32857 Commits (e1126ffc10efcb6f44822937836ebfbed9d109ec)

 

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Author	SHA1	Message	Date
Ben Ford	e1126ffc10	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	b046e960af	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
Ivan Poddubnyi	4d8fc97e4a	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	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
Alexander Traud	5894535fed	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	b0f349a330	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
Sebastien Duthil	6e695c867f	app_mixmonitor: Add AMI events MixMonitorStart, -Stop and -Mute. ... ASTERISK-29244 Change-Id: I1862d58264c2c8b5d8983272cb29734b184d67c5	5 years ago
Kevin Harwell	5e998d8bd3	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	5 years ago
Alexander Traud	389b8b0774	rtp: Enable srtp replay protection ... Add option "srtpreplayprotection" rtp.conf to enable srtp replay protection. ASTERISK-29260 Reported by: Alexander Traud Change-Id: I5cd346e3c6b6812039d1901aa4b7be688173b458	5 years ago
Ivan Poddubnyi	7d15655f9d	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	5 years ago
Sean Bright	e7b13df394	res_rtp_asterisk.c: Fix signed mismatch that leads to overflow ... ASTERISK-29205 #close Change-Id: Ib7aa65644e8df76e2378d7613ee7cf751b9d0bea	5 years ago
Joshua C. Colp	492945ac60	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	5 years ago
George Joseph	15b4080679	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	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
Boris P. Korzun	a96eb6de6c	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	5 years ago
Alexander Traud	1f77c33c02	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	5 years ago
George Joseph	91b0778791	chan_iax2.c: Require secret and auth method if encryption is enabled ... If there's no secret specified for an iax2 peer and there's no secret specified in the dial string, Asterisk will crash if the auth method requested by the peer is MD5 or plaintext. You also couldn't specify a default auth method in the [general] section of iax.conf so if you don't have static peers defined and just use the dial string, Asterisk will still crash even if you have a secret specified in the dial string. * Added logic to iax2_call() and authenticate_reply() to print a warning and hanhup the call if encryption is requested and there's no secret or auth method. This prevents the crash. * Added the ability to specify a default "auth" in the [general] section of iax.conf. ASTERISK-29624 Reported by: N A Change-Id: I5928e16137581f7d383fcc7fa04ad96c919e6254	5 years ago
Sean Bright	4a71b08091	app_read: Release tone zone reference on early return. ... Change-Id: I350939f2220f9e5d44ddf4c8d9a4c99fde4d169a	6 years ago
Alexander Traud	620d9f4782	chan_sip: Set up calls without audio (text+video), again. ... The previous commit 6d980de fixed this issue in the core of Asterisk. With that, each channel technology can be used without audio theoretically. Practically, the channel-technology driver chan_sip turned out to have an invalid check preventing that. chan_sip tested whether there is at least one audio format. However, chan_sip has to test whether there is at least one format. More cannot be tested while requesting chan_sip because only the [general] capabilities but not the [peer] caps are known yet. And the [peer] caps might not be a subset or show any intersection with the [general] caps. This change here fixes this. The original commit f04d5fb, thirteen years ago, contained a software bug as it passed ANY audio capability to the channel-technology driver. Instead, it should have passed NO audio format. Therefore, this addressed issue here was not noticed in Asterisk 1.6.x and Asterisk 1.8. Then, Asterisk 10 changed that from ANY to NO, but nobody reported since then. ASTERISK-29265 Change-Id: Ic16a3bf13cd1b5c4fc4041ed74961177d96b600f	6 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	6 years ago
Alexander Traud	6d980de282	channel: Set up calls without audio (text+video), again. ... ASTERISK-29259 Change-Id: Ib6a6550e0e08355745d66da8e60ef49e81f9c6c5	6 years ago
