asterisk

Commit Graph

Author	SHA1	Message	Date
Sean Bright	9ff548f1db	app_voicemail: Prevent deadlocks when out of ODBC database connections ASTERISK-28992 #close Change-Id: Ia7d608924036139ee2520b840d077762d02668d0	5 years ago
Kevin Harwell	d9aef0e6e5	pbx_realtime: wrong type stored on publish of ast_channel_snapshot_type A prior patch segmented channel snapshots, and changed the underlying data object type associated with ast_channel_snapshot_type stasis messages. Prior to Asterisk 18 it was a type ast_channel_snapshot, but now it type ast_channel_snapshot_update. When publishing ast_channel_snapshot_type in pbx_realtime the ast_channel_snapshot was being passed in as the message data object. When a handler, expecting a data object type of ast_channel_snapshot_update, dereferenced this value a crash would occur. This patch makes it so pbx_realtime now uses the expected type, and channel snapshot publish method when publishing. ASTERISK-29168 #close Change-Id: I9a2cfa0ec285169317f4b9146e4027da8a4fe896	5 years ago
Sean Bright	68d3d3af6f	asterisk: Export additional manager functions Rename check_manager_enabled() and check_webmanager_enabled() to begin with ast_ so that the symbols are automatically exported by the linker. ASTERISK~29184 Change-Id: I85762b9a5d14500c15f6bad6507138c8858644c9	5 years ago
Nick French	3c8598ffef	res_pjsip: Prevent segfault in UDP registration with flow transports Segfault occurs during outbound UDP registration when all transport states are being iterated over. The transport object in the transport is accessed, but flow transports have a NULL transport object. Modify to not iterate over any flow transport ASTERISK-29210 #close Change-Id: If28dc3a18bdcbd0a49598b09b7fe4404d45c996a	5 years ago
Richard Mudgett	3d379845e6	chan_vpb.cc: Fix compile errors. Fix the usual compile problem when someone adds a new callback to struct ast_channel_tech. Change-Id: I9bdeb8a8cc65f03b2d6e4f2eb5809af47c906c32	5 years ago
Richard Mudgett	027f4e3a21	res_pjsip_session.c: Fix compiler warnings. AST_VECTOR_SIZE() returns a size_t. This is not always equivalent to an unsigned long on all machines. Change-Id: I0a4189a104e6e3a2e2273de06620eaef19df9338	5 years ago
Sungtae Kim	d8b7a6f599	res_pjsip_session: Fixed NULL active media topology handle Added NULL pointer check to prevent Asterisk crash. ASTERISK-29215 Change-Id: If07e50ea8d78cb610af9195fc13b5dca4bfcef95	5 years ago
Torrey Searle	a7aea71e60	res/res_pjsip_diversion: prevent crash on tel: uri in History-Info Add a check to see if the URI is a Tel URI and prevent crashing on trying to retrieve the reason parameter. ASTERISK-29191 ASTERISK-29219 Change-Id: I0320aa205f22cda511d60a2edf2b037e8fd6cc37	5 years ago
Sean Bright	13682210e2	app_chanspy: Spyee information missing in ChanSpyStop AMI Event The documentation in the wiki says there should be spyee-channel information elements in the ChanSpyStop AMI event. https://wiki.asterisk.org/wiki/x/Xc5uAg However, this is not the case in Asterisk <= 16.10.0 Version. We're using these Spyee* arguments since Asterisk 11.x, so these arguments vanished in Asterisk 12 or higher. For maximum compatibility, we still send the ChanSpyStop event even if we are not able to find any 'Spyee' information. ASTERISK-28883 #close Change-Id: I81ce397a3fd614c094d043ffe5b1b1d76188835f	5 years ago
Sungtae Kim	4b450b4334	res_ari: Fix wrong media uri handle for channel play Fixed wrong null object handle in /channels/<channel_id>/play request handler. ASTERISK-29188 Change-Id: I6691c640247a51ad15f23e4a203ca8430809bafe	5 years ago
Pirmin Walthert	7a6cfde4db	res_pjsip_nat.c: Create deep copies of strings when appropriate In rewrite_uri asterisk was not making deep copies of strings when changing the uri. This was in some cases causing garbage in the route header and in other cases even crashing asterisk when receiving a message with a record-route header set. Thanks to Ralf Kubis for pointing out why this happens. A similar problem was found in res_pjsip_transport_websocket.c. Pjproject needs as well to be patched to avoid garbage in CANCEL messages. ASTERISK-29024 #close Change-Id: Ic5acd7fa2fbda3080f5f36ef12e46804939b198b	5 years ago
