asterisk

Commit Graph

Author	SHA1	Message	Date
Asterisk Development Team	a2ed8a604d	Update for 13.36.0	6 years ago
Asterisk Development Team	4fb6a151f9	Update for 13.36.0-rc2	6 years ago
Joshua C. Colp	aca3c550b5	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	6 years ago
Sean Bright	7f46923ebd	app_voicemail: Fix pollmailboxes The name of the voicemail context was overwriting the name of the subscribed mailbox. Fix by simplifying how we create the MWI subscription. ASTERISK-29029 #close Change-Id: Ie8a7db6a0b68f3995b0846bbb733a21909ba44e5	6 years ago
Joshua C. Colp	8d6c2221a9	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	6 years ago
Asterisk Development Team	96106ac286	Update for 13.36.0-rc1	6 years ago
Dennis Buteyn	56f446482b	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	6 years ago
sungtae kim	53f9bbdc2d	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	7379a54424	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	6 years ago
Sean Bright	46fb9fb663	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	6 years ago
Michael Neuhauser	dd739f8a08	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
Joshua C. Colp	5c0d247d69	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
Sean Bright	718efe9365	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
Sean Bright	4fec6da25b	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	f6afac1b37	CI: Force publishAsteriskDocs to use python2 Change-Id: I7d951e75ad2d472fa096647dfb55670b11105e23	6 years ago
Sean Bright	ff90e00677	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
Joshua C. Colp	bcdc3ec645	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
Nickolay Shmyrev	f8f934b545	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	77a5383957	Update CHANGES and UPGRADE.txt for 13.35.0	6 years ago
George Joseph	c705e79d79	Scope Trace: Update stub defines Reminder: 13 doesn't support scope tracing but we're keeping the stub defines consistent across branches. Change-Id: If00733a4d5a4e829cbb4c934bcc2f4c1a80dab81	6 years ago
sungtae kim	52e1e362f0	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	f323ac334b	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	ef19d69c6b	PJSIP_MEDIA_OFFER: override what's specified on configuration When using the PSJIP_MEDIA_OFFER dialplan function it was not overriding an endpoint's configured codecs 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 definition no matter what. ASTERISK-28878 #close Change-Id: I0f7dc86fd0fb607c308e6f98ede303c54d1eacb6	6 years ago
Kevin Harwell	4bbc5be798	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	9fce541643	logger: Fix scope trace defines Although not implemented in the 13 branch, we need to keep the defines consistent across all branches. Change-Id: Id88ad71ca36818a35f1f14a5acb450e240366301	6 years ago
Joshua C. Colp	df509590a1	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	2513235859	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
Frederic LE FOLL	745ae35f0e	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	09370b0a2b	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	635885ec33	res_http_websocket: Add payload masking to the websocket client ASTERISK-28949 Change-Id: Id465030f2b1997b83d408933fdbabe01827469ca	6 years ago
Guido Falsi	a3af7ef7e7	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	1a40a6d55a	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	8576603faf	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	bcc7766bf6	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	50ba662b6a	res_pjsip: Include <pjsip_ua.h> instead of internal "pjsua-lib/pjsua.h" Change-Id: I24b5453df412232cf7f9a171ea4a34b35ad3ae78	6 years ago
Kevin Harwell	58eaf41e28	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
George Joseph	dd983edafe	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
Kevin Harwell	654ec4b908	Compiler fixes for gcc 10 This patch fixes a few compile warnings/errors that now occur when using gcc 10+. Also, the Makefile.rules check to turn off partial inlining in gcc versions greater or equal to 8.2.1 had a bug where it only it only checked against versions with at least 3 numbers (ex: 8.2.1 vs 10). This patch now ensures any version above the specified version is correctly compared. Change-Id: I54718496eb0c3ce5bd6d427cd279a29e8d2825f9	6 years ago
Walter Doekes	1e5ea08e0c	pjsip: Prevent invalid memory access when attempting to contact a non-sip URI You cannot cast a pjsip_uri to a pjsip_sip_uri using pjsip_uri_get_uri, without checking that it's a PJSIP_URI_SCHEME_IS_SIP(S). ASTERISK-28936 Change-Id: I9f572b3677e4730458e9402719e580f8681afe2a	6 years ago
Joshua C. Colp	f30a61022a	res_fax: Don't consume frames given to fax gateway on write. In a particular fax gateway scenario whereby it would have to translate using the read translation path on a channel the frame being translated would be consumed. When the frame is in the write path it is not permitted to free the frame as the caller expects it to continue to exist. This change makes it so that the frame is only consumed on the read path where it is acceptable to free it. ASTERISK-28900 Change-Id: I011c321288a1b056d92b37c85e229f4a28ee737d	6 years ago
