4215 Commits (4c49c84deee820a4062eda2cfb3dfd357d405f79)

Author	SHA1	Message	Date
Rijnhard Hessel	f13eef719c	res_statsd: handle non-standard meter type safely ... Meter types are not well supported, lacking support in telegraf, datadog and the official statsd servers. We deprecate meters and provide a compliant fallback for any existing usages. A flag has been introduced to allow meters to fallback to counters. ASTERISK-29513 Change-Id: I5fcb385983a1b88f03696ff30a26b55c546a1dd7	5 years ago
Andre Barbosa	a47308ccb2	res_stasis_playback: Send PlaybackFinish event only once for errors ... When we try to play a list of sound files in the same Play command, we get only one PlaybackFinish event, after all sounds are played. But in the case where the Play fails (because channel is destroyed for example), Asterisk will send one PlaybackFinish event for each sound file still to be played. If the list is big, Asterisk is sending many events. This patch adds a failed state so we can understand that the play failed. On that case we don't send the event, if we still have a list of sounds to be played. When we reach the last sound, we send the PlaybackFinish with the failed state. ASTERISK-29464 #close Change-Id: I4c2e5921cc597702513af0d7c6c2c982e1798322	5 years ago
Joshua C. Colp	5382b9dbb8	core: Don't play silence for Busy() and Congestion() applications. ... When using the Busy() and Congestion() applications the function ast_safe_sleep is used by wait_for_hangup to safely wait on the channel. This function may send silence if Asterisk is configured to do so using the transmit_silence option. In a scenario where an answered channel dials a Local channel either directly or through call forwarding and the Busy() or Congestion() dialplan applications were executed with the transmit_silence option enabled the busy or congestion tone would not be heard. This is because inband generation of tones (such as busy and congestion) is stopped when other audio is sent to the channel they are being played to. In the given scenario the transmit_silence option would result in silence being sent to the channel, thus stopping the inband generation. This change adds a variant of ast_safe_sleep which can be used when silence should not be played to the channel. The wait_for_hangup function has been updated to use this resulting in the tones being generated as expected. ASTERISK-29485 Change-Id: I066bfc987a3ad6f0ccc88e0af4cd63f6a4729133	5 years ago
Ben Ford	12e8600849	STIR/SHAKEN: Add Date header, dest->tn, and URL checking. ... STIR/SHAKEN requires a Date header alongside the Identity header, so that has been added. Still on the outgoing side, we were missing the dest->tn section of the JSON payload, so that has been added as well. Moving to the incoming side, URL checking has been added to the public cert URL to ensure that it starts with http. https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021 Change-Id: Idee5b1b5e45bc3b483b3070e46ce322dca5b3f1c	5 years ago
George Joseph	9cc1d6fc22	res_pjsip_outbound_authenticator_digest: Be tolerant of RFC8760 UASs ... RFC7616 and RFC8760 allow more than one WWW-Authenticate or Proxy-Authenticate header per realm, each with different digest algorithms (including new ones like SHA-256 and SHA-512-256). Thankfully however a UAS can NOT send back multiple Authenticate headers for the same realm with the same digest algorithm. The UAS is also supposed to send the headers in order of preference with the first one being the most preferred. We're supposed to send an Authorization header for the first one we encounter for a realm that we can support. The UAS can also send multiple realms, especially when it's a proxy that has forked the request in which case the proxy will aggregate all of the Authenticate headers and then send them all back to the UAC. It doesn't stop there though... Each realm can require a different username from the others. There's also nothing preventing each digest algorithm from having a unique password although I'm not sure if that adds any benefit. So now... For each Authenticate header we encounter, we have to determine if we support the digest algorithm and, if not, just skip the header. We then have to find an auth object that matches the realm AND the digest algorithm or find a wildcard object that matches the digest algorithm. If we find one, we add it to the results