roadkill	9b5d20e3d5	res/res_pjsip.c: allow user=phone when number contain *# ... if From number contain * or # asterisk will not add user=phone Currently only number that uses AST_DIGIT_ANYNUM can have "user=phone" but the validation should use AST_DIGIT_ANY this is a problem when you want to send call to ISUP as they will disregard the From header and either replace From with anonymous or with p-asserted-identity ASTERISK-29261 Reported by: Mark Petersen Tested by: Mark Petersen Change-Id: I3307bdbf757582740bfee4110e85f7b6c9291cc4	6 years ago
Alexander Traud	4aff42b274	chan_sip: SDP: Reject audio streams correctly. ... This completes the fix for ASTERISK_24543. Only when the call is an outgoing call, consult and append the configured format capabilities (p->caps). When all audio formats got rejected the negotiated format capabilities (p->jointcaps) contain no audio formats for incoming calls. This is required when there are other accepted media streams. ASTERISK-29258 Change-Id: I8bab31c7f3f3700dce204b429ad238a524efebb9	6 years ago
Ivan Poddubnyi	05472da92b	main/frame: Add missing control frame names to ast_frame_subclass2str ... Log proper control frame names instead of "Unknown control '14'", etc. Change-Id: I1724f2f4d1b064b25a5c93a7da0cb03be5143935	6 years ago
Boris P. Korzun	92f5cf7f2d	res_musiconhold: Add support of various URL-schemes by MoH. ... Provided a support of variuos URL-schemes for res_musiconhold, registered by ast_bucket_scheme_register(). ASTERISK-29262 #close Change-Id: If0ea8697587353dce358a70035d82649fd4632b6	6 years ago
Jaco Kroon	060ce10163	AC_HEADER_STDC causes a compile failure with autoconf 2.70 ... From https://www.mail-archive.com/bug-autoconf@gnu.org/msg04408.html > ... the long-obsolete AC_HEADER_STDC, previously used internally by > AC_INCLUDES_DEFAULT, used AC_EGREP_HEADER. The AC_HEADER_STDC macro > is now a no-op (and is not used at all within Autoconf anymore), so > that change is likely what made the first use of AC_EGREP_HEADER the > one inside the if condition, causing the observed results. The implication is that the test does nothing anyway, and due to it being a no-op from 2.70 onwards, results in the required not being set to yes, resulting in ./configure to fail. Change-Id: Ic1ff38d87f791fbf1f2a80512f81bb7110392460 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Alexander Traud	10a0a0c59b	pjsip_scheduler: Fix pjsip show scheduled_tasks like for compiler Clang. ... Otherwise, Clang 10 warned because of logical-not-parentheses. Change-Id: Ia8fb493f727b08070eb2dcf520c08df34ed11d79	6 years ago
Alexander Traud	df6afadf26	res_pjsip_session: Avoid sometimes-uninitialized warning with Clang. ... ASTERISK-29248 Change-Id: I2b17bd5ffb246bc64c463402c9831413da78a556	6 years ago
Sean Bright	6d2bec7028	res_pjsip_pubsub: Fix truncation of persisted SUBSCRIBE packet ... The last argument to ast_copy_string() is the buffer size, not the number of characters, so we add 1 to avoid stamping out the final \n in the persisted SUBSCRIBE message. Change-Id: I019b78942836f57965299af15d173911fcead5b2	6 years ago
Ben Ford	948ceb1228	chan_pjsip.c: Add parameters to frame in indicate. ... There are a couple of parameters (datalen and data) that do not get set in chan_pjsip_indicate which could cause an Invalid message to pop up for things such as fax. This patch adds them to the frame. Change-Id: Ia51be086a0708be905e73d1f433572c49c7e38f8	6 years ago
Robert Cripps	24e678b9bb	res/res_pjsip_session.c: Check that media type matches in ... function ast_sip_session_media_state_add. Check ast_media_type matches when a ast_sip_session_media is found otherwise when transitioning from say image to audio, the wrong session is returned in the first if statement. ASTERISK-29220 #close Change-Id: I6f6efa9b821ebe8881bb4c8c957f8802ddcb4b5d	6 years ago
Jean Aunis	c559667868	Stasis/messaging: tech subscriptions conflict with endpoint subscriptions. ... When both a tech subscription and an endpoint subscription exist for a given endpoint, TextMessageReceived events are dispatched to the tech subscription only. ASTERISK-29229 Change-Id: I9eac4cba5f9e27285a282509395347abc58fc2b8	6 years ago
