asterisk

Commit Graph

Author	SHA1	Message	Date
Asterisk Development Team	52108ad084	Update for 17.7.0	5 years ago
Asterisk Development Team	81ccbe6a8c	Update for 17.7.0-rc2	5 years ago
Joshua C. Colp	b489557c46	parking: Copy parker UUID as well. When fixing issues uncovered by GCC10 a copy of the parker UUID was removed accidentally. This change restores it so that the subscription has the data it needs. ASTERISK-29042 Change-Id: I7d396a14ea648bd26d3c363dd78e78bd386b544a	5 years ago
Joshua C. Colp	309895efcd	pbx: Fix hints deadlock between reload and ExtensionState. When the ExtensionState AMI action is executed on a pattern matched hint it can end up adding a new hint if one does not already exist. This results in a locking order of contexts -> hints -> contexts. If at the same time a reload is occurring and adding its own hint it will have a locking order of hints -> contexts. This results in a deadlock as one thread wants a lock on contexts that the other has, and the other thread wants a lock on hints that the other has. This change enforces a hints -> contexts locking order by explicitly locking hints in the places where a hint is added when queried for. This matches the order seen through normal adding of hints. ASTERISK-29046 Change-Id: I49f027f4aab5d2d50855ae937bcf5e2fd8bfc504	5 years ago
Asterisk Development Team	25c644736c	Update for 17.7.0-rc1	5 years ago
Dennis Buteyn	53c06791fd	chan_sip: Clear ToHost property on peer when changing to dynamic host The ToHost parameter was not cleared when a peer's host value was changed to dynamic. This causes invites to be sent to the original host. ASTERISK-29011 #close Change-Id: I9678d512741f71baca8f131a65b7523020b07d5c	5 years ago
cmaj	2a96c52c0c	Makefile: Fix certified version numbers Adds sed before awk to produce reasonable ASTERISKVERSIONNUM on certified versions of Asterisk eg. 16.8-cert3 is 160803 instead of the previous 00800. ASTERISK-29021 #close Change-Id: Icf241df0ff6db09011b8c936a317a84b0b634e16	5 years ago
George Joseph	50a98c37c8	scope_trace: Add/update utilities * Added a AST_STREAM_STATE_END sentinel * Add ast_stream_to_str() * Add ast_stream_state_to_str() * Add ast_stream_get_format_count() * Add ast_stream_topology_to_str() * Add ast_stream_topology_get_active_count() * Add ast_format_cap_append_names() * Add ast_sip_session_get_name() Change-Id: I132eb5971ea41509c660f64e9113cda8c9013b0b	5 years ago
Sean Bright	4167cc1be4	res_musiconhold.c: Prevent crash with realtime MoH The MoH class internal file vector is potentially being manipulated by multiple threads at the same time without sufficient locking. Switch to a reference counted list and operate on copies where necessary. ASTERISK-28927 #close Change-Id: I479c5dcf88db670956e8cac177b5826c986b0217	5 years ago
Sean Bright	f606312b8f	vector.h: Fix implementation of AST_VECTOR_COMPACT() for empty vectors The assumed behavior of realloc() - that it was effectively a free() if its second argument was 0 - is Linux specific behavior and is not guaranteed by either POSIX or the C specification. Instead, if we want to resize a vector to 0, do it explicitly. Change-Id: Ife31d4b510ebab41cb5477fdc7ea4e3138ca8b4f	5 years ago
Michael Neuhauser	bfee7e1550	pjproject: clone sdp to protect against (nat) modifications PJSIP, UDP transport with external_media_address and session timers enabled. Connected to SIP server that is not in local net. Asterisk initiated the connection and is refreshing the session after 150s (timeout 300s). The 2nd refresh-INVITE triggered by the pjsip timer has a malformed IP address in its SDP (garbage string). This only happens when the SDP is modified by the nat-code to replace the local IP address with the configured external_media_address. Analysis: the code to modify the SDP (in res_pjsip_session.c:session_outgoing_nat_hook() and also (redundantly?) in res_pjsip_sdp_rtp.c:change_outgoing_sdp_stream_media_address()) uses the tdata->pool to allocate the replacement string. But the same pjmedia_sdp_stream that was modified for the 1st refresh-INVITE is also used for the 2nd refresh-INVITE (because it is stored in pjmedia's pjmedia_sdp_neg structure). The problem is, that at that moment, the tdata->pool that holds the stringified external_media_address from the 1. refresh-INVITE has long been reused for something else. Fix by Sauw Ming of pjproject (see https://github.com/pjsip/pjproject/pull/2476): the local, potentially modified pjmedia_sdp_stream is cloned in pjproject/source/pjsip/src/pjmedia/sip_neg.c:process_answer() and the clone is stored, thereby detaching from the tdata->pool (which is only released after process_answer()) ASTERISK-28973 Reported-by: Michael Neuhauser Change-Id: I272ac22436076596e06aa51b9fa23fd1c7734a0e	6 years ago
