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Author	SHA1	Message	Date
Joshua Colp	f1493f900e	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	d16347f33b	Merge "Documentation: A couple of trivial fixes in sip.conf.sample and func_cdr.c" into 13	10 years ago
Rusty Newton	62c64c3bd1	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	d9094ddd73	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: I06660ba672c0a343814af4cec838e6025cafd54b	10 years ago
Matt Jordan	34207887e6	Merge "chan_pjsip: Don't change formats when frame of unsupported format is received." into 13	10 years ago
Matt Jordan	8a1d9a9f83	Merge "res/res_musiconhold: Add a warning when MOH does not exist" into 13	10 years ago
Joshua Colp	29de5b497f	Merge "pbx.c: Post AMI VarSet event if delete a non-empty dialplan variable." into 13	10 years ago
Michael Cargile	a23adcca3d	res/res_musiconhold: Add a warning when MOH does not exist ... Change-Id: Ifdfbd0b97cf31478d29923ec30aabce28d01740b	10 years ago
Matt Jordan	03064daeb2	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
Mark Michelson	1915aa5a54	Merge "sig_pri.h: force_restart_unavailable_chans in wrong scope" into 13	10 years ago
Joshua Colp	2c626ceb64	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
Patric Marschall	abb14ac5b8	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
Richard Mudgett	875aee4c09	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
Matt Jordan	9d7f689b4b	Merge "ARI: Add support for push configuration of dynamic object" into 13	10 years ago
Matt Jordan	325d83f37f	Merge "strings.h: Fix issues with escape string functions." into 13	10 years ago
Matt Jordan	8bcf6d2801	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	80eaf0b025	Merge "res_pjsip_session.c: Extract sip_session_defer_termination_stop_timer()." into 13	10 years ago
Matt Jordan	174f0e9d4d	Merge "res_pjsip_session.c: Add some helpful comments and minor tweaks." into 13	10 years ago
Matt Jordan	bba2c44ac4	Merge "res_pjsip_session.c: Fix off nominal crash potential in debug message." into 13	10 years ago
Richard Mudgett	e31cb6b248	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	243c0d1609	parking_applications.c: Fix ast_verb() line terminator. ... Change-Id: I8797238c71563e243c48c6145b4f1ae58f91f775	10 years ago
Richard Mudgett	c782320c68	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	2735dd5b2d	res_pjsip_session.c: Extract sip_session_defer_termination_stop_timer(). ... Change-Id: I9e115dee74bd72e06081d0ee73ecdeb886caa5fb	10 years ago
Richard Mudgett	3d0ca343ca	res_pjsip_session.c: Add some helpful comments and minor tweaks. ... Change-Id: I742aeeaf5f760593f323a00fb691affe22e35743	10 years ago
Richard Mudgett	8d08bb179c	res_pjsip_session.c: Fix off nominal crash potential in debug message. ... Change-Id: I09928297927ee85f7655289acee3a586816466bc	10 years ago
Joshua Colp	5086bdacfb	Merge "apps/app_dictate: Fix typo in attribution" into 13	10 years ago
Mark Michelson	ca41785774	Merge "ARI: Fixed unload mode for unload module." into 13	10 years ago
Matt Jordan	0a1a550593	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	3384e64ef6	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	00d858da87	Merge "res_pjsip_session.c: Fix crash on call disconnect." into 13	10 years ago
Matt Jordan	0b6ff77afb	res/res_sorcery_astdb: Add a debugging message for when retrieval by ID fails ... Having a debug message tell us that we attempted to look up an item but failed is nice in circumstances when it isn't clear if the wizard was queried correctly or not. Change-Id: I2600c3bbea87f252196358f62e73f4c7da8632f7	10 years ago
Matt Jordan	2f0d6d346c	res/res_pjsip_outbound_registration: Fix WARNING message ... Newlines are nice. Change-Id: Icf0d915db02882e47cd9077ed9009f5d44140d42	10 years ago
Matt Jordan	cd2213f1ae	res_pjsip/configuration: Fix a variety of default value problems ... This patch fixes some bad default value handling in the following settings: * The 'message_context' and 'accountcode' settings are not mandatory. As such, we can allow their stringfield values to be empty. * The 'media_encryption' setting applies a default value of 'none' to the setting, which it then can't parse or understand. Since the value is documented to be 'no', this will now apply that as the default value. Change-Id: Ib9be7f97a7a5b9bc7aee868edf5acf38774cff83	10 years ago
Matt Jordan	2e4bdbd78a	main/sorcery: Provide log messages when a wizard does not support an operation ... If a sorcery wizard does not support one of the 'optional' CRUD operations (namely the CUD), log a WARNING message so we are aware of why the operation failed. This also removes an assert in this case, as the CUD operation may have been triggered by an external system, in which case it is not a programming error but a configuration error. Change-Id: Ifecd9df946d9deaa86235257b49c6e5e24423b53	10 years ago
Richard Mudgett	653f2087e0	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	1aafadf814	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
Benjamin Ford	9dcae23cfc	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	c219a98d2b	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
