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Author	SHA1	Message	Date
Alexander Traud	4f7f8a7e95	chan_sip: Support nat=auto_comedia or nat=force_rport,auto_comedia. ... In the SIP channel driver chan_sip, auto_comedia was expected to be used in tandem with auto_force_rport. Or stated differently: Only when auto_force_rport was chosen (the default), auto_comedia worked. This change allows auto_comedia to be set independently of the state of (auto_)force_rport. For example, nat=force_rport,auto_comedia is useful for IPv4/IPv6 Dual Stack deployments when IPv6 clients are behind a Firewall. ASTERISK-26457 #close Change-Id: Ib29d66c6dbb61648e371e01fc36c6978ddae5bc2	9 years ago
Badalyan Vyacheslav	17031f12fe	vector: After remove element recheck index ... Small fix. It is necessary to double-check the index that we just removed because there is a new element. ASTERISK-26453 #close Change-Id: Ib947fa94dc91dcd9341f357f1084782c64434eb7	9 years ago
Joshua Colp	31a3e75025	Merge "cel_odbc: Fix memory leak on module unload"	9 years ago
Joshua Colp	28c0e7ab18	Merge "pjproject_bundled: Add MALLOC_DEBUG capability"	9 years ago
Torrey Searle	cc269766b8	res_rtp_asterisk: Fix infinite DTMF issue when switching to P2P bridge ... If a bridge switched to P2P when a DTMF was in progress it was possible for the DTMF to continue being sent indefinitely. Change-Id: I7e2a3efe0d59d4b214ed50cd0b5d0317e2d92e29	9 years ago
Corey Farrell	fafdde322c	logger: Prevent output of verbose messages initiated from rasterisk. ... Remote asterisk consoles should only display verbose log messages created by the daemon. The first patch for ASTERISK-26410 caused a couple verbose messages to be printed when the rasterisk process ended. ASTERISK-26410 Change-Id: Ie2a1bb3753ad2724c0349ec1a336f52f7117b52a	9 years ago
Michael Walton	7af7490e42	audiohooks: Remove redundant codec translations when using audiohooks ... The main frame read and write handlers in main/channel.c don't use the optimum placement in the processing flow for calling audiohooks callbacks, as far as codec translation is concerned. This change places the audiohooks callback code: * After the channel read translation if the frame is not linear before the translation, thereby increasing the chance that the frame is linear as required by audiohooks * Before the channel write translation if the frame is linear at this point This prevents the audiohooks code from instantiating additional translation paths to/from linear where a linear frame format is already available, saving valuable CPU cycles ASTERISK-26419 Change-Id: I6edd5771f0740e758e7eb42558b953f046c01f8f	9 years ago
Badalyan Vyacheslav	3ab7fae96b	res_pjsip_config_wizard: Memory leak in module_unload ... Fixed a memory leak. It removes only the first element. Added a useful feature in vector.h to remove all items under the CMP through a callback function / macro. ASTERISK-26453 #close Change-Id: I84508353463456d2495678f125738e20052da950	9 years ago
Ludovic Gasc (GMLudo)	9f62feca60	res_calendar: Add support for fetching calendars when reloading ... We use a lot res_calendar, we are very happy with that, especially because you use libical, the almost alone opensource library that supports really ical format with all types of recurrency. Nevertheless, some features are missed for our business use cases. This first patch adds a new option in calendar.conf: fetch_again_at_reload. Be my guest for a better name. If it's true, when you'll launch "module reload res_calendar.so", Asterisk will download again the calendar. The business use case is that we have a WebUI with a scheduler planner, we know when the calendars are modified. For now, we need to define 1 minute of timeout to have a chance that our user doesn't wait too long between the modification and the real test. But it generates a lot of useless HTTP traffic. ASTERISK-26422 #close Change-Id: I384b02ebfa42b142bbbd5b7221458c7f4dee7077	9 years ago
Joshua Colp	7126194187	Merge "Revert "Packet-Loss Concealment (PLC) for supporting codecs.""	9 years ago
Badalyan Vyacheslav	ca2f3e5b99	cel_odbc: Fix memory leak on module unload ... Change-Id: Ic7a1236eba2408090fdabb5f717b5fa455ead715	9 years ago
