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2018-02-05Merge "res_pjsip/config_domain_aliases.c: Add check for missing domain."Jenkins2
2018-02-05app_confbridge: ConfbridgeList event has standard channel shapshot headers.Richard Mudgett
* Made the AMI ConfbridgeList action's ConfbridgeList events output all the standard channel snapshot headers instead of a few hand-coded channel snapshot headers. The benefit is that the CallerIDName gets disruptive characters like CR, LF, Tab, and a few others escaped. However, an empty CallerIDName is now output as "<unknown>" instead of "<no name>". ASTERISK-27651 Change-Id: Iaf7d54a9d40194c2db060bc9b4979fab6720d977
2018-02-05app_confbridge: Add the Muted header to ConfbridgeJoin AMI event.Richard Mudgett
ASTERISK-27651 Change-Id: Idef2ca54d242d1b894efd3fc7b360bc6fd5bdc34
2018-02-05Merge "res_sorcery_realtime.c: Fix ref leak if object failed to apply."Jenkins2
2018-02-05Merge "manager.c: Fixed "(null):" header in AMI AsyncAGIEnd event"Jenkins2
2018-02-03Merge "res_pjsip.c: Fix documentation typos."Jenkins2
2018-02-03Merge "manager_channels.c: Reordered ↵Jenkins2
ast_manager_build_channel_state_string_prefix()"
2018-02-03Merge "res_pjsip_mwi.c: Fix null pointer crash"Jenkins2
2018-02-03Merge "manager.c: Fix potential memory leak and corruption."Jenkins2
2018-02-03chan_console: don't read and write at the same timeOron Peled
It seems that the ALSA backend of PortAudio doesn't know how to both read and write at the same time by adding a per-device mutex. FIXME: currently only a draft version. Need to either auto-detect we work with the ALSA backend or add an extra configuration option to use this mutex. ASTERISK-27426 #close Change-Id: I635eacee45f5413faa18f5a3b606af03b926dacb
2018-02-02endpoint identifiers: Some code cleanup.Richard Mudgett
res_pjsip_endpoint_identifier_user.c: * Fix copy/paste error in find_endpoint(). We were using a constant "anonymous" string instead of the passed in endpoint_name when checking the transport domain for an endpoint match. * Eliminate RAII_VAR in find_endpoint(). * Remove always true check in find_transport_state_in_use(). * Remove useless CMD_STOP in find_transport_state_in_use(). res_pjsip_endpoint_identifier_anonymous.c: * Eliminate RAII_VAR in anonymous_identify(). * Remove always true check in find_transport_state_in_use(). * Remove useless CMD_STOP in find_transport_state_in_use(). Change-Id: I86924c31db5bd225ca0c1219c761b668c6f91189
2018-02-02res_pjsip/config_domain_aliases.c: Add check for missing domain.Richard Mudgett
What is the point of defining an alias and not saying what is being aliased? Change-Id: I98a892016ed61dcf5efeb6619fd748925103f0be
2018-02-02res_pjsip.c: Fix documentation typos.Richard Mudgett
Change-Id: I82ae0b92bfa2ece84a5c684efd9eefdc83ebd068
2018-02-02res_sorcery_realtime.c: Fix ref leak if object failed to apply.Richard Mudgett
Change-Id: I3c7106ff77009754725cee790eadf5da44154ab6
2018-02-02Merge "appdocsxml.xslt: Add Language to channel snapshot transformation"Richard Mudgett
2018-02-02Merge "bridge_softmix.c: Report not talking immediately when muted."Joshua Colp
2018-02-01manager.c: Fixed "(null):" header in AMI AsyncAGIEnd eventSungtae Kim
* Changed to create ami_event string only when the given blob is not json_null(). * Fixed bad expression. ASTERISK-27621 Change-Id: Ice58c16361f9d9e8648261c9ed5d6c8245fb0d8f
2018-02-01res_pjsip_mwi.c: Fix null pointer crashJoshua Elson
ASTERISK-27652 #close Change-Id: I78a0d38bfd8d0d82830f3d53da04872d6b67284d
2018-02-01appdocsxml.xslt: Add Language to channel snapshot transformationSean Bright
Change-Id: I8f494b0c895a69b8bc94656d0c6ceebecb0394d8
2018-02-01manager.c: Fix potential memory leak and corruption.Richard Mudgett
ast_str_append_event_header() could potentially leak and corrupt memory if the ast_str needed to expand to add the AMI event header. * Fixed to return error if the ast_str_append() failed. Change-Id: I92f36b855540743b208d76e274152ee2d758176d
2018-02-01manager_channels.c: Reordered ast_manager_build_channel_state_string_prefix()Richard Mudgett
* Made not allocate memory if the channel snapshot is an internal channel. * Free memory earlier when no longer needed. Change-Id: Ia06e0c065f1bd095781aa3f4a626d58fa4d28b38
2018-02-01Merge "app_confbridge: Update dsp_silence_threshold and ↵Jenkins2
dsp_talking_threshold docs."
