Commit 0415120ced9aece945567327cb5525d977bf3db7 - asterisk

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.version less more

0		16.15.1⏎
	0	16.16.1⏎

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CHANGES less more

10	10	=== and the other UPGRADE files for older releases.
11	11	===
12	12	==============================================================================
	13
	14	------------------------------------------------------------------------------
	15	--- Functionality changes from Asterisk 16.15.0 to Asterisk 16.16.0 ----------
	16	------------------------------------------------------------------------------
	17
	18	Core
	19	------------------
	20	* The location where the media cache stores its temporary files
	21	is no longer hardcoded to /tmp but can now be configured separately
	22	via the astcachedir config variable in asterisk.conf. To retain
	23	backwards compatibility, the default location remains /tmp.
	24
	25	app_voicemail
	26	------------------
	27	* The VoiceMail application can now be configured to send greetings and
	28	instructions via early media and only answering the channel when it is
	29	time for the caller to record their message. This behavior can be
	30	activated by passing the new 'e' option to VoiceMail.
13	31
14	32	------------------------------------------------------------------------------
15	33	--- Functionality changes from Asterisk 16.13.0 to Asterisk 16.14.0 ----------

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ChangeLog less more

0		2020-12-22 21:02 +0000 Asterisk Development Team <asteriskteam@digium.com>
1
2		* asterisk 16.15.1 Released.
3
4		2020-12-22 15:02 +0000 [49fbd57889] Asterisk Development Team <asteriskteam@digium.com>
5
6		* Update for 16.15.1
7
8		2020-12-22 02:58 +0000 [354049e055] Torrey Searle <tsearle@voxbone.com>
	0	2021-02-18 16:48 +0000 Asterisk Development Team <asteriskteam@digium.com>
	1
	2	* asterisk 16.16.1 Released.
	3
	4	2021-02-01 15:24 +0000 [a5619097cd] Kevin Harwell <kharwell@sangoma.com>
	5
	6	* AST-2021-002: Remote crash possible when negotiating T.38
	7
	8	When an endpoint requests to re-negotiate for fax and the incoming
	9	re-invite is received prior to Asterisk sending out the 200 OK for
	10	the initial invite the re-invite gets delayed. When Asterisk does
	11	finally send the re-inivite the SDP includes streams for both audio
	12	and T.38.
	13
	14	This happens because when the pending topology and active topologies
	15	differ (pending stream is not in the active) in the delayed scenario
	16	the pending stream is appended to the active topology. However, in
	17	the fax case the pending stream should replace the active.
	18
	19	This patch makes it so when a delay occurs during fax negotiation,
	20	to or from, the audio stream is replaced by the T.38 stream, or vice
	21	versa instead of being appended.
	22
	23	Further when Asterisk sent the re-invite with both audio and T.38,
	24	and the endpoint responded with a declined T.38 stream then Asterisk
	25	would crash when attempting to change the T.38 state.
	26
	27	This patch also puts in a check that ensures the media state has a
	28	valid fax session (associated udptl object) before changing the
	29	T.38 state internally.
	30
	31	ASTERISK-29203 #close
	32
	33	Change-Id: I407f4fa58651255b6a9030d34fd6578cf65ccf09
	34
	35	2021-01-26 11:09 +0000 [3f4dfd5c02] Alexander Traud <pabstraud@compuserve.com>
	36
	37	* rtp: Enable srtp replay protection
	38
	39	Add option "srtpreplayprotection" rtp.conf to enable srtp
	40	replay protection.
	41
	42	ASTERISK-29260
	43	Reported by: Alexander Traud
	44
	45	Change-Id: I5cd346e3c6b6812039d1901aa4b7be688173b458
	46
	47	2020-12-28 06:43 +0000 [17561b5e64] Ivan Poddubnyi <ivan.poddubny@gmail.com>
	48
	49	* res_pjsip_diversion: Fix adding more than one histinfo to Supported
	50
	51	New responses sent within a PJSIP sessions are based on those that were
	52	sent before. Therefore, adding/modifying a header once causes it to be
	53	sent on all responses that follow.
	54
	55	Sending 181 Call Is Being Forwarded many times first adds "histinfo"
	56	duplicated more and more, and eventually overflows past the array
	57	boundary.
	58
	59	This commit adds a check preventing adding "histinfo" more than once,
	60	and skipping it if there is no more space in the header.
	61
	62	Similar overflow situations can also occur in res_pjsip_path and
	63	res_pjsip_outbound_registration so those were also modified to
	64	check the bounds and suppress duplicate Supported values.
	65
	66	ASTERISK-29227
	67	Reported by: Ivan Poddubny
	68
	69	Change-Id: Id43704a1f1a0293e35cc7f844026f0b04f2ac322
	70
	71	2020-12-11 14:49 +0000 [4cea145aa9] Sean Bright <sean.bright@gmail.com>
	72
	73	* res_rtp_asterisk.c: Fix signed mismatch that leads to overflow
	74
	75	ASTERISK-29205 #close
	76
	77	Change-Id: Ib7aa65644e8df76e2378d7613ee7cf751b9d0bea
	78
	79	2021-02-05 05:26 +0000 [321632b02e] Joshua C. Colp <jcolp@sangoma.com>
	80
	81	* pjsip: Make modify_local_offer2 tolerate previous failed SDP.
	82
	83	If a remote side is broken and sends an SDP that can not be
	84	negotiated the call will be torn down but there is a window
	85	where a second 183 Session Progress or 200 OK that is forked
	86	can be received that also attempts to negotiate SDP. Since
	87	the code marked the SDP negotiation as being done and complete
	88	prior to this it assumes that there is an active local and remote
	89	SDP which it can modify, while in fact there is not as the SDP
	90	did not successfully negotiate. Since there is no local or remote
	91	SDP a crash occurs.
	92
	93	This patch changes the pjmedia_sdp_neg_modify_local_offer2
	94	function to no longer assume that a previous SDP negotiation
	95	was successful.
	96
	97	ASTERISK-29196
	98
	99	Change-Id: I22de45916d3b05fdc2a67da92b3a38271ee5949e
	100
	101	2021-01-21 16:28 +0000 Asterisk Development Team <asteriskteam@digium.com>
	102
	103	* asterisk 16.16.0 Released.
	104
	105	2021-01-14 16:20 +0000 Asterisk Development Team <asteriskteam@digium.com>
	106
	107	* asterisk 16.16.0-rc1 Released.
	108
	109	2021-01-14 09:51 +0000 [6056818467] Asterisk Development Team <asteriskteam@digium.com>
	110
	111	* Update CHANGES and UPGRADE.txt for 16.16.0
	112	2020-12-30 07:56 +0000 [45e1d89135] Jean Aunis <jean.aunis@prescom.fr>
	113
	114	* Stasis/messaging: tech subscriptions conflict with endpoint subscriptions.
	115
	116	When both a tech subscription and an endpoint subscription exist for a given
	117	endpoint, TextMessageReceived events are dispatched to the tech subscription
	118	only.
	119
	120	ASTERISK-29229
	121
	122	Change-Id: I9eac4cba5f9e27285a282509395347abc58fc2b8
	123
	124	2020-12-23 08:44 +0000 [34b0960310] Alexander Traud <pabstraud@compuserve.com>
	125
	126	* chan_sip: SDP: Sidestep stream parsing when its media is disabled.
	127
	128	Previously, chan_sip parsed all known media streams in an SDP offer
	129	like video (and text) even when videosupport=no (and textsupport=no).
	130	This wasted processor power. Furthermore, chan_sip accepted SDP offers,
	131	including no audio but just video (or text) streams although
	132	videosupport=no (or textsupport=no). Finally, chan_sip denied the whole
	133	offer instead of individual streams when they had encryption (SDES-sRTP)
	134	unexpectedly enabled.
	135
	136	ASTERISK-29238
	137	ASTERISK-29237
	138	ASTERISK-29222
	139
	140	Change-Id: Ie49e4e2a11f0265f914b684738348ba8c0f89755
	141
	142	2020-12-29 12:16 +0000 [97afc9055f] Ivan Poddubnyi <ivan.poddubny@gmail.com>
	143
	144	* chan_pjsip: Assign SIPDOMAIN after creating a channel
	145
	146	session->channel doesn't exist until chan_pjsip creates it, so intead of
	147	setting a channel variable every new incoming call sets one and the same
	148	global variable.
	149
	150	This patch moves the code to chan_pjsip so that SIPDOMAIN is set on
	151	a newly created channel, it also removes a misleading reference to
	152	channel->session used to fetch call pickup configuraion.
	153
	154	ASTERISK-29240
	155
	156	Change-Id: I90c9bbbed01f5d8863585631a29322ae4e046755
	157
	158	2020-12-31 05:53 +0000 [17fa9c93d0] Ivan Poddubnyi <ivan.poddubny@gmail.com>
	159
	160	* chan_pjsip: Stop queueing control frames twice on outgoing channels
	161
	162	The fix for ASTERISK-27902 made chan_pjsip process SIP responses twice.
	163	This resulted in extra noise in logs (for example, "is making progress"
	164	and "is ringing" get logged twice by app_dial), as well as in noise in
	165	signalling: one incoming 183 Session Progress results in 2 outgoing 183-s.
	166
	167	This change splits the response handler into 2 functions:
	168	- one for updating HANGUPCAUSE, which is still called twice,
	169	- another that does the rest, which is called only once as before.
	170
	171	ASTERISK-28016
	172	Reported-by: Alex Hermann
	173
	174	ASTERISK-28549
	175	Reported-by: Gant Liu
	176
	177	ASTERISK-28185
	178	Reported-by: Julien
	179
	180	Change-Id: I0a1874be5bb5ed12d572d17c7f80de6e5e542940
	181
	182	2020-12-23 11:41 +0000 [7df88c98d0] Jaco Kroon <jaco@uls.co.za>
	183
	184	* contrib/systemd: Added note on common issues with systemd and asterisk
	185
	186	With newer version of linux /var/run/ is a symlink to /run/ that has
	187	been turned into tmpfs.
	188
	189	Added note that if asterisk has to bind to a specific IP that
	190	systemd has to wait until the network is up.
	191
	192	Added note on how to make sure that the environment variable
	193	HOSTNAME is included.
	194
	195	ASTERISK-29216
	196	Reported by: Mark Petersen
	197	Tested by: Mark Petersen
	198
	199	Change-Id: Ib3e560655befd3e99eec743687144f5569533379
	200
	201	2021-01-07 08:39 +0000 [7ed20b9d3b] George Joseph <gjoseph@digium.com>
	202
	203	* Revert "res_pjsip_outbound_registration.c: Use our own scheduler and other stuff"
	204
	205	This reverts commit e1fd51cd2c183a3eb61803f52277d1a7d7a18c9c.
	206
	207	Reason for revert: Too many issues reported. Need to research and correct.
	208
	209	ASTERISK-29230
	210	ASTERISK-29231
	211	Reported by: Michael Maier
	212
	213	Change-Id: Ie2356c1cd74436603600599885286b98055b8a17
	214
	215	2020-12-23 11:41 +0000 [aca435dfe7] Jaco Kroon <jaco@uls.co.za>
	216
	217	* pbx_lua: Add LUA_VERSIONS environment variable to ./configure.
	218
	219	On Gentoo it's possible to have multiple lua versions installed, all
	220	with a path of /usr, so it's not possible to use the current --with-lua
	221	option to determisticly pin to a specific version as is required by the
	222	Gentoo PMS standards.
	223
	224	This environment variable allows to lock to specific versions,
	225	unversioned check will be skipped if this variable is supplied.
	226
	227	Change-Id: I8c403eda05df25ee0193960262ce849c7d2fd088
	228	Signed-off-by: Jaco Kroon <jaco@uls.co.za>
	229
	230	2020-12-18 13:06 +0000 [32e36144c7] Jaco Kroon <jaco@uls.co.za>
	231
	232	* func_lock: fix multiple-channel-grant problems.
	233
	234	Under contention it becomes possible that multiple channels will be told
	235	they successfully obtained the lock, which is a bug. Please refer
	236
	237	ASTERISK-29217
	238
	239	This introduces a couple of changes.
	240
	241	1. Replaces requesters ao2 container with simple counter (we don't
	242	really care who is waiting for the lock, only how many). This is
	243	updated undex ->mutex to prevent memory access races.
	244	2. Correct semantics for ast_cond_timedwait() as described in
	245	pthread_cond_broadcast(3P) is used (multiple threads can be released
	246	on a single _signal()).
	247	3. Module unload races are taken care of and memory properly cleaned
	248	up.
	249
	250	Change-Id: I6f68b5ec82ff25b2909daf6e4d19ca864a463e29
	251	Signed-off-by: Jaco Kroon <jaco@uls.co.za>
	252
	253	2020-12-07 16:59 +0000 [e127a57761] Dan Cropp <dan@amtelco.com>
	254
	255	* chan_pjsip: Incorporate channel reference count into transfer_refer().
	256
	257	Add channel reference count for PJSIP REFER. The call could be terminated
	258	prior to the result of the transfer. In that scenario, when the SUBSCRIBE/NOTIFY
	259	occurred several minutes later, it would attempt to access a session which was
	260	no longer valid. Terminate event subscription if pjsip_xfer_initiate() or
	261	pjsip_xfer_send_request() fails in transfer_refer().
	262
	263	ASTERISK-29201 #close
	264	Reported-by: Dan Cropp
	265
	266	Change-Id: I3fd92fd14b4e3844d3d7b0f60fe417a4df5f2435
	267
	268	2020-12-23 13:06 +0000 [e96f744816] Kevin Harwell <kharwell@sangoma.com>
	269
	270	* app_mixmonitor: cleanup datastore when monitor thread fails to launch
	271
	272	launch_monitor_thread is responsible for creating and initializing
	273	the mixmonitor, and dependent data structures. There was one off
	274	nominal path after the datastore gets created that triggers when
	275	the channel being monitored is hung up prior to monitor starting
	276	itself.
	277
	278	If this happened the monitor thread would not "launch", and the
	279	mixmonitor object and associated objects are freed, including the
	280	underlying datastore data object. However, the datastore itself was
	281	not removed from the channel, so when the channel eventually gets
	282	destroyed it tries to access the previously freed datastore data
	283	and crashes.
	284
	285	This patch removes and frees datastore object itself from the channel
	286	before freeing the mixmonitor object thus ensuring the channel does
	287	not call it when destroyed.
	288
	289	ASTERISK-28947 #close
	290
	291	Change-Id: Id4f9e958956d62473ed5ff06c98ae3436e839ff8
	292
	293	2020-12-24 09:03 +0000 [2b7af3eb27] Sean Bright <sean.bright@gmail.com>
	294
	295	* app_voicemail: Prevent deadlocks when out of ODBC database connections
	296
	297	ASTERISK-28992 #close
	298
	299	Change-Id: Ia7d608924036139ee2520b840d077762d02668d0
	300
	301	2020-12-18 09:16 +0000 [4c5bffb217] Sean Bright <sean.bright@gmail.com>
	302
	303	* asterisk: Export additional manager functions
	304
	305	Rename check_manager_enabled() and check_webmanager_enabled() to begin
	306	with ast_ so that the symbols are automatically exported by the
	307	linker.
	308
	309	ASTERISK~29184
	310
	311	Change-Id: I85762b9a5d14500c15f6bad6507138c8858644c9
	312
	313	2020-12-26 11:42 +0000 [89cf7899be] Richard Mudgett <rmudgett@digium.com>
	314
	315	* res_pjsip_session.c: Fix compiler warnings.
	316
	317	AST_VECTOR_SIZE() returns a size_t. This is not always equivalent to an
	318	unsigned long on all machines.
	319
	320	Change-Id: I0a4189a104e6e3a2e2273de06620eaef19df9338
	321
	322	2020-12-13 06:03 +0000 [ab3f57d88f] Sungtae Kim <pchero21@gmail.com>
	323
	324	* res_pjsip_session: Fixed NULL active media topology handle
	325
	326	Added NULL pointer check to prevent Asterisk crash.
	327
	328	ASTERISK-29215
	329
	330	Change-Id: If07e50ea8d78cb610af9195fc13b5dca4bfcef95
	331
	332	2020-12-22 02:58 +0000 [9196e0d1d5] Torrey Searle <tsearle@voxbone.com>
9	333
10	334	* res/res_pjsip_diversion: prevent crash on tel: uri in History-Info
11	335

17	341
18	342	Change-Id: I0320aa205f22cda511d60a2edf2b037e8fd6cc37
19	343
20		2020-11-19 12:34 +0000 Asterisk Development Team <asteriskteam@digium.com>
21
22		* asterisk 16.15.0 Released.
	344	2020-12-11 13:27 +0000 [0a23296834] Sean Bright <sean.bright@gmail.com>
	345
	346	* app_chanspy: Spyee information missing in ChanSpyStop AMI Event
	347
	348	The documentation in the wiki says there should be spyee-channel
	349	information elements in the ChanSpyStop AMI event.
	350
	351	https://wiki.asterisk.org/wiki/x/Xc5uAg
	352
	353	However, this is not the case in Asterisk <= 16.10.0 Version. We're
	354	using these Spyee* arguments since Asterisk 11.x, so these arguments
	355	vanished in Asterisk 12 or higher.
	356
	357	For maximum compatibility, we still send the ChanSpyStop event even if
	358	we are not able to find any 'Spyee' information.
	359
	360	ASTERISK-28883 #close
	361
	362	Change-Id: I81ce397a3fd614c094d043ffe5b1b1d76188835f
	363
	364	2020-11-30 19:27 +0000 [a47e6965b3] Sungtae Kim <pchero21@gmail.com>
	365
	366	* res_ari: Fix wrong media uri handle for channel play
	367
	368	Fixed wrong null object handle in
	369	/channels/<channel_id>/play request handler.
	370
	371	ASTERISK-29188
	372
	373	Change-Id: I6691c640247a51ad15f23e4a203ca8430809bafe
	374
	375	2020-12-10 09:09 +0000 [5a2867efa9] George Joseph <gjoseph@digium.com>
	376
	377	* logger.c: Automatically add a newline to formats that don't have one
	378
	379	Scope tracing allows you to not specify a format string or
	380	variable, in which case it just prints the indent, file,
	381	function, and line number. The trace output automatically
	382	adds a newline to the end in this case. If you also have
	383	debugging turned on for the module, a debug message is
	384	also printed but the standard log functionality which
	385	prints it doesn't add the newline so you have messages
	386	that don't break correctly.
	387
	388	* format_log_message_ap(), which is the common log
	389	message formatter for all channels, now adds a
	390	newline to the end of format strings that don't
	391	already have a newline.
	392
	393	ASTERISK-29209
	394	Reported by: Alexander Traud
	395
	396	Change-Id: I994a7df27f88df343b7d19f3e81a4b562d9d41da
	397
	398	2020-12-08 11:37 +0000 [11def974a8] Pirmin Walthert <infos@nappsoft.ch>
	399
	400	* res_pjsip_nat.c: Create deep copies of strings when appropriate
	401
	402	In rewrite_uri asterisk was not making deep copies of strings when
	403	changing the uri. This was in some cases causing garbage in the route
	404	header and in other cases even crashing asterisk when receiving a
	405	message with a record-route header set. Thanks to Ralf Kubis for
	406	pointing out why this happens. A similar problem was found in
	407	res_pjsip_transport_websocket.c. Pjproject needs as well to be patched
	408	to avoid garbage in CANCEL messages.
	409
	410	ASTERISK-29024 #close
	411
	412	Change-Id: Ic5acd7fa2fbda3080f5f36ef12e46804939b198b
	413
	414	2020-12-16 06:17 +0000 [7e4bb4ed11] Joshua C. Colp <jcolp@sangoma.com>
	415
	416	* res_pjsip_pidf_digium_body_supplement: Support Sangoma user agent.
	417
	418	This adds support for both Digium and Sangoma user agent strings
	419	for the Sangoma specific body supplement.
	420
	421	Change-Id: Ib99362b24b91d3cbe888d8b2fce3fad5515d9482
	422
	423	2020-12-10 17:06 +0000 [bb46595799] Nathan Bruning <nathan@iperity.com>
	424
	425	* res_musiconhold: Don't crash when real-time doesn't return any entries
	426
	427	ASTERISK-29211 #close
	428
	429	Change-Id: Ifbf0a4f786ab2a52342f9d1a1db4c9907f069877
	430
	431	2020-10-29 06:25 +0000 [8d2558209b] laszlovl <digium@lvlconsultancy.nl>
	432
	433	* Introduce astcachedir, to be used for temporary bucket files
	434
	435	As described in the issue, /tmp is not a suitable location for a
	436	large amount of cached media files, since most distributions make
	437	/tmp a RAM-based tmpfs mount with limited capacity.
	438
	439	I opted for a location that can be configured separately, as opposed
	440	to using a subdirectory of spooldir, given the different storage
	441	profile (transient files vs files that might stay there indefinitely).
	442
	443	This commit just makes the cache directory configurable, but leaves
	444	it at /tmp by default, to ensure backwards compatibility.
	445
	446	A future commit that only targets master could change the default
	447	location to something more sensible such as /var/tmp/asterisk. At
	448	that point, the cachedir could be created and cleaned up during
	449	uninstall by the Makefile script.
	450
	451	ASTERISK-29143
	452
	453	Change-Id: Ic54e95199405abacd9e509cef5f08fa14c510b5d
	454
	455	2020-10-29 12:21 +0000 [ea744ca7c2] Joshua C. Colp <jcolp@sangoma.com>
	456
	457	* pjsip: Match lifetime of INVITE session to our session.
	458
	459	In some circumstances it was possible for an INVITE
	460	session to be destroyed while we were still using it.
	461	This occurred due to the reference on the INVITE session
	462	being released internally as a result of its state
	463	changing to DISCONNECTED.
	464
	465	This change adds a reference to the INVITE session
	466	which is released when our own session is destroyed,
	467	ensuring that the INVITE session remains valid for
	468	the lifetime of our session.
	469
	470	ASTERISK-29022
	471
	472	Change-Id: I300c6d9005ff0e6efbe1132daefc7e47ca6228c9
	473
	474	2020-11-21 11:51 +0000 [0c185c9e21] Sean Bright <sean.bright@gmail.com>
	475
	476	* res_http_media_cache.c: Set reasonable number of redirects
	477
	478	By default libcurl does not follow redirects, so we explicitly enable
	479	it by setting CURLOPT_FOLLOWLOCATION. Once that is enabled, libcurl
	480	will follow up to CURLOPT_MAXREDIRS redirects, which by default is
	481	configured to be unlimited.
	482
	483	This patch sets CURLOPT_MAXREDIRS to a more reasonable default (8). If
	484	we determine at some point that this needs to be increased on
	485	configurable it is a trivial change.
	486
	487	ASTERISK-29173 #close
	488
	489	Change-Id: I4925ebbcf0c7d728bb9252b3795b3479ae225b30
	490
	491	2020-11-23 14:56 +0000 [ddbf3a7f73] Sean Bright <sean.bright@gmail.com>
	492
	493	* media_cache: Fix reference leak with bucket file metadata
	494
	495	Change-Id: Ia0e4124110df613ce5fdfa9ef8780016ebaa52c6
	496
	497	2020-11-24 00:55 +0000 [159522003a] Stanislav <stas.abramenkov@gmail.com>
	498
	499	* res_pjsip_stir_shaken: Fix module description
	500
	501	the 'J' is missing in module description.
	502	"PSIP STIR/SHAKEN Module for Asterisk" -> "PJSIP STIR/SHAKEN Module for Asterisk"
	503
	504	ASTERISK-29175 #close
	505
	506	Change-Id: I17da008540ee2e8496b644d05f995b320b54ad7a
	507
	508	2020-10-12 05:30 +0000 [15566494f9] Joshua C. Colp <jcolp@sangoma.com>
	509
	510	* voicemail: add option 'e' to play greetings as early media
	511
	512	When using this option, answering the channel is deferred until
	513	all prompts/greetings have been played and the caller is about
	514	to leave their message.
	515
	516	ASTERISK-29118 #close
	517
	518	Change-Id: I41b9f0428783c0bd697c8c994f906d1e75ce9ddb
	519
	520	2020-11-02 02:02 +0000 [4c79bc19d1] Alexander Traud <pabstraud@compuserve.com>
	521
	522	* loader: Sync load- and build-time deps.
	523
	524	In MODULEINFO, each depend has to be listed in .requires of AST_MODULE_INFO.
	525
	526	ASTERISK-29148
	527
	528	Change-Id: I254dd33194ae38d2877b8021c57c2a5deb6bbcd2
	529
	530	2020-11-18 13:11 +0000 [a360150ee0] Sean Bright <sean.bright@gmail.com>
	531
	532	* CHANGES: Remove already applied CHANGES update
	533
	534	Change-Id: Iee7163bc732d58c5cbaa2cfab1f5aab4a412060a
	535
	536	2020-11-04 07:39 +0000 [f667c5a781] Alexander Traud <pabstraud@compuserve.com>
	537
	538	* chan_sip: Remove unused sip_socket->port.
	539
	540	12 years ago, with ASTERISK_12115 the last four get/uses of socket.port
	541	vanished. However, the struct member itself and all seven set/uses
	542	remained as dead code.
	543
	544	ASTERISK-28798
	545
	546	Change-Id: Ib90516a49eca3d724a70191278aaf2144fb58c59
	547
	548	2020-11-17 14:19 +0000 [a8f6238cc8] Alexander Greiner-Baer <alex+asterisk@greiner-baer.de>
	549
	550	* res_pjsip: set Accept-Encoding to identity in OPTIONS response
	551
	552	RFC 3261 says that the Accept-Encoding header should be present
	553	in an options response. Permitted values according to RFC 2616
	554	are only compression algorithms like gzip or the default identity
	555	encoding. Therefore "text/plain" is not a correct value here.
	556	As long as the header is hard coded, it should be set to "identity".
	557
	558	Without this fix an Alcatel OmniPCX periodically logs warnings like
	559	"[sip_acceptIncorrectHeader] Header Accept-Encoding is malformed"
	560	on a SIP Trunk.
	561
	562	ASTERISK-29165 #close
	563
	564	Change-Id: I0aa2211ebf0b4c2ed554ac7cda794523803a3840
	565
	566	2020-11-13 06:19 +0000 [89d3de37ca] Boris P. Korzun <drtr0jan@yandex.ru>
	567
	568	* bridge_basic: Fixed setup of recall channels
	569
	570	Fixed a bug (like a typo) in retransfer_enter()
	571	at main/bridge_basic.c:2641. common_recall_channel_setup() setups
	572	common things on the recalled transfer target, but used same target
	573	as source instead trasfered.
	574
	575	ASTERISK-29161 #close
	576
	577	Change-Id: Ieb549654a621c38b1ad5e9d15b9f18823d9cc31f
	578
	579	2020-11-16 04:38 +0000 [d1a78e047d] Alexander Traud <pabstraud@compuserve.com>
	580
	581	* modules.conf: Align the comments for more conclusiveness.
	582
	583	Change-Id: I79cc693cd5a6e5dd7d403b7e91d970ff1ddf8306
	584
	585	2020-11-11 08:55 +0000 [8d8c9db618] George Joseph <gjoseph@digium.com>
	586
	587	* app_queue: Fix deadlock between update and show queues
	588
	589	Operations that update queues when shared_lastcall is set lock the
	590	queue in question, then have to lock the queues container to find the
	591	other queues with the same member. On the other hand, __queues_show
	592	(which is called by both the CLI and AMI) does the reverse. It locks
	593	the queues container, then iterates over the queues locking each in
	594	turn to display them. This creates a deadlock.
	595
	596	* Moved queue print logic from __queues_show to a separate function
	597	that can be called for a single queue.
	598
	599	* Updated __queues_show so it doesn't need to lock or traverse
	600	the queues container to show a single queue.
	601
	602	* Updated __queues_show to snap a copy of the queues container and iterate
	603	over that instead of locking the queues container and iterating over
	604	it while locked. This prevents us from having to hold both the
	605	container lock and the queue locks at the same time. This also
	606	allows us to sort the queue entries.
	607
	608	ASTERISK-29155
	609
	610	Change-Id: I78d4dc36728c2d7bc187b97d82673fc77f2bcf41
23	611
24	612	2020-11-12 12:03 +0000 Asterisk Development Team <asteriskteam@digium.com>
25	613