George Joseph	5fb848eebd	bundled_pjproject: Add tests for programs used by the Makefile, et al. ... Added tests for bzip2, tar, patch, sed and nm to configure.ac. Set DOWNLOAD_TO_STDOUT to a working command line regardless of whether the download program is wget, curl or fetch. Added a 'configure.m4' file to the third-party directory which takes care of calling any third-party project setup. Had to move some pjproject_bundled stuff up in configure.ac so it was called before the third-party configure macro. The pjproject tarball is now downloaded to the externals_cache_dir if it was specified on the ./configure command line Removed regeneration of the pjproject aconfigure file. It was only needed for an old patch that no longer applies. Converted the tests for symbols to explicit tests since we know that they're now available in the bundled version. Saves a little time during configure. ASTERISK-26416 #close Reported-by: Corey Farrell Change-Id: Id1d94251c0155f8dd41b7de7067f35cfbaafbb9b (cherry picked from commit e6b0053d75) (cherry picked from commit a0d02f3832)	9 years ago
Joshua Colp	73f75c246b	Revert "Packet-Loss Concealment (PLC) for supporting codecs." ... This change introduced some fax test failures that have not yet been addressed. So this is not forgotten I'm submitting a change which reverts it. This reverts: d56fc3b36b. ASTERISK-25629 Change-Id: Ibc2f23c38643f5a2c89cf8915ae2d805b81bc3d5	9 years ago
George Joseph	c5e8f50169	pjproject_bundled: Add MALLOC_DEBUG capability ... pjproject_bundled will now use the asterisk memory debugging APIs if MALLOC_DEBUG is turned on in menuselect. Because this required stubs for the executable programs and the python bindings, some Makefile reorganization was needed to properly handle the dependencies. As a result, the makefile now individually makes each of the pjproject libraries separately instead of making them all in 1 shot. The only visible change is that there are separate status lines printed for each library instead oif 1 for all libs. Also, the making of the pjproject dependency files was eliminated. They're not needed for building unless you're actively modifying pjproject source files and it makes the build process faster. Finally, any issues with parallel builds should be resolved again making the build faster. Change-Id: Icc5e3d658fbfb00e0a46b44c66dcc2522d5171b0	9 years ago
George Joseph	442b597929	alembic: Allow cdr, config and voicemail to exist in the same schema ... cdr, config and voicemail are all separate alembic trees. Because alembic's default is to use a table named 'alembic_version' to store the current tree revision, the 3 trees can't exist in the same schema without stepping on each other. Now each tree uses 'alembic_version_<tree_name>' as the version table. Each tree's env.py script now first checks for 'alembic_version'. If it finds it AND its revision is in the tree's history, the script renames it to 'alembic_version_<tree_name>'. Regardless, the script then continues with the migration using 'alembic_version_<tree_name>' and creates that table if it's not found. The result is that if an existing 'alembic_version' table was found but it didn't belong to this tree, it's left alone and 'alembic_version_<tree_name>' is used or created. WARNING: If multiple trees are using the same schema, they MUST NOT CRU or D any objects with names that might exist in the other trees. An example would be 'yesno_values' type. If two trees perform operations on it, one tree could pull it out from under the other. Thankfully we currently don't share any names among cdr, config and voicemail. NOTE: Since the env.py scripts in each tree were identical, a common env.py has been placed in the ast-db-manage directory and a symlink to it has been placed in each tree directory. ASTERISK-24311 #close Reported-by: Dafi Ni Change-Id: I4d593f000350deb5d21a14fa1e9bc3896844d898	9 years ago
Joshua Colp	55e8c9eff2	Merge "chan_sip: Honor support of Symmetric Response (rport) for SIP requests."	9 years ago
Alexander Traud	c4268ec734	chan_sip: Honor support of Symmetric Response (rport) for SIP requests. ... In the SIP channel driver chan_sip, the default is "auto_force_rport". When no NAT was detected, for example in case of IPv6, Asterisk uses the IP address from the headers within the SIP-REGISTER for subsequent SIP signaling. When the remote party specifies support for Symmetric Response (RFC 3581) via the parameter "rport", Asterisk should not extract the port from the SIP headers but reuse the port of the transport. This did not happen because of a typo. ASTERISK-26438 #close Change-Id: If6e7891848aaf96666dee5305695f7c6667cd5a6	9 years ago
Joshua Colp	ac3c51f284	Merge "app_queue: Update dynamic members ringinuse on reload."	9 years ago
frahaase	c455823657	Binaural synthesis (confbridge): interleaved two-channel audio. ... Asterisk only supports mono audio at the moment. This patch adds interleaved two-channel audio to Asterisk's channels. ASTERISK-26292 Change-Id: I7a547cea0fd3c6d1e502709d9e7e39605035757a	9 years ago
