asterisk

Commit Graph

Author	SHA1	Message	Date
Sean Bright	ddbf3a7f73	media_cache: Fix reference leak with bucket file metadata Change-Id: Ia0e4124110df613ce5fdfa9ef8780016ebaa52c6	5 years ago
Stanislav	159522003a	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	15566494f9	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	4c79bc19d1	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	a360150ee0	CHANGES: Remove already applied CHANGES update Change-Id: Iee7163bc732d58c5cbaa2cfab1f5aab4a412060a	5 years ago
Alexander Traud	f667c5a781	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
Alexander Greiner-Baer	a8f6238cc8	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
Boris P. Korzun	89d3de37ca	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	d1a78e047d	modules.conf: Align the comments for more conclusiveness. Change-Id: I79cc693cd5a6e5dd7d403b7e91d970ff1ddf8306	5 years ago
George Joseph	8d8c9db618	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	3366956139	Update CHANGES and UPGRADE.txt for 16.15.0	5 years ago
George Joseph	e1fd51cd2c	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	80f116c156	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	728cd55cde	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	ddfb76a864	res_pjsip/config_transport: Load and run without OpenSSL. ASTERISK-28933 Reported-by: Walter Doekes Change-Id: I65eac49e5b0a79261ea80e2b9b38a836886ed59f	5 years ago
Alexander Traud	277aa0ced6	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	5046e1fb06	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	8973fe5cf3	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	58aa6a7057	AST-2020-002 - res_pjsip: Stop sending INVITEs after challenge limit. If Asterisk sends out an INVITE and receives a challenge with a different nonce value each time, it will continuously send out INVITEs, even if the call is hung up. The endpoint must be configured for outbound authentication 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	e067d5c8fd	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	13b56c4be6	Compiler fixes for GCC with -Og ASTERISK-29144 Change-Id: I2a72c072083b4492a223c6f9d73d21f4f424db62	5 years ago
Alexander Traud	334661601a	Compiler fixes for GCC when printf %s is NULL ASTERISK-29146 Change-Id: Ib04bdad87d729f805f5fc620ef9952f58ea96d41	5 years ago
Alexander Traud	92ca48d54c	Compiler fixes for GCC with -Os ASTERISK-29145 Change-Id: I9af705f2b9725c53141aef5d0ff512a1800f073c	5 years ago
Alexander Traud	951ce0524d	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	f98eed17c1	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	92e1de458a	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	65426f4312	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
Alexander Traud	fb721ce82c	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	e326b133dc	features.conf.sample: Sample sound files incorrectly quoted ASTERISK-29136 #close Change-Id: I3186536d65a50014c8da4780c9224919caa81440	5 years ago
Asterisk Development Team	69356a7895	Update CHANGES and UPGRADE.txt for 16.14.0	5 years ago
Andrew Siplas	606bd35060	logger.conf.sample: add missing comment mark Add missing comment mark from stock configuration. ASTERISK-29123 #close Change-Id: I4f94eb4544166bca8af4c17fd11edee3c6980620	5 years ago
Kevin Harwell	e051806e80	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
Jean Aunis	0b835f2156	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	d0313d8b12	tcptls.c: Don't close TCP client file descriptors more than once ASTERISK-28430 #close Change-Id: Ib556b0a0c95cca939e956886214ec8d828d89606	5 years ago
Ben Ford	681a1624b5	utils.c: NULL terminate ast_base64decode_string. With the addition of STIR/SHAKEN, the function ast_base64decode_string was added for convenience since there is a lot of converting done during the STIR/SHAKEN process. This function returned the decoded string for you, but did not NULL terminate it, causing some issues (specifically with MALLOC_DEBUG). Now, the returned string is NULL terminated, and the documentation has been updated to reflect this. Change-Id: Icdd7d05b323b0c47ff6ed43492937a03641bdcf5	5 years ago
Ben Ford	df7c4ed0ed	res_stir_shaken: Fix memory allocation error in curl.c Fixed a memory allocation that was not passing in the correct size for the struct in curl.c. Change-Id: I5fb92fbbe84b075fa6aefa2423786df80e114c3a	5 years ago
Ben Ford	21ab0a450b	res_stir_shaken: Add stir_shaken option and general improvements. Added a new configuration option for PJSIP endpoints - stir_shaken. If set to yes, then STIR/SHAKEN support will be added to inbound and outbound INVITEs. The default is no. Alembic has been updated to include this option. Previously the dialplan function was not trimming the whitespace from the parameters it recieved. Now it does. Also added a conditional that, when TEST_FRAMEWORK is enabled, the timestamp in the identity header will be overlooked. This is just for testing, since the testsuite will rely on a SIPp scenario with a preset identity header to trigger the MISMATCH result. Change-Id: I43d67f1489b8c1c5729ed3ca8d71e35ddf438df1	5 years ago
Ben Ford	d979bdf87a	res_stir_shaken: Add outbound INVITE support. Integrated STIR/SHAKEN support with outgoing INVITEs. When an INVITE is sent, the caller ID will be checked to see if there is a certificate that corresponds to it. If so, that information will be retrieved and an Identity header will be added to the SIP message. The format is: header.payload.signature;info=<public_key_url>alg=ES256;ppt=shaken Header, payload, and signature are all BASE64 encoded. The public key URL is retrieved from the certificate. Currently the algorithm and ppt are ES256 and shaken, respectively. This message is signed and can be used for verification on the receiving end. Two new configuration options have been added to the certificate object: attestation and origid. The attestation is required and must be A, B, or C. origid is the origination identifier. A new utility function has been added as well that takes a string, allocates space, BASE64 encodes it, then returns it, eliminating the need to calculate the size yourself. Change-Id: I1f84d6a5839cb2ed152ef4255b380cfc2de662b4	5 years ago
