asterisk

Commit Graph

Author	SHA1	Message	Date
Sungtae Kim	519bb01d5f	res_stasis.c: Added video_single option for bridge creation Currently, it was not possible to create bridge with video_mode single. This made hard to put the bridge in a vidoe_single mode. So, added video_single option for Bridge creation using the ARI. This allows create a bridge with video_mode single. ASTERISK-29055 Change-Id: I43e720e5c83fc75fafe10fe22808ae7f055da2ae	5 years ago
Patrick Verzele	d2cc36e4a8	res_pjsip_session: Deferred re-INVITE without SDP send a=sendrecv instead of a=sendonly Building on ASTERISK-25854. When the device requests hold by sending SDP with attribute recvonly, asterisk places the session in sendonly mode. When the device later requests to resume the call by using a re-INVITE excluding SDP, asterisk needs to change the sendonly mode to sendrecv again. Change-Id: I60341ce3d87f95869f3bc6dc358bd3e8286477a6	5 years ago
Joshua C. Colp	cc60a0fb6f	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
Nickolay Shmyrev	7f5cc9ad28	res_speech: Bump reference on format object Properly bump reference on format object to avoid memory corruption on double free ASTERISK-29040 #close Change-Id: Ic5a7faabfe2ef965ddb024186e1de7ca4542e2a3	6 years ago
Torrey Searle	25f0588c0f	res_pjsip_diversion: handle 181 Adapt the response handler so it also called when 181 is received. In the case 181 is received, also generate the 181 response. ASTERISK-29001 #close Change-Id: I73cfee46a8ca85371280ebdb38674f8fde7510df	6 years ago
Joshua C. Colp	a815f02597	res_pjsip_session: Don't aggressively terminate on failed re-INVITE. Per the RFC when an outgoing re-INVITE is done we should only terminate the dialog if a 481 or 408 is received. ASTERISK-29033 Change-Id: I6c3ff513aa41005d02de0396ba820083e9b18503	6 years ago
Sean Bright	315b5f3929	res_musiconhold.c: Use ast_file_read_dir to scan MoH directory Two changes of note in this patch: * Use ast_file_read_dir instead of opendir/readdir/closedir * If the files list should be sorted, do that at the end rather than as we go which improves performance for large lists Change-Id: Ic7e9c913c0f85754c99c74c9cf6dd3514b1b941f	6 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	6 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	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
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
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
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
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
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
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
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	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
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
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
Kevin Harwell	3726992e6b	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
sungtae kim	a9dea6a38e	res_ari: Fix create request body parameter parsing. If parameters were passed in the body as JSON to the create route they were not being parsed before checking to ensure that required fields were set. This change moves the parsing so it occurs before checking. ASTERISK-28940 Change-Id: I898b4c3c7ae1cde19a6840e59f498822701cf5cf	6 years ago
Walter Doekes	cd82d5ab42	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	7a151eae74	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
Pirmin Walthert	a2b5ca0479	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
Pirmin Walthert	5bb6d2766c	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
Joshua C. Colp	62611df412	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	10c5562e3b	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
Alexander Traud	ab0ab19993	res_srtp: Set all possible flags while selecting the Crypto Suite. The flags of a previous selection could have been set within the object 'srtp', for example, when the previous selection returned failure after setting just 'some' flags. Now, not to clutter the code, all possible flags are cleared first, and then the selected flags are set as before. ASTERISK-28903 Change-Id: I1b9d7aade7d5120244ce7e3a8865518cbd6e0eee	6 years ago
Joshua C. Colp	441b614fa0	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	d7d2d13b2c	res_pjsip_logger: Expand functionality to improve logging. The PJSIP packet logger now has the following CLI commands: pjsip set logger pcap <filename> When used this will create a pcap file containing the incoming and outgoing SIP packets, in unencrypted form. pjsip set logger verbose <on / off> This allows you to toggle logging to verbose on and off. pjsip set logger host <IP/subnet mask> add This allows you to add an additional IP address or subnet mask to logging, allowing you to log multiple instead of just a single IP address or all traffic. The normal "pjsip set logger host" CLI command has also been expanded to allow subnet masks as well. ASTERISK-28895 Change-Id: If5859161a72b0d7dd2d1f92d45bed88e0cd07d0e	6 years ago
