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2015-11-12Merge "res_pjsip: Deny requests when threadpool queue is backed up."Joshua Colp
2015-11-12Merge "format_cap: Don't append the 'none' format when appending all."Matt Jordan
2015-11-12res_pjsip: Deny requests when threadpool queue is backed up.Mark Michelson
We have observed situations where the SIP threadpool may become deadlocked. However, because incoming traffic is still arriving, the SIP threadpool's queue can continue to grow, eventually running the system out of memory. This change makes it so that incoming traffic gets rejected with a 503 response if the queue is backed up too much. Change-Id: I4e736d48a2ba79fd1f8056c0dcd330e38e6a3816
2015-11-12Merge "Further fixes to improper usage of scheduler"Joshua Colp
2015-11-12format_cap: Don't append the 'none' format when appending all.Joshua Colp
When appending all formats of a type all the codecs are iterated and added. This operation was incorrectly adding the ast_format_none format which is special in that it is supposed to be used when no format is present. It shouldn't be appended. ASTERISK-25535 Change-Id: I7b00f3bdf4a5f3022e483d6ece602b1e8b12827c
2015-11-12Further fixes to improper usage of schedulerSteve Davies
When ASTERISK-25449 was closed, a number of scheduler issues mentioned in the comments were missed. These have since beed raised in ASTERISK-25476 and elsewhere. This patch attempts to collect all of the scheduler issues discovered so far and address them sensibly. ASTERISK-25476 #close Change-Id: I87a77d581e2e0d91d33b4b2fbff80f64a566d05b
2015-11-11threadpool: Handle worker thread transitioning to dead when going active.Joshua Colp
This change adds handling of dead worker threads when moving them to be active. When this happens the worker thread is removed from both the active and idle threads container. If no threads are able to be moved to active then the pool grows as configured. A unit test has also been added which thrashes the idle timeout and thread activation to exploit any race conditions between the two. ASTERISK-25546 #close Change-Id: I6c455f9a40de60d9e86458d447b548fb52ba1143
2015-11-11Merge "rtp_engine: Init a format-attribute module to its RFC defaults."Matt Jordan
2015-11-11Merge "Increase account code maximum length to 80."Matt Jordan
2015-11-11Merge "dns: Use ntohl for ans->ttl in dns_parse_answer_ex"Matt Jordan
2015-11-11Merge "res_pjsip_sdp_rtp: Enable Opus to be negotiated via SIP/SDP."Matt Jordan
2015-11-11Merge "ast_format_cap: Avoid format creation on module load, use cache instead."Matt Jordan
2015-11-11Merge "xmldoc: Improve xmldoc wrapping of 'core show ...' output."Matt Jordan
2015-11-11rtp_engine: Init a format-attribute module to its RFC defaults.Alexander Traud
Previously, format-attribute modules relied on an existing fmtp line in SDP negotiation. However, fmtp is optional for several formats like the Opus Codec. Now, the format-attribute module is called with an empty fmtp, which allows the module to initialise itself to RFC defaults. Furthermore now, Asterisk is able to differentiate between internally and externally created formats. ASTERISK-25537 #close Change-Id: I28f680cef7fdf51c0969ff8da71548edad72ec52
2015-11-11Merge "taskprocessor: Add high water mark warnings"Joshua Colp
2015-11-10Merge "Remove ABI compatibility stub functions."Joshua Colp
2015-11-10Merge "ast_format_cap_get_names: To display all formats, the buffer was ↵Joshua Colp
increased."