Corey Farrell	2a03575c30	astobj2: Add backtrace to log_bad_ao2. ... * Compile __ast_assert_failed unconditionally. * Use __ast_assert_failed to log messages from log_bad_ao2 * Remove calls to ast_assert(0) that happen after log_bad_ao2 was run. Change-Id: I48f1af44b2718ad74a421ff75cb6397b924a9751	9 years ago
Rodrigo Ramírez Norambuena	79532bca75	Add text of cdr directory into README.md for ast-db-manage ... Change-Id: I68321c4bea50730c39fdb486e5f23aeadd1ad636	9 years ago
Etienne Lessard	806d08b675	app_queue: Update dynamic members ringinuse on reload. ... Previously, when reloading the members of a queue, the members added statically (i.e. defined in queues.conf) would see their "ringinuse" value updated but not the members added dynamically. This change makes dynamic members ringuse value to be updated on reload. Note that it's impossible to add a dynamic member with a specific ringinuse value. For both static and dynamic members, the ringinuse value can always be changed later on with command like "queue set ringinuse" or with the AMI action "QueueMemberRingInUse". So it's possible this commit could break a user workflow if he was changing the ringinuse value of dynamic members via such commands and was also relying on the fact that a queue reload would not update the dynamic members ringinuse value. ASTERISK-26330 Change-Id: I3745cc9a06ba7e02c399636f1ee9e58c04081f3f	9 years ago
Joshua Colp	224c295292	Merge "core: Remove ABI effects of LOW_MEMORY."	9 years ago
Joshua Colp	73061091b5	Merge "Remove "format_ogg_opus: New format""	9 years ago
Joshua Colp	222f8b132a	Merge "download_externals: Fix issue with re-install"	9 years ago
Kevin Harwell	d31ffb421c	Remove "format_ogg_opus: New format" ... This reverts commit 40aa28131b. ASTERISK-26426 #close Change-Id: I81e55c3c512f1dd6f49896f0c6b97a07d74fd8f5	9 years ago
Corey Farrell	8c5c95ad89	core: Remove ABI effects of LOW_MEMORY. ... This allows asterisk to compiled with LOW_MEMORY to load modules built without LOW_MEMORY. ASTERISK-26398 #close Change-Id: I24b78ac9493ab933b11087a8b6794f3c96d4872d	9 years ago
George Joseph	a77ebb2017	download_externals: Fix issue with re-install ... Needed to ignore an xmlstarlet return code for optional element. Change-Id: I6a96f709b4b38c9a3f3dda4e8b07903787e16873 Reported-by: Dan Jenkins	9 years ago
Corey Farrell	2d2a8944be	logger: Output early verbose messages to console. ... Verbose messages should be printed to the console if the sublevel is less than option_verbose. This fix ensures the welcome message with copyright and license are printed at daemon and interactive rasterisk startup. ASTERISK-26410 #close Change-Id: Ia44235e30ec328aba92ea2c8a837b094e65c9a03	9 years ago
zuul	3f62485ba7	Merge "chan_sip: Resolve externhost not to IPv6; instead go for IPv4."	9 years ago
George Joseph	6d548e736b	Merge "codec_opus: Add download ability to menuselect"	9 years ago
George Joseph	c7e4e035a2	Merge "codec_opus: Replace res_format_attr_opus with the one from codec_opus"	9 years ago
George Joseph	d205d8d4a1	Merge "format_ogg_opus: New format"	9 years ago
George Joseph	c7ef1e0af3	codec_opus: Add download ability to menuselect ... Updated codecs/codecs.xml to add codec_opus to the external download list. ASTERISK-26409 Change-Id: Ia07b36539f30e852125fb2b94147dc9774df31a4 (cherry picked from commit 2cdab0e36eec4997ca3bd85aa09efc477038e31c) (cherry picked from commit e9684f3acd0e8def0df582c1505dd39dd3fd1610)	9 years ago
George Joseph	5cc3c6679f	codec_opus: Replace res_format_attr_opus with the one from codec_opus ... Preparation ASTERISK-26409 Change-Id: I9f20e7cce00c32464d9a180e81283d49d199d0a3 (cherry picked from commit 59f7662a93)	9 years ago
George Joseph	40aa28131b	format_ogg_opus: New format ... Add Ogg/Opus playback support. This uses libopusfile in order to be able to read .opus files and play them back. Writing/recording support is not present at this time. ASTERISK-26409 Change-Id: I8815d23345108d8ca7c0bd640f6a1ce6b4f56955 (cherry picked from commit daee8bbd5209b4158bc1785eede845a26e6cbeaa)	9 years ago
George Joseph	43901e9418	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
zuul	eeeff9487f	Merge "chan_sip: Address runaway when realtime peers subscribe to mailboxes"	9 years ago
zuul	91513c5e8d	Merge "channels/chan_pjsip: fix HANGUPCAUSE function bug."	9 years ago
