27759 Commits (65d0c0beb12e28df23532a80b6fa8300b83da39c)

 

All Branches   

Search

Author	SHA1	Message	Date
George Joseph	8dc8d6ceb8	res_rtp_asterisk: Fix placement of txcount increment ... Commit 1bce690ccb was incrementing txcount for rtcp packets as well as rtp packets and that was causing sender reports to be generated instead of receiver reports in cases where no rtp was actually being sent. Moved the txcount increment from __rtp_sento, which handles both rtp and rtcp, to rtp_sento which only handles rtp packets. Discovered by the hep/rtcp-receiver test. Change-Id: Ie442e4bb947a68847a676497021ba10ffaf376d5	9 years ago
George Joseph	c7eb18d865	chan_pjsip: Add 'pjsip show channelstats' ... Added the ability to show channel statistics to chan_pjsip (cli_functions.c) Moved the existing 'pjsip show channel(s)' functionality from pjsip_configuration to cli_functions.c. The stats needed chan_pjsip's private header so it made sense to move the existing channel commands as well. Now using stasis_cache_dump to get the channel snapshots rather than retrieving all endpoints, then getting each one's channel snapshots. Much more efficient. Change-Id: I03b114522126d27434030b285bf6d531ddd79869	9 years ago
zuul	e595f02f83	Merge "res_rtp_asterisk: Fix packet stats on bridged connection" into 13	9 years ago
George Joseph	1583559a06	res_pjsip/pjsip_options: Fix From generation on outgoing OPTIONS ... No one seemed to notice but every time an OPTIONS goes out, it goes out with a From of "asterisk" (or whatever the default from_user is set to), even if you specify an endpoint. The issue had several causes... qualify_contact is only called with an endpoint if called from the CLI. If the endpoint is NULL, qualify_contact only looks up the endpoint if authenticate_qualify=yes. Even then, it never passes it on to ast_sip_create_request where the From header is set. Therefore From is always "asterisk" (or whatever the default from_user is set to). Even if ast_sip_create_request were to get an endpoint, it only sets the From if endpoint->from_user is set. The fix is 4 parts... First, create_out_of_dialog_request was modified to use the endpoint id if endpoint was specified and from_user is not set. Second, qualify_contact was modified to always look up an endpoint if one wasn't specified regardless of authenticate_qualify. It then passes the endpoint on to create_out_of_dialog_request. Third (and most importantly), find_an_endpoint was modified to find an endpoint by using an "aors LIKE %contact->aor%" predicate with ast_sorcery_retrieve_by_fields. As such, this patch will only work if the sorcery realtime optimizations patch goes in. Otherwise we'd be pulling the entire endpoints database every time we send an OPTIONS. Since we already know the contact's aor, the on_endpoint callback was also modified to just check if the contact->aor is an exact match to one of the endpoint's. Finally, since we now have an endpoint for every OPTIONS request, res_pjsip/endpt_send_request (which handles out-of-dialog reqests) was updated to get the transport from the endpoint and set it on tdata. Now the correct transport is used. Change-Id: I2207e12bb435e373bd1e03ad091d82e5aba011af	9 years ago
Joshua Colp	7b7e3909e4	Merge "sorcery/res_pjsip: Refactor for realtime performance" into 13	9 years ago
Joshua Colp	e0a916d6f5	Merge "app_echo: forward and generate VIDUPDATE frames" into 13	9 years ago
Jacek Konieczny	0cfab30b28	res_rtp_asterisk: Use separate SRTP session for RTCP with DTLS ... Asterisk uses separate UDP ports for RTP and RTCP traffic and RFC 5764 explicitly states: There MUST be a separate DTLS-SRTP session for each distinct pair of source and destination ports used by a media session This means RTP keying material cannot be used for DTLS RTCP, which was the reason why RTCP encryption would fail. ASTERISK-25642 Change-Id: I7e8779d8b63e371088081bb113131361b2847e3a	9 years ago
zuul	3528a9a42e	Merge "res_parking: Misc fixes." into 13	9 years ago
Jacek Konieczny	6a9c18fb59	app_echo: forward and generate VIDUPDATE frames ... When using app_echo via WebRTC with VP8 video the video would appear only after a few minutes, because there would be nothing to request a full reference frame. This fixes the problem in both ways: - echos any VIDUPDATE frames received on the channel - sends one such frame when first video frame is to be forwarded This makes the echo work with Firefox and Chrome WebRTC implementation. ASTERISK-25867 #close Change-Id: I73bda87bf7532ee8bfb28d917045a21034908c1e	9 years ago
