32932 Commits (c5c5171ec8b0b81d4699a86b6aedf785e8045b32)

 

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Author	SHA1	Message	Date
Ben Ford	464ecc1d82	logger.conf.sample: Add more debug documentation. ... Change-Id: Iff0e713f2120d8dce8e1e26924b99ed17f9d9dff	5 years ago
Sean Bright	cd090efde6	queues.conf.sample: Correct 'context' documentation. ... ASTERISK-24631 #close Change-Id: I8bf8776906a72ee02f24de6a85345940b9ff6b6f	5 years ago
Sean Bright	7215895590	app_queue.c: Remove dead 'updatecdr' code. ... Also removed the sample documentation, and some oddly-placed documentation about the timeout argument to the Queue() application itself. There is a large section on the timeout behavior below. ASTERISK-26614 #close Change-Id: I8f84e8304b50305b7c4cba2d9787a5d77c3a6217	5 years ago
Mark Murawski	98d149b4ce	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
Sean Bright	9e19d98f45	app_queue.c: Don't crash when realtime queue name is empty. ... ASTERISK-27542 #close Change-Id: If0b9719380a25533d2aed1053cff845dc3a4854a	5 years ago
George Joseph	3c8a7597b3	res_pjsip_session: Make reschedule_reinvite check for NULL topologies ... When the check for equal topologies was added to reschedule_reinvite() it was assumed that both the pending and active media states would actually have non-NULL topologies. We since discovered this isn't the case. We now only test for equal topologies if both media states have non-NULL topologies. The logic had to be rearranged a bit to make sure that we cloned the media states if their topologies were non-NULL but weren't equal. ASTERISK-29215 Change-Id: I61313cca7fc571144338aac826091791b87b6e17	5 years ago
Joshua C. Colp	a8042f2a4c	app_queue: Only send QueueMemberStatus if status changes. ... If a queue member was updated with the same status multiple times each time a QueueMemberStatus event would be sent which would be a duplicate of the previous. This change makes it so that the QueueMemberStatus event is only sent if the status actually changes. ASTERISK-29355 Change-Id: I580c60d992a0a8f2bea8b91c868771b3b490d116	5 years ago
Joshua C. Colp	73a52c53ce	core_unreal: Fix deadlock with T.38 control frames. ... When using the ast_unreal_lock_all function no channel locks can be held before calling it. This change unlocks the channel that indicate was called on before doing so and then relocks it afterwards. ASTERISK-29035 Change-Id: Id65016201b5f9c9519a216e250f9101c629e19e9	5 years ago
Joshua C. Colp	720a39eb9e	res_pjsip: Add support for partial transport reload. ... Some configuration items for a transport do not result in the underlying transport changing, but instead are just state we keep ourselves and use. It is perfectly reasonable to change these items. These include local_net and external_* information. ASTERISK-29354 Change-Id: I027857ccfe4419f460243e562b5f098434b3d43a	5 years ago
Jaco Kroon	eff26ffe42	menuselect: exit non-zero in case of failure on --enable|disable options. ... ASTERISK-29348 Change-Id: I77e3466435f5a51a57538b29addb68d811af238d Signed-off-by: Jaco Kroon <jaco@uls.co.za>	5 years ago
Joshua C. Colp	7f29dd0a53	res_rtp_asterisk: Force resync on SSRC change. ... When an SSRC change occurs the timestamps are likely to change as well. As a result we need to reset the timestamp mapping done in the calc_rxstamp function so that they map properly from timestamp to real time. This previously occurred but due to packet retransmission support the explicit setting of the marker bit was not effective. ASTERISK-29352 Change-Id: I2d4c8f93ea24abc1030196706de2d70facf05a5a	5 years ago
Joshua C. Colp	0ae922e48b	menuselect: Add ability to set deprecated and removed versions. ... The "deprecated_in" and "removed_in" information can now be set in MODULEINFO for a module and is then displayed in menuselect so users can be aware of when a module is slated to be deprecated and then removed. ASTERISK-29337 Change-Id: I6952889cf08e0e9e99cf8b43f99b3cef4688087a	5 years ago