Alexander Traud	5dd99465d3	chan_sip: Resolve externhost not to IPv6; instead go for IPv4. ... For the channel driver chan_sip, you specify externhost=example.com in sip.conf when your Asterisk is behind a NAT and your IP address is assigned dynamically. Or stated differently: You do not have a static IP address to use "externaddr" directly. This NAT support is quite handy but just about IPv4. Previously, Asterisk resolved "externhost" to any IP version. When the first DNS answer resolved to an IPv6, Asterisk sent an IPv6 in SIP/SDP for origin (o=) and connection (c=). This happened in outgoing SIP-REGISTER and while answering SIP-INVITE. If the remote peer is IPv4-only, it might not handle o=/c= with an IPv6. This change makes sure, no IPv6 is resolved anymore for "externhost". ASTERISK-18232 #close Reported by: Jacek Kowalski Tested by: Alexander Traud patches: changes.patch submitted by Alessandro Crespi Change-Id: If68eedbeff65bd1c1d8a9ed921c02ba464b32dac	9 years ago
George Joseph	d425971009	chan_sip: Address runaway when realtime peers subscribe to mailboxes ... Users upgrading from asterisk 13.5 to a later version and who use realtime with peers that have mailboxes were experiencing runaway situations that manifested as a continuous stream of taskprocessor congestion errors, memory leaks and an unresponsive chan_sip. A related issue was that setting rtcachefriends=no NEVER worked in asterisk 13 (since the move to stasis). In 13.5 and earlier, when a peer tried to register, all of the stasis threads would block and chan_sip would again become unresponsive. After 13.5, the runaway would happen. There were a number of causes... * mwi_event_cb was (indirectly) calling build_peer even though calls to mwi_event_cb are often caused by build_peer. * In an effort to prevent chan_sip from being unloaded while messages were still in flight, destroy_mailboxes was calling stasis_unsubscribe_and_join but in some cases waited forever for the final message. * add_peer_mailboxes wasn't properly marking the existing mailboxes on a peer as "keep" so build_peer would always delete them all. * add_peer_mwi_subs was unsubscribing existing mailbox subscriptions then just creating them again. All of this was causing a flood of subscribes and unsubscribes on multiple threads all for the same peer and mailbox. Fixes... * add_peer_mailboxes now marks mailboxes correctly and build_peer only deletes the ones that really are no longer needed by the peer. * add_peer_mwi_subs now only adds subscriptions marked as "new" instead of unsubscribing and resubscribing everything. It also adds the peer object's address to the mailbox instead of its name to the subscription userdata so mwi_event_cb doesn't have to call build_peer. With these changes, with rtcachefriends=yes (the most common setting), there are no leaks, locks, loops or crashes at shutdown. rtcachefriends=no still causes leaks but at least it doesn't lock, loop or crash. Since making rtcachefriends=no work wasnt in scope for this issue, further work will have to be deferred to a separate patch. Side fixes... * The ast_lock_track structure had a member named "thread" which gdb doesn't like since it conflicts with it's "thread" command. That member was renamed to "thread_id". ASTERISK-25468 #close Change-Id: I07519ef7f092629e1e844f855abd279d6475cdd0	9 years ago
Joshua Colp	8d5219e6b8	Merge "core: Ensure presencestate subtype and message are NULL."	9 years ago
Joshua Colp	cdf2522160	Merge "res_odbc: Make pooling option deprecation notice more useful."	9 years ago
Joshua Colp	8966c8ec5a	Merge "cdr_mysql: fix UTC support"	9 years ago
Aaron An	18a8ca06eb	channels/chan_pjsip: fix HANGUPCAUSE function bug. ... HANGUPCAUSE not return 'SIP 200 Ok' when dialed channel answered. This patch change the call order of ast_queue_control_data and ast_queue_control in chan_pjsip_incoming_response. ASTERISK-26396 #close Reported by: AaronAn Tested by: AaronAn Change-Id: Ide2d31723d8d425961e985de7de625694580be61	9 years ago
zuul	9ef0eb6487	Merge "logger: Simplify ast_callid handling code."	9 years ago
Joshua Colp	57b29f3b69	Merge "logger: Always enable verbose for console channel."	9 years ago
Joshua Colp	a805d779e8	core: Ensure presencestate subtype and message are NULL. ... When retrieving presence state information there is no guarantee that the subtype and message passed in are set to NULL. This change ensures they are. ASTERISK-26397 #close Change-Id: If38cd730e409e9a9b6eb9adef6591d15a9e61f86	9 years ago
zuul	ccc0bfa69c	Merge "logger: Fix default console settings."	9 years ago
zuul	4caee4a11b	Merge "core: Fix LOW_MEMORY missing symbol ast_pbx_uuid_get."	9 years ago