26860 Commits (077c58cd5c421edac3eb0f7cb453334bc51fa0ac)

 

All Branches   

Search

Author	SHA1	Message	Date
Richard Mudgett	077c58cd5c	res_pjsip_session.c: Fix crashes seen when call cancelled. ... Two testsuite tests crashed in the same place as a result of an INVITE being CANCELed. tests/channels/pjsip/resolver/srv/failover/in_dialog/transport_unspecified tests/channels/pjsip/resolver/srv/failover/in_dialog/transport_tcp The session pointer is no longer in the inv->mod_data[session_module.id] location because the INVITE transaction has reached the terminated state. ASTERISK-25297 #close Reported by: Richard Mudgett Change-Id: Idb75fdca0321f5447d5dac737a632a5f03614427	10 years ago
Jonathan Rose	687597ca8c	holding_bridge: ensure moh participants get frames ... Currently, if a blank musiconhold.conf is used, musiconhold will fail to start for a channel going into a holding bridge with an anticipation of getting music on hold. That being the case, no frames will be written to the channel and that can pose a problem for blind transfers in PJSIP which may rely on frames being written to get past the REFER framehook. This patch makes holding bridges start a silence generator if starting music on hold fails and makes it so that if no music on hold functions are installed that the ast_moh_start function will report a failure so that consumers of that function will be able to respond appropriately. ASTERISK-25271 #close Change-Id: I06f066728604943cba0bb0b39fa7cf658a21cd99 (cherry picked from commit 8458b8d441)	10 years ago
Joshua Colp	309dd2a409	pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options. ... This change adds support for the 'rtp_timeout' and 'rtp_timeout_hold' endpoint options. These allow the channel to be hung up if RTP is not received from the remote endpoint for a specified number of seconds. ASTERISK-25259 #close Change-Id: I3f39daaa7da2596b5022737b77799d16204175b9	10 years ago
Joshua Colp	a105461f9e	Merge "res_pjsip: Add rtp_keepalive to sample config file."	10 years ago
Mark Michelson	a0c31c7a05	res_pjsip: Add rtp_keepalive to sample config file. ... Change-Id: I5f62d0c5684f8b2335f9f8ac2d79ee04fbdafb19	10 years ago
Mark Michelson	d97bed46b7	Local channels: Alternate solution to ringback problem. ... Commit 54b25c80c8 solved an issue where a specific scenario involving local channels and a native local RTP bridge could result in ringback still being heard on a calling channel even after the call is bridged. That commit caused many tests in the testsuite to fail with alarming consequences, such as not sending DialBegin and DialEnd events, and giving incorrect hangup causes during calls. This commit reverts the previous commit and implements and alternate solution. This new solution involves only passing AST_CONTROL_RINGING frames across local channels if the local channel is in AST_STATE_RING. Otherwise, the frame does not traverse the local channels. By doing this, we can ensure that a playtones generator does not get started on the calling channel but rather is started on the local channel on which the ringing frame was initially indicated. ASTERISK-25250 #close Reported by Etienne Lessard Change-Id: I3bc87a18a38eb2b68064f732d098edceb5c19f39	10 years ago
Matt Jordan	ffb679f7b5	Merge "audiohook: Use manipulated frame instead of dropping it."	10 years ago
Mark Michelson	43173f073c	Merge "audiohook: Read the correct number of samples based on audiohook format."	10 years ago
Joshua Colp	1cc99ba8b6	audiohook: Use manipulated frame instead of dropping it. ... Previous changes to sample rate support in audiohooks accidentally removed code responsible for allowing the manipulate audiohooks to work. Without this code the manipulated frame would be dropped and not used. This change restores it. ASTERISK-25253 #close Change-Id: I3ff50664cd82faac8941f976fcdcb3918a50fe13	10 years ago