vector and read the next Authenticate header. If the next header is for the same realm AND we already added an auth object for that realm, we skip the header. Otherwise we repeat the process for the next header. In the end, we'll have accumulated a list of credentials we can pass to pjproject that it can use to add Authentication headers to a request. NOTE: Neither we nor pjproject can currently handle digest algorithms other than MD5. We don't even have a place for it in the ast_sip_auth object. For this reason, we just skip processing any Authenticate header that's not MD5. When we support the others, we'll move the check into the loop that searches the objects. Changes: * Added a new API ast_sip_retrieve_auths_vector() that takes in a vector of auth ids (usually supplied on a call to ast_sip_create_request_with_auth()) and populates another vector with the actual objects. * Refactored res_pjsip_outbound_authenticator_digest to handle multiple Authenticate headers and set the stage for handling additional digest algorithms. * Added a pjproject patch that allows them to ignore digest algorithms they don't support. This patch has already been merged upstream. * Updated documentation for auth objects in the XML and in pjsip.conf.sample. * Although res_pjsip_authenticator_digest isn't affected by this change, some debugging and a testsuite AMI event was added to facilitate testing. Discovered during OpenSIPit 2021. ASTERISK-29397 Change-Id: I3aef5ce4fe1d27e48d61268520f284d15d650281	5 years ago
Naveen Albert	04454fc238	AMI: Add AMI event to expose hook flash events ... Although Asterisk can receive and propogate flash events, it currently provides no mechanism for doing anything with them itself. This AMI event allows flash events to be processed by Asterisk. Additionally, AST_CONTROL_FLASH is included in a switch statement in channel.c to avoid throwing a warning when we shouldn't. ASTERISK-29380 Change-Id: Ie17ffe65086e0282c88542e38eed6a461ec79e81	5 years ago
Ben Ford	0564d12280	STIR/SHAKEN: Switch to base64 URL encoding. ... STIR/SHAKEN encodes using base64 URL format. Currently, we just use base64. New functions have been added that convert to and from base64 encoding. The origid field should also be an UUID. This means there's no reason to have it as an option in stir_shaken.conf, as we can simply generate one when creating the Identity header. https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021 Change-Id: Icf094a2a54e87db91d6b12244c9f5ba4fc2e0b8c	5 years ago
Ben Ford	259ecfa289	STIR/SHAKEN: Fix certificate type and storage. ... During OpenSIPit, we found out that the public certificates must be of type X.509. When reading in public keys, we use the corresponding X.509 functions now. We also discovered that we needed a better naming scheme for the certificates since certificates with the same name would cause issues (overwriting certs, etc.). Now when we download a public certificate, we get the serial number from it and use that as the name of the cached certificate. The configuration option public_key_url in stir_shaken.conf has also been renamed to public_cert_url, which better describes what the option is for. https://wiki.asterisk.org/wiki/display/AST/OpenSIPit+2021 Change-Id: Ia00b20835f5f976e3603797f2f2fb19672d8114d	5 years ago
Sean Bright	4a843e00ef	res_pjsip.c: OPTIONS processing can now optionally skip authentication ... ASTERISK-27477 #close Change-Id: I68f6715bba92a525149e35d142a49377a34a1193	5 years ago
Kevin Harwell	eb92fb7298	time: Add timeval create and unit conversion functions ... Added a TIME_UNIT enumeration, and a function that converts a string to one of the enumerated values. Also, added functions that create and initialize a timeval object using a specified value, and unit type. Change-Id: Ic31a1c3262a44f77a5ef78bfc85dcf69a8d47392	5 years ago
Mark Murawski	b4347c4861	logger: Console sessions will now respect logger.conf dateformat= option ... The 'core' console (ie: asterisk -c) does read logger.conf and does use the dateformat= option. Whereas 'remote' consoles (ie: asterisk -r -T) does not read logger.conf and uses a hard coded dateformat option for printing received verbose messages: main/logger.c: static char dateformat[256] = "%b %e %T" This change will load logger.conf for each remote console session and use the dateformat= option to set the per-line timestamp for verbose messages Change-Id: I3ea10990dbd920e9f7ce8ff771bc65aa7f4ea8c1 ASTERISK-25358: #close Reported-by: Igor Liferenko	5 years ago
