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Author	SHA1	Message	Date
George Joseph	b39bbc0c6a	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	741c2e1570	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
Corey Farrell	d236973657	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
Alexander Traud	16d9d67329	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	21ee185f57	Merge "Remove "format_ogg_opus: New format"" into 14	9 years ago
Joshua Colp	79adec5d93	Merge "download_externals: Fix issue with re-install" into 14	9 years ago
Kevin Harwell	65d0cbea1b	Remove "format_ogg_opus: New format" ... This reverts commit 40aa28131b. ASTERISK-26426 #close Change-Id: I81e55c3c512f1dd6f49896f0c6b97a07d74fd8f5	9 years ago
George Joseph	ea29ff9023	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	013414ca4b	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	0b62db30ce	Merge "chan_sip: Resolve externhost not to IPv6; instead go for IPv4." into 14	9 years ago
George Joseph	af399b0cb2	Merge "codec_opus: Add download ability to menuselect" into 14	9 years ago
George Joseph	ff35463aba	Merge "codec_opus: Replace res_format_attr_opus with the one from codec_opus" into 14	9 years ago
George Joseph	9a1f7dab49	Merge "format_ogg_opus: New format" into 14	9 years ago
George Joseph	f243be6a8c	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)	9 years ago
George Joseph	550608db04	codec_opus: Replace res_format_attr_opus with the one from codec_opus ... Preparation ASTERISK-26409 Change-Id: I9f20e7cce00c32464d9a180e81283d49d199d0a3	9 years ago
George Joseph	b4642fd6a6	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	9 years ago
George Joseph	ce6405a234	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	f029ba6564	Merge "channels/chan_pjsip: fix HANGUPCAUSE function bug." into 14	9 years ago
Aaron An	7bcb1fc940	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
Alexander Traud	175975e994	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	d64edafa63	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
zuul	d4422a2253	Merge "core: Ensure presencestate subtype and message are NULL." into 14	9 years ago
Joshua Colp	94b655ab05	Merge "res_odbc: Make pooling option deprecation notice more useful." into 14	9 years ago
zuul	5b27d064af	Merge "cdr_mysql: fix UTC support" into 14	9 years ago
zuul	c526e3d94c	Merge "logger: Simplify ast_callid handling code." into 14	9 years ago
Joshua Colp	64df75b02c	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	79070e486d	Merge "logger: Always enable verbose for console channel." into 14	9 years ago
zuul	6b8de13946	Merge "logger: Fix default console settings." into 14	9 years ago
Joshua Colp	f63ff293eb	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	5325e7a500	Merge "core: Fix LOW_MEMORY missing symbol ast_pbx_uuid_get." into 14	9 years ago
Tzafrir Cohen	fc16739600	cdr_mysql: fix UTC support ... * Make 'cdrzone=UTC' work properly. * Fix the documentation of cdr_mysql.conf: it's cdrzone and not timezone ASTERISK-26359 #close Change-Id: I2a6f67b71bbbe77cac31a34d0bbfb1d67c933778	9 years ago
Joshua Colp	2684ffd9cd	odbc: Remove options that are no longer applicable. ... The pooling, shared_connection, limit, and idlecheck options are no longer used in res_odbc. ASTERISK-26389 Change-Id: I2fde7b467d01f9d1c82cc0a339bb4f7e1dd6bbe6	9 years ago
zuul	7f8d570c8d	Merge "asterisk.c: Non-root users also get the astcanary after core restart." into 14	9 years ago
Corey Farrell	ba3269f12b	core: Fix LOW_MEMORY missing symbol ast_pbx_uuid_get. ... Move the function outside the conditional block that excludes LOW_MEMORY. ASTERISK-26273 #close Change-Id: Ic290fa128222c410c3531107e30efacabc8493b4	9 years ago
zuul	381cafe335	Merge "sd_notify (systemd status notifications) support" into 14	9 years ago
Corey Farrell	1a7d39818f	logger: Simplify ast_callid handling code. ... Routines responsible for managing ast_callid's are overly complicated. This is left-over code from when ast_callid was an AO2 object. Now that it is an integer the code can be reduced. ast_callid handler code no longer prints it's own error message upon failure to allocate threadstorage as ast_calloc would have already printed a message. Debug messages that were printed when TEST_FRAMEWORK was enabled have been also been removed. Change-Id: I65a768a78dc6cf3cfa071e97f33ce3dce280258e	9 years ago
zuul	2c8b8a9ecb	Merge "res_pjsip_multihomed: Change Contact port to listening port." into 14	9 years ago
