3645 Commits (0674f319b9393a9265185a9023c3cdec539247ba)

Author	SHA1	Message	Date
George Joseph	0ab443007b	build_tools: Add ability to download variants to download_externals ... Some external packages have multiple variants that apply to different builds of asterisk. The DPMA for instance has a "bundled" variant that needs to be downloaded if asterisk was configured with --with-pjproject-bundled. There are 2 ways to specify variants: If you need the user to make the decision about which variant to download, simply create multiple menuselect "member" entries like so... <member name="res_digium_phone" displayname="..snipped.."> <support_level>external</support_level> <depend>xmlstarlet</depend> <depend>bash</depend> <defaultenabled>no</defaultenabled> </member> <member name="res_digium_phone-bundled" displayname="..snipped.."> <support_level>external</support_level> <depend>xmlstarlet</depend> <depend>bash</depend> <defaultenabled>no</defaultenabled> </member> Note that the second entry has "-<variant>" appended to the name. You can then use the existing menuselect facilities to restrict which members to enable or disable. Youy probably don't want the user to enable multiple at the same time. If you want to hide the details of the variants, the better way to do it is to create 1 member with "variant" elements. <member name="res_digium_phone" displayname="..snipped.."> <support_level>external</support_level> <depend>xmlstarlet</depend> <depend>bash</depend> <defaultenabled>no</defaultenabled> <member_data> <downloader> <variants> <variant tag="bundled" condition='[[ "$PJPROJECT_BUNDLED" = "yes" ]]'/> </variants> </downloader> </member_data> </member> The condition must be a bash expression suitable for use with an "if" statement. Any environment variable can be used plus those available in makeopts. In this case, if asterisk was configured with --with-pjproject-bundled the bundled variant will be automatically downloaded. Otherwise the normal version will be downloaded. Change-Id: I4de23e06d4492b0a65e105c8369966547d0faa3e	9 years ago
Joshua Colp	10c180760c	res_odbc: Make pooling option deprecation notice more useful. ... This changes the notice for the deprecation of the old pooling options to point to the new option for doing pooling. This gives a clearer direction as to what to look into. ASTERISK-26389 #close Change-Id: I2ca9cdfdcd75aec170a7db9d5ff69a4cd25b7c10	9 years ago
zuul	177557bc00	Merge "res_pjsip_multihomed: Change Contact port to listening port." into 13	9 years ago
Richard Mudgett	cdbad152c7	res_config_odbc.c: Fix buffer size limitation creating invalid SQL. ... Creating ODBC SQL queries resulted in queries too large to fit into the supplied buffer. The resulting truncated buffer contained an invalid SQL query. * Made SQL query generation code use a thread storage buffer that can increase in size as needed. * Fixed bad multi-line warning messages. ASTERISK-26263 #close Reported by: Jeppe Ryskov Larsen Change-Id: I23f3cdd43c2dac80bed3ded4dd77d18cb17f21ae	9 years ago
Joshua Colp	449719be00	res_pjsip_multihomed: Change Contact port to listening port. ... The res_pjsip_multihomed module determines what interface and transport a request is going out on and updates the SIP message accordingly with the address information. This currently incorrectly updates the Contact header for connectionful protocols to the ephemeral connection port, instead of the bound address for the listening socket which can actually accept the connection back. If the remote side attempts to connect back on the epehemeral port it will fail. This change makes it so the port is updated to the bound port on connectionful protocols and is maintained on UDP (as there can be multiple of those). ASTERISK-26374 #close Change-Id: I50f8dab65b9f75117d73ba5f6bbcf6c9871854ab	9 years ago
zuul	11d05fc369	Merge "res_pjsip_transport_management: Convert time in log message to seconds." into 13	9 years ago
Joshua Colp	1cac856e17	rtp: Preserve timestamps on video frames. ... Currently when receiving video over RTP we store only a calculated samples on the frame. When starting the video it can take some time for this calculation to actually yield a value as it requires constant changing timestamps. As well if a video frame passes over multiple RTP packets this calculation will fail as the timestamp is the same as the previous RTP packet and the number of samples calculated will be 0. This change preserves the timestamp on the frame and allows it to pass through the core. When sending the video this timestamp is used instead of a new one being calculated. ASTERISK-26367 #close Change-Id: Iba8179fb5c14c9443aee4baf670d2185da3ecfbd	9 years ago