Jaco Kroon	8acb4fbd1e	app.h: Fix -Werror=zero-length-bounds compile errors in dev mode. ... Change-Id: I5c104dc1f8417ccd3d01faf86e84ccbf89bc3b31 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	5 years ago
Sean Bright	30e509c2f9	strings.h: ast_str_to_upper() and _to_lower() are not pure. ... Because they modify their argument they are not pure functions and should not be marked as such, otherwise the compiler may optimize them away. ASTERISK-29306 #close Change-Id: Ibec03a08522dd39e8a137ece9bc6a3059dfaad5f	5 years ago
Joshua C. Colp	304f8ddfb2	sorcery: Add support for more intelligent reloading. ... Some sorcery objects actually contain dynamic content that can change despite the underlying configuration itself not changing. A good example of this is the res_pjsip_endpoint_identifier_ip module which allows specifying hostnames. While the configuration may not change between reloads the DNS information of the hostnames can. This change adds the ability for a sorcery object to be marked as having dynamic contents which is then taken into account when reloading by the sorcery file based config module. If there is an object with dynamic content then a reload will be forced while if there are none then the existing behavior of not reloading occurs. ASTERISK-29321 Change-Id: I9342dc55be46cc00204533c266a68d972760a0b1	5 years ago
Jaco Kroon	725eca3bfa	app.h: Restore C++ compatibility for macro AST_DECLARE_APP_ARGS ... This partially reverts commit 3d1bf3c537, specifically for app.h. This works with both gcc 9.3.0 and 10.2.0 now, both for C and C++ (as tested with external modules). ASTERISK-29287 Change-Id: I5b9f02a9b290675682a1d13f1788fdda597c9fca Signed-off-by: Jaco Kroon <jaco@uls.co.za>	5 years ago
Sebastien Duthil	6e695c867f	app_mixmonitor: Add AMI events MixMonitorStart, -Stop and -Mute. ... ASTERISK-29244 Change-Id: I1862d58264c2c8b5d8983272cb29734b184d67c5	5 years ago
Ben Ford	00b229c69c	core_unreal: Fix T.38 faxing when using local channels. ... After some changes to streams and topologies, receiving fax through local channels stopped working. This change adds a stream topology with a stream of type IMAGE to the local channel pair and allows fax to be received. ASTERISK-29035 #close Change-Id: Id103cc5c9295295d8e68d5628e76220f8f17e9fb	5 years ago
Dan Cropp	55891227e8	chan_pjsip, app_transfer: Add TRANSFERSTATUSPROTOCOL variable ... When a Transfer/REFER is executed, TRANSFERSTATUSPROTOCOL variable is 0 when no protocl specific error SIP example of failure, 3xx-6xx for the SIP error code received This allows applications to perform actions based on the failure reason. ASTERISK-29252 #close Reported-by: Dan Cropp Change-Id: Ia6a94784b4925628af122409cdd733c9f29abfc4	5 years ago
Sean Bright	1b74555fcf	asterisk: Export additional manager functions ... Rename check_manager_enabled() and check_webmanager_enabled() to begin with ast_ so that the symbols are automatically exported by the linker. ASTERISK~29184 Change-Id: I85762b9a5d14500c15f6bad6507138c8858644c9	5 years ago
Alexander Traud	80c14f74bc	codecs: Remove test-law. ... This was dead code, test code introduced with Asterisk 13. This was found while analyzing ASTERISK_28416 and ASTERISK_29185. This change partly fixes, not closes those two issues. Change-Id: I42d0daa37f6f334c7d86672f06f085858a3f3940	5 years ago
lvl	b08427134f	Introduce astcachedir, to be used for temporary bucket files ... As described in the issue, /tmp is not a suitable location for a large amount of cached media files, since most distributions make /tmp a RAM-based tmpfs mount with limited capacity. I opted for a location that can be configured separately, as opposed to using a subdirectory of spooldir, given the different storage profile (transient files vs files that might stay there indefinitely). This commit just makes the cache directory configurable, and changes the default location from /tmp to /var/cache/asterisk. ASTERISK-29143 Change-Id: Ic54e95199405abacd9e509cef5f08fa14c510b5d	5 years ago
