asterisk

Commit Graph

Author	SHA1	Message	Date
Kevin Harwell	e8addc07f5	AST-2022-005: pjproject - undefined behavior after freeing a dialog set ASTERISK-29945 #close Change-Id: Ia8ce6d82b115c82c1138747c72a0adcaa42b718c	5 years ago
Jaco Kroon	fc0294c702	main/backtrace: binutils-2.34 fix. My tester missed this one previously, have confirmed a positive build this time round. Change-Id: Id06853375954a200f03f9a1b9c97fe0b10d31fbf	6 years ago
Jaco Kroon	62f0d1a79e	Update main/backtrace.c to deal with changes in binutils 2.34. binutils 2.34 merged this commit: https://sourceware.org/git/gitweb.cgi?p=binutils-gdb.git;a=commitdiff;\ h=fd3619828e94a24a92cddec42cbc0ab33352eeb4 Which effectively does things like: -#define bfd_section_size(bfd, ptr) ((ptr)->size) -#define bfd_get_section_size(ptr) ((ptr)->size) +#define bfd_section_size(sec) ((sec)->size) So in order to remain backwards compatible we need to detect this API change, and adjust accordingly. The simplest is to notice that the bfd_get_section_size and bfd_get_section_vma MACROs are no longer defined, and define then onto the new API. The alternative is to litter the code with a number of #ifdef #else #endif splatters right through the code. Change-Id: I3efe0f8e8f3e338d16fcbc2b26a505367b6e172f (cherry picked from commit `33b2c7f79b`)	6 years ago
Kevin Harwell	25990bdd28	channel: write to a stream on multi-frame writes If a frame handling routine returns a list of frames (vs. a single frame) those frames are never passed to a tech's write_stream handler even if one is available. For instance, if a codec translation occurred and that codec returned multiple frames then those particular frames were always only sent to the tech's "write" handler. If that tech (pjsip for example) was stream capable then those frames were essentially ignored. Thus resulting in bad audio. This patch makes it so the "write_stream" handler is appropriately called for all cases, and for all frames if available. ASTERISK-28795 #close Change-Id: I868faea0b73a07ed5a32c2b05bb9cf4b586f739d	6 years ago
Kevin Harwell	e48f63cf24	res_rtp_asterisk: bad audio (static) due to incomplete dtls/srtp setup There was a race condition between client initiated DTLS setup, and handling of server side ice completion that caused the underlying SSL object to get cleared during DTLS initialization. If this happened Asterisk would be left in a partial DTLS setup state. RTP packets were sent and received, but were not being encrypted and decrypted. This resulted in no audio, or static. Specifically, this occurred when '__rtp_recvfrom' was processing the handshake sequence from the client to the server, and then 'ast_rtp_on_ice_complete' gets called from another thread and clears the SSL object when calling the 'dtls_perform_setup' function. The timing had to be just right in the sense that from the external SSL library perspective SSL initialization completed (rtp recv), Asterisk clears/resets the SSL object (ice done), and then checks to see if SSL is intialized (rtp recv). Since it was cleared, Asterisk thinks it is not finished, thus not completing 'dtls_srtp_setup'. This patch removes calls to 'dtls_perform_setup', which clears the SSL object, in 'ast_rtp_on_ice_complete'. When ice completes, there is no reason to clear the underlying SSL object. If an ice candidate changes a full protocol level renegotiation occurs. Also, in the case of bundled ICE candidates are reused when a stream is added. So no real reason to have to clear, and reset in this instance. Also, this patch adds a bit of extra logging to aid in diagnosis of any future problems. ASTERISK-28742 #close Change-Id: I34c9e6bad5a39b087164646e2836e3e48fe6892f	7 years ago
Asterisk Development Team	3e3c298318	Update for certified/16.3-cert1	7 years ago
Joshua C. Colp	9880a899c4	Merge "confbridge: Add support for specifying maximum sample rate." into certified/16.3	7 years ago
