asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	02295456ef	chan_rtp: Add unicast RTP support. This module supports sending both unicast and multicast RTP to a specified target. Multicast functionality is the same as chan_multicast_rtp was. In the case of unicast a specific IP address and port can be specified, along with optional RTP engine and format in the form of: UnicastRTP/<ip address>:<port>/<engine>/<format> This can be useful for sending a copy of a media stream to another application for processing. Review: https://reviewboard.asterisk.org/r/3981/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423004 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	dd6bdede7d	Realtime: Fix a bug that caused realtime destroy command to crash Also has could affect with anything that goes through ast_destroy_realtime. If a CLI user used the command 'realtime destroy <family>' with only a single column/value pair, Asterisk would crash when trying to create a variable list from a NULL value. ASTERISK-24231 #close Reported by: Niklas Larsson Review: https://reviewboard.asterisk.org/r/3985/ ........ Merged revisions 422984 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422985 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422991 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	c212a71f0b	Remove undocumented default behavior of ast_play_and_record_full acceptdtmf. ast_play_and_record_full() has a parameter called "acceptdtmf" that is a string of acceptable DTMF digits that may be pressed by a caller to end and accept the recording. ARI uses this function in order to perform recording, and it provides options for what is passed as acceptdtmf to ast_play_and_record_full(). By default, ARI passes an empty string, with the intention that no DTMF can be used to end the recording. The problem is that ast_play_and_record_full() attempts to be "helpful" by setting "#" as the acceptdtmf if an empty string or NULL pointer has been passed in. With ARI, this results in unexpected behavior occurring if you have attempted to intercept "#" yourself in order to perform some other manipulation of the live recording. This change removes the "helpful" behavior by no longer accepting "#" as a default acceptdtmf if none is specified by the caller of ast_play_and_record_full(). This makes the ARI scenario work as expected. The other callers of ast_play_and_record_full() are app_voicemail and app_minivm, and in both cases, they pass an explicit "#" to ast_play_and_record_full() as acceptdtmf, so they are unaffected by this change. ........ Merged revisions 422964 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422965 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422967 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	93894d53c4	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 ........ Merged revisions 422905 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422906 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Sean Bright	7bd3287a11	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 ........ Merged revisions 422901 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422902 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a47873168a	Update CHANGES for CHANNEL(onhold). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422885 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Rusty Newton	51f082af34	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 ........ Merged revisions 422883 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422884 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	9183416fe2	func_channel: Add CHANNEL(onhold) item to get the current hold status of the channel. It would be useful to get the current hold status of a channel. Added CHANNEL(onhold) item that returns 1 (onhold) and 0 (not-onhold) for the hold status of a channel. ASTERISK-24038 Reported by: Matt Jordan AFS-113 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3983/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	baf99dffac	Add note about configuring list_items on a single line. ........ Merged revisions 422855 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	5ad0edacb6	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. ........ Merged revisions 422853 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422854 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	c6bc44f700	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 ........ Merged revisions 422851 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422852 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	ef5f7a0e32	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 ........ Merged revisions 422836 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422837 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	71acca4de2	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 ........ Merged revisions 422770 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422771 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	e4591f98b1	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 ........ Merged revisions 422767 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422768 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	fd8010de2b	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 ........ Merged revisions 422747 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422748 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	d42b116925	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 ........ Merged revisions 422719 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422720 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	4499eb05d8	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 ........ Merged revisions 422716 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	025bd1bf3f	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. ........ Merged revisions 422700 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	85878c4dd8	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/ ........ Merged revisions 422684 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422697 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	e19017fc00	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 ........ Merged revisions 422665 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422683 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	5a1de68b9a	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. ........ Merged revisions 422661 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	2362d88a18	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 ........ Merged revisions 422646 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422647 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	af75e45da1	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 ........ Merged revisions 422631 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422632 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	3cd36d0e10	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 ........ Merged revisions 422558 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422559 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	1b64f353f1	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 ........ Merged revisions 422542 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422543 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	897cbf6a4f	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 ........ Merged revisions 422507 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422524 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	df5dbbd878	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 ........ Merged revisions 422504 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422505 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	5aefecd81e	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 ........ Merged revisions 422445 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422446 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	59d4dbd3d0	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 ........ Merged revisions 422442 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422443 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	664f83a03b	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 ........ Merged revisions 422379 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422380 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	81598fa082	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 ........ Merged revisions 422374 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422375 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	2df2d785b7	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422358 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	3194892ea2	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 ........ Merged revisions 422296 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422297 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Michael L. Young	c5916fb39f	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 ........ Merged revisions 422276 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	4e750a26fd	Added ConfBridge AMI event note to UPGRADE.txt. ........ Merged revisions 422255 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 422256 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Paul Belanger	ef28cc0d43	chan_sip.c: Add 'rtpbindaddr' setting Users now have the ability to bind the rtpengine instance to a specific IP address. For example, you want chan_sip (call control) on eth0 but rtp (media) on eth1. ASTERISK-24280 #close Reported by: Paul Belanger Tested by: Paul Belanger Review: https://reviewboard.asterisk.org/r/3952/ Patches: rtpengine.diff uploaded by Paul Belanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422241 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	327d67270f	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. ........ Merged revisions 422239 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422240 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	94e1b4a8a4	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 ........ Merged revisions 422215 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422216 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	4728c05957	sched: Fix typo and whitespace change. ........ Merged revisions 422200 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422201 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	7c1a22fba7	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 ........ Merged revisions 422177 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422178 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	bf85018107	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 ........ Merged revisions 422154 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	d199536a04	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 ........ Merged revisions 422091 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422092 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	c5ab4adf17	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 ........ Merged revisions 422071 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	fefa6fba82	res_musiconhold.c: Release any format refs before memset(). * Clear the channel music_state pointer before destroying the music_state object for safety. ........ Merged revisions 422037 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@422038 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	2b19d94a71	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 ........ Merged revisions 421979 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421980 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	497a92d079	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 ........ Merged revisions 421956 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421957 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	477e2e6edb	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 ........ Merged revisions 421945 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421950 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	2c0cbf8e64	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 ........ Merged revisions 421932 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421933 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	cee660dadf	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 ........ Merged revisions 421911 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421912 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	dcfffce66d	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. ........ Merged revisions 421882 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@421883 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

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