George Joseph	ccb4951bf8	logger.c: Automatically add a newline to formats that don't have one Scope tracing allows you to not specify a format string or variable, in which case it just prints the indent, file, function, and line number. The trace output automatically adds a newline to the end in this case. If you also have debugging turned on for the module, a debug message is also printed but the standard log functionality which prints it doesn't add the newline so you have messages that don't break correctly. * format_log_message_ap(), which is the common log message formatter for all channels, now adds a newline to the end of format strings that don't already have a newline. ASTERISK-29209 Reported by: Alexander Traud Change-Id: I994a7df27f88df343b7d19f3e81a4b562d9d41da	5 years ago
Joshua C. Colp	938a240793	res_pjsip_pidf_digium_body_supplement: Support Sangoma user agent. This adds support for both Digium and Sangoma user agent strings for the Sangoma specific body supplement. Change-Id: Ib99362b24b91d3cbe888d8b2fce3fad5515d9482	5 years ago
Nathan Bruning	0774d9f9aa	res_musiconhold: Don't crash when real-time doesn't return any entries ASTERISK-29211 #close Change-Id: Ifbf0a4f786ab2a52342f9d1a1db4c9907f069877	5 years ago
Joshua C. Colp	5b4e71fa0a	pjsip: Match lifetime of INVITE session to our session. In some circumstances it was possible for an INVITE session to be destroyed while we were still using it. This occurred due to the reference on the INVITE session being released internally as a result of its state changing to DISCONNECTED. This change adds a reference to the INVITE session which is released when our own session is destroyed, ensuring that the INVITE session remains valid for the lifetime of our session. ASTERISK-29022 Change-Id: I300c6d9005ff0e6efbe1132daefc7e47ca6228c9	5 years ago
lvl	92fcd4edba	Introduce astcachedir, to be used for temporary bucket files As described in the issue, /tmp is not a suitable location for a large amount of cached media files, since most distributions make /tmp a RAM-based tmpfs mount with limited capacity. I opted for a location that can be configured separately, as opposed to using a subdirectory of spooldir, given the different storage profile (transient files vs files that might stay there indefinitely). This commit just makes the cache directory configurable, but leaves it at /tmp by default, to ensure backwards compatibility. A future commit that only targets master could change the default location to something more sensible such as /var/tmp/asterisk. At that point, the cachedir could be created and cleaned up during uninstall by the Makefile script. ASTERISK-29143 Change-Id: Ic54e95199405abacd9e509cef5f08fa14c510b5d	5 years ago
Sean Bright	f39d5ea7cd	res_http_media_cache.c: Set reasonable number of redirects By default libcurl does not follow redirects, so we explicitly enable it by setting CURLOPT_FOLLOWLOCATION. Once that is enabled, libcurl will follow up to CURLOPT_MAXREDIRS redirects, which by default is configured to be unlimited. This patch sets CURLOPT_MAXREDIRS to a more reasonable default (8). If we determine at some point that this needs to be increased on configurable it is a trivial change. ASTERISK-29173 #close Change-Id: I4925ebbcf0c7d728bb9252b3795b3479ae225b30	5 years ago
Sean Bright	f9438e6457	media_cache: Fix reference leak with bucket file metadata Change-Id: Ia0e4124110df613ce5fdfa9ef8780016ebaa52c6	5 years ago
Stanislav	6a85dc860f	res_pjsip_stir_shaken: Fix module description the 'J' is missing in module description. "PSIP STIR/SHAKEN Module for Asterisk" -> "PJSIP STIR/SHAKEN Module for Asterisk" ASTERISK-29175 #close Change-Id: I17da008540ee2e8496b644d05f995b320b54ad7a	5 years ago
Joshua C. Colp	fd57fae048	voicemail: add option 'e' to play greetings as early media When using this option, answering the channel is deferred until all prompts/greetings have been played and the caller is about to leave their message. ASTERISK-29118 #close Change-Id: I41b9f0428783c0bd697c8c994f906d1e75ce9ddb	5 years ago
Alexander Traud	bf9f0f13c4	loader: Sync load- and build-time deps. In MODULEINFO, each depend has to be listed in .requires of AST_MODULE_INFO. ASTERISK-29148 Change-Id: I254dd33194ae38d2877b8021c57c2a5deb6bbcd2	5 years ago
Sean Bright	994fbdaf48	CHANGES: Remove already applied CHANGES update Change-Id: Iee7163bc732d58c5cbaa2cfab1f5aab4a412060a	5 years ago