George Joseph	5042ebf4ff	res_pjsip_session: Ensure reused streams have correct bundle group When a bundled stream is removed, its bundle_group is reset to -1. If that stream is later reused, the bundle parameters on session media need to be reset correctly it could mistakenly be rebundled with a stream that was removed and never reused. Since the removed stream has no rtp instance, a crash will result. Change-Id: Ie2b792220f9291587ab5f9fd123145559dba96d7	6 years ago
Sean Bright	87e79e6f9c	utf8.c: Add UTF-8 validation and utility functions There are various places in Asterisk - specifically in regards to database integration - where having some kind of UTF-8 validation would be beneficial. This patch adds: * Functions to validate that a given string contains only valid UTF-8 sequences. * A function to copy a string (similar to ast_copy_string) stopping when an invalid UTF-8 sequence is encountered. * A UTF-8 validator that allows for progressive validation. All of this is based on the excellent UTF-8 decoder by Björn Höhrmann. More information is available here: https://bjoern.hoehrmann.de/utf-8/decoder/dfa/ The API was written in such a way that should allow us to replace the implementation later should we determine that we need something more comprehensive. Change-Id: I3555d787a79e7c780a7800cd26e0b5056368abf9	6 years ago
Joshua C. Colp	4ac3d4cdbd	res_pjsip_registrar: Don't specify an expiration for static contacts. Statically configured contacts on an AOR don't have an expiration time so when adding them to the resulting 200 OK if an endpoint registers ensure they are marked as such. ASTERISK-28995 Change-Id: I9f0e45eb2ccdedc9a0df5358634a19ccab0ad596	6 years ago
sungtae kim	ade20df66b	stasis_bridge.c: Fixed wrong video_mode shown Currently, if the bridge has created by the ARI, the video_mode parameter was not shown in the BridgeCreated event correctly. Fixed it and added video_mode shown in the 'bridge show <bridge id>' cli. ASTERISK-28987 Change-Id: I8c205126724e34c2bdab9380f523eb62478e4295	6 years ago
Sean Bright	7907c24e6a	vector.h: Add AST_VECTOR_SORT() Allows a vector to be sorted in-place, rather than only during insertion. Change-Id: I22cba9ddf556a7e44dacc53c4431bd81dd2fa780	6 years ago
George Joseph	a3aed44d12	CI: Force publishAsteriskDocs to use python2 Change-Id: I7d951e75ad2d472fa096647dfb55670b11105e23	6 years ago
Joshua C. Colp	a613e07e76	websocket / pjsip: Increase maximum packet size. When dealing with a lot of video streams on WebRTC the resulting SDPs can grow to be quite large. This effectively doubles the maximum size to allow more streams to exist. The res_http_websocket module has also been changed to use a buffer on the session for reading in packets to ensure that the stack space usage is not excessive. Change-Id: I31d4351d70c8e2c11564807a7528b984f3fbdd01	6 years ago
Joshua C. Colp	ba1472b960	pjsip: Include timer patch to prevent cancelling timer 0. I noticed this while looking at another issue and brought it up with Teluu. It was possible for an uninitialized timer to be cancelled, resulting in the invalid timer id of 0 being placed into the timer heap causing issues. This change is a backport from the pjproject repository preventing this from happening. Change-Id: I1ba318b1f153a6dd7458846396e2867282b428e7	6 years ago
Sean Bright	7d487b8cb6	acl.c: Coerce a NULL pointer into the empty string If an ACL is misconfigured in the realtime database (for instance, the "rule" is blank) and Asterisk attempts to read the ACL, Asterisk will crash. ASTERISK-28978 #close Change-Id: Ic1536c4df856231bfd2da00128f7822224d77610	6 years ago