Nicholas John Koch	55d76ae663	res_musiconhold: Added check for dot character in path of playlist entries to avoid warnings A warning was triggered that there may be a problem regarding file extension (which is correct and should not be set anyway). The warning also appeared if there was dot within the path itself. E.g. [sales-queue-hold] mode=playlist entry=/var/www/domain.tld/moh/funky_music The music played correctly but you get a warning message. Now there will be a check if the position of a potential dot character is after the last position of a slash character. This dot charachter will be treated as a extension naming. Dots within the path then ignored. ASTERISK-28892 Reported-By: Nicholas John Koch Change-Id: I2ec35a613413affbf5fcc01c8c181eba24865b9e	6 years ago
sungtae kim	e2c5d2908b	res_rtp_asterisk.c: Fixed memory leak Added freeifaddrs() for memory releasing. ASTERISK-28904 Change-Id: I109403866e85a30659351946903a679de9727a8f	6 years ago
Joshua C. Colp	ace1000ff3	ari: Allow variables to be set on channel create. This change adds the same variable functionality that is available for originating a channel to the create call. Now when creating a channel you can specify dialplan variables to set instead of having to do another API call. ASTERISK-28896 Change-Id: If13997ba818136d7c070585504fc4164378aa992	6 years ago
Roger James	e0c6e1b71b	pjsip_resolver.c: Ensure AAAA dns requests are made. 1. Modify sip_resolve and sip_resolve_callback to request AAAA lookups when an IPV6 transport type has been requested. 2. Rename all occurrences of pjsip_transport_get_type_name to pjsip_transport_get_type_desc. This ensures that the log/debug info shows whether the transport is IPv6 or IPv4. 3. Do not add the constant PJSIP_TRANSPORT_IPV6 to existing transport types. This results in invalid values. Use a bitwise or instead. ASTERISK-26780 Patches: pjsip_resolver.c uploaded by Peter Sokolov (License #7070) Change-Id: I8b1e298f8efa682d0a7644113258fe76d9889c58	6 years ago
Roger James	29910aa451	res_pjsip_history.c: Fix to stop SIGSEGV when IPv6 addresses are encountered. Changed source and destination address fields in struct pjsip_history_entry so that they are long enough to hold an IPv6 address. ASTERISK-28854 Change-Id: Id65bb9aa961e9ecbcb500815e18170f774e34d3e	6 years ago
Guido Falsi	fa2b05a9e0	res_rtp_asterisk: Protect access to nochecksums with #ifdef Recently code accessing nochecksums variable has been added without including #ifdef SO_NO_CHECK protection, while the variable is created only when such constant is defined. ASTERISK-28852 #close Change-Id: I381718893b80599ab8635f2b594a10c1000d595e	6 years ago
Joshua C. Colp	fe11bc7a2a	stream: Enforce formats immutability and ensure formats exist. Some places in Asterisk did not treat the formats on a stream as immutable when they are. The ast_stream_get_formats function is now const to enforce this and parts of Asterisk have been updated to take this into account. Some violations of this were also fixed along the way. An additional minor tweak is that streams are now allocated with an empty format capabilities structure removing the need in various places to check that one is present on the stream. ASTERISK-28846 Change-Id: I32f29715330db4ff48edd6f1f359090458a9bfbe	6 years ago
sungtae kim	9a90ab3722	res_ari_channels: Fixed endpoint 80 characters limit Fixed it to copy the entire string from the requested endpoint body except tech-prefix. ASTERISK-28847 Change-Id: I91b5f6708a1200363f3267b847dd6a0915222c25	6 years ago
Joshua C. Colp	e7b6736ba3	fax: Fix crashes in PJSIP re-negotiation scenarios. This change fixes a few re-negotiation issues uncovered with fax. 1. The fax support uses its own mechanism for re-negotiation by conveying T.38 information in its own frames. The new support for re-negotiating when adding/removing/changing streams was also being triggered for this causing multiple re-INVITEs. The new support will no longer trigger when transitioning between fax. 2. In off-nominal re-negotiation cases it was possible for some state information to be left over and used by the next re-negotiation. This is now cleared. ASTERISK-28811 ASTERISK-28839 Change-Id: I8ed5924b53be9fe06a385c58817e5584b0f25cc2	6 years ago
DanielYK	139cd912af	res_pjsip: Fixed format of IPv6 addresses for external media addresses ASTERISK-28835 Change-Id: I66289afd164c5cdd6c5caa39e79d629a467e7a26	6 years ago
Alexander Traud	71c70dfbfb	res_pjsip: Sync load- and build-time deps. MODULEINFO is checked while buidling/linking the module. AST_MODULE_INFO is checked while loading/running the module. ASTERISK-28838 Change-Id: I4bb868532ca217fec1351885d99eb55c21b58042	6 years ago
Alexander Traud	c4c259c3af	res_pjsip_refer: Add build-time dependency. ASTERISK-28838 Change-Id: Ic693c3f464e35ec0db242afdb0a1415806af4e25	6 years ago

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5070 Commits (519bb01d5f94b34a5a088dd570725aee3c5c8da2)