Alexander Greiner-Baer	c79bd583d9	res_pjsip: set Accept-Encoding to identity in OPTIONS response RFC 3261 says that the Accept-Encoding header should be present in an options response. Permitted values according to RFC 2616 are only compression algorithms like gzip or the default identity encoding. Therefore "text/plain" is not a correct value here. As long as the header is hard coded, it should be set to "identity". Without this fix an Alcatel OmniPCX periodically logs warnings like "[sip_acceptIncorrectHeader] Header Accept-Encoding is malformed" on a SIP Trunk. ASTERISK-29165 #close Change-Id: I0aa2211ebf0b4c2ed554ac7cda794523803a3840	5 years ago
Alexander Traud	e884d935f6	chan_sip: Remove unused sip_socket->port. 12 years ago, with ASTERISK_12115 the last four get/uses of socket.port vanished. However, the struct member itself and all seven set/uses remained as dead code. ASTERISK-28798 Change-Id: Ib90516a49eca3d724a70191278aaf2144fb58c59	5 years ago
Boris P. Korzun	33e3542132	bridge_basic: Fixed setup of recall channels Fixed a bug (like a typo) in retransfer_enter() at main/bridge_basic.c:2641. common_recall_channel_setup() setups common things on the recalled transfer target, but used same target as source instead trasfered. ASTERISK-29161 #close Change-Id: Ieb549654a621c38b1ad5e9d15b9f18823d9cc31f	5 years ago
Alexander Traud	6e1fb58183	modules.conf: Align the comments for more conclusiveness. Change-Id: I79cc693cd5a6e5dd7d403b7e91d970ff1ddf8306	5 years ago
George Joseph	2413598705	app_queue: Fix deadlock between update and show queues Operations that update queues when shared_lastcall is set lock the queue in question, then have to lock the queues container to find the other queues with the same member. On the other hand, __queues_show (which is called by both the CLI and AMI) does the reverse. It locks the queues container, then iterates over the queues locking each in turn to display them. This creates a deadlock. * Moved queue print logic from __queues_show to a separate function that can be called for a single queue. * Updated __queues_show so it doesn't need to lock or traverse the queues container to show a single queue. * Updated __queues_show to snap a copy of the queues container and iterate over that instead of locking the queues container and iterating over it while locked. This prevents us from having to hold both the container lock and the queue locks at the same time. This also allows us to sort the queue entries. ASTERISK-29155 Change-Id: I78d4dc36728c2d7bc187b97d82673fc77f2bcf41	5 years ago
Asterisk Development Team	98d1537c1e	Update CHANGES and UPGRADE.txt for 18.1.0	5 years ago
George Joseph	860e40dd80	res_pjsip_outbound_registration.c: Use our own scheduler and other stuff * Instead of using the pjproject timer heap, we now use our own pjsip_scheduler. This allows us to more easily debug and allows us to see times in "pjsip show/list registrations" as well as being able to see the registrations in "pjsip show scheduled_tasks". * Added the last registration time, registration interval, and the next registration time to the CLI output. * Removed calls to pjsip_regc_info() except where absolutely necessary. Most of the calls were just to get the server and client URIs for log messages so we now just save them on the client_state object when we create it. * Added log messages where needed and updated most of the existong ones to include the registration object name at the start of the message. Change-Id: I4534a0fc78c7cb69f23b7b449dda9748c90daca2	5 years ago
George Joseph	569fc28966	pjsip_scheduler.c: Add type ONESHOT and enhance cli show command * Added a ONESHOT type that never reschedules. * Added "like" capability to "pjsip show scheduled_tasks" so you can do the following: CLI> pjsip show scheduled_tasks like outreg PJSIP Scheduled Tasks: Task Name Interval Times Run ... ============================================= ========= ========= ... pjsip/outreg/testtrunk-reg-0-00000074 50.000 oneshot ... pjsip/outreg/voipms-reg-0-00000073 110.000 oneshot ... * Fixed incorrect display of "Next Start". * Compacted the displays of times in the CLI. * Added two new functions (ast_sip_sched_task_get_times2, ast_sip_sched_task_get_times_by_name2) that retrieve the interval, next start time, and next run time in addition to the times already returned by ast_sip_sched_task_get_times(). Change-Id: Ie718ca9fd30490b8a167bedf6b0b06d619dc52f3	5 years ago