2015-11-10Remove ABI compatibility stub functions.Corey Farrell
ABI compatibility stubs existed for ast_app_separate_args and ast_verbose, this is not needed in master. Change-Id: I07b4d2c16079da3c2c6efa55df4a74368e0bd453
2015-11-10Remove execute permission from dns_system_resolver.cCorey Farrell
Change-Id: I3185735db42064bab00d3e073aed703385a00bf4
2015-11-10Merge "func_callerid: Document that CALLERID(pres) is available."Joshua Colp
2015-11-09ast_format_cap_get_names: To display all formats, the buffer was increased.Alexander Traud
ASTERISK-25533 #close Change-Id: Ie1a9d1a6511b3f1a56b93d04475fbf8a4e40010a
2015-11-09ast_format_cap: Avoid format creation on module load, use cache instead.Alexander Traud
Since Asterisk 13, formats are immutable and cached. However while loading a module like chan_sip, some formats were created instead using cached ones. ASTERISK-25535 #close Change-Id: I479cdc220d5617c840a98f3389b3bd91e91fbd9b
2015-11-06func_callerid: Document that CALLERID(pres) is available.Walter Doekes
CALLERPRES() says that it's deprecated in favor of CALLERID(num-pres) and CALLERID(name-pres). But for channel driver that don't make a distinction between the two (e.g. SIP), it makes more sense to get/set both at once. This change reveals the availability of CALLERID(pres), CONNECTEDLINE(pres), REDIRECTING(orig-pres), REDIRECTING(to-pres) and REDIRECTING(from-pres). ASTERISK-25373 #close Change-Id: I5614ae4ab7d3bbe9c791c1adf147e10de8698d7a
2015-11-06docs: Fix a few typo's in app docs (more then, resourse).Walter Doekes
Change-Id: Iba57efadf6c0b822e762c7a001bc89611d98afd7
2015-11-06dns: Use ntohl for ans->ttl in dns_parse_answer_exGeorge Joseph
dns_parse_answer_ex was not converting ans->ttl from network by order to host byte order which was causing certain ttls it to go negative. In turn this was causing answer edit checks to fail. ASTERISK-25528 #close Reported-by: Daniel Tryba Tested-by: George Joseph Change-Id: I31505132d6321c46d2f39fd06c20ee808a864037
2015-11-06xmldoc: Improve xmldoc wrapping of 'core show ...' output.Walter Doekes
Previously, the wrapping did both lookahead and lookback, which, together with color escape sequences, caused some lines to be wrapped way earlier than other lines. This led to inconsistent output. This simplifies the wrapping code and makes it more sane: if maxcolumns is hit, we simply jump back to the last space and wrap there. ASTERISK-25527 #close Change-Id: I56d01c6f9a812642b1b05535c98d4db48d17c957
2015-11-06res_pjsip_sdp_rtp: Enable Opus to be negotiated via SIP/SDP.Alexander Traud
In SIP/SDP, Opus has two channels always (see RFC 7587 section 7). The actual amount of channels is negotiated in-band. Therefore now, the Opus codec and its attribute rtpmap are registered with two channels. ASTERISK-24779 #close Reported by: PowerPBX Tested by: Alexander Traud patches: asterisk-24779.patch submitted by Sean Bright (license #5060) Change-Id: Ic7ac13cafa1d3450b4fa4987350924b42cbb657b
2015-11-05taskprocessor: Add high water mark warningsJonathan Rose
If a taskprocessor's queue grows large, this can indicate that there may be a problem with tasks not leaving the processor or else that the number of available task processors for a given type of task is too low. This patch makes it so that if a taskprocessor's task queue grows above 100 queued tasks that it will emit a warning message. Warning messages are emitted only once per task processor. ASTERISK-25518 #close Reported by: Jonathan Rose Change-Id: Ib1607c35d18c1d6a0575b3f0e3ff5d932fd6600c
2015-11-05Increase account code maximum length to 80.Corey Farrell
This increases the maximum length of account code's to match extensions. This ensures it is always possible to set an accountcode to ${EXTEN} without truncation. ASTERISK-23904 Reported by: Ben Merrills Change-Id: If122602304ce03362722eb213a3111b32da5eeb9
2015-11-04Merge "StatsD: Add res_statsd compatibility"Joshua Colp
2015-11-04StatsD: Add res_statsd compatibilitytcambron
Added a new api to res_statsd.c to allow it to receive a character pointer for the value argument. This allows for a '+' and a '-' to easily be sent with the value. ASTERISK-25419 Reported By: Ashley Sanders Change-Id: Id6bb53600943d27347d2bcae26c0bd5643567611
2015-11-04main/dial: Protect access to the format_cap structure of the requesting channelMatt Jordan
When a dial attempt is made that involves a requesting channel, we previously were not: a) Protecting access to the native format capabilities structure on the requesting channel. That is inherently unsafe. b) Reference bumping the lifetime of the format capabilities structure. In both cases, something else could sneak in, blow away the format capabilities, and we'd be holding onto an invalid format_cap structure. When the newly created channel attempts to construct its format capabilities, things go poorly. This patch: a) Ensures that we get a reference to the native format capabilities while the requesting channel is locked b) Holds a reference to the native format capabilities during the creation of the new channel. ASTERISK-25522 #close Change-Id: I0bfb7ba8b9711f4158cbeaae96edf9626e88a54f
2015-11-04Fix cli display of build options.Corey Farrell
A previous commit reduced the AST_BUILDOPTS compiler define to only include options that affected ABI. This included some options that were previously displayed by cli "core show settings". This change corrects the CLI display while still restricting buildopts.h to ABI effecting options only. ASTERISK-25434 #close Reported by: Rusty Newton Change-Id: Id07af6bedd1d7d325878023e403fbd9d3607e325