+4

-2

Makefile less more

48	48	export DESTDIR
49	49
50	50	export INSTALL_PATH # Additional prefix for the following paths
	51	export ASTCACHEDIR
51	52	export ASTETCDIR # Path for config files
52	53	export ASTVARRUNDIR
53	54	export ASTSPOOLDIR

554	555
555	556	NEWHEADERS=$(notdir $(wildcard include/asterisk/*.h))
556	557	OLDHEADERS=$(filter-out $(NEWHEADERS) $(notdir $(DESTDIR)$(ASTHEADERDIR)),$(notdir $(wildcard $(DESTDIR)$(ASTHEADERDIR)/*.h)))
557		INSTALLDIRS="$(ASTLIBDIR)" "$(ASTMODDIR)" "$(ASTSBINDIR)" "$(ASTETCDIR)" "$(ASTVARRUNDIR)" \
	558	INSTALLDIRS="$(ASTLIBDIR)" "$(ASTMODDIR)" "$(ASTSBINDIR)" "$(ASTCACHEDIR)" "$(ASTETCDIR)" "$(ASTVARRUNDIR)" \
558	559	"$(ASTSPOOLDIR)" "$(ASTSPOOLDIR)/dictate" "$(ASTSPOOLDIR)/meetme" \
559	560	"$(ASTSPOOLDIR)/monitor" "$(ASTSPOOLDIR)/system" "$(ASTSPOOLDIR)/tmp" \
560	561	"$(ASTSPOOLDIR)/voicemail" "$(ASTSPOOLDIR)/recording" \

778	779	done ; \
779	780	if [ "$(OVERWRITE)" = "y" ]; then \
780	781	echo "Updating asterisk.conf" ; \
781		sed -e 's\|^astetcdir.*$$\|astetcdir => $(ASTETCDIR)\|' \
	782	sed -e 's\|^astcachedir.*$$\|astcachedir => $(ASTCACHEDIR)\|' \
	783	-e 's\|^astetcdir.*$$\|astetcdir => $(ASTETCDIR)\|' \
782	784	-e 's\|^astmoddir.*$$\|astmoddir => $(ASTMODDIR)\|' \
783	785	-e 's\|^astvarlibdir.*$$\|astvarlibdir => $(ASTVARLIBDIR)\|' \
784	786	-e 's\|^astdbdir.*$$\|astdbdir => $(ASTDBDIR)\|' \

+3

-3

apps/app_chanspy.c less more

844	844	ast_audiohook_unlock(&csth.spy_audiohook);
845	845	ast_audiohook_destroy(&csth.spy_audiohook);
846	846
	847	ast_verb(2, "Done Spying on channel %s\n", name);
	848	publish_chanspy_message(chan, spyee_autochan->chan, 0);
	849
847	850	if (spyee_bridge_autochan) {
848	851	ast_autochan_destroy(spyee_bridge_autochan);
849	852	}
850
851		ast_verb(2, "Done Spying on channel %s\n", name);
852		publish_chanspy_message(chan, NULL, 0);
853	853
854	854	return running;
855	855	}

+22

-1

apps/app_mixmonitor.c less more

861	861	return 0;
862	862	}
863	863
	864	static void mixmonitor_ds_remove_and_free(struct ast_channel chan, const char datastore_id)
	865	{
	866	struct ast_datastore *datastore;
	867
	868	ast_channel_lock(chan);
	869
	870	datastore = ast_channel_datastore_find(chan, &mixmonitor_ds_info, datastore_id);
	871
	872	/*
	873	* Currently the one place this function is called from guarantees a
	874	* datastore is present, thus return checks can be avoided here.
	875	*/
	876	ast_channel_datastore_remove(chan, datastore);
	877	ast_datastore_free(datastore);
	878
	879	ast_channel_unlock(chan);
	880	}
	881
864	882	static int launch_monitor_thread(struct ast_channel chan, const char filename,
865	883	unsigned int flags, int readvol, int writevol,
866	884	const char post_process, const char filename_write,

936	954	pbx_builtin_setvar_helper(chan, uid_channel_var, datastore_id);
937	955	}
938	956	}
939		ast_free(datastore_id);
940	957
941	958	mixmonitor->name = ast_strdup(ast_channel_name(chan));
942	959

986	1003	if (startmon(chan, &mixmonitor->audiohook)) {
987	1004	ast_log(LOG_WARNING, "Unable to add '%s' spy to channel '%s'\n",
988	1005	mixmonitor_spy_type, ast_channel_name(chan));
	1006	mixmonitor_ds_remove_and_free(chan, datastore_id);
	1007	ast_free(datastore_id);
989	1008	ast_autochan_destroy(mixmonitor->autochan);
990	1009	ast_audiohook_destroy(&mixmonitor->audiohook);
991	1010	mixmonitor_free(mixmonitor);
992	1011	return -1;
993	1012	}
	1013
	1014	ast_free(datastore_id);
994	1015
995	1016	/* reference be released at mixmonitor destruction */
996	1017	mixmonitor->callid = ast_read_threadstorage_callid();

+138

-111

apps/app_queue.c less more

9641	9641	}
9642	9642	}
9643	9643
	9644	/! \brief Print a single queue to AMI or the CLI /
	9645	static void print_queue(struct mansession s, int fd, struct call_queue q)
	9646	{
	9647	float sl;
	9648	float sl2;
	9649	struct ao2_iterator mem_iter;
	9650	struct ast_str *out = ast_str_alloca(512);
	9651	time_t now = time(NULL);
	9652
	9653	ast_str_set(&out, 0, "%s has %d calls (max ", q->name, q->count);
	9654	if (q->maxlen) {
	9655	ast_str_append(&out, 0, "%d", q->maxlen);
	9656	} else {
	9657	ast_str_append(&out, 0, "unlimited");
	9658	}
	9659	sl = 0;
	9660	sl2 = 0;
	9661	if (q->callscompleted > 0) {
	9662	sl = 100 * ((float) q->callscompletedinsl / (float) q->callscompleted);
	9663	}
	9664	if (q->callscompleted + q->callsabandoned > 0) {
	9665	sl2 =100 * (((float)q->callsabandonedinsl + (float)q->callscompletedinsl) / ((float)q->callsabandoned + (float)q->callscompleted));
	9666	}
	9667
	9668	ast_str_append(&out, 0, ") in '%s' strategy (%ds holdtime, %ds talktime), W:%d, C:%d, A:%d, SL:%2.1f%%, SL2:%2.1f%% within %ds",
	9669	int2strat(q->strategy), q->holdtime, q->talktime, q->weight, q->callscompleted, q->callsabandoned, sl, sl2, q->servicelevel);
	9670	do_print(s, fd, ast_str_buffer(out));
	9671	if (!ao2_container_count(q->members)) {
	9672	do_print(s, fd, " No Members");
	9673	} else {
	9674	struct member *mem;
	9675
	9676	do_print(s, fd, " Members: ");
	9677	mem_iter = ao2_iterator_init(q->members, 0);
	9678	while ((mem = ao2_iterator_next(&mem_iter))) {
	9679	ast_str_set(&out, 0, " %s", mem->membername);
	9680	if (strcasecmp(mem->membername, mem->interface)) {
	9681	ast_str_append(&out, 0, " (%s", mem->interface);
	9682	if (!ast_strlen_zero(mem->state_interface)
	9683	&& strcmp(mem->state_interface, mem->interface)) {
	9684	ast_str_append(&out, 0, " from %s", mem->state_interface);
	9685	}
	9686	ast_str_append(&out, 0, ")");
	9687	}
	9688	if (mem->penalty) {
	9689	ast_str_append(&out, 0, " with penalty %d", mem->penalty);
	9690	}
	9691
	9692	ast_str_append(&out, 0, " (ringinuse %s)", mem->ringinuse ? "enabled" : "disabled");
	9693
	9694	ast_str_append(&out, 0, "%s%s%s%s%s%s%s%s%s",
	9695	mem->dynamic ? ast_term_color(COLOR_CYAN, COLOR_BLACK) : "", mem->dynamic ? " (dynamic)" : "", ast_term_reset(),
	9696	mem->realtime ? ast_term_color(COLOR_MAGENTA, COLOR_BLACK) : "", mem->realtime ? " (realtime)" : "", ast_term_reset(),
	9697	mem->starttime ? ast_term_color(COLOR_BROWN, COLOR_BLACK) : "", mem->starttime ? " (in call)" : "", ast_term_reset());
	9698
	9699	if (mem->paused) {
	9700	ast_str_append(&out, 0, " %s(paused%s%s was %ld secs ago)%s",
	9701	ast_term_color(COLOR_BROWN, COLOR_BLACK),
	9702	ast_strlen_zero(mem->reason_paused) ? "" : ":",
	9703	ast_strlen_zero(mem->reason_paused) ? "" : mem->reason_paused,
	9704	(long) (now - mem->lastpause),
	9705	ast_term_reset());
	9706	}
	9707
	9708	ast_str_append(&out, 0, " (%s%s%s)",
	9709	ast_term_color(
	9710	mem->status == AST_DEVICE_UNAVAILABLE \|\| mem->status == AST_DEVICE_UNKNOWN ?
	9711	COLOR_RED : COLOR_GREEN, COLOR_BLACK),
	9712	ast_devstate2str(mem->status), ast_term_reset());
	9713	if (mem->calls) {
	9714	ast_str_append(&out, 0, " has taken %d calls (last was %ld secs ago)",
	9715	mem->calls, (long) (now - mem->lastcall));
	9716	} else {
	9717	ast_str_append(&out, 0, " has taken no calls yet");
	9718	}
	9719	do_print(s, fd, ast_str_buffer(out));
	9720	ao2_ref(mem, -1);
	9721	}
	9722	ao2_iterator_destroy(&mem_iter);
	9723	}
	9724	if (!q->head) {
	9725	do_print(s, fd, " No Callers");
	9726	} else {
	9727	struct queue_ent *qe;
	9728	int pos = 1;
	9729
	9730	do_print(s, fd, " Callers: ");
	9731	for (qe = q->head; qe; qe = qe->next) {
	9732	ast_str_set(&out, 0, " %d. %s (wait: %ld:%2.2ld, prio: %d)",
	9733	pos++, ast_channel_name(qe->chan), (long) (now - qe->start) / 60,
	9734	(long) (now - qe->start) % 60, qe->prio);
	9735	do_print(s, fd, ast_str_buffer(out));
	9736	}
	9737	}
	9738	do_print(s, fd, ""); /* blank line between entries */
	9739	}
	9740
	9741	AO2_STRING_FIELD_SORT_FN(call_queue, name);
	9742
9644	9743	/*!
9645	9744	* \brief Show queue(s) status and statistics
9646	9745	*

9651	9750	{
9652	9751	struct call_queue *q;
9653	9752	struct ast_str *out = ast_str_alloca(512);
	9753	struct ao2_container *sorted_queues;
	9754
	9755	struct ao2_iterator queue_iter;
9654	9756	int found = 0;
9655		time_t now = time(NULL);
9656		struct ao2_iterator queue_iter;
9657		struct ao2_iterator mem_iter;
9658	9757
9659	9758	if (argc != 2 && argc != 3) {
9660	9759	return CLI_SHOWUSAGE;

9662	9761
9663	9762	if (argc == 3) { /* specific queue */
9664	9763	if ((q = find_load_queue_rt_friendly(argv[2]))) {
9665		queue_t_unref(q, "Done with temporary pointer");
9666		}
9667		} else if (ast_check_realtime("queues")) {
	9764	ao2_lock(q);
	9765	print_queue(s, fd, q);
	9766	ao2_unlock(q);
	9767	queue_unref(q);
	9768	} else {
	9769	ast_str_set(&out, 0, "No such queue: %s.", argv[2]);
	9770	do_print(s, fd, ast_str_buffer(out));
	9771	}
	9772	return CLI_SUCCESS;
	9773	}
	9774
	9775	if (ast_check_realtime("queues")) {
9668	9776	/* This block is to find any queues which are defined in realtime but
9669	9777	* which have not yet been added to the in-core container
9670	9778	*/

9681	9789	}
9682	9790	}
9683	9791
9684		ao2_lock(queues);
9685		queue_iter = ao2_iterator_init(queues, AO2_ITERATOR_DONTLOCK);
	9792	/*
	9793	* Snapping a copy of the container prevents having to lock both the queues container
	9794	* and the queue itself at the same time. It also allows us to sort the entries.
	9795	*/
	9796	sorted_queues = ao2_container_alloc_rbtree(AO2_ALLOC_OPT_LOCK_NOLOCK, 0, call_queue_sort_fn, NULL);
	9797	if (!sorted_queues) {
	9798	return CLI_SUCCESS;
	9799	}
	9800	if (ao2_container_dup(sorted_queues, queues, 0)) {
	9801	ao2_ref(sorted_queues, -1);
	9802	return CLI_SUCCESS;
	9803	}
	9804
	9805	/*
	9806	* No need to lock the container since it's temporary and static.
	9807	* We also unlink the entries as we use them so the container is
	9808	* empty when the iterator finishes. We can then just unref the container.
	9809	*/
	9810	queue_iter = ao2_iterator_init(sorted_queues, AO2_ITERATOR_DONTLOCK \| AO2_ITERATOR_UNLINK);
9686	9811	while ((q = ao2_t_iterator_next(&queue_iter, "Iterate through queues"))) {
9687		float sl;
9688		float sl2;
9689
9690	9812	struct call_queue *realtime_queue = NULL;
9691
9692	9813	ao2_lock(q);
9693	9814	/* This check is to make sure we don't print information for realtime
9694	9815	* queues which have been deleted from realtime but which have not yet
9695	9816	* been deleted from the in-core container. Only do this if we're not
9696	9817	* looking for a specific queue.
9697	9818	*/
9698		if (argc < 3 && q->realtime) {
	9819	if (q->realtime) {
9699	9820	realtime_queue = find_load_queue_rt_friendly(q->name);
9700	9821	if (!realtime_queue) {
9701	9822	ao2_unlock(q);

9705	9826	queue_t_unref(realtime_queue, "Queue is already in memory");
9706	9827	}
9707	9828
9708		if (argc == 3 && strcasecmp(q->name, argv[2])) {
9709		ao2_unlock(q);
9710		queue_t_unref(q, "Done with iterator");
9711		continue;
9712		}
9713	9829	found = 1;
9714
9715		ast_str_set(&out, 0, "%s has %d calls (max ", q->name, q->count);
9716		if (q->maxlen) {
9717		ast_str_append(&out, 0, "%d", q->maxlen);
9718		} else {
9719		ast_str_append(&out, 0, "unlimited");
9720		}
9721		sl = 0;
9722		sl2 = 0;
9723		if (q->callscompleted > 0) {
9724		sl = 100 * ((float) q->callscompletedinsl / (float) q->callscompleted);
9725		}
9726		if (q->callscompleted + q->callsabandoned > 0) {
9727		sl2 =100 * (((float)q->callsabandonedinsl + (float)q->callscompletedinsl) / ((float)q->callsabandoned + (float)q->callscompleted));
9728		}
9729
9730		ast_str_append(&out, 0, ") in '%s' strategy (%ds holdtime, %ds talktime), W:%d, C:%d, A:%d, SL:%2.1f%%, SL2:%2.1f%% within %ds",
9731		int2strat(q->strategy), q->holdtime, q->talktime, q->weight, q->callscompleted, q->callsabandoned, sl, sl2, q->servicelevel);
9732		do_print(s, fd, ast_str_buffer(out));
9733		if (!ao2_container_count(q->members)) {
9734		do_print(s, fd, " No Members");
9735		} else {
9736		struct member *mem;
9737
9738		do_print(s, fd, " Members: ");
9739		mem_iter = ao2_iterator_init(q->members, 0);
9740		while ((mem = ao2_iterator_next(&mem_iter))) {
9741		ast_str_set(&out, 0, " %s", mem->membername);
9742		if (strcasecmp(mem->membername, mem->interface)) {
9743		ast_str_append(&out, 0, " (%s", mem->interface);
9744		if (!ast_strlen_zero(mem->state_interface)
9745		&& strcmp(mem->state_interface, mem->interface)) {
9746		ast_str_append(&out, 0, " from %s", mem->state_interface);
9747		}
9748		ast_str_append(&out, 0, ")");
9749		}
9750		if (mem->penalty) {
9751		ast_str_append(&out, 0, " with penalty %d", mem->penalty);
9752		}
9753
9754		ast_str_append(&out, 0, " (ringinuse %s)", mem->ringinuse ? "enabled" : "disabled");
9755
9756		ast_str_append(&out, 0, "%s%s%s%s%s%s%s%s%s",
9757		mem->dynamic ? ast_term_color(COLOR_CYAN, COLOR_BLACK) : "", mem->dynamic ? " (dynamic)" : "", ast_term_reset(),
9758		mem->realtime ? ast_term_color(COLOR_MAGENTA, COLOR_BLACK) : "", mem->realtime ? " (realtime)" : "", ast_term_reset(),
9759		mem->starttime ? ast_term_color(COLOR_BROWN, COLOR_BLACK) : "", mem->starttime ? " (in call)" : "", ast_term_reset());
9760
9761		if (mem->paused) {
9762		ast_str_append(&out, 0, " %s(paused%s%s was %ld secs ago)%s",
9763		ast_term_color(COLOR_BROWN, COLOR_BLACK),
9764		ast_strlen_zero(mem->reason_paused) ? "" : ":",
9765		ast_strlen_zero(mem->reason_paused) ? "" : mem->reason_paused,
9766		(long) (now - mem->lastpause),
9767		ast_term_reset());
9768		}
9769
9770		ast_str_append(&out, 0, " (%s%s%s)",
9771		ast_term_color(
9772		mem->status == AST_DEVICE_UNAVAILABLE \|\| mem->status == AST_DEVICE_UNKNOWN ?
9773		COLOR_RED : COLOR_GREEN, COLOR_BLACK),
9774		ast_devstate2str(mem->status), ast_term_reset());
9775		if (mem->calls) {
9776		ast_str_append(&out, 0, " has taken %d calls (last was %ld secs ago)",
9777		mem->calls, (long) (now - mem->lastcall));
9778		} else {
9779		ast_str_append(&out, 0, " has taken no calls yet");
9780		}
9781		do_print(s, fd, ast_str_buffer(out));
9782		ao2_ref(mem, -1);
9783		}
9784		ao2_iterator_destroy(&mem_iter);
9785		}
9786		if (!q->head) {
9787		do_print(s, fd, " No Callers");
9788		} else {
9789		struct queue_ent *qe;
9790		int pos = 1;
9791
9792		do_print(s, fd, " Callers: ");
9793		for (qe = q->head; qe; qe = qe->next) {
9794		ast_str_set(&out, 0, " %d. %s (wait: %ld:%2.2ld, prio: %d)",
9795		pos++, ast_channel_name(qe->chan), (long) (now - qe->start) / 60,
9796		(long) (now - qe->start) % 60, qe->prio);
9797		do_print(s, fd, ast_str_buffer(out));
9798		}
9799		}
9800		do_print(s, fd, ""); /* blank line between entries */
	9830	print_queue(s, fd, q);
	9831
9801	9832	ao2_unlock(q);
9802	9833	queue_t_unref(q, "Done with iterator"); /* Unref the iterator's reference */
9803	9834	}
9804	9835	ao2_iterator_destroy(&queue_iter);
9805		ao2_unlock(queues);
	9836	ao2_ref(sorted_queues, -1);
9806	9837	if (!found) {
9807		if (argc == 3) {
9808		ast_str_set(&out, 0, "No such queue: %s.", argv[2]);
9809		} else {
9810		ast_str_set(&out, 0, "No queues.");
9811		}
	9838	ast_str_set(&out, 0, "No queues.");
9812	9839	do_print(s, fd, ast_str_buffer(out));
9813	9840	}
9814	9841	return CLI_SUCCESS;

+29

-7

apps/app_voicemail.c less more

166	166	<argument name="c" />
167	167	<para>Accept digits for a new extension in context <replaceable>c</replaceable>,
168	168	if played during the greeting. Context defaults to the current context.</para>
	169	</option>
	170	<option name="e">
	171	<para>Play greetings as early media -- only answer the channel just
	172	before accepting the voice message.</para>
169	173	</option>
170	174	<option name="g">
171	175	<argument name="#" required="true" />

659	663	OPT_AUTOPLAY = (1 << 6),
660	664	OPT_DTMFEXIT = (1 << 7),
661	665	OPT_MESSAGE_Urgent = (1 << 8),
662		OPT_MESSAGE_PRIORITY = (1 << 9)
	666	OPT_MESSAGE_PRIORITY = (1 << 9),
	667	OPT_EARLYM_GREETING = (1 << 10)
663	668	};
664	669
665	670	enum vm_option_args {

685	690	AST_APP_OPTION('p', OPT_PREPEND_MAILBOX),
686	691	AST_APP_OPTION_ARG('a', OPT_AUTOPLAY, OPT_ARG_PLAYFOLDER),
687	692	AST_APP_OPTION('U', OPT_MESSAGE_Urgent),
688		AST_APP_OPTION('P', OPT_MESSAGE_PRIORITY)
	693	AST_APP_OPTION('P', OPT_MESSAGE_PRIORITY),
	694	AST_APP_OPTION('e', OPT_EARLYM_GREETING)
689	695	});
690	696
691	697	static const char * const mailbox_folders[] = {

3932	3938	char fn[PATH_MAX];
3933	3939	char full_fn[PATH_MAX];
3934	3940	char msgnums[80];
	3941	char msg_id[MSG_ID_LEN] = "";
3935	3942	char *argv[] = { dir, msgnums };
3936	3943	struct generic_prepare_struct gps = { .sql = sql, .argc = 2, .argv = argv };
3937	3944	struct odbc_obj *obj;