Alexander Traud	1c05667cfc	chan_sip: SDP: Sidestep stream parsing when its media is disabled. ... Previously, chan_sip parsed all known media streams in an SDP offer like video (and text) even when videosupport=no (and textsupport=no). This wasted processor power. Furthermore, chan_sip accepted SDP offers, including no audio but just video (or text) streams although videosupport=no (or textsupport=no). Finally, chan_sip denied the whole offer instead of individual streams when they had encryption (SDES-sRTP) unexpectedly enabled. ASTERISK-29238 ASTERISK-29237 ASTERISK-29222 Change-Id: Ie49e4e2a11f0265f914b684738348ba8c0f89755	6 years ago
Ivan Poddubnyi	f2aa6c7017	chan_pjsip: Assign SIPDOMAIN after creating a channel ... session->channel doesn't exist until chan_pjsip creates it, so intead of setting a channel variable every new incoming call sets one and the same global variable. This patch moves the code to chan_pjsip so that SIPDOMAIN is set on a newly created channel, it also removes a misleading reference to channel->session used to fetch call pickup configuraion. ASTERISK-29240 Change-Id: I90c9bbbed01f5d8863585631a29322ae4e046755	6 years ago
Ivan Poddubnyi	134d2e729d	chan_pjsip: Stop queueing control frames twice on outgoing channels ... The fix for ASTERISK-27902 made chan_pjsip process SIP responses twice. This resulted in extra noise in logs (for example, "is making progress" and "is ringing" get logged twice by app_dial), as well as in noise in signalling: one incoming 183 Session Progress results in 2 outgoing 183-s. This change splits the response handler into 2 functions: - one for updating HANGUPCAUSE, which is still called twice, - another that does the rest, which is called only once as before. ASTERISK-28016 Reported-by: Alex Hermann ASTERISK-28549 Reported-by: Gant Liu ASTERISK-28185 Reported-by: Julien Change-Id: I0a1874be5bb5ed12d572d17c7f80de6e5e542940	6 years ago
Jaco Kroon	2d3441772b	contrib/systemd: Added note on common issues with systemd and asterisk ... With newer version of linux /var/run/ is a symlink to /run/ that has been turned into tmpfs. Added note that if asterisk has to bind to a specific IP that systemd has to wait until the network is up. Added note on how to make sure that the environment variable HOSTNAME is included. ASTERISK-29216 Reported by: Mark Petersen Tested by: Mark Petersen Change-Id: Ib3e560655befd3e99eec743687144f5569533379	6 years ago
George Joseph	9a4486e9fb	Revert "res_pjsip_outbound_registration.c: Use our own scheduler and other stuff" ... This reverts commit 2fe76dd816. Reason for revert: Too many issues reported. Need to research and correct. ASTERISK-29230 ASTERISK-29231 Reported by: Michael Maier Change-Id: I6453af680e17d8ffe7af2c5de7e1b2a58c8793cb	6 years ago
Jaco Kroon	c797500956	func_lock: fix multiple-channel-grant problems. ... Under contention it becomes possible that multiple channels will be told they successfully obtained the lock, which is a bug. Please refer ASTERISK-29217 This introduces a couple of changes. 1. Replaces requesters ao2 container with simple counter (we don't really care who is waiting for the lock, only how many). This is updated undex ->mutex to prevent memory access races. 2. Correct semantics for ast_cond_timedwait() as described in pthread_cond_broadcast(3P) is used (multiple threads can be released on a single _signal()). 3. Module unload races are taken care of and memory properly cleaned up. Change-Id: I6f68b5ec82ff25b2909daf6e4d19ca864a463e29 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Jaco Kroon	4e038c1eaa	pbx_lua: Add LUA_VERSIONS environment variable to ./configure. ... On Gentoo it's possible to have multiple lua versions installed, all with a path of /usr, so it's not possible to use the current --with-lua option to determisticly pin to a specific version as is required by the Gentoo PMS standards. This environment variable allows to lock to specific versions, unversioned check will be skipped if this variable is supplied. Change-Id: I8c403eda05df25ee0193960262ce849c7d2fd088 Signed-off-by: Jaco Kroon <jaco@uls.co.za>	6 years ago