Mark Michelson	585d98fbb6	Merge "ARI: Added new functionality to get information on a single module." into 13	10 years ago
Mark Michelson	ca65ddcd19	Merge "bridge.c: Fixed race condition during attended transfer" into 13	10 years ago
Benjamin Ford	73e35d20de	ARI: Added new functionality to get information on a single module. ... An http request can be sent to retrieve information on a single module, including the resource name, description, use count, status, and support level. The command "curl -v -u user:pass -X GET '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 * Information on a single module can now be retrieved ASTERISK-25173 Change-Id: Ibce5a94e70ecdf4e90329cf0ba66c33a62d37463	10 years ago
Kevin Harwell	97ee0ee6c6	bridge.c: Fixed race condition during attended transfer ... During an attended transfer a thread is started that handles imparting the bridge channel. From the start of the thread to when the bridge channel is ready exists a gap that can potentially cause problems (for instance, the channel being swapped is hung up before the replacement channel enters the bridge thus stopping the transfer). This patch adds a condition that waits for the impart thread to get to a point of acceptable readiness before allowing the initiating thread to continue. ASTERISK-24782 Reported by: John Bigelow Change-Id: I08fe33a2560da924e676df55b181e46fca604577	10 years ago
Matt Jordan	6176ef782c	Merge "main/format_cap: Parse capabilities generated by ast_format_cap_get_names" into 13	10 years ago
Matt Jordan	bb76b88baf	main/sorcery: Don't fail object set creation from JSON if field fails ... Some individual fields may fail their conversion due to their default values being invalid for their custom handlers. In particular, configuration values that depend on others being enabled (and thus have an empty default value) are notorious for tripping this routine up. An example of this are any of the DTLS options for endpoints. Any of the DTLS options will fail to be applied (as DTLS is not enabled), causing the entire object set to be aborted. This patch makes it so that we log a debug message when skipping a field, and rumble on anyway. ASTERISK-25238 Change-Id: I0bea13de79f66bf9f9ae6ece0e94a2dc1c026a76	10 years ago
Matt Jordan	5f13c2226a	main/format_cap: Parse capabilities generated by ast_format_cap_get_names ... We have a strange relationship between the parsing of format capabilities from a string and their representation as a string. We expect the format capabilities to be expressed as a string in the following format: allow = !all,ulaw,alaw disallow = g722 While we would generate the string representation of those formats as: allow = (ulaw|alaw) disallow = (ulaw|alaw|g729...) When the configuration framework needs to store values as a string, it generates the format capabilities using the second representation; this representation however cannot be parsed when the entry is rehydrated. This patch fixes that by updating ast_format_cap_update_by_allow_disallow to parse an entry as if it were in the generated format if it has a leading '(' and a trailing ')'. ASTERISK-25238 Change-Id: I904d43caf4cf45af06f6aee0c9e58556eb91d6ca	10 years ago
Matt Jordan	5c491a6295	Merge "tests/test_devicestate: Add additional tests for the device state API" into 13	10 years ago
Matt Jordan	4a5f23e716	Merge "res/res_sorcery_memory_cache: Fix test registration issues" into 13	10 years ago
Matt Jordan	3645ed0971	Merge "tests/test_sorcery_memory_cache_thrash: Fix test loading problems" into 13	10 years ago
Matt Jordan	2325b106fd	tests/test_devicestate: Add additional tests for the device state API ... This patch adds more tests that exercise the device state API. This includes: * Tests that cover adding a device state provider, as well as deleting a device state provider. This also verifies that you cannot add an already added device state provider, and cannot delete an already deleted device state provider. * A test that covers changing device state and receiving said updates from a device state subscriber. This also covers hitting both the device state cache as well as a custom device state provider. * A test that covers converting device state to channel state and device state values to a string representation and back. * A test that covers obtaining device state from an active channel and a channel driver that provides its own device state. Change-Id: I2adca67ffb405cd8625a5d6df1e3f9b3d945c08d	10 years ago
Matt Jordan	328f0be806	main/devicestate: Prevent duplicate registration of device state providers ... Currently, the device state provider API will allow you to register a device state provider with the same case insensitive name more than once. This could cause strange issues, as the duplicate device state providers will not be queried when a device's state has to be polled. This patch updates the API such that a device state provider with the same name as one that has already registered will be rejected. Change-Id: I4a418a12280b7b6e4960bd44f302e27cd036ceb2	10 years ago