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Author	SHA1	Message	Date
Joshua C. Colp	baa6e8f112	res_pjsip_session: Fix stream name memory leak. ... When constructing a stream name based on the media type and position the allocated name was not being freed causing a leak. Change-Id: I52510863b24a2f531f0a55b440bb2c81844029de	6 years ago
Sean Bright	799426cd58	func_curl.c: Prevent crash when using CURLOPT(httpheader) ... Because we use shared thread-local cURL instances, we need to ensure that the state of the cURL instance is correct before each invocation. In the case of custom headers, we were not resetting cURL's internal HTTP header pointer which could result in a crash if subsequent requests do not configure custom headers. ASTERISK-29085 #close Change-Id: I8b4ab34038156dfba613030a45f10e932d2e992d	6 years ago
Sean Bright	4a7bbac0ed	res_musiconhold: Start playlist after initial announcement ... Only track our sample offset if we are playing a non-announcement file, otherwise we will skip that number of samples when we start playing the first MoH file. ASTERISK-24329 #close Change-Id: Ib6b3c84fcaa1063889ab38ba7e7fc50050a3ccfc	6 years ago
Joshua C. Colp	e6ed74347c	res_pjsip_session: Fix session reference leak. ... The ast_sip_dialog_get_session function returns the session with reference count increased. This was not taken into account and was causing sessions to remain around when they should not be. ASTERISK-29089 Change-Id: I430fa721b0a824311a59effec6056e9ec528e3e8	6 years ago
Michal Hajek	f7285140b4	res_stasis.c: Add compare function for bridges moh container ... Sometimes not play MOH on bridge. ASTERISK-29081 Reported-by: Michal Hajek <michal.hajek@daktela.com> Change-Id: I760c73e0c9be1d340303b5d1c18a00c4759e8232	6 years ago
George Joseph	fb6f2157e7	logger.h: Fix ast_trace to respect scope_level ... ast_trace() was always emitting messages when it's level was set to -1 because it was ignoring scope_level. Change-Id: I849c8f4f4613899c37f82be0202024e7d117e506	6 years ago
George Joseph	8d9633074e	bridge_softmix/sfu_topologies_on_join: Ignore topology change failures ... When a channel joins a bridge, we do topology change requests on all existing channels to add the new participant to them. However the announcer channel will return an error because it doesn't support topology in the first place. Unfortunately, there doesn't seem to be a reliable way to tell if the error is expected or not so the error is ignored for all channels. If the request fails on a "real" channel, that channel just won't get the new participant's video. Change-Id: Ic95db4683f27d224c1869fe887795d6b9fdea4f0	6 years ago
Sean Bright	9458577f68	res_pjsip_session.c: Fix build when TEST_FRAMEWORK is not defined ... Change-Id: Id4852c26e9c412af8e37b5dd3c15da9453ad3276	6 years ago
Torrey Searle	5a12463c07	res_pjsip_diversion: implement support for History-Info ... Implemention of History-Info capable of interworking with Diversion Header following RFC7544 ASTERISK-29027 #close Change-Id: I2296369582d4b295c5ea1e60bec391dd1d318fa6	6 years ago
Sean Bright	c9cc281484	format_cap: Perform codec lookups by pointer instead of name ... ASTERISK-28416 #close Change-Id: I069420875ebdbcaada52d92599a5f7de3cb2cdf4	6 years ago
George Joseph	df429c97a1	res_pjsip_session: Fix issue with COLP and 491 ... The recent 491 changes introduced a check to determine if the active and pending topologies were equal and to suppress the re-invite if they were. When a re-invite is sent for a COLP-only change, the pending topology is NULL so that check doesn't happen and the re-invite is correctly sent. Of course, sending the re-invite sets the pending topology. If a 491 is received, when we resend the re-invite, the pending topology is set and since we didn't request a change to the topology in the first place, pending and active topologies are equal so the topologies-equal check causes the re-invite to be erroneously suppressed. This change checks if the topologies are equal before we run the media state resolver (which recreates the pending topology) so that when we do the final topologies-equal check we know if this was a topology change request. If it wasn't a change request, we don't suppress the re-invite even though the topologies are equal. ASTERISK-29014 Change-Id: Iffd7dd0500301156a566119ebde528d1a9573314	6 years ago
George Joseph	6abf6f345d	debugging: Add enough to choke a mule ... Added to: * bridges/bridge_softmix.c * channels/chan_pjsip.c * include/asterisk/res_pjsip_session.h * main/channel.c * res/res_pjsip_session.c There NO functional changes in this commit. Change-Id: I06af034d1ff3ea1feb56596fd7bd6d7939dfdcc3	6 years ago
George Joseph	65088494cb	res_pjsip_session: Handle multi-stream re-invites better ... When both Asterisk and a UA send re-invites at the same time, both send 491 "Transaction in progress" responses to each other and back off a specified amount of time before retrying. When Asterisk prepares to send its re-invite, it sets up the session's pending media state with the new topology it wants, then sends the re-invite. Unfortunately, when it received the re-invite from the UA, it partially processed the media in the re-invite and reset the pending media state before sending the 491 losing the state it set in its own re-invite. Asterisk also was not tracking re-invites received while an existing re-invite was queued resulting in sending stale SDP with missing or duplicated streams, or no re-invite at all because we erroneously determined that a re-invite wasn't needed. There was also an issue in bridge_softmix where we were using a stream from the wrong topology to determine if a stream was added. This also caused us to erroneously