asterisk

Commit Graph

Author	SHA1	Message	Date
Jenkins2	2d69137bf4	Merge "app_record: Resolve some absolute vs. relative filename bugs" into 13	8 years ago
Jenkins2	378e0f7764	Merge "voicemail: Fix various abuses of mkstemp" into 13	8 years ago
Jenkins2	71cc740828	Merge "main/app: Only look to end of file if ':end' is specified, and not just ':'" into 13	8 years ago
Jenkins2	06e7d6e750	Merge "res/res_pjsip_session: allow SDP answer to be regenerated" into 13	8 years ago
Jenkins2	a3b36b5043	Merge "alembic: Add dtls_fingerprint column in ps_endpoints table" into 13	8 years ago
Sean Bright	9e6efcace5	voicemail: Fix various abuses of mkstemp mkstemp() returns a unique filename, but appending an extension to that filename does not guarantee uniqueness. Instead, use mkdtemp() and we can put whatever extension we want on the files that we create inside the directory. In the case of app_minivm, we also now properly clean up any temporary files that we create. ASTERISK-20858 #close Reported by: Walter Doekes Change-Id: I30ad04f0e115f0b11693ff678ba5184d8b938e43	8 years ago
Sean Bright	01b5913ce0	app_record: Resolve some absolute vs. relative filename bugs If the Record() application is called with a relative filename that includes directories, we were not properly creating the intermediate directories and Record() would fail. Secondarily, updated the documentation for RECORDED_FILE to mention that it does not include a filename extension. Finally, rewrote the '%d' functionality to be a bit more straight forward and less noisy. ASTERISK-16777 #close Reported by: klaus3000 Change-Id: Ibc2640cba3a8c7f17d97b02f76b7608b1e7ffde2	8 years ago
Jenkins2	28857047da	Merge "app_queue: Evaluate realtime queues when running dialplan functions" into 13	8 years ago
Jenkins2	c0251ae004	Merge "app_voicemail: Honor escape digits in "greeting only" mode" into 13	8 years ago
Florian Floimair	bf178a0f4f	alembic: Add dtls_fingerprint column in ps_endpoints table The ps_endpoints table was missing the dtls_fingerprint column introduced with commit `adba2a8d7f`. ASTERISK-27168 #close Change-Id: I9cb5006f7f50718b5239919562773adabb334cfd	8 years ago
Matt Jordan	fff2f68616	main/app: Only look to end of file if ':end' is specified, and not just ':' There is a little known feature in app_controlplayback that will cause the specified offset to be used relative to the end of a file if a ':end' is detected within the filename. This feature is pretty bad, but okay. However, a bug exists in this code where a ':' detected in the filename will cause the end pointer to be non-NULL, even if the full ':end' isn't specified. This causes us to treat an unspecified offset (0) as being "start playing from the end of the file", resulting in no file playback occurring. This patch fixes this bug by resetting the end pointer if ':end' is not found in the filename. ASTERISK-23608 #close Reported by: Jonathan White Change-Id: Ib4c7b1b45283e4effd622a970055c51146892f35 (cherry picked from commit `13efea24f7`)	8 years ago
Sean Bright	579d4593ac	app_queue: Evaluate realtime queues when running dialplan functions ASTERISK-19103 #close Reported by: Jim Van Meggelen Change-Id: I4bd32a9d1fcebb8ac56bff0e084d4f53e31b692b	8 years ago
Sean Bright	0af145de2d	app_voicemail: Honor escape digits in "greeting only" mode ASTERISK-21241 #close Reported by: Eelco Brolman Patches: Patch uploaded by Eelco Brolman (License 6442) Change-Id: Icbe39b5c82a49b46cf1d168dc17766f3d84f54fe	8 years ago
Sean Bright	d251a961ac	res_smdi: Clean up memory leak Change-Id: I1e33290929e1aa7c5b9cb513f8254f2884974de8	8 years ago
Jenkins2	e8cfd5a80f	Merge "bridge_softmix.c: Remove always true test." into 13	8 years ago
Richard Mudgett	3f22b53349	bridge_softmix.c: Remove always true test. Change-Id: I26238df2ff0d0f6dfe95c3aa35da588f1ee71727	8 years ago