zuul	a5f7b07579	Merge "res_pjsip: Add ignore_uri_user_options option." into 13	9 years ago
Joshua Colp	9df4056d70	res_pjsip_transport_management: Convert time in log message to seconds. ... ASTERISK-26375 #close Change-Id: I46496af5cae41413e76d44d2068a7431279f09dc	9 years ago
zuul	3d7cbaa675	Merge "res_pjsip_messaging.c: Misc cleanups and fixes." into 13	9 years ago
Richard Mudgett	30af92e78d	res_pjsip: Add ignore_uri_user_options option. ... This implements the chan_sip legacy_useroption_parsing option but with a better name. * Made the caller-id number and redirecting number strings obtained from incoming SIP URI user fields always truncated at the first semicolon. People don't care about anything after the semicolon showing up on their displays even though the RFC allows the semicolon. ASTERISK-26316 #close Reported by: Kevin Harwell Change-Id: Ib42b0e940dd34d84c7b14bc2e90d1ba392624f62	9 years ago
Joshua Colp	7580a736bb	res_pjsip: Only invoke unidentified endpoint logic when unidentified. ... The code was incorrectly invoking the unidentified logic when an endpoint had actually been identified, causing log messages to be output. ASTERISK-26349 #close Change-Id: Id8104fc9e3d138d5e8b6f6977ecc08765fd17d4f	9 years ago
Mark Michelson	f1ffc22933	res_pjsip: Do not crash on ACKs from unknown endpoints. ... The endpoint identification PJSIP module is intended to identify which endpoint an incoming request is from. If an endpoint is not identified, then an artificial endpoint is used in its place when proceeding. The problem is that the ACK request type is an exception to the rule. The artificial endpoint is not used when processing an ACK. This results in the possibility of having a NULL endpoint being used further on. The reason ACK is an exception is an attempt not to spam security logs with unidentified requests. Presumably, you've already logged the unidentified request on the preceeding INVITE. Up until Asterisk 13.10, retrieving a NULL endpoint in this fashion didn't cause an issue. A new change in 13.10 added endpoint ACL checking shortly after endpoint identification. Because we are accessing a NULL endpoint, this ACL check resulted in a crash. The fix here is to be sure to retrieve the artificial endpoint for all request types. ACKs still do not generate unidentified request security events. ASTERISK-26264 #close Reported by nappsoft AST-2016-006 Change-Id: Ie0c795ae2d72273decb972dd74b6a1489fb6b703	9 years ago
Richard Mudgett	23d6ec7417	res_pjsip_messaging.c: Misc cleanups and fixes. ... * Eliminated RAII_VAR in get_outbound_endpoint(). * Simplify update_to() coding. However, this function can only be a NoOp because the To string can only be a URI and not a name-address formatted string. * Simplify update_from() coding. Also fixed a code path modifying the from string when the caller could still want to use the original string. * Fixed msg_data_create() incompletely removing the "pjsip:" to then add back the "sip:" string if needed. The code didn't handle the "pjsip:sip:" case because it left the colon after pjsip in the string. Change-Id: I68a09a665f6d4daa9eaa59069045ab69122e28db	9 years ago
Joshua Colp	5f19657710	res_pjsip: Allow global headers to be overridden. ... Currently when you add global headers from the dialplan both the header in the dialplan and the globally configured header are added to the resulting SIP INVITE. This change makes it so the headers in the dialplan take precedence and are the only ones added. Change-Id: I36f864298f38db3632ad503edc11267cb8ffb3ad	9 years ago
zuul	8925367291	Merge "res_pjsip_session: segfault on already disconnected session" into 13	9 years ago
Joshua Colp	89f7cd8182	Merge "build: Add download capability for external packages" into 13	9 years ago
zuul	9e874d2cc8	Merge "res_pjsip_registrar.c: Reduce stack usage in find_aor_name()." into 13	9 years ago
zuul	6de392eb17	Merge "pjsip_configuration.c: Ignore repeated identify by methods." into 13	9 years ago