George Joseph	1bce690ccb	res_rtp_asterisk: Fix packet stats on bridged connection ... rxcount, txcount, rxoctetcount and txoctetcount weren't being calculated for bridged streams because the calulations were being done after the bridged short-circuit. Actually, rxoctetcount wasn't ever being calculated. Moved the calculations so they occur for all valid received packets and all transmitted packets. Also added rxoctetcount and txoctetcount to ast_rtp_instance_stat. Change-Id: I08fb06011a82d38c3b4068867a615068fbe59cbb	9 years ago
Joshua Colp	c88c09a920	Merge "res_parking: Fix blind transfer dynamic lots creation." into 13	9 years ago
zuul	3974bf1ef0	Merge "res_parking: Cleanup find_channel_parking_lot_name() usage." into 13	9 years ago
Richard Mudgett	50f90d4099	res_parking: Fix blind transfer dynamic lots creation. ... Blind transfers to a recognized parking extension need to use the parker's channel variable values to create the dynamic parking lot. This is because there is always only one parker while the parkee may actually be a multi-party bridge. A multi-party bridge can never supply the needed channel variables to create the dynamic parking lot. In the multi-party bridge blind transfer scenario, the parker's CHANNEL(parkinglot) value and channel variables are inherited by the local channel used to park the bridge. * In park_common_setup(), make use the parker instead of the parkee to supply the dynamic parking lot channel variable values. In all but one case, the parkee is the same as the parker. However, in the recognized parking extension blind transfer scenario for a two party bridge they are different channels. For consistency, we need to use the parker channel. * In park_local_transfer(), pass the CHANNEL(parkinglot) value to the local channel when blind transferring a multi-party bridge to a recognized parking extension. * When a local channel starts a call, the Local;2 side needs to inherit the CHANNEL(parkinglot) value from Local;1. The DTMF one-touch parking case wasn't even trying to create dynamic parking lots before it aborted the attempt. * In parking_park_call(), add missing code to create a dynamic parking lot. A DTMF bridge hook is documented as returning -1 to remove the hook. Though the hook caller is really coded to accept non-zero. See the ast_bridge_hook_callback typedef. * In feature_park_call(), don't remove the DTMF one-touch parking hook because of an error. ASTERISK-24605 #close Reported by: Philip Correia Patches: call_park.patch (license #6672) patch uploaded by Philip Correia Change-Id: I221d3a8fcc181877a1158d17004474d35d8016c9	9 years ago
George Joseph	5aa5c49413	sorcery/res_pjsip: Refactor for realtime performance ... There were a number of places in the res_pjsip stack that were getting all endpoints or all aors, and then filtering them locally. A good example is pjsip_options which, on startup, retrieves all endpoints, then the aors for those endpoints, then tests the aors to see if the qualify_frequency is > 0. One issue was that it never did anything with the endpoints other than retrieve the aors so we probably could have skipped a step and just retrieved all aors. But nevermind. This worked reasonably well with local config files but with a realtime backend and thousands of objects, this was a nightmare. The issue really boiled down to the fact that while realtime supports predicates that are passed to the database engine, the non-realtime sorcery backends didn't. They do now. The realtime engines have a scheme for doing simple comparisons. They take in an ast_variable (or list) for matching, and the name of each variable can contain an operator. For instance, a name of "qualify_frequency >" and a value of "0" would create a SQL predicate that looks like "where qualify_frequency > '0'". If there's no operator after the name, the engines add an '=' so a simple name of "qualify_frequency" and a value of "10" would return exact matches. The non-realtime backends decide whether to include an object in a result set by calling ast_sorcery_changeset_create on every object in the internal container. However, ast_sorcery_changeset_create only does exact string matches though so a name of "qualify_frequency >" and a value of "0" returns nothing because the literal "qualify_frequency >" doesn't match any name in the objset set. So, the real task was to create a generic string matcher that can take a left value, operator and a right value and perform the match. To that end, strings.c has a new ast_strings_match(left, operator, right) function. Left and right are the strings to operate on and the operator can be a string containing any of the following: = (or NULL or ""), !