George Joseph	3f770e50d6	res_rtp_asterisk: Add frame list cleanups to ast_rtp_read In Asterisk 16+, there are a few places in ast_rtp_read where we've allocated a frame list but return a null frame instead of the list. In these cases, any frames left in the list won't be freed. In the vast majority of the cases, the list is empty when we return so there's nothing to free but there have been leaks reported in the wild that can be traced back to frames left in the list before returning. The escape paths now all have logic to free frames left in the list. ASTERISK-28609 Reported by: Ted G Change-Id: Ia1d7075857ebd26b47183c44b1aebb0d8f985f7a	7 years ago
Joshua C. Colp	b6572d35e4	confbridge: Add support for specifying maximum sample rate. ConfBridge has the ability to move between different sample rates for mixing the conference bridge. Up until now there has only been the ability to set the conference bridge to mix at a specific sample rate, or to let it move between sample rates as necessary. This change adds the ability to configure a conference bridge with a maximum sample rate so it can move between sample rates but only up to the configured maximum. ASTERISK-28658 Change-Id: Idff80896ccfb8a58a816e4ce9ac4ebde785963ee	7 years ago
George Joseph	b969773ad9	Merge "res_pjsip_outbound_registration: add support for SRV failover" into certified/16.3	7 years ago
George Joseph	f9b6586020	channel.c: Resolve issue with receiving SIP INFO packets for DTMF The problem is essentially the same as in ASTERISK~28245. Besides the direct media scenario we have an additional scenario where a special client is involved. This device mutes audio by default in transmit direction (no rtp frames) and activates audio only by a foot switch. In this situation dtmf input (pin for conferences, transfer features codes , etc) using SIP INFO mode is not understood properly especially when SIP INFO messages are sent quickly. This patch ensures that SIP INFO frames are properly queued and processed in the above scenario. The patch also corrects situations where successive dtmf events are received quicker than the signalled event duration (plus minimum gap/pause) allows, i.e. DTMF events have to be buffered in the ast channel read queue and emulation has to be processed asynchronously at slower speed. Reported by: Thomas Arimont patches: trigger_dtmf_emulation.patch submitted by Thomas Arimont (license 5525) Change-Id: I309bf61dd065c9978c8e48f5b9a936ab47de64c2	7 years ago
Kevin Harwell	572c088e66	Merge "parking: Fall back to parker channel name even if it matches parkee." into certified/16.3	7 years ago
Joshua Colp	bafd5f9908	parking: Fall back to parker channel name even if it matches parkee. ASTERISK-28631 Change-Id: Ia74d084799fbb9bee3403e30d2391aacd46243cc	7 years ago
Ben Ford	e4b024ec11	gitreview: Point to correct branch. Change-Id: I6f709a45a0d4ebadf8d51f64bd3509b1382d0496	7 years ago
Benjamin Keith Ford	bf9b0104d6	Merge "manager.c: Prevent the Originate action from running the Originate app" into certified/16.3	7 years ago
Ben Ford	e989cb4590	chan_sip.c: Prevent address change on unauthenticated SIP request. If the name of a peer is known and a SIP request is sent using that peer's name, the address of the peer will change even if the request fails the authentication challenge. This means that an endpoint can be altered and even rendered unusuable, even if it was in a working state previously. This can only occur when the nat option is set to the default, or auto_force_rport. This change checks the result of authentication first to ensure it is successful before setting the address and the nat option. ASTERISK-28589 #close Change-Id: I581c5ed1da60ca89f590bd70872de2b660de02df	7 years ago