4036	4043	} else {
4037	4044	res = SQLGetData(stmt, x + 1, SQL_CHAR, rowdata, sizeof(rowdata), NULL);
4038	4045	if (res == SQL_NULL_DATA && !strcasecmp(coltitle, "msg_id")) {
4039		char msg_id[MSG_ID_LEN];
	4046	/* Generate msg_id now, but don't store it until we're done with this
	4047	connection */
4040	4048	generate_msg_id(msg_id);
4041	4049	snprintf(rowdata, sizeof(rowdata), "%s", msg_id);
4042		odbc_update_msg_id(dir, msgnum, msg_id);
4043	4050	} else if (res == SQL_NULL_DATA && !strcasecmp(coltitle, "category")) {
4044	4051	/* Ignore null column value for category */
4045	4052	ast_debug(3, "Ignoring null category column in ODBC voicemail retrieve_file.\n");

4064	4071	close(fd);
4065	4072
4066	4073	ast_odbc_release_obj(obj);
	4074
	4075	/* If res_odbc is configured to only allow a single database connection, we
	4076	will deadlock if we try to do this before releasing the connection we
	4077	were just using. */
	4078	if (!ast_strlen_zero(msg_id)) {
	4079	odbc_update_msg_id(dir, msgnum, msg_id);
	4080	}
4067	4081
4068	4082	return x - 1;
4069	4083	}

6958	6972	return -1;
6959	6973	}
6960	6974	/* The meat of recording the message... All the announcements and beeps have been played*/
	6975	if (ast_channel_state(chan) != AST_STATE_UP) {
	6976	ast_answer(chan);
	6977	}
6961	6978	ast_copy_string(fmt, vmfmts, sizeof(fmt));
6962	6979	if (!ast_strlen_zero(fmt)) {
6963	6980	char msg_id[MSG_ID_LEN] = "";

12363	12380
12364	12381	memset(&leave_options, 0, sizeof(leave_options));
12365	12382
12366		if (ast_channel_state(chan) != AST_STATE_UP)
12367		ast_answer(chan);
12368
12369	12383	if (!ast_strlen_zero(data)) {
12370	12384	tmp = ast_strdupa(data);
12371	12385	AST_STANDARD_APP_ARGS(args, tmp);

12396	12410	if (ast_strlen_zero(temp))
12397	12411	return 0;
12398	12412	args.argv0 = ast_strdupa(temp);
	12413	}
	12414
	12415	if (ast_channel_state(chan) != AST_STATE_UP) {
	12416	if (ast_test_flag(&flags, OPT_EARLYM_GREETING)) {
	12417	ast_indicate(chan, AST_CONTROL_PROGRESS);
	12418	} else {
	12419	ast_answer(chan);
	12420	}
12399	12421	}
12400	12422
12401	12423	res = leave_voicemail(chan, args.argv0, &leave_options);

+0

-27

~~asterisk-16.15.1-summary.html~~ less more

0		<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"><html xmlns="http://www.w3.org/1999/xhtml"><title>Release Summary - asterisk-16.15.1</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-16.15.1</h3><h3 align="center">Date: 2020-12-22</h3><h3 align="center"><asteriskteam@digium.com></h3><hr><h2 align="center">Table of Contents</h2><ol>
1		<li><a href="#summary">Summary</a></li>
2		<li><a href="#contributors">Contributors</a></li>
3		<li><a href="#open_issues">Open Issues</a></li>
4		<li><a href="#commits">Other Changes</a></li>
5		<li><a href="#diffstat">Diffstat</a></li>
6		</ol><hr><a name="summary"><h2 align="center">Summary</h2></a><center><a href="#top">[Back to Top]</a></center><p>This release has been made to address one or more security vulnerabilities that have been identified. A security advisory document has been published for each vulnerability that includes additional information. Users of versions of Asterisk that are affected are strongly encouraged to review the advisories and determine what action they should take to protect their systems from these issues.</p><p>Security Advisories:</p><ul>
7		<li><a href="http://downloads.asterisk.org/pub/security/AST-2020-003,AST-2020-004.html">AST-2020-003,AST-2020-004</a></li>
8		</ul><p>The data in this summary reflects changes that have been made since the previous release, asterisk-16.15.0.</p><hr><a name="contributors"><h2 align="center">Contributors</h2></a><center><a href="#top">[Back to Top]</a></center><p>This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were affected by commits that went into this release.</p><table width="100%" border="0">
9		<tr><th width="33%">Coders</th><th width="33%">Testers</th><th width="33%">Reporters</th></tr>
10		<tr valign="top"><td width="33%">1 Torrey Searle <tsearle@voxbone.com><br/>1 Asterisk Development Team <asteriskteam@digium.com><br/></td><td width="33%"><td width="33%">1 Mikhail Ivanov <mivanov@lanta-net.ru><br/>1 Torrey Searle <tsearle@gmail.com><br/></td></tr>
11		</table><hr><a name="open_issues"><h2 align="center">Open Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all open issues from the issue tracker that were referenced by changes that went into this release.</p><h3>Security</h3><h4>Category: Resources/res_pjsip_diversion</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29219">ASTERISK-29219</a>: res_pjsip_diversion: Crash if Tel URI contains History-Info<br/>Reported by: Torrey Searle<ul>
12		<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=354049e055a2c0162d510f2c75d1580f80d0d6e6">[354049e055]</a> Torrey Searle -- res/res_pjsip_diversion: prevent crash on tel: uri in History-Info</li>
13		</ul><br><h3>Bug</h3><h4>Category: Resources/res_pjsip_diversion</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29191">ASTERISK-29191</a>: tel: URI in Diversion header causes crash<br/>Reported by: Mikhail Ivanov<ul>
14		<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=354049e055a2c0162d510f2c75d1580f80d0d6e6">[354049e055]</a> Torrey Searle -- res/res_pjsip_diversion: prevent crash on tel: uri in History-Info</li>
15		</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29191">ASTERISK-29191</a>: tel: URI in Diversion header causes crash<br/>Reported by: Mikhail Ivanov<ul>
16		<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=354049e055a2c0162d510f2c75d1580f80d0d6e6">[354049e055]</a> Torrey Searle -- res/res_pjsip_diversion: prevent crash on tel: uri in History-Info</li>
17		</ul><br><hr><a name="commits"><h2 align="center">Commits Not Associated with an Issue</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all changes that went into this release that did not reference a JIRA issue.</p><table width="100%" border="1">
18		<tr><th>Revision</th><th>Author</th><th>Summary</th></tr>
19		<tr><td><a href="https://code.asterisk.org/code/changelog/asterisk?cs=49fbd5788924f4d3d05165194b380ade69cacafb">49fbd57889</a></td><td>Asterisk Development Team</td><td>Update for 16.15.1</td></tr>
20		</table><hr><a name="diffstat"><h2 align="center">Diffstat Results</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p><pre>asterisk-16.15.0-summary.html \| 213 ---------------
21		asterisk-16.15.0-summary.txt \| 543 ----------------------------------------
22		b/.version \| 2
23		b/ChangeLog \| 16 +
24		b/asterisk-16.15.1-summary.html \| 17 +
25		b/asterisk-16.15.1-summary.txt \| 47 +++
26		6 files changed, 81 insertions(+), 757 deletions(-)</pre><br></html>⏎

+0

-123

~~asterisk-16.15.1-summary.txt~~ less more

0		Release Summary
1
2		asterisk-16.15.1
3
4		Date: 2020-12-22
5
6		<asteriskteam@digium.com>
7
8		----------------------------------------------------------------------
9
10		Table of Contents
11
12		1. Summary
13		2. Contributors
14		3. Open Issues
15		4. Other Changes
16		5. Diffstat
17
18		----------------------------------------------------------------------
19
20		Summary
21
22		[Back to Top]
23
24		This release has been made to address one or more security vulnerabilities
25		that have been identified. A security advisory document has been published
26		for each vulnerability that includes additional information. Users of
27		versions of Asterisk that are affected are strongly encouraged to review
28		the advisories and determine what action they should take to protect their
29		systems from these issues.
30
31		Security Advisories:
32
33		* AST-2020-003,AST-2020-004
34
35		The data in this summary reflects changes that have been made since the
36		previous release, asterisk-16.15.0.
37
38		----------------------------------------------------------------------
39
40		Contributors
41
42		[Back to Top]
43
44		This table lists the people who have submitted code, those that have
45		tested patches, as well as those that reported issues on the issue tracker
46		that were resolved in this release. For coders, the number is how many of
47		their patches (of any size) were committed into this release. For testers,
48		the number is the number of times their name was listed as assisting with
49		testing a patch. Finally, for reporters, the number is the number of
50		issues that they reported that were affected by commits that went into
51		this release.
52
53		Coders Testers Reporters
54		1 Torrey Searle 1 Mikhail Ivanov
55		1 Asterisk Development Team 1 Torrey Searle
56
57		----------------------------------------------------------------------
58
59		Open Issues
60
61		[Back to Top]
62
63		This is a list of all open issues from the issue tracker that were
64		referenced by changes that went into this release.
65
66		Security
67
68		Category: Resources/res_pjsip_diversion
69
70		ASTERISK-29219: res_pjsip_diversion: Crash if Tel URI contains
71		History-Info
72		Reported by: Torrey Searle
73		* [354049e055] Torrey Searle -- res/res_pjsip_diversion: prevent crash
74		on tel: uri in History-Info
75
76		Bug
77
78		Category: Resources/res_pjsip_diversion
79
80		ASTERISK-29191: tel: URI in Diversion header causes crash
81		Reported by: Mikhail Ivanov
82		* [354049e055] Torrey Searle -- res/res_pjsip_diversion: prevent crash
83		on tel: uri in History-Info
84
85		Category: pjproject/pjsip
86
87		ASTERISK-29191: tel: URI in Diversion header causes crash
88		Reported by: Mikhail Ivanov
89		* [354049e055] Torrey Searle -- res/res_pjsip_diversion: prevent crash
90		on tel: uri in History-Info
91
92		----------------------------------------------------------------------
93
94		Commits Not Associated with an Issue
95
96		[Back to Top]
97
98		This is a list of all changes that went into this release that did not
99		reference a JIRA issue.
100
101		+------------------------------------------------------------------------+
102		\| Revision \| Author \| Summary \|
103		\|----------------+-------------------------------+-----------------------\|
104		\| 49fbd57889 \| Asterisk Development Team \| Update for 16.15.1 \|
105		+------------------------------------------------------------------------+
106
107		----------------------------------------------------------------------
108
109		Diffstat Results
110
111		[Back to Top]
112
113		This is a summary of the changes to the source code that went into this
114		release that was generated using the diffstat utility.
115
116		asterisk-16.15.0-summary.html \| 213 ---------------
117		asterisk-16.15.0-summary.txt \| 543 ----------------------------------------
118		b/.version \| 2
119		b/ChangeLog \| 16 +
120		b/asterisk-16.15.1-summary.html \| 17 +
121		b/asterisk-16.15.1-summary.txt \| 47 +++
122		6 files changed, 81 insertions(+), 757 deletions(-)

+34

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asterisk-16.16.1-summary.html less more

	0	<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd"><html xmlns="http://www.w3.org/1999/xhtml"><title>Release Summary - asterisk-16.16.1</title><h1 align="center"><a name="top">Release Summary</a></h1><h3 align="center">asterisk-16.16.1</h3><h3 align="center">Date: 2021-02-18</h3><h3 align="center"><asteriskteam@digium.com></h3><hr><h2 align="center">Table of Contents</h2><ol>
	1	<li><a href="#summary">Summary</a></li>
	2	<li><a href="#contributors">Contributors</a></li>
	3	<li><a href="#closed_issues">Closed Issues</a></li>
	4	<li><a href="#diffstat">Diffstat</a></li>
	5	</ol><hr><a name="summary"><h2 align="center">Summary</h2></a><center><a href="#top">[Back to Top]</a></center><p>This release has been made to address one or more security vulnerabilities that have been identified. A security advisory document has been published for each vulnerability that includes additional information. Users of versions of Asterisk that are affected are strongly encouraged to review the advisories and determine what action they should take to protect their systems from these issues.</p><p>Security Advisories:</p><ul>
	6	<li><a href="http://downloads.asterisk.org/pub/security/AST-2021-001,AST-2021-002,AST-2021-003,AST-2021-004,AST-2021-005.html">AST-2021-001,AST-2021-002,AST-2021-003,AST-2021-004,AST-2021-005</a></li>
	7	</ul><p>The data in this summary reflects changes that have been made since the previous release, asterisk-16.16.0.</p><hr><a name="contributors"><h2 align="center">Contributors</h2></a><center><a href="#top">[Back to Top]</a></center><p>This table lists the people who have submitted code, those that have tested patches, as well as those that reported issues on the issue tracker that were resolved in this release. For coders, the number is how many of their patches (of any size) were committed into this release. For testers, the number is the number of times their name was listed as assisting with testing a patch. Finally, for reporters, the number is the number of issues that they reported that were affected by commits that went into this release.</p><table width="100%" border="0">
	8	<tr><th width="33%">Coders</th><th width="33%">Testers</th><th width="33%">Reporters</th></tr>
	9	<tr valign="top"><td width="33%">1 Ivan Poddubnyi <ivan.poddubny@gmail.com><br/>1 Sean Bright <sean.bright@gmail.com><br/>1 Kevin Harwell <kharwell@sangoma.com><br/>1 Alexander Traud <pabstraud@compuserve.com><br/>1 Joshua C. Colp <jcolp@sangoma.com><br/></td><td width="33%"><td width="33%">1 Mauri de Souza Meneguzzo (3CPlus) <mauri.nunes@fluxoti.com><br/>1 Ivan Poddubny<br/>1 Ivan Poddubny <ivan.poddubny@gmail.com><br/>1 Edvin Vidmar <edvinvidmar@hotmail.com><br/>1 Alexander Traud <pabstraud@compuserve.com><br/>1 Gregory Massel <greg@csurf.co.za><br/>1 Alexander Traud<br/></td></tr>
	10	</table><hr><a name="closed_issues"><h2 align="center">Closed Issues</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a list of all issues from the issue tracker that were closed by changes that went into this release.</p><h3>Security</h3><h4>Category: Resources/res_srtp</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29260">ASTERISK-29260</a>: sRTP Replay Protection ignored; even tears down long calls<br/>Reported by: Alexander Traud<ul>
	11	<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=3f4dfd5c022c82265cbaba7fa726d1e27dd21cce">[3f4dfd5c02]</a> Alexander Traud -- rtp: Enable srtp replay protection</li>
	12	</ul><br><h4>Category: pjproject/pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29227">ASTERISK-29227</a>: res_pjsip_diversion: sending multiple 181 responses causes memory corruption and crash<br/>Reported by: Ivan Poddubny<ul>
	13	<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=17561b5e643092b64a1e8351b25017b28ed690b4">[17561b5e64]</a> Ivan Poddubnyi -- res_pjsip_diversion: Fix adding more than one histinfo to Supported</li>
	14	</ul><br><h3>Bug</h3><h4>Category: Resources/res_pjsip</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29196">ASTERISK-29196</a>: res_pjsip: Segmentation fault<br/>Reported by: Mauri de Souza Meneguzzo (3CPlus)<ul>
	15	<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=321632b02e828406d45bc4be2793399f1746c32a">[321632b02e]</a> Joshua C. Colp -- pjsip: Make modify_local_offer2 tolerate previous failed SDP.</li>
	16	</ul><br><h4>Category: Resources/res_pjsip_session</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29203">ASTERISK-29203</a>: res_pjsip_t38: Crash when changing state<br/>Reported by: Gregory Massel<ul>
	17	<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a5619097cd7e4da4d64a513d9659871ea217a706">[a5619097cd]</a> Kevin Harwell -- AST-2021-002: Remote crash possible when negotiating T.38</li>
	18	</ul><br><h4>Category: Resources/res_pjsip_t38</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29203">ASTERISK-29203</a>: res_pjsip_t38: Crash when changing state<br/>Reported by: Gregory Massel<ul>
	19	<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=a5619097cd7e4da4d64a513d9659871ea217a706">[a5619097cd]</a> Kevin Harwell -- AST-2021-002: Remote crash possible when negotiating T.38</li>
	20	</ul><br><h4>Category: Resources/res_rtp_asterisk</h4><a href="https://issues.asterisk.org/jira/browse/ASTERISK-29205">ASTERISK-29205</a>: res_rtp_asterisk: Asterisk crashes when making hold/unhold from webrtc client<br/>Reported by: Edvin Vidmar<ul>
	21	<li><a href="https://code.asterisk.org/code/changelog/asterisk?cs=4cea145aa9a7dc90eabdd1f63cb3162f3e1d91e8">[4cea145aa9]</a> Sean Bright -- res_rtp_asterisk.c: Fix signed mismatch that leads to overflow</li>
	22	</ul><br><hr><a name="diffstat"><h2 align="center">Diffstat Results</h2></a><center><a href="#top">[Back to Top]</a></center><p>This is a summary of the changes to the source code that went into this release that was generated using the diffstat utility.</p><pre>configs/samples/rtp.conf.sample \| 12 +++++++
	23	doc/CHANGES-staging/srtp_replay_protection.txt \| 9 +++++
	24	doc/UPGRADE-staging/srtp_replay_protection.txt \| 9 +++++
	25	res/res_pjsip_diversion.c \| 14 ++++++++
	26	res/res_pjsip_outbound_registration.c \| 12 +++++++
	27	res/res_pjsip_path.c \| 12 +++++++
	28	res/res_pjsip_session.c \| 9 +++++
	29	res/res_pjsip_t38.c \| 9 +++++
	30	res/res_rtp_asterisk.c \| 16 +++++++---
	31	res/res_srtp.c \| 5 +--
	32	third-party/pjproject/patches/0080-fix-sdp-neg-modify-local-offer.patch \| 15 +++++++++
	33	11 files changed, 115 insertions(+), 7 deletions(-)</pre><br></html>⏎

+138

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asterisk-16.16.1-summary.txt less more

	0	Release Summary
	1
	2	asterisk-16.16.1
	3
	4	Date: 2021-02-18
	5
	6	<asteriskteam@digium.com>
	7
	8	----------------------------------------------------------------------
	9
	10	Table of Contents
	11
	12	1. Summary
	13	2. Contributors
	14	3. Closed Issues
	15	4. Diffstat
	16
	17	----------------------------------------------------------------------
	18
	19	Summary
	20
	21	[Back to Top]
	22
	23	This release has been made to address one or more security vulnerabilities
	24	that have been identified. A security advisory document has been published
	25	for each vulnerability that includes additional information. Users of
	26	versions of Asterisk that are affected are strongly encouraged to review
	27	the advisories and determine what action they should take to protect their
	28	systems from these issues.
	29
	30	Security Advisories:
	31
	32	* AST-2021-001,AST-2021-002,AST-2021-003,AST-2021-004,AST-2021-005
	33
	34	The data in this summary reflects changes that have been made since the
	35	previous release, asterisk-16.16.0.
	36
	37	----------------------------------------------------------------------
	38
	39	Contributors
	40
	41	[Back to Top]
	42
	43	This table lists the people who have submitted code, those that have
	44	tested patches, as well as those that reported issues on the issue tracker
	45	that were resolved in this release. For coders, the number is how many of
	46	their patches (of any size) were committed into this release. For testers,
	47	the number is the number of times their name was listed as assisting with
	48	testing a patch. Finally, for reporters, the number is the number of
	49	issues that they reported that were affected by commits that went into
	50	this release.
	51
	52	Coders Testers Reporters
	53	1 Ivan Poddubnyi 1 Mauri de Souza Meneguzzo (3CPlus)
	54	1 Sean Bright 1 Ivan Poddubny
	55	1 Kevin Harwell 1 Ivan Poddubny
	56	1 Alexander Traud 1 Edvin Vidmar
	57	1 Joshua C. Colp 1 Alexander Traud
	58	1 Gregory Massel
	59	1 Alexander Traud
	60
	61	----------------------------------------------------------------------
	62
	63	Closed Issues
	64
	65	[Back to Top]
	66
	67	This is a list of all issues from the issue tracker that were closed by
	68	changes that went into this release.
	69
	70	Security
	71
	72	Category: Resources/res_srtp
	73
	74	ASTERISK-29260: sRTP Replay Protection ignored; even tears down long calls
	75	Reported by: Alexander Traud
	76	* [3f4dfd5c02] Alexander Traud -- rtp: Enable srtp replay protection
	77
	78	Category: pjproject/pjsip
	79
	80	ASTERISK-29227: res_pjsip_diversion: sending multiple 181 responses causes
	81	memory corruption and crash
	82	Reported by: Ivan Poddubny
	83	* [17561b5e64] Ivan Poddubnyi -- res_pjsip_diversion: Fix adding more
	84	than one histinfo to Supported
	85
	86	Bug
	87
	88	Category: Resources/res_pjsip
	89
	90	ASTERISK-29196: res_pjsip: Segmentation fault
	91	Reported by: Mauri de Souza Meneguzzo (3CPlus)
	92	* [321632b02e] Joshua C. Colp -- pjsip: Make modify_local_offer2
	93	tolerate previous failed SDP.
	94
	95	Category: Resources/res_pjsip_session
	96
	97	ASTERISK-29203: res_pjsip_t38: Crash when changing state
	98	Reported by: Gregory Massel
	99	* [a5619097cd] Kevin Harwell -- AST-2021-002: Remote crash possible when
	100	negotiating T.38
	101
	102	Category: Resources/res_pjsip_t38
	103
	104	ASTERISK-29203: res_pjsip_t38: Crash when changing state
	105	Reported by: Gregory Massel
	106	* [a5619097cd] Kevin Harwell -- AST-2021-002: Remote crash possible when
	107	negotiating T.38
	108
	109	Category: Resources/res_rtp_asterisk
	110
	111	ASTERISK-29205: res_rtp_asterisk: Asterisk crashes when making hold/unhold
	112	from webrtc client
	113	Reported by: Edvin Vidmar
	114	* [4cea145aa9] Sean Bright -- res_rtp_asterisk.c: Fix signed mismatch
	115	that leads to overflow
	116
	117	----------------------------------------------------------------------
	118
	119	Diffstat Results
	120
	121	[Back to Top]
	122
	123	This is a summary of the changes to the source code that went into this
	124	release that was generated using the diffstat utility.
	125
	126	configs/samples/rtp.conf.sample \| 12 +++++++
	127	doc/CHANGES-staging/srtp_replay_protection.txt \| 9 +++++
	128	doc/UPGRADE-staging/srtp_replay_protection.txt \| 9 +++++
	129	res/res_pjsip_diversion.c \| 14 ++++++++
	130	res/res_pjsip_outbound_registration.c \| 12 +++++++
	131	res/res_pjsip_path.c \| 12 +++++++
	132	res/res_pjsip_session.c \| 9 +++++
	133	res/res_pjsip_t38.c \| 9 +++++
	134	res/res_rtp_asterisk.c \| 16 +++++++---
	135	res/res_srtp.c \| 5 +--
	136	third-party/pjproject/patches/0080-fix-sdp-neg-modify-local-offer.patch \| 15 +++++++++
	137	11 files changed, 115 insertions(+), 7 deletions(-)

+1

-0

build_tools/install_subst less more

25	25	src="$1"
26	26	dst="$2"
27	27	sed <"$src" \
	28	-e "s\|__ASTERISK_CACHE_DIR__\|$ASTCACHEDIR\|g" \
28	29	-e "s\|__ASTERISK_DATA_DIR__\|$ASTDATADIR\|g" \
29	30	-e "s\|__ASTERISK_DB_DIR__\|$ASTDBDIR\|g" \
30	31	-e "s\|__ASTERISK_ETC_DIR__\|$ASTETCDIR\|g" \

+1

-0

build_tools/make_defaults_h less more

6	6	*/
7	7	#define DEFAULT_CONFIG_FILE "${INSTALL_PATH}${ASTCONFPATH}"
8	8
	9	#define DEFAULT_CACHE_DIR "${INSTALL_PATH}${ASTCACHEDIR}"
9	10	#define DEFAULT_CONFIG_DIR "${INSTALL_PATH}${ASTETCDIR}"
10	11	#define DEFAULT_MODULE_DIR "${INSTALL_PATH}${ASTMODDIR}"
11	12	#define DEFAULT_AGI_DIR "${INSTALL_PATH}${AGI_DIR}"

+1

-0

build_tools/mkpkgconfig less more

31	31	cat <<EOF > "$PPATH/asterisk.pc"
32	32	install_prefix=$INSTALL_PREFIX
33	33	version_number=$ASTERISKVERSIONNUM
	34	cachedir=$ASTCACHEDIR
34	35	etcdir=$ASTETCDIR
35	36	libdir=$ASTLIBDIR
36	37	varlibdir=$ASTVARLIBDIR

