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Author	SHA1	Message	Date
George Joseph	43c4529f15	config: bug: fix truncation of included config files on permissions error ... ast_config_text_file_save() currently truncates include files as they are processed. If a subsequent include file or the main config file has a permissions error that prevents writing, earlier include files are left truncated resulting in a frantic search for backups. This patch causes ast_config_text_file_save to check for write access on all files before it truncates any of them. Will be applied 1.8 > trunk. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3986/ ........ Merged revisions 422900 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 422903 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422904 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422905 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Sean Bright	8f15395e09	pjsip/config_auth.c: Add missing whitespace to log messages. ... The errors generated when validating 'auth' settings are missing a space which makes the messages a little confusing. ........ Merged revisions 422899 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422901 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Rusty Newton	a8f290e208	Sounds/BuildSystem: Modifications to include new releases and Japanese language. ... Modifying Makefile and sounds.xml to include new core 1.4.26 and extra 1.4.15 sound prompt releases, plus the new Japanese core sound prompts contributed by QLOOG. ASTERISK-23324 Reported by: Kevin McCoy Tested by: Rusty Newton ........ Merged revisions 422789 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 422790 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422791 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422883 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	048560f330	Add note about configuring list_items on a single line. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422855 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	a8ef508f5e	Add sample configuration for resource lists. ... On review /r/3977, it was recommended to note in the sample configuration about the size limitation for resource lists. However, since there was no section in the sample configuration at all for resource list subscriptions, I decided to make a separate commit where I have added the necessary sample configuration as well as the size limitation warning. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422853 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	9a5ee513d5	Pre-allocate transmission data buffer for RLS NOTIFY requests. ... PJSIP, unless a constant is modified at compilation time, limits SIP requests to 4000 bytes. Full-state RLS notifications can easily exceed this limit with moderately small lists. This changeset allows for Asterisk to work around this size limit by performing its own allocation of the transmission data buffer. This way, Asterisk can allocate a buffer that exceeds the built-in maximum. We still impose our own limit of 64000 bytes, mainly because making allocations larger than that is a bit absurd. ASTERISK-24181 #close Reported by Mark Michelson Review: https://reviewboard.asterisk.org/r/3977 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422851 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	346877f6c9	res_pjsip_pubsub: Check supported headers for eventlist when subscribing to ... resource list https://wiki.asterisk.org/wiki/display/AST/Resource+List+Subscription+Test+Plan According to the off-nominal plan, if evenlist support is not specified in a SUBSCRIBE's supported header(s), that subscription should be rejected with an error. ASTERISK-23871 Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3960/diff/#index_header git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422836 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	128d187f38	main/cdr: Copy over location information during a fork ... When a CDR is forked, a new CDR is created and appended to the CDR chain for the Party A. The forked CDR starts life off as a clone of the last non-finalized for the particular Party A. In the past, merely copying over the snapshots for Party A/Party B would be sufficient. However, as the CDRs now contain cached information from Party A - specifically application/data, context, and extension - we need to copy that over during a fork as well. Huzzah for unit tests catching this when the context/extension were derived from a cached value on the CDR instead of on Party A. ........ Merged revisions 422769 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	8302bc7f0a	main/rtp_engine: Format NTP timestamps as unsigned ints ... On some systems, a timeval's tv_sec/tv_usec will be unsigned lont ints, as opposed to long ints. When the RTP engine formats these as strings, it was previously formatting them as signed integers, which can result in some odd negative timestamp values (particularly on 32-bit systems). This patch formats the values as unsigned long integers. ........ Merged revisions 422766 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422767 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	df77a7c5f0	res_pjsip_sdp_rtp: Fix retrieval of "ice-pwd" attribute if in session and not media stream. ... ........ Merged revisions 422746 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422747 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	0fbd9947e2	main/cdrs: Preserve context/extension when executing a Macro or GoSub ... The context/extension in a CDR is generally considered the destination of a call. When looking at a 2-party call CDR, users will typically be presented with the following: context exten channel dest_channel app data default 1000 SIP/8675309 SIP/1000 Dial SIP/1000,,20 However, if the Dial actually takes place in a Macro, the current behaviour in 12 will result in the following CDR: context exten channel dest_channel app data macro-dial s SIP/8675309 SIP/1000 Dial SIP/1000,,20 The same is true of a GoSub: context exten channel dest_channel app data subs dial_stuff SIP/8675309 SIP/1000 Dial SIP/1000,,20 This generally makes the context/exten fields less than useful. It isn't hard to preserve these values in the CDR state machine; however, we need to have something that informs us when a channel is executing a subroutine. Prior to this patch, there isn't anything that does this. This patch solves this problem by adding a new channel flag, AST_FLAG_SUBROUTINE_EXEC. This flag is set on a channel when it executes a Macro or a GoSub. The CDR engine looks for this value when updating a Party A snapshot; if the flag is present, we don't override the context/exten on the main CDR object. In a funny quirk, executing a hangup handler must *not* abide by this logic, as the endbeforehexten logic assumes that the user wants to see data that occurs in hangup logic, which includes those subroutines. Since those execute outside of a typical Dial operation (and will typically have their own dedicated CDR anyway), this is unlikely to cause any heartburn. Review: https://reviewboard.asterisk.org/r/3962/ ASTERISK-24254 #close Reported by: tm1000, Tony Lewis Tested by: Tony Lewis ........ Merged revisions 422718 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422719 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	ffffc0efd8	main/cdr: Fix crash/memory consumption in CDRs in multi-party bridge scenarios ... This patch fixes an issue where CDRs would get stuck generating an infinite number of CDRs, eventually crashing Asterisk (and consuming a lot of memory along the way). When a channel enters into a multi-party bridge, the CDR engine creates mappings of each participant to each other participant, picking the 'A' party as it goes. So, if we have four channels in a multi-party bridge (Alice, Bob, Charlie, Denise), we would have something like: Alice => Bob Alice => Charlie Alice => Denise Bob => Charlie Bob => Denise Charlie => Denise This works fine when participants enter the bridge a single time. When a participant leaves a bridge, the CDRs for that channel are transitioned to a finalized state. The bug occurs if Bob rejoins. When the CDR engine creates mappings between the channels, it walks through all the participants currently in the bridge, and realizes that no one in the bridge can create a CDR with the channel (Bob). As such it creates a new CDR for the candidate and appends it to that candidate's chain. Unfortunately, on this particular code path, it doesn't stop traversing the candidate's chain. Since we just added ourselves to the chain, this causes the loop to keep going, constantly adding new CDRs. This patch makes it so the engine bails when it creates a CDR match in this case. Review: https://reviewboard.asterisk.org/r/3964/ ASTERISK-24241 #close Reported by: Deepak Singh Rawat Tested by: Deepak Singh Rawat ASTERISK-24208 Reported by: Frankie Chin ........ Merged revisions 422715 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422716 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	b06cd2f048	func_channel.c: Add missing locking to some CHANNEL() requests. ... * The CHANNEL() audionativeformat, videonativeformat, audioreadformat, and audiowriteformat now need locking since the media format rework when accessing the channel's format pointers. * Increased the buffer size for CHANNEL() audionativeformat and videonativeformat output strings since the allow=all can be a lengthy list. * Tweaked the CHANNEL() XML documentation for secure_bridge_signaling, secure_bridge_media, and state. * Ensured the output buffer is initialized for secure_bridge_signaling and secure_bridge_media. * Made use the locked_copy_string() macro instead of inlining it for trace and checkhangup. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422700 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	c56aa2d8f6	Dial API: Add a dial option to indicate the dialed channel will replace dialer ... Adds an option to the dial API that marks an outgoing dial as replacing the dialing channel for the purpose of propagating accountcode. When it is used, AST_CHANNEL_REQUESTOR_REPLACEMENT is used instead of AST_CHANNEL_REQUESTOR_BRIDGE_PEER when setting accountcodes on the involved channels with ast_channel_req_accountcodes. Review: https://reviewboard.asterisk.org/r/3968/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422684 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	43a35c2407	Call IDs: Fix appearance of call ID in core show channels when NULL ... NULL call IDs were meant to appear as '(none)' but instead were showing the contents of an uninitialized character buffer. ASTERISK-24223 Review: https://reviewboard.asterisk.org/r/3979/ ........ Merged revisions 422664 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	3ce9a8b4f4	devicestate.c: Minor tweaks ... * In ast_state_chan2dev() use ARRAY_LEN() instead of a sentinel value in chan2dev[]. * Fix some comments in chan_iax2.c. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422661 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	e59ae8b0c9	Menuselect: Fix incorrect enabling on failed deps ... This corrects a situation where menuselect can incorrectly enable a module by default that has defaultenabled set to "no" and has failed/non-selected dependencies. The bug is due to an inverted test when checking for whether the given module should be set to enabled by default on load. Review: https://reviewboard.asterisk.org/r/3975/ Reported by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422646 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	de07c80ede	Manager: Require read permission for SYSTEM in order to send FullyBooted ... Review: https://reviewboard.asterisk.org/r/3969/ ........ Merged revisions 422584 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 422625 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422626 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422631 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	1bcb46c578	res_pjsip_transport_websocket: Fix crash when the Contact header is not a URI. ... The code for changing the Contact header wrongly assumed that the Contact would always contain a URI. This is incorrect. ASTERISK-24271 Reported by: Dafi Ni ........ Merged revisions 422557 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422558 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	c98e04753b	Resolve race condition where channels enter dialplan application before media has been negotiated. ... Testsuite tests will occasionally fail because on reception of a 200 OK SIP response, an AST_CONTROL_ANSWER frame is queued prior to when media has finished being negotiated. This is because session supplements are called into before PJSIP's inv_session code has told us that media has been updated. Sometimes the queued answer frame is handled by the PBX thread before the ensuing media negotiations occur, causing a test failure. As it turns out, there is another place that session supplements could be called into, which is after media has finished getting negotiated. What this commit introduces is a means for session supplements to indicate when they wish to be called into when handling an incoming SIP response. By default, all session supplements will be run at the same point that they were prior to this commit. However, session supplements may indicate that they wish to be handled earlier than normal on redirects, or they may indicate they wish to be handled after media has been negotiated. In this changeset, two session supplements have been updated to indicate a preference for when they should be run: res_pjsip_diversion executes before handling redirection in order to get information from the Diversion header, and chan_pjsip now handles responses to INVITEs after media negotiation to fix the race condition mentioned previously. ASTERISK-24212 #close Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/3930 ........ Merged revisions 422536 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	6033c16fc3	main/cli: Do not attempt to show CDR data for internal channels ... Internal channels don't have CDRs. Querying the CDR engine for their variables will make it cranky. ........ Merged revisions 422506 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422507 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	2d8c38cf9a	res_stasis: Don't play MoH to channels by default when added to holding bridges ... When ARI manipulates a bridge, it generally doesn't care what the mixing technology is. Operations on a bridge initiated through ARI should perform their action in generally the same way, regardless of the bridge's mixing technology. While the mixing technology may determine how media flows to channels, the actual operations on a bridge themselves should be the same. Currently, this isn't the case with holding bridges. When a channel joins without a role, MoH is started on that channel automatically. Subsequent bridge operations that would stop MoH would fail (as there is no Announcer channel playing MoH to the bridge). Starting MoH on the bridge will also create two MoH streams: one from the MoH being played on the participant channel, and one from the announcer channel. From the perspective of ARI users, this is counter-intuitive - I would not expect MoH to be started for me. The mixing technology determines how media is shared between participants, not the application experience. This patch does the following: * The Stasis bridge class now inspects channels as they are going into a bridge. If the bridge has a holding capability, and the channel has no roles, we give it a participant role and mark the default behaviour to have no entertainment. This allows addChannel operations to continue to set a participant role with an entertainment option if it felt like it (or could do it). * The music on hold channel is now Stasis approved (tm) Review: https://reviewboard.asterisk.org/r/3929/ ASTERISK-24264 #close Reported by: Samuel Galarneau Tested by: Samuel Galarneau ........ Merged revisions 422503 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422504 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	e0c66f5f14	confbridge: Add Duration to ConfbridgeList event ... The ConfbridgeList event doesn't include how long the user has been a member of the conference. This patch adds Duration (seconds) which is based on user->chan->answertime. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3955/ ........ Merged revisions 422444 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422445 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	d4dd19cb77	manager: Make WaitEvent action respect eventfilters ... A WaitEvent issued via an http session isn't respecting eventfilters defined for the user. I just added a match_filter to the predicate that controls astman_append. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3958/ ........ Merged revisions 422439 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 422440 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422441 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422442 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	34ea694094	doc: Add a manpage for the smsq utility ... This patch adds a manpage for the smsq utility. Note that this is one of the patches the Debian distro applies for the Asterisk project, as per ASTERISK-24191. Review: https://reviewboard.asterisk.org/r/3895/ ASTERISK-24171 #close Reported by: Jeremy Laine patches: smsq.8 uploaded by Jeremy Laine (License 6561) ........ Merged revisions 422376 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 422377 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422378 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422379 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	636db18496	doc: Add a manpage for the aelparse utility ... This patch adds a manpage for the aelparse utility. Note that this is one of the patches the Debian distro applies for the Asterisk project, as per ASTERISK-24191. Review: https://reviewboard.asterisk.org/r/3896/ ASTERISK-24171 #close Reported by: Jeremy Laine patches: aelparse.8 uploaded by Jeremy Laine (License 6561) ........ Merged revisions 422371 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 422372 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422373 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422374 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	bd99a96b21	The assertion that peer was not found on final event ... message was being triggered on configuration reload. This patch changes that case to just return instead. Review: https://reviewboard.asterisk.org/r/3953/ Commited in trunk revision 422358 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422359 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	ed4c696319	LICENSE: Clarify language in Asterisk's LICENSE to allow for linking to UniMRCP ... The UniMRCP project distributes Asterisk modules that integrate Asterisk with UniMRCP, and other Asterisk users use the UniMRCP library as well. Unfortunately, the UniMRCP license is Apache 2.0, which per the Free Software Foundation, is not a compatible license with the GPLv2. "Please note that this license is not compatible with GPL version 2, because it has some requirements that are not in that GPL version. These include certain patent termination and indemnification provisions. The patent termination provision is a good thing, which is why we recommend the Apache 2.0 license for substantial programs over other lax permissive licenses." On the other hand, UniMRCP is a great project and we'd like to let people use it with Asterisk. This patch updates the LICENSE text to allow users to link Asterisk with UniMRCP and distribute the resulting binaries. ........ Merged revisions 422293 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 422294 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422295 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422296 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Michael L. Young	64f1a6e830	chan_iax2: Fix Dynamic IAX2 Registrations After Temporary DNS Failure ... The reporter on the issue found some issues when upgrading from version 10 to 11 on 55 hosts. Two situations that can occur with dynamic registrations. 1. With dnsmgr disabled, if the host is not resolvable we are not trying to resolve the host again when it is time to attempt to register again. This results in never registering to the host. 2. With dnsmgr enabled, when the host is temporarily not resolvable the address is set to 0.0.0.0:0 and then when the host is resolvable the port is not being restored and stays set to 0. This patch resolves these two issues by: * Storing the hostname so that it can be used for resolving with DNS. * Resolve the hostname on the next scheduled attempt to register. * Storing the port used to reach the host so that when the hostname is resolvable again, we can set the port again if the port is still unset after looking up the host. ASTERISK-23767 #close Reported by: David Herselman Tested by: David Herselman, Michael L. Young Patches: asterisk-23767-dns_reg_retry_and_set_port_11_v3.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/3856/ ........ Merged revisions 422274 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422275 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422276 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	6e698ef440	Added ConfBridge AMI event note to UPGRADE.txt. ... ........ Merged revisions 422255 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422256 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	7918e67d35	Fix bug that did not allow for multiple batched RLS notifications to be sent. ... A misunderstanding of how the scheduler worked caused further batched notifications beyond the first not to get scheduled. Now we reset our scheduler ID to -1 after the batched notification is sent. This way, further notifications can be scheduled when they arise. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422239 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a4700eee6a	res/res_pjsip/pjsip_options.c: Eliminate excessive RAII_VAR usage. ... * Fix off nominal ref leak in find_or_create_contact_status(). * Add missing NULL check of status in update_contact_status() and init_start_time(). ........ Merged revisions 422214 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422215 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a02d8a0681	sched: Fix typo and whitespace change. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422200 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	372236cc2a	confbridge: Add 'Admin' param to join, leave, mute, unmute and talking events ... Currently there's no way to tell if a user is an admin or not when receiving the join, leave, mute, unmute and talking events. This patch adds that capability. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3950/ ........ Merged revisions 422176 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422177 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	a4a58c2771	CallerID: Fix parsing of malformed callerid ... This allows the callerid parsing function to handle malformed input strings and strings containing escaped and unescaped double quotes. This also adds a unittest to cover many of the cases where the parsing algorithm previously failed. Review: https://reviewboard.asterisk.org/r/3923/ Review: https://reviewboard.asterisk.org/r/3933/ ........ Merged revisions 422112 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 422113 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 422114 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422154 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	e985cb076d	confbridge: Make kick, mute and unmute handle channel targets consistently. ... Kick, mute and unmute were a little inconsistent in their handling of channel targets. This patch cleans that up by insuring they all handle the 'all' target consistently and adds the 'participants' target which acts on non-admins. Documentation for kick was also cleaned up as it never supported partial channel names. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3944/ ........ Merged revisions 422090 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422091 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	7c4ed8cc89	Fix race condition in the scheduler when deleting a running entry. ... When scheduled tasks run, they are removed from the heap (or hashtab). When a scheduled task is deleted, if the task can't be found in the heap (or hashtab), an assertion is triggered. If DO_CRASH is enabled, this assertion causes a crash. The problem is, sometimes it just so happens that someone attempts to delete a scheduled task at the time that it is running, leading to a crash. This change corrects the issue by tracking which task is currently running. If that task is attempted to be deleted, then we mark the task, and then wait for the task to complete. This way, we can be sure to coordinate task deletion and memory freeing. ASTERISK-24212 Reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/3927 ........ Merged revisions 422070 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422071 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	00ffbc40e1	res_musiconhold.c: Release any format refs before memset(). ... * Clear the channel music_state pointer before destroying the music_state object for safety. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422037 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	d6ea6f5848	res_musiconhold: Fix MOH restarting where it left off from the last hold. ... Restore code removed by https://reviewboard.asterisk.org/r/3536/ that introduced a regression that prevents MOH from restarting were it left off the last time. ASTERISK-24019 #close Reported by: Jason Richards Patches: jira_asterisk_24019_v1.8.patch (license #5621) patch uploaded by rmudgett Review: https://reviewboard.asterisk.org/r/3928/ ........ Merged revisions 421976 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421977 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421978 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421979 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	42fe127009	res_pjsip_transport_websocket: Attach the Websocket module on outgoing INVITEs. ... In order to alter the Contact header on in-dialog requests and responses the Websocket module must be attached on outgoing INVITEs. The Contact header is modified so that the PJSIP transport layer can find and use the existing Websocket connection based on the source IP address, port, and transport. ASTERISK-24143 #close Reported by: Aleksei Kulakov ........ Merged revisions 421955 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421956 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	e86ee8e76b	res_pjsip_transport_websocket: Fix a progressive memory growth. ... The packet structure used to receive messages was using the transport pool. This meant that for each parsing the pool would grow accordingly. Since memory can not be reclaimed without resetting it this would cause the memory pool to grow and grow. This change uses a specific memory pool for the packet structure and resets it to a fresh state after the message has been received and handled. ........ Merged revisions 421939 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421945 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	3592d4b398	res_pjsip_transport_websocket: Ensure secure Websocket clients can be called. ... This change enforces the transport in the Contact header for Websocket clients. Previously a client may provide a transport of 'ws' when it is actually using a transport of 'wss'. This would cause outgoing calls to fail as the existing connection could not be found. ........ Merged revisions 421931 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421932 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	6fa02d1bfd	chan_sip: Use the server reflexive ICE candidate RTCP port as provided. ... This code originally worked around an issue within res_rtp_asterisk itself. The wrong socket was being used for the STUN check for RTCP, causing the port to be the same as RTP. This was subsequently fixed and the RTCP port provided for the ICE candidate is correct and does not need to be incremented. ASTERISK-23997 #close Reported by: Badalian Vyacheslav Patches: plus1.diff submitted by Badalian Vyacheslav (license 5249) ........ Merged revisions 421909 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421910 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421911 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	16c96760e7	Fix a locking inversion in MixMonitor. ... We need to unlock the audiohook before trying to lock the channel, since the correct locking order is channel then audiohook. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421882 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	712907eec6	ARI: Fix a crash caused by hanging during playback to a channel in a bridge ... ASTERISK-24147 #close Reported by: Edvin Vidmar Review: https://reviewboard.asterisk.org/r/3908/ ........ Merged revisions 421879 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	50381d2c77	main/message: Add a new-line to a DEBUG message ... ........ Merged revisions 421859 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421860 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	6684442945	res_musiconhold.c: Remove obsolete REF_DEBUG code. ... Remove unneeded code that writes to the wrong file location in an obsolete format. ........ Merged revisions 421799 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421800 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421801 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421802 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	a457acff46	Switch from hostname to an IP address in the SDP origin line. ... Using the hostname in the SDP origin line may not satisfy the requirement of RFC 4566 that we use a FQDN or IP address. This change has us use the same information from the SDP connection line if possible. If not possible, we'll use the configured media address. And if that's not possible, we use the result of a PJLIB call to get the IP address of ourself. ASTERISK-23994 #close Reported by Private Name Review: https://reviewboard.asterisk.org/r/3925 ........ Merged revisions 421796 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421797 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	5de3fa2c60	Ensure after-bridge behavior is correct when moving from Stasis to a non-Stasis bridge. ... Because of the departable state of channels that enter Stasis bridges, Stasis has to take responsibility for directing the channel to its intended after-bridge destination if the channel moves from a Stasis bridge to a non-Stasis bridge. This change ensures that when such a move occurs, when the channel leaves the bridging system, any after bridge gotos are honored. Review: https://reviewboard.asterisk.org/r/3920 ........ Merged revisions 421792 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421794 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	a9befb9eec	Let's try checking the name and number, instead of the name twice. ... ........ Merged revisions 421789 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421790 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago