asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua C. Colp	6aac148d59	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	4 years ago
Joshua C. Colp	be3e469f98	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	4 years ago
Joshua C. Colp	60fb559ccc	xml: Allow deprecated_in and removed_in for MODULEINFO. ASTERISK-29337 Change-Id: I2211b7da8d29369f8649aeabce07679da0787f2b	4 years ago
Joshua C. Colp	60800b038a	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	4 years ago
Kevin Harwell	263f906af4	manager: Increase the non breaking AMI version number ASTERISK~29244 added three new AMI events, so bump the version number. Change-Id: I0e77fa36d38fb27dec3481d4ef08131330da0632	4 years ago
Asterisk Development Team	0afd37e3b5	Update CHANGES and UPGRADE.txt for 18.3.0	4 years ago
Joshua C. Colp	f7bda066bb	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	4 years ago
Jaco Kroon	23e41313a8	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>	4 years ago
Jaco Kroon	52707fba7f	app.h: Fix -Werror=zero-length-bounds compile errors in dev mode. Change-Id: I5c104dc1f8417ccd3d01faf86e84ccbf89bc3b31 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	4 years ago
Sean Bright	94debe5085	app_dial.c: Only send DTMF on first progress event. ASTERISK-29329 #close Change-Id: Ic58e7a17f1ff3f785a5b21dced88682581149601	4 years ago
Alexander Traud	262473c6d9	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	4 years ago
Alexander Traud	4fc0e16838	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	4 years ago
Sean Bright	3084084648	modules.conf: Fix differing usage of assignment operators. ASTERISK-24434 #close Change-Id: I0144e8d65d878128da59dcf3df12ca8cee47d6db	4 years ago
Sean Bright	e4cd7a7d0b	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	4 years ago
Sean Bright	16e4d1f36f	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	4 years ago
Torrey Searle	90ef6a14a7	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	4 years ago
Joshua C. Colp	a9acbd19f3	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	4 years ago
George Joseph	269bb08ea2	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	4 years ago
Joshua C. Colp	5f1c21e4ca	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	4 years ago
Joshua C. Colp	682f7d9437	asterisk: Update copyright. ASTERISK-29326 Change-Id: Ia95dbfb66e2d11ac4d1228444283bb2e4d77396a	4 years ago
Ben Ford	77328142b4	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	4 years ago
Alexander Traud	0323293142	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	4 years ago
Joshua C. Colp	976b1a1d7a	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	4 years ago
Joshua C. Colp	b43b81d953	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	4 years ago
Salah Ahmed	df8d335ad1	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	4 years ago
Holger Hans Peter Freyther	3286c04856	pjsip: Generate progress (once) when receiving a 180 with a SDP ASTERISK-29105 Change-Id: If1615fe7115fe544ef974b044d3cea5c48b94a38	4 years ago
Nico Kooijman	7b052ec965	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	4 years ago
Nick French	dedfb334bd	res_pjsip: dont return early from registration if init auth fails If set_outbound_initial_authentication_credentials() fails, handle_client_registration() bails early without creating or sending a register message. [set_outbound_initial_authentication_credentials() failures can occur during the process of retrieving an oauth access token.] The return from handle_client_registration is ignored, so returning an error doesn't do any good. This is a real problem when the registration request is a re-register, because then the registration will still be marked 'active' despite the re-register never being sent at all. So instead, log a warning but let the registration be created and sent (and probably fail) and follow the normal registration failed retry/abort logic. ASTERISK-29315 #close Change-Id: I2e03b1ea7fba1fa1a8279086aa4b17679e7fa7fa	4 years ago
Alexei Gradinari	d5e73d2121	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	4 years ago
Sean Bright	6673c1b177	app_page.c: Don't fail to Page if beep sound file is missing ASTERISK-16799 #close Change-Id: I40367b0d6dbf66a39721bde060c8b2d734a61cf4	4 years ago
George Joseph	15afabdf8e	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	4 years ago
Kevin Harwell	be0a61bc3d	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	4 years ago
Boris P. Korzun	beb579bc99	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	4 years ago
Ben Ford	83b0f5963f	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	4 years ago
Jaco Kroon	7ab53fce7a	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>	4 years ago
Joshua C. Colp	1af2a84c8b	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	4 years ago
Jaco Kroon	916d5d5e45	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>	4 years ago
Ivan Poddubnyi	985d3e4940	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	4 years ago
Alexander Traud	1adf9368ee	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	4 years ago
Jaco Kroon	bee35fe04a	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>	4 years ago
