df89	9 years ago
Tzafrir Cohen	b62f84bfb1	test_voicemail_api: order of params to VERIFY macros ... Fix order of parameters in calls to VM_API_INT_VERIFY and VM_API_STRING_VERIFY ASTERISK-26739 #close Change-Id: I30dc6b36893aadad6012be3f16f93aa5720870d6 Note: status: builds. Not tested any further.	9 years ago
George Joseph	48178e5918	Merge "res_pjsip_endpoint_identifier_ip: Read settings before resolving." into 13	9 years ago
Richard Mudgett	e3dcb9ddd9	res_pjsip_pubsub.c: Implement "pjsip show subscriptions" commands. ... ASTERISK-23828 #close Change-Id: Ifb8a3b61f447aedc58a8e6b36a810f7566018567	9 years ago