Alexei Gradinari	da0f2ea99e	sched: AST_SCHED_REPLACE_UNREF can lead to use after free of data The data can be freed if the old object '_data' is the same object as new 'data'. Because at first the object is unreferenced which can lead to destroying it. This could happened in res_pjsip_pubsub when the publication is updated which could lead to segfault in function publish_expire. Change-Id: I0164f57c387243510bdbd2f8dcf33377b6c202da	5 years ago
Alexander Traud	5a6037778b	res_pjsip/config_transport: Load and run without OpenSSL. ASTERISK-28933 Reported-by: Walter Doekes Change-Id: I65eac49e5b0a79261ea80e2b9b38a836886ed59f	5 years ago
Alexander Traud	be54c7e9ea	res_stir_shaken: Include OpenSSL headers where used actually. This avoids the inclusion of the OpenSSL headers in the public header, which avoids one external library dependency in res_pjsip_stir_shaken. Change-Id: I6a07e2d81d2b5442e24e99b8cc733a99f881dcf4	5 years ago
Dovid Bender	c635c78265	func_curl.c: Allow user to set what return codes constitute a failure. Currently any response from res_curl where we get an answer from the web server, regardless of what the response is (404, 403 etc.) Asterisk currently treats it as a success. This patch allows you to set which codes should be considered as a failure by Asterisk. If say we set failurecodes=404,403 then when using curl in realtime if a server gives a 404 error Asterisk will try to failover to the next option set in extconfig.conf ASTERISK-28825 Reported by: Dovid Bender Code by: Gobinda Paul Change-Id: I94443e508343e0a3e535e51ea6e0562767639987	5 years ago
Kevin Harwell	6baa4b53be	AST-2020-001 - res_pjsip: Return dialog locked and referenced pjproject returns the dialog locked and with a reference. However, in Asterisk the method that handles this decrements the reference and removes the lock prior to returning. This makes it possible, under some circumstances, for another thread to free said dialog before the thread that created it attempts to use it again. Of course when the thread that created it tries to use a freed dialog a crash can occur. This patch makes it so Asterisk now returns the newly created dialog both locked, and with an added reference. This allows the caller to de-reference, and unlock the dialog when it is safe to do so. In the case of a new SIP Invite the lock, and reference are now held for the entirety of the new invite handling process. Otherwise it's possible for the dialog, or its dependent objects, like the transaction, to disappear. For example if there is a TCP transport error. ASTERISK-29057 #close Change-Id: I5ef645a47829596f402cf383dc02c629c618969e	5 years ago
Ben Ford	82325ba58b	AST-2020-002 - res_pjsip: Stop sending INVITEs after challenge limit. If Asterisk sends out and INVITE and receives a challenge with a different nonce value each time, it will continually send out INVITEs, even if the call is hung up. The endpoint must be configured for outbound authentication in order for this to occur. A limit has been set on outbound INVITEs so that, once reached, Asterisk will stop sending INVITEs and the transaction will terminate. ASTERISK-29013 Change-Id: I2d001ca745b00ca8aa12030f2240cd72363b46f7	5 years ago
Sean Bright	fe540d0326	sip_to_pjsip.py: Handle #include globs and other fixes * Wildcards in #includes are now properly expanded * Implement operators for Section class to allow sorting ASTERISK-29142 #close Change-Id: I9b9cd95f4cbe5c24506b75d17173c5aa1a83e5df	5 years ago
Alexander Traud	e0ee53dc9c	Compiler fixes for GCC with -Og ASTERISK-29144 Change-Id: I2a72c072083b4492a223c6f9d73d21f4f424db62	5 years ago
Alexander Traud	2dacadd9df	Compiler fixes for GCC with -Os ASTERISK-29145 Change-Id: I9af705f2b9725c53141aef5d0ff512a1800f073c	5 years ago
Alexander Traud	f86af1fbd0	Compiler fixes for GCC when printf %s is NULL ASTERISK-29146 Change-Id: Ib04bdad87d729f805f5fc620ef9952f58ea96d41	5 years ago
Alexander Traud	5b25c75d7b	chan_sip: On authentication, pick MD5 for sure. RFC 8760 added new digest-access-authentication schemes. Testing revealed that chan_sip does not pick MD5 if several schemes are offered by the User Agent Server (UAS). This change does not implement any of the new schemes like SHA-256. This change makes sure, MD5 is picked so UAS with SHA-2 enabled, like the service www.linphone.org/freesip, can still be used. This should have worked since day one because SIP/2.0 already envisioned several schemes (see RFC 3261 and its augmented BNF for 'algorithm' which includes 'token' as third alternative; note: if 'algorithm' was not present, MD5 is still assumed even in RFC 7616). Change-Id: I61ca0b1f74b5ec2b5f3062c2d661cafeaf597fcd	5 years ago