Sebastien Duthil	092628c982	app_mixmonitor: Add AMI events MixMonitorStart, -Stop and -Mute. ASTERISK-29244 Change-Id: I1862d58264c2c8b5d8983272cb29734b184d67c5	4 years ago
George Joseph	dbd8908f8d	res_pjsip_refer: Always serialize calls to refer_progress_notify refer_progress_notify wasn't always being called from the progress serializer. This could allow clearing notification->progress->sub in one thread while another was trying to use it. * Instances where refer_progress_notify was being called in-line, have been changed to use ast_sip_push_task(). Change-Id: Idcf1934c4e873f2c82e2d106f8d9f040caf9fa1e	4 years ago
Kevin Harwell	fad0cf12e6	AST-2021-002: Remote crash possible when negotiating T.38 When an endpoint requests to re-negotiate for fax and the incoming re-invite is received prior to Asterisk sending out the 200 OK for the initial invite the re-invite gets delayed. When Asterisk does finally send the re-inivite the SDP includes streams for both audio and T.38. This happens because when the pending topology and active topologies differ (pending stream is not in the active) in the delayed scenario the pending stream is appended to the active topology. However, in the fax case the pending stream should replace the active. This patch makes it so when a delay occurs during fax negotiation, to or from, the audio stream is replaced by the T.38 stream, or vice versa instead of being appended. Further when Asterisk sent the re-invite with both audio and T.38, and the endpoint responded with a declined T.38 stream then Asterisk would crash when attempting to change the T.38 state. This patch also puts in a check that ensures the media state has a valid fax session (associated udptl object) before changing the T.38 state internally. ASTERISK-29203 #close Change-Id: I407f4fa58651255b6a9030d34fd6578cf65ccf09	4 years ago
Alexander Traud	703158b903	rtp: Enable srtp replay protection Add option "srtpreplayprotection" rtp.conf to enable srtp replay protection. ASTERISK-29260 Reported by: Alexander Traud Change-Id: I5cd346e3c6b6812039d1901aa4b7be688173b458	4 years ago
Ivan Poddubnyi	2770cc5872	res_pjsip_diversion: Fix adding more than one histinfo to Supported New responses sent within a PJSIP sessions are based on those that were sent before. Therefore, adding/modifying a header once causes it to be sent on all responses that follow. Sending 181 Call Is Being Forwarded many times first adds "histinfo" duplicated more and more, and eventually overflows past the array boundary. This commit adds a check preventing adding "histinfo" more than once, and skipping it if there is no more space in the header. Similar overflow situations can also occur in res_pjsip_path and res_pjsip_outbound_registration so those were also modified to check the bounds and suppress duplicate Supported values. ASTERISK-29227 Reported by: Ivan Poddubny Change-Id: Id43704a1f1a0293e35cc7f844026f0b04f2ac322	4 years ago
Sean Bright	5a6f2f913b	res_rtp_asterisk.c: Fix signed mismatch that leads to overflow ASTERISK-29205 #close Change-Id: Ib7aa65644e8df76e2378d7613ee7cf751b9d0bea	4 years ago
Joshua C. Colp	acb7ce4fe7	pjsip: Make modify_local_offer2 tolerate previous failed SDP. If a remote side is broken and sends an SDP that can not be negotiated the call will be torn down but there is a window where a second 183 Session Progress or 200 OK that is forked can be received that also attempts to negotiate SDP. Since the code marked the SDP negotiation as being done and complete prior to this it assumes that there is an active local and remote SDP which it can modify, while in fact there is not as the SDP did not successfully negotiate. Since there is no local or remote SDP a crash occurs. This patch changes the pjmedia_sdp_neg_modify_local_offer2 function to no longer assume that a previous SDP negotiation was successful. ASTERISK-29196 Change-Id: I22de45916d3b05fdc2a67da92b3a38271ee5949e	4 years ago
Ben Ford	62e2dd484d	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	4 years ago
Boris P. Korzun	57d130d3aa	format_wav: Support of MIME-type for wav16 Provided a support of a MIME-type for wav16. Added new MIME-type for classic wav. ASTERISK-29275 #close Change-Id: I749bda287ba1ab20c1e0af5e4c0153817d47873b	4 years ago
Alexander Traud	45e48e387c	chan_sip: Allow [peer] without audio (text+video). Two previous commits, `620d9f4` and `6d980de`, allow to set up a call without audio, again. That was introduced originally with commit `f04d5fb` but changed and broke over time. The original commit missed one scenario: A [peer] section in sip.conf, which does not allow audio at all. In that case, chan_sip rejected the call, although even when the requester offered no audio. Now, chan_sip does not check whether there is no audio format but checks whether there is no format in general. In other words, if there is at least one format to offer, the call succeeds. However, to prevent calls with no-audio, chan_sip still rejects calls when both call parties (caller = requester of the call and callee = [peer] section in sip.conf) included audio. In such a case, it is expected that the call should have audio. ASTERISK-29280 Change-Id: I0fb74faf51ef22a60c10b467df6a4d1c1943b73e	4 years ago

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