Ben Ford	746ce16b16	res_stir_shaken: Add inbound INVITE support. Integrated STIR/SHAKEN support with incoming INVITES. Upon receiving an INVITE, the Identity header is retrieved, parsing the message to verify the signature. If any of the parsing fails, AST_STIR_SHAKEN_VERIFY_NOT_PRESENT will be added to the channel for this caller ID. If verification itself fails, AST_STIR_SHAKEN_VERIFY_SIGNATURE_FAILED will be added. If anything in the payload does not line up with the SIP signaling, AST_STIR_SHAKEN_VERIFY_MISMATCH will be added. If all of the above steps pass, then AST_STIR_SHAKEN_VERIFY_PASSED will be added, completing the verification process. A new config option has been added to the general section for stir_shaken.conf. "signature_timeout" is the amount of time a signature will be considered valid. If an INVITE is received and the amount of time between when it was received and when it was signed is greater than signature_timeout, verification will fail. Some changes were also made to signing and verification. There was an error where the whole JSON string was being signed rather than the header combined with the payload. This has been changed to sign the correct thing. Verification has been changed to do this as well, and the unit tests have been updated to reflect these changes. A couple of utility functions have also been added. One decodes a BASE64 string and returns the decoded string, doing all the length calculations for you. The other retrieves a string value from a header in a rdata object. Change-Id: I855f857be3d1c63b64812ac35d9ce0534085b913	5 years ago
Ben Ford	9d7628829c	res_stir_shaken: Add unit tests for signing and verification. Added two unit tests, one for signing and another for verifying. stir_shaken_sign checks to make sure that all the required parameters are passed in and then signs the actual payload. If a signature is produced and a payload returned as a result, the test passes. stir_shaken_verify takes the signature from a signed payload to verify. This unit test also verifies that all the required information is passed in, and then attempts to verify the signature. If verification is successful and a payload is returned, the test passes. Change-Id: I9fa43380f861ccf710cd0f6b6c102a517c86ea13	5 years ago
Ben Ford	035b463c93	res_stir_shaken: Added dialplan function and API call. Adds the "STIR_SHAKEN" dialplan function and an API call to add a STIR_SHAKEN verification result to a channel. This information will be held in a datastore on the channel that can later be queried through the "STIR_SHAKEN" dialplan funtion to get information on STIR_SHAKEN results including identity, attestation, and verify_result. Here are some examples: STIR_SHAKEN(count) STIR_SHAKEN(0, identity) STIR_SHAKEN(1, attestation) STIR_SHAKEN(2, verify_result) Getting the count can be used to iterate through the results and pull information by specifying the index and the field you want to retrieve. Change-Id: Ice6d52a3a7d6e4607c9c35b28a1f7c25f5284a82	5 years ago
Joshua C. Colp	0392a8e620	res_stir_shaken: Use ast_asprintf for creating file path. Change-Id: Ice5d92ecea2f1101c80487484f48ef98be2f1824	5 years ago
Ben Ford	70af7e1311	res_stir_shaken: Implemented signature verification. There are a lot of moving parts in this patch, but the focus of it is on the verification of the signature using a public key located at the public key URL provided in the JSON payload. First, we check the database to see if we have already downloaded the key. If so, check to see if it has expired. If it has, redownload from the URL. If we don't have an entry in the database, just go ahead and download the public key. The expiration is tested each time we download the file. After that, read the public key from the file and use it to verify the signature. All sanity checking is done when the payload is first received, so the verification is complete once this point is reached. The XML has also been added since a new config option was added to general (curl_timeout). The maximum amount of time to wait for a download can be configured through this option, with a low value by default. Change-Id: I3ba4c63880493bf8c7d17a9cfca1af0e934d1a1c	5 years ago
Alexander Traud	971b125fc0	res_stir_shaken: Do not build without OpenSSL. Change-Id: Idba5151a3079f9dcc0076d635422c5df5845114f	5 years ago
Ben Ford	e9ee9a381b	res_stir_shaken: Implemented signing of JSON payload. This change provides functions that take in a JSON payload, verify that the contents contain all the mandatory fields and required values (if any), and signs the payload with the private key. Four fields are added to the payload: x5u, attest, iat, and origid. As of now, these are just placeholder values that will be set to actual values once the logic is implemented for what to do when an actual payload is received, but the functions to add these values have all been implemented and are ready to use. Upon successful signing and the addition of those four values, a ast_stir_shaken_payload is returned, containing other useful information such as the algorithm and signature. Change-Id: I74fa41c0640ab2a64a1a80110155bd7062f13393	5 years ago
Ben Ford	716e51a3f3	res_stir_shaken: Initial commit and reading private key. This commit sets up some of the initial framework for the module and adds a way to read the private key from the specified file, which will then be appended to the certificate object. This works fine for now, but eventually some other structure will likely need to be used to store all this information. Similarly, the caller_id_number is specified on the certificate config object, but in the end we will want that information to be tied to the certificate itself and read it from there. A method has been added that will retrieve the private key associated with the caller_id_number passed in. Tab completion for certificates and stores has also been added. Change-Id: Ic4bc1416fab5d6afe15a8e2d32f7ddd4e023295f	5 years ago
Sean Bright	7a64868118	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
Sean Bright	9ac933fbba	app_voicemail.c: Document VMSayName interruption behavior ASTERISK-26424 #close Change-Id: I797ad0ed302d0b3d2c90543eff5b7207ed08ecf0	5 years ago
George Joseph	3b0a53f257	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
lvl	374d18cb97	res_musiconhold: Load all realtime entries, not just the first ASTERISK-29099 Change-Id: I45636679c0fb5a5f59114c8741626631a604e8a6	5 years ago

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