Nickolay Shmyrev	c3b3b681fb	res_http_websocket: Avoid reading past end of string We read beyond the end of the buffer when copying the string out of the buffer when we used ast_copy_string() because the original string was not null terminated. Instead switch to ast_strndup() which does not exhibit the same behavior. ASTERISK-28975 #close Change-Id: Ib4a75cffeb1eb8cf01136ef30306bd623e531a2a	6 years ago
Asterisk Development Team	11cee28e4f	Update CHANGES and UPGRADE.txt for 17.6.0	6 years ago
George Joseph	e61d045d96	frame.c: Make debugging easier * ast_frame_subclass2str() and ast_frame_type2str() now return a pointer to the buffer that was passed in instead of void. This makes it easier to use these functions inline in printf-style debugging statements. * Added many missing control frame entries in ast_frame_subclass2str. Change-Id: Ifd0d6578e758cd644c96d17a5383ff2128c572fc	6 years ago
George Joseph	bb1af1b6fc	Scope Trace: Make it easier to trace through synchronous tasks Tracing through synchronous tasks was a little troublesome because the new thread's stack counter reset to 0. This change allows a synchronous task to set its trace level to be the same as the thread that pushed the task. For now, the task's level has to be passed in the task's data structure but a future enhancement to the taskprocessor subsystem could automatically set the trace level of the servant to be that of the caller. This doesn't really make sense for async tasks because you never know when they're going to run anyway. Change-Id: Ib8049c0b815063a45d8c7b0cb4e30b7b87b1d825	6 years ago
sungtae kim	7c12303c4c	res_pjsip.c: Added disable_rport option for pjsip.conf Currently when the pjsip making an outgoing request, it keep adding the rport parameter in a request message as a default. This causes unexpected rport handle at the other end. Added option for disable this behaviour in the pjsip.conf. This is a system option, but working as a gloabl option. ASTERISK-28959 Change-Id: I9596675e52a742774738b5aad5d1fec32f477abc	6 years ago
Nickolay Shmyrev	b96b4ec438	res_http_websocket.c: Continue reading after ping/pong Do not return error if the client received ping frame while looking for a string and just wait for another frame. ASTERISK-28958 #close Change-Id: I4d06b4827bd71e56cbaafc011ffdcef9f0332922	6 years ago
Kevin Harwell	bdcaa275da	PJSIP_MEDIA_OFFER: override configuration on refresh When using the PSJIP_MEDIA_OFFER dialplan function it was not overriding an endpoint's configured codecs on refresh unless they had a shared codec between the two. This patch makes it so whatever is set using PJSIP_MEDIA_OFFER is used when creating the SDP for a refresh no matter what. ASTERISK-28878 #close Change-Id: I0f7dc86fd0fb607c308e6f98ede303c54d1eacb6	6 years ago
Kevin Harwell	ef78162780	manager - Add Content-Type parameter to the SendText action This patch allows a user of AMI to now specify the type of message content contained within by setting the 'Content-Type' parameter. Note, the AMI version has been bumped for this change. ASTERISK-28945 #close Change-Id: Ibb5315702532c6b954e1498beddc8855fabdf4bb	6 years ago
George Joseph	86f6238e08	Scope Trace: Add some new tracing macros and an ast_str helper Created new SCOPE_ functions that don't depend on RAII_VAR. Besides generating less code, the use of the explicit SCOPE_EXIT macros capture the line number where the scope exited. The RAII_VAR versions can't do that. * SCOPE_ENTER(level, ...): Like SCOPE_TRACE but doesn't use RAII_VAR and therefore needs needs one of... * SCOPE_EXIT(...): Decrements the trace stack counter and optionally prints a message. * SCOPE_EXIT_EXPR(__expr, ...): Decrements the trace stack counter, optionally prints a message, then executes the expression. SCOPE_EXIT_EXPR(break, "My while got broken\n"); * SCOPE_EXIT_RTN(, ...): Decrements the trace stack counter, optionally prints a message, then returns without a value. SCOPE_EXIT_RTN("Bye\n"); * SCOPE_EXIT_RTN_VALUE(__return_value, ...): Decrements the trace stack counter, optionally prints a message, then returns the value specified. SCOPE_EXIT_RTN_VALUE(rc, "Returning with RC: %d\n", rc); Create an ast_str helper ast_str_tmp() that