Joshua C. Colp	ea3d4e0343	xml: Allow deprecated_in and removed_in for MODULEINFO. ... ASTERISK-29337 Change-Id: I2211b7da8d29369f8649aeabce07679da0787f2b	5 years ago
Joshua C. Colp	e42417e910	xml: Embed module information into core XML documentation. ... This change embeds the MODULEINFO block of modules into the core XML documentation. This provides a shared mechanism for use by both menuselect and Asterisk for information and a definitive source of truth. ASTERISK-29335 Change-Id: Ifbfd5c700049cf320a3e45351ac65dd89bc99d90	5 years ago
Joshua C. Colp	ee8c9017a7	documentation: Fix non-matching module support levels. ... Some modules have a different support level documented in their MODULEINFO XML and Asterisk module definition. This change brings the two in sync for the modules which were not matching. ASTERISK-29336 Change-Id: If2f819103d4a271e2e0624ef4db365e897fa3d35	5 years ago
Kevin Harwell	ec8b030bd7	manager: Increase the non breaking AMI version number ... ASTERISK~29244 added three new AMI events, so bump the version number. Change-Id: I0e77fa36d38fb27dec3481d4ef08131330da0632	5 years ago
Asterisk Development Team	fb17a41641	Update CHANGES and UPGRADE.txt for 16.17.0	5 years ago
Joshua C. Colp	ceb8404667	channel: Fix crash in suppress API. ... There exists an inconsistency with framehook usage such that it is only on reads that the frame should be freed, not on writes as well. ASTERISK-29071 Change-Id: I5ef918ebe4debac8a469e8d43bf9d6b673e8e472	5 years ago
Jaco Kroon	59129bfab8	func_callerid+res_agi: Fix compile errors related to -Werror=zero-length-bounds ... Change-Id: I75152cece8a00b7523d542e5ac22796f9595692b Signed-off-by: Jaco Kroon <jaco@uls.co.za>	5 years ago
Jaco Kroon	fce875cf46	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	dcfcf7d70a	app_dial.c: Only send DTMF on first progress event. ... ASTERISK-29329 #close Change-Id: Ic58e7a17f1ff3f785a5b21dced88682581149601	5 years ago
Alexander Traud	f875175815	res_format_attr_*: Parameter Names are Case-Insensitive. ... see RFC 4855: parameter names are case-insensitive both in media type strings and in the default mapping to the SDP a=fmtp attribute. This change is required for H.263+ because some implementations are known to use even mixed-case. This does not fix ASTERISK~29268 because H.264 was not fixed. This approach here lowers/uppers both parameter names and parameter values. H.264 needs a different approach because one of its parameter values is not case-insensitive: sprop-parameter-sets is Base64. Change-Id: Idf2a73457be231647aed3c87b1da197afba86892	5 years ago
Alexander Traud	10c0d9e711	chan_iax2: System Header strings is included via asterisk.h/compat.h. ... The system header strings was included mistakenly with commit 3de0204. That header is included via asterisk.h and there via the compat.h. Change-Id: I3dc49060e275295f785670c87cc65fd3c3abd24a	5 years ago
Sean Bright	526917bf61	modules.conf: Fix differing usage of assignment operators. ... ASTERISK-24434 #close Change-Id: I0144e8d65d878128da59dcf3df12ca8cee47d6db	5 years ago
Sean Bright	08b94d5837	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
Sean Bright	1fc4ab38b8	res_musiconhold.c: Plug ref leak caused by ao2_replace() misuse. ... ao2_replace() bumps the reference count of the object that is doing the replacing, which is not what we want. We just want to drop the old ref on the old object and update the pointer to point to the new object. Pointed out by George Joseph in #asterisk-dev Change-Id: Ie8167ed3d4b52b9d1ea2d785f885e8c27206743d	5 years ago
Torrey Searle	b90ecc6b31	res/res_rtp_asterisk: generate new SSRC on native bridge end ... For RTCP to work, we update the ssrc to be the one corresponding to the native bridge while active. However when the bridge ends we should generate a new SSRC as the sequence numbers will not continue from the native bridge left off. ASTERISK-29300 #close Change-Id: I23334b6934d2bf6490bda4bbf6414d96b8d17d10	5 years ago