+69

-149

channels/chan_pjsip.c less more

136	136	static void chan_pjsip_session_end(struct ast_sip_session *session);
137	137	static int chan_pjsip_incoming_request(struct ast_sip_session session, struct pjsip_rx_data rdata);
138	138	static void chan_pjsip_incoming_response(struct ast_sip_session session, struct pjsip_rx_data rdata);
	139	static void chan_pjsip_incoming_response_update_cause(struct ast_sip_session session, struct pjsip_rx_data rdata);
139	140
140	141	/! \brief SIP session supplement structure /
141	142	static struct ast_sip_session_supplement chan_pjsip_supplement = {

144	145	.session_begin = chan_pjsip_session_begin,
145	146	.session_end = chan_pjsip_session_end,
146	147	.incoming_request = chan_pjsip_incoming_request,
	148	.incoming_response = chan_pjsip_incoming_response,
147	149	/* It is important that this supplement runs after media has been negotiated */
148	150	.response_priority = AST_SIP_SESSION_AFTER_MEDIA,
149	151	};

152	154	static struct ast_sip_session_supplement chan_pjsip_supplement_response = {
153	155	.method = "INVITE",
154	156	.priority = AST_SIP_SUPPLEMENT_PRIORITY_CHANNEL,
155		.incoming_response = chan_pjsip_incoming_response,
	157	.incoming_response = chan_pjsip_incoming_response_update_cause,
156	158	.response_priority = AST_SIP_SESSION_BEFORE_MEDIA \| AST_SIP_SESSION_AFTER_MEDIA,
157	159	};
158	160

672	674	ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
673	675	session->inv_session->cause,
674	676	pjsip_get_status_text(session->inv_session->cause)->ptr);
675		#ifdef HAVE_PJSIP_INV_SESSION_REF
676		pjsip_inv_dec_ref(session->inv_session);
677		#endif
678	677	return 0;
679	678	}
680	679

690	689	if (status == PJ_SUCCESS && packet) {
691	690	ast_sip_session_send_response(session, packet);
692	691	}
693
694		#ifdef HAVE_PJSIP_INV_SESSION_REF
695		pjsip_inv_dec_ref(session->inv_session);
696		#endif
697	692
698	693	if (status != PJ_SUCCESS) {
699	694	char err[PJ_ERR_MSG_SIZE];

723	718
724	719	ast_setstate(ast, AST_STATE_UP);
725	720	session = ao2_bump(channel->session);
726
727		#ifdef HAVE_PJSIP_INV_SESSION_REF
728		if (pjsip_inv_add_ref(session->inv_session) != PJ_SUCCESS) {
729		ast_log(LOG_ERROR, "Can't increase the session reference counter\n");
730		ao2_ref(session, -1);
731		return -1;
732		}
733		#endif
734	721
735	722	/* the answer task needs to be pushed synchronously otherwise a race condition
736	723	can occur between this thread and bridging (specifically when native bridging

741	728	if (res == -1) {
742	729	ast_log(LOG_ERROR,"Cannot answer '%s': Unable to push answer task to the threadpool.\n",
743	730	ast_channel_name(session->channel));
744		#ifdef HAVE_PJSIP_INV_SESSION_REF
745		pjsip_inv_dec_ref(session->inv_session);
746		#endif
747	731	}
748	732	ao2_ref(session, -1);
749	733	ast_channel_lock(ast);

1342	1326	ast_sip_session_send_response(session, packet);
1343	1327	}
1344	1328
1345		#ifdef HAVE_PJSIP_INV_SESSION_REF
1346		pjsip_inv_dec_ref(session->inv_session);
1347		#endif
1348	1329	ao2_ref(ind_data, -1);
1349	1330
1350	1331	return 0;

1376	1357	ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
1377	1358	session->inv_session->cause,
1378	1359	pjsip_get_status_text(session->inv_session->cause)->ptr);
1379		goto failure;
	1360	return -1;
1380	1361	}
1381	1362
1382	1363	if (ast_sip_create_request("INFO", session->inv_session->dlg, session->endpoint, NULL, NULL, &tdata)) {
1383	1364	ast_log(LOG_ERROR, "Could not create text video update INFO request\n");
1384		goto failure;
	1365	return -1;
1385	1366	}
1386	1367	if (ast_sip_add_body(tdata, &body)) {
1387	1368	ast_log(LOG_ERROR, "Could not add body to text video update INFO request\n");
1388		goto failure;
	1369	return -1;
1389	1370	}
1390	1371	ast_sip_session_send_request(session, tdata);
1391	1372
1392		#ifdef HAVE_PJSIP_INV_SESSION_REF
1393		pjsip_inv_dec_ref(session->inv_session);
1394		#endif
1395
1396	1373	return 0;
1397
1398		failure:
1399		#ifdef HAVE_PJSIP_INV_SESSION_REF
1400		pjsip_inv_dec_ref(session->inv_session);
1401		#endif
1402		return -1;
1403
1404	1374	}
1405	1375
1406	1376	/*!

1448	1418	ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
1449	1419	session->inv_session->cause,
1450	1420	pjsip_get_status_text(session->inv_session->cause)->ptr);
1451		#ifdef HAVE_PJSIP_INV_SESSION_REF
1452		pjsip_inv_dec_ref(session->inv_session);
1453		#endif
1454	1421	ao2_ref(session, -1);
1455	1422	return -1;
1456	1423	}

1490	1457	}
1491	1458	}
1492	1459	}
1493
1494		#ifdef HAVE_PJSIP_INV_SESSION_REF
1495		pjsip_inv_dec_ref(session->inv_session);
1496		#endif
1497	1460
1498	1461	ao2_ref(session, -1);
1499	1462	return 0;

1725	1688	res = ast_rtp_instance_write(media->rtp, &fr);
1726	1689	} else {
1727	1690	ao2_ref(channel->session, +1);
1728		#ifdef HAVE_PJSIP_INV_SESSION_REF
1729		if (pjsip_inv_add_ref(channel->session->inv_session) != PJ_SUCCESS) {
1730		ast_log(LOG_ERROR, "Can't increase the session reference counter\n");
	1691	if (ast_sip_push_task(channel->session->serializer, transmit_info_with_vidupdate, channel->session)) {
1731	1692	ao2_cleanup(channel->session);
1732		} else {
1733		#endif
1734		if (ast_sip_push_task(channel->session->serializer, transmit_info_with_vidupdate, channel->session)) {
1735		ao2_cleanup(channel->session);
1736		}
1737		#ifdef HAVE_PJSIP_INV_SESSION_REF
1738	1693	}
1739		#endif
1740	1694	}
1741	1695	ast_test_suite_event_notify("AST_CONTROL_VIDUPDATE", "Result: Success");
1742	1696	} else {

1750	1704	break;
1751	1705	case AST_CONTROL_CONNECTED_LINE:
1752	1706	ao2_ref(channel->session, +1);
1753		#ifdef HAVE_PJSIP_INV_SESSION_REF
1754		if (pjsip_inv_add_ref(channel->session->inv_session) != PJ_SUCCESS) {
1755		ao2_cleanup(channel->session);
1756		SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Couldn't increase the session reference counter\n",
1757		ast_channel_name(ast));
1758		}
1759		#endif
1760	1707	if (ast_sip_push_task(channel->session->serializer, update_connected_line_information, channel->session)) {
1761		#ifdef HAVE_PJSIP_INV_SESSION_REF
1762		pjsip_inv_dec_ref(channel->session->inv_session);
1763		#endif
1764	1708	ao2_cleanup(channel->session);
1765	1709	}
1766	1710	break;

1867	1811	if (!ind_data) {
1868	1812	SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Couldn't alloc indicate data\n", ast_channel_name(ast));
1869	1813	}
1870		#ifdef HAVE_PJSIP_INV_SESSION_REF
1871		if (pjsip_inv_add_ref(ind_data->session->inv_session) != PJ_SUCCESS) {
1872		ao2_cleanup(ind_data);
1873		SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Couldn't increase the session reference counter\n", ast_channel_name(ast));
1874		}
1875		#endif
	1814
1876	1815	if (ast_sip_push_task(channel->session->serializer, indicate, ind_data)) {
1877	1816	ast_log(LOG_ERROR, "%s: Cannot send response code %d to endpoint %s. Could not queue task properly\n",
1878	1817	ast_channel_name(ast), response_code, ast_sorcery_object_get_id(channel->session->endpoint));
1879		#ifdef HAVE_PJSIP_INV_SESSION_REF
1880		pjsip_inv_dec_ref(ind_data->session->inv_session);
1881		#endif
1882	1818	ao2_cleanup(ind_data);
1883	1819	res = -1;
1884	1820	}

1971	1907	*/
1972	1908	static void xfer_client_on_evsub_state(pjsip_evsub sub, pjsip_event event)
1973	1909	{
1974		struct ast_sip_session *session;
1975		struct ast_channel *chan = NULL;
	1910	struct ast_channel *chan;
1976	1911	enum ast_control_transfer message = AST_TRANSFER_SUCCESS;
1977	1912	int res = 0;
1978	1913

1980	1915	return;
1981	1916	}
1982	1917
1983		session = pjsip_evsub_get_mod_data(sub, refer_callback_module.id);
1984		if (!session) {
1985		return;
1986		}
1987
1988		chan = session->channel;
	1918	chan = pjsip_evsub_get_mod_data(sub, refer_callback_module.id);
1989	1919	if (!chan) {
1990	1920	return;
1991	1921	}

2011	1941	*/
2012	1942	if (refer_sub && !pj_stricmp2(&refer_sub->hvalue, "false")) {
2013	1943	/* Since no subscription is desired, assume that call has been transferred successfully. */
	1944	/* Channel reference will be released at end of function */
2014	1945	/* Terminate subscription. */
	1946	pjsip_evsub_set_mod_data(sub, refer_callback_module.id, NULL);
2015	1947	pjsip_evsub_terminate(sub, PJ_TRUE);
2016	1948	res = -1;
2017	1949	}

2076	2008	}
2077	2009
2078	2010	if (res) {
2079		ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
	2011	ast_queue_control_data(chan, AST_CONTROL_TRANSFER, &message, sizeof(message));
	2012	ao2_ref(chan, -1);
2080	2013	}
2081	2014	}
2082	2015

2089	2022	const char *ref_by_val;
2090	2023	char local_info[pj_strlen(&session->inv_session->dlg->local.info_str) + 1];
2091	2024	struct pjsip_evsub_user xfer_cb;
	2025	struct ast_channel *chan = session->channel;
2092	2026
2093	2027	pj_bzero(&xfer_cb, sizeof(xfer_cb));
2094	2028	xfer_cb.on_evsub_state = &xfer_client_on_evsub_state;
2095	2029
2096	2030	if (pjsip_xfer_create_uac(session->inv_session->dlg, &xfer_cb, &sub) != PJ_SUCCESS) {
2097	2031	message = AST_TRANSFER_FAILED;
2098		ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
	2032	ast_queue_control_data(chan, AST_CONTROL_TRANSFER, &message, sizeof(message));
2099	2033
2100	2034	return;
2101	2035	}
2102	2036
2103		pjsip_evsub_set_mod_data(sub, refer_callback_module.id, session);
	2037	/* refer_callback_module requires a reference to chan
	2038	* which will be released in xfer_client_on_evsub_state()
	2039	* when the implicit REFER subscription terminates */
	2040	pjsip_evsub_set_mod_data(sub, refer_callback_module.id, chan);
	2041	ao2_ref(chan, +1);
2104	2042
2105	2043	if (pjsip_xfer_initiate(sub, pj_cstr(&tmp, target), &packet) != PJ_SUCCESS) {
2106		message = AST_TRANSFER_FAILED;
2107		ast_queue_control_data(session->channel, AST_CONTROL_TRANSFER, &message, sizeof(message));
2108		pjsip_evsub_terminate(sub, PJ_FALSE);
2109
2110		return;
2111		}
2112
2113		ref_by_val = pbx_builtin_getvar_helper(session->channel, "SIPREFERREDBYHDR");
	2044	goto failure;
	2045	}
	2046
	2047	ref_by_val = pbx_builtin_getvar_helper(chan, "SIPREFERREDBYHDR");
2114	2048	if (!ast_strlen_zero(ref_by_val)) {
2115	2049	ast_sip_add_header(packet, "Referred-By", ref_by_val);
2116	2050	} else {

2118	2052	ast_sip_add_header(packet, "Referred-By", local_info);
2119	2053	}
2120	2054
2121		pjsip_xfer_send_request(sub, packet);
	2055	if (pjsip_xfer_send_request(sub, packet) == PJ_SUCCESS) {
	2056	return;
	2057	}
	2058
	2059	failure:
	2060	message = AST_TRANSFER_FAILED;
	2061	ast_queue_control_data(chan, AST_CONTROL_TRANSFER, &message, sizeof(message));
	2062	pjsip_evsub_set_mod_data(sub, refer_callback_module.id, NULL);
	2063	pjsip_evsub_terminate(sub, PJ_FALSE);
	2064
	2065	ao2_ref(chan, -1);
2122	2066	}
2123	2067
2124	2068	static int transfer(void *data)

2149	2093	}
2150	2094	}
2151	2095
2152		#ifdef HAVE_PJSIP_INV_SESSION_REF
2153		pjsip_inv_dec_ref(trnf_data->session->inv_session);
2154		#endif
2155
2156	2096	ao2_ref(trnf_data, -1);
2157	2097	ao2_cleanup(endpoint);
2158	2098	ao2_cleanup(contact);

2169	2109	return -1;
2170	2110	}
2171	2111
2172		#ifdef HAVE_PJSIP_INV_SESSION_REF
2173		if (pjsip_inv_add_ref(trnf_data->session->inv_session) != PJ_SUCCESS) {
2174		ast_log(LOG_ERROR, "Can't increase the session reference counter\n");
2175		ao2_cleanup(trnf_data);
2176		return -1;
2177		}
2178		#endif
2179
2180	2112	if (ast_sip_push_task(channel->session->serializer, transfer, trnf_data)) {
2181	2113	ast_log(LOG_WARNING, "Error requesting transfer\n");
2182		#ifdef HAVE_PJSIP_INV_SESSION_REF
2183		pjsip_inv_dec_ref(trnf_data->session->inv_session);
2184		#endif
2185	2114	ao2_cleanup(trnf_data);
2186	2115	return -1;
2187	2116	}

2276	2205	ast_log(LOG_ERROR, "Session already DISCONNECTED [reason=%d (%s)]\n",
2277	2206	session->inv_session->cause,
2278	2207	pjsip_get_status_text(session->inv_session->cause)->ptr);
2279		goto failure;
	2208	return -1;
2280	2209	}
2281	2210
2282	2211	if (!(body_text = ast_str_create(32))) {
2283	2212	ast_log(LOG_ERROR, "Could not allocate buffer for INFO DTMF.\n");
2284		goto failure;
	2213	return -1;
2285	2214	}
2286	2215	ast_str_set(&body_text, 0, "Signal=%c\r\nDuration=%u\r\n", dtmf_data->digit, dtmf_data->duration);
2287	2216

2289	2218
2290	2219	if (ast_sip_create_request("INFO", session->inv_session->dlg, session->endpoint, NULL, NULL, &tdata)) {
2291	2220	ast_log(LOG_ERROR, "Could not create DTMF INFO request\n");
2292		goto failure;
	2221	return -1;
2293	2222	}
2294	2223	if (ast_sip_add_body(tdata, &body)) {
2295	2224	ast_log(LOG_ERROR, "Could not add body to DTMF INFO request\n");
2296	2225	pjsip_tx_data_dec_ref(tdata);
2297		goto failure;
	2226	return -1;
2298	2227	}
2299	2228	ast_sip_session_send_request(session, tdata);
2300	2229
2301		#ifdef HAVE_PJSIP_INV_SESSION_REF
2302		pjsip_inv_dec_ref(session->inv_session);
2303		#endif
2304
2305	2230	return 0;
2306
2307		failure:
2308		#ifdef HAVE_PJSIP_INV_SESSION_REF
2309		pjsip_inv_dec_ref(session->inv_session);
2310		#endif
2311		return -1;
2312
2313	2231	}
2314	2232
2315	2233	/! \brief Function called by core to stop a DTMF digit /

2350	2268	return -1;
2351	2269	}
2352	2270
2353		#ifdef HAVE_PJSIP_INV_SESSION_REF
2354		if (pjsip_inv_add_ref(dtmf_data->session->inv_session) != PJ_SUCCESS) {
2355		ast_log(LOG_ERROR, "Can't increase the session reference counter\n");
2356		ao2_cleanup(dtmf_data);
2357		return -1;
2358		}
2359		#endif
2360
2361	2271	if (ast_sip_push_task(channel->session->serializer, transmit_info_dtmf, dtmf_data)) {
2362	2272	ast_log(LOG_WARNING, "Error sending DTMF via INFO.\n");
2363		#ifdef HAVE_PJSIP_INV_SESSION_REF
2364		pjsip_inv_dec_ref(dtmf_data->session->inv_session);
2365		#endif
2366	2273	ao2_cleanup(dtmf_data);
2367	2274	return -1;
2368	2275	}

2857	2764	ast_sip_send_request(tdata, data->session->inv_session->dlg, data->session->endpoint, NULL, NULL);
2858	2765	}
2859	2766
2860		#ifdef HAVE_PJSIP_INV_SESSION_REF
2861		pjsip_inv_dec_ref(data->session->inv_session);
2862		#endif
2863
2864	2767	ao2_cleanup(data);
2865	2768
2866	2769	return 0;

2882	2785	return -1;
2883	2786	}
2884	2787
2885		#ifdef HAVE_PJSIP_INV_SESSION_REF
2886		if (pjsip_inv_add_ref(data->session->inv_session) != PJ_SUCCESS) {
2887		ast_log(LOG_ERROR, "Can't increase the session reference counter\n");
2888		ao2_ref(data, -1);
2889		return -1;
2890		}
2891		#endif
2892
2893	2788	if (ast_sip_push_task(channel->session->serializer, sendtext, data)) {
2894		#ifdef HAVE_PJSIP_INV_SESSION_REF
2895		pjsip_inv_dec_ref(data->session->inv_session);
2896		#endif
2897	2789	ao2_ref(data, -1);
2898	2790	return -1;
2899	2791	}

3051	2943	}
3052	2944	}
3053	2945
	2946	static void set_sipdomain_variable(struct ast_sip_session *session)
	2947	{
	2948	pjsip_sip_uri *sip_ruri = pjsip_uri_get_uri(session->request_uri);
	2949	size_t size = pj_strlen(&sip_ruri->host) + 1;
	2950	char *domain = ast_alloca(size);
	2951
	2952	ast_copy_pj_str(domain, &sip_ruri->host, size);
	2953
	2954	pbx_builtin_setvar_helper(session->channel, "SIPDOMAIN", domain);
	2955	return;
	2956	}
	2957
3054	2958	/! \brief Function called when a request is received on the session /
3055	2959	static int chan_pjsip_incoming_request(struct ast_sip_session session, struct pjsip_rx_data rdata)
3056	2960	{

3102	3006	SCOPE_EXIT_LOG_RTN_VALUE(-1, LOG_ERROR, "%s: Failed to allocate new PJSIP channel on incoming SIP INVITE\n",
3103	3007	ast_sip_session_get_name(session));
3104	3008	}
	3009
	3010	set_sipdomain_variable(session);
	3011
3105	3012	/* channel gets created on incoming request, but we wait to call start
3106	3013	so other supplements have a chance to run */
3107	3014	SCOPE_EXIT_RTN_VALUE(0, "%s\n", ast_sip_session_get_name(session));

3197	3104	};
3198	3105
3199	3106	/! \brief Function called when a response is received on the session /
3200		static void chan_pjsip_incoming_response(struct ast_sip_session session, struct pjsip_rx_data rdata)
	3107	static void chan_pjsip_incoming_response_update_cause(struct ast_sip_session session, struct pjsip_rx_data rdata)
3201	3108	{
3202	3109	struct pjsip_status_line status = rdata->msg_info.msg->line.status;
3203	3110	struct ast_control_pvt_cause_code *cause_code;

3222	3129	cause_code->ast_cause = hangup_sip2cause(status.code);
3223	3130	ast_queue_control_data(session->channel, AST_CONTROL_PVT_CAUSE_CODE, cause_code, data_size);
3224	3131	ast_channel_hangupcause_hash_set(session->channel, cause_code, data_size);
	3132
	3133	SCOPE_EXIT_RTN("%s\n", ast_sip_session_get_name(session));
	3134	}
	3135
	3136	/! \brief Function called when a response is received on the session /
	3137	static void chan_pjsip_incoming_response(struct ast_sip_session session, struct pjsip_rx_data rdata)
	3138	{
	3139	struct pjsip_status_line status = rdata->msg_info.msg->line.status;
	3140	SCOPE_ENTER(3, "%s: Status: %d\n", ast_sip_session_get_name(session), status.code);
	3141
	3142	if (!session->channel) {
	3143	SCOPE_EXIT_RTN("%s: No channel\n", ast_sip_session_get_name(session));
	3144	}
3225	3145
3226	3146	switch (status.code) {
3227	3147	case 180:

+6

-26

channels/chan_sip.c less more

3119	3119
3120	3120	if (ast_iostream_get_ssl(tcptls_session->stream)) {
3121	3121	set_socket_transport(&req.socket, AST_TRANSPORT_TLS);
3122		req.socket.port = htons(ourport_tls);
3123	3122	} else {
3124	3123	set_socket_transport(&req.socket, AST_TRANSPORT_TCP);
3125		req.socket.port = htons(ourport_tcp);
3126	3124	}
3127	3125	req.socket.fd = ast_iostream_get_fd(tcptls_session->stream);
3128	3126

6401	6399	}
6402	6400	}
6403	6401
6404		if (!dialog->socket.type)
	6402	if (!dialog->socket.type) {
6405	6403	set_socket_transport(&dialog->socket, AST_TRANSPORT_UDP);
6406		if (!dialog->socket.port) {
6407		dialog->socket.port = htons(ast_sockaddr_port(&bindaddr));
6408	6404	}
6409	6405
6410	6406	if (!ast_sockaddr_port(&dialog->sa)) {

10444	10440	rtcp_mux_offered = has_media_level_attribute(iterator, req, "rtcp-mux");
10445	10441
10446	10442	/* Check for 'audio' media offer */
10447		if (strncmp(m, "audio ", 6) == 0) {
	10443	if (p->rtp && strncmp(m, "audio ", 6) == 0) {
10448	10444	if ((sscanf(m, "audio %30u/%30u %17s %n", &x, &numberofports, protocol, &len) == 3 && len > 0) \|\|
10449	10445	(sscanf(m, "audio %30u %17s %n", &x, protocol, &len) == 2 && len > 0)) {
10450	10446	codecs = m + len;

10554	10550	}
10555	10551	}
10556	10552	/* Check for 'video' media offer */
10557		else if (strncmp(m, "video ", 6) == 0) {
	10553	else if (p->vrtp && strncmp(m, "video ", 6) == 0) {
10558	10554	if ((sscanf(m, "video %30u/%30u %17s %n", &x, &numberofports, protocol, &len) == 3 && len > 0) \|\|
10559	10555	(sscanf(m, "video %30u %17s %n", &x, protocol, &len) == 2 && len > 0)) {
10560	10556	codecs = m + len;

10633	10629	}
10634	10630	}
10635	10631	/* Check for 'text' media offer */
10636		else if (strncmp(m, "text ", 5) == 0) {
	10632	else if (p->trtp && strncmp(m, "text ", 5) == 0) {
10637	10633	if ((sscanf(m, "text %30u/%30u %17s %n", &x, &numberofports, protocol, &len) == 3 && len > 0) \|\|
10638	10634	(sscanf(m, "text %30u %17s %n", &x, protocol, &len) == 2 && len > 0)) {
10639	10635	codecs = m + len;

15147	15143	ast_string_field_set(mwi->call, peersecret, mwi->secret);
15148	15144	}
15149	15145	set_socket_transport(&mwi->call->socket, mwi->transport);
15150		mwi->call->socket.port = htons(mwi->portno);
15151	15146	ast_sip_ouraddrfor(&mwi->call->sa, &mwi->call->ourip, mwi->call);
15152	15147	build_via(mwi->call);
15153	15148

16261	16256	ast_string_field_set(p, exten, r->callback);
16262	16257	}
16263	16258
16264		/* Set transport and port so the correct contact is built */
	16259	/* Set transport so the correct contact is built */
16265	16260	set_socket_transport(&p->socket, r->transport);
16266		if (r->transport == AST_TRANSPORT_TLS \|\| r->transport == AST_TRANSPORT_TCP) {
16267		if (ast_sockaddr_isnull(&sip_tcp_desc.local_address)) {
16268		ast_log(LOG_ERROR,
16269		"TCP/TLS clients without server were not tested.\n");
16270		ast_log(LOG_ERROR,
16271		"Please, follow-up and report at issue 28798.\n");
16272		} else {
16273		p->socket.port =
16274		htons(ast_sockaddr_port(&sip_tcp_desc.local_address));
16275		}
16276		}
16277	16261
16278	16262	/*
16279	16263	check which address we should use in our contact header

