asterisk

Commit Graph

Author	SHA1	Message	Date
Torrey Searle	83140c9fed	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	5 years ago
Sean Bright	4964302984	format_cap: Perform codec lookups by pointer instead of name ASTERISK-28416 #close Change-Id: I069420875ebdbcaada52d92599a5f7de3cb2cdf4	5 years ago
George Joseph	cc71be0078	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	5 years ago
George Joseph	ad4f2a8c99	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	5 years ago
George Joseph	d4f3b17dd3	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	5 years ago
Sungtae Kim	1fd12b88c7	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	5 years ago
Sungtae Kim	a0d41a27d4	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
Ben Ford	7eaae4e7b6	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	5 years ago
Asterisk Development Team	f589985840	Update CHANGES and UPGRADE.txt for 18.0.0	5 years ago
Patrick Verzele	5a49757e40	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
Kevin Harwell	ec03909831	conversions: Add string to signed integer conversion functions Change-Id: Id603b0b03b78eb84c7fca030a08b343c0d5973f9	5 years ago
Kfir Itzhak	c83e4821e5	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	5 years ago
George Joseph	e32815dddb	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	5 years ago
Joshua C. Colp	4f0766dcda	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
Alexander Traud	9ed1b1452d	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	5 years ago
Alexander Traud	217449a1e5	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	5 years ago
Kevin Harwell	31fbfc5e95	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	5 years ago
Joshua C. Colp	6d50d152d8	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
George Joseph	5a8cacb93d	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	5 years ago
Nickolay Shmyrev	0319e0b07f	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	5 years ago
Torrey Searle	addd295cda	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	5 years ago
Evandro César Arruda	36dd15c659	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	5 years ago
Sean Bright	b575868000	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	5 years ago
Joshua C. Colp	3c074038fe	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	5 years ago
Sean Bright	5ec7099312	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	5 years ago
Sean Bright	5dfeeba623	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	5 years ago
George Joseph	c4c72d55a2	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	5 years ago
George Joseph	d26ab7f8f9	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	5 years ago
Dennis Buteyn	9058d9e591	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
George Joseph	6faf76308d	ACN: Changes specific to the core Allow passing a topology from the called channel back to the calling channel. * Added a new function ast_queue_answer() that accepts a stream topology and queues an ANSWER CONTROL frame with it as the data. This allows the called channel to indicate its resolved topology. * Added a new virtual function to the channel tech structure answer_with_stream_topology() that allows the calling channel to receive the called channel's topology. Added ast_raw_answer_with_stream_topology() that invokes that virtual function. * Modified app_dial.c and features.c to grab the topology from the ANSWER frame queued by the answering channel and send it to the calling channel with ast_raw_answer_with_stream_topology(). * Modified frame.c to automatically cleanup the reference to the topology on ANSWER frames. Added a few debugging messages to stream.c. Change-Id: I0115d2ed68d6bae0f87e85abcf16c771bdaf992c	5 years ago
cmaj	543f936147	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
Sean Bright	57554c2834	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
Joshua C. Colp	a3d87f78ed	res_pjsip: Fix codec preference defaults. When reading in a codec preference configuration option the value would be set on the respective option before applying any default adjustments, resulting in the configuration not being as expected. This was exposed by the REST API push configuration as it used the configuration returned by Asterisk to then do a modification. In the case of codec preferences one of the options had a transcode value of "unspecified" when the defaults should have ensured it would be "allow" instead. This also renames the options in other places that were missed. Change-Id: I4ad42e74fdf181be2e17bc75901c62591d403964	5 years ago
Sean Bright	da8a617dc9	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	6482ab5bea	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	5 years ago
Ben Ford	769a9611e7	utils.c: NULL terminate ast_base64decode_string. With the addition of STIR/SHAKEN, the function ast_base64decode_string was added for convenience since there is a lot of converting done during the STIR/SHAKEN process. This function returned the decoded string for you, but did not NULL terminate it, causing some issues (specifically with MALLOC_DEBUG). Now, the returned string is NULL terminated, and the documentation has been updated to reflect this. Change-Id: Icdd7d05b323b0c47ff6ed43492937a03641bdcf5	5 years ago
George Joseph	802aa97fa0	ACN: Configuration renaming for pjsip endpoint This change renames the codec preference endpoint options. incoming_offer_codec_prefs becomes codec_prefs_incoming_offer to keep the options together when showing an endpoint. Change-Id: I6202965b4723777f22a83afcbbafcdafb1d11c8d	5 years ago
Ben Ford	de23cb4002	res_stir_shaken: Fix memory allocation error in curl.c Fixed a memory allocation that was not passing in the correct size for the struct in curl.c. Change-Id: I5fb92fbbe84b075fa6aefa2423786df80e114c3a (cherry picked from commit `deaa3742dc`)	5 years ago
George Joseph	71446b68fc	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	5 years ago
Joshua C. Colp	99eafe5771	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	5 years ago
Sean Bright	d9ae902f52	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	5 years ago
sungtae kim	2e32b56bdb	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	5 years ago
Sean Bright	9022f35f09	vector.h: Add AST_VECTOR_SORT() Allows a vector to be sorted in-place, rather than only during insertion. Change-Id: I22cba9ddf556a7e44dacc53c4431bd81dd2fa780	5 years ago
George Joseph	a678dafac8	CI: Force publishAsteriskDocs to use python2 Change-Id: I7d951e75ad2d472fa096647dfb55670b11105e23	5 years ago
Joshua C. Colp	af70bbb13a	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	5 years ago
Sean Bright	7a43bedd72	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	5 years ago
Joshua C. Colp	8d15f72721	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	5 years ago
George Joseph	3330764213	Update .gitreview defaultbranch to 18 Change-Id: Ib2c42fc2d46563e2fbadbd5513cb029b4042791e	5 years ago
Asterisk Development Team	1f5e6805bf	Update CHANGES and UPGRADE.txt for 18.0.0	5 years ago
Nickolay Shmyrev	e4d24f5137	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	5 years ago

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