Joshua C. Colp	11b53aecc8	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
George Joseph	90a456e6d2	res_pjsip_refer: Move the progress dlg release to a serializer ... Although the dlg session count was incremented in a pjsip servant thread, there's no guarantee that the last thread to unref this progress object was one. Before we decrement, we need to make sure that this is either a servant thread or that we push the decrement to a serializer that is one. Because pjsip_dlg_dec_session requires the dialog lock, we don't want to wait on the task to complete if we had to push it to a serializer. Change-Id: I8ff2d5d94be3ff04298394070434e22a7d3cbc41	5 years ago
Joshua C. Colp	13b1ee8aa5	res_pjsip_registrar: Include source IP and port in log messages. ... When registering it can be useful to see the source IP address and port in cases where multiple devices are using the same endpoint or when anonymous is in use. ASTERISK-29325 Change-Id: Ie178a6f55f53f8473035854c411bc3d056e0a2e0	5 years ago
Joshua C. Colp	4083aa6546	asterisk: Update copyright. ... ASTERISK-29326 Change-Id: Ia95dbfb66e2d11ac4d1228444283bb2e4d77396a	5 years ago
Ben Ford	b409fa1fff	AST-2021-006 - res_pjsip_t38.c: Check for session_media on reinvite. ... When Asterisk sends a reinvite negotiating T38 faxing, it's possible a crash can occur if the response contains a m=image and zero port. The reinvite callback code now checks session_media to see if it is null or not before trying to access the udptl variable on it. ASTERISK-29305 Change-Id: I1dfc51c5fa586e38579ede4bc228edee213ccaa9	5 years ago
Alexander Traud	fcf1bec363	res_format_attr_h263: Generate valid SDP fmtp for H.263+. ... Fixed: * RFC 4629 does not allow the value "0" for MPI, K, and N. * Allow value "0" for PAR. * BPP is printed only when specified because "0" has a meaning. New: * Added CPCF and MaxBR. * Some implementations provide CIF without MPI: a=fmtp:xx CIF;F=1 Although a violation of RFC 3555 section 3, we can support that. Changed: * Resorts the CIFs from large to small which partly fixes ASTERISK~29267. Change-Id: I95a650c715007b8dde11a77cb37d9c6c123a441e	5 years ago
Joshua C. Colp	d65041ebea	res_pjsip_nat: Don't rewrite Contact on REGISTER responses. ... When sending a SIP response to an incoming REGISTER request we don't want to change the Contact header as it will contain the Contacts registered to the AOR and not our own Contact URI. ASTERISK-29235 Change-Id: I35a0723545281dd01fcd5cae497baab58720478c	5 years ago
Joshua C. Colp	c81c4f3ae2	channel: Fix memory leak in suppress API. ... A frame suppression API exists as part of channels which allows audio frames to or from a channel to be dropped. The MuteAudio AMI action uses this API to perform its job. This API uses a framehook to intercept flowing audio and drop it when appropriate. It is the responsibility of the framehook to free the frame it is given if it changes the frame. The suppression API failed to do this resulting in a leak of audio frames. This change adds the freeing of these frames. ASTERISK-29071 Change-Id: Ie50acd454d672d36af914050c327d2e120d8ba7b	5 years ago
Salah Ahmed	6c065088be	res_rtp_asterisk: Check remote ICE reset and reset local ice attrb ... This change will check is the remote ICE session got reset or not by checking the offered ufrag and password with session. If the remote ICE reset session then Asterisk reset its local ufrag and password to reject binding request with Old ufrag and Password. ASTERISK-29266 Change-Id: I9c55e79a7af98a8fbb497d336b828ba41bc34eeb	5 years ago
Holger Hans Peter Freyther	4ceb07434c	pjsip: Generate progress (once) when receiving a 180 with a SDP ... ASTERISK-29105 Change-Id: If1615fe7115fe544ef974b044d3cea5c48b94a38	5 years ago