George Joseph	117a7741c8	build: Add download capability for external packages ... The DPMA and g729a, silk, siren7 and siren14 codecs hosted at http://downloads.digium.com/pub/telephony/ are now listed in the "External" sections of the "Resource Modules" and "Codec Translators" pages in menuselect. Any that are selected will automatically be downloaded and installed when "make install" is run. Their LICENSE and README (if avaialble) files will be installed to ASTVARLIBDIR/documentation/thirdparty/<product_name>. Example use with codecs: The codecs/codecs.xml file is a menuselect style xml file that lists the codecs to be included. Their support levels are 'external', which triggers the download and install, and defaultenabled is no. Also because codec_g729a is actually in a directory named codec_g729 on the download server, the newly added 'member_data' element is used to override the default of the directory name being the package name. You can use the 'directory_name' attribute to keep default base URL (http://downloads.digium.com/pub/telephony/) but use the new directory, or you use the 'remote_url' attribute to specify a full URL to the download directory. In this case, you must still follow the same subdirectory naming conventions as that used for the packages located at 'http://downloads.digium.com/pub/telephony'. A new configure option '--with-externals-cache' was added and like '--with-sounds-cache' it allows the installer to cache tarballs so they're not downloaded every time. To assist with the download and install process, each external package now has a manifest.xml file that, among other things, contains a package version and checksums for each file in the tarball. The manifest is saved to both the cache directory and ASTMODDIR and together with the manifest.xml on the downloads site, tells the install scripts whether a download and/or update is needed. bash and xmlstarlet are required for downloader operation. If they're not installed, the external items in menuselect will be unavailable. Change-Id: Id3dcf1289ffd3cb0bbd7dfab3cafbb87be60323a	9 years ago
Richard Mudgett	03fc438f6e	res_pjsip_registrar.c: Reduce stack usage in find_aor_name(). ... Change-Id: I8aebad1fdcf303bd115b59a4b57fbbd5b2267f09	9 years ago
Richard Mudgett	b5e753227d	pjsip_configuration.c: Ignore repeated identify by methods. ... Change-Id: Ied0c06043d1dfef8fdc9c9a808cf89b118119838	9 years ago
Richard Mudgett	9b7501b6ad	config_global.c: Comments and a default expression adjustment. ... Change-Id: Ia6a58f8c73a30da6874b3f94364dce162d6f1ad3	9 years ago
Alexei Gradinari	9bca895469	res_pjsip_session: segfault on already disconnected session ... On heavy loaded system the TCP/TLS incoming calls could be disconnected by pjproject while these calls are being processed by asterisk which could use the session's memory pools. If the session in the disconnected state then the session memory pools were already freed, so we get segfault. This patch adds a lifetime control on an INVITE session to pjproject. The lifetime of the session is manipulated by calling pjsip_inv_add_ref/pjsip_inv_dec_ref. This patch uses these functions to inform pjproject that the session is in use. This patch adds check if the session state is not disconnected and also checks if the memory pool is not NULL. This patch also places tasks 'session_end' and 'session_end_completion' into session's serializer to avoid race condition. ASTERISK-26291 #close Change-Id: I4d28b1fb3b91f0492a911d110049d670fdc3c8d7	9 years ago
zuul	1bd571ef75	Merge "res_pjsip: qualify/unqualify added/deleted realtime endpoints" into 13	9 years ago
Alexei Gradinari	308a65fe6c	res_pjsip: qualify/unqualify added/deleted realtime endpoints ... If the PJSIP endpoint's AOR with the permanent contact was deleted from the realtime storage the res_pjsip module continues trying to qualify this contact. The error 'Unable to find an endpoint to qualify contact' appeares every 'qualify_frequency' seconds. This patch deletes this contact in this case. The PJSIP endpoint's AOR with the permanent contact is never qualified if it is added to realtime storage after asterisk started. This patch adds qualifying for the AOR's permanent contacts on the first handling of this AOR. ASTERISK-26319 #close Change-Id: Ib93dded9121edb113076903d1aa95402f799f8fe	9 years ago
zuul	27989f22f3	Merge "res_pjsip: Default endpoints to the "offline" status." into 13	9 years ago
Joshua Colp	1b91adf7a1	Merge "res_pjsip: Cache global config options." into 13	9 years ago
Richard Mudgett	5cd583d7a2	res_pjsip: Cache global config options. ... We may check a global config option hundreds of times a second or more. Asking sorcery for the global configuration from the config files backend involves several allocations and container traversals. Using realtime without a memory cache is a lot worse because you have to lookup in the realtime database each time to reconstitute the sorcery object. With a memory cache for realtime, there is about the same amount of overhead as for config files. Either way, it is still fairly expensive to access the sorcery object that much. * Cache the global config options so we can access them faster. You must now always perform a res_pjsip reload to change the global options. Change-Id: Ice16c7a4cbca4614da344aaea21a072b86263ef7	9 years ago