Sungtae Kim	b88c3a4209	app_queue: Fix initial hold time queue statistic Fixed to use correct initial value and fixed to use the correct queue info to check the first value. ASTERISK-27204 Change-Id: Ia9e36c828e566e1cc25c66f73307566e4acb8e73	8 years ago
Jenkins2	71753d67f4	Merge "res_xmpp: fix inverted return code check in OAuth" into 13	8 years ago
Torrey Searle	8e99969000	res/res_pjsip_session: allow SDP answer to be regenerated If an SDP answer hasn't been sent yet, it's legal to change it. This is required for PJSIP_DTMF_MODE to work correctly, and can also have use in the future for updating codecs too. ASTERISK-27209 #close Change-Id: Idbbfb7cb3f72fbd96c94d10d93540f69bd51e7a1	8 years ago
Jenkins2	3fffa7ae86	Merge "res_calendar_icalendar: Properly handle recurring events" into 13	8 years ago
Michael Kuron	4faf77feec	res_xmpp: fix inverted return code check in OAuth fetch_access_token calls func_curl via ast_func_read. The latter returns 0 upon success and -1 if the function is not available. This commit inverts the return code check so that an error is printed if the module is not loaded and not if it is loaded. ASTERISK-27207 #close Change-Id: I9ef903f80702d1218e8701f65a4e5e918e6548fb	8 years ago
Sean Bright	a6251ec373	res_calendar_icalendar: Properly handle recurring events When looking for recurring events, use the correct end time based on the configured 'timeframe.' ASTERISK-27174 #close Reported by: Mark Thompson Change-Id: Id90c3cfc79d561a5521d79be176683e225f2edef	8 years ago
George Joseph	572b5307e0	Fix downloader not working with curl The codec/dpma downloader wasn't handling curl correctly. The logic that transforms makeopts into a bash-sourceable file wasn't handling the make 'or' command in DOWNLOAD_TIMEOUT so bash was looking for an 'or' command. That logic has been eliminated. Instead of trying to transform and source makeopts, the downloader now calls a make scriptlet to print the value of a specific variable. This way, make handles the ors (or any other make construct that happens to creep into that file). ASTERISK-27202 Reported by: Sean McCord Change-Id: Iadfb6693528e4d4da7b8bb201fa66da2c71c7f99	8 years ago
Jenkins2	b461fe8bcd	Merge "configure: Check cache for valid pjproject tarball before downloading." into 13	8 years ago
Richard Mudgett	8594f73a81	configure: Check cache for valid pjproject tarball before downloading. On a fresh Asterisk source directory, the bundled pjproject tarball is unconditionally downloaded even if the tarball is already in a specified cache directory. * Made check if the pjproject tarball is valid in the cache directory before downloading the tarball on a fresh source directory. Change-Id: Ic7ec842d3c97ecd8dafbad6f056b7fdbce41cae5	8 years ago
Richard Mudgett	d08342b0cb	res_pjsip: Fix prune_on_boot to remove only contacts for the host. * Check that the contact's reg_server matches the host's name before deleting any prune_on_boot contacts. We don't want to delete reliable transport contacts made with other servers if the ps_contacts database table is shared with other servers. Thanks to Ross Beer for pointing out that the original prune logic would delete reliable transport contacts from other servers. ASTERISK-27147 Change-Id: I8e439d0d1c266ffdfd7b73d1e5e466180a689bd0	8 years ago
Jenkins2	fa50a3def9	Merge "res_xmpp: Google OAuth 2.0 protocol support for XMPP / Motif" into 13	8 years ago
Andrey Egorov	54e3ac402f	res_xmpp: Google OAuth 2.0 protocol support for XMPP / Motif Add ability to use tokens instead of passwords according to Google OAuth 2.0 protocol. ASTERISK-27169 Reported by: Andrey Egorov Tested by: Andrey Egorov Change-Id: I07f7052a502457ab55010a4d3686653b60f4c8db	8 years ago
George Joseph	b81de2c896	Merge "res_pjsip_outbound_registration.c: Re-REGISTER on transport shutdown." into 13	8 years ago
