8669 Commits (16)

Author	SHA1	Message	Date
George Joseph	0b0cb8733b	Fix incorrect application and function documentation references ... There were a few references in the embedded documentation XML where the case didn't match or where the referenced app or function simply didn't exist any more. These were causing 404 responses in docs.asterisk.org.	1 year ago
Naveen Albert	83afb94f79	chan_dahdi: Resolve format truncation warning. ... Fixes a format truncation warning in notify_message. ASTERISK-30256 #close Change-Id: I983a423c0214641ca4f8c9dfe0b19c47448fdee1	3 years ago
Ben Ford	369f45c432	res_pjsip: Add TEL URI support for basic calls. ... This change allows TEL URI requests to come through for basic calls. The allowed requests are INVITE, ACK, BYE, and CANCEL. The From and To headers will now allow TEL URIs, as well as the request URI. Support is only for TEL URIs present in traffic from a remote party. Asterisk does not generate any TEL URIs on its own. ASTERISK-26894 Change-Id: If5729e6cd583be7acf666373bf9f1b9d653ec29a	3 years ago
Philip Prindeville	b1cf0e0069	res_crypto: Use EVP API's instead of legacy API's ... ASTERISK-30046 #close Change-Id: I5c738756de75fd27ebad54be144c0ac6193f21b2	3 years ago
Sean Bright	1e2dfd0c32	chan_dahdi.c: Resolve a format-truncation build warning. ... With gcc (Ubuntu 11.2.0-19ubuntu1) 11.2.0: > chan_dahdi.c:4129:18: error: ‘%s’ directive output may be truncated > writing up to 255 bytes into a region of size between 242 and 252 > [-Werror=format-truncation=] This removes the error-prone sizeof(...) calculations in favor of just doubling the size of the base buffer. Change-Id: I2d276785286730d3d5d0a921bcea2e065dbf27c5	3 years ago
Naveen Albert	859b84476e	chan_iax2: Add missing options documentation. ... Adds missing dial resource option documentation. ASTERISK-30164 #close Change-Id: I674e1fc9b1e5d67a20599bd4b418ce294d48fc83	3 years ago
Naveen Albert	7703330142	general: Improve logging levels of some log messages. ... Adjusts some logging levels to be more or less important, that is more prominent when actual problems occur and less prominent for less noteworthy things. ASTERISK-30153 #close Change-Id: Ifc8f7df427aa018627db462125ae744986d3261b	3 years ago
Naveen Albert	5177bb9f34	chan_dahdi: Fix buggy and missing Caller ID parameters ... There are several things wrong with analog Caller ID handling that are fixed by this commit: callerid.c's Caller ID generation function contains the logic to use the presentation to properly send the proper Caller ID. However, currently, DAHDI does not pass any presentation information to the Caller ID module, which means that presentation is completely ignored on all calls. This means that lines could be getting Caller ID information they aren't supposed to. Part of the reason this has been obscured is because the simple switch logic for handling the built in *67 and *82 is completely wrong. Rather than modifying the presentation for the call accordingly (which is what it's supposed to do), it simply blanks out the Caller ID or fills it in. This is wrong, so wrong that it makes a mockery of the specification. Additionally, it would leave to the "UNAVAILABLE" disposition being used for Caller ID generation as opposed to the "PRIVATE" disposition that it should have been using. This is now fixed to only update the presentation and not modify the number and name, so that the simple switch *67/*82 work correctly. Next, sig_analog currently only copies over the name and number, nothing else, when it is filling in a duplicated caller id structure. Thus, we also now copy over the presentation information so that is available for the Caller ID spill. Additionally, this meant that "valid" was implicitly 0, and as such presentation would always fail to "Unavailable". The validity is therefore also copied over so it can be used by ast_party_id_presentation. As part of this fix, new API is added so that all the relevant Caller ID information can be passed in to the Caller ID generation functions. Parameters that are also completely missing from the Caller ID spill have also been added, to enhance the compatibility, correctness, and completeness of the Asterisk Caller ID implementation. ASTERISK-29991 #close Change-Id: Icc44a5e09979916f4c18a440f96e10dc1c76ae15	3 years ago
