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Author	SHA1	Message	Date
Joshua C. Colp	6b017a51e0	Merge "utils: Wrap socket() and pipe() to reduce syscalls"	7 years ago
Sean Bright	00b36bb045	build: Update config.guess and config.sub ... Pulled from the authoritative respository at: https://git.savannah.gnu.org/cgit/config.git/tree/ Change-Id: I748708ce24d4d47ff1f395075d0b08d3da3355e0	7 years ago
George Joseph	d1598dbc7d	Revert "RTP: reset DTMF last seqno/timestamp on voice packet with marker bit" ... This reverts commit 3f53041267. Pending resolution of ASTERISK_28200 Change-Id: Iad4f3614cac95b00fdbb2b799aab8ae6285ec988	7 years ago
Sebastian Damm	a24bb1c4b6	res/res_ari: Add additional hangup reasons ... The ARI DELETE /channels command takes a "reason" parameter Previously, there were only five reasons implemented This patch adds more reasons to choose from for more complex setups ASTERISK-28198 #close Change-Id: I85996f1076c9946d65c778413f040a845a90fecc	7 years ago
Joshua C. Colp	fe49d04519	Merge "chan_sip: Fix leak using contact ACL"	7 years ago
Sean Bright	6d69fb3cc2	utils: Wrap socket() and pipe() to reduce syscalls ... Some platforms provide an implementation of socket() and pipe2() that allow the caller to specify that the resulting file descriptors should be non-blocking. Using these allows us to potentially elide 3 calls into 1 by avoiding extraneous calls to fcntl() to set the O_NONBLOCK flag afterwards. In passing, change ast_alertpipe_init() to use pipe2() directly instead of the wrapper if it is available. Change-Id: I3ebe654fb549587537161506c6c950f4ab298bb0	7 years ago
George Joseph	3f3dd992a2	stasis: Allow filtering by formatter ... A subscriber can now indicate that it only wants messages that have formatters of a specific type. For instance, manager can indicate that it only wants messages that have a "to_ami" formatter. You can combine this with the existing filter for message type to get only messages with specific formatters or messages of specific types. ASTERISK-28186 Change-Id: Ifdb7a222a73b6b56c6bb9e4ee93dc8a394a5494c	7 years ago
David M. Lee	b899119a5d	Removing registrar_expire from basic-pbx config ... The module has been removed, so it shouldn't be in the default config any more. Change-Id: Ie7e09f00f9c9a885574e29478250de4c2cefd9f1	7 years ago
Giuseppe Sucameli	0bde3751a0	chan_sip: Fix leak using contact ACL ... Free old peer's contactacl before overwrite it within build_peer. ASTERISK-28194 Change-Id: Ie580db6494e50cee0e2a44b38e568e34116ff54c	7 years ago
George Joseph	19c4e0f592	CI: Various updates to buildAsterisk.sh ... * Added ---no-configure, --no-menuselect, --no-make and --no-alembic options that prevent those actions from being performed. Useful for testing and re-running portions of the build after fixing earlier failures. * Added "set -e" to abort the script on command failure. Not sure why this wasn't there in the first place. * Fixed a few echos that were redirecting to stderr when they shouldn't have been. * Catch more alembic failures by actually trying to generate the SQL. Change-Id: I9f395fa4e9254be7299e7c1014f1a13db78faffb	7 years ago
George Joseph	64b0bcacb5	Merge "test_websocket_client.c: Disable websocket_client_create_and_connect test."	7 years ago
Kevin Harwell	cbb7633ad3	pjsip_add_use_callerid_contact: fixed alembic script ... Change-Id: I413f1583c797fb79651786cd8d0b003599f8ed10	7 years ago
Sean Bright	8f5df046f6	core: Add some documentation to the malloc_trim code ... This adds documentation to handle_cli_malloc_trim() indicating how it can be useful when debugging OOM conditions. Change-Id: I1936185e78035bf123cd5e097b793a55eeebdc78	7 years ago
George Joseph	ff02c93d9e	Merge "core: Merge malloc_trim patch"	7 years ago
Chris-Savinovich	58e50e56cb	core: Merge malloc_trim patch ... We've had multiple opportunities where Richard Mudgett's malloc_trim patch has been useful. Let's get it pushed up to gerrit and merged. Since malloc_trim is only available in libc, an entry is added to configure.ac to create a definition for HAVE_MALLOC_TRIM. Change-Id: Ia38308c550149d9d6eae4ca414a649957de9700c	7 years ago
Sungtae Kim	8644511cbf	res_pjsip: Patch for res_pjsip_* module load/reload crash ... The session_supplements for the pjsip makes crashes when the module load/unload. ASTERISK-28157 Change-Id: I5b82be3a75d702cf1933d8d1417f44aa10ad1029	7 years ago