Nico Kooijman	980cc0d364	main: With Dutch language year after 2020 is not spoken in say.c ... Implemented the english way of saying the year in ast_say_date_with_format_nl. Currently the numbers are spoken correctly until 2020 and stopped working this year. ASTERISK-29297 #close Reported-by: Jacek Konieczny Change-Id: If5918eed5ab05df31df4dd23f08a909a60f6aba4	5 years ago
Alexei Gradinari	d7eaf44501	res_fax: validate the remote/local Station ID for UTF-8 format ... If the remote Station ID contains invalid UTF-8 characters the asterisk fails to publish the Stasis and ReceiveFax status messages. json.c: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string. 0: /usr/sbin/asterisk(ast_json_vpack+0x98) [0x4f3f28] 1: /usr/sbin/asterisk(ast_json_pack+0x8c) [0x4f3fcc] 2: /usr/sbin/asterisk(ast_channel_publish_varset+0x2b) [0x57aa0b] 3: /usr/sbin/asterisk(pbx_builtin_setvar_helper+0x121) [0x530641] 4: /usr/lib64/asterisk/modules/res_fax.so(+0x44fe) [0x7f27f4bff4fe] ... stasis_channels.c: Error creating message json.c: Error building JSON from '{s: s, s: s, s: s, s: s, s: s, s: s, s: o}': Invalid UTF-8 string. 0: /usr/sbin/asterisk(ast_json_vpack+0x98) [0x4f3f28] 1: /usr/sbin/asterisk(ast_json_pack+0x8c) [0x4f3fcc] 2: /usr/lib64/asterisk/modules/res_fax.so(+0x5acd) [0x7f27f4c00acd] ... res_fax.c: Error publishing ReceiveFax status message This patch replaces the invalid UTF-8 Station IDs with an empty string. ASTERISK-29312 #close Change-Id: Ieb00b6ecf67db3bfca787649caa8517f29d987db	5 years ago
Sean Bright	8604867857	app_page.c: Don't fail to Page if beep sound file is missing ... ASTERISK-16799 #close Change-Id: I40367b0d6dbf66a39721bde060c8b2d734a61cf4	5 years ago
George Joseph	1c1d4b2c68	res_pjsip_refer: Refactor progress locking and serialization ... Although refer_progress_notify() always runs in the progress serializer, the pjproject evsub module itself can cause the subscription to be destroyed which then triggers refer_progress_on_evsub_state() to clean it up. In this case, it's possible that refer_progress_notify() could get the subscription pulled out from under it while it's trying to use it. At one point we tried to have refer_progress_on_evsub_state() push the cleanup to the serializer and wait for its return before returning to pjproject but since pjproject calls its state callbacks with the dialog locked, this required us to unlock the dialog while waiting for the serialized cleanup, then lock it again before returning to pjproject. There were also still some cases where other callers of refer_progress_notify() weren't using the serializer and crashes were resulting. Although all callers of refer_progress_notify() now use the progress serializer, we decided to simplify the locking so we didn't have to unlock and relock the dialog in refer_progress_on_evsub_state(). Now, refer_progress_notify() holds the dialog lock for its duration and since pjproject also holds the dialog lock while calling refer_progress_on_evsub_state() (which does the cleanup), there should be no more chances for the subscription to be cleaned up while still being used to send NOTIFYs. To be extra safe, we also now increment the session count on the dialog when we create a progress object and decrement the count when the progress is destroyed. ASTERISK-29313 Change-Id: I97a8bb01771a3c85345649b8124507f7622a8480	5 years ago
Kevin Harwell	11fa7f6e0c	res_rtp_asterisk: Add packet subtype during RTCP debug when relevant ... For some RTCP packet types the report count is actually the packet's subtype. This was not being reflected in the packet debug output. This patch makes it so for some RTCP packet types a "Packet Subtype" is now output in the debug replacing the "Reception reports" (i.e count). Change-Id: Id4f4b77bb37077a4c4f039abd6a069287bfefcb8	5 years ago