2018-02-01Merge "res_pjsip_pubsub: Prune subs with reliable transports at startup"George Joseph
2018-02-01Merge "res_pjsip_registrar_expire: Delete empty module."Jenkins2
2018-02-01Merge "BuildSystem: Raise autoconf version requirement to 2.60a."Jenkins2
2018-01-31Merge "res_pjsip_session: Prevent crash during shutdown."Jenkins2
2018-01-31Merge "core: Create ast_atomic macro's."Jenkins2
2018-01-31Merge "app_voicemail: Avoid always true when using pointer address."Jenkins2
2018-01-31res_pjsip_registrar_expire: Delete empty module.Corey Farrell
Verified nothing in the testsuite lists this module as a dependency. Change-Id: I90c7d52c7e15e85fde3389d5eaccb05b97848813
2018-01-31bridge_softmix.c: Report not talking immediately when muted.Richard Mudgett
Currently in app_confbridge if someone mutes a channel while that channel is talking, the talk detection code is suspended while the channel is muted. As far an an external observer is concerned, the muted channel's talk status is still "talking" even though the channel is not contributing audio to the conference bridge. When the channel is later unmuted, it takes the usual 'dsp_silence_threshold' option time to clear the talking status even though the channel may have stopped talking while the channel was muted. * In bridge_softmix.c, clear the talking status and report talking stopped if the channel was talking when the channel is muted. When the channel is unmuted and the channel is still talking then report the channel as talking since it is contributing audio to the bridge again. ASTERISK-27647 Change-Id: Ie4fdbc05a0bc7343c2972bab012e2567917b3d4e
2018-01-31app_confbridge: Update dsp_silence_threshold and dsp_talking_threshold docs.Richard Mudgett
The dsp_talking_threshold does not represent time in milliseconds. It represents the average magnitude per sample in the audio packets. This is what the DSP uses to determine if a packet is silence or talking/noise. Change-Id: If6f939c100eb92a5ac6c21236559018eeaf58443
2018-01-31res_pjsip_registrar.c: Fix compiler error.Richard Mudgett
Need to include signal.h to define pthread_kill() and SIGURG. Change-Id: I10ae3aa4bf8e7386ac29ade78c0f2caed8e674fa
2018-01-31Merge "loader: Use ast_cli_completion_add for 'module load' completion."Jenkins2
2018-01-31Merge "res_pjsip_registrar_expire: Refactor into res_pjsip_register"Jenkins2
2018-01-31Merge "pbx_variables.c: Misc fixes in variable substitution."Jenkins2
2018-01-31Merge "install_prereq: Update RHEL/CentOS/Fedora libraries."Jenkins2
2018-01-30res_pjsip_session: Prevent crash during shutdown.Corey Farrell
pjproject does not have a function to reverse pjsip_inv_usage_init. This means we need to ignore any calls to the functions once shutdown is final. ASTERISK-27571 #close Change-Id: Ia550fcba563e2328f03162d79fb185f16b7c9b9d
2018-01-30core: Create ast_atomic macro's.Corey Farrell
Create ast_atomic macro's to provide a consistent interface to the common functionality of __atomic and __sync built-in functions. ASTERISK-27619 Change-Id: Ieba3f81832a0e25c5725ea067e5d6f742d33eb5b
2018-01-30res_pjsip_pubsub: Prune subs with reliable transports at startupGeorge Joseph
In an earlier release, inbound registrations on a reliable transport were pruned on Asterisk restart since the TCP connection would have been torn down and become unusable when Asterisk stopped. This same process is now also applied to inbound subscriptions. Also fixed issues in res_pjsip_registrar where it wasn't handling the monitoring correctly when multiple registrations came in over the same transport. To accomplish this, the pjsip_transport_event feature needed to be refactored to allow multiple monitors (multiple subcriptions or registrations from the same endpoint) to exist on the same transport. Since this changed the API, any external modules that may have used the transport monitor feature (highly unlikey) will need to be changed. ASTERISK-27612 Reported by: Ross Beer Change-Id: Iee87cf4eb9b7b2b93d5739a72af52d6ca8fbbe36
2018-01-30Merge "Build System: Require __sync or __atomic functions."Jenkins2
2018-01-30Merge "Sample modules.conf: comment out example load statement."Jenkins2
2018-01-30Merge "Build System: Add support for __atomic built-in operators."Jenkins2
2018-01-29res_pjsip_registrar_expire: Refactor into res_pjsip_registerGeorge Joseph
res_pjsip_registrar_expire remains as an empty module for now. Change-Id: Ib93698938bae548d2199cb542f3692d1a171239f
2018-01-29Sample modules.conf: comment out example load statement.Corey Farrell
The sample modules.conf explicitly loaded res_musiconhold.so. This is redundent as autoload=yes is already set. It causes warnings if res_musiconhold.so was not installed and results in an unexpected load if the admin disables autoload without remembering to remove the res_musiconhold load statement. Also remove reference to unknown module pbx_gtkconsole. Change-Id: Ib01888994d9f1364b14d3c9fb6ff96774a6e580a
2018-01-29Merge "Remove redundant module checks and references."Jenkins2
2018-01-29Merge "Update sounds release to fix siren7 and siren14 files."Jenkins2
2018-01-29Merge "core: Fix unused variable error in handle_show_sysinfo."Jenkins2
2018-01-29Merge "core: Tweak startup order."Jenkins2
2018-01-29Merge "editline: Avoid shifting a negative signed value."Jenkins2
2018-01-29BuildSystem: Enable autotools in FreeBSD.Alexander Traud
In the current versions of FreeBSD, the apps of GNU autotools do not need to be called with a version anymore. The latest version can be invoked directly. Additionally, the script ./bootstrap.sh asked for autoconf 2.62 and automake 1.9, versions which are not available as port anymore. ASTERISK-27637 Change-Id: Id7b94b80e78cc943a40ba79b697e3f70019820a7
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