allocates a temporary ast_str that can be passed to a function that needs it, then frees it. This makes using the above macros easier. Example: SCOPE_ENTER(1, Format Caps 1: %s Format Caps 2: %s\n", ast_str_tmp(32, ast_format_cap_get_names(cap1, &STR_TMP), ast_str_tmp(32, ast_format_cap_get_names(cap2, &STR_TMP)); The calls to ast_str_tmp create an ast_str of the specified initial length which can be referenced as STR_TMP. It then calls the expression, which must return a char *, ast_strdupa's it, frees STR_TMP, then returns the ast_strdupa'd string. That string is freed when the function returns. Change-Id: I44059b20d55a889aa91440d2f8a590865998be51	6 years ago
Joshua C. Colp	f129bbe387	res_pjsip: Apply AOR outbound proxy to static contacts. The outbound proxy for an AOR was not being applied to any statically configured Contacts. This resulted in the OPTIONS requests being sent to the wrong target. This change sets the outbound proxy on statically configured contacts once the AOR configuration is done being applied. ASTERISK-28965 Change-Id: Ia60f3e93ea63f819c5a46bc8b54be2e588dfa9e0	6 years ago
Joshua C. Colp	ec90811e3e	menuselect: Resolve infinite loop in dependency scenario. Given a scenario where a module has a dependency on both an external library and a module if the external library was available and the module was not an infinite loop would occur. This happened due to the code changing the dependecy status to no failure on each dependency checking loop iteration, resulting in the code thinking that it had gone from no failure to failure each time triggering another dependency check. This change makes it so that the old dependency status is preserved throughout the dependency checking allowing it to determine that after the first iteration the dependency status does not transition from no failure to failure. ASTERISK-28930 Change-Id: Iea06d45d9fd6d8bfd068882a0bb7e23a53ec3e84	6 years ago
Kevin Harwell	d62a5b5be9	chan_pjsip: don't use PJSIP_SC_NULL as it only exists pjproject 2.8+ A patch made a reference to the PJSIP_SC_NULL enumeration value, which was added to pjproject 2.8 and above thus making it so Asterisk would fail to compile with prior versions of pjproject. This patch removes the reference, and instead initializes the value to '0'. ASTERISK-28886 #close Change-Id: I68491c80da1a0154b2286c9458440141c98db9d7	6 years ago
Frederic LE FOLL	1dabc61bb5	chan_sip: chan_sip does not process 400 response to an INVITE. chan_sip handle_response() function, for a 400 response to an INVITE, calls handle_response_invite() and does not generate ACK. handle_response_invite() does not recognize 400 response and has no default response processing for unexpected responses, thus it does not generate ACK either. The ACK on response repetition comes from handle_response() mechanism "We must re-send ACKs to re-transmitted final responses". According to code history, 400 response specific processing was introduced with commit "channels/chan_sip: Add improved support for 4xx error codes" This commit added support for : - 400/414/493 in handle_response_subscribe() handle_response_register() and handle_response(). - 414/493 only in handle_response_invite(). This fix adds 400 response support in handle_response_invite(). ASTERISK-28957 Change-Id: Ic71a087e5398dfc7273946b9ec6f9a36960218ad	6 years ago
Università di Bologna - CESIA VoIP	6462987a03	res_corosync: Fix crash in huge distributed environment. 1) Fix memory-leaks Added code to release ast_events extracted from corosync and stasis messages 2) Clean stasis cache when a member of the corosync cluster leaves the group Added code to remove from the stasis cache of the members remained on the group all the messages with the EID of the left member. If the device states of the left member remain in the stasis cache of other members, they will not be updated anymore and high priority cached values, like BUSY, will take precedence over current device states. 3) Stop corosync event propagation when node is not joined to the group Updated dispatch_thread_handler code to detect when asterisk is not joined to the corosync group and added some condition in publish_event_to_corosync code to send corosync messages only when joined. When a node is not joined its corosync daemon can't send messages: the cpg_mcast_joined function append new messages to the FIFO buffer until it's full and then it blocks indefinitely. In this scenario if the stasis_message_cb callback, registered by res_corosync to handle stasis messages, try to send a corosync messages, the thread of the stasis thread-pool will be blocked until the node join the corosync cluster. ASTERISK-28888 Reported by: Università di Bologna - CESIA VoIP Change-Id: Ie8e99bc23f141a73c13ae6fb1948d148d4de17f2	6 years ago