Joshua C. Colp	7acb6e2194	res_pjsip_session: Always produce offer on re-INVITE without SDP. ... When PJSIP receives a re-INVITE without an SDP offer the INVITE session library will first call the on_create_offer callback and if unavailable then use the active negotiated SDP as the offer. In some cases this would result in a different SDP then was previously used without an incremented SDP version number. The two known cases are: 1. Sending an initial INVITE with a set of codecs and having the remote side answer with a subset. The active negotiated SDP would have the pruned list but would not have an incremented SDP version number. 2. Using re-INVITE for unhold. We would modify the active negotiated SDP but would not increment the SDP version. To solve these, and potential other unknown cases, the on_create_offer callback has now been implemented which produces a fresh offer with incremented SDP version number. This better fits within the model provided by the INVITE session library. ASTERISK-28452 Change-Id: I2d81048d54edcb80fe38fdbb954a86f0a58281a1	5 years ago
Ben Ford	88461a23bb	res_pjsip_session.c: Check topology on re-invite. ... Removes an unnecessary check for the conditional that compares the stream topologies to see if they are equal to suppress re-invites. This was a problem when a Digium phone received an INVITE that offered codecs different than what it supported, causing Asterisk to send the re-invite. ASTERISK-29303 Change-Id: I04dc91befb2387904e28a9aaeaa3bcdbcaa7fa63	5 years ago
Boris P. Korzun	3096dcee15	res_config_pgsql: Limit realtime_pgsql() to return one (no more) record. ... Added a SELECT 'LIMIT' clause to realtime_pgsql() and refactored the function. ASTERISK-29293 #close Change-Id: If5a6d4b1072ea2e6e89059b21139d554a74b34f5	5 years ago
Jaco Kroon	9a1cf7fb77	res_odbc_transaction: correctly initialise forcecommit value from DSN. ... Also improve the in-process documentation to clarify that the value is initialised from the DSN and not default false, but that the DSN's value is default false if unset. ASTERISK-29311 #close Change-Id: I46e2379f7b0656034442bce77cb37ccd4e61098d Signed-off-by: Jaco Kroon <jaco@uls.co.za>	5 years ago
Ivan Poddubnyi	c63ab5580c	app_queue: Fix conversion of complex extension states into device states ... Queue members using dialplan hints as a state interface must handle INUSE+RINGING hint as RINGINUSE devstate, and INUSE + ONHOLD as INUSE. ASTERISK-28369 Change-Id: I127e06943d4b4f1afc518f9e396de77449992b9f	5 years ago
Jaco Kroon	a63c7883b4	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
Alexander Traud	99666117be	chan_sip: Filter pass-through audio/video formats away, again. ... Instead of looking for pass-through formats in the list of transcodable formats (which is going to find nothing), go through the result which is going to be the jointcaps of the tech_pvt of the channel. Finally, only with that list, ast_format_cap_remove(.) is going to succeed. This restores the behaviour of Asterisk 1.8. However, it does not fix ASTERISK_29282 because that issue report is about chan_sip and PJSIP. Here, only chan_sip is fixed because PJSIP does not even call ast_rtp_instance_available_formats -> ast_translate_available_format. Change-Id: Icade2366ac2b82935b95a9981678c987da2e8c34	5 years ago
Jaco Kroon	d847f7e8f8	func_odbc: Introduce minargs config and expose ARGC in addition to ARGn. ... minargs enables enforcing of minimum count of arguments to pass to func_odbc, so if you're unconditionally using ARG1 through ARG4 then this should be set to 4. func_odbc will generate an error in this case, so for example [FOO] minargs = 4 and ODBC_FOO(a,b,c) in dialplan will now error out instead of using a potentially leaked ARG4 from Gosub(). ARGC is needed if you're using optional argument, to verify whether or not an argument has been passed, else it's possible to use a leaked ARGn from Gosub (app_stack). So now you can safely do ${IF($[${ARGC}>3]?${ARGV}:default value)} kind of thing. Change-Id: I6ca0b137d90b03f6aa9c496991f6cbf1518f6c24 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	5 years ago