summaryrefslogtreecommitdiff
AgeCommit message (Collapse)Author
2014-10-13manager/config: Support templates and non-unique category names via AMIGeorge Joseph
This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-12res_rtp_asterisk: Make the ICE transport check case insensitive as some ↵Joshua Colp
implementations use 'udp'. ........ Merged revisions 425360 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425361 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425362 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-12chan_sip: Fix so asterisk won't send reINVITE after a BYE.Walter Doekes
After a reINVITE glare situation, Asterisk would re-send the reINVITE even though the call had been hung up in the mean time. This patch unschedules the reinvite when handling the BYE. ASTERISK-22791 #close Reported by: Paolo Compagnini Tested by: Paolo Compagnini Review: https://reviewboard.asterisk.org/r/4056/ (testcase is in review r4055) ........ Merged revisions 425296 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 425297 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425298 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425299 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-12build: Relax badshell tilde test to allow for ~ in middle of DESTDIR.Walter Doekes
The main Makefile has a target test called 'badshell' that tests if DESTDIR does not happen to have an an-expanded tilde (~). This might be the case if you run: make install DESTDIR=~/somewhere/ That test also disallowed valid tildes in directory names. The test is now changed to only trigger on a tilde at the start of the path. ASTERISK-13797 #close Reported by: Tzafrir Cohen Review: https://reviewboard.asterisk.org/r/4064/ ........ Merged revisions 425291 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 425292 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425293 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425294 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-12res_calendar_ews: Relax neon version check to work with 0.30 too.Walter Doekes
Allow res_calendar_ews to work not only with libneon-0.29 but also with 0.30. ASTERISK-24325 #close Reported by: Tzafrir Cohen Review: https://reviewboard.asterisk.org/r/4068/ ........ Merged revisions 425286 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 425287 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425288 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425289 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-11res_phoneprov: Cleanup module load error handlingGeorge Joseph
Tested module load/reload interaction between res_phoneprov and res_pjsip_phoneprov_provider in cases where res_phoneprov didn't load correctly (usually misconfiguration or missing phoneprov.conf) Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4069/ ........ Merged revisions 425264 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425265 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-10bridge: During a smart bridge operation provide a more complete bridge to ↵Joshua Colp
the old technology. When a smart bridge operation occurs and a bridge transitions from one technology to another the old technology is provided the channels formerly in it and told that they are leaving. Unfortunately the bridge provided along with them is incomplete. The bridge, despite there being channels in it, contains none. This forces technology implementations to have additional logic when channels are leaving or to store their own duplicated state. This change makes the bridge more complete so it contains the expected channels. Now that the bridge is complete special logic within bridge_native_rtp is no longer needed and has been removed. Review: https://reviewboard.asterisk.org/r/4057/ ........ Merged revisions 425242 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425243 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-10res/res_phoneprov: Bail on registration if res_phoneprov didn't loadMatthew Jordan
If res_phoneprov failed to fully load (due to not being configured), the providers container will be NULL. If a module attempts to register a phone provisioning provider, it should check for the presence of the container. If there is no providers container, it should return an error. This patch makes the ast_phoneprov_provider_register function do that... otherwise this would be a silly commit message. ........ Merged revisions 425220 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-10res_pjsip_phoneprov_provider: Add missing dependency on pjproject.Joshua Colp
........ Merged revisions 425216 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-10CallerID: Fix parsing regressionKinsey Moore
This fixes a regression in callerid parsing introduced when another bug was fixed. This bug occurred when the name was composed entirely of DTMF keys and quoted without a number section (<>). ASTERISK-24406 #close Reported by: Etienne Lessard Tested by: Etienne Lessard Patches: callerid_fix.diff uploaded by Kinsey Moore Review: https://reviewboard.asterisk.org/r/4067/ ........ Merged revisions 425152 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 425153 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425154 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425155 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-10res_pjsip_nat: Place source port into rport of responses if 'force_rport' is on.Joshua Colp