29425	29409
29426	29410	req.socket.fd = sipsock;
29427	29411	set_socket_transport(&req.socket, AST_TRANSPORT_UDP);
29428		req.socket.tcptls_session = NULL;
29429		req.socket.port = htons(ast_sockaddr_port(&bindaddr));
	29412	req.socket.tcptls_session = NULL;
29430	29413
29431	29414	handle_request_do(&req, &addr);
29432	29415	deinit_req(&req);

32367	32350	(peer->socket.type & AST_TRANSPORT_TLS) ?
32368	32351	STANDARD_TLS_PORT : STANDARD_SIP_PORT);
32369	32352	}
32370		if (!peer->socket.port) {
32371		peer->socket.port = htons(((peer->socket.type & AST_TRANSPORT_TLS) ? STANDARD_TLS_PORT : STANDARD_SIP_PORT));
32372		}
32373	32353
32374	32354	if (realtime) {
32375	32355	int enablepoke = 1;

+1

-1

channels/sip/include/sip.h less more

789	789	struct sip_socket {
790	790	enum ast_transport type; /!< UDP, TCP or TLS /
791	791	int fd; /!< Filed descriptor, the actual socket /
792		uint16_t port;
	792	uint16_t unused; /* since 1.6.2, retained not to change order/size of struct */
793	793	struct ast_tcptls_session_instance tcptls_session; / If tcp or tls, a socket manager */
794	794	struct ast_websocket ws_session; /! If ws or wss, a WebSocket session */
795	795	};

+5

-3

configs/basic-pbx/modules.conf less more

0	0	[modules]
1	1	autoload = no
2	2
3		; This is a minimal module load. We are loading only the modules required for
4		; the Asterisk features used in the Super Awesome Company configuration.
	3	; This is a minimal module load. We are loading only the modules
	4	; required for the Asterisk features used in the "Super Awesome
	5	; Company" configuration.
5	6
6	7	; Applications
7	8

109	110	load = res_sorcery_realtime.so
110	111	load = res_timing_timerfd.so
111	112
112		; Don't load res_hep.so and kin unless you are using hep monitoring in your network
	113	; Do not load res_hep and kin unless you are using HEP monitoring
	114	; <http://sipcapture.org> in your network.
113	115
114	116	noload = res_hep.so
115	117	noload = res_hep_pjsip.so

+1

-0

configs/samples/asterisk.conf.sample less more

0	0	[directories](!)
	1	astcachedir => /tmp
1	2	astetcdir => /etc/asterisk
2	3	astmoddir => /usr/lib/asterisk/modules
3	4	astvarlibdir => /var/lib/asterisk

+6

-3

configs/samples/modules.conf.sample less more

23	23	; the "require" keyword. Asterisk will exit with a status code of 2
24	24	; if a required module does not load.
25	25	;
26		; require = chan_pjsip.so
	26	;require = chan_pjsip.so
27	27	;
28	28	; If you want you can combine with preload
29	29	; preload-require = your_special_module.so

34	34	; By default, load chan_oss only (automatically).
35	35	;
36	36	noload => chan_alsa.so
	37	noload => chan_console.so
37	38	;noload => chan_oss.so
38		noload => chan_console.so
39
	39	;
	40	; Do not load res_hep and kin unless you are using HEP monitoring
	41	; <http://sipcapture.org> in your network.
	42	;
40	43	noload => res_hep.so
41	44	noload => res_hep_pjsip.so
42	45	noload => res_hep_rtcp.so

+12

-0

configs/samples/rtp.conf.sample less more

43	43	; if rtp packets are dropped from one or both ends after a call is
44	44	; connected. This option is set to 4 by default.
45	45	; probation=8
	46	;
	47	; Enable sRTP replay protection. Buggy SIP user agents (UAs) reset the
	48	; sequence number (RTP-SEQ) on a re-INVITE, for example, with Session Timers
	49	; or on Call Hold/Resume, but keep the synchronization source (RTP-SSRC). If
	50	; the new RTP-SEQ is higher than the previous one, the call continues if the
	51	; roll-over counter (sRTP-ROC) is zero (the call lasted less than 22 minutes).
	52	; In all other cases, the call faces one-way audio or even no audio at all.
	53	; "replay check failed (index too old)" gets printed continuously. This is a
	54	; software bug. You have to report this to the creator of that UA. Until it is
	55	; fixed, you could disable sRTP replay protection (see RFC 3711 section 3.3.2).
	56	; This option is enabled by default.
	57	; srtpreplayprotection=yes
46	58	;
47	59	; Whether to enable or disable ICE support. This option is enabled by default.
48	60	; icesupport=false

+10

-1

configure less more

1042	1042	MISDN_DIR
1043	1043	MISDN_INCLUDE
1044	1044	MISDN_LIB
	1045	LUA_VERSIONS
1045	1046	PBX_LUA
1046	1047	LUA_DIR
1047	1048	LUA_INCLUDE

1286	1287	BUILD_VENDOR
1287	1288	BUILD_CPU
1288	1289	BUILD_PLATFORM
	1290	astcachedir
1289	1291	astvarrundir
1290	1292	astlogdir
1291	1293	astspooldir

1467	1469	CXXCPP
1468	1470	JANSSON_CONFIGURE_OPTS
1469	1471	PJPROJECT_CONFIGURE_OPTS
	1472	LUA_VERSIONS
1470	1473	PKG_CONFIG
1471	1474	PKG_CONFIG_PATH
1472	1475	PKG_CONFIG_LIBDIR

2225	2228	Additional configure options to pass to bundled jansson
2226	2229	PJPROJECT_CONFIGURE_OPTS
2227	2230	Additional configure options to pass to bundled pjproject
	2231	LUA_VERSIONS
	2232	A space separated list of target lua versions to test.
2228	2233	PKG_CONFIG path to pkg-config utility
2229	2234	PKG_CONFIG_PATH
2230	2235	directories to add to pkg-config's search path

4651	4656
4652	4657	# System default paths
4653	4658	astsbindir='${sbindir}'
	4659	astcachedir='/tmp'
4654	4660	astetcdir='${sysconfdir}/asterisk'
4655	4661	astheaderdir='${includedir}/asterisk'
4656	4662	astlibdir='${libdir}'

4719	4725	;;
4720	4726	solaris*)
4721	4727	if test ${prefix} = 'NONE'; then
	4728	astcachedir=/tmp
4722	4729	astetcdir=/var/etc/asterisk
4723	4730	astsbindir=/opt/asterisk/sbin
4724	4731	astlibdir=/opt/asterisk/lib

29511	29518
29512	29519
29513	29520
29514		for ver in 5.4 5.3 5.2 5.1; do
	29521	for ver in ${LUA_VERSIONS:-5.4 5.3 5.2 5.1}; do
29515	29522
29516	29523	if test "x${PBX_LUA}" != "x1" -a "${USE_LUA}" != "no"; then
29517	29524	pbxlibdir=""

29622	29629	done
29623	29630
29624	29631	# Some distributions (like openSUSE and NetBSD) remove the 5.x suffix.
	29632	if test "x${LUA_VERSIONS}" = "x"; then
29625	29633
29626	29634	if test "x${PBX_LUA}" != "x1" -a "${USE_LUA}" != "no"; then
29627	29635	pbxlibdir=""

29718	29726	fi
29719	29727
29720	29728
	29729	fi
29721	29730
29722	29731	# Accept either RADIUS client library, their APIs are fully compatible,
29723	29732	# just different header filenames and different SONAMEs

+7

-2

configure.ac less more

33	33
34	34	# System default paths
35	35	AC_SUBST([astsbindir], ['${sbindir}'])dnl
	36	AC_SUBST([astcachedir], ['/tmp'])dnl
36	37	AC_SUBST([astetcdir], ['${sysconfdir}/asterisk'])dnl
37	38	AC_SUBST([astheaderdir], ['${includedir}/asterisk'])dnl
38	39	AC_SUBST([astlibdir], ['${libdir}'])dnl

97	98	;;
98	99	solaris*)
99	100	if test ${prefix} = 'NONE'; then
	101	astcachedir=/tmp
100	102	astetcdir=/var/etc/asterisk
101	103	astsbindir=/opt/asterisk/sbin
102	104	astlibdir=/opt/asterisk/lib

508	510	AST_EXT_LIB_SETUP([LIBXSLT], [LibXSLT], [libxslt])
509	511	AST_EXT_LIB_SETUP_OPTIONAL([LIBXSLT_CLEANUP], [LibXSLT Library Cleanup Function], [LIBXSLT], [libxslt])
510	512	AST_EXT_LIB_SETUP([LUA], [Lua], [lua])
	513	AC_ARG_VAR([LUA_VERSIONS],[A space separated list of target lua versions to test.])
511	514	AST_EXT_LIB_SETUP([MISDN], [mISDN user], [misdn])
512	515	AST_EXT_LIB_SETUP([MYSQLCLIENT], [MySQL client], [mysqlclient])
513	516	AST_EXT_LIB_SETUP([NBS], [Network Broadcast Sound], [nbs])

2552	2555	fi
2553	2556	AST_EXT_LIB_CHECK([OPUSFILE], [opusfile], [op_open_callbacks], [opus/opusfile.h], [], [$__opus_include])
2554	2557
2555		for ver in 5.4 5.3 5.2 5.1; do
	2558	for ver in ${LUA_VERSIONS:-5.4 5.3 5.2 5.1}; do
2556	2559	AST_EXT_LIB_CHECK([LUA], lua${ver}, [luaL_newstate], lua${ver}/lua.h, [-lm])
2557	2560	if test "x${PBX_LUA}" = "x1" ; then
2558	2561	if test x"${LUA_DIR}" = x; then

2565	2568	done
2566	2569
2567	2570	# Some distributions (like openSUSE and NetBSD) remove the 5.x suffix.
2568		AST_EXT_LIB_CHECK([LUA], [lua], [luaL_newstate], [lua.h], [-lm])
	2571	if test "x${LUA_VERSIONS}" = "x"; then
	2572	AST_EXT_LIB_CHECK([LUA], [lua], [luaL_newstate], [lua.h], [-lm])
	2573	fi
2569	2574
2570	2575	# Accept either RADIUS client library, their APIs are fully compatible,
2571	2576	# just different header filenames and different SONAMEs

+7

-0

contrib/systemd/asterisk.service less more

0	0	[Unit]
1	1	Description=Asterisk PBX and telephony daemon.
2	2	After=network.target
	3	#include these if asterisk need to bind to a specific IP (other than 0.0.0.0)
	4	#Wants=network-online.target
	5	#After=network-online.target network.target
3	6
4	7	[Service]
5	8	Type=notify
6	9	Environment=HOME=/var/lib/asterisk
	10	#if systemd do not provide hostname and you need to use ${ENV(HOSTNAME)}
	11	#Environment=HOSTNAME=%H
7	12	WorkingDirectory=/var/lib/asterisk
8	13	User=asterisk
9	14	Group=asterisk
10	15	ExecStart=/usr/sbin/asterisk -mqf -C /etc/asterisk/asterisk.conf
11	16	ExecReload=/usr/sbin/asterisk -rx 'core reload'
	17	#if /var/run is a tmpfs, this will create /var/run/asterisk on start
	18	#RuntimeDirectory=asterisk
12	19
13	20	#Nice=0
14	21	#UMask=0002

+0

-11

~~doc/CHANGES-staging/hide_messaging_ami_events~~ less more

0		Subject: Messaging
1
2		In order to reduce the amount of AMI and ARI events generated,
3		the global "Message/ast_msg_queue" channel can be set to suppress
4		it's normal channel housekeeping events such as "Newexten",
5		"VarSet", etc. This can greatly reduce load on the manager
6		and ARI applications when the Digium Phone Module for Asterisk
7		is in use. To enable, set "hide_messaging_ami_events" in
8		asterisk.conf to "yes" In Asterisk versions <18, the default
9		is "no" preserving existing behavior. Beginning with
10		Asterisk 18, the option will default to "yes".

+9

-0

doc/CHANGES-staging/srtp_replay_protection.txt less more

	0	Subject: res_srtp
	1
	2	SRTP replay protection has been added to res_srtp and
	3	a new configuration option "srtpreplayprotection" has
	4	been added to the rtp.conf config file. For security
	5	reasons, the default setting is "yes". Buggy clients
	6	may not handle this correctly which could result in
	7	no, or one way, audio and Asterisk error messages like
	8	"replay check failed".

+9

-0

doc/UPGRADE-staging/srtp_replay_protection.txt less more

	0	Subject: res_srtp
	1
	2	SRTP replay protection has been added to res_srtp and
	3	a new configuration option "srtpreplayprotection" has
	4	been added to the rtp.conf config file. For security
	5	reasons, the default setting is "yes". Buggy clients
	6	may not handle this correctly which could result in
	7	no, or one way, audio and Asterisk error messages like
	8	"replay check failed".

+60

-111

funcs/func_lock.c less more

109	109	static void lock_free(void *data);
110	110	static void lock_fixup(void data, struct ast_channel oldchan, struct ast_channel *newchan);
111	111	static int unloading = 0;
112		static pthread_t broker_tid = AST_PTHREADT_NULL;
113	112
114	113	static const struct ast_datastore_info lock_info = {
115	114	.type = "MUTEX",

123	122	ast_cond_t cond;
124	123	/! count is needed so if a recursive mutex exits early, we know how many times to unlock it. /
125	124	unsigned int count;
126		/! Container of requesters for the named lock /
127		struct ao2_container *requesters;
	125	/! Count of waiting of requesters for the named lock /
	126	unsigned int requesters;
128	127	/! who owns us /
129	128	struct ast_channel *owner;
130	129	/! name of the lock /

146	145	while ((clframe = AST_LIST_REMOVE_HEAD(oldlist, list))) {
147	146	/* Only unlock if we own the lock */
148	147	if (clframe->channel == clframe->lock_frame->owner) {
	148	ast_mutex_lock(&clframe->lock_frame->mutex);
149	149	clframe->lock_frame->count = 0;
150	150	clframe->lock_frame->owner = NULL;
	151	ast_cond_signal(&clframe->lock_frame->cond);
	152	ast_mutex_unlock(&clframe->lock_frame->mutex);
151	153	}
152	154	ast_free(clframe);
153	155	}

172	174	if (clframe->lock_frame->owner == oldchan) {
173	175	clframe->lock_frame->owner = newchan;
174	176	}
175		/* We don't move requesters, because the thread stack is different */
176	177	clframe->channel = newchan;
177	178	}
178	179	AST_LIST_UNLOCK(list);
179		}
180
181		static void lock_broker(void unused)
182		{
183		struct lock_frame *frame;
184		struct timespec forever = { 1000000, 0 };
185		for (;;) {
186		int found_requester = 0;
187
188		/* Test for cancel outside of the lock */
189		pthread_testcancel();
190		AST_LIST_LOCK(&locklist);
191
192		AST_LIST_TRAVERSE(&locklist, frame, entries) {
193		if (ao2_container_count(frame->requesters)) {
194		found_requester++;
195		ast_mutex_lock(&frame->mutex);
196		if (!frame->owner) {
197		ast_cond_signal(&frame->cond);
198		}
199		ast_mutex_unlock(&frame->mutex);
200		}
201		}
202
203		AST_LIST_UNLOCK(&locklist);
204		pthread_testcancel();
205
206		/* If there are no requesters, then wait for a signal */
207		if (!found_requester) {
208		nanosleep(&forever, NULL);
209		} else {
210		sched_yield();
211		}
212		}
213		/* Not reached */
214		return NULL;
215		}
216
217		static int ast_channel_cmp_cb(void obj, void arg, int flags)
218		{
219		struct ast_channel chan = obj, cmp_args = arg;
220		return strcasecmp(ast_channel_name(chan), ast_channel_name(cmp_args)) ? 0 : CMP_MATCH;
221	180	}
222	181
223	182	static int get_lock(struct ast_channel chan, char lockname, int trylock)

289	248	AST_LIST_UNLOCK(&locklist);
290	249	return -1;
291	250	}
292		current->requesters = ao2_container_alloc_list(AO2_ALLOC_OPT_LOCK_MUTEX, 0,
293		NULL, ast_channel_cmp_cb);
294		if (!current->requesters) {
295		ast_mutex_destroy(&current->mutex);
296		ast_cond_destroy(&current->cond);
297		ast_free(current);
298		AST_LIST_UNLOCK(&locklist);
299		return -1;
300		}
	251	current->requesters = 0;
301	252	AST_LIST_INSERT_TAIL(&locklist, current, entries);
302	253	}
	254	/* Add to requester list */
	255	ast_mutex_lock(&current->mutex);
	256	current->requesters++;
	257	ast_mutex_unlock(&current->mutex);
303	258	AST_LIST_UNLOCK(&locklist);
304	259
305	260	/* Found lock or created one - now find or create the corresponding link in the channel */

336	291	* the same amount, before we'll release this one.
337	292	*/
338	293	if (current->owner == chan) {
	294	/* We're not a requester, we already have it */
	295	ast_mutex_lock(&current->mutex);
	296	current->requesters--;
	297	ast_mutex_unlock(&current->mutex);
339	298	current->count++;
340	299	return 0;
341	300	}
342
343		/* Okay, we have both frames, so now we need to try to lock.
344		*
345		* Locking order: always lock locklist first. We need the
346		* locklist lock because the broker thread counts whether
347		* there are requesters with the locklist lock held, and we
348		* need to hold it, so that when we send our signal, below,
349		* to wake up the broker thread, it definitely will see that
350		* a requester exists at that point in time. Otherwise, we
351		* could add to the requesters after it has already seen that
352		* that lock is unoccupied and wait forever for another signal.
353		*/
354		AST_LIST_LOCK(&locklist);
355		ast_mutex_lock(&current->mutex);
356		/* Add to requester list */
357		ao2_link(current->requesters, chan);
358		pthread_kill(broker_tid, SIGURG);
359		AST_LIST_UNLOCK(&locklist);
360	301
361	302	/* Wait up to three seconds from now for LOCK. */
362	303	now = ast_tvnow();
363	304	timeout.tv_sec = now.tv_sec + 3;
364	305	timeout.tv_nsec = now.tv_usec * 1000;
365	306
366		if (!current->owner
367		\|\| (!trylock
368		&& !(res = ast_cond_timedwait(&current->cond, &current->mutex, &timeout)))) {
369		res = 0;
	307	ast_mutex_lock(&current->mutex);
	308
	309	res = 0;
	310	while (!trylock && !res && current->owner) {
	311	res = ast_cond_timedwait(&current->cond, &current->mutex, &timeout);
	312	}
	313	if (current->owner) {
	314	/* timeout;
	315	* trylock; or
	316	* cond_timedwait failed.
	317	*
	318	* either way, we fail to obtain the lock.
	319	*/
	320	res = -1;
	321	} else {
370	322	current->owner = chan;
371	323	current->count++;
372		} else {
373		res = -1;
	324	res = 0;
374	325	}
375	326	/* Remove from requester list */
376		ao2_unlink(current->requesters, chan);
	327	current->requesters--;
	328	if (res && unloading)
	329	ast_cond_signal(&current->cond);
377	330	ast_mutex_unlock(&current->mutex);
378	331
379	332	return res;

421	374	}
422	375
423	376	if (--clframe->lock_frame->count == 0) {
	377	ast_mutex_lock(&clframe->lock_frame->mutex);
424	378	clframe->lock_frame->owner = NULL;
	379	ast_cond_signal(&clframe->lock_frame->cond);
	380	ast_mutex_unlock(&clframe->lock_frame->mutex);
425	381	}
426	382
427	383	ast_copy_string(buf, "1", len);

477	433	/* Module flag */
478	434	unloading = 1;
479	435
480		AST_LIST_LOCK(&locklist);
481		while ((current = AST_LIST_REMOVE_HEAD(&locklist, entries))) {
482		/* If any locks are currently in use, then we cannot unload this module */
483		if (current->owner \|\| ao2_container_count(current->requesters)) {
484		/* Put it back */
485		AST_LIST_INSERT_HEAD(&locklist, current, entries);
486		AST_LIST_UNLOCK(&locklist);
487		unloading = 0;
488		return -1;
489		}
490		ast_mutex_destroy(&current->mutex);
491		ao2_ref(current->requesters, -1);
492		ast_free(current);
493		}
494
495		/* No locks left, unregister functions */
	436	/* Make it impossible for new requesters to be added
	437	* NOTE: channels could already be in get_lock() */
496	438	ast_custom_function_unregister(&lock_function);
497	439	ast_custom_function_unregister(&trylock_function);
	440
	441	AST_LIST_LOCK(&locklist);
	442	AST_LIST_TRAVERSE(&locklist, current, entries) {
	443	ast_mutex_lock(&current->mutex);
	444	while (current->owner \|\| current->requesters) {
	445	/* either the mutex is locked, or other parties are currently in get_lock,
	446	* we need to wait for all of those to clear first */
	447	ast_cond_wait(&current->cond, &current->mutex);
	448	}
	449	ast_mutex_unlock(&current->mutex);
	450	/* At this point we know:
	451	* 1. the lock has been released,
	452	* 2. there are no requesters (nor should any be able to sneak in).
	453	*/
	454	ast_mutex_destroy(&current->mutex);
	455	ast_cond_destroy(&current->cond);
	456	ast_free(current);
	457	}
	458	AST_LIST_UNLOCK(&locklist);
	459	AST_LIST_HEAD_DESTROY(&locklist);
	460
	461	/* At this point we can safely stop access to UNLOCK */
498	462	ast_custom_function_unregister(&unlock_function);
499
500		if (broker_tid != AST_PTHREADT_NULL) {
501		pthread_cancel(broker_tid);
502		pthread_kill(broker_tid, SIGURG);
503		pthread_join(broker_tid, NULL);
504		}
505
506		AST_LIST_UNLOCK(&locklist);
507	463
508	464	return 0;
509	465	}

514	470	res \|= ast_custom_function_register_escalating(&trylock_function, AST_CFE_READ);
515	471	res \|= ast_custom_function_register_escalating(&unlock_function, AST_CFE_READ);
516	472
517		if (ast_pthread_create_background(&broker_tid, NULL, lock_broker, NULL)) {
518		ast_log(LOG_ERROR, "Failed to start lock broker thread. Unloading func_lock module.\n");
519		broker_tid = AST_PTHREADT_NULL;
520		unload_module();
521		return AST_MODULE_LOAD_DECLINE;
522		}
523
524	473	return res;
525	474	}
526	475

+1

-0

funcs/func_odbc.c less more

1946	1946	.load = load_module,
1947	1947	.unload = unload_module,
1948	1948	.reload = reload,
	1949	.requires = "res_odbc",
1949	1950	);

+1

-0

funcs/func_periodic_hook.c less more

518	518	.support_level = AST_MODULE_SUPPORT_CORE,
519	519	.load = load_module,
520	520	.unload = unload_module,
	521	.requires = "app_chanspy,func_cut,func_groupcount,func_uri",
521	522	);

+2

-2

include/asterisk/manager.h less more

114	114	};
115	115
116	116	/! \brief Check if AMI is enabled /
117		int check_manager_enabled(void);
	117	int ast_manager_check_enabled(void);
118	118
119	119	/! \brief Check if AMI/HTTP is enabled /
120		int check_webmanager_enabled(void);
	120	int ast_webmanager_check_enabled(void);
121	121
122	122	/*! Add a custom hook to be called when an event is fired
123	123	\param hook struct manager_custom_hook object to add

+1

-0

include/asterisk/paths.h less more

17	17	#ifndef _ASTERISK_PATHS_H
18	18	#define _ASTERISK_PATHS_H
19	19
	20	extern const char *ast_config_AST_CACHE_DIR;
20	21	extern const char *ast_config_AST_CONFIG_DIR;
21	22	extern const char *ast_config_AST_CONFIG_FILE;
22	23	extern const char *ast_config_AST_MODULE_DIR;

+2

-2

main/asterisk.c less more

533	533
534	534	ast_cli(a->fd, "\n* Subsystems\n");
535	535	ast_cli(a->fd, " -------------\n");
536		ast_cli(a->fd, " Manager (AMI): %s\n", check_manager_enabled() ? "Enabled" : "Disabled");
537		ast_cli(a->fd, " Web Manager (AMI/HTTP): %s\n", check_webmanager_enabled() ? "Enabled" : "Disabled");
	536	ast_cli(a->fd, " Manager (AMI): %s\n", ast_manager_check_enabled() ? "Enabled" : "Disabled");
	537	ast_cli(a->fd, " Web Manager (AMI/HTTP): %s\n", ast_webmanager_check_enabled() ? "Enabled" : "Disabled");
538	538	ast_cli(a->fd, " Call data records: %s\n", ast_cdr_is_enabled() ? "Enabled" : "Disabled");
539	539	ast_cli(a->fd, " Realtime Architecture (ARA): %s\n", ast_realtime_enabled() ? "Enabled" : "Disabled");
540	540

+1

-1

main/bridge_basic.c less more

2637	2637	&props->original_transferer_colp);
2638	2638	ast_party_id_reset(&ast_channel_connected(props->recall_target)->priv);
2639	2639
2640		common_recall_channel_setup(props->recall_target, props->recall_target);
	2640	common_recall_channel_setup(props->recall_target, props->transferer);
2641	2641	ast_channel_unlock(props->recall_target);
2642	2642	ast_channel_unlock(props->transferer);
2643	2643

+2

-1

main/bucket.c less more

72	72	#include "asterisk/json.h"
73	73	#include "asterisk/file.h"
74	74	#include "asterisk/module.h"
	75	#include "asterisk/paths.h"
75	76
76	77	/! \brief Number of buckets for the container of schemes /
77	78	#define SCHEME_BUCKETS 53

898	899	{
899	900	int fd;
900	901
901		ast_copy_string(file->path, "/tmp/bucket-XXXXXX", sizeof(file->path));
	902	snprintf(file->path, sizeof(file->path), "%s/bucket-XXXXXX", ast_config_AST_CACHE_DIR);
902	903
903	904	fd = mkstemp(file->path);
904	905	if (fd < 0) {

+5

-0

main/logger.c less more

1671	1671	return NULL;
1672	1672	}
1673	1673
	1674	/* Automatically add a newline to format strings that don't have one */
	1675	if (!ast_ends_with(ast_str_buffer(buf), "\n")) {
	1676	ast_str_append(&buf, 0, "\n");
	1677	}
	1678
1674	1679	/* Create a new logging message */
1675	1680	if (!(logmsg = ast_calloc_with_stringfields(1, struct logmsg, res + 128))) {
1676	1681	return NULL;

+3

-3

main/manager.c less more

1945	1945	AST_RWLIST_UNLOCK(&manager_hooks);
1946	1946	}
1947	1947
1948		int check_manager_enabled(void)
	1948	int ast_manager_check_enabled(void)
1949	1949	{
1950	1950	return manager_enabled;
1951	1951	}
1952	1952
1953		int check_webmanager_enabled(void)
	1953	int ast_webmanager_check_enabled(void)
1954	1954	{
1955	1955	return (webmanager_enabled && manager_enabled);
1956	1956	}

6308	6308	ast_option_maxfiles,
6309	6309	AST_CLI_YESNO(ast_realtime_enabled()),
6310	6310	AST_CLI_YESNO(ast_cdr_is_enabled()),
6311		AST_CLI_YESNO(check_webmanager_enabled())
	6311	AST_CLI_YESNO(ast_webmanager_check_enabled())
6312	6312	);
6313	6313	return 0;
6314	6314	}

+15

-3

main/manager_channels.c less more

824	824	struct stasis_message *message)
825	825	{
826	826	RAII_VAR(struct ast_str *, spyer_channel_string, NULL, ast_free);
	827	RAII_VAR(struct ast_str *, spyee_channel_string, NULL, ast_free);
827	828	struct ast_channel_snapshot *spyer;
	829	struct ast_channel_snapshot *spyee;
	830	const char *spyee_info = "";
828	831	struct ast_multi_channel_blob *payload = stasis_message_data(message);
829	832
830	833	spyer = ast_multi_channel_blob_get_channel(payload, "spyer_channel");
831	834	if (!spyer) {
832		ast_log(AST_LOG_WARNING, "Received ChanSpy Start event with no spyer channel!\n");
	835	ast_log(AST_LOG_WARNING, "Received ChanSpy Stop event with no spyer channel!\n");
833	836	return;
834	837	}
835	838

838	841	return;
839	842	}
840	843
	844	spyee = ast_multi_channel_blob_get_channel(payload, "spyee_channel");
	845	if (spyee) {
	846	spyee_channel_string = ast_manager_build_channel_state_string_prefix(spyee, "Spyee");
	847	if (spyee_channel_string) {
	848	spyee_info = ast_str_buffer(spyee_channel_string);
	849	}
	850	}
	851
841	852	manager_event(EVENT_FLAG_CALL, "ChanSpyStop",
842		"%s",
843		ast_str_buffer(spyer_channel_string));
	853	"%s%s",
	854	ast_str_buffer(spyer_channel_string),
	855	spyee_info);
844	856	}
845	857
846	858	static void channel_chanspy_start_cb(void data, struct stasis_subscription sub,

+1

-0

main/media_cache.c less more

183	183	}
184	184	ast_free(mime_type);
185	185	}
	186	ao2_ref(header, -1);
186	187	}
187	188	}
188	189

+6

-1

main/options.c less more

97	97	char ast_defaultlanguage[MAX_LANGUAGE] = DEFAULT_LANGUAGE;
98	98
99	99	struct _cfg_paths {
	100	char cache_dir[PATH_MAX];
100	101	char config_dir[PATH_MAX];
101	102	char module_dir[PATH_MAX];
102	103	char spool_dir[PATH_MAX];

124	125	};
125	126
126	127	static struct _cfg_paths cfg_paths = {
	128	.cache_dir = DEFAULT_CACHE_DIR,
127	129	.config_dir = DEFAULT_CONFIG_DIR,
128	130	.module_dir = DEFAULT_MODULE_DIR,
129	131	.spool_dir = DEFAULT_SPOOL_DIR,

144	146	.ctl_file = "asterisk.ctl",
145	147	};
146	148
	149	const char *ast_config_AST_CACHE_DIR = cfg_paths.cache_dir;
147	150	const char *ast_config_AST_CONFIG_DIR = cfg_paths.config_dir;
148	151	const char *ast_config_AST_CONFIG_FILE = cfg_paths.config_file;
149	152	const char *ast_config_AST_MODULE_DIR = cfg_paths.module_dir;

253	256	}
254	257
255	258	for (v = ast_variable_browse(cfg, "directories"); v; v = v->next) {
256		if (!strcasecmp(v->name, "astetcdir")) {
	259	if (!strcasecmp(v->name, "astcachedir")) {
	260	ast_copy_string(cfg_paths.cache_dir, v->value, sizeof(cfg_paths.cache_dir));
	261	} else if (!strcasecmp(v->name, "astetcdir")) {
257	262	ast_copy_string(cfg_paths.config_dir, v->value, sizeof(cfg_paths.config_dir));
258	263	} else if (!strcasecmp(v->name, "astspooldir")) {
259	264	ast_copy_string(cfg_paths.spool_dir, v->value, sizeof(cfg_paths.spool_dir));

+2

-0

main/pbx_variables.c less more

326	326	s = ast_str_buffer(*str);
327	327	} else if (!strcmp(var, "SYSTEMNAME")) {
328	328	s = ast_config_AST_SYSTEM_NAME;
	329	} else if (!strcmp(var, "ASTCACHEDIR")) {
	330	s = ast_config_AST_CACHE_DIR;
329	331	} else if (!strcmp(var, "ASTETCDIR")) {
330	332	s = ast_config_AST_CONFIG_DIR;
331	333	} else if (!strcmp(var, "ASTMODDIR")) {

+1

-0

makeopts.in less more

96	96	ASTMANDIR = @astmandir@
97	97	astvarlibdir = @astvarlibdir@
98	98	ASTVARLIBDIR = @astvarlibdir@
	99	ASTCACHEDIR = @astcachedir@
99	100	ASTDATADIR = @astdatadir@
100	101	ASTDBDIR = @astdbdir@
101	102	ASTKEYDIR = @astkeydir@

+1

-1

res/res_hep_pjsip.c less more

250	250	.support_level = AST_MODULE_SUPPORT_EXTENDED,
251	251	.load = load_module,
252	252	.unload = unload_module,
253		.requires = "res_hep,res_pjsip",
	253	.requires = "res_pjsip,res_pjsip_session,res_hep",
254	254	);

+1

-0

res/res_http_media_cache.c less more

214	214	curl_easy_setopt(curl, CURLOPT_HEADERFUNCTION, curl_header_callback);
215	215	curl_easy_setopt(curl, CURLOPT_USERAGENT, GLOBAL_USERAGENT);
216	216	curl_easy_setopt(curl, CURLOPT_FOLLOWLOCATION, 1);
	217	curl_easy_setopt(curl, CURLOPT_MAXREDIRS, 8);
217	218	curl_easy_setopt(curl, CURLOPT_URL, ast_sorcery_object_get_id(cb_data->bucket_file));
218	219	curl_easy_setopt(curl, CURLOPT_HEADERDATA, cb_data);
219	220

+13

-10

res/res_musiconhold.c less more

1612	1612	char *category = NULL;
1613	1613	size_t entry_count = 0;
1614	1614
1615		while ((category = ast_category_browse(entries, category))) {
1616		const char *entry = ast_variable_retrieve(entries, category, "entry");
1617
1618		if (entry) {
1619		struct ast_variable *dup = ast_variable_new("entry", entry, "");
1620		if (dup) {
1621		entry_count++;
1622		ast_variable_list_append(&var, dup);
	1615	/* entries is NULL if there are no results */
	1616	if (entries) {
	1617	while ((category = ast_category_browse(entries, category))) {
	1618	const char *entry = ast_variable_retrieve(entries, category, "entry");
	1619
	1620	if (entry) {
	1621	struct ast_variable *dup = ast_variable_new("entry", entry, "");
	1622	if (dup) {
	1623	entry_count++;
	1624	ast_variable_list_append(&var, dup);
	1625	}
1623	1626	}
1624	1627	}
1625		}
1626		ast_config_destroy(entries);
	1628	ast_config_destroy(entries);
	1629	}
1627	1630
1628	1631	if (entry_count == 0) {
1629	1632	/* Behave as though this class doesn't exist */

+1

-0

res/res_odbc.c less more

1142	1142	.unload = unload_module,
1143	1143	.reload = reload,
1144	1144	.load_pri = AST_MODPRI_REALTIME_DEPEND,
	1145	.requires = "res_odbc_transaction",
1145	1146	);

+2

-0

res/res_pjproject.c less more

33	33
34	34	/*** MODULEINFO
35	35	<depend>pjproject</depend>
	36	<depend>res_sorcery_config</depend>
36	37	<support_level>core</support_level>
37	38	***/
38	39

766	767	.unload = unload_module,
767	768	.reload = reload_module,
768	769	.load_pri = AST_MODPRI_CHANNEL_DEPEND - 6,
	770	.requires = "res_sorcery_config",
769	771	);

+1

-1

res/res_pjsip/pjsip_options.c less more

93	93	*/
94	94
95	95	#define DEFAULT_LANGUAGE "en"
96		#define DEFAULT_ENCODING "text/plain"
	96	#define DEFAULT_ENCODING "identity"
97	97
98	98	/! \brief These are the number of buckets to store AORs in /
99	99	#ifdef LOW_MEMORY

+1

-1

res/res_pjsip.c less more

5428	5428	.unload = unload_module,
5429	5429	.reload = reload_module,
5430	5430	.load_pri = AST_MODPRI_CHANNEL_DEPEND - 5,
5431		.requires = "dnsmgr,res_pjproject",
	5431	.requires = "dnsmgr,res_pjproject,res_sorcery_config,res_sorcery_memory,res_sorcery_astdb",
5432	5432	.optional_modules = "res_statsd",
5433	5433	);

+14

-0

res/res_pjsip_diversion.c less more

119	119	static int add_supported(pjsip_tx_data *tdata)
120	120	{
121	121	pjsip_supported_hdr *hdr;
	122	unsigned int i;
122	123
123	124	hdr = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_SUPPORTED, NULL);
124	125	if (!hdr) {

129	130	}
130	131
131	132	pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *)hdr);
	133	}
	134
	135	/* Asterisk can send multiple "181 Call forwarded" in a single session,
	136	* we might have already modified Supported before
	137	*/
	138	for (i = 0; i < hdr->count; ++i) {
	139	if (pj_stricmp(&hdr->values[i], &HISTINFO_SUPPORTED_NAME) == 0) {
	140	return 0;
	141	}
	142	}
	143
	144	if (hdr->count >= PJSIP_GENERIC_ARRAY_MAX_COUNT) {
	145	return -1;
132	146	}
133	147
134	148	/* add on to the existing Supported header */

+1

-1

res/res_pjsip_dlg_options.c less more

33	33	#include "asterisk/res_pjsip_session.h"
34	34
35	35	#define DEFAULT_LANGUAGE "en"
36		#define DEFAULT_ENCODING "text/plain"
	36	#define DEFAULT_ENCODING "identity"
37	37
38	38	static int options_incoming_request(struct ast_sip_session session, pjsip_rx_data rdata)
39	39	{

+5

-5

res/res_pjsip_nat.c less more

65	65	return;
66	66	}
67	67
68		static void rewrite_uri(pjsip_rx_data rdata, pjsip_sip_uri uri)
	68	static void rewrite_uri(pjsip_rx_data rdata, pjsip_sip_uri uri, pj_pool_t *pool)
69	69	{
70	70
71	71	if (pj_strcmp2(&uri->host, rdata->pkt_info.src_name) != 0) {
72	72	save_orig_contact_host(rdata, uri);
73	73	}
74	74
75		pj_cstr(&uri->host, rdata->pkt_info.src_name);
	75	pj_strdup2(pool, &uri->host, rdata->pkt_info.src_name);
76	76	uri->port = rdata->pkt_info.src_port;
77	77	if (!strcasecmp("WSS", rdata->tp_info.transport->type_name)) {
78	78	/* WSS is special, we don't want to overwrite the URI at all as it needs to be ws */

150	150
151	151	if (rr) {
152	152	uri = pjsip_uri_get_uri(&rr->name_addr);
153		rewrite_uri(rdata, uri);
	153	rewrite_uri(rdata, uri, rdata->tp_info.pool);
154	154	res = 0;
155	155	}
156	156
157	157	if (dlg && !pj_list_empty(&dlg->route_set) && !dlg->route_set_frozen) {
158	158	pjsip_routing_hdr *route = dlg->route_set.next;
159	159	uri = pjsip_uri_get_uri(&route->name_addr);
160		rewrite_uri(rdata, uri);
	160	rewrite_uri(rdata, uri, dlg->pool);
161	161	res = 0;
162	162	}
163	163

183	183	if (contact && !contact->star && (PJSIP_URI_SCHEME_IS_SIP(contact->uri) \|\| PJSIP_URI_SCHEME_IS_SIPS(contact->uri))) {
184	184	pjsip_sip_uri *uri = pjsip_uri_get_uri(contact->uri);
185	185
186		rewrite_uri(rdata, uri);
	186	rewrite_uri(rdata, uri, rdata->tp_info.pool);
187	187
188	188	if (dlg && pj_list_empty(&dlg->route_set) && (!dlg->remote.contact
189	189	\|\| pjsip_uri_cmp(PJSIP_URI_IN_REQ_URI, dlg->remote.contact->uri, contact->uri))) {

+139

-177

res/res_pjsip_outbound_registration.c less more

342	342	*/
343	343	pjsip_tx_data *last_tdata;
344	344	/! \brief Timer entry for retrying on temporal responses /
345		struct ast_sip_sched_task *sched_task;
	345	pj_timer_entry timer;
346	346	/! \brief Optional line parameter placed into Contact /
347	347	char line[LINE_PARAMETER_SIZE];
348	348	/! \brief Current number of retries /

373	373	char *transport_name;
374	374	/! \brief The name of the registration sorcery object /
375	375	char *registration_name;
376		/! \brief Server URI /
377		char *server_uri;
378		/! \brief Client URI /
379		char *client_uri;
380		/! \brief Contact URI /
381		char *contact_uri;
382	376	};
383	377
384	378	/! \brief Outbound registration state information (persists for lifetime that registration should exist) /

517	511	return NULL;
518	512	}
519	513
520		ast_debug(3, "%s: Line '%s', using configured endpoint '%s'\n",
521		state->client_state->registration_name, state->client_state->line, state->registration->endpoint);
	514	ast_debug(3, "Determined relationship to outbound registration '%s' based on line '%s', using configured endpoint '%s'\n",
	515	ast_sorcery_object_get_id(state->registration), state->client_state->line, state->registration->endpoint);
522	516
523	517	return ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", state->registration->endpoint);
524	518	}

530	524	/! \brief Helper function which cancels the timer on a client /
531	525	static void cancel_registration(struct sip_outbound_registration_client_state *client_state)
532	526	{
533		if (client_state->sched_task) {
534		ast_sip_sched_task_cancel(client_state->sched_task);
535		ao2_cleanup(client_state->sched_task);
536		client_state->sched_task = NULL;
	527	if (pj_timer_heap_cancel_if_active(pjsip_endpt_get_timer_heap(ast_sip_get_pjsip_endpoint()),
	528	&client_state->timer, client_state->timer.id)) {
	529	/* The timer was successfully cancelled, drop the refcount of client_state */
	530	ao2_ref(client_state, -1);
537	531	}
538	532	}
539	533

550	544	callback_invoked = ast_threadstorage_get(&register_callback_invoked, sizeof(int));
551	545	if (!callback_invoked) {
552	546	pjsip_tx_data_dec_ref(tdata);
553		ast_log(LOG_ERROR, "%s: Failed to get threadstorage for registration to server '%s' from client '%s'\n",
554		client_state->registration_name, client_state->server_uri, client_state->client_uri);
555	547	return PJ_ENOMEM;
556	548	}
557	549	*callback_invoked = 0;

579	571	* drop the references we just added
580	572	*/
581	573	if ((status != PJ_SUCCESS) && !(*callback_invoked)) {
582		ast_log(LOG_ERROR, "%s: Failed send registration to server '%s' from client '%s'\n",
583		client_state->registration_name, client_state->server_uri, client_state->client_uri);
584	574	pjsip_tx_data_dec_ref(tdata);
585	575	ao2_ref(client_state, -1);
586	576	return status;

602	592	/! \brief Callback function for registering /
603	593	static int handle_client_registration(void *data)
604	594	{
605		struct sip_outbound_registration_client_state *client_state = data;
	595	RAII_VAR(struct sip_outbound_registration_client_state *, client_state, data, ao2_cleanup);
606	596	pjsip_tx_data *tdata;
607	597
608		if (client_state->status == SIP_REGISTRATION_STOPPED) {
	598	if (client_state->status == SIP_REGISTRATION_STOPPED
	599	\|\| pjsip_regc_register(client_state->client, PJ_FALSE, &tdata) != PJ_SUCCESS) {
609	600	return 0;
610	601	}
611	602
612		if (pjsip_regc_register(client_state->client, PJ_FALSE, &tdata) != PJ_SUCCESS) {
613		ast_log(LOG_ERROR, "%s: Failed to create registration to server '%s' from client '%s'\n",
614		client_state->registration_name, client_state->server_uri, client_state->client_uri);
615		return 0;
616		}
617
618		ast_debug(1, "%s: Outbound REGISTER attempt %u to '%s' with client '%s'\n",
619		client_state->registration_name,
620		client_state->retries + 1,
621		client_state->server_uri, client_state->client_uri);
	603	if (DEBUG_ATLEAST(1)) {
	604	pjsip_regc_info info;
	605
	606	pjsip_regc_get_info(client_state->client, &info);
	607	ast_log(LOG_DEBUG, "Outbound REGISTER attempt %u to '%.s' with client '%.s'\n",
	608	client_state->retries + 1,
	609	(int) info.server_uri.slen, info.server_uri.ptr,
	610	(int) info.client_uri.slen, info.client_uri.ptr);
	611	}
622	612
623	613	if (client_state->support_path) {
624	614	pjsip_supported_hdr *hdr;
	615	int i;
625	616
626	617	hdr = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_SUPPORTED, NULL);
627	618	if (!hdr) {

629	620	hdr = pjsip_supported_hdr_create(tdata->pool);
630	621	if (!hdr) {
631	622	pjsip_tx_data_dec_ref(tdata);
632		ast_log(LOG_ERROR, "%s: Failed to add SUPPORTED header to registration to server '%s' from client '%s'\n",
633		client_state->registration_name, client_state->server_uri, client_state->client_uri);
634		return 0;
	623	return -1;
635	624	}
636	625
637	626	pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *)hdr);
638	627	}
639	628
	629	/* Don't add the value if it's already there */
	630	for (i = 0; i < hdr->count; ++i) {
	631	if (pj_stricmp(&hdr->values[i], &PATH_NAME) == 0) {
	632	return 1;
	633	}
	634	}
	635
	636	if (hdr->count >= PJSIP_GENERIC_ARRAY_MAX_COUNT) {
	637	return 0;
	638	}
	639
640	640	/* add on to the existing Supported header */
641	641	pj_strassign(&hdr->values[hdr->count++], &PATH_NAME);
642	642	}

646	646	return 0;
647	647	}
648	648
	649	/! \brief Timer callback function, used just for registrations /
	650	static void sip_outbound_registration_timer_cb(pj_timer_heap_t timer_heap, struct pj_timer_entry entry)
	651	{
	652	struct sip_outbound_registration_client_state *client_state = entry->user_data;
	653
	654	entry->id = 0;
	655
	656	/*
	657	* Transfer client_state reference to serializer task so the
	658	* nominal path will not dec the client_state ref in this
	659	* pjproject callback thread.
	660	*/
	661	if (ast_sip_push_task(client_state->serializer, handle_client_registration, client_state)) {
	662	ast_log(LOG_WARNING, "Scheduled outbound registration could not be executed.\n");
	663	ao2_ref(client_state, -1);
	664	}
	665	}
	666
649	667	/! \brief Helper function which sets up the timer to re-register in a specific amount of time /
650		static void schedule_registration(struct sip_outbound_registration_client_state *client_state, unsigned int ms)
651		{
652
653		ast_debug(1, "%s: Scheduling outbound registration to server '%s' from client '%s' in %d ms\n",
654		client_state->registration_name, client_state->server_uri, client_state->client_uri, ms);
	668	static void schedule_registration(struct sip_outbound_registration_client_state *client_state, unsigned int seconds)
	669	{
	670	pj_time_val delay = { .sec = seconds, };
	671	pjsip_regc_info info;
655	672
656	673	cancel_registration(client_state);
657	674
658		client_state->sched_task = ast_sip_schedule_task(client_state->serializer,
659		ms, handle_client_registration, ast_taskprocessor_name(client_state->serializer),
660		client_state,
661		AST_SIP_SCHED_TASK_ONESHOT \| AST_SIP_SCHED_TASK_DATA_AO2);
662
663		if (!client_state->sched_task) {
664		ast_log(LOG_WARNING, "%s: Failed to schedule registration to server '%s' from client '%s'\n",
665		client_state->registration_name, client_state->server_uri, client_state->client_uri);
	675	pjsip_regc_get_info(client_state->client, &info);
	676	ast_debug(1, "Scheduling outbound registration to server '%.s' from client '%.s' in %d seconds\n",
	677	(int) info.server_uri.slen, info.server_uri.ptr,
	678	(int) info.client_uri.slen, info.client_uri.ptr,
	679	seconds);
	680
	681	ao2_ref(client_state, +1);
	682	if (pjsip_endpt_schedule_timer(ast_sip_get_pjsip_endpoint(), &client_state->timer, &delay) != PJ_SUCCESS) {
	683	ast_log(LOG_WARNING, "Failed to schedule registration to server '%.s' from client '%.s'\n",
	684	(int) info.server_uri.slen, info.server_uri.ptr,
	685	(int) info.client_uri.slen, info.client_uri.ptr);
	686	ao2_ref(client_state, -1);
666	687	}
667	688	}
668	689

699	720	{
700	721	struct sip_outbound_registration_client_state *client_state = data;
701	722
	723	cancel_registration(client_state);
	724
702	725	if (client_state->client) {
703	726	pjsip_regc_info info;
704	727	pjsip_tx_data *tdata;
705	728
706		/* We need to get the live state to test "is_busy" */
707	729	pjsip_regc_get_info(client_state->client, &info);
708	730
709	731	if (info.is_busy == PJ_TRUE) {
710	732	/* If a client transaction is in progress we defer until it is complete */
711	733	ast_debug(1,
712		"%s: Registration transaction is busy with server '%.s' from client '%.s'.\n",
713		client_state->registration_name,
	734	"Registration transaction is busy with server '%.s' from client '%.s'.\n",
714	735	(int) info.server_uri.slen, info.server_uri.ptr,
715	736	(int) info.client_uri.slen, info.client_uri.ptr);
716	737	client_state->destroy = 1;

723	744	break;
724	745	case SIP_REGISTRATION_REGISTERED:
725	746	ast_debug(1,
726		"%s: Trying to unregister with server '%.s' from client '%.s' before destruction.\n",
727		client_state->registration_name,
	747	"Trying to unregister with server '%.s' from client '%.s' before destruction.\n",
728	748	(int) info.server_uri.slen, info.server_uri.ptr,
729	749	(int) info.client_uri.slen, info.client_uri.ptr);
730		cancel_registration(client_state);
	750
731	751	update_client_state_status(client_state, SIP_REGISTRATION_STOPPING);
732	752	client_state->destroy = 1;
733	753	if (pjsip_regc_unregister(client_state->client, &tdata) == PJ_SUCCESS