Mark Michelson	0b7148e262	Local channels: Do not block control -1 payloads. ... Control frames with a -1 payload are used as a special signal to stop playtones generators on channels. This indication is sent both by app_dial as well as by ast_answer() when a call is answered in case any tones were being generated on a calling channel. This control frame type was made to stop traversing local channel pairs as an optimization, because it was thought that it was unnecessary to send these indications, and allowing such unnecessary control frames to traverse the local channels would cause the local channels to optimize away less quickly. As it turns out, through some special magic dialplan code, it is possible to have a tones being played on a non-local channel, and it is important for the local channel to convey that the tones should be stopped. The result of having tones continue to be played on the non-local channel is that the tones play even once the channel has been bridged. By not blocking the -1 control frame type, we can ensure that this situation does not happen. ASTERISK-25250 #close Reported by Etienne Lessard Change-Id: I0bcaac3d70b619afdbd0ca8a8dd708f33fd2f815	10 years ago
Joshua Colp	e5fe8d40c8	audiohook: Read the correct number of samples based on audiohook format. ... Due to changes in audiohooks to support different sample rates the underlying storage of samples is in the format of the audiohook itself and not of the format being requested. This means that if a channel is using G722 the samples stored will be at 16kHz. If something subsequently reads from the audiohook at a format which is not the same sample rate as the audiohook the number of samples needs to be adjusted. Given the following example: 1. Channel writing into audiohook at 16kHz (as it is using G722). 2. Chanspy reading from audiohook at 8kHz. The original code would read 160 samples from the audiohook for each 20ms of audio. This is incorrect. Since the audio in the audiohook is at 16kHz the actual number needing to be read is 320. Failure to read this much would cause the audiohook to reset itself constantly as the buffer became full. This change adjusts the requested number of samples by determining the duration of audio requested and then calculating how many samples that would be in the audiohook format. ASTERISK-25247 #close Change-Id: Ia91ce516121882387a315fd8ee116b118b90653d	10 years ago
Joshua Colp	3143bb2e88	Merge "Documentation: A couple of trivial fixes in sip.conf.sample and func_cdr.c"	10 years ago
Matt Jordan	d482361a40	Merge "cdr/cdr_adaptive_odbc.c: Fix quoted identifier usage when inserting CDR records"	10 years ago
Elazar Broad	293c9f6894	cdr/cdr_adaptive_odbc.c: Fix quoted identifier usage when inserting CDR records ... Commit a24ce38 added support for the use of quoted indentifiers when inserting CDR records into the database. However, the if statement logic responsible for determining whether to use those identifiers is reversed, resulting in a reference to the quoted identifier character buffer which will be null, hence null terminating the SQL query, resulting in a truncated statement which fails to execute. ASTERISK-25263 #close Reported by: Elazar Broad Tested by: Elazar Broad Change-Id: I40da47309b67cc1572207b1515dcc08ec9b1f644	10 years ago
Joshua Colp	f7f3ae1815	Merge "res_pjsip: Add rtp_keepalive endpoint option."	10 years ago
Rusty Newton	d02196448b	Documentation: A couple of trivial fixes in sip.conf.sample and func_cdr.c ... * In sip.conf.sample fix sentence where we said that WS or WSS are supported transports for use in an outbound register definition. They are not supported in that case. * In func_cdr.c made it clear that the Disable option for CDR_PROP can be used to enable CDR on a channel. ASTERISK-24867 #close Reported by: Rusty Newton ASTERISK-24853 #close Reported by: PSDK Change-Id: I3d698bc6302b9d00a0a995b5c4ad9a42d69b48ca	10 years ago