=, >, >=, <, <=, like or regex. If the operator is like or regex, the right string should be a %-pattern or a regex expression. If both left and right can be converted to float, then a numeric comparison is performed, otherwise a string comparison is performed. To use this new function on ast_variables, 2 new functions were added to config.c. One that compares 2 ast_variables, and one that compares 2 ast_variable lists. The former is useful when you want to compare 2 ast_variables that happen to be in a list but don't want to traverse the list. The latter will traverse the right list and return true if all the variables in it match the left list. Now, the backends' fields_cmp functions call ast_variable_lists_match instead of ast_sorcery_changeset_create and they can now process the same syntax as the realtime engines. The realtime backend just passes the variable list unaltered to the engine. The only gotcha is that there's no common realtime engine support for regex so that's been noted in the api docs for ast_sorcery_retrieve_by_fields. Only one more change to sorcery was done... A new config flag "allow_unqualified_fetch" was added to reg_sorcery_realtime. "no": ignore fetches if no predicate fields were supplied. "error": same as no but emit an error. (good for testing) "yes": allow (the default); "warn": allow but emit a warning. (good for testing) Now on to res_pjsip... pjsip_options was modified to retrieve aors with qualify_frequency > 0 rather than all endpoints then all aors. Not only was this a big improvement in realtime retrieval but even for config files there's an improvement because we're not going through endpoints anymore. res_pjsip_mwi was modified to retieve only endpoints with something in the mailboxes field instead of all endpoints then testing mailboxes. res_pjsip_registrar_expire was completely refactored. It was retrieving all contacts then setting up scheduler entries to check for expiration. Now, it's a single thread (like keepalive) that periodically retrieves only contacts whose expiration time is < now and deletes them. A new contact_expiration_check_interval was added to global with a default of 30 seconds. Ross Beer reports that with this patch, his Asterisk startup time dropped from around an hour to under 30 seconds. There are still objects that can't be filtered at the database like identifies, transports, and registrations. These are not going to be anywhere near as numerous as endpoints, aors, auths, contacts however. Back to allow_unqualified_fetch. If this is set to yes and you have a very large number of objects in the database, the pjsip CLI commands will attempt to retrive ALL of them if not qualified with a LIKE. Worse, if you type "pjsip show endpoint <tab>" guess what's going to happen? :) Having a cache helps but all the objects will have to be retrieved at least once to fill the cache. Setting allow_unqualified_fetch=no prevents the mass retrieve and should be used on endpoints, auths, aors, and contacts. It should NOT be used for identifies, registrations and transports since these MUST be retrieved in bulk. Example sorcery.conf: [res_pjsip] endpoint=config,pjsip.conf,criteria=type=endpoint endpoint=realtime,ps_endpoints,allow_unqualified_fetch=error ASTERISK-25826 #close Reported-by: Ross Beer Tested-by: Ross Beer Change-Id: Id2691e447db90892890036e663aaf907b2dc1c67	9 years ago
Richard Mudgett	05fc3a96d1	res_parking: Cleanup find_channel_parking_lot_name() usage. ... Change-Id: I8f7a8890aef27824301c642d4d15407ac83e6f02	9 years ago