Walter Doekes	fb3b14ab7d	main/say: Work around gcc 9 format-truncation false positive Version: gcc (Ubuntu 9.3.0-10ubuntu2) 9.3.0 Warning: say.c:2371:24: error: ‘%d’ directive output may be truncated writing between 1 and 11 bytes into a region of size 10 [-Werror=format-truncation=] 2371 \| snprintf(buf, 10, "%d", num); say.c:2371:23: note: directive argument in the range [-2147483648, 9] That's not possible though, as the if() starts out checking for (num < 0), making this Warning a false positive. (Also replaced some else<TAB>if with else<SP>if while in the vicinity.) Change-Id: Ic7a70120188c9aa525a6d70289385bfce878438a	5 years ago
Kevin Harwell	439f7bb848	res_pjsip, res_pjsip_session: initialize local variables This patch initializes a couple of local variables to some default values. Interestingly, in the 'pj_status_t dlg_status' case the value not being initialized caused memory to grow, and not be recovered, in the off nominal path (at least on my machine). Change-Id: I22ee65e1e1bff8efacea8a167c6c8428898523f7	5 years ago
Alexander Traud	f89531cb98	install_prereq: Add GMime 3.0. Ubuntu 20.10 does not come with GMime 2.6. Ubuntu 16.04 LTS does not come with GMime 3.0. aptitude ignores any missing package. Therefore, it installs the correct package(s). However, in Ubuntu 18.04 LTS and Ubuntu 20.04 LTS, both versions are installed alongside although only one is really needed. Change-Id: Ic58aa9f2e131d94671f286f17dbd61e1ccbabcb7	5 years ago
Nick French	f041763e3b	res_pjsip_session: Restore calls to ast_sip_message_apply_transport() Commit `44bb0858cb` ("debugging: Add enough to choke a mule") accidentally removed calls to ast_sip_message_apply_transport when it was attempting to just add debugging code. The kiss of death was saying that there were no functional changes in the commit comment. This makes outbound calls that use the 'flow' transport mechanism fail, since this call is used to relay headers into the outbound INVITE requests. ASTERISK-29124 #close Change-Id: I0f3e32c2e8ac415e30b1d29966d75a1546f0526a	5 years ago
Alexander Traud	2773f93154	BuildSystem: Enable Lua 5.4. Note to maintainers: Lua 5.4, Lua 5.3, and Lua 5.2 have not been tested at runtime with pbx_lua. Until then, use the lowest available version of Lua, if you enabled the module pbx_lua at all. Change-Id: Ie5270448b11fcb4e2a53d899e4fe7fea793ce7e0	5 years ago
Sean Bright	6f321b561a	features.conf.sample: Sample sound files incorrectly quoted ASTERISK-29136 #close Change-Id: I3186536d65a50014c8da4780c9224919caa81440	5 years ago
Andrew Siplas	ff33f7f44f	logger.conf.sample: add missing comment mark Add missing comment mark from stock configuration. ASTERISK-29123 #close Change-Id: I4f94eb4544166bca8af4c17fd11edee3c6980620	5 years ago
Joshua C. Colp	412b385de5	res_pjsip: Adjust outgoing offer call pref. This changes the outgoing offer call preference default option to match the behavior of previous versions of Asterisk. The additional advanced codec negotiation options have also been removed from the sample configuration and marked as reserved for future functionality in XML documentation. The codec preference options have also been fixed to enforce local codec configuration. ASTERISK-29109 Change-Id: Iad19347bd5f3d89900c15ecddfebf5e20950a1c2	5 years ago
Kevin Harwell	6255e7976c	Logging: Add debug logging categories Added debug logging categories that allow a user to output debug information based on a specified category. This lets the user limit, and filter debug output to data relevant to a particular context, or topic. For instance the following categories are now available for debug logging purposes: dtls, dtls_packet, ice, rtcp, rtcp_packet, rtp, rtp_packet, stun, stun_packet These debug categories can be enable/disable via an Asterisk CLI command. While this overrides, and outputs debug data, core system debugging is not affected by this patch. Statements still output at their appropriate debug level. As well backwards compatibility has been maintained with past debug groups that could be enabled using the CLI (e.g. rtpdebug, stundebug, etc.). ASTERISK-29054 #close Change-Id: I6e6cb247bb1f01dbf34750b2cd98e5b5b41a1849 (cherry picked from commit `56028426de`)	5 years ago

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