George Joseph	5a4640d208	pjsip_scheduler.c: Add type ONESHOT and enhance cli show command ... * Added a ONESHOT type that never reschedules. * Added "like" capability to "pjsip show scheduled_tasks" so you can do the following: CLI> pjsip show scheduled_tasks like outreg PJSIP Scheduled Tasks: Task Name Interval Times Run ... ============================================= ========= ========= ... pjsip/outreg/testtrunk-reg-0-00000074 50.000 oneshot ... pjsip/outreg/voipms-reg-0-00000073 110.000 oneshot ... * Fixed incorrect display of "Next Start". * Compacted the displays of times in the CLI. * Added two new functions (ast_sip_sched_task_get_times2, ast_sip_sched_task_get_times_by_name2) that retrieve the interval, next start time, and next run time in addition to the times already returned by ast_sip_sched_task_get_times(). Change-Id: Ie718ca9fd30490b8a167bedf6b0b06d619dc52f3	5 years ago
Alexei Gradinari	cc7eb72f65	sched: AST_SCHED_REPLACE_UNREF can lead to use after free of data ... The data can be freed if the old object '_data' is the same object as new 'data'. Because at first the object is unreferenced which can lead to destroying it. This could happened in res_pjsip_pubsub when the publication is updated which could lead to segfault in function publish_expire. Change-Id: I0164f57c387243510bdbd2f8dcf33377b6c202da	5 years ago
Alexander Traud	64d2de19ee	res_stir_shaken: Include OpenSSL headers where used actually. ... This avoids the inclusion of the OpenSSL headers in the public header, which avoids one external library dependency in res_pjsip_stir_shaken. Change-Id: I6a07e2d81d2b5442e24e99b8cc733a99f881dcf4	5 years ago
Kevin Harwell	b82f880647	AST-2020-001 - res_pjsip: Return dialog locked and referenced ... pjproject returns the dialog locked and with a reference. However, in Asterisk the method that handles this decrements the reference and removes the lock prior to returning. This makes it possible, under some circumstances, for another thread to free said dialog before the thread that created it attempts to use it again. Of course when the thread that created it tries to use a freed dialog a crash can occur. This patch makes it so Asterisk now returns the newly created dialog both locked, and with an added reference. This allows the caller to de-reference, and unlock the dialog when it is safe to do so. In the case of a new SIP Invite the lock, and reference are now held for the entirety of the new invite handling process. Otherwise it's possible for the dialog, or its dependent objects, like the transaction, to disappear. For example if there is a TCP transport error. ASTERISK-29057 #close Change-Id: I5ef645a47829596f402cf383dc02c629c618969e (cherry picked from commit 6baa4b53be)	5 years ago
Ben Ford	cd8f8b94f8	AST-2020-002 - res_pjsip: Stop sending INVITEs after challenge limit. ... If Asterisk sends out and INVITE and receives a challenge with a different nonce value each time, it will continually send out INVITEs, even if the call is hung up. The endpoint must be configured for outbound authentication in order for this to occur. A limit has been set on outbound INVITEs so that, once reached, Asterisk will stop sending INVITEs and the transaction will terminate. ASTERISK-29013 Change-Id: I2d001ca745b00ca8aa12030f2240cd72363b46f7	5 years ago
Kevin Harwell	56028426de	Logging: Add debug logging categories ... Added debug logging categories that allow a user to output debug information based on a specified category. This lets the user limit, and filter debug output to data relevant to a particular context, or topic. For instance the following categories are now available for debug logging purposes: dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet, stun, stun_packet These debug categories can be enable/disable via an Asterisk CLI command. While this overrides, and outputs debug data, core system debugging is not affected by this patch. Statements still output at their appropriate debug level. As well backwards compatibility has been maintained with past debug groups that could be enabled using the CLI (e.g. rtpdebug, stundebug, etc.). ASTERISK-29054 #close Change-Id: I6e6cb247bb1f01dbf34750b2cd98e5b5b41a1849	5 years ago
Sean Bright	51cba591e3	pbx.c: On error, ast_add_extension2_lockopt should always free 'data' ... In the event that the desired extension already exists, ast_add_extension2_lockopt() will free the 'data' it is passed before returning an error, so we should not be freeing it ourselves. Additionally, there were two places where ast_add_extension2_lockopt() could return an error without also freeing the 'data' pointer, so we add that. ASTERISK-29097 #close Change-Id: I904707aae55169feda050a5ed7c6793b53fe6eae	5 years ago