Moises Silva	6533d02e14	res_http_websocket: Add payload masking to the websocket client ASTERISK-28949 Change-Id: Id465030f2b1997b83d408933fdbabe01827469ca	6 years ago
Guido Falsi	075510fd41	chan_dadhi: Fix setvar in dahdi channels The change to how setvar works for various channels performed in ASTERISK~23756 missed some required change in the dahdi channel, where the variables are actually set while reading configuration. This change should fix the issue. ASTERISK-28955 Change-Id: Ibfeb7f8cbdd735346dc4028de6a265f24f9df274	6 years ago
Joshua C. Colp	bc7ce46199	app_stream_echo: Fix state of added streams. When stream support was added to Asterisk the stream state was used inconsistently, resulting in odd behavior. This was then standardized to be the state of a stream from the perspective of Asterisk. This change updates the StreamEcho dialplan application to use the correct state, send only, since we are only sending to the endpoint and not expecting them to send us multiple video streams. ASTERISK-28954 Change-Id: I35bfd533ef1184ffe62586b22bbd253c82872a56	6 years ago
Joshua C. Colp	9589f97c9e	res_pjsip_session: Preserve label on incoming re-INVITE. When a re-INVITE is received we create a new set of streams that are then swapped in as the active streams. We did not preserve the SDP label from the previous streams, resulting in the label getting lost. This change ensures that if an SDP label is present on the previous stream then it is set on the new stream. ASTERISK-28953 Change-Id: I9dd63b88b562fe96ce5c791a3dae5bcaca258445	6 years ago
Joshua C. Colp	79ebbca3a7	res_sorcery_memory_cache: Disallow per-object expire with full backend. The AMI action and CLI command did not take into account the properties of full backend caching. This resulted in an expired object remaining removed until a full backend update occurred, instead of having the object updated when needed. This change makes it so that the AMI action and CLI command for object expire will now fail instead of putting the cache into an undesired state. If full backend caching is enabled then only operations which act on the entire cache are available. ASTERISK-28942 Change-Id: Id662d888f177ab566c8e802ad583083b742d21f4	6 years ago
Walter Doekes	266d81c296	app_queue: Read latest wrapuptime instead of (possibly stale) copy Before this changeset, it was possible that a queue member (agent) was called even though they just got out of a call, and wrapuptime seconds hadn't passed yet. This could happen if a member ended a call _between_ a new call attempt and asterisk trying that particular member for a new call. In that case, Asterisk would check the hangup time of the call-before-the-last-call instead of the hangup time of the-last-call. ASTERISK-28952 Change-Id: Ie0cab8f0e8d639c01cba633d4968ba19873d80b3	6 years ago
Walter Doekes	1684418a65	app_queue: Remove stale code in try_calling Because ring_entry() is not called, outgoing->chan is not touched here either. ASTERISK-28950 ASTERISK-28644 Change-Id: I564613715dfaf45af868251eb75a451f512af90f	6 years ago
Walter Doekes	4bc0f24e97	res_pjsip: Include <pjsip_ua.h> instead of internal "pjsua-lib/pjsua.h" Change-Id: I24b5453df412232cf7f9a171ea4a34b35ad3ae78	6 years ago