George Joseph	4e80a32ebd	manager.c: Prevent the Originate action from running the Originate app If an AMI user without the "system" authorization calls the Originate AMI command with the Originate application, the second Originate could run the "System" command. Action: Originate Channel: Local/1111 Application: Originate Data: Local/2222,app,System,touch /tmp/owned If the "system" authorization isn't set, we now block the Originate app as well as the System, Exec, etc. apps. ASTERISK-28580 Reported by: Eliel Sardañons Change-Id: Ic4c9dedc34c426f03c8c14fce334a71386d8a5fa	7 years ago
Kevin Harwell	c551fd3ec9	res_pjsip_outbound_registration: add support for SRV failover ASTERISK-28624 Change-Id: I8da7c300dd985ab7b10dbd5194aff2f737808561	7 years ago
Joshua Colp	de568ddcd4	parking: Fix case where we can't get the parker. ASTERISK-28616 Change-Id: Iabe31ae38d01604284fcc5c2438d44e29a32ea4d	7 years ago
Joshua Colp	728a1ba77a	parking: Use channel snapshot instead of channel. There exists a scenario where a thread can hold a lock on the channels container while trying to lock a bridge. At the same time another thread can hold the lock for said bridge while attempting to retrieve a channel. This causes a deadlock. This change fixes this scenario by retrieving a channel snapshot instead of a channel, as information present in the snapshot is all that is needed. ASTERISK-28616 Change-Id: I68ceb1d62c7378addcd286e21be08a660a7cecf2	7 years ago
Kevin Harwell	a801a7dae2	res_pjsip_pubsub: change warning to debug The following message: "Subscription request from endpoint <blah> rejected. Expiration of 0 is invalid" Would sometimes spam the log with warnings if Asterisk restarted and a bunch of clients sent unsubscribes. This patch changes it from a warning to a debug message. Change-Id: I841ec42f65559f3135e037df0e55f89b6447a467	7 years ago
Kevin Harwell	d811289498	res_sorcery_memory_cache: stale item update leak When a stale item was being updated the object was being retrieved, but its reference was not being decremented after the update. This patch makes it so the object is now appropriately de-referenced. ASTERISK-28523 Change-Id: I9d8173d3a0416a242f4eba92fa0853279c500ec7	7 years ago
Joshua Colp	38c4d43820	chan_pjsip: Relock correct channel during "fax" redirect. When fax detection occurs on an outbound PJSIP channel the redirect operation will result in a masquerade occurring and the underlying channel on the session changing. The code incorrectly relocked the new channel instead of the old channel when returning. This resulted in the new channel being locked indefinitely. The code now always acts on the expected channel. ASTERISK-28538 Change-Id: I2b2e60d07e74383ae7e90d752c036c4b02d6b3a3	7 years ago
George Joseph	7535c54da1	Merge "test_utils.c: Skip test adsi_loaded_test if module not loaded." into certified/16.3	7 years ago
Kevin Harwell	93736ffc7f	AST-2019-004 - res_pjsip_t38.c: Add NULL checks before using session media After receiving a 200 OK with a declined stream in response to a T.38 initiated re-invite Asterisk would crash when attempting to dereference a NULL session media object. This patch checks to make sure the session media object is not NULL before attempting to use it. ASTERISK-28495 patches: ast-2019-004.patch submitted by Alexei Gradinari (license 5691) Change-Id: I168f45f4da29cfe739acf87e597baa2aae7aa572 (cherry picked from commit `965df3c228`)	7 years ago
Chris-Savinovich	eac6f2b0a2	test_utils.c: Skip test adsi_loaded_test if module not loaded. Module res_adsi.so is deprecated, therefore it does not load by default. Module not loaded causes it to yield a FAIL when tested by tests/test_utils.c. This fix checks if the corresponding module is loaded at the start of the test, and if not, it passes the test and exits with a message. This fix is applied to all versions where the module is marked deprecated. Change-Id: I52be64c8f6af222e15148a856d1f10cb113e1e94	7 years ago
George Joseph	e5b33bb979	CI: Escape backslashes in printenv/sort/tr Change-Id: I52be64c8f6af2bbe15148a856d1f10cb113e1e94	7 years ago