Naveen Albert	15dc2f8c16	chan_dahdi: Add POLARITY function. ... Adds a POLARITY function which can be used to retrieve the current polarity of an FXS channel as well as set the polarity of an FXS channel to idle or reverse at any point during a call. ASTERISK-30000 #close Change-Id: If6f50998f723e4484bf68e2473f5cedfeaf9b8f1	3 years ago
Naveen Albert	a8395c9420	general: Fix various typos. ... ASTERISK-30089 #close Change-Id: I1f5db911fd05a3a211c522c13e990fa1d0e62275	3 years ago
Naveen Albert	f812dfb68c	chan_iax2: Allow compiling without OpenSSL. ... ASTERISK_30007 accidentally made OpenSSL a required depdendency. This adds an ifdef so the relevant code is compiled only if OpenSSL is available, since it only needs to be executed if OpenSSL is available anyways. ASTERISK-30083 #close Change-Id: Iad05c1a9a8bd2a48e7edf8d234eaa9f80779e34d	3 years ago
Naveen Albert	a57e36edfc	sig_analog: Fix broken three-way conferencing. ... Three-way calling for analog lines is currently broken. If party A is on a call with party B and initiates a three-way call to party C, the behavior differs depending on whether the call is conferenced prior to party C answering. The post-answer case is correct. However, if A flashes before C answers, then the next flash disconnects B rather than C, which is incorrect. This error occurs because the subs are not swapped in the misbehaving case. This is because the flash handler only swaps the subs if C has answered already, which is wrong. To fix this, we swap the subs regardless of whether C has answered or not when the call is conferenced. This ensures that C is disconnected on the next hook flash, rather than B as can happen currently. ASTERISK-30043 #close Change-Id: I96c5bf6c9b7eb2636136b716c677c82c079b6f06	3 years ago
Naveen Albert	82cebaa023	chan_iax2: Prevent deadlock due to duplicate autoservice. ... If a switch is invoked using chan_iax2, deadlock can result because the PBX core is autoservicing the channel while chan_iax2 also then attempts to service it while waiting for the result of the switch. This removes servicing of the channel to prevent any conflicts. ASTERISK-30064 #close Change-Id: Ie92f206d32f9a36924af734ddde652b21106af22	3 years ago
Maximilian Fridrich	9ae06885fc	chan_pjsip: Only set default audio stream on hold. ... When a PJSIP channel is set on hold or off hold, all streams were set on/off hold. This is not the desired behaviour and caused issues when there were multiple streams in the topology. Now, only the default audio stream is set on/off hold when a hold is indicated. ASTERISK-30051 Change-Id: I04f1110565fd05fea565f5539b534b54549d4f71	3 years ago
Moritz Fain	4f2bd069a4	ari: expose channel driver's unique id to ARI channel resource ... This change exposes the channel driver's unique id (i.e. the Call-ID for chan_sip/chan_pjsip based channels) to ARI channel resources as `protocol_id`. ASTERISK-30027 Reported by: Moritz Fain Tested by: Moritz Fain Change-Id: I7cc6e7a9d29efe74bc27811d788dac20fe559b87	3 years ago