determine that a re-invite wasn't needed. Regardless of how the delayed re-invite was triggered, we need to reconcile the topology that was active at the time the delayed request was queued, the pending topology of the queued request, and the topology currently active on the session. To do this we need a topology resolver AND we need to make stream named unique so we can accurately tell what a stream has been added or removed and if we can re-use a slot in the topology. Summary of changes: * bridge_softmix: * We no longer reset the stream name to "removed" in remove_all_original_streams(). That was causing multiple streams to have the same name and wrecked the checks for duplicate streams. * softmix_bridge_stream_sources_update() was checking the old_stream to see if it had the softmix prefix and not considering the stream as "new" if it did. If the stream in that slot has something in it because another re-invite happened, then that slot in old might have a softmix stream but the same stream in new might actually be a new one. Now we check the new_stream's name instead of the old_stream's. * stream: * Instead of using plain media type name ("audio", "video", etc) as the default stream name, we now append the stream position to it to make it unique. We need to do this so we can distinguish multiple streams of the same type from each other. * When we set a stream's state to REMOVED, we no longer reset its name to "removed" or destroy its metadata. Again, we need to do this so we can distinguish multiple streams of the same type from each other. * res_pjsip_session: * Added resolve_refresh_media_states() that takes in 3 media states and creates an up-to-date pending media state that includes the changes that might have happened while a delayed session refresh was in the delayed queue. * Added is_media_state_valid() that checks the consistency of a media state and returns a true/false value. A valid state has: * The same number of stream entries as media session entries. Some media session entries can be NULL however. * No duplicate streams. * A valid stream for each non-NULL media session. * A stream that matches each media session's stream_num and media type. * Updated handle_incoming_sdp() to set the stream name to include the stream position number in the name to make it unique. * Updated the ast_sip_session_delayed_request structure to include both the pending and active media states and updated the associated delay functions to process them. * Updated sip_session_refresh() to accept both the pending and active media states that were in effect when the request was originally queued and to pass them on should the request need to be delayed again. * Updated sip_session_refresh() to call resolve_refresh_media_states() and substitute its results for the pending state passed in. * Updated sip_session_refresh() with additional debugging. * Updated session_reinvite_on_rx_request() to simply return PJ_FALSE to pjproject if a transaction is in progress. This stops us from creating a partial pending media state that would be invalid later on. * Updated reschedule_reinvite() to clone both the current pending and active media states and pass them to delay_request() so the resolver can tell what the original intention of the re-invite was. * Added a large unit test for the resolver. ASTERISK-29014 Change-Id: Id3440972943c611a15f652c6c569fa0e4536bfcb	6 years ago
Sungtae Kim	af339d0adb	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	6 years ago
Sungtae Kim	a353b76c75	realtime: Increased reg_server character size ... Currently, the ps_contacts table's reg_server column in realtime database type is varchar(20). This is fine for normal cases, but if the hostname is longer than 20, it returns error and then failed to register the contact address of the peer. Normally, 20 characters limitation for the hostname is fine, but with the cloud env. So, increased the size to 255. ASTERISK-29056 Change-Id: Iac52c8c35030303cfa551bb39f410b33bffc507d	6 years ago
Ben Ford	e7620d034a	Bridging: Use a ref to bridge_channel's channel to prevent crash. ... There's a race condition with bridging where a bridge can be torn down causing the bridge_channel's ast_channel to become NULL when it's still needed. This particular case happened with attended transfers, but the crash occurred when trying to publish a stasis message. Now, the bridge_channel is locked, a ref to the ast_channel is obtained, and that ref is passed down the chain. Change-Id: Ic48715c0c041615d17d286790ae3e8c61bb28814	6 years ago
Asterisk Development Team	bd0724c7ed	Update CHANGES and UPGRADE.txt for 16.14.0	6 years ago
Patrick Verzele	ab34417f7e	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	6 years ago
Kevin Harwell	7b441505dc	conversions: Add string to signed integer conversion functions ... Change-Id: Id603b0b03b78eb84c7fca030a08b343c0d5973f9	6 years ago
Kfir Itzhak	ae718c975a	app_queue: Fix leave-empty not recording a call as abandoned ... This fixes a bug introduced mistakenly in ASTERISK-25665: If leave-empty is enabled, a call may sometimes be removed from a queue without recording it as abandoned. This causes Asterisk to not generate an abandon event for that call, and for the queue abandoned counter to be incorrect. ASTERISK-29043 #close Change-Id: I1a71b81df78adff59af587f1d8483cf57df430c7	6 years ago