Jenkins2	1bfb4047ae	Merge "res_pjsip: Remove ephemeral registered contacts on transport shutdown." into 13	8 years ago
Jenkins2	cc4735742c	Merge "res_pjsip: PJSIP Transport state monitor refactor." into 13	8 years ago
Jenkins2	1ff6082da9	Merge "res_pjsip_transport_management.c: Rename some variables." into 13	8 years ago
Richard Mudgett	bac3e8c08b	STUN/netsock2: Fix some valgrind uninitialized memory findings. * netsock2.c: Test the addr->len member first as it may be the only member initialized in the struct. * stun.c:ast_stun_handle_packet(): The combinded[] local array could get used uninitialized by ast_stun_request(). The uninitialized string gets copied to another location and could overflow the destination memory buffer. These valgrind findings were found for ASTERISK_27150 but are not necessarily a fix for the issue. Change-Id: I55f8687ba4ffc0f69578fd850af006a56cbc9a57	8 years ago
Richard Mudgett	1cf2c79f37	res_pjsip_outbound_registration.c: Re-REGISTER on transport shutdown. The fix for the issue is broken up into three parts. This is part three which handles the client side of REGISTER requests. The registered contact may no longer be valid on the server when the transport used is reliable and the connection is broken. * Re-REGISTER our contact if the reliable transport is broken after registration completes. We attempt to re-REGISTER immediately to minimize the time we are unreachable. Time may have already passed between the connection being broken and the loss being detected. * Reorder sip_outbound_registration_state_alloc() so the STATSD_GUAGE's are still correct if an allocation failure happens. ASTERISK-27147 Change-Id: I3668405b1ee75dfefb07c0d637826176f741ce83	8 years ago
Richard Mudgett	07d026b4cd	res_pjsip: Remove ephemeral registered contacts on transport shutdown. The fix for the issue is broken up into three parts. This is part two which handles the server side of REGISTER requests when rewrite_contact is enabled. Any registered reliable transport contact becomes invalid when the transport connection becomes disconnected. * Monitor the rewrite_contact's reliable transport REGISTER contact for shutdown. If it is shutdown then the contact must be removed because it is no longer valid. Otherwise, when the client attempts to re-REGISTER it may be blocked because the invalid contact is there. Also if we try to send a call to the endpoint using the invalid contact then the endpoint is not likely to see the request. The endpoint either won't be listening on that port for new connections or a NAT/firewall will block it. * Prune any rewrite_contact's registered reliable transport contacts on boot. The reliable transport no longer exists so the contact is invalid. * Websockets always rewrite the REGISTER contact address and the transport needs to be monitored for shutdown. * Made the websocket transport set a unique name since that is what we use as the ao2 container key. Otherwise, we would not know which transport we find when one of them shuts down. The names are also used for PJPROJECT debug logging. * Made the websocket transport post the PJSIP_TP_STATE_CONNECTED state event. Now the global keep_alive_interval option, initially idle shutdown timer, and the server REGISTER contact monitor can work on wetsocket transports. * Made the websocket transport set the PJSIP_TP_DIR_INCOMING direction. Now initially idle websockets will automatically shutdown. ASTERISK-27147 Change-Id: I397a5e7d18476830f7ffe1726adf9ee6c15964f4	8 years ago