Kevin Harwell	3bcf483373	app_mixmonitor: cleanup datastore when monitor thread fails to launch ... launch_monitor_thread is responsible for creating and initializing the mixmonitor, and dependent data structures. There was one off nominal path after the datastore gets created that triggers when the channel being monitored is hung up prior to monitor starting itself. If this happened the monitor thread would not "launch", and the mixmonitor object and associated objects are freed, including the underlying datastore data object. However, the datastore itself was not removed from the channel, so when the channel eventually gets destroyed it tries to access the previously freed datastore data and crashes. This patch removes and frees datastore object itself from the channel before freeing the mixmonitor object thus ensuring the channel does not call it when destroyed. ASTERISK-28947 #close Change-Id: Id4f9e958956d62473ed5ff06c98ae3436e839ff8	6 years ago
Sean Bright	44d68bd56b	app_voicemail: Prevent deadlocks when out of ODBC database connections ... ASTERISK-28992 #close Change-Id: Ia7d608924036139ee2520b840d077762d02668d0	6 years ago
Dan Cropp	ffa87ecade	chan_pjsip: Incorporate channel reference count into transfer_refer(). ... Add channel reference count for PJSIP REFER. The call could be terminated prior to the result of the transfer. In that scenario, when the SUBSCRIBE/NOTIFY occurred several minutes later, it would attempt to access a session which was no longer valid. Terminate event subscription if pjsip_xfer_initiate() or pjsip_xfer_send_request() fails in transfer_refer(). ASTERISK-29201 #close Reported-by: Dan Cropp Change-Id: I3fd92fd14b4e3844d3d7b0f60fe417a4df5f2435	6 years ago
Kevin Harwell	4274a4a7dd	pbx_realtime: wrong type stored on publish of ast_channel_snapshot_type ... A prior patch segmented channel snapshots, and changed the underlying data object type associated with ast_channel_snapshot_type stasis messages. Prior to Asterisk 18 it was a type ast_channel_snapshot, but now it type ast_channel_snapshot_update. When publishing ast_channel_snapshot_type in pbx_realtime the ast_channel_snapshot was being passed in as the message data object. When a handler, expecting a data object type of ast_channel_snapshot_update, dereferenced this value a crash would occur. This patch makes it so pbx_realtime now uses the expected type, and channel snapshot publish method when publishing. ASTERISK-29168 #close Change-Id: I9a2cfa0ec285169317f4b9146e4027da8a4fe896	6 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	6 years ago
Nick French	505939c9ed	res_pjsip: Prevent segfault in UDP registration with flow transports ... Segfault occurs during outbound UDP registration when all transport states are being iterated over. The transport object in the transport is accessed, but flow transports have a NULL transport object. Modify to not iterate over any flow transport ASTERISK-29210 #close Change-Id: If28dc3a18bdcbd0a49598b09b7fe4404d45c996a	6 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	6 years ago
Torrey Searle	51e2187a14	res/res_pjsip_diversion: prevent crash on tel: uri in History-Info ... Add a check to see if the URI is a Tel URI and prevent crashing on trying to retrieve the reason parameter. ASTERISK-29191 ASTERISK-29219 Change-Id: I0320aa205f22cda511d60a2edf2b037e8fd6cc37 (cherry picked from commit a7aea71e60)	6 years ago
Richard Mudgett	058bc0d593	chan_vpb.cc: Fix compile errors. ... Fix the usual compile problem when someone adds a new callback to struct ast_channel_tech. Change-Id: I9bdeb8a8cc65f03b2d6e4f2eb5809af47c906c32	6 years ago
Richard Mudgett	6d7af72559	res_pjsip_session.c: Fix compiler warnings. ... AST_VECTOR_SIZE() returns a size_t. This is not always equivalent to an unsigned long on all machines. Change-Id: I0a4189a104e6e3a2e2273de06620eaef19df9338	6 years ago
Sungtae Kim	02c4b2ac60	res_pjsip_session: Fixed NULL active media topology handle ... Added NULL pointer check to prevent Asterisk crash. ASTERISK-29215 Change-Id: If07e50ea8d78cb610af9195fc13b5dca4bfcef95	6 years ago