Naveen Albert	6722136531	chan_dahdi: Fix broken operator mode clearing. ... Currently, the operator services mode in DAHDI is broken and unusable. The actual operator recall functionality works properly; however, when the operator hangs up (which is the only way that such a call is allowed to end), both lines are permanently taken out of service until "dahdi restart" is run. This prevents this feature from being used. Operator mode is one of the few factors that can cause the general analog event handling in sig_analog not to be used. Several years back, much of the analog handling was moved from chan_dahdi to sig_analog. However, this was not done fully or consistently at the time, and when operator mode is active, sig_analog does not get used. Generally this is correct, but in the case of hangup it should be using sig_analog regardless of the operator mode; otherwise, the lines do not properly clear and they become unusable. This bug is fixed so the operator can now hang up and properly release the call. It is treated just like any other hangup. The operator mode functionality continues to work as it did before. ASTERISK-29993 #close Change-Id: Ib2e3ddb40d9c71e8801e0b4bb0a12e2b52f51d24	3 years ago
George Joseph	01dc630b8c	GCC12: Fixes for 16+ ... Most issues were in stringfields and had to do with comparing a pointer to an constant/interned string with NULL. Since the string was a constant, a pointer to it could never be NULL so the comparison was always "true". gcc now complains about that. There were also a few issues where determining if there was enough space for a memcpy or s(n)printf which were fixed by defining some of the involved variables as "volatile". There were also a few other miscellaneous fixes. ASTERISK-30044 Change-Id: Ia081ca1bcfb329df6487c4660aaf1944309eb570	3 years ago
Naveen Albert	e5dc9e0979	chan_dahdi: Document dial resource options. ... Documents the Dial syntax for DAHDI, namely the channel group, distinctive ring, answer confirmation, and digital call options that are specified in the resource itself. ASTERISK-24827 #close Change-Id: Ib95e78497fb00dc5cbfde1c93a69f034bfd08c30	3 years ago
Naveen Albert	3dc982bdcc	chan_dahdi: Don't allow MWI FSK if channel not idle. ... For lines that have mailboxes configured on them, with FSK MWI, DAHDI will periodically try to dispatch FSK to update MWI. However, this is never supposed to be done when a channel is not idle. There is currently an edge case where MWI FSK can extraneously get spooled for the channel if a caller hook flashes and hangs up, which triggers a recall ring. After one ring, the on hook time threshold in this if condition has been satisfied and an MWI update is spooled. This means that when the phone is picked up again, the answerer gets an FSK spill before being reconnected to the party on hold. To prevent this, we now explicitly check to ensure that subchannel 0 has no owner. There is no owner when DAHDI channels are idle, but if the channel is "in use" in some way (such as in the aforementioned scenario), then there is an owner, and we shouldn't process MWI at this time. ASTERISK-28518 #close Change-Id: Ia3904434fd81688d71742f7e84358b7e1c38e92a	3 years ago
Naveen Albert	4350eee04b	chan_dahdi: Don't append cadences on dahdi restart. ... Currently, if any custom ring cadences are specified, they are appended to the array of cadences from wherever we left off last time. This works properly the first time, but on subsequent dahdi restarts, it means that the existing cadences are left alone and (most likely) the same cadences are then re-added afterwards. In short order, the cadence array gets maxed out and the user begins seeing warnings that the array is full and no more cadences may be added. This buggy behavior persists until Asterisk is completely restarted; however, if and when dahdi restart is run again, then the same problem is reintroduced. This fixes this behavior so that cadence parsing is more idempotent, that is so running dahdi restart multiple times starts adding cadences from the beginning, rather than from wherever the last cadence was added. As before, it is still not possible to revert to the default cadences by simply removing all cadences in this manner, nor is it possible to delete existing cadences. However, this does make it possible to update existing cadences, which was not possible before, and also ensures that the cadences remain unchanged if the config remains unchanged. ASTERISK-29990 #close Change-Id: Ie32ea3e8a243b766756b1afce684d4a31ee7421d	3 years ago