lvl	140702ba2d	app_queue: Revert broken queue channel reference patch ... Revert commit 6409e7b11a, and add NULL checks for all app_queue event handling code. Related issues: ASTERISK~25185, ASTERISK~27006, ASTERISK~25844 ASTERISK-28125 Change-Id: I37334ea184ebb56e54471496b82937d4927815a0	7 years ago
Chris-Savinovich	6c13b20803	test_websocket_client.c: Disable websocket_client_create_and_connect test. ... This test was occasionally failing, with: WARNING[5812]: http.c:1939 httpd_helper_thread: Failed to set TCP_NODELAY on HTTP connection: Bad file descriptor ERROR[5812]: iostream.c:91 ast_iostream_nonblock: Failed to get fcntl() flags for file descriptor: Bad file descriptor ERROR[5812]: iostream.c:569 ast_iostream_close: close() failed: Bad file descriptor Disabled for now by making the test explicit only. Change-Id: I778f6cbb6104c6b4e89737a2eaf1a9540888d351	7 years ago
Pirmin Walthert	ecb9ed0958	pjproject_bundled: check whether UPDATE is supported on outgoing calls ... In ASTERISK-27095 an issue had been fixed because of which chan_pjsip was not trying to send UPDATE messages when connected_line_method was set to invite. However this only solved the issue for incoming INVITES. For outgoing INVITES (important when transferring calls) the options variable needs to be updated at a different place. ASTERISK-28182 #close Reported-by: nappsoft Change-Id: I76cc06da4ca76ddd6dce814a8b97cc66b98aaf29	7 years ago
George Joseph	efeab21b52	Merge "Revert "app_voicemail: Remove need to subscribe to stasis""	7 years ago
George Joseph	945451af90	Merge "bridges: Remove reliance on stasis caching"	7 years ago
George Joseph	4f0bf0270e	Revert "app_voicemail: Remove need to subscribe to stasis" ... This reverts commit 29115e2384. That commit closed a long standing hole which allowed subscriptions to mailboxes that weren't configured in voicemail.conf. This caused an issue with FreePBX which depdended on that behavior. The commit is being reverted until FreePBX can handle the new behavior. ASTERISK-28151 Reported by: Ronald Raikes Change-Id: I57b7b85e75d7dd97c742b5c69d718a0f61260c15	7 years ago
Kevin Harwell	7d37967e7e	Merge "jansson: Upgrade to 2.12."	7 years ago
George Joseph	f4924d40db	test_cel: Plug a few ref leaks ... These are only a few of the leaks. The large number of macros and return paths in this file would make a weeks worth of work to plug them all. Change-Id: Ie2369fa944023d44767871c5c30974cb077ffb56	7 years ago
George Joseph	3667c5e1d2	bridges: Remove reliance on stasis caching ... * The bridging core no longer uses the stasis cache for bridge snapshots. The latest bridge snapshot is now stored on the ast_bridge structure itself. * The following APIs are no longer available since the stasis cache is no longer used: ast_bridge_topic_cached() ast_bridge_topic_all_cached() * A topic pool is now used for individual bridge topics. * The ast_bridge_cache() function was removed since there's no longer a separate container of snapshots. * A new function "ast_bridges()" was created to retrieve the container of all bridges. Users formerly calling ast_bridge_cache() can use the new function to iterate over bridges and retrieve the latest snapshot directly from the bridge. * The ast_bridge_snapshot_get_latest() function was renamed to ast_bridge_get_snapshot_by_uniqueid(). * A new function "ast_bridge_get_snapshot()" was created to retrieve the bridge snapshot directly from the bridge structure. * The ast_bridge_topic_all() function now returns a normal topic not a cached one so you can't use stasis cache functions on it either. * The ast_bridge_snapshot_type() stasis message now has the ast_bridge_snapshot_update structure as it's data. It contains the last snapshot and the new one. * cdr, cel, manager and ari have been updated to use the new arrangement. Change-Id: I7049b80efa88676ce5c4666f818fa18ad1985369	7 years ago
Jenkins2	4ca709768d	Merge "stasis: Segment channel snapshot to reduce creation cost."	7 years ago
Joshua Colp	a6b37e5c43	Merge "astobj2: Create function to copy weak proxied objects from container."	7 years ago
Joshua Colp	b80c9071e3	Merge "RTP: need to reset DTMF last seqno/timestamp on voice packet with marker bit"	7 years ago