Richard Mudgett	8b4b2500ee	res_fax: Fix deadlock in ast_channel_get_t38_state(). ... ast_channel_get_t38_state() calls ast_channel_queryoption() with AST_OPTION_T38_STATE. If the passed in channel is a local channel then a deadlock can happen if a channel lock is held when called. * Made ast_channel_get_t38_state() callers not hold a channel lock before calling. * Update ast_channel_get_t38_state() doxygen to note that no channel locks can be held when calling the function. ASTERISK-26203 #close Reported by: Etienne Lessard ASTERISK-24822 #close Reported by: David Brillert ASTERISK-22732 #close Reported by: Richard Mudgett Change-Id: I49fd76fa9af628b4198009b5c0b82c8b03681214	9 years ago
Richard Mudgett	e8d4f40022	res_fax: Fix deadlock setting FAXMODE channel variable. ... ASTERISK-25980 added the FAXMODE channel variable to res_fax.c. Unfortunately, it also introduced a deadlock potential because set_channel_variables() which sets FAXMODE can be called during a masquerade. The ast_channel_get_t38_state() which gets the value used to set FAXMODE cannot be called with the channel locked. As a result, local channels can deadlock because of how they must acquire the locks necessary to operate. The intent of FAXMODE is for dialplan to know how a fax was transferred after the fax completes. However, the previous patch sets FAXMODE to the channel's current T.38 state AFTER the fax has completed and where T.38 may have already disconnected. * Set FAXMODE based upon T.38 negotiations exchanged either with the fax applications or the fax framehooks. ASTERISK-26203 Reported by: Etienne Lessard ASTERISK-24822 Reported by: David Brillert ASTERISK-22732 Reported by: Richard Mudgett Change-Id: Id525747254b64c1efe8b1b5973d52ff9719c2ae1	9 years ago
Richard Mudgett	35cf6c7702	res_fax.c: Fix deadlock in fax_gateway_indicate_t38(). ... fax_gateway_indicate_t38() calls ast_indicate_data() which cannot be called with any channel locks already held. A deadlock can happen if the function is operating on a local channel. * Made fax_gateway_indicate_t38() unlock the channel before calling ast_indicate_data() since fax_gateway_indicate_t38() is always called with the channel locked. * Made fax_gateway_indicate_t38() return void since nothing cared about its return value. ASTERISK-26203 Reported by: Etienne Lessard ASTERISK-24822 Reported by: David Brillert ASTERISK-22732 Reported by: Richard Mudgett Change-Id: I701ff2d26c5fc23e0d5a48a3fd98759a9fd09407	9 years ago
Richard Mudgett	50b2aa506f	res_fax.c: Add chan locked precondition comments. ... Change-Id: Ic10ae434536bbf7fb7055d6ab36cc50b8748a4e7	9 years ago
Richard Mudgett	038cbc0215	ast_framehook_detach() must be called with the channel locked. ... The framehook container could become corrupted if the channel lock is not held before calling. Change-Id: If0a1c7ba0484ed3a191106a7516526b905952584	9 years ago
Richard Mudgett	88e9d05ef7	ast_framehook_attach() must be called with the channel locked. ... The framehook container could become corrupted if the channel lock is not held before calling. Change-Id: I1a6b957a1f7b899eb29a186915f8cccab886a438	9 years ago
George Joseph	c9e83f6d0b	res_rtp_multicast: Fix SEGV in ast_multicast_rtp_create_options ... ast_multicast_rtp_create_options now checks for NULL or empty options Change-Id: Ib845eae46a67a9787e89a87ebd1027344e5e0362	9 years ago
Mark Michelson	c16ef02318	res_pjsip: Default endpoints to the "offline" status. ... A recent change attempted to optimize startup by not updating contact status. Instead, code responsible for qualifying contacts updates the status as it becomes known. The code even accounts for contacts/AORs that are not set to be qualified. The problem, though, is when there are no contacts associated with an endpoint. A common case is when an endpoint is set to register its contacts but has not done so yet. In this case, prior to registration, the endpoint's device state will appear to be "not in use" and hints associated with that device will appear to be "idle". In actuality, the device state and hint should both appear as "unavailable". The reason for the failure is that the optimization change made all persistent endpoint states set to "unknown". The fix here is to change the hard-coded "unknown" to be "offline" instead. The default state will be offline until the qualifying code determines that the contact is actually online. This way, if there are no contacts at all, then the state stays as offline, and device state and hints appear correctly. ASTERISK-26269 #close Reported by nappsoft Change-Id: Ie99b84169393983453076f5e9c0d35ff313a456a	9 years ago