Mark Michelson	2b42264e66	res_pjsip: Add rtp_keepalive endpoint option. ... This adds an "rtp_keepalive" option for PJSIP endpoints. Similar to the chan_sip option, this specifies an interval, in seconds, at which we will send RTP comfort noise frames. This can be useful for keeping RTP sessions alive as well as keeping NAT associations alive during lulls. ASTERISK-25242 #close Reported by Mark Michelson Change-Id: I3b9903d99e35fe5d0b53ecc46df82c750776bc8d	10 years ago
Matt Jordan	6741eedece	Merge "chan_pjsip: Don't change formats when frame of unsupported format is received."	10 years ago
Matt Jordan	0047ca8c84	Merge "res/res_musiconhold: Add a warning when MOH does not exist"	10 years ago
Joshua Colp	bd3515abaf	Merge "pbx.c: Post AMI VarSet event if delete a non-empty dialplan variable."	10 years ago
Michael Cargile	8b503f2a10	res/res_musiconhold: Add a warning when MOH does not exist ... Change-Id: Ifdfbd0b97cf31478d29923ec30aabce28d01740b	10 years ago
Matt Jordan	9475dc9492	res/res_sorcery_config: Prevent crash from misconfigured sorcery.conf ... Misconfiguring sorcery.conf with a 'config' wizard with no extra data will currently crash Asterisk on startup, as the wizard requires a comma delineated list to parse. This patch updates res_sorcery_config to check for the presence of the data before it starts manipulating it. Change-Id: I4c97512e8258bc82abe190627a9206c28f5d3847	10 years ago
Joshua Colp	649460aa44	chan_pjsip: Don't change formats when frame of unsupported format is received. ... Receipt of an RTP packet currently causes the formats on an PJSIP channel to change to the format of the RTP packet. In some off-nominal cases it's possible for this to be a format that has not been configured or negotiated. This change makes it so only formats explicitly configured on the endpoint are allowed. ASTERISK-25258 #close Change-Id: If93d641fb6418a285928839300d7854cab8c1020	10 years ago
Richard Mudgett	4a875e8082	pbx.c: Post AMI VarSet event if delete a non-empty dialplan variable. ... ASTERISK-25256 #close Reported by: Richard Mudgett Change-Id: I0b6be720b66fa956f6a798cd22ef8934eb0c0ff3	10 years ago
Patric Marschall	7908ae4934	sig_pri.h: force_restart_unavailable_chans in wrong scope ... In channels/sig_pri.h, struct sig_pri_span, the field force_restart_unavailable_chans is only defined if #if defined(HAVE_PRI_MCID) is true. All other occurences of force_restart_unavailable_chans are outside of the #if defined(HAVE_PRI_MCID) endif scope. ASTERISK-25257 #close Reported by: Patric Marschall Change-Id: I071de89cc2cd0d85927a013036e235851f672549	10 years ago
Matt Jordan	a2bfd663a1	Merge "ARI: Add support for push configuration of dynamic object"	10 years ago
Matt Jordan	f90beeaea6	Merge "strings.h: Fix issues with escape string functions."	10 years ago
Matt Jordan	daf284944b	Merge "media cache: Add CLI commands"	10 years ago
Matthew Jordan	3b39dbe38b	Merge "media cache: Add a core API and facade for a backend agnostic media cache"	10 years ago
Matt Jordan	254d07b15b	ARI: Add support for push configuration of dynamic object ... This patch adds support for push configuration of dynamic, i.e., sorcery, objects in Asterisk. It adds three new REST API calls to the 'asterisk' resource: * GET /asterisk/{configClass}/{objectType}/{id}: retrieve the current object given its ID. This returns back a list of ConfigTuples, which define the fields and their present values that make up the object. * PUT /asterisk/{configClass}/{objectType}/{id}: create or update an object. A body may be passed with the request that contains fields to populate in the object. The same format as what is retrieved using the GET operation is used for the body, save that we specify that the list of fields to update are contained in the "fields" attribute. * DELETE /asterisk/{configClass}/{objectType}/{id}: remove a dynamic object from its backing storage. Note that the success/failure of these operations is somewhat configuration dependent, i.e., you must be using a sorcery wizard that supports the operation in question. If a sorcery wizard does not support the create or delete mechanisms, then the REST API call will fail with a 403 forbidden. ASTERISK-25238 #close Change-Id: I28cd5c7bf6f67f8e9e437ff097f8fd171d30ff5c	10 years ago