When the 'force_rport' option is enabled the behavior should be the same as if the remote side placed rport into the message themselves. Therefore any responses we send should include the source port of the request in the rport of the Via header. #SIPit31 ASTERISK-24387 #close Reported by: Matt Jordan ........ Merged revisions 425131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425132 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-10chan_sip: Fix dialog leak resulting from missing ACK to re-INVITE.Walter Doekes
If a device re-INVITEs at the same time as the dialog is hung up, and if then the ACK to the re-INVITE never reaches Asterisk, chan_sip would fail to destroy the dialog after a while. This resulted in (most prominently) file handle leaks. (Patch reindented by me.) ASTERISK-20784 #close ASTERISK-15879 #close Reported by: Torrey Searle, Nitesh Bansal Patches: reinvite_ack_timeout.patch uploaded by Torrey Searle (License #5334) patch_asterisk_20784.txt uploaded by Nitesh Bansal (License #6418) Reviewboard: https://reviewboard.asterisk.org/r/4052/ (testcase can be found at r4051) ........ Merged revisions 425068 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 425069 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 425070 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-09res_pjsip_phoneprov_provider: fix compile breakage on AST_VECTORGeorge Joseph
endpoint->inbound_auths was changed to a vector in 13 and I committed the 12 patch instead of the 13 patch. Tested-by: George Joseph git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-09res_rtp_asterisk: Crash if no candidates received for componentKevin Harwell
When starting ice if there is not at least one remote ice candidate with an RTP component asterisk will crash. This is due to an assertion in pjnath as it expects at least one candidate with an RTP component. Added a check to make sure at least one candidate contains an RTP component and at least one candidate has an RTCP component. ASTERISK-24383 #close Review: https://reviewboard.asterisk.org/r/4039/ ........ Merged revisions 425030 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425031 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-09res_pjsip_phoneprov_provider: Provides pjsip integration with res_phoneprovGeorge Joseph
This module allows res_pjsip to integrate with res_phoneprov. It handles the pjsip 'phoneprov' object type. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/3976/ ........ Merged revisions 425007 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-09res/res_phoneprov: Don't cancel Asterisk load on module load failureMatthew Jordan
........ Merged revisions 424985 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424986 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-09res_phoneprov: Refactor phoneprov to allow pluggable config providersGeorge Joseph
This patch makes res_phoneprov more modular so other modules (like pjsip) can provide configuration information instead of res_phoneprov relying solely on users.conf and sip.conf. To accomplish this a new ast_phoneprov public API is now exposed which allows config providers to register themselves, set defaults (server profile, etc) and add user extensions. * ast_phoneprov_provider_register registers the provider and provides callbacks for loading default settings and loading users. * ast_phoneprov_provider_unregister clears the defaults and users. * ast_phoneprov_add_extension should be called once for each user/extension by the provider's load_users callback to add them. * ast_phoneprov_delete_extension deletes one extension. * ast_phoneprov_delete_extensions deletes all extensions for the provider. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/3970/ ........ Merged revisions 424963 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-09cdr.c: Make turning on CDR debug a one step process instead of two.Richard Mudgett
Now "cdr set debug on" doesn't also require "core set verbose 1" to see CDR debug output. ........ Merged revisions 424941 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424942 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-09safe_asterisk: Don't automatically exceed MAXFILES value of 2^20.Walter Doekes
On systems with lots of RAM (e.g. 24GB) /proc/sys/fs/file-max divided by two can exceed the per-process file limit of 2^20. This patch ensures the value is capped. (Patch cleaned up by me.) ASTERISK-24011 #close Reported by: Michael Myles Patches: safe_asterisk-ulimit.diff uploaded by Michael Myles (License #6626) ........ Merged revisions 424875 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 424878 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424879 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424880 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-08res_rtp_asterisk: Allow only UDP ICE candidates.Joshua Colp