Richard Mudgett	a4189763ab	res_parking: Misc fixes. ... res/parking/parking_applications.c: * Add malloc fail checks in setup_park_common_datastore(). * Fix playing parking failed announcement to only happen on non-blind transfers in park_app_exec(). It could never go out before because a test was provedly always false. res/parking/parking_bridge.c: * Fix NULL tolerance in generate_parked_user() because bridge_parking_push() can theoretically pass a NULL parker channel if the parker channel went away for some reason. * Clarify some weird code dealing with blind_transfer in bridge_parking_push(). res/parking/parking_bridge_features.c: * Made park_local_transfer() set BLINDTRANSFER on the Local;1 channel which will be bulk copied to the Local;2 channel on the subsequent ast_call(). The additional advantage is if the parker channel has the BLINDTRANSFER and ATTENDEDTRANSFER variables set they are now guaranteed to be overridden. res/parking/parking_manager.c: * Fix AMI Park action input range checking of the Timeout header in manager_park(). * Reduced locking scope to where needed in manager_park(). res/res_parking.c: * Fix some off nominal missing unlocks by eliminating the returns. Change-Id: Ib64945bc285acb05a306dc12e6f16854898915ca	9 years ago
Philip Correia	6f95b5eda1	res_parking: Update parking documentation for dynamic parking lots. ... * Remove duplicate res_parking.conf courtesytone config option documentation. ASTERISK-24596 #close Reported by: Philip Correia ASTERISK-24605 Reported by: Philip Correia Patches: call_park_app_doc.patch (license #6672) patch uploaded by Philip Correia Change-Id: I90a92a891c6494dc08173e675856afcc4764c5b5	9 years ago
zuul	01150597f6	Merge "pjproject-bundled: Cleanups for reported issues" into 13	9 years ago
zuul	f3b154d7d2	Merge "progdocs: Exclude ./third-party from documentation generation" into 13	9 years ago
zuul	096e7a88ce	Merge "core/logging: Fix broken syslog levels on older glibc." into 13	9 years ago
zuul	36d016b1ab	Merge "musiconhold: Only warn if music class is not found in memory and database." into 13	9 years ago
zuul	8271a06dde	Merge "Restrict CLI/AMI commands on shutdown." into 13	9 years ago
Alexander Traud	81ce60f6d4	chan_sip: Do not send all codecs on INVITE. Do not break on Session-Timers. ... Asterisk 13.7.0 included a fix for ASTERISK-24543, not to send all those codecs, which the caller did not request/support. That fix was not complete because on the second Session Timer all codecs were sent again. Some VoIP/SIP clients interpreted that complete codec-list as a change in the SIP session. Because of that, Asterisk did not send the RTP audio via NAT anymore which created a non-audio scenario after the second Session Timer fired. ASTERISK-24543 #close Change-Id: I1881827816ab7fd47eb4287a95961179b34a0b66	9 years ago
zuul	d3af5320d4	Merge "config: fix flags in uint option handler" into 13	9 years ago
Gianluca Merlo	c6e4c48e67	config: fix flags in uint option handler ... The configuration unsigned integer option handler sets flags for the parser as if the option should be a signed integer (PARSE_INT32), leading to errors on "out of range" values. Fix flags (PARSE_UINT32). A fix to res_pjsip is also present which stops invalid flags from being passed when registering sorcery object fields for qualify status. ASTERISK-25612 #close Change-Id: I96b539336275e0e72a8e8033487d2c3344debd3e	9 years ago
Mark Michelson	59c8e189fd	Restrict CLI/AMI commands on shutdown. ... During stress testing, we have frequently seen crashes occur because a CLI or AMI command attempts to access information that is in the process of being destroyed. When addressing how to fix this issue, we initially considered fixing individual crashes we observed. However, the changes required to fix those problems would introduce considerable overhead to the nominal case. This is not reasonable in order to prevent a crash from occurring while Asterisk is already shutting down. Instead, this change makes it so AMI and CLI commands cannot be executed if Asterisk is being shut down. For AMI, this is absolute. For CLI, though, certain commands can be registered so that they may be run during Asterisk shutdown. ASTERISK-25825 #close Change-Id: I8887e215ac352fadf7f4c1e082da9089b1421990	9 years ago
Walter Doekes	ff3eebf454	musiconhold: Only warn if music class is not found in memory and database. ... The log message when a MusicOnHold music class was not found was changed from debug level to WARNING level in Asterisk 11.19 and 13.5. For those using realtime musiconhold, this message is wrong because it warns before checking the database. This changeset delays the warning until after the database has been checked. Reported-by: Conrad de Wet ASTERISK-25444 #close Change-Id: I6cfb2db2f9cfbd2bb3d30566ecae361c4abf6dbf	9 years ago
Walter Doekes	82e55e4883	core/logging: Fix broken syslog levels on older glibc. ... The fix to ASTERISK-25407 introduced the usage of LOG_MAKEPRI. However this macro is broken in older glibc (< 2.17); it would left-shift the facility a second time, causing the resultant priority to become invalid. The syslog manpage mentions nothing about LOG_MAKEPRI and suggests this: The priority argument is formed by ORing the facility and the level values [...]. ASTERISK-25510 #close Reported by: Michael Newton Change-Id: Ia89debe7fac5ad090c7ef595c0707f31bb1e3d03	9 years ago
George Joseph	d963a33749	pjproject-bundled: Cleanups for reported issues ... PortAudio should no longer be required PJSIP_MAX_PKT_LEN is now 6000 Older autoconf issue fixed. (CentOS 6) Change-Id: I463fa9586cbe7c6b3b603289f535bd8e361611dd	9 years ago
Francesco Castellano	c5170677e7	chan_sip.c: Space after port causes unnecessary resolution attempt ... check_via() already skips leading blanks where the sent-by address (with the optional port) should be placed. Since RFC 3261 allows for blanks between the port ant the Via parameters: > https://tools.ietf.org/html/rfc3261#section-20.42 (actually it allows a lot of blanks more ;-)). I just switched from ast_skip_blanks() to ast_strip() on the local copy of the string. ASTERISK-21301 #close Change-Id: Ie5b8fe5a07067b7c0dc9bcdd1707e99b23b02b06	9 years ago
zuul	7d6782cc0f	Merge "func_aes: fix misuse of strlen on binary data" into 13	9 years ago
George Joseph	51deadee38	progdocs: Exclude ./third-party from documentation generation ... We don't need pjproject's documentation embedded in Asterisk's. Change-Id: Iea6f5a621c0f4e3168dda3321eaab258d9f24a17	9 years ago
Gianluca Merlo	aa2fcd244e	func_aes: fix misuse of strlen on binary data ... The encryption code for AES_ENCRYPT evaluates the length of the data to be encoded in base64 using strlen. The data is binary, thus the length of it can be underestimated at the first NULL character. Reuse the write pointer offset to evaluate it, instead. ASTERISK-25857 #close Change-Id: If686b5d570473eb926693c73461177b35b13b186	9 years ago
Kevin Harwell	6a40520fe9	chan_pjsip: ref leak when checking direct_media_glare ... Fix the reference leak introduced in the following commit: 9444ddadf8 ASTERISK-25849 Change-Id: I5cfefd5ee6c1c3a1715c050330aaa10e4d2a5e85	9 years ago
Kevin Harwell	9444ddadf8	chan_pjsip: transfers with direct media reinvite has wrong address/port ... During a transfer involving direct media a race occurs between when the transferer channel is swapped out, initiating rtp changes/updates, and the subsequent reinvites. When Alice, after speaking with Charlie (Bob is on hold), connects Bob and Charlie invites are sent to each in order to establish the call between them. Bob is taken off hold and Charlie is told to have his media flow through Asterisk. However, if before those invites go out the bridge updates Bob's and/or Charlie's rtp information with direct media data (i.e. address, port) then the invite(s) will contain the remote data in the SDP instead of the Asterisk data. The race occurs in the native bridge glue code when updating the peer. The direct_media_address can get set twice before sending out the first invite during call connection. This can happen because the checking/setting of the direct_media_address happened in one thread while the sending of the invite(s) happened in another thread. This fix removes the race condition by moving the checking/setting of the direct_media_address to be in the same thread as the sending of the invites(s). This serializes the checking/setting and sending so they can no longer happen out of order. ASTERISK-25849 #close Change-Id: Idfea590175e74f401929a601dba0c91ca1a7f873	9 years ago
zuul	65ad761130	Merge "install_prereq: Check if is installed aptitude otherwise to install." into 13	9 years ago
zuul	d444a2713f	Merge "install_prereq: Update repositories before install on Debian systems" into 13	9 years ago
Rodrigo Ramírez Norambuena	88240f98d9	install_prereq: Update repositories before install on Debian systems ... When to install packages the indexed local is more old of the version of software on the repository they have been upgraded by security update then get the package will give 404 not found. The patch prevent by update local index to repository for aptitude before install. ASTERISK-25495 #close Reporte by: Rodrigo Ramírez Norambuena Change-Id: I645959e553aac542805ced394cac2dca964051fa (cherry picked from commit 88f3dbaec9)	9 years ago