Matt Jordan	af9ee2910d	Merge "parking_applications.c: Fix ast_verb() line terminator."	10 years ago
Matt Jordan	f99322ab21	Merge "res_pjsip_session.c: Extract sip_session_defer_termination_stop_timer()."	10 years ago
Matt Jordan	3613babd99	Merge "res_pjsip_session.c: Add some helpful comments and minor tweaks."	10 years ago
Matt Jordan	f25660c99d	Merge "res_pjsip_session.c: Fix off nominal crash potential in debug message."	10 years ago
Richard Mudgett	b34c4528ab	strings.h: Fix issues with escape string functions. ... Fixes for issues with the ASTERISK-24934 patch. * Fixed ast_escape_alloc() and ast_escape_c_alloc() if the s parameter is an empty string. If it were an empty string the functions returned NULL as if there were a memory allocation failure. This failure caused the AMI VarSet event to not get posted if the new value was an empty string. * Fixed dest buffer overwrite potential in ast_escape() and ast_escape_c(). If the dest buffer size is smaller than the space needed by the escaped s parameter string then the dest buffer would be written beyond the end by the nul string terminator. The num parameter was really the dest buffer size parameter so I renamed it to size. * Made nul terminate the dest buffer if the source string parameter s was an empty string in ast_escape() and ast_escape_c(). * Updated ast_escape() and ast_escape_c() doxygen function description comments to reflect reality. * Added some more unit test cases to /main/strings/escape to cover the empty source string issues. ASTERISK-25255 #close Reported by: Richard Mudgett Change-Id: Id77fc704600ebcce81615c1200296f74de254104	10 years ago
Richard Mudgett	097c15ac51	parking_applications.c: Fix ast_verb() line terminator. ... Change-Id: I8797238c71563e243c48c6145b4f1ae58f91f775	10 years ago
Richard Mudgett	8b620c555b	res_parking: Fix crash if ATTENDEDTRANSFER set empty before Park. ... setup_park_common_datastore() was assuming that a non-NULL string returned for the ATTENDEDTRANSFER and BLINDTRANSFER channel variables are not empty strings. Things got crashy as a result. * Made setup_park_common_datastore() treat the channel variable values the same whether they are NULL or empty for ATTENDEDTRANSFER and BLINDTRANSFER. ASTERISK-25254 #close Reported by: Richard Mudgett Change-Id: I9a9c174b33f354f35f82cc6b7cea8303adbaf9c2	10 years ago
Richard Mudgett	4af24ec74b	res_pjsip_session.c: Extract sip_session_defer_termination_stop_timer(). ... Change-Id: I9e115dee74bd72e06081d0ee73ecdeb886caa5fb	10 years ago
Richard Mudgett	71b3bcf5e0	res_pjsip_session.c: Add some helpful comments and minor tweaks. ... Change-Id: I742aeeaf5f760593f323a00fb691affe22e35743	10 years ago
Richard Mudgett	53c91737a5	res_pjsip_session.c: Fix off nominal crash potential in debug message. ... Change-Id: I09928297927ee85f7655289acee3a586816466bc	10 years ago
Joshua Colp	605c5d23a8	Merge "ARI: Fixed unload mode for unload module."	10 years ago
Matt Jordan	eff6a88a88	apps/app_dictate: Fix typo in attribution ... Last time I checked, it's "Sangoma", not "Samgoma". Thanks to Brian (GameGamer43) for pointing that out. Change-Id: I43d7b196f6d7a2b2517b84915e3a8dfbc2894106	10 years ago
Benjamin Ford	e01d93e092	ARI: Fixed unload mode for unload module. ... Changed the unload mode to AST_FORCE_SOFT from AST_FORCE_FIRM, which would unload a module even if it was in use. * Changed unload mode to proper mode ASTERISK-25173 Change-Id: If2402487b5bce05d9770f25f65f5c8e292ad5533	10 years ago
Matt Jordan	b188eb788d	Merge "res_pjsip_session.c: Fix crash on call disconnect."	10 years ago