Joshua Colp	50ac85cb40	stasis: Segment channel snapshot to reduce creation cost. ... When a channel snapshot was created it used to be done from scratch, copying all data (many strings). This incurs a cost when doing so. This change segments the channel snapshot into different components which can be reused if unchanged from the previous snapshot creation, reducing the cost. In normal cases this results in some pointers being copied with reference count being bumped, some integers being set, and a string or two copied. The other benefit is that it is now possible to determine if a channel snapshot update is redundant and thus stop it before a message is published to stasis. The specific segments in the channel snapshot were split up based on whether they are changed together, how often they are changed, and their general grouping. In practice only 1 (or 0) of the segments actually get changed in normal operation. Invalidation is done by setting a flag on the channel when the segment source is changed, forcing creation of a new segment when the channel snapshot is created. ASTERISK-28119 Change-Id: I5d7ef3df963a88ac47bc187d73c5225c315f8423	7 years ago
Joshua Colp	d0ccbb3377	stasis: Use an implementation specific channel snapshot cache. ... Channels no longer use the Stasis cache for channel snapshots. Instead they are stored in a hash table in stasis_channels which reduces the number of Stasis messages created and allows better storage. As a result the following APIs are no longer available since the stasis cache is no longer used: ast_channel_topic_cached() ast_channel_topic_all_cached() The ast_channel_cache_all() and ast_channel_cache_by_name() functions now return an ao2_container of ast_channel_snapshots rather than a container of stasis_messages therefore you can't (and don't need to) call stasis_cache functions on it. The ast_channel_topic_all() function now returns a normal topic not a cached one so you can't use stasis cache functions on it either. The ast_channel_snapshot_type() stasis message now has the ast_channel_snapshot_update structure as it's data. It contains the last snapshot and the new one. ast_channel_snapshot_get_latest() still returns the latest snapshot. The latest snapshot is now stored on the channel itself to eliminate cache hits when Stasis messages that have the snapshot as a payload are created. ASTERISK-28102 Change-Id: I9334febff60a82d7c39703e49059fa3a68825786	7 years ago
Joshua Colp	0a60bc1a68	Merge "func_strings: HASHKEY - negative array index can cause corruption"	7 years ago
Corey Farrell	8e1ab4f11c	jansson: Upgrade to 2.12. ... This brings in jansson-2.12, removes all patches that were merged upstream. README is created in third-party/jansson/patches to explain how to add patches but also because the patches folder must exist for the build process to succeed. Change-Id: If0f2d541c50997690660c21fb7b03d625a5cdadd	7 years ago
Alexei Gradinari	3f53041267	RTP: need to reset DTMF last seqno/timestamp on voice packet with marker bit ... The marker bit set on the voice packet indicates the start of a new stream and a new time stamp. Need to reset the DTMF last sequence number and the timestamp of the last END packet. If the new time stamp is lower then the timestamp of the last DTMF END packet the asterisk drops all DTMF frames as out of order. This bug was caught using Cisco ip-phone SPA50X and codec g722. On SIP session update the SPA50X resets stream indicating it with market bit and a new timestamp is twice smaller then the previous. ASTERISK-28162 #close Change-Id: If9c5742158fa836ad549713a9814d46a5d2b1620	7 years ago
Corey Farrell	021ce938ca	astobj2: Remove legacy ao2_container_alloc routine. ... Replace usage of ao2_container_alloc with ao2_container_alloc_hash or ao2_container_alloc_list. Remove ao2_container_alloc macro. Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088	7 years ago
Corey Farrell	bc7f4f4db3	astobj2: Create function to copy weak proxied objects from container. ... Create ao2_container_dup_weakproxy_objs to perform a similar function to ao2_container_dup. This function expects the source container to have weakproxy objects, inserts the associated non-weak objects into the destination container. Orphaned weakproxy objects are ignored. Create test for this new function and for ao2_weakproxy_find. Change-Id: I898387f058057e08696fe9070f8cd94ef3a27482	7 years ago
Joshua Colp	b7af9c8b19	Merge "stasis: Remove stringfields and lock from change message."	7 years ago
Joshua Colp	9b0808e2d0	Merge "app_queue: Cleanup queue_ref / queue_unref routines."	7 years ago
