asterisk

Commit Graph

Author	SHA1	Message	Date
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
Sean Bright	a6faa53af0	tcptls.c: Don't close TCP client file descriptors more than once ASTERISK-28430 #close Change-Id: Ib556b0a0c95cca939e956886214ec8d828d89606	5 years ago
Jean Aunis	7ced144867	resource_endpoints.c: memory leak when providing a 404 response When handling a send_message request to a non-existing endpoint, the response's body is overriden and not properly freed. ASTERISK-29108 Change-Id: Ie1d3d70065f80793445b60f5e4a7eb31b4b9c5c8	5 years ago
Sean Bright	abee490639	app_voicemail.c: Document VMSayName interruption behavior ASTERISK-26424 #close Change-Id: I797ad0ed302d0b3d2c90543eff5b7207ed08ecf0	5 years ago
Sean Bright	5a0b19a4f3	pbx.c: On error, ast_add_extension2_lockopt should always free 'data' In the event that the desired extension already exists, ast_add_extension2_lockopt() will free the 'data' it is passed before returning an error, so we should not be freeing it ourselves. Additionally, there were two places where ast_add_extension2_lockopt() could return an error without also freeing the 'data' pointer, so we add that. ASTERISK-29097 #close Change-Id: I904707aae55169feda050a5ed7c6793b53fe6eae	5 years ago
George Joseph	4a049ad510	app_confbridge/bridge_softmix: Add ability to force estimated bitrate app_confbridge now has the ability to set the estimated bitrate on an SFU bridge. To use it, set a bridge profile's remb_behavior to "force" and set remb_estimated_bitrate to a rate in bits per second. The remb_estimated_bitrate parameter is ignored if remb_behavior is something other than "force". Change-Id: Idce6464ff014a37ea3b82944452e56cc4d75ab0a	5 years ago
Jasper van der Neut	08ccfd4588	channels: Don't dereference NULL pointer Check result of ast_translator_build_path against NULL before dereferencing. ASTERISK-29091 Change-Id: Ia3538ea190bd371f70c9dd49984b021765691b29	5 years ago
Holger Hans Peter Freyther	4499fbc819	res_pjsip_sdp_rtp: Fix accidentally native bridging calls Stop advertising RFC2833 support on the rtp_engine when DTMF mode is auto but no tel_event was found inside SDP file. On an incoming call create_rtp will be called and when session->dtmf is set to AST_SIP_DTMF_AUTO, the AST_RTP_PROPERTY_DTMF will be set without looking at the SDP file. Once get_codecs gets called we move the DTMF mode from RFC2833 to INBAND but continued to advertise RFC2833 support. This meant the native_rtp bridge would falsely consider the two channels as compatible. In addition to changing the DTMF mode we now set or remove the AST_RTP_PROPERTY_DTMF. The property is checked in ast_rtp_dtmf_compatible and called by native_rtp_bridge_compatible. ASTERISK-29051 #close Change-Id: I1e0c1e324598a437932c0b7836bcb626aba8e287	5 years ago
lvl	b3b6b5e9f7	res_musiconhold: Load all realtime entries, not just the first ASTERISK-29099 Change-Id: I45636679c0fb5a5f59114c8741626631a604e8a6	5 years ago
Torrey Searle	c470327e6c	res_pjsip_diversion: fix double 181 Arming response to both AST_SIP_SESSION_BEFORE_REDIRECTING and AST_SIP_SESSION_BEFORE_MEDIA causes 302 to to be handled twice, resulting in to 181 being generated. Change-Id: I866e5461564644ffb8a5e12b6f1330b50a7b63ab	5 years ago
Sean Bright	5929e0ccbd	res_musiconhold: Clarify that playlist mode only supports HTTP(S) URLs Change-Id: I41e77a04e4a523f4ed61a7a20b738ffd42be441e	5 years ago
Sean Bright	9b08eddf90	dsp.c: Update calls to ast_format_cmp to check result properly ASTERISK-28311 #close Change-Id: Ib1ce8fc1a8752751f5bf3615c59245532dfd9aa2	5 years ago
Sean Bright	d0644faa5a	res_musiconhold: Start playlist after initial announcement Only track our sample offset if we are playing a non-announcement file, otherwise we will skip that number of samples when we start playing the first MoH file. ASTERISK-24329 #close Change-Id: Ib6b3c84fcaa1063889ab38ba7e7fc50050a3ccfc	5 years ago
Joshua C. Colp	9eeb40af33	res_pjsip_session: Fix stream name memory leak. When constructing a stream name based on the media type and position the allocated name was not being freed causing a leak. Change-Id: I52510863b24a2f531f0a55b440bb2c81844029de	5 years ago
Sean Bright	28c88e8fe2	func_curl.c: Prevent crash when using CURLOPT(httpheader) Because we use shared thread-local cURL instances, we need to ensure that the state of the cURL instance is correct before each invocation. In the case of custom headers, we were not resetting cURL's internal HTTP header pointer which could result in a crash if subsequent requests do not configure custom headers. ASTERISK-29085 #close Change-Id: I8b4ab34038156dfba613030a45f10e932d2e992d	5 years ago
Joshua C. Colp	957aff751d	res_pjsip_session: Fix session reference leak. The ast_sip_dialog_get_session function returns the session with reference count increased. This was not taken into account and was causing sessions to remain around when they should not be. ASTERISK-29089 Change-Id: I430fa721b0a824311a59effec6056e9ec528e3e8	5 years ago
Michal Hajek	2bce21da88	res_stasis.c: Add compare function for bridges moh container Sometimes not play MOH on bridge. ASTERISK-29081 Reported-by: Michal Hajek <michal.hajek@daktela.com> Change-Id: I760c73e0c9be1d340303b5d1c18a00c4759e8232	5 years ago
George Joseph	99bd7d95de	logger.h: Fix ast_trace to respect scope_level ast_trace() was always emitting messages when it's level was set to -1 because it was ignoring scope_level. Change-Id: I849c8f4f4613899c37f82be0202024e7d117e506	5 years ago
Sean Bright	c90c182932	audiosocket: Fix module menuselect descriptions The module description needs to be on the same line as the AST_MODULE_INFO or it is not parsed correctly. Change-Id: I9ba11df1415369790e8656fcb527bb2749373c21	5 years ago
George Joseph	fdc13060df	bridge_softmix/sfu_topologies_on_join: Ignore topology change failures When a channel joins a bridge, we do topology change requests on all existing channels to add the new participant to them. However the announcer channel will return an error because it doesn't support topology in the first place. Unfortunately, there doesn't seem to be a reliable way to tell if the error is expected or not so the error is ignored for all channels. If the request fails on a "real" channel, that channel just won't get the new participant's video. Change-Id: Ic95db4683f27d224c1869fe887795d6b9fdea4f0	5 years ago
Sean Bright	6f32c254be	res_pjsip_session.c: Fix build when TEST_FRAMEWORK is not defined Change-Id: Id4852c26e9c412af8e37b5dd3c15da9453ad3276	5 years ago

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