2015-11-04Merge "res_pjsip/location: Destroy contact_status objects on contact deletion"Matt Jordan
2015-11-04Merge "pjsip_configuration: On delete, remove the persistent version of an ↵Matt Jordan
endpoint"
2015-11-03pjsip_configuration: On delete, remove the persistent version of an endpointMatt Jordan
When an endpoint is deleted (such as through an API), the persistent endpoint currently continues to lurk around. While this isn't harmful from a memory consumption perspective - as all persistent endpoints are reclaimed on shutdown - it does cause Stasis endpoint related operations to continue to believe that the endpoint may or may not exist. This patch causes the persistent endpoint related to a PJSIP endpoint to be destroyed if the PJSIP endpoint is deleted. Change-Id: I85ac707b4d5e6aad882ac275b0c2e2154affa5bb
2015-11-03res_pjsip/location: Destroy contact_status objects on contact deletionMatt Jordan
The contact_status Sorcery objects are currently not destroyed when a contact is deleted. This causes the contact's last known RTT/status to be 'sticky' when the contact itself may no longer exist. This patch causes the contact_status objects associated with both dynamic and static contacts to be destroyed if the AoR holding those contacts is also destroyed (or via other paths where a contact may be deleted.) Change-Id: I7feec8b9278cac3c5263a4c0483f4a0f3b62426e
2015-11-03main/stasis_endpoints: Fix ContactStatusChange JSON for roundtrip_usec fieldMatt Jordan
The JSON packing for the ContactStatusChange event forgot to include the roundtrip_usec field. As a result, the field never showed up in any event, even when the data was available. This patch corrects that error by properly packing the JSON blob with the data. Change-Id: I8df80da659a44010afbd48f645967518ff5daa17
2015-11-03chan_sip: Allow websockets to be disabled.Corey Farrell
This patch adds a new setting "websockets_enabled" to sip.conf. Setting this to false allows chan_sip to be used without causing conflicts with res_pjsip_transport_websocket. ASTERISK-24106 #close Reported by: Andrew Nagy Change-Id: I04fe8c4f2d57b2d7375e0e25826c91a72e93bea7
2015-11-02res_pjsip: Set threadpool max size default to 50.Mark Michelson
During a stress test of subscriptions, a huge blast of subscription-related traffic resulted in the threadpool expanding to a ridiculous number of threads. The balooning of threads resulted in an increase of memory, which led to a crash due to being out of memory. An easy fix for the particular test was to limit the size of the threadpool, thus reining in the amount of memory that would be used. It was decided that there really is no downside to having a non-infinite default value for the maximum size of the threadpool, so this change introduces 50 threads as the maximum threadpool size for the SIP threadpool. ASTERISK-25513 #close Reported by John Bigelow Change-Id: If0b9514f1d9b172540ce1a6e2f2ffa1f2b6119be
2015-11-02Merge "pjsip_options: Schedule/unschedule qualifies on AoR creation/destruction"Joshua Colp
2015-11-02Merge "StatsD: Send stuff to the StatsD server and test"Joshua Colp
2015-11-02StatsD: Send stuff to the StatsD server and testtcambron
Added code to allow the StatsD dialplan application to send data to the server specified in statsd.conf. ASTERISK-25419 Change-Id: I400db2f37c6ddf61515ff5a019646e36dcd0f922
2015-11-02pjsip_options: Schedule/unschedule qualifies on AoR creation/destructionMatt Jordan
When an AoR is created or destroyed dynamically, the scheduled OPTIONS requests that qualify the contacts on the AoR are not necessarily started or destroyed, particularly for persistent contacts created for that AoR. This patch adds create/update/delete sorcery observers for an AoR, which schedule/unschedule the qualifies as expected. Change-Id: Ic287ed2e2952a7808ee068776fe966f9554bdf7d
2015-11-02Merge "app_queue: Added reason pause of member"Matt Jordan
2015-10-31Makefile: Add a rule 'basic-pbx' that installs the Basic PBX configsMatt Jordan
This patch adds a rule for installing the Super Awesome Company based 'Basic PBX' configuration files. As part of adding this rule, a bit of the content that makes up installing the configuration files under the 'samples' target was refactored into a make subroutine for usage by additional later config make targets. Change-Id: I6c2e27906f73e2919a2b691da0be20ae70302404
2015-10-29res_pjsip_pubsub: Fix assertion when UAS dialog creation fails.Joshua Colp
When compiled with assertions enabled one will occur when destroying the subscription tree when UAS dialog creation fails. This is because the code assumes that a dialog will always exist on a subscription tree when in reality during this specific scenario it won't. This change makes it so a dialog is not removed from the subscription tree if it is not present. ASTERISK-25505 #close Change-Id: Id5c182b055aacc5e66c80546c64804ce19218dee
2015-10-29Merge "chan_sip: Do not send all codecs on INVITE."Matt Jordan
2015-10-28Merge "StatsD: Add user input validation to the application"Joshua Colp
2015-10-28StatsD: Add user input validation to the applicationtcambron
Added code to accept user input and validate it before allowing it to be sent to the StatsD server. ASTERISK-25419 Reported By: Ashley Sanders Change-Id: I55c7ce44326a68ad6c5c1514b9575ac50f25bbc3
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