Kevin Harwell	4b5d11ec17	func_strings: HASHKEY - negative array index can cause corruption ... This patch makes it so only matching non-empty key names, and keys created by the HASH function are eligible for inclusion in the comma separated string. It also fixes a bug where it was possible to write to a negative index if the result buffer was empty. ASTERISK-28159 patches: ASTERISK-28159.diff submitted by Michael Walton (license 6502) Change-Id: I6e57fe7307dfd856271753aed5ba64c59b511487	7 years ago
George Joseph	bcdfb90362	CI: Get job timeouts from environment ... The job timeouts were hard coded in the jenkinsfiles which means changes had to go through gerrit. Now they are taken from the following environment variables (and their defaults) that can be set in Jenkins configuration... TIMEOUT_GATES = "60 MINUTES" TIMEOUT_DAILIES = "3 HOURS" TIMEOUT_REF_DEBUG = "24 HOURS" TIMEOUT_UNITTESTS = "30 MINUTES" Change-Id: I673a551c1780bf665a3bc160b245da574aa4bbab	7 years ago
Joshua Colp	9df772530f	Merge "test_res_pjsip_scheduler: Fix possible write after free in scheduler_policy."	7 years ago
Joshua Colp	9f996ca1f0	Merge "res/res_ari: Fix null endpoint handle"	7 years ago
Joshua Colp	e0816637a4	Merge "bridge_native_rtp.c: Fail native bridge if no framing match."	7 years ago
Joshua Colp	b285333673	Merge "res_pjsip_caller_id: Use static pj_str_t for fromto header names."	7 years ago
Joshua Colp	614c8a60d7	Merge "stasis: Add internal filtering of messages."	7 years ago
Corey Farrell	64e21c9ea9	app_queue: Cleanup queue_ref / queue_unref routines. ... This replaces the inline functions with macros. This removes the need to directly use __ao2_ref, opts instead for standard ao2_bump and ao2_cleanup macros. Change-Id: If4e04e9bab2e3c883188437cb9f487b3e498a21b	7 years ago
George Joseph	ece5f8015f	backtrace: Refactor ast_bt_get_symbols so it doesn't crash ... We've been seeing crashes in libbfd when we attempt to generate a stack trace from multiple threads. It turns out that libbfd is NOT thread-safe. It can cache the bfd structure and give it to multiple threads without protecting itself. To get around this, we've added a global mutex around the bfd functions and also have refactored the use of those functions to be more efficient and to provide more information about inlined functions. Also added a few more tests to test_pbx.c. One just calls ast_assert() and the other calls ast_log_backtrace(). Neither are run by default. WARNING: This change necessitated changing the return value of ast_bt_get_symbols() from an array of strings to a VECTOR of strings. However, the use of this function outside Asterisk is not likely. ASTERISK-28140 Change-Id: I79d02862ddaa2423a0809caa4b3b85c128131621	7 years ago
Joshua C. Colp	56eb18f395	stasis: Remove stringfields and lock from change message. ... When a subscribe or unsubscribe occurs a message is published containing this information. This change makes it so that the message no longer uses stringfields or a lock, as both are not really needed for the message. Change-Id: I3f4831931d79f94fd979baf48048738df5dc1632	7 years ago
Alexei Gradinari	fa048183aa	pjsip: New function PJSIP_PARSE_URI to parse URI and return part of URI ... New dialplan function PJSIP_PARSE_URI added to parse an URI and return a specified part of the URI. This is useful when need to get part of the URI instead of cutting it using a CUT function. For example to get 'user' part of Remote URI ${PJSIP_PARSE_URI(${CHANNEL(pjsip,remote_uri)},user)} ASTERISK-28144 #close Change-Id: I5d828fb87f6803b6c1152bb7b44835f027bb9d5a	7 years ago
George Joseph	d81a43893d	Merge "pjproject-bundled: Use AST_DEVMODE for conditional compilation."	7 years ago
Joshua Colp	3077ad0c24	stasis: Add internal filtering of messages. ... This change adds the ability for subscriptions to indicate which message types they are interested in accepting. By doing so the filtering is done before being dispatched to the subscriber, reducing the amount of work that has to be done. This is optional and if a subscriber does not add message types they wish to accept and set the subscription to selective filtering the previous behavior is preserved and they receive all messages. There is also the ability to explicitly force the reception of all messages for cases such as AMI or ARI where a large number of messages are expected that are then generically converted into a different format. ASTERISK-28103 Change-Id: I99bee23895baa0a117985d51683f7963b77aa190	7 years ago