Naveen Albert	59a8cdaca2	chan_iax2: Prevent crash if dialing RSA-only call without outkey. ... Currently, if attempting to place a call to a peer that only allows RSA authentication, if we fail to provide an outkey when placing the call, Asterisk will crash. This exposes the broader issue that IAX2 is prone to causing a crash if encryption or decryption is attempted but we never initialized the encryption and decryption keys. In other words, if the logic to use encryption in chan_iax2 is not perfectly aligned with the decision to build keys in the first place, then a crash is not only possible but probable. This was demonstrated by ASTERISK_29264, for instance. This permanently prevents such events from causing a crash by explicitly checking that keys are initialized properly before setting the flags to use encryption for the call. Instead of crashing, the call will now abort. ASTERISK-30007 #close Change-Id: If925c3d86099ceac7f621804f2532baac5050c9a	3 years ago
Naveen Albert	d1c23c2cec	chan_dahdi: Fix insufficient array size for round robin. ... According to chan_dahdi.conf, up to 64 groups (numbered 0 through 63) can be used when dialing DAHDI channels. However, currently dialing round robin with a group number greater than 31 fails because the array for the round robin structure is only size 32, instead of 64 as it should be. This fixes that so the round robin array size is consistent with the actual groups capacity. ASTERISK-29994 Change-Id: I4caa08d7025f78ac75a0539f71aaf3eb3e85b3b7	3 years ago
Mark Petersen	a107ee8616	chan_sip.c Session timers get removed on UPDATE ... If Asterisk receives a SIP REFER with Session-Timers UAC maintain Session-Timers when sending UPDATE" ASTERISK-29843 Change-Id: I8e9a21c13bf757fa34d778f49ba3cf859b29ae5c	3 years ago
Naveen Albert	f97386bd54	chan_pjsip: Add ability to send flash events. ... PJSIP currently is capable of receiving flash events and converting them to FLASH control frames, but it currently lacks support for doing the reverse: taking a FLASH control frame and converting it into a flash event in the SIP domain. This adds the ability for PJSIP to process flash control frames by converting them into the appropriate SIP INFO message, which can then be sent to the peer. This allows, for example, flash events to be sent between Asterisk systems using PJSIP. ASTERISK-29941 #close Change-Id: I1590221a4d238597f79672fa5825dd4a920c94dd	3 years ago
Naveen Albert	7e18d42479	documentation: Adds versioning information. ... Adds version information for applications, functions, and manager events/actions. This is not completely exhaustive by any means but covers most new things added that have release versioning information in the issue tracker. ASTERISK-29940 #close Change-Id: I506401e93c799715dbbe97c0a8ba18af2bf5e131	3 years ago
Mark Petersen	edaf12872b	chan_sip: SIP route header is missing on UPDATE ... if Asterisk need to send an UPDATE before answer on a channel that uses Record-Route: it will not include a Route header ASTERISK-29955 Change-Id: Id1920ecbfea7739a038b14dc94487ecfe7b57eef	3 years ago
Mark Petersen	bd5cc4c81e	chan_pjsip: add allow_sending_180_after_183 option ... added new global config option "allow_sending_180_after_183" that if enabled will preserve 180 after a 183 ASTERISK-29842 Change-Id: I8a53f8c35595b6d16d8e86e241b5f110d92f3d18	3 years ago