George Joseph	a14c5301e7	ast_coredumper: Fix issues with naming ... If you run ast_coredumper --tarball-coredumps in the same directory as the actual coredump, tar can fail because the link to the actual coredump becomes recursive. The resulting tarball will have everything _except_ the coredump (which is usually what you need) There's also an issue that the directory name in the tarball is the same as the coredump so if you extract the tarball the directory it creates will overwrite the coredump. So: * Made the link to the coredump use the absolute path to the file instead of a relative one. This prevents the recursive link and allows tar to add the coredump. * The tarballed directory is now named <coredump>.output instead of just <coredump> so if you expand the tarball it won't overwrite the coredump. Change-Id: I8b3eeb26e09a577c702ff966924bb0a2f9a759ea	6 years ago
Joshua C. Colp	075a7ee4a6	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	53a88df076	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
Alexander Traud	afc5cf8d67	samples: Fix keep_alive_interval default in pjsip.conf. ... Since ASTERISK_27978 the default is not off but 90 seconds. That change happened because ASTERISK_27347 disabled the keep-alives in the bundled PJProject and Asterisk should behave the same as before. Change-Id: Ie63dc558ade6a5a2b969c30a4bd492d63730dc46	6 years ago
Alexander Traud	0fa20c8df6	sip_nat_settings: Update script for latest Linux. ... With the latest Linux, 'ifconfig' is not installed on default anymore. Furthermore, the output of the current net-tools 'ifconfig' changed. Therefore, parsing failed. This update uses 'ip addr show' instead. Finally, the service for the external IP changed. Change-Id: I9b1a7c3f457e3553b50a3e9a55524e40d70245a0	6 years ago
Kevin Harwell	17258e0cdf	chan_pjsip: disallow PJSIP_SEND_SESSION_REFRESH pre-answer execution ... This patch makes it so if the PJSIP_SEND_SESSION_REFRESH dialplan function is called on a channel prior to answering a warning is issued and the function returns unsuccessful. ASTERISK-28878 #close Change-Id: I053f767d10cf3b2b898fa9e3e7c35ff07e23c9bb	6 years ago
Joshua C. Colp	0452717645	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
George Joseph	d28a44c33d	logger.c: Added a new log formatter called "plain" ... Added a new log formatter called "plain" that always prints file, function and line number if available (even for verbose messages) and never prints color control characters. It also doesn't apply any special formatting for verbose messages. Most suitable for file output but can be used for other channels as well. You use it in logger.conf like so: debug => [plain]debug console => [plain]error,warning,debug,notice,pjsip_history messages => [plain]warning,error,verbose Change-Id: I4fdfe4089f66ce2f9cb29f3005522090dbb5243d	6 years ago
Nickolay Shmyrev	444c606161	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	75756ab850	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
Evandro César Arruda	aa93107444	app_queue: Member lastpause time reseting ... This fixes the reseting members lastpause problem when realtime members is being used, the function rt_handle_member_record was forcing the reset members lastpause because it does not exist in realtime ASTERISK-29034 #close Change-Id: Ic9107e4456732a1f78412a32adb2ef87f5da40b5	6 years ago
Sean Bright	788b60d935	app_voicemail: Process urgent messages with mailcmd ... Rather than putting messages into INBOX and then moving them to Urgent later, put them directly in to the Urgent folder. This prevents mailcmd from being skipped. ASTERISK-27273 #close Change-Id: I49934e093290d308506ab8d45a40ef705c5ae4f5	6 years ago
Joshua C. Colp	eaed23b340	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	32b593c325	bridge_channel: Ensure text messages are zero terminated ... T.140 data in RTP is not zero terminated, so when we are queuing a text frame on a bridge we need to ensure that we are passing a zero terminated string. ASTERISK-28974 #close Change-Id: Ic10057387ce30b2094613ea67e3ae8c5c431dda3	6 years ago
Sean Bright	55358b1276	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	b9ba0f8da8	scope_trace: Added debug messages and added additional macros ... The SCOPE_ENTER and SCOPE_EXIT* macros now print debug messages at the same level as the scope level. This allows the same messages to be printed to the debug log when AST_DEVMODE isn't enabled. Also added a few variants of the SCOPE_EXIT macros that will also call ast_log instead of ast_debug to make it easier to use scope tracing and still print error messages. Change-Id: I7fe55f7ec28069919a0fc0b11a82235ce904cc21	6 years ago
George Joseph	377caaed3c	stream.c: Added 2 more debugging utils and added pos to stream string ... * Added ast_stream_to_stra and ast_stream_topology_to_stra() macros which are shortcuts for ast_str_tmp(256, ast_stream_to_str(stream, &STR_TMP)) * Added the stream position to the string representation of the stream. * Fixed some formatting in ast_stream_to_str(). Change-Id: Idaf4cb0affa46d4dce58a73a111f35435331cc4b	6 years ago
Dennis Buteyn	9c6d14b26b	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
cmaj	5609d008da	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	6 years ago
George Joseph	16afb0a05d	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	cd8e011670	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	abad395098	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	1e58da7814	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	8c54be8fc9	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	2196511121	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	e9e441c399	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
sungtae kim	15a3318f1f	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	7892bdec53	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	e91e2ec0da	CI: Force publishAsteriskDocs to use python2 ... Change-Id: I7d951e75ad2d472fa096647dfb55670b11105e23	6 years ago
Joshua C. Colp	e8deea38bd	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