Richard Mudgett	1b666549f3	res_pjsip_session.c: Fix crash on call disconnect. ... The crash fix for ASTERISK-25183 backported some code from master to try to make sure that a BYE response is processed by the same serializer used by the BYE request. The identified race condition causing that backport was the BYE request code had not finished processing after sending the BYE before the BYE response came in for processing under a different thread. Unfortunately, there is still a race condition. Now the race condition is between destroying the call session's serializer in ast_taskprocessor_unreference() and using ast_taskprocessor_get() to get a reference to the serializer for a BYE response. Even worse, the new race condition is a design limitation of the taskprocessor implementation that didn't matter in versions before v12. Back then, taskprocessors were only destroyed when a module unloaded. Now res_pjsip can destroy them when a call ends. However, as noted on the ASTERISK-25183 commit, session_inv_on_state_changed() is disassociating the dialog from the session when the invite dialog state becomes PJSIP_INV_STATE_DISCONNECTED. This is a tad too soon because our BYE request transaction has not completed yet. * Split session_end() that is called by session_inv_on_state_changed() to hold off session destruction until the BYE transaction timeout occurs or a failed initial INVITE transaction timeout occurs in session_inv_on_tsx_state_changed(). ASTERISK-25201 #close Reported by: Matt Jordan Change-Id: Iaf8dc8485fd8392a2a3ee4ad3b7f7f04a0dcc961	10 years ago
Benjamin Ford	9d458b8311	ARI: Added new functionality to reload a single module. ... An http request can be sent to reload an Asterisk module. If the module can not be reloaded or is not already loaded, an error response will be returned. The command "curl -v -u user:pass -X PUT 'http://localhost:8088 /ari/asterisk/modules/{moduleName}'" (or something similar, based on configuration) can be run in the terminal to access this new functionality. For more information, see: https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource * Added new ARI functionality * Asterisk modules can be reloaded through http requests ASTERISK-25173 Change-Id: I289188bcae182b2083bdbd9ebfffd50b62f58ae1	10 years ago
Mark Michelson	3ebe5cd766	Merge "ARI: Added new functionality to unload a single module."	10 years ago
Matt Jordan	f35998e0ef	Merge "main/bucket: Add a callback function for ast_bucket_file objects"	10 years ago
Benjamin Ford	f64f1c2772	ARI: Added new functionality to unload a single module. ... An http request can be sent to unload an Asterisk module. If the module can not be unloaded or is already unloaded, an error response will be returned. The command "curl -v -u user:pass -X DELETE 'http://localhost:8088 /ari/asterisk/modules/{moduleName}'" (or something similar, depending on configuration) can be run in the terminal to access this new functionality. For more information, see: https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource * Added new ARI functionality * Asterisk modules can be unloaded through http requests ASTERISK-25173 Change-Id: I535a95f5676deb02651522761ecbdc0b00b5ac57	10 years ago
Benjamin Ford	aa5707b889	ARI: Added new functionality to load a single module. ... An http request can be sent to load an Asterisk module. If the module can not be loaded or is loaded already, an error response will be returned. The command curl -v -u user:pass -X POST 'http://localhost:8088/ari /asterisk/modules/{moduleName}'" (or something similar, depending on configuration) can be run in the terminal to access this new functionality. For more information, see: https://wiki.asterisk.org/wiki.display/~bford/Asterisk+ARI+Resource * Added new ARI functionality * Asterisk modules can be loaded through http requests ASTERISK-25173 Change-Id: I9e05d5b8c5c666ecfef341504f9edc1aa84fda33	10 years ago