Kevin Harwell	3b99479934	pjproject: Upgrade bundled version to pjproject 2.10 This patch makes the usual necessary changes when upgrading to a new version pjproject. For instance, version number bump, patches removed from third-party, new *.md5 file added, etc.. This patch also includes a change to the Asterisk pjproject Makefile to explicitly create the 'source/pjsip-apps/lib' directory. This directory is no longer there by default so needs to be added so the Asterisk malloc debug can be built. This patch also includes some minor changes to Asterisk that were a result of the upgrade. Specifically, there was a backward incompatibility change made in 2.10 that modified the "expires header" variable field from a signed to an unsigned value. This potentially effects comparison. Namely, those check for a value less than zero. This patch modified a few locations in the Asterisk code that may have been affected. Lastly, this patch adds a new macro PJSIP_MINVERSION that can be used to check a minimum version of pjproject at compile time. ASTERISK-28899 #close Change-Id: Iec8821c6cbbc08c369d0e3cd2f14e691b41d0c81	6 years ago
Joshua C. Colp	47c4a3becf	core_unreal / core_local: Add multistream and re-negotiation. When requesting a Local channel the requested stream topology or a converted stream topology will now be placed onto the resulting channels. Frames written in on streams will now also preserve the stream identifier as they are queued on the opposite channel. Finally when a stream topology change is requested it is immediately accepted and reflected on both channels. Each channel also receives a queued frame to indicate that the topology has changed. ASTERISK-28938 Change-Id: I4e9d94da5230d4bd046dc755651493fce1d87186	6 years ago
sungtae kim	37f8adacd1	res_ari: Fix create channel request channelId parameter parsing If channelId parameters were passed in the body, the Asterisk doesn't parsing it correctly. Fixed it to parse the channelId, other_channel_id parameter correclty. ASTERISK-28948 Change-Id: I59b49161a94869169ee19c1ffab5afcef7026157	6 years ago
Joshua C. Colp	3dbbdfa8c3	res_rtp_asterisk: Don't assume setting retrans props means to enable. The "value" passed in when setting an RTP property determines whether it should be enabled or disabled. The RTP send and receive retrans props did not examine this to know if the buffers should be enabled. They assumed they always should be. This change makes it so that the "value" passed in is respected. ASTERISK-28939 Change-Id: I9244cdbdc5fd065c7f6b02cbfa572bc55c7123dc	6 years ago
Joshua C. Colp	9be19243f6	bridge_softmix: Add additional old states for adding new source. There are three states that an old stream can be in to allow becoming a source stream in a new stream: 1. Removed 2. Inactive 3. Sendonly This change adds the two missing ones, inactive and sendonly, so if a stream transitions from those to a state where they are providing video to Asterisk we properly re-negotiate the other participants. ASTERISK-28944 Change-Id: Id8256b9b254b403411586284bbaedbf50452de01	6 years ago
George Joseph	816a082470	res_fax: Don't start a gateway if either channel is hung up When fax_gateway_framehook is called and a gateway hasn't already been started, the framehook gets the t38 state for both the current channel and the peer. That call trickles down to the channel driver which determines the state. If either channel is hung up (or in the process of being hung up), the channel driver's tech_pvt is going to be NULL which, in the case of chan_pjsip, will cause a segfault. * Added a hangup check for both the channel and peer channel before starting a fax gateway. * Added a check for NULL tech_pvt to chan_pjsip_queryoption so we don't attempt to reference a tech_pvt that's already gone. ASTERISK-28923 Reported by: Yury Kirsanov Change-Id: I4e10e63b667bbb68c1c8623f977488f5d807897c	6 years ago
George Joseph	bc0207d0dd	app_confbridge: Plug ref leak of bridge channel with send_events When send_events is enabled for a user, we were leaking a reference to the bridge channel in confbridge_manager.c:send_message(). This also caused the bridge snapshot to not be destroyed. Change-Id: I87a7ae9175e3cd29f6d6a8750e0ec5427bd98e97	6 years ago
Ben Ford	246615925f	cli.c: Fix compiler error. Added default variable value to fix a compiler error. Change-Id: I7b592adbb1274dc5464dea1c5e5de0685c928553	6 years ago

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32691 Commits (17.7) All Branches Search

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