The underlying library, pjnath, that res_rtp_asterisk uses for ICE support does not have support for ICE-TCP. As candidates are passed through directly to it this can cause error messages to occur when it receives something unexpected (such as a TCP candidate). This change merely ignores all non-UDP candidates so they never reach pjnath. ASTERISK-24326 #close Reported by: Joshua Colp ........ Merged revisions 424852 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424853 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424854 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-08Stasis: Relegate log message to dev-modeKinsey Moore
This error message primarily applies to development tasks and will now only show up when dev-mode is enabled via configure. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-08Indexer: Format message types may not existKinsey Moore
In Asterisk 13+, any given message type is not guaranteed to exist even if Asterisk comes up correctly since creation of the message type could be declined. The indexer should not prevent Asterisk from starting under these conditions. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424833 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-07Stasis: Only log errors for non-declined typesKinsey Moore
When message type creation is declined via stasis.conf, certain operations log errors assuming that the declined type is being used before initialization or after destruction. These error messages get quite spammy for oft used message types and should not be logged in the first place since the message type is validly NULL. Reported by: Matt DiMeo git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-07data: Properly access formats in capabilities structure when adding codecs.Joshua Colp
Formats within a capabilities structure are addressed starting at 0, not 1. Assuming 1 causes it to exceed an array. ASTERISK-24389 #close Reported by: Kevin Harwell git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424752 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-07res/res_pjsip_outbound_registration: Initialize auth_reject_permanent parameterMatthew Jordan
Prior to this patch, the auth_reject_permanent parameter was not initialized on the registration client state, leading to the parameter being disabled regardless of the value specified in pjsip.conf. This patch initialized the setting on the registration client state to the provided configuration value. ASTERISK-24398 #close ........ Merged revisions 424730 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-07res/res_pjsip_pubsub: Fix typo in WARNING messageMatthew Jordan
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424713 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-06message: Don't close an AMI connection on SendMessage action errorMatthew Jordan
If SendMessage encounters an error (such as incorrect input provided to the action), it will currently return -1. Actions should only return -1 if the connection to the AMI client should be closed. In this case, SendMessage causing the client to disconnect is inappropriate. This patch causes the action to return 0, which simply causes the action to fail. Review: https://reviewboard.asterisk.org/r/4024 ASTERISK-24354 #close Reported by: Peter Katzmann patches: sendMessage.patch uploaded by Peter Katzmann (License 5968) ........ Merged revisions 424690 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424691 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424692 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-06features.c: Fix lingering channel ref while Bridge() application is active.Richard Mudgett
Using the Bridge application to bridge a channel that is executing an applicaiton such as Wait results in a lingering Surrogate channel in the CLI "core show channels" output even though it has already hungup. * Fix bridge_exec() to not hold onto the current_dest_chan ref once it has been put into the bridge. * Eliminated bridge_exec()'s use of RAII_VAR(). ASTERISK-24224 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4041/ ........ Merged revisions 424668 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424669 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-06sdp_srtp: Add new lines to some WARNING messagesMatthew Jordan
........ Merged revisions 424646 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424647 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-06res_pjsip/pjsip_options: Do not 404 an OPTIONS request not sent to an endpointMatthew Jordan
An OPTIONS request that is sent to Asterisk but not to a specific endpoint is currently sent a 404 in response. This is because, not surprisingly, an empty extension is never going to be found in the dialplan. This patch makes it so that we only attempt to look up the endpoint in the dialplan if it is specified in the OPTIONS request URI. #SIPit31 ASTERISK-24370 #close Reported by: Matt Jordan ........ Merged revisions 424624 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424625 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-06pjsip/dialplan_functions: Handle PJSIP_MEDIA_OFFER called on non-PJSIP channelsMatthew Jordan