Richard Mudgett	ca261d4b70	res_pjsip: PJSIP Transport state monitor refactor. The fix for the issue is broken up into three parts. This is part one which refactors the transport state monitor code to allow more modules to be able to monitor transports. * Pull the management of PJPROJECT's transport state callback code from res_pjsip_transport_management.c into res_pjsip. Now other modules can dynamically add and remove themselves from transport monitoring without worrying about breaking PJPROJECT's callback chain. * Add the ability for other modules to get a callback whenever a specific transport is shutdown. ASTERISK-27147 Change-Id: I7d9a31371eb1487c9b7050cf82a9af5180a57912	8 years ago
Richard Mudgett	162f6ab845	res_pjsip_transport_management.c: Rename some variables. * Use monitored instead of the misleading keepalive name. Change-Id: I9e5bcbb4ab2b82d49bcd0f06dfe85d15e0b552b6	8 years ago
Jenkins2	7ad750477d	Merge "res_pjsip_messaging: IPv6 receive address needs brackets" into 13	8 years ago
Scott Griepentrog	22575b6342	res_pjsip_messaging: IPv6 receive address needs brackets When handling an incoming SIP MESSAGE, PJSIP attaches the IP address that the message was received from to the message in the variable PJSIP_RECVADDR. When the IP address is IPv6 the :PORT appended results in an unparseable mess. By using an additional bit flag on the pj_sockaddr_print call, the conventional use of brackets around the address is achieved. ASTERISK-27193 #close Change-Id: I12342521f2ce87a5b6e4883d480a3fd957aa9fd9	8 years ago
Joshua Colp	0cf2e60161	Merge "configure: Add --with-download-cache option" into 13	8 years ago
Jenkins2	9bc114ffba	Merge "res_rtp_asterisk: enable rtcp & QOS stats on native bridge" into 13	8 years ago
Joshua Colp	b5eca43b29	Merge "res_rtp_asterisk: Make P2P bridge Asymmetric codec aware" into 13	8 years ago
George Joseph	363d61ef58	configure: Add --with-download-cache option To make building without an internet connection easier, a new ./configure option '--with-download-cache' was added that sets the cache for externals (like pjproject, the codecs and the DPMA), AND the sounds files. It can also be specified as an environment variable named "AST_DOWNLOAD_CACHE". The existing '--with-sounds-cache' option / SOUNDS_CACHE_DIR env variable and '--with-externals-cache' option / EXTERNALS_CACHE_DIR env variable remain and if specified, will override '--with-downloads-cache'. Change-Id: I5c3cf15ee61e8fe191b52732303e969854f8d861	8 years ago
Jenkins2	0493f08468	Merge "res_pjsip_session: Release media resources on session end quicker." into 13	8 years ago
Jenkins2	f75f6c71f8	Merge "res_pjsip_nat.c: Remove unnecessary CMP_STOP." into 13	8 years ago
Jenkins2	3cf30b4544	Merge "Support GMIME 3.0" into 13	8 years ago
Torrey Searle	3608f96ea3	res_rtp_asterisk: enable rtcp & QOS stats on native bridge Asterisk wasn't generating or forwarding RTCP packets when native bridge was activated. Also the stats weren't available via CHANNEL(qos). Now the RTCP stats are always calculated. ASTERISK-27158 #close Change-Id: I46fb8f61c95e836b9d2dda6054b0cf205c16037b	8 years ago
Joshua Colp	0de7312fac	res_pjsip_session: Release media resources on session end quicker. A change was made long ago where the session was kept around until the underlying INVITE session had been destroyed. This had the side effect of also keeping the underlying media resources around for this time as well. This change ensures that when we know the session is ending we release the media resources immediately. ASTERISK-27110 Change-Id: I643e431d5c3bf05cda220c1d39e824a505a29b82	8 years ago
Corey Farrell	905c4ca3dc	app_privacy: remove unused header asterisk/image.h Change-Id: I56ed530633a642633b18383821069e806c92ae82	8 years ago
Jenkins2	6dd12aaacd	Merge "Fix compile error for old versions of GCC." into 13	8 years ago

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29115 Commits (2d69137bf446aba4b690e6e07ff3ba1108250f34) All Branches Search

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