Alexander Traud	439f782452	pjproject_bundled: Honor --without-pjproject. The previous change missed that 'make' uses 'PJPROJECT_BUNDLED' anyway. ASTERISK-28929 Change-Id: I7ef0e78a06ea391b59d95b99d46bbed3fec4fed9	6 years ago
Pirmin Walthert	84da8622f6	res_pjsip_logger: use the correct pointer when logging tx_messages to pcap When writing tx messages to pcap files, Asterisk is using the wrong pointer resulting in lots of wasted space. This patch fixes it to use the correct pointer. ASTERISK-28932 #close Change-Id: I5b8253dd59a083a2ca2c81f232f1d14d33c6fd23	6 years ago
George Joseph	edf9471e05	Scope Tracing: Add stubs to allow easier cherry-picking Scope tracing isn't supported in Asterisk 13 due to changes made to logging between 13 and 16 but since the scope tracing macros may be present in the 16, 17 and master branches, those macros are defined here as NOOPs so cherry-picking changes downward to 13 can still be seamless. Change-Id: I0390ce5651374d8f3e06d7620050acb22c5440a2	6 years ago
Joshua C. Colp	e7f7c8c9de	res_rtp_asterisk: Re-order RTP destruction. The destructor for RTP deallocated transport resources before terminating the ICE support. This could result in a crash as the thread handling ICE would access already freed parts of the RTP data. This change re-orders the destruction so that ICE is stopped before destroying things. ASTERISK-28885 Change-Id: Ie71add549f12b6cdea7e9dbf976d1bd1d2fc0bdc	6 years ago
Pirmin Walthert	81b86702fb	res_pjsip_logger.c: correct the return value checks when writing to pcap files fwrite() does return the number of elements written and not the number of bytes. However asterisk is currently comparing the return value to the size of the written element what means that asterisk logs five WARNING messages on every packet written to the pcap file. This patch changes the code to check for the correct value, which will always be 1. ASTERISK-28921 #close Change-Id: I2455032d9cb4c5a500692923f9e2a22e68b08fc2	6 years ago
Asterisk Development Team	07bf7890dc	Update CHANGES and UPGRADE.txt for 13.34.0	6 years ago
Joshua C. Colp	c8cabb08aa	res_pjsip: Use correct pool for storing the contact_user value. When replacing the user portion of the Contact URI the code was using the ephemeral pool instead of the tdata pool. This could cause the Contact user value to become invalid after a period of time. The code will now use the tdata pool which persists for the lifetime of the message instead. ASTERISK-28794 Change-Id: I31e7b958e397cbdaeedd0ebb70bcf8dd2ed3c4d5	6 years ago
Pirmin Walthert	4b7293f44e	res_pjsip_nat.c: remove x-ast-orig-host from request URI and To header While asterisk is filtering out the x-ast-orig-host parameter from the contact on response messages, it is not filtering it out from the request URI and the to header on SIP requests (for example INVITE). ASTERISK-28884 #close Change-Id: Id032b33098a1befea9b243ca994184baecccc59e	6 years ago
Joshua C. Colp	0d86f49f7f	res_sorcery_config: Always reload configuration on errors. When a configuration file in Asterisk is loaded information about it is stored such that on a reload it is not reloaded if nothing has changed. This can be problematic when an error exists in a configuration file in PJSIP since the error will be output at start and not subsequently on reload if the file is unchanged. This change makes it so that if an error is encountered when res_sorcery_config is loading a configuration file a reload will always read in the configuration file, allowing the error to be seen easier. Change-Id: If2e05a017570f1f5f4f49120da09601e9ecdf9ed	6 years ago
Joshua C. Colp	17a878dd99	bridge_softmix: Always remove audio from mixed frame. When receiving audio from a channel we determine if it is talking or silence based on a threshold value. If this threshold is met we always mix the audio into the conference bridge. If this threshold is not met we also mix the audio into the conference bridge UNLESS the drop silence option is enabled. The code that removed the audio from the mixed frame assumed that it was always not present if it did not meet the threshold to be considered talking. This is incorrect. If it has been stated that the audio was mixed into the mixed frame then it has been mixed into the mixed frame. By not removing audio that was considered non-talking it was possible for a channel to receive a slight echo of audio of itself at times. This change ensures that the audio is always removed from the mixed frame going back to the channel so it no longer receives the slight echo. ASTERISK-28898 Change-Id: I7b1b582cc1bcdb318ecc60c9d2e3d87ae31d55cb	6 years ago

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31482 Commits (13.36) All Branches Search

31482 Commits (13.36)

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