737	757	}
738	758	break;
739	759	case SIP_REGISTRATION_REJECTED_TEMPORARY:
	760	case SIP_REGISTRATION_REJECTED_PERMANENT:
740	761	case SIP_REGISTRATION_STOPPING:
741		cancel_registration(client_state);
742		break;
743		case SIP_REGISTRATION_REJECTED_PERMANENT:
744	762	case SIP_REGISTRATION_STOPPED:
745	763	break;
746	764	}

751	769
752	770	update_client_state_status(client_state, SIP_REGISTRATION_STOPPED);
753	771	ast_sip_auth_vector_destroy(&client_state->outbound_auths);
754
755	772	ao2_ref(client_state, -1);
756	773
757	774	return 0;

809	826	}
810	827	}
811	828
812		static void schedule_retry(struct registration_response *response, unsigned int interval)
	829	static void schedule_retry(struct registration_response *response, unsigned int interval,
	830	const char server_uri, const char client_uri)
813	831	{
814	832	update_client_state_status(response->client_state, SIP_REGISTRATION_REJECTED_TEMPORARY);
815		schedule_registration(response->client_state, interval * 1000);
	833	schedule_registration(response->client_state, interval);
816	834
817	835	if (response->rdata) {
818		ast_log(LOG_WARNING, "%s: Temporal response '%d' received from '%s' on "
	836	ast_log(LOG_WARNING, "Temporal response '%d' received from '%s' on "
819	837	"registration attempt to '%s', retrying in '%u'\n",
820		response->client_state->registration_name,
821		response->code, response->client_state->server_uri, response->client_state->client_uri, interval);
	838	response->code, server_uri, client_uri, interval);
822	839	} else {
823		ast_log(LOG_WARNING, "%s: No response received from '%s' on "
	840	ast_log(LOG_WARNING, "No response received from '%s' on "
824	841	"registration attempt to '%s', retrying in '%u'\n",
825		response->client_state->registration_name,
826		response->client_state->server_uri, response->client_state->client_uri, interval);
	842	server_uri, client_uri, interval);
827	843	}
828	844	}
829	845

836	852	return 0;
837	853	}
838	854
839		ast_debug(1,
840		"%s: Outbound registration transport to server '%s' from client '%s' shutdown\n",
841		state->client_state->registration_name, state->client_state->server_uri, state->client_state->client_uri);
	855	if (DEBUG_ATLEAST(1)) {
	856	pjsip_regc_info info;
	857
	858	pjsip_regc_get_info(state->client_state->client, &info);
	859	ast_log(LOG_DEBUG,
	860	"Outbound registration transport to server '%.s' from client '%.s' shutdown\n",
	861	(int) info.server_uri.slen, info.server_uri.ptr,
	862	(int) info.client_uri.slen, info.client_uri.ptr);
	863	}
842	864
843	865	cancel_registration(state->client_state);
844	866

911	933	static int handle_registration_response(void *data)
912	934	{
913	935	struct registration_response *response = data;
	936	pjsip_regc_info info;
	937	char server_uri[PJSIP_MAX_URL_SIZE];
	938	char client_uri[PJSIP_MAX_URL_SIZE];
914	939
915	940	if (response->client_state->status == SIP_REGISTRATION_STOPPED) {
916	941	ao2_ref(response, -1);
917	942	return 0;
918	943	}
919	944
920		ast_debug(1, "%s: Processing REGISTER response %d from server '%s' for client '%s' with expiration '%d'\n",
921		response->client_state->registration_name, response->code, response->client_state->server_uri,
922		response->client_state->client_uri, response->expiration);
	945	pjsip_regc_get_info(response->client_state->client, &info);
	946	ast_copy_pj_str(server_uri, &info.server_uri, sizeof(server_uri));
	947	ast_copy_pj_str(client_uri, &info.client_uri, sizeof(client_uri));
	948
	949	ast_debug(1, "Processing REGISTER response %d from server '%s' for client '%s'\n",
	950	response->code, server_uri, client_uri);
923	951
924	952	if (response->code == 408 \|\| response->code == 503) {
925	953	if ((ast_sip_failover_request(response->old_request))) {

941	969	if (!ast_sip_create_request_with_auth(&response->client_state->outbound_auths,
942	970	response->rdata, response->old_request, &tdata)) {
943	971	response->client_state->auth_attempted = 1;
944		ast_debug(1, "%s: Sending authenticated REGISTER to server '%s' from client '%s'\n",
945		response->client_state->registration_name, response->client_state->server_uri,
946		response->client_state->client_uri);
	972	ast_debug(1, "Sending authenticated REGISTER to server '%s' from client '%s'\n",
	973	server_uri, client_uri);
947	974	pjsip_tx_data_add_ref(tdata);
948	975	res = registration_client_send(response->client_state, tdata);
949	976

957	984	return 0;
958	985	}
959	986	} else {
960		ast_log(LOG_WARNING, "%s: Failed to create authenticated REGISTER request to server '%s' from client '%s'\n",
961		response->client_state->registration_name, response->client_state->server_uri,
962		response->client_state->client_uri);
	987	ast_log(LOG_WARNING, "Failed to create authenticated REGISTER request to server '%s' from client '%s'\n",
	988	server_uri, client_uri);
963	989	}
964	990	/* Otherwise, fall through so the failure is processed appropriately */
965	991	}

972	998	int next_registration_round;
973	999
974	1000	/* If the registration went fine simply reschedule registration for the future */
975		ast_debug(1, "%s: Outbound registration to '%s' with client '%s' successful\n",
976		response->client_state->registration_name, response->client_state->server_uri,
977		response->client_state->client_uri);
	1001	ast_debug(1, "Outbound registration to '%s' with client '%s' successful\n", server_uri, client_uri);
978	1002	update_client_state_status(response->client_state, SIP_REGISTRATION_REGISTERED);
979	1003	response->client_state->retries = 0;
980		next_registration_round = (response->expiration - REREGISTER_BUFFER_TIME) * 1000;
	1004	next_registration_round = response->expiration - REREGISTER_BUFFER_TIME;
981	1005	if (next_registration_round < 0) {
982		/* Re-register no sooner than the buffer time. */
983		next_registration_round = REREGISTER_BUFFER_TIME * 1000;
	1006	/* Re-register immediately. */
	1007	next_registration_round = 0;
984	1008	}
985	1009	schedule_registration(response->client_state, next_registration_round);
986	1010

988	1012	registration_transport_monitor_setup(response->rdata->tp_info.transport,
989	1013	response->client_state->registration_name);
990	1014	} else {
991		ast_debug(1, "%s: Outbound unregistration to '%s' with client '%s' successful\n",
992		response->client_state->registration_name, response->client_state->server_uri,
993		response->client_state->client_uri);
	1015	ast_debug(1, "Outbound unregistration to '%s' with client '%s' successful\n", server_uri, client_uri);
994	1016	update_client_state_status(response->client_state, SIP_REGISTRATION_UNREGISTERED);
995	1017	ast_sip_transport_monitor_unregister(response->rdata->tp_info.transport,
996	1018	registration_transport_shutdown_cb, response->client_state->registration_name,

1000	1022	/* We need to deal with the pending destruction instead. */
1001	1023	} else if (response->retry_after) {
1002	1024	/* If we have been instructed to retry after a period of time, schedule it as such */
1003		schedule_retry(response, response->retry_after);
	1025	schedule_retry(response, response->retry_after, server_uri, client_uri);
1004	1026	} else if (response->client_state->retry_interval
1005	1027	&& sip_outbound_registration_is_temporal(response->code, response->client_state)) {
1006	1028	if (response->client_state->retries == response->client_state->max_retries) {
1007	1029	/* If we received enough temporal responses to exceed our maximum give up permanently */
1008	1030	update_client_state_status(response->client_state, SIP_REGISTRATION_REJECTED_PERMANENT);
1009		ast_log(LOG_WARNING, "%s: Maximum retries reached when attempting outbound registration to '%s' with client '%s', stopping registration attempt\n",
1010		response->client_state->registration_name, response->client_state->server_uri,
1011		response->client_state->client_uri);
	1031	ast_log(LOG_WARNING, "Maximum retries reached when attempting outbound registration to '%s' with client '%s', stopping registration attempt\n",
	1032	server_uri, client_uri);
1012	1033	} else {
1013	1034	/* On the other hand if we can still try some more do so */
1014	1035	response->client_state->retries++;
1015		schedule_retry(response, response->client_state->retry_interval);
	1036	schedule_retry(response, response->client_state->retry_interval, server_uri, client_uri);
1016	1037	}
1017	1038	} else {
1018	1039	if (response->code == 403

1021	1042	/* A forbidden response retry interval is configured and there are retries remaining */
1022	1043	update_client_state_status(response->client_state, SIP_REGISTRATION_REJECTED_TEMPORARY);
1023	1044	response->client_state->retries++;
1024		schedule_registration(response->client_state, response->client_state->forbidden_retry_interval * 1000);
1025		ast_log(LOG_WARNING, "%s: 403 Forbidden fatal response received from '%s' on registration attempt to '%s', retrying in '%u' seconds\n",
1026		response->client_state->registration_name, response->client_state->server_uri,
1027		response->client_state->client_uri, response->client_state->forbidden_retry_interval);
	1045	schedule_registration(response->client_state, response->client_state->forbidden_retry_interval);
	1046	ast_log(LOG_WARNING, "403 Forbidden fatal response received from '%s' on registration attempt to '%s', retrying in '%u' seconds\n",
	1047	server_uri, client_uri, response->client_state->forbidden_retry_interval);
1028	1048	} else if (response->client_state->fatal_retry_interval
1029	1049	&& response->client_state->retries < response->client_state->max_retries) {
1030	1050	/* Some kind of fatal failure response received, so retry according to configured interval */
1031	1051	update_client_state_status(response->client_state, SIP_REGISTRATION_REJECTED_TEMPORARY);
1032	1052	response->client_state->retries++;
1033		schedule_registration(response->client_state, response->client_state->fatal_retry_interval * 1000);
1034		ast_log(LOG_WARNING, "%s: '%d' fatal response received from '%s' on registration attempt to '%s', retrying in '%u' seconds\n",
1035		response->client_state->registration_name, response->code, response->client_state->server_uri,
1036		response->client_state->client_uri, response->client_state->fatal_retry_interval);
	1053	schedule_registration(response->client_state, response->client_state->fatal_retry_interval);
	1054	ast_log(LOG_WARNING, "'%d' fatal response received from '%s' on registration attempt to '%s', retrying in '%u' seconds\n",
	1055	response->code, server_uri, client_uri, response->client_state->fatal_retry_interval);
1037	1056	} else {
1038	1057	/* Finally if there's no hope of registering give up */
1039	1058	update_client_state_status(response->client_state, SIP_REGISTRATION_REJECTED_PERMANENT);
1040	1059	if (response->rdata) {
1041		ast_log(LOG_WARNING, "%s: Fatal response '%d' received from '%s' on registration attempt to '%s', stopping outbound registration\n",
1042		response->client_state->registration_name, response->code, response->client_state->server_uri,
1043		response->client_state->client_uri);
	1060	ast_log(LOG_WARNING, "Fatal response '%d' received from '%s' on registration attempt to '%s', stopping outbound registration\n",
	1061	response->code, server_uri, client_uri);
1044	1062	} else {
1045		ast_log(LOG_WARNING, "%s: Fatal registration attempt to '%s', stopping outbound registration\n",
1046		response->client_state->registration_name, response->client_state->client_uri);
	1063	ast_log(LOG_WARNING, "Fatal registration attempt to '%s', stopping outbound registration\n", client_uri);
1047	1064	}
1048	1065	}
1049	1066	}
1050	1067
1051		ast_system_publish_registry("PJSIP", response->client_state->client_uri, response->client_state->server_uri,
	1068	ast_system_publish_registry("PJSIP", client_uri, server_uri,
1052	1069	sip_outbound_registration_status_str(response->client_state->status), NULL);
1053	1070
1054	1071	if (response->client_state->destroy) {

1089	1106	*/
1090	1107	response->client_state = client_state;
1091	1108
1092		ast_debug(1, "%s: Received REGISTER response %d(%.*s) from '%s': Expires: %d\n",
1093		client_state->registration_name,
1094		param->code, (int) param->reason.slen, param->reason.ptr,
1095		client_state->server_uri, param->expiration);
	1109	ast_debug(1, "Received REGISTER response %d(%.*s)\n",
	1110	param->code, (int) param->reason.slen, param->reason.ptr);
1096	1111
1097	1112	if (param->rdata) {
1098	1113	struct pjsip_retry_after_hdr *retry_after;

1127	1142	* pjproject callback thread.
1128	1143	*/
1129	1144	if (ast_sip_push_task(client_state->serializer, handle_registration_response, response)) {
1130		ast_log(LOG_WARNING, "%s: Failed to pass incoming registration response to threadpool\n",
1131		client_state->registration_name);
	1145	ast_log(LOG_WARNING, "Failed to pass incoming registration response to threadpool\n");
1132	1146	ao2_cleanup(response);
1133	1147	}
1134	1148	}

1138	1152	{
1139	1153	struct sip_outbound_registration_state *state = obj;
1140	1154
1141		ast_debug(3, "%s: Destroying registration state for registration to server '%s' from client '%s'\n",
1142		state->client_state->registration_name,
	1155	ast_debug(3, "Destroying registration state for registration to server '%s' from client '%s'\n",
1143	1156	state->registration ? state->registration->server_uri : "",
1144	1157	state->registration ? state->registration->client_uri : "");
1145	1158	ao2_cleanup(state->registration);

1150	1163	ao2_ref(state->client_state, -1);
1151	1164	} else if (ast_sip_push_task(state->client_state->serializer,
1152	1165	handle_client_state_destruction, state->client_state)) {
1153		ast_log(LOG_WARNING, "%s: Failed to pass outbound registration client destruction to threadpool\n",
1154		state->client_state->registration_name);
	1166	ast_log(LOG_WARNING, "Failed to pass outbound registration client destruction to threadpool\n");
1155	1167	ao2_ref(state->client_state, -1);
1156	1168	}
1157	1169	}

1160	1172	static void sip_outbound_registration_client_state_destroy(void *obj)
1161	1173	{
1162	1174	struct sip_outbound_registration_client_state *client_state = obj;
1163
1164		ast_debug(3, "%s: Destroying registration client state\n",
1165		client_state->registration_name);
1166	1175
1167	1176	ast_statsd_log_string("PJSIP.registrations.count", AST_STATSD_GAUGE, "-1", 1.0);
1168	1177	ast_statsd_log_string_va("PJSIP.registrations.state.%s", AST_STATSD_GAUGE, "-1", 1.0,

1171	1180	ast_taskprocessor_unreference(client_state->serializer);
1172	1181	ast_free(client_state->transport_name);
1173	1182	ast_free(client_state->registration_name);
1174		ast_free(client_state->server_uri);
1175		ast_free(client_state->client_uri);
1176		ast_free(client_state->contact_uri);
1177	1183	if (client_state->last_tdata) {
1178	1184	pjsip_tx_data_dec_ref(client_state->last_tdata);
1179	1185	}

1197	1203	}
1198	1204
1199	1205	state->client_state->status = SIP_REGISTRATION_UNREGISTERED;
	1206	pj_timer_entry_init(&state->client_state->timer, 0, state->client_state,
	1207	sip_outbound_registration_timer_cb);
1200	1208	state->client_state->transport_name = ast_strdup(registration->transport);
1201	1209	state->client_state->registration_name =
1202	1210	ast_strdup(ast_sorcery_object_get_id(registration));

1431	1439	}
1432	1440
1433	1441	pj_cstr(&server_uri, registration->server_uri);
1434		state->client_state->server_uri = ast_strdup(registration->server_uri);
	1442
1435	1443
1436	1444	if (sip_dialog_create_contact(pjsip_regc_get_pool(state->client_state->client),
1437	1445	&contact_uri, S_OR(registration->contact_user, "s"), &server_uri, &selector,
1438	1446	state->client_state->line)) {
1439	1447	return -1;
1440	1448	}
1441		ast_copy_pj_str2(&state->client_state->contact_uri, &contact_uri);
1442	1449
1443	1450	pj_cstr(&client_uri, registration->client_uri);
1444		state->client_state->client_uri = ast_strdup(registration->client_uri);
1445
1446	1451	if (pjsip_regc_init(state->client_state->client, &server_uri, &client_uri,
1447	1452	&client_uri, 1, &contact_uri, registration->expiration) != PJ_SUCCESS) {
1448	1453	return -1;
1449	1454	}
1450	1455
1451
1452	1456	return 0;
1453	1457	}
1454	1458

1458	1462	struct sip_outbound_registration_state *state = data;
1459	1463	struct sip_outbound_registration *registration = ao2_bump(state->registration);
1460	1464	size_t i;
1461
1462		ast_debug(1, "%s: Performing register to server '%s' from client '%s' Expiration: %d\n",
1463		state->client_state->registration_name,
1464		state->registration->server_uri, state->registration->client_uri, registration->expiration);
1465	1465
1466	1466	/* Just in case the client state is being reused for this registration, free the auth information */
1467	1467	ast_sip_auth_vector_destroy(&state->client_state->outbound_auths);

1484	1484
1485	1485	pjsip_regc_update_expires(state->client_state->client, registration->expiration);
1486	1486
1487		schedule_registration(state->client_state, ((ast_random() % 10) + 1) * 1000);
	1487	schedule_registration(state->client_state, (ast_random() % 10) + 1);
1488	1488
1489	1489	ao2_ref(registration, -1);
1490	1490	ao2_ref(state, -1);

1613	1613	struct sip_outbound_registration_state *state = obj;
1614	1614	struct pjsip_regc *client = state->client_state->client;
1615	1615	pjsip_tx_data *tdata;
1616
1617		ast_debug(1, "%s: Unregistering contacts with server '%s' from client '%s'\n",
1618		state->client_state->registration_name,
	1616	pjsip_regc_info info;
	1617
	1618	pjsip_regc_get_info(client, &info);
	1619	ast_debug(1, "Unregistering contacts with server '%s' from client '%s'\n",
1619	1620	state->registration->server_uri, state->registration->client_uri);
1620	1621
1621	1622	cancel_registration(state->client_state);

1642	1643	static int queue_register(struct sip_outbound_registration_state *state)
1643	1644	{
1644	1645	ao2_ref(state, +1);
1645
1646		ast_debug(1, "%s: Queueing register to server '%s' from client '%s'\n",
1647		state->client_state->registration_name,
1648		state->registration->server_uri, state->registration->client_uri);
1649
1650	1646	if (ast_sip_push_task(state->client_state->serializer, sip_outbound_registration_perform, state)) {
1651	1647	ao2_ref(state, -1);
1652	1648	return -1;

1900	1896	ast_sip_sorcery_object_to_ami(ami->registration, &buf);
1901	1897
1902	1898	if ((state = get_state(ast_sorcery_object_get_id(ami->registration)))) {
1903		int next_reg = 0;
	1899	pjsip_regc_info info;
1904	1900
1905	1901	if (state->client_state->status == SIP_REGISTRATION_REGISTERED) {
1906	1902	++ami->registered;

1911	1907	ast_str_append(&buf, 0, "Status: %s\r\n",
1912	1908	sip_outbound_registration_status_str(state->client_state->status));
1913	1909
1914		if (state->client_state->status == SIP_REGISTRATION_REGISTERED && state->client_state->sched_task) {
1915		ast_sip_sched_task_get_times2(state->client_state->sched_task, NULL, NULL, NULL, NULL,
1916		&next_reg, NULL);
1917		}
1918
1919		ast_str_append(&buf, 0, "NextReg: %d\r\n", next_reg / 1000);
	1910	pjsip_regc_get_info(state->client_state->client, &info);
	1911	ast_str_append(&buf, 0, "NextReg: %d\r\n", info.next_reg);
1920	1912	ao2_ref(state, -1);
1921	1913	}
1922	1914

2021	2013	ast_assert(context->output_buffer != NULL);
2022	2014
2023	2015	ast_str_append(&context->output_buffer, 0,
2024		" <Registration/ServerURI..............................> <Auth..........> <Status.......>");
2025		ast_str_append(&context->output_buffer, 0,
2026		" <Last Reg..> <Intvl> <Next Start.....secs>\n"
2027		"==============================================");
	2016	" <Registration/ServerURI..............................> <Auth..........> <Status.......>\n");
2028	2017
2029	2018	return 0;
2030	2019	}
2031
2032		#define TIME_FORMAT "%a %T"
2033	2020
2034	2021	static int cli_print_body(void obj, void arg, int flags)
2035	2022	{

2037	2024	struct ast_sip_cli_context *context = arg;
2038	2025	const char *id = ast_sorcery_object_get_id(registration);
2039	2026	struct sip_outbound_registration_state *state = get_state(id);
2040		int next_run_ms = 0;
2041		struct timeval next = { 0, 0 };
2042		struct timeval last = { 0, 0 };
2043		struct ast_tm tm;
2044		char next_start[32] = "";
2045		char last_start[32] = "";
2046		int interval;
2047
2048	2027	#define REGISTRATION_URI_FIELD_LEN 53
2049	2028
2050	2029	ast_assert(context->output_buffer != NULL);
2051	2030
2052		if (state->client_state->sched_task) {
2053		ast_sip_sched_task_get_times2(state->client_state->sched_task, &last, NULL, NULL, &interval,
2054		&next_run_ms, &next);
2055
2056		ast_localtime(&last, &tm, NULL);
2057		ast_strftime(last_start, sizeof(last_start), TIME_FORMAT, &tm);
2058
2059		ast_localtime(&next, &tm, NULL);
2060		ast_strftime(next_start, sizeof(next_start), TIME_FORMAT, &tm);
2061		}
2062
2063		ast_str_append(&context->output_buffer, 0, " %-s/%-.s %-16s %-16s",
	2031	ast_str_append(&context->output_buffer, 0, " %-s/%-.s %-16s %-16s\n",
2064	2032	id,
2065	2033	(int) (REGISTRATION_URI_FIELD_LEN - strlen(id)),
2066	2034	(int) (REGISTRATION_URI_FIELD_LEN - strlen(id)),

2068	2036	AST_VECTOR_SIZE(&registration->outbound_auths)
2069	2037	? AST_VECTOR_GET(&registration->outbound_auths, 0)
2070	2038	: "n/a",
2071		(state ? sip_outbound_registration_status_str(state->client_state->status) : "Unregistered")
2072		);
2073
2074		ast_str_append(&context->output_buffer, 0, " %-12s %7d %-12s %8d\n",
2075		last_start, interval / 1000, next_start, next_run_ms / 1000);
2076
	2039	(state ? sip_outbound_registration_status_str(state->client_state->status) : "Unregistered"));
2077	2040	ao2_cleanup(state);
2078
2079	2041
2080	2042	if (context->show_details
2081	2043	\|\| (context->show_details_only_level_0 && context->indent_level == 0)) {

+12

-0

res/res_pjsip_path.c less more

122	122	static int add_supported(pjsip_tx_data *tdata)
123	123	{
124	124	pjsip_supported_hdr *hdr;
	125	int i;
125	126
126	127	hdr = pjsip_msg_find_hdr(tdata->msg, PJSIP_H_SUPPORTED, NULL);
127	128	if (!hdr) {

132	133	}
133	134
134	135	pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr *)hdr);
	136	}
	137
	138	/* Don't add the value if it's already there */
	139	for (i = 0; i < hdr->count; ++i) {
	140	if (pj_stricmp(&hdr->values[i], &PATH_SUPPORTED_NAME) == 0) {
	141	return 0;
	142	}
	143	}
	144
	145	if (hdr->count >= PJSIP_GENERIC_ARRAY_MAX_COUNT) {
	146	return -1;
135	147	}
136	148
137	149	/* add on to the existing Supported header */

+5

-3

res/res_pjsip_pidf_digium_body_supplement.c less more

41	41	pj_xml_node *node;
42	42	char sanitized[1024];
43	43
	44	/* The res_pjsip_exten_state module converts the user agent to lower case */
44	45	if (ast_strlen_zero(state_data->user_agent) \|\|
45		!strstr(state_data->user_agent, "digium")) {
46		/* not a digium phone */
	46	(!strstr(state_data->user_agent, "digium") &&
	47	!strstr(state_data->user_agent, "sangoma"))) {
	48	/* not a Sangoma phone */
47	49	return 0;
48	50	}
49	51

106	108	return 0;
107	109	}
108	110
109		AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP PIDF Digium presence supplement",
	111	AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP PIDF Sangoma presence supplement",
110	112	.support_level = AST_MODULE_SUPPORT_CORE,
111	113	.load = load_module,
112	114	.unload = unload_module,