Calling PJSIP_MEDIA_OFFER on a non-PJSIP channel is hazardous to your health. It will treat the channels as a PJSIP channel, eventually hitting an ao2 error, FRACKing on assertion error, and quite likely crashing. This patch adds checks to the read/write callbacks that ensure that the channel technology is of type 'PJSIP' before attempting to operate on the channel. #SIPit31 ASTERISK-24382 #close Reported by: Matt Jordan ........ Merged revisions 424621 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-06res_pjsip: Prevent crashes when PJPROJECT presents an rdata with no messageMatthew Jordan
When a message that exceeds the PJ_MAX_PKT_SIZE is sent over a reliable transport, it is possible (although it shouldn't occur) for pjproject to pass up an rdata object with a NULL msg in the msg_info. Needless to say, things that attempt to dereference this are in for a rough ride. In particular, this caused crashes in three different locations, all of which are 'low level' enough to intercept an rdata object early in processing: (1) res_pjsip_logger (2) res_hep_pjsip (3) res_pjsip/distributor Anything that can intercept an rdata object before res_pjsip/distributor should be defensive when looking at the received packet. #SIPit31 ASTERISK-24369 #close Reported by: Matt Jordan ........ Merged revisions 424618 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424619 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-06res/res_pjsip_pubsub: Gracefully handle errors when re-creating subscriptionsMatthew Jordan
A subscription that has been persisted can - for various reasons - fail to be re-created on startup. This patch resolves a number of crashes that occurred when a subscription cannot be re-created on several off-nominal paths. #SIPit31 ASTERISK-24368 #close Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424601 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-05Release AMI connections on shutdown.Corey Farrell
ASTERISK-24378 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4037/ ........ Merged revisions 424578 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424579 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-05Blocked revisions 424575Corey Farrell
........ chan_sip: Clean leak on error path of process_sdp Resolve leak in process_sdp that occurs in 2 error path's where crypto lines are expected but not provided. ASTERISK-24385 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4045/ ........ Merged revisions 424569 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424577 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-05chan_motif: Correct last commit to use ao2_cleanup to free format capCorey Farrell
This fix applies to 13 and trunk. ASTERISK-24384 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4043/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-05chan_motif: Release format capabilities and config on module load errorCorey Farrell
ASTERISK-24384 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4043/ ........ Merged revisions 424550 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 424551 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424552 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-03res_pjsip: Fix XML typo and update CHANGES.Richard Mudgett
ASTERISK-24199 ........ Merged revisions 424528 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424529 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-03audiohooks: Reevaluate the bridge technology when an audiohook is added or ↵Richard Mudgett
removed. Adding a mixmonitor to a channel causes the bridge to change technologies from native to simple_bridge so the call can be recorded. However, when the mixmonitor is stopped the bridge does not switch back to the native technology. * Added unbridge requests to reevaluate the bridge when a channel audiohook is removed. * Moved the unbridge request into ast_audiohook_attach() ensure that the bridge reevaluates whenever an audiohook is attached. This simplified the mixmonitor and chan_spy start code as well. * Added defensive code to stop_mixmonitor_full() in case additional arguments are ever added to the StopMixMonitor application. * Made ast_framehook_detach() not do an unbridge request if the framehook does not exist. * Made ast_framehook_list_fixup() do an unbridge request if there are any framehooks. Also simplified the loop. ASTERISK-24195 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4046/ ........ Merged revisions 424506 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424507 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-03chan_pjsip: Fix deadlock when masquerading PJSIP channels.Richard Mudgett
Performing a directed call pickup resulted in a deadlock when PJSIP channels were involved. A masquerade needs to hold onto the channel locks while it swaps channel information between the two channels involved in the masquerade. With PJSIP channels, the fixup routine needed to push a fixup task onto the PJSIP channel's serializer. Unfortunately, if the serializer was also processing a task that needed to lock the channel, you get deadlock. * Added a new control frame that is used to notify the channels that a masquerade is about to start and when it has completed. * Added the ability to query taskprocessors if the current thread is the taskprocessor thread. * Added the ability to suspend/unsuspend the PJSIP serializer thread so a masquerade could fixup the PJSIP channel without using the serializer. ASTERISK-24356 #close Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/4034/ ........ Merged revisions 424471 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424472 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-03sorcery: Prevent SEGV in sorcery_wizard_create when there's no create functionGeorge Joseph