Naveen Albert	a6f9499c71	chan_iax2: Fix spacing in netstats command ... The iax2 show netstats command previously didn't contain enough spacing in the header to properly align the table header with the table body. This caused column headers to not align with the values on longer channel names. Some spacing is added to account for the longest channel names that display (before truncation occurs) so that columns are always properly aligned. ASTERISK-29895 #close patches: 61205_misaligned2.patch submitted by Birger Harzenetter (license 5870) Change-Id: I450ce6bb81157b9d6d149007e53b749f237b6d9f	3 years ago
Naveen Albert	669f9da5ce	chan_iax2: Fix perceived showing host address. ... ASTERISK_22025 introduced a regression that shows the host IP and port as the perceived IP and port again, as opposed to showing the actual perceived address. This fixes this by showing the correct information. ASTERISK-29048 #close Change-Id: I0ad3e25bc6b449e83ce72ea5d1a1cdba72aa304a	3 years ago
Boris P. Korzun	5565d86fa6	res_pjsip_sdp_rtp: Improve detecting of lack of RTP activity ... Change RTP timer behavior for detecting RTP only after two-way SDP channel establishment. Ignore detecting after receiving 183 with SDP or while direct media is used. Make rtp_timeout and rtp_timeout_hold options consistent to rtptimeout and rtpholdtimeout options in chan_sip. ASTERISK-26689 #close ASTERISK-29929 #close Change-Id: I07326d5b9c40f25db717fd6075f6f3a8d77279eb	3 years ago
Mark Petersen	e03b313fe9	chan_sip.c Fix pickup on channel that are in AST_STATE_DOWN ... resolve issue with pickup on device that uses "183" and not "180" ASTERISK-29832 Change-Id: I4c7d223870f8ce9a7354e0f73d4e4cb2e8b58841	3 years ago
Sean Bright	6060b9ccb0	utils.c: Remove all usages of ast_gethostbyname() ... gethostbyname() and gethostbyname_r() are deprecated in favor of getaddrinfo() which we use in the ast_sockaddr family of functions. ASTERISK-29819 #close Change-Id: Ie277c0ef768d753b169c121ef570a71665692ab7	3 years ago
Naveen Albert	1ac89e0875	chan_sip: Fix crash when accessing RURI before initiating outgoing call ... Attempting to access ${CHANNEL(ruri)} in a pre-dial handler before initiating an outgoing call will cause Asterisk to crash. This is because a null field is accessed, resulting in an offset from null and subsequent memory access violation. Since RURI is not guaranteed to exist, we now check if the base pointer is non-null before calculating an offset. ASTERISK-29772 Change-Id: Icd3b02f07256bbe6615854af5717074087b95a83	3 years ago
Alexander Traud	3489f926e7	progdocs: Fix for Doxygen, the hidden parts. ... ASTERISK-29779 Change-Id: If338163488498f65fa7248b60e80299c0a928e4b	3 years ago
Alexander Traud	0398daf4cb	progdocs: Fix grouping for latest Doxygen. ... Since Doxygen 1.8.16, a special comment block is required. Otherwise (pure C comment), the group command is ignored. Additionally, several unbalanced group commands were fixed. ASTERISK-29732 Change-Id: I780ed8e0e8acdbdd8f8b75fb7baf5ef657af7a3f	3 years ago
Naveen Albert	6296f645a2	documentation: Standardize examples ... Most examples in the XML documentation use the example tag to demonstrate examples, which gets parsed specially in the Wiki to make it easier to follow for users. This fixes a few modules to use the example tag instead of vanilla para tags to bring them in line with the standard syntax. ASTERISK-29777 #close Change-Id: I9acb6cc5faf1d220e73c6dd28592371d768d279b	3 years ago
Alexander Traud	5f9c9ba8c3	chan_misdn: Fix for Doxygen. ... ASTERISK-29764 Change-Id: I6e5466cce03e25695c5c7d8b68c305184dcf5375	4 years ago
Alexander Traud	10a0e3d4d6	channels: Fix for Doxygen. ... ASTERISK-29762 Change-Id: Ia8811ac12b93ff8c18164699c6fbc604cb0a23f7	4 years ago
Alexander Traud	c29bf45863	chan_iax2: Fix for Doxygen. ... ASTERISK-29737 Change-Id: I282003cc553989fd5c19ceeac9e478fa4ee06cec	4 years ago
Josh Soref	636d3dc4be	channels: Spelling fixes ... Correct typos of the following word families: appease permanently overriding residue silliness extension channels globally reference japanese group coordinate registry information inconvenience attempts cadence payloads presence provisioning mimics behavior width natively syslabel not owning unquelch mostly constants interesting active unequipped brodmann commanding backlogged without bitstream firmware maintain exclusive practically structs appearance range retransmission indication provisional associating always whether cyrillic distinctive components reinitialized initialized capability switches occurring happened outbound ASTERISK-29714 Change-Id: Ife52ee89cd2170b684fa651ca72b1cb911a57339	4 years ago
Naveen Albert	6d242723bb	sig_analog: Fix truncated buffer copy ... Fixes compiler warning caused by a truncated copy of the ANI2 into a buffer of size 10. This could prevent the null terminator from being copied if the copy value exceeds the size of the buffer. This increases the buffer size to 101 to ensure there is no way for truncation to occur. ASTERISK-29702 #close Change-Id: Ief9052212952840fa44de6463b8699fdb3e163d0	4 years ago
Naveen Albert	530e89655e	chan_iax2: Allow both secret and outkey at dial time ... Historically, the dial syntax for IAX2 has held that an outkey (used only for RSA authenticated calls) and a secret (used only for plain text and MD5 authenticated calls, historically) were mutually exclusive, and thus the same position in the dial string was used for both values. Now that encryption is possible with RSA authentication, this poses a limitation, since encryption requires a secret and RSA authentication requires an outkey. Thus, the dial syntax is extended so that both a secret and an outkey can be specified. The new extended syntax is backwards compatible with the old syntax. However, a secret can now be specified after the outkey, or the outkey can be specified after the secret. This makes it possible to spawn an encrypted RSA authenticated call without a corresponding peer being predefined in iax.conf. ASTERISK-29707 #close Change-Id: I1f8149313ed760169d604afbb07720a8b07dd00e	4 years ago
Mike Bradeen	8031b6c9e6	various: Fix GCC 11 compilation issues. ... test_voicemail_api: Use empty char* for empty_msg_ids. chan_skinny: Fix size of calledParty to be maximum extension. menuselect: Change Makefile to stop deprecated warnings. Added comments test_linkedlist: 'bogus' variable was manually allocated from a macro and the test fails if this happens but the compiler couldn't 'see' this and returns a warning. memset to all 0's after allocation. chan_ooh323: Fixed various indentation issues that triggered misleading indentation warnings. ASTERISK-29682 Reported by: George Joseph Change-Id: If4fe42222c8444dc16828a42731ee53b4ce5cbbe	4 years ago
Naveen Albert	8ad4cb901b	chan_iax2: Add encryption for RSA authentication ... Adds support for encryption to RSA-authenticated calls. Also prevents crashes if an RSA IAX2 call is initiated to a switch requiring encryption but no secret is provided. ASTERISK-20219 Change-Id: I18f1f9d7c59b4f9cffa00f3b94a4c875846efd40	4 years ago
Naveen Albert	7feee9c65b	res_pjsip_caller_id: Add ANI2/OLI parsing ... Adds parsing of ANI II digits (Originating Line Information) to PJSIP, on par with what currently exists in chan_sip. ASTERISK-29472 Change-Id: Ifc938a7a7d45ce33999ebf3656a542226f6d3847	4 years ago
Naveen Albert	38cc3160d3	chan_iax2: Add ANI2/OLI information element ... Adds an information element for ANI2 so that Originating Line Information can be transmitted over IAX2 channels. ASTERISK-29605 #close Change-Id: Iaeacdf6ccde18eaff7f776a0f49fee87dcb549d2	4 years ago