George Joseph	a9aa952e0f	CI: Add "throttle" label and "skip_gate" capability To make throttling by label fully active, the "throttle" option has to be specified with a specific label. You can now specify "skip_gate" in the Gerrit comments when you do a +2 code review to tell Jenkins not to actually run the gate. You'd do this if you plan to manually merge the change. Also updated the "printenv" debug output to better sort multi-line comments. Change-Id: I4c0b1085acec4805f2ca207eebac50aad81f27e2	7 years ago
George Joseph	475a8214fe	CI: Make node labels job-specific Originally, the eligible nodes for a job were labelled only by "swdev-docker". So basically any node could run any job. We had found that allowing a node to run more than 1 gate at a time was problematic so we limited the nodes to processing 1 job at a time. With the creation of the Asterisk 17 branches however, we now have so many active branches that getting checks and gates through in a timely manner is problematic when a node can run only 1 job at a time. Now the nodes are also labelled by the job type they can run. For instance: "asterisk-check", "asterisk-gate", etc. With the "Throttle Concurrent Builds" plugin, we can now allow a node to run more than 1 job BUT throttle by job type. For instance: Allow 2 jobs but only 1 asterisk-gate at a time. Now a node can run 2 checks or 1 check and 1 gate or 1 gate but not 2 gates at a time. Change-Id: I2032bf6afbcec5c341d9b852214c0c812d3d6db5	7 years ago
Joshua Colp	9bc22ef5ce	Merge ".gitreview: Update defaultbranch for certified" into certified/16.3	7 years ago
George Joseph	0a9a04338d	.gitreview: Update defaultbranch for certified Change-Id: I3dc35bdd6df92595f6941e92257986943fb50c2e	7 years ago
George Joseph	e6622f6697	CI: Don't enable non-core modules in Certified branches We don't support non-core modules for Certified releases but we were enabling them for CI builds which was causing lots of test failures. Now we don't. Change-Id: I0b3254c08a2479f3d39151690350cce5ce5ad766	7 years ago
Kevin Harwell	03d402a5ca	Merge "res_pjsip_messaging: Check for body in in-dialog message" into certified/16.3	7 years ago
George Joseph	e3180a9ef0	res_pjsip_messaging: Check for body in in-dialog message We now check that a body exists and it has a length > 0 before attempting to process it. ASTERISK-28447 Reported-by: Gil Richard Change-Id: Ic469544b22ab848734636588d4c93426cc6f4b1f	7 years ago
Francesco Castellano	f087d70ed0	chan_sip: Handle invalid SDP answer to T.38 re-invite The chan_sip module performs a T.38 re-invite using a single media stream of udptl, and expects the SDP answer to be the same. If an SDP answer is received instead that contains an additional media stream with no joint codec a crash will occur as the code assumes that at least one joint codec will exist in this scenario. This change removes this assumption. ASTERISK-28465 Change-Id: I8b02845b53344c6babe867a3f0a5231045c7ac87	7 years ago
Friendly Automation	79c18cb4c3	Merge "res_rtp_asterisk: Add ability to propose local address in ICE" into certified/16.3	7 years ago
George Joseph	3489fd015e	res_rtp_asterisk: Add ability to propose local address in ICE You can now add the "include_local_address" flag to an entry in rtp.conf "[ice_host_candidates]" to include both the advertized address and the local address in ICE negotiation: [ice_host_candidates] 192.168.1.1 = 1.2.3.4,include_local_address This causes both 192.168.1.1 and 1.2.3.4 to be advertized. Change-Id: Ide492cd45ce84546175ca7d557de80d9770513db	7 years ago
Joshua Colp	b62f3acd83	pjproject-bundled: Add upstream timer fixes Fixed #2191: - Stricter double timer entry scheduling prevention. - Integrate group lock in SIP transport, e.g: for add/dec ref, for timer scheduling. ASTERISK-28161 Reported-by: Ross Beer Change-Id: I2e09aa66de0dda9414d8a8259a649c4d2d96a9f5	7 years ago
George Joseph	43e1757b10	Prepare certified/16.3 * Update all extended and deprecated modules to set defaultenabled = no * Change .version to certified/16.3 Change-Id: Idc92d5c797678fba755536fe4a27297d959f67e9	7 years ago