zuul	e0c807c33d	Merge "compilation failed with -Werror=maybe-uninitialized" into 13	9 years ago
David M. Lee	e54dcf4fd5	res_odbc_transaction: add dep on generic_odbc ... When res_odbc_transaction depended on res_odbc, it got the generic_odbc headers and libs implicitly. Now that it no longer depends on res_odbc, its dependency on generic_odbc must be explicit. Change-Id: I9db88f7af7388437f49903d3008ba8d4890d5911	9 years ago
zuul	fcc49d760f	Merge "res_ari: Add http prefix to generated docs" into 13	9 years ago
zuul	6b3e96e479	Merge "res_odbc: Correct the dependency relationship with res_odbc_transaction" into 13	9 years ago
zuul	16cb686fac	Merge "rest-api: Swagger scripts were not replacing format variable in file brief" into 13	9 years ago
zuul	d49043c092	Merge "res_format_attr_g729: Add annexb=no format parameter to SDPs" into 13	9 years ago
Alexei Gradinari	b494b9f88c	compilation failed with -Werror=maybe-uninitialized ... The compilation failed for devmode --enable DONT_OPTIMIZE --enable BETTER_BACKTRACES --enable DO_CRASH --enable TEST_FRAMEWORK res_pjsip/pjsip_configuration.c: In function dtls_handler: res_pjsip/pjsip_configuration.c:974:20: error: back may be used uninitialized in this function [-Werror=maybe-uninitialized] int size = strlen(front); ^ cc1: all warnings being treated as errors Change-Id: I7f082ead0312792a577ec7c73015ba64dabca580	9 years ago
zuul	504da21d47	Merge "res_pjsip: Add contact_user to endpoint" into 13	9 years ago
Kevin Harwell	ff2378c735	rest-api: Swagger scripts were not replacing format variable in file brief ... Given resource paths did not have 'json' substituted in for the '{format}'. For some auto generated documentation/comment strings it resulted in something like the following: "... REST handler for /api-docs/sounds.{format}" This patch makes sure the resource api's path is properly substituted. ASTERISK-25472 #close Change-Id: Ie3e950a35db4043e284019d6c9061f3b03922e23	9 years ago
Kevin Harwell	43f400ef95	res_format_attr_g729: Add annexb=no format parameter to SDPs ... Historically, Asterisk has always specified annexb=no for the g729 format. However, when using res_pjsip no format attribute was specified. This patch makes it so the SDP now contains a format attribute line with annexb=no. Note, that this means only g729a is negotiated. Even for pass through support. According to rfc7261 the type of annex used (a or b) is dependent upon the answerer. However, Asterisk being a back to back user agent makes this tricky to support at this time, thus we only allow annex 'a' for now. ASTERISK-26228 #close patches: res_format_attr_g729.c submitted by Jason Parker (license 4993) Change-Id: I76bc20cc0a01af01536e9915afef319c269c22d0	9 years ago
George Joseph	4c1ae07d51	res_odbc: Correct the dependency relationship with res_odbc_transaction ... The MODULEINFO dependencies between these 2 modules was reversed. res_odbc should depend on res_odbc_transaction, not the other way around. ASTERISK-25984 #close Change-Id: Ifcfbb49c0b51cf6640a5446d47cd6c48caf1331f	9 years ago
Richard Mudgett	1cd12d73a6	res_pjsip_session.c: Fix unbound srv failover tests. ... Commit 1b666549f3 broke the srv failover functionality if a TCP connection gets disconnected. Under these conditions, session_inv_on_state_changed() gets a PJSIP_EVENT_TRANSPORT_ERROR and restarts the INVITE transaction on a new transport. Unfortunately, session_inv_on_tsx_state_changed() also gets the same PJSIP_EVENT_TRANSPORT_ERROR event and unconditionally terminates the session. * Made session_inv_on_tsx_state_changed() complete terminating the session on PJSIP_EVENT_TRANSPORT_ERROR only if the session state is still PJSIP_INV_STATE_DISCONNECTED. ASTERISK-26305 #close Reported by: Richard Mudgett Change-Id: If736e766b5c55b970fa38ca6c8a885caf27b897d	9 years ago
George Joseph	329507fe20	res_pjsip: Add contact_user to endpoint ... contact_user, when specified on an endpoint, will override the user portion of the Contact header on outgoing requests. Change-Id: Icd4ebfda2f2e44d3ac749d0b4066630e988407d4	9 years ago