Sarah Autumn	b731b810b9	sig_analog: Changes to improve electromechanical signalling compatibility ... This changeset is intended to address compatibility issues encountered when interfacing Asterisk to electromechanical telephone switches that implement ANI-B, ANI-C, or ANI-D. In particular the behaviours that this impacts include: - FGC-CAMA did not work at all when using MF signaling. Modified the switch case block to send calls to the correct part of the signaling-handling state machine. - For FGC-CAMA operation, the delay between called number ST and second wink for ANI spill has been made configurable; previously all calls were made to wait for one full second. - After the ANI spill, previous behavior was to require a 'ST' tone to advance the call. This has been changed to allow 'STP' 'ST2P' or 'ST3P' as well, for compatibility with ANI-D. - Store ANI2 (ANI INFO) digits in the CALLERID(ANI2) channel variable. - For calls with an ANI failure, No. 1 Crossbar switches will send forward a single-digit failure code, with no calling number digits and no ST pulse to terminate the spill. I've made the ANI timeout configurable so to reduce dead air time on calls with ANI fail. - ANI info digits configurable. Modern digital switches will send 2 digits, but ANI-B sends only a single info digit. This caused the ANI reported by Asterisk to be misaligned. - Changed a confusing log message to be more informative. ASTERISK-29518 Change-Id: Ib7e27d987aee4ed9bc3663c57ef413e21b404256	4 years ago
Sean Bright	b02538de2e	mgcp: Remove dead debug code ... ASTERISK-20339 #close Change-Id: I36f364aaa1971241d8f3ea1a5909b463d185a2d5	4 years ago
Joshua C. Colp	68adc2304f	policy: Add deprecation and removal versions to modules. ... app_meetme is deprecated in 19, to be removed in 21. app_osplookup is deprecated in 19, to be removed in 21. chan_alsa is deprecated in 19, to be removed in 21. chan_mgcp is deprecated in 19, to be removed in 21. chan_skinny is deprecated in 19, to be removed in 21. res_pktccops is deprecated in 19, to be removed in 21. cdr_mysql was deprecated in 1.8, to be removed in 19. app_mysql was deprecated in 1.8, to be removed in 19. app_ices was deprecated in 16, to be removed in 19. app_macro was deprecated in 16, to be removed in 21. app_fax was deprecated in 16, to be removed in 19. app_url was deprecated in 16, to be removed in 19. app_image was deprecated in 16, to be removed in 19. app_nbscat was deprecated in 16, to be removed in 19. app_dahdiras was deprecated in 16, to be removed in 19. cdr_syslog was deprecated in 16, to be removed in 19. chan_oss was deprecated in 16, to be removed in 19. chan_phone was deprecated in 16, to be removed in 19. chan_sip was deprecated in 17, to be removed in 21. chan_nbs was deprecated in 16, to be removed in 19. chan_misdn was deprecated in 16, to be removed in 19. chan_vpb was deprecated in 16, to be removed in 19. res_config_sqlite was deprecated in 16, to be removed in 19. res_monitor was deprecated in 16, to be removed in 21. conf2ael was deprecated in 16, to be removed in 19. muted was deprecated in 16, to be removed in 19. ASTERISK-29548 ASTERISK-29549 ASTERISK-29550 ASTERISK-29551 ASTERISK-29552 ASTERISK-29553 ASTERISK-29554 ASTERISK-29555 ASTERISK-29557 ASTERISK-29558 ASTERISK-29559 ASTERISK-29560 ASTERISK-29561 ASTERISK-29562 ASTERISK-29563 ASTERISK-29564 ASTERISK-29565 ASTERISK-29566 ASTERISK-29567 ASTERISK-29568 ASTERISK-29569 ASTERISK-29570 ASTERISK-29571 ASTERISK-29572 ASTERISK-29573 ASTERISK-29574 Change-Id: Ic3bee31a10d42c4b3bbc913d893f7b2a28a27131	4 years ago
Kevin Harwell	dbfde090ea	AST-2021-008 - chan_iax2: remote crash on unsupported media format ... If chan_iax2 received a packet with an unsupported media format, for example vp9, then it would set the frame's format to NULL. This could then result in a crash later when an attempt was made to access the format. This patch makes it so chan_iax2 now ignores/drops frames received with unsupported media format types. ASTERISK-29392 #close Change-Id: Ifa869a90dafe33eed8fd9463574fe6f1c0ad3eb1	4 years ago