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Author	SHA1	Message	Date
George Joseph	3464093f85	Merge "res_pjsip: Patch for res_pjsip_* module load/reload crash"	7 years ago
George Joseph	2e21910ca1	Merge "res_rtp_asterisk: Remove some unused structure fields."	7 years ago
George Joseph	c23c8d92d5	app_voicemail: Don't delete mailbox state unless mailbox is deleted ... The free_user function was automatically deleting the stasis mailbox state but this only makes sense when the mailbox is actually deleted, not just the structure freed. This was causing issues where leave_voicemail would publish the mwi message to stasis and delete the state before the message could be processed by res_pjsip_mwi. * Removed the delete of state from free_user(). * Created a new free_user_final() function that both frees the data structure and deletes the state. This function is only called during module load/unload where it's appropriate to delete the state. ASTERISK-28215 Change-Id: I305e8b3c930e9ac41d901e5dc8a58fd7904d98dd	7 years ago
Joshua C. Colp	768be60fbe	Merge "res_format_attr_h264.c: Make sure profile-level-id fmtp attribute is set"	7 years ago
Sean Bright	357219dfb3	res_rtp_asterisk: Remove some unused structure fields. ... All of the fields that were removed were no longer referenced except for 'lastrxts' and 'rxseqno' which were only ever written to. Change-Id: I5a5d31eb33e97663843698f58d0d97f22a76627c	7 years ago
Joshua C. Colp	0eaa736541	Merge "bridge_builtin_features.c: Set auto(mix)mon variables on both channels"	7 years ago
Sean Bright	5b12dfa6dd	res_format_attr_h264.c: Make sure profile-level-id fmtp attribute is set ... The profile-iop octet (the 2nd) of profile-level-id can be zero according to RFC 6184 Section 8.1. So we ignore its value when deciding to include profile-level-id in the outgoing SDP. ASTERISK-27959 #close Reported by: David Kuehling Change-Id: Id28cd916a3d7748058fe9609b585d07d9e243f73	7 years ago
Joshua C. Colp	b701c8b8a0	Merge "confbridge: announce to the marked users when they join an empty conference"	7 years ago
Sean Bright	3db1df301e	bridge_builtin_features.c: Set auto(mix)mon variables on both channels ... This is how features behaved up through Asterisk 11, but was changed when the new bridging framework was implemented in Asterisk 12. Reported by rrittgarn in #asterisk. Change-Id: I72cf86223947a8118c75f46e2c603dbc11e3125b	7 years ago
Joshua C. Colp	e12ba0a600	Merge "utils: Don't set or clear flags that don't need setting or clearing"	7 years ago
Friendly Automation	4e8aa3b68c	Merge "stasis: Add statistics gathering in developer mode."	7 years ago
Joshua C. Colp	a28f0382e8	Merge "Use non-blocking socket() and pipe() wrappers"	7 years ago
Alexei Gradinari	cb1a08bdcb	confbridge: announce to the marked users when they join an empty conference ... Currently the file sound_only_person is not played when a marked user (with announce_only_user=yes) joins an empty conference. This patch fixes it. ASTERISK-28201 #close Change-Id: I85b67687e6b220939c3af8091d83a70a7b174cf4	7 years ago
Joshua C. Colp	fe07093660	stasis: Add statistics gathering in developer mode. ... This change adds statistics gathering to Stasis topics, subscriptions, and message types. These can be viewed using CLI commands and provide insight into how Stasis is used and how long certain operations take to execute. These are only available when Asterisk is compiled in developer mode and do not have any impact under normal operation. ASTERISK-28117 Change-Id: I94411b53767f89ee01714daaecf0c2f1666e863f	7 years ago
Friendly Automation	1f8062c6a6	Merge "stasis: Allow filtering by formatter"	7 years ago
Joshua C. Colp	3ee040e9cd	Merge "build: Update config.guess and config.sub"	7 years ago
George Joseph	5a0a292843	Merge "pjproject_bundled: check whether UPDATE is supported on outgoing calls"	7 years ago
George Joseph	5d4d723844	Merge "Revert "RTP: reset DTMF last seqno/timestamp on voice packet with marker bit""	7 years ago
Sean Bright	42ff856216	Use non-blocking socket() and pipe() wrappers ... Change-Id: I050ceffe5a133d5add2dab46687209813d58f597	7 years ago
Sean Bright	bedf16b041	utils: Don't set or clear flags that don't need setting or clearing ... Change-Id: I0e7fb507ac09b15e45e1ff8501ecfca67afa5217	7 years ago
George Joseph	82473227be	Merge "CI: Various updates to buildAsterisk.sh"	7 years ago
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