Corey Farrell	1b61a21757	logger: Always enable verbose for console channel. ... Previous versions of Asterisk did not require verbose to be specified in logger.conf for the console channel, if it was requested by command line or asterisk.conf it just worked. This change causes Asterisk to always enable verbose in the console channel level mask. Verbose is displayed on consoles if requested by command line, option_verbose or 'core set verbose'. This also delays initialization of the logger until after threadstorage is initialized. Initializing too early can cause messages to be printed multiple times to the console (stdout). ASTERISK-26391 #close Change-Id: I52187d67c2fcb3efd5561bf04b3e5e23e5ee8a04	9 years ago
Corey Farrell	68ff7362ae	logger: Fix default console settings. ... When logger.conf is missing or invalid we should be printing notices, warnings and errors to the console. The logmask was incorrectly calculated. Change-Id: Ibaa9465a8682854bc1a5e9ba07079bea1bfb6bb3	9 years ago
zuul	5205b0196b	Merge "rtp: Only accept the first payload for a format in SDP." into 14	9 years ago
Tzafrir Cohen	29e096cd13	sd_notify (systemd status notifications) support ... sd_notify() is used to notify systemd of changes to the status of the process. This allows the systemd daemon to know when the process finished loading (and thus only start another program after Asterisk has finished loading). To use this, use a systemd unit with 'Type=notify' for Asterisk. This commit also adds the function ast_sd_notify(), a wrapper around sd_notify that does nothing if not built with systemd support. Also adds support for libsystemd detection in the configure script. Change-Id: Ied6a59dafd5ef331c5c7ae8f3ccd2dfc94be7811 (cherry picked from commit 07b95f7c65)	9 years ago
Walter Doekes	2b75559e6c	asterisk.c: Non-root users also get the astcanary after core restart. ... Without this change, a 'core restart' would kill the astcanary forever if you're not running as root. Both with and without this patch, the scheduling priority was still SCHED_RR after restart. Additionally, the astcanary is now spawned if you start with high priority and Asterisk doesn't get a chance to lower it. For example through: `chrt -r 10 sudo -u asterisk asterisk -c` Also reap killed astcanary processes on core restart. ASTERISK-26352 #close Change-Id: Iacb49f26491a0717084ad46ed96b0bea5f627a55	9 years ago
zuul	b56717b396	Merge "asterisk.c: When astcanary dies on linux, reset priority on all threads." into 14	9 years ago
zuul	7340a63adf	Merge "Fix showing of swap details when sysinfo() is available" into 14	9 years ago
zuul	2acd3e38ae	Merge "res_config_odbc.c: Fix buffer size limitation creating invalid SQL." into 14	9 years ago
Walter Doekes	bd39d2c0ab	asterisk.c: When astcanary dies on linux, reset priority on all threads. ... Previously only the canary checking thread itself had its priority set to SCHED_OTHER. Now all threads are traversed and adjusted. ASTERISK-19867 #close Reported by: Xavier Hienne Change-Id: Ie0dd02a3ec42f66a78303e9c1aac28f7ed9aae39	9 years ago
Joshua Colp	64d2ea44d8	Merge "pjproject_bundled: Prevent SERVFAIL from marking name server bad" into 14	9 years ago
Richard Mudgett	4c61d9f629	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
Timo Teräs	1e9ae453fd	Fix showing of swap details when sysinfo() is available ... If sysinfo() is available, but not sysctl() or swapctl() the printing code for swap buffer sizes is incorrectly omitted. The above condition happens with musl c-library. Fix #if rule to consider defined(HAVE_SYSINFO). And also remove the redundant || defined(HAVE_SYSCTL) which was incorrectly there to start with. Now swap information is displayed only if an actual libc function to get it is available. This also fixes warnings previously seen with musl libc: [CC] asterisk.c -> asterisk.o asterisk.c: In function 'handle_show_sysinfo': asterisk.c:773:6: warning: variable 'totalswap' set but not used [-Wunused-but-set-variable] int totalswap = 0; ^~~~~~~~~ asterisk.c:770:11: warning: variable 'freeswap' set but not used [-Wunused-but-set-variable] uint64_t freeswap = 0; ^~~~~~~~ Change-Id: I1fb21dad8f27e416c60f138c6f2bff03fb626eca	9 years ago
Joshua Colp	696824df89	rtp: Only accept the first payload for a format in SDP. ... When receiving an SDP offer with multiple payloads for the same format we would generate an answer with the first payload, but during the payload crossover operation (to set the payloads for receiving) we would remove all payloads but the last. This would result in incoming traffic being matched against the wrong format and outgoing traffic being sent using the wrong payload. This change makes it so that once a format has a payload number put into the mapping all subsequent ones are ignored. This ensures there is only ever one payload in the mapping and that it is the payload placed into the answer SDP. ASTERISK-26365 #close Change-Id: I1e8150860a3518cab36d00b1fab50f9352b64e60	9 years ago