+41

-32

res/res_pjsip_session.c less more

2267	2267	ast_sip_session_get_name(session));
2268	2268	}
2269	2269
2270		if (active_media_state) {
	2270	/*
	2271	* Attempt to resolve only if objects are available, and it's not
	2272	* switching to or from an image type.
	2273	*/
	2274	if (active_media_state && active_media_state->topology &&
	2275	(!active_media_state->default_session[AST_MEDIA_TYPE_IMAGE] ==
	2276	!pending_media_state->default_session[AST_MEDIA_TYPE_IMAGE])) {
	2277
2271	2278	struct ast_sip_session_media_state *new_pending_state;
2272	2279	/*
2273	2280	* We need to check if the passed in active and pending states are equal

2978	2985	ast_dsp_free(session->dsp);
2979	2986
2980	2987	if (session->inv_session) {
2981		pjsip_dlg_dec_session(session->inv_session->dlg, &session_module);
	2988	struct pjsip_dialog *dlg = session->inv_session->dlg;
	2989
	2990	/* The INVITE session uses the dialog pool for memory, so we need to
	2991	* decrement its reference first before that of the dialog.
	2992	*/
	2993
	2994	#ifdef HAVE_PJSIP_INV_SESSION_REF
	2995	pjsip_inv_dec_ref(session->inv_session);
	2996	#endif
	2997	pjsip_dlg_dec_session(dlg, &session_module);
2982	2998	}
2983	2999
2984	3000	ast_test_suite_event_notify("SESSION_DESTROYED", "Endpoint: %s", endpoint_name);

3086	3102	}
3087	3103	ast_sip_dialog_set_serializer(inv_session->dlg, session->serializer);
3088	3104	ast_sip_dialog_set_endpoint(inv_session->dlg, endpoint);
	3105
	3106	/* When a PJSIP INVITE session is created it is created with a reference
	3107	* count of 1, with that reference being managed by the underlying state
	3108	* of the INVITE session itself. When the INVITE session transitions to
	3109	* a DISCONNECTED state that reference is released. This means we can not
	3110	* rely on that reference to ensure the INVITE session remains for the
	3111	* lifetime of our session. To ensure it does we add our own reference
	3112	* and release it when our own session goes away, ensuring that the INVITE
	3113	* session remains for the lifetime of session.
	3114	*/
	3115
	3116	#ifdef HAVE_PJSIP_INV_SESSION_REF
	3117	if (pjsip_inv_add_ref(inv_session) != PJ_SUCCESS) {
	3118	ast_log(LOG_ERROR, "Can't increase the session reference counter\n");
	3119	return NULL;
	3120	}
	3121	#endif
	3122
3089	3123	pjsip_dlg_inc_session(inv_session->dlg, &session_module);
3090	3124	inv_session->mod_data[session_module.id] = ao2_bump(session);
3091	3125	session->contact = ao2_bump(contact);

3711	3745	*/
3712	3746	AST_SIP_USER_OPTIONS_TRUNCATE_CHECK(session->exten);
3713	3747
3714		pickup_cfg = ast_get_chan_features_pickup_config(session->channel);
	3748	pickup_cfg = ast_get_chan_features_pickup_config(NULL); /* session->channel doesn't exist yet, using NULL */
3715	3749	if (!pickup_cfg) {
3716	3750	ast_log(LOG_ERROR, "%s: Unable to retrieve pickup configuration options. Unable to detect call pickup extension\n",
3717	3751	ast_sip_session_get_name(session));

3723	3757
3724	3758	if (!strcmp(session->exten, pickupexten) \|\|
3725	3759	ast_exists_extension(NULL, session->endpoint->context, session->exten, 1, NULL)) {
3726		size_t size = pj_strlen(&sip_ruri->host) + 1;
3727		char *domain = ast_alloca(size);
3728
3729		ast_copy_pj_str(domain, &sip_ruri->host, size);
3730		pbx_builtin_setvar_helper(session->channel, "SIPDOMAIN", domain);
3731
3732	3760	/*
3733	3761	* Save off the INVITE Request-URI in case it is
3734	3762	* needed: CHANNEL(pjsip,request_uri)

3961	3989	ast_sip_session_get_name(invite->session),
3962	3990	invite->session->inv_session->cause,
3963	3991	pjsip_get_status_text(invite->session->inv_session->cause)->ptr);
3964		#ifdef HAVE_PJSIP_INV_SESSION_REF
3965		pjsip_inv_dec_ref(invite->session->inv_session);
3966		#endif
3967	3992	SCOPE_EXIT_RTN_VALUE(-1);
3968	3993	}
3969	3994

4081	4106	handle_incoming_request(invite->session, invite->rdata);
4082	4107
4083	4108	end:
4084		#ifdef HAVE_PJSIP_INV_SESSION_REF
4085		pjsip_inv_dec_ref(invite->session->inv_session);
4086		#endif
4087	4109	SCOPE_EXIT_RTN_VALUE(0, "%s\n", ast_sip_session_get_name(invite->session));
4088	4110	}
4089	4111

4126	4148	* process handling has successfully completed.
4127	4149	*/
4128	4150
4129		#ifdef HAVE_PJSIP_INV_SESSION_REF
4130		if (pjsip_inv_add_ref(inv_session) != PJ_SUCCESS) {
4131		ast_log(LOG_ERROR, "Can't increase the session reference counter\n");
4132		/* Dialog's lock and a reference are removed in new_invite_initial_answer */
4133		if (!new_invite_initial_answer(inv_session, rdata, 500, 500, PJ_FALSE)) {
4134		/* Terminate the session if it wasn't done in the answer */
4135		pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
4136		}
4137		return;
4138		}
4139		#endif
4140	4151	session = ast_sip_session_alloc(endpoint, NULL, inv_session, rdata);
4141	4152	if (!session) {
4142	4153	/* Dialog's lock and reference are removed in new_invite_initial_answer */

4144	4155	/* Terminate the session if it wasn't done in the answer */
4145	4156	pjsip_inv_terminate(inv_session, 500, PJ_FALSE);
4146	4157	}
4147
4148		#ifdef HAVE_PJSIP_INV_SESSION_REF
4149		pjsip_inv_dec_ref(inv_session);
4150		#endif
4151	4158	return;
4152	4159	}
4153	4160

5514	5521	}
5515	5522	if (AST_VECTOR_SIZE(&left->sessions) != AST_VECTOR_SIZE(&right->sessions)) {
5516	5523	ast_assert(!assert_on_failure);
5517		SCOPE_EXIT_RTN_VALUE(0, "session vector sizes different: left %lu != right %lu\n", AST_VECTOR_SIZE(&left->sessions),
	5524	SCOPE_EXIT_RTN_VALUE(0, "session vector sizes different: left %zu != right %zu\n",
	5525	AST_VECTOR_SIZE(&left->sessions),
5518	5526	AST_VECTOR_SIZE(&right->sessions));
5519	5527	}
5520	5528	if (AST_VECTOR_SIZE(&left->read_callbacks) != AST_VECTOR_SIZE(&right->read_callbacks)) {
5521	5529	ast_assert(!assert_on_failure);
5522		SCOPE_EXIT_RTN_VALUE(0, "read_callback vector sizes different: left %lu != right %lu\n", AST_VECTOR_SIZE(&left->read_callbacks),
	5530	SCOPE_EXIT_RTN_VALUE(0, "read_callback vector sizes different: left %zu != right %zu\n",
	5531	AST_VECTOR_SIZE(&left->read_callbacks),
5523	5532	AST_VECTOR_SIZE(&right->read_callbacks));
5524	5533	}
5525	5534

+2

-2

res/res_pjsip_stir_shaken.c less more

322	322	#define AST_BUILDOPT_SUM ""
323	323
324	324	AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS \| AST_MODFLAG_LOAD_ORDER,
325		"PSIP STIR/SHAKEN Module for Asterisk",
	325	"PJSIP STIR/SHAKEN Module for Asterisk",
326	326	.support_level = AST_MODULE_SUPPORT_CORE,
327	327	.load = load_module,
328	328	.unload = unload_module,
329	329	.load_pri = AST_MODPRI_DEFAULT,
330		.requires = "res_pjsip,res_stir_shaken",
	330	.requires = "res_pjsip,res_pjsip_session,res_stir_shaken",
331	331	);

+9

-0

res/res_pjsip_t38.c less more

319	319	int index;
320	320
321	321	session_media = session->active_media_state->default_session[AST_MEDIA_TYPE_IMAGE];
	322
	323	/*
	324	* If there is a session_media object, but no udptl object available
	325	* then it's assumed the stream was declined.
	326	*/
	327	if (!session_media->udptl) {
	328	session_media = NULL;
	329	}
	330
322	331	if (!session_media) {
323	332	ast_log(LOG_WARNING, "Received %d response to T.38 re-invite on '%s' but no active session media\n",
324	333	status.code, session->channel ? ast_channel_name(session->channel) : "unknown channel");

+1

-1

res/res_pjsip_transport_websocket.c less more

453	453	pj_strbuf(txp_str));
454	454	}
455	455
456		pj_cstr(&uri->host, rdata->pkt_info.src_name);
	456	pj_strdup2(rdata->tp_info.pool, &uri->host, rdata->pkt_info.src_name);
457	457	uri->port = rdata->pkt_info.src_port;
458	458	pj_strdup(rdata->tp_info.pool, &uri->transport_param, txp_str);
459	459	}

+12

-4

res/res_rtp_asterisk.c less more

180	180	#define STRICT_RTP_LEARN_TIMEOUT 5000
181	181
182	182	#define DEFAULT_STRICT_RTP STRICT_RTP_YES /!< Enabled by default /
	183	#define DEFAULT_SRTP_REPLAY_PROTECTION 1
183	184	#define DEFAULT_ICESUPPORT 1
184	185	#define DEFAULT_DTLS_MTU 1200
185	186

202	203	static int strictrtp = DEFAULT_STRICT_RTP; /!< Only accept RTP frames from a defined source. If we receive an indication of a changing source, enter learning mode. /
203	204	static int learning_min_sequential = DEFAULT_LEARNING_MIN_SEQUENTIAL; /!< Number of sequential RTP frames needed from a single source during learning mode to accept new source. /
204	205	static int learning_min_duration = DEFAULT_LEARNING_MIN_DURATION; /!< Lowest acceptable timeout between the first and the last sequential RTP frame. /
	206	static int srtp_replay_protection = DEFAULT_SRTP_REPLAY_PROTECTION;
205	207	#if defined(HAVE_OPENSSL) && (OPENSSL_VERSION_NUMBER >= 0x10001000L) && !defined(OPENSSL_NO_SRTP)
206	208	static int dtls_mtu = DEFAULT_DTLS_MTU;
207	209	#endif

6081	6083
6082	6084	/* If this is encrypted then decrypt the payload */
6083	6085	if ((*rtcpheader & 0xC0) && res_srtp && srtp && res_srtp->unprotect(
6084		srtp, rtcpheader, &len, 1) < 0) {
	6086	srtp, rtcpheader, &len, 1 \| (srtp_replay_protection << 1)) < 0) {
6085	6087	return &ast_null_frame;
6086	6088	}
6087	6089

6931	6933	/* If there is no more room left for storing packets stop now, we leave 20
6932	6934	* extra bits at the end just in case.
6933	6935	*/
6934		if ((sizeof(bdata) - (packet_len + delta_len + 20)) < 0) {
	6936	if (packet_len + delta_len + 20 > sizeof(bdata)) {
6935	6937	res = -1;
6936	6938	break;
6937	6939	}

6965	6967	previous_packet = statistics;
6966	6968
6967	6969	/* If there is no more room left in the packet stop handling of any subsequent packets */
6968		if ((sizeof(bdata) - (packet_len + delta_len + 20)) < 0) {
	6970	if (packet_len + delta_len + 20 > sizeof(bdata)) {
6969	6971	break;
6970	6972	}
6971	6973	}

7169	7171
7170	7172	/* If this payload is encrypted then decrypt it using the given SRTP instance */
7171	7173	if ((*read_area & 0xC0) && res_srtp && srtp && res_srtp->unprotect(
7172		srtp, read_area, &res, 0) < 0) {
	7174	srtp, read_area, &res, 0 \| (srtp_replay_protection << 1)) < 0) {
7173	7175	return &ast_null_frame;
7174	7176	}
7175	7177

8957	8959	if (strictrtp) {
8958	8960	ast_cli(a->fd, " Probation: %d frames\n", learning_min_sequential);
8959	8961	}
	8962
	8963	ast_cli(a->fd, " Replay Protect: %s\n", AST_CLI_YESNO(srtp_replay_protection));
8960	8964	#ifdef HAVE_PJPROJECT
8961	8965	ast_cli(a->fd, " ICE support: %s\n", AST_CLI_YESNO(icesupport));
8962	8966	#endif

9059	9063	strictrtp = DEFAULT_STRICT_RTP;
9060	9064	learning_min_sequential = DEFAULT_LEARNING_MIN_SEQUENTIAL;
9061	9065	learning_min_duration = DEFAULT_LEARNING_MIN_DURATION;
	9066	srtp_replay_protection = DEFAULT_SRTP_REPLAY_PROTECTION;
9062	9067
9063	9068	/** This resource is not "reloaded" so much as unloaded and loaded again.
9064	9069	* In the case of the TURN related variables, the memory referenced by a

9137	9142	learning_min_sequential = DEFAULT_LEARNING_MIN_SEQUENTIAL;
9138	9143	}
9139	9144	learning_min_duration = CALC_LEARNING_MIN_DURATION(learning_min_sequential);
	9145	}
	9146	if ((s = ast_variable_retrieve(cfg, "general", "srtpreplayprotection"))) {
	9147	srtp_replay_protection = ast_true(s);
9140	9148	}
9141	9149	#ifdef HAVE_PJPROJECT
9142	9150	if ((s = ast_variable_retrieve(cfg, "general", "icesupport"))) {

+3

-2

res/res_srtp.c less more

363	363	}
364	364
365	365	/* Vtable functions */
366		static int ast_srtp_unprotect(struct ast_srtp srtp, void buf, int *len, int rtcp)
	366	static int ast_srtp_unprotect(struct ast_srtp srtp, void buf, int *len, int flags)
367	367	{
368	368	int res = 0;
369	369	int i;
370		int retry = 0;
	370	int rtcp = (flags & 0x01) >> 0;
	371	int retry = (flags & 0x02) >> 1;
371	372	struct ast_rtp_instance_stats stats = {0,};
372	373
373	374	tryagain:

+7

-0

res/res_stasis_playback.c less more

488	488	for (i = 0; i < media_count; i++) {
489	489	char *media_uri;
490	490
	491	if (ast_strlen_zero(media[i])) {
	492	ast_log(LOG_ERROR, "Attempted to play media on channel '%s' but no media URI was provided.\n",
	493	stasis_app_control_get_channel_id(control));
	494	ao2_ref(playback, -1);
	495	return NULL;
	496	}
	497
491	498	media_uri = ast_malloc(strlen(media[i]) + 1);
492	499	if (!media_uri) {
493	500	ao2_ref(playback, -1);

+7

-5

res/res_stasis_snoop.c less more

65	65	struct ast_str *app;
66	66	/! \brief Snoop channel /
67	67	struct ast_channel *chan;
	68	/! \brief The channel that the Snoop channel is snooping on /
	69	struct ast_channel *spyee_chan;
68	70	/! \brief Whether the spy capability is active or not /
69	71	unsigned int spy_active:1;
70	72	/! \brief Whether the whisper capability is active or not /
71	73	unsigned int whisper_active:1;
72		/! \brief Uniqueid of the channel this snoop is snooping on /
73		char uniqueid[AST_MAX_UNIQUEID];
74	74	/! \brief A frame of silence to use when the audiohook returns null /
75	75	struct ast_frame silence;
76	76	};

99	99
100	100	ast_free(snoop->app);
101	101
	102	ast_channel_cleanup(snoop->spyee_chan);
102	103	ast_channel_cleanup(snoop->chan);
103	104	}
104	105

133	134	}
134	135	ast_multi_channel_blob_add_channel(payload, "spyer_channel", snoop_snapshot);
135	136
136		spyee_snapshot = ast_channel_snapshot_get_latest(snoop->uniqueid);
	137	spyee_snapshot = ast_channel_snapshot_get_latest(ast_channel_uniqueid(snoop->spyee_chan));
137	138	if (spyee_snapshot) {
138	139	ast_multi_channel_blob_add_channel(payload, "spyee_channel", spyee_snapshot);
139	140	}

353	354	return NULL;
354	355	}
355	356
356		ast_copy_string(snoop->uniqueid, ast_channel_uniqueid(chan), sizeof(snoop->uniqueid));
357
358	357	/* To keep the channel valid on the Snoop structure until it is destroyed we bump the ref up here */
359	358	ast_channel_ref(snoop->chan);
360	359

424	423	return NULL;
425	424	}
426	425
	426	/* Keep a reference to the channel we are spying on */
	427	snoop->spyee_chan = ast_channel_ref(chan);
	428
427	429	publish_chanspy_message(snoop, 1);
428	430
429	431	/* The caller of this has a reference as well */

+41

-27

res/stasis/messaging.c less more

288	288	return json_obj;
289	289	}
290	290
	291	static void dispatch_message(struct message_subscription sub, const char endpoint_name, struct ast_json *json_msg)
	292	{
	293	int i;
	294
	295	ast_debug(3, "Dispatching message to subscription %s for endpoint %s\n",
	296	sub->token,
	297	endpoint_name);
	298	for (i = 0; i < AST_VECTOR_SIZE(&sub->applications); i++) {
	299	struct application_tuple *tuple = AST_VECTOR_GET(&sub->applications, i);
	300
	301	tuple->callback(endpoint_name, json_msg, tuple->pvt);
	302	}
	303	}
	304
291	305	static int handle_msg_cb(struct ast_msg *msg)
292	306	{
	307	/* We have at most 3 subscriptions: TECH_WILDCARD, tech itself, and endpoint. */
	308	struct message_subscription *matching_subscriptions[3];
293	309	struct message_subscription *sub;
294		int i;
	310	int i, j;
	311	int result;
295	312	char buf[256];
296	313	const char *endpoint_name;
297	314	struct ast_json *json_msg;
298	315
299	316	msg_to_endpoint(msg, buf, sizeof(buf));
300
	317	endpoint_name = buf;
	318	json_msg = msg_to_json(msg);
	319	if (!json_msg) {
	320	return -1;
	321	}
	322	result = -1;
	323
	324	/* Find subscriptions to TECH_WILDCARD and to the endpoint's technology. */
301	325	ast_rwlock_rdlock(&tech_subscriptions_lock);
302		for (i = 0; i < AST_VECTOR_SIZE(&tech_subscriptions); i++) {
	326	for (i = 0, j = 0; i < AST_VECTOR_SIZE(&tech_subscriptions) && j < 2; i++) {
303	327	sub = AST_VECTOR_GET(&tech_subscriptions, i);
304	328
305	329	if (!sub) {

308	332
309	333	if (!strcmp(sub->token, TECH_WILDCARD)
310	334	\|\| !strncasecmp(sub->token, buf, strlen(sub->token))) {
311		ast_rwlock_unlock(&tech_subscriptions_lock);
312		ao2_bump(sub);
313		endpoint_name = buf;
314		goto match;
	335	ao2_ref(sub, +1);
	336	matching_subscriptions[j++] = sub;
315	337	}
316	338	}
317	339	ast_rwlock_unlock(&tech_subscriptions_lock);
318	340
	341	/* Find the subscription to this particular endpoint. */
319	342	sub = ao2_find(endpoint_subscriptions, buf, OBJ_SEARCH_KEY);
320	343	if (sub) {
321		endpoint_name = buf;
322		goto match;
323		}
324
325		return -1;
326
327		match:
328		ast_debug(3, "Dispatching message for %s\n", endpoint_name);
329
330		json_msg = msg_to_json(msg);
331		if (!json_msg) {
	344	matching_subscriptions[j++] = sub;
	345	}
	346
	347	/* Dispatch the message to all matching subscriptions. */
	348	for (i = 0; i < j; i++) {
	349	sub = matching_subscriptions[i];
	350
	351	dispatch_message(sub, endpoint_name, json_msg);
	352
332	353	ao2_ref(sub, -1);
333		return -1;
334		}
335
336		for (i = 0; i < AST_VECTOR_SIZE(&sub->applications); i++) {
337		struct application_tuple *tuple = AST_VECTOR_GET(&sub->applications, i);
338
339		tuple->callback(endpoint_name, json_msg, tuple->pvt);
	354	result = 0;
340	355	}
341	356
342	357	ast_json_unref(json_msg);
343		ao2_ref(sub, -1);
344		return 0;
	358	return result;
345	359	}
346	360
347	361	struct ast_msg_handler ari_msg_handler = {

+1

-0

tests/test_http_media_cache.c less more

696	696	.load = load_module,
697	697	.reload = reload_module,
698	698	.unload = unload_module,
	699	.requires = "res_http_media_cache",
699	700	);

+37

-0

third-party/pjproject/patches/0070-fix-incorrect-copying-when-creating-cancel.patch less more

	0	From ce18018cc17bef8f80c08686e3a7b28384ef3ba5 Mon Sep 17 00:00:00 2001
	1	From: sauwming <ming@teluu.com>
	2	Date: Mon, 12 Oct 2020 13:31:25 +0800
	3	Subject: [PATCH] Fix incorrect copying of destination info when creating
	4	CANCEL (#2546)
	5
	6	---
	7	pjsip/src/pjsip/sip_util.c \| 10 +++++-----
	8	1 file changed, 5 insertions(+), 5 deletions(-)
	9
	10	diff --git a/pjsip/src/pjsip/sip_util.c b/pjsip/src/pjsip/sip_util.c
	11	index d10a6fa30..a1bf878ea 100644
	12	--- a/pjsip/src/pjsip/sip_util.c
	13	+++ b/pjsip/src/pjsip/sip_util.c
	14	@@ -779,14 +779,14 @@ PJ_DEF(pj_status_t) pjsip_endpt_create_cancel( pjsip_endpoint *endpt,
	15	pjsip_hdr_clone(cancel_tdata->pool, req_tdata->saved_strict_route);
	16	}
	17
	18	- /* Copy the destination host name from the original request */
	19	- pj_strdup(cancel_tdata->pool, &cancel_tdata->dest_info.name,
	20	- &req_tdata->dest_info.name);
	21	-
	22	- /* Finally copy the destination info from the original request */
	23	+ /* Copy the destination info from the original request */
	24	pj_memcpy(&cancel_tdata->dest_info, &req_tdata->dest_info,
	25	sizeof(req_tdata->dest_info));
	26
	27	+ /* Finally, copy the destination host name from the original request */
	28	+ pj_strdup(cancel_tdata->pool, &cancel_tdata->dest_info.name,
	29	+ &req_tdata->dest_info.name);
	30	+
	31	/* Done.
	32	* Return the transmit buffer containing the CANCEL request.
	33	*/
	34	--
	35	2.25.1
	36

+33

-0

third-party/pjproject/patches/0080-fix-sdp-neg-modify-local-offer.patch less more

	0	diff --git a/pjmedia/src/pjmedia/sdp_neg.c b/pjmedia/src/pjmedia/sdp_neg.c
	1	index 3b85b4273..a14009662 100644
	2	--- a/pjmedia/src/pjmedia/sdp_neg.c
	3	+++ b/pjmedia/src/pjmedia/sdp_neg.c
	4	@@ -304,7 +304,6 @@ PJ_DEF(pj_status_t) pjmedia_sdp_neg_modify_local_offer2(
	5	{
	6	pjmedia_sdp_session *new_offer;
	7	pjmedia_sdp_session *old_offer;
	8	- char media_used[PJMEDIA_MAX_SDP_MEDIA];
	9	unsigned oi; /* old offer media index */
	10	pj_status_t status;
	11
	12	@@ -323,8 +322,19 @@ PJ_DEF(pj_status_t) pjmedia_sdp_neg_modify_local_offer2(
	13	/* Change state to STATE_LOCAL_OFFER */
	14	neg->state = PJMEDIA_SDP_NEG_STATE_LOCAL_OFFER;
	15
	16	+ /* When there is no active local SDP in state PJMEDIA_SDP_NEG_STATE_DONE,
	17	+ * it means that the previous initial SDP nego must have been failed,
	18	+ * so we'll just set the local SDP offer here.
	19	+ */
	20	+ if (!neg->active_local_sdp) {
	21	+ neg->initial_sdp_tmp = NULL;
	22	+ neg->initial_sdp = pjmedia_sdp_session_clone(pool, local);
	23	+ neg->neg_local_sdp = pjmedia_sdp_session_clone(pool, local);
	24	+
	25	+ return PJ_SUCCESS;
	26	+ }
	27	+
	28	/* Init vars */
	29	- pj_bzero(media_used, sizeof(media_used));
	30	old_offer = neg->active_local_sdp;
	31	new_offer = pjmedia_sdp_session_clone(pool, local);
	32