Asterisk Development Team	60a3800ed2	Update for 16.3.0	7 years ago
Ben Ford	c4027dbf2d	build: Fix compiler warnings/errors. The compiler complained about a couple of variables that weren't initialized but were being used. Initializing them to NULL resolves the warnings/errors. ASTERISK-28362 #close Change-Id: I6243afc5459b416edff6bbf571b0489f6b852e4b	7 years ago
Asterisk Development Team	481e41a537	Update for 16.3.0-rc1	7 years ago
Sean Bright	6f5a6072ab	app_queue: Fix a few member pause bugs * Always set member->lastpause when setting member->paused * Fixed typo (using member->lastcall instead of member->lastpause) in 'queue show' output. * Use a constant 'now' in 'queue show' output for a better point-in-time view of time based stats. ASTERISK-27541 #close Reported by: César Benjamín García Martínez Change-Id: Ib41ced90cfdb66f9bb1e7b263d0f6fc1ac6e18fa	7 years ago
Ben Ford	64279e5992	alembic: Fix errors during upgrade head. When trying to upgrade using alembic, a couple different errors kept popping up that prevented the upgrade. An additional parameter was needed when changing the schema for mwi_subscribe_replaces_unsolicited from an integer to an enum. When changing from a string to an enum, the type needed to be cast for postgresql. The other issue was a parameter being used during column creation that did not exist. After fixing the upgrade process, it revealed errors with the downgrade process. One was a variable not being defined in the downgrade function, and the other was tables not existing when using MySQL. This was due to a context check that should have encompassed MySQL, but in the end was not doing so. Change-Id: Ib4d70cf3ce5080023a50be496272a777b55d6c8e	7 years ago
George Joseph	bc06a13123	Merge "pjsip: restrict function PJSIP_PARSE_URI to parse only SIP/SIPS URIs" into 16	7 years ago
Friendly Automation	eebc98fa44	Merge "manager: Use separate lock for session event notification." into 16	7 years ago
George Joseph	0bcbf97cb7	Merge "build: Add staging directories for future changes." into 16	7 years ago
Friendly Automation	5592ae465e	Merge "app_queue: Fix documentation for QUEUE_MEMBER function." into 16	7 years ago
Ben Ford	636e301f91	build: Add staging directories for future changes. This is the first step in changing the release process so that changes made to the CHANGES and UPGRADE.txt files do not result in merge conflicts every time multiple people modify these files. The changes made will go in these new directories: doc/CHANGES-staging and doc/UPGRADE-staging. The README.md files explain how things will work, but here's a little overview. When you make a change that would go in either CHANGES or UPGRADE.txt, this should instead be documented in a new file in the doc/CHANGES-staging or doc/UPGRADE-staging directory, respectively. The format will look like this: Subject: res_pjsip A description that explains the changes made and why. The release script will handle the bulleting and section separators! The 'Subject:' header is case-sensitive. You can still separate with new lines within your description. Subject: res_ari Master-Only: true You can have more than one subject, and they don't have to be the same! Also, the 'Master-Only' header should always be true and is also case-sensitive (but the value is not - you can have 'true' or 'True'). This header will only ever be present in the master branch. For more information, check out the wiki page: https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt This is an initial change for ASTERISK_28111. Functionally, this will make no difference, but it will prep the directories for when the changes from CHANGES and UPGRADE.txt are extracted. Change-Id: I8d852f284f66ac456b26dcb899ee46babf7d15b6	7 years ago
Friendly Automation	e1b0961fce	Merge "chan_sip: Ensure 'qualifygap' isn't negative" into 16	7 years ago

1 2 3 4 5 ...

31977 Commits (certified/16.3) All Branches Search

31977 Commits (certified/16.3)

All Branches