George Joseph	773f424c7f	app_confbridge/bridge_softmix: Add ability to force estimated bitrate ... app_confbridge now has the ability to set the estimated bitrate on an SFU bridge. To use it, set a bridge profile's remb_behavior to "force" and set remb_estimated_bitrate to a rate in bits per second. The remb_estimated_bitrate parameter is ignored if remb_behavior is something other than "force". Change-Id: Idce6464ff014a37ea3b82944452e56cc4d75ab0a	5 years ago
George Joseph	923d95cc84	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	5 years ago
Torrey Searle	888090ab18	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	30e08ce1bb	format_cap: Perform codec lookups by pointer instead of name ... ASTERISK-28416 #close Change-Id: I069420875ebdbcaada52d92599a5f7de3cb2cdf4	5 years ago
George Joseph	44bb0858cb	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	86f1bce186	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
Ben Ford	80a609fcce	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
Kevin Harwell	1a5597741f	conversions: Add string to signed integer conversion functions ... Change-Id: Id603b0b03b78eb84c7fca030a08b343c0d5973f9	5 years ago
Sean Bright	3553192900	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
George Joseph	64ca2d48da	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	118cb3f0dd	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
George Joseph	647c53c41f	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	6 years ago
Sean Bright	048b12b59d	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
Ben Ford	9ed6387c14	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	6 years ago
George Joseph	a15e64aaf5	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	6 years ago
Sean Bright	7d96b3e437	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	b5bb4a7a0d	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	9c3b57822a	Prepare master for the next Asterisk version ... * Updated AMI version to 8.0.0 * Updated ARI version to 7.0.0 * Update make_ari_stubs.py to "Asterisk 19" Change-Id: I51fb38c2e29f2db785f64a8bbd5565d56bea5af5	6 years ago
Ben Ford	5fbed5af24	res_stir_shaken: Add stir_shaken option and general improvements. ... Added a new configuration option for PJSIP endpoints - stir_shaken. If set to yes, then STIR/SHAKEN support will be added to inbound and outbound INVITEs. The default is no. Alembic has been updated to include this option. Previously the dialplan function was not trimming the whitespace from the parameters it recieved. Now it does. Also added a conditional that, when TEST_FRAMEWORK is enabled, the timestamp in the identity header will be overlooked. This is just for testing, since the testsuite will rely on a SIPp scenario with a preset identity header to trigger the MISMATCH result. Change-Id: I43d67f1489b8c1c5729ed3ca8d71e35ddf438df1	6 years ago
George Joseph	9bd1d686a1	ACN: Add tracing to existing code ... Prior to making any modifications to the pjsip infrastructure for ACN, I've added the tracing functions to the existing code. This should make the final commit easier to review, but we can also now run a "before and after" trace. No functional changes were made with this commit. Change-Id: Ia83a1a2687ccb96f2bc8a2a3928a5214c4be775c	6 years ago
George Joseph	2d22e34206	ACN: res_pjsip endpoint options ... This commit adds the endpoint options required to control Advanced Codec Negotiation. incoming_offer_codec_prefs outgoing_offer_codec_prefs incoming_answer_codec_prefs outgoing_answer_codec_prefs The documentation may need tweaking and some additional edits added, especially for the "answer" prefs. That'll be handled when things finalize. This commit is safe to merge as it doens't alter any existing functionality nor does it alter the previous codec negotiation work which may now be obsolete. Change-Id: I920ba925d7dd36430dfd2ebd9d82d23f123d0e11	6 years ago
George Joseph	d093e44b1e	frame.c: Make debugging easier ... * ast_frame_subclass2str() and ast_frame_type2str() now return a pointer to the buffer that was passed in instead of void. This makes it easier to use these functions inline in printf-style debugging statements. * Added many missing control frame entries in ast_frame_subclass2str. Change-Id: Ifd0d6578e758cd644c96d17a5383ff2128c572fc	6 years ago