Rodrigo Ramírez Norambuena	efcf9a96db	install_prereq: Check if is installed aptitude otherwise to install. ... If in Debian or system based, dont have aptitude installed the script do nothing. This patch checked if aptitude installed, if not installed. Also, if execute script with all packages installed yet, the script not show nothing and return exit 1 because the command 'grep' get nothing from pipe from 'awk'. ASTERISK-25113 #close Reported By: Rodrigo Ramírez Norambuena <decipher.hk@gmail.com> Change-Id: Iebdff55805d3917166e5e08e0a1e2176f36ff27f (cherry picked from commit 6737ded058)	9 years ago
Sergio Medina Toledo	2b1b8e382a	res_pjsip_refer.c: Fix seg fault in process of Refer-to header. ... The "Refer-to" header of an incoming REFER request is parsed by pjsip_parse_uri(). That function requires the URI parameter to be NULL terminated. Unfortunately, the previous code added the NULL terminator by overwriting memory that may not be safe. The overwritten memory results could be benign, memory corruption, or a segmentation fault. Now the URI is NULL terminated safely by copying the URI to a new chunk of memory with the correct size to be NULL terminated. ASTERISK-25814 #close Change-Id: I32565496684a5a49c3278fce06474b8c94b37342	9 years ago
zuul	aeef52db9f	Merge "Add initial support to build Docker images" into 13	9 years ago
Richard Mudgett	de04308ae4	chan_sip.c: Fix mwi resub deadlock potential. ... This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 #close Change-Id: I96d429c57a48861fd8bde63dd93db4e92dc3adb6	9 years ago
Richard Mudgett	5f6627a8a4	chan_sip.c: Fix registration timeout and expire deadlock potential. ... This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: I2e40de89efc8ae6e8850771d089ca44bc604b508	9 years ago
Richard Mudgett	32bd7a64f9	chan_sip.c: Fix t38id deadlock potential. ... This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: If595e4456cd059d7171880c7f354e844c21b5f5f	9 years ago
Richard Mudgett	43556b800b	chan_sip.c: Fix reinviteid deadlock potential. ... This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: I9c11b9d597468f63916c99e1dabff9f4a46f84c1	9 years ago
Richard Mudgett	38c1cdab2c	chan_sip.c: Fix packet retransid deadlock potential. ... This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. * Fix retrans_pkt() to call check_pendings() with both the owner channel and the private objects locked as required. * Refactor dialog retransmission packet list to safely remove packet nodes. The list nodes are now ao2 objects. The list has a ref and the scheduled entry has a ref. ASTERISK-25023 Change-Id: I50926d81be53f4cd3d572a3292cd25f563f59641	9 years ago
Richard Mudgett	e4ad55c888	chan_sip.c: Fix waitid deadlock potential. ... This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. * Made always run check_pendings() under the scheduler thread so scheduler ids can be checked safely. ASTERISK-25023 Change-Id: Ia834d6edd5bdb47c163e4ecf884428a4a8b17d52	9 years ago
Richard Mudgett	98d5669c28	chan_sip.c: Fix session timers deadlock potential. ... This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: I6d65269151ba95e0d8fe4e9e611881cde2ab4900	9 years ago
Richard Mudgett	9cb8f73226	chan_sip.c: Fix autokillid deadlock potential. ... This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. * Fix clearing autokillid in __sip_autodestruct() even though we could reschedule. ASTERISK-25023 Change-Id: I450580dbf26e2e3952ee6628c735b001565c368f	9 years ago
Richard Mudgett	c5c7f48a15	chan_sip.c: Fix provisional_keepalive_sched_id deadlock. ... This patch is part of a series to resolve deadlocks in chan_sip.c. Stopping a scheduled event can result in a deadlock if the scheduled event is running when you try to stop the event. If you hold a lock needed by the scheduled event while trying to stop the scheduled event then a deadlock can happen. The general strategy for resolving the deadlock potential is to push the actual starting and stopping of the scheduled events off onto the scheduler/do_monitor() thread by scheduling an immediate one shot scheduled event. Some restructuring may be needed because the code may assume that the start/stop of the scheduled events is immediate. ASTERISK-25023 Change-Id: I98a694fd42bc81436c83aa92de03226e6e4e3f48	9 years ago