When you call ast_sorcery_create() you don't necessarily know which wizard is going to be invoked. If it happens to be a wizard like 'config' that doesn't have a 'create' virtual function you get a segfault in the sorcery_wizard_create callback. This patch catches the null function pointer, does an ast_assert, and logs an error. Review: https://reviewboard.asterisk.org/r/4044/ ........ Merged revisions 424447 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424448 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-03PJSIP: Restore functional default for callerid_privacyKinsey Moore
The pjsip config option default fixups from r424263 altered the functional default from "allowed_not_screened" to "allowed". This change restores the functional default value when none is provided. ........ Merged revisions 424426 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-03Manager: Add missing fields and documentation for CoreShowChannelsKinsey Moore
This corrects some issues introduced in the responses to the CoreShowChannels AMI command as well as adding documentation for the responses. The command in Asterisk 12 was missing the following fields: Duration, Application, ApplicationData, and BridgedChannel and BridgedUniqueID (replaced with BridgeId). ASTERISK-24262 #close Reported by: Mitch Claborn Review: https://reviewboard.asterisk.org/r/4040/ ........ Merged revisions 424423 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-03res_pjsip_session: Reduce SDP size by removing duplicate connection lines.Joshua Colp
Due to the architecture of how media streams are handled each individual handler adds connection details (IP address) for it. The first media stream is then used as the top level SDP connection line. In practice each line ends up being the same so to reduce the SDP size stream-level connection information is also added to the SDP if it differs from the top level SDP connection line. ........ Merged revisions 424414 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-02res_pjsip: Make transport cipher option accept a comma separated list of ↵Richard Mudgett
cipher names. Improvements to the res_pjsip transport cipher option. * Made the cipher option accept a comma separated list of OpenSSL cipher names. Users of realtime will be glad if they have more than one name to list. * Added the CLI command 'pjsip list ciphers' so a user can know what OpenSSL names are available for the cipher option. * Updated the cipher option online XML documentation to specify what is expected for the value. * Updated pjsip.conf.sample to not indicate that ALL is acceptable since ALL does not imply a preference order for the ciphers and PJSIP does not simply pass the string to OpenSSL for interpretation. ASTERISK-24199 #close Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/4018/ ........ Merged revisions 424393 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424394 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-02Alembic: Add enumerator value to sippeers -> directmedia - 'outgoing'Jonathan Rose
The 'outgoing' value was left off of the enumerator when first creating the column. This patch adds it, and should gracefully upgrade keeping the existing data in tact. ASTERISK-23781 #close Reported by: Stephen More Review: https://reviewboard.asterisk.org/r/4013/ ........ Merged revisions 424372 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424373 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-02res_pjsip: document use of rewrite_contact in sample confScott Griepentrog
Without setting rewrite_contact, an invite to an endpoint behind NAT will not reach it - unless the endpoint itself uses STUN or TURN to discover it's public URI. Thus, the use of this should be in the sample documentation. Review: https://reviewboard.asterisk.org/r/4036/ ........ Merged revisions 424337 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-01chan_pjsip: Fix an assertion for channels that lack formats on creationJonathan Rose
ASTERISK-24222 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/4017/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-01res_hep: Release allocation reference to configuration.Corey Farrell
ASTERISK-24362 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4026/ ........ Merged revisions 424312 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-01res_pjsip: Add 'dtls_fingerprint' option to configure DTLS fingerprint hash.Joshua Colp
During the latest update to DTLS-SRTP support the ability to configure the hash used for fingerprints was added. This gave us two supported ones: SHA-1 and SHA-256. The default was accordingly updated to SHA-256. Unfortunately this configuration ability was not exposed within res_pjsip. This change adds a dtls_fingerprint option that controls it. #SIPit31 ........ Merged revisions 424290 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-07 22:15:33 +0000