asterisk

Commit Graph

Author	SHA1	Message	Date
David M. Lee	42891b9a08	Removed codecs/g722/*.i on make clean ........ Merged revisions 383340 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@383341 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	c379172cae	Fix a bug where resources were not found due to hashing on the priority itself. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@383266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	4a50764715	tcptls: Prevent unsupported options from being set AMI, HTTP, and chan_sip all support TLS in some way, but none of them support all the options that Asterisk's TLS core is capable of interpreting. This prevents consumers of the TLS/SSL layer from setting TLS/SSL options that they do not support. This also gets tlsverifyclient closer to a working state by requesting the client certificate when tlsverifyclient is set. Currently, there is no consumer of main/tcptls.c in Asterisk that supports this feature and so it can not be properly tested. Review: https://reviewboard.asterisk.org/r/2370/ Reported-by: John Bigelow Patch-by: Kinsey Moore (closes issue AST-1093) ........ Merged revisions 383165 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@383166 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	fb8760d679	When a session timer expires during a T.38 call, re-invite with correct SDP When a session timer expires during a dialog that has re-negotiated to T.38 and Asterisk is the refresher, Asterisk will send a re-INVITE with an SDP containing audio media only. This causes some hilarity with the poor fax session under weigh. This patch corrects that by sending T.38 parameters if we are in the middle of a T.38 session. (closes issue ASTERISK-21232) Reported by: Nitesh Bansal patches: dont-send-audio-reinvite-for-sess-timer-in-t38-call.patch uploaded by nbansal (License 6418) ........ Merged revisions 383124 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@383125 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	e76649b53c	Fix processing of call files when using KQueue on OS X In certain situations, call files are not processed when using KQueue with pbx_spool. Asterisk was sending an invalid timeout value when the spool directory is empty, causing the call to kevent to error immediately. This can create a tight loop, increasing the CPU load on the system. (closes issue ASTERISK-21176) Reported by: Carlton O'Riley patches: kqueue_osx.patch uploaded by coriley (License 6473) ........ Merged revisions 383120 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@383121 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	e77ab46721	Fix whitespace in AST_EXT_LIB_CHECK macro. ........ Merged revisions 383061 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@383062 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	d4e02135a3	Fix Sorting Order For Parking Lots Stored In Static Realtime When retrieving the parking lots from a MySQL database table, the current order is "filename, cat_metric desc, var_metric asc, category". If there are multiple parking lots with the same cat_metric but different categories, everything is being sorted on cat_metric first resulting in errors when loading the parking lots. This patch fixes the problem by sorting on the category field first, then the cat_metric field. (closes issue ASTERISK-21035) Reported by: Alex Epshteyn Patches: asterisk-21035-orderby.diff Michael L. Young (license 5026) ........ Merged revisions 382942 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382943 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	4a72692442	Update Contributed Realtime Schema Files - IPv6 Addresses This commit updates some fields in the contributed realtime schema files to handle IPv6 addresses. (closes issue ASTERISK-21173) Reported by: Torrey Searle Patches: realtime_sql.patch Torrey Searle (license 5334) asterisk-21173-update-ip-fields.diff Michael L. Young (license 5026) ........ Merged revisions 382939 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382940 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	7031ad62e8	Fix a crash when res_xmpp is configured using a username without a domain. (closes issue ASTERISK-21156) Reported by: amsoft2001 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382923 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	77ca918044	Include the Username field in SIP Registry events when Status is registered In ASTERISK-17888, the AMI Registry event during SIP registrations was supposed to include the Username field. Somehow, one of the events was missed. This patch corrects that - the Username field should be included in all AMI Registry events involving SIP registrations. (issue ASTERISK-17888) (closes issue ASTERISK-21201) Reported by: Dmitriy Serov patches: chan_sip.c.diff uploaded by Dmitriy Serov (license 6479) ........ Merged revisions 382847 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382848 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Igor Goncharovskiy	1531ef77a8	Fix core dump on CLI usage Fix issue with 'unistim show info' CLI command when device connected not configured git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382827 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	96c231fc18	chan_sip: Update the via header when relaying SMS MESSAGE Prior to this change, certain conditions for sending the message would result in an address of '(null)' being used in the via header of the SIP message because a NULl value of pvt->ourip was used when initially generating the via header. This is fixed by adding a call to build_via when the address is set before sending the message. (closes issue ASTERISK-21148) Reported by: Zhi Cheng Patches: 700-sip_msg_send_via_fix.patch uploaded by Zhi Cheng (license 6475) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382739 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	0b37e777f6	Let vm_mailbox_snapshot combine "Urgent" when no folder is specified r381835 fixed a bug in vm_mailbox_snapshot where combining INBOX and Old forgot that Urgent also "counts" as new messages. This fixed the problem when any of the three folders was specified and the combine option was used. It missed the case where the folder isn't specified and we build a snapshot of all folders. This patch corrects that. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382617 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	3a0a627a00	Ensure that logmsgs are freed properly Messages sent while the logger thread is shutting down will now have their associated callid freed properly. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382574 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	5b1533ca7b	Add a 'secret' probation strictrtp mode to handle delayed changes in RTP source Often, Asterisk may realize that a change in the source of an RTP stream is about to occur and ask that the RTP engine reset it's lock on the current RTP source. In certain scenarios, it may take awhile for the new remote system to send RTP packets, while the old remote system may continue providing RTP during that time period. This causes Asterisk to re-lock onto the old source, thereby rejecting the new source when the old source stops sending RTP and the new source begins. This patch prevents that by having a constant secondary, 'secret' probation mode enabled when an RTP source has been chosen. RTP packets from other sources are always considered, but never chosen unless the current RTP source stops sending RTP. Review: https://reviewboard.asterisk.org/r/2364 (closes issue AST-1124) Reported by: John Bigelow Tested by: John Bigelow (closes issue AST-1125) Reported by: John Bigelow Tested by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382573 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	dde15421b5	Recorded merge of revisions 382513 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Correct app_page documentation The 'A' and 'n' options for Page() mention that the announcement will be played simultaneously. This is not necessarily the case. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382514 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Igor Goncharovskiy	d4cb37c956	Fix several unreleased mutex locks that cause problem with processing calls (Closes issue ASTERISK-21119) Reported by: Daniel Bohling Tested by: Daniel Bohling ........ Merged revisions 382409 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382410 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	0667c05261	Fix comparison of presence state in event subsystem. Several new IEs were not given types (or names), causing the comparison function to improperly succeed. This adds those. (closes issue AST-1128) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382390 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	04100feed7	Confbridge CLI new record file name check. This fix checks to make sure that if a confbridge record start command is issued from the CLI it will always use the file name given on the CLI even if it changes between start/stop records for a conference. Previously it had been reusing the same file between start/stops even if a new filename was given. (issue AST-1088) Reported by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382385 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	2109e47109	Fix / Clean Up Some Items To Handle The New auto_* NAT Options The original report had to do with a realtime peer behind NAT being pruned and the peer's private address being used instead of its external address. Upon debugging, it was discovered that this was being caused by the addition of the auto_force_rport and auto_comedia settings. This patch does the following: * Adds a missing note to the CHANGES file indicating that the default global nat setting is auto_force_rport * Constify the 'req' parameter for check_via() * Add calls to check_via() in a couple of places in order for the auto_* settings to do their job in attempting to determine if NAT is involved * Set the flags SIP_NAT_FORCE_RPORT and SIP_PAGE2_SYMMETRICRTP if the auto_* settings are in use where it was needed * Moves the copying of peer flags up in build_peer() to before they are used; this fixes the realtime prune issue * Update the contrib/realtime schemas to allow the nat column to handle the different nat setting combinations we have This patch received a review and "Ship It!" on the issue itself. (closes issue ASTERISK-20904) Reported by: JoshE Tested by: JoshE, Michael L. Young Patches: asterisk-20904-nat-auto-and-rt-peersv2.diff Michael L. Young (license 5026) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382322 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	a78bb96d94	While the ICE negotiation is occurring leave strictrtp in an open state, media can and will come from different places. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382298 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	f6b368216a	Fix a bug with ICE and strictrtp where media could get dropped. If the end result of the ICE negotiation resulted in the path for media changing it was possible for the strictrtp code to discard the RTP packets. This change causes strictrtp to enter learning mode once again when the ICE negotiation has completed successfully. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382296 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	b056a88e08	Prevent deadlock in chan_iax2 when attempting to set caller ID A deadlock can occur in chan_iax2 when it attempts to set the caller ID, as it already holds the iax2 private lock and improperly fails to obtain the channel lock before calling ast_set_callerid. By not safely obtaining the channel lock, a locking inversion can take place, causing a deadlock. This patch solves this by calling the required deadlock avoidance functions that obtain the channel lock before setting the caller ID. Thanks to Pavel for fixing my syntax errors and testing this patch out. (closes issue ASTERISK-21128) Reported by: Pavel Troller Tested by: Pavel Troller patches: ASTERISK-21128-1.8.diff uploaded by mjordan (license 6283) ASTERISK-21128-modified-1.8.diff uploaded by Pavel Troller (license 6302) ........ Merged revisions 382233 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	47bd918dad	Let channels joining a MeetMe conference opt out of the denoiser For some channel drivers, specifically those that have a varying rate in the number of audio samples, the audio quality for a MeetMe conference can be exceedingly poor. This is due to a unilateral application of the DENOISE function in func_speex to channels joining the conference. The denoiser function in the speex library is initialized with the number of audio samples in each sample that will be provided to it. If the number of audio samples changes, the denoiser has to be thrown away and re-initialized. While this could be worked around by removing func_speex, that doesn't help if you actually use the denoiser with other channels on the system. This patches does the following: * Checks for the presence of func_speex as opposed to codec_speex when determining if the DENOISE function is present (which is where the function is actually implemented) * Adds an option to MeetMe 'n' that causes the denoiser to not be applied to a channel when it joins. This keeps the current behavior the default, but let's users disable the denoiser if it causes problems on their system. Review: https://reviewboard.asterisk.org/r/2358 (closes issue AST-1062) Reported by: Thomas Arimont ........ Merged revisions 382227 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382230 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	e26bd56ff4	Relax dialog checking in get_sip_pvt_byid_locked so it works when the dialog is forked. (closes issue ASTERISK-20638) Reported by: eelcob Patches: pedantic-call-pickup-from-tag.patch uploaded by eelcob (license 6442) ........ Merged revisions 382171 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382174 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	2d95e2884e	Regenerate the configure script. The one in the tree was not working for me at all. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382151 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Tzafrir Cohen	6d5b04d708	Consider linux-gnuspe as linux-gnu * The powerpcspe Linux port uses linux-gnuspe as the OS string. * Our build system shouldn't really care for that, so just call it linux-gnu. * Original report: Roland Stigge , http://bugs.debian.org/701505 Review: https://reviewboard.asterisk.org/r/2357/ ........ Merged revisions 382110 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382111 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Walter Doekes	ce9bc4e9a1	Correct RPID parsing for unquoted display-name. Parsing Remote-Party-ID will now succeed if display-name is of the *(token LWS) kind and not just the quoted-string kind. Review: https://reviewboard.asterisk.org/r/2341/ ........ Merged revisions 382107 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382108 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Tzafrir Cohen	35fce97da3	Remove unneeded linux-gnueabi* As of r380521 the configure scripts converts the value of linux-gnueabi* of OSARCH to "linux-gnu". So no point in testing for those values. ........ Merged revisions 382087 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382096 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	4163b04c42	Fix typo in r382068 Well, that was embarrassing. Removed an '-l' that somehow got in there. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382069 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	d9f20617d0	Clean up ConfBridge commands to account for wait_marked users When ConfBridge was refactored to better handle the concept of marked, wait_marked, and normal users co-existing in a conference (thereby implementing a state machine for the conference), the wait_marked users were put into their own list of conference participants, separate from the active users. This list is used for wait_marked users when they are waiting in a conference but no marked user has joined; normal users may have joined at this point however. There are several AMI/CLI commands that affect conference users that were not checking the wait_marked users list: * CLI/AMI commands that mute/unmute a participant. In this case, wait_marked users have to remain in their particular state and should not be affected - however, the commands would return "Channel not found" as opposed to the appropriate error condition. * CLI/AMI commands that kick a participant. An admin should always be able to kick a participant out of the conference. This patch fixes both sets of commands, and cleans up the CLI commands slightly by allowing them to complete a participant name (this was supposed to have been added, but the function call was commented out and wasn't implemented). Review: https://reviewboard.asterisk.org/r/2346/ (closes issue AST-1114) Reported by: John Bigelow Tested by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382068 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	b679563222	Ensure that the default bridge/user profiles are always available ConfBridge and Page require that there always be a default bridge and user profile available. While properties of the default profiles can be overriden in the configuration file, removing them can create situations where neither application can function properly. This patch ensures that if an administrator removes the profiles from the confbridge.conf configuration file, the profiles are added upon load. Documentation clarifying this has been added to the confbridge.conf.sample file. Review: https://reviewboard.asterisk.org/r/2356/ (closes issue AST-1115) Reported by: John Bigelow Tested by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382066 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	586efe0d7f	Clean up use of va_end/va_args in res_config_mysql There were several problems using variadic argument macros in res_config_mysql. * Improper use of va_end. Multiple calls to va_end were possible resulting in an unbalanced matching of va_start/va_end. * Calls to va_arg after a possible encounter of a SENTINEL value. This patch corrects those errors. (closes issue ASTERISK-19451) Reported by: wdoekes patches: ASTERISK-19451-1.8--2.diff uploaded by wdoekes (License 5674) ........ Merged revisions 382021 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382022 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	fec0881135	Set the sin_family on the bind address socket during initialization Somehow, chan_jingle has managed to operate for years without setting the sin_family on its bindaddr socket. This patch properly sets the field during initial module load to AF_INET. Note that the patch on the issue was modified slightly to change the initialization of the socket from allocation of a chan_jingle private to the module initialization, as the bindaddr object (which is static) only needs to have the address set once. (closes issue ASTERISK-19341) Reported by: andre valentin patches: 0105-chan_jingle.patch uploaded by avalentin (License 6064) ........ Merged revisions 381975 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381976 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	e469bc5237	Don't display the AMI ALL class authorization for users if they don't have it When converting AMI class authorizations to a string representation, the method always appends the ALL class authorization. This is especially important for events, as they should always communicate that class authorization - even if the event itself does not specify ALL as a class authorization for itself. (Events have always assumed that the ALL class authorization is implied when they are raised) Unfortunately, this did mean that specifying a user with restricted class authorizations would show up in the 'manager show user' CLI command as having the ALL class authorization. Rather then modifying the existing string manipulation function, this patch adds a function that will only return a string if the field being compared explicitly matches class authorization field it is being compared against. This prevents ALL from being returned unless it is actually specified for the user. (closes issue ASTERISK-20397) Reported by: Johan Wilfer ........ Merged revisions 381939 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381943 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	dfe7451e1c	Make ParkAndAnnounce return to priority + 1 when return context is not defined The ParkAndAnnounce application documentation for the optional return_context parameter states the following: return_context The goto-style label to jump the call back into after timeout. Default 'priority+1'. Unfortunately, the application was sending the channel back into the dialplan at 'priority', which is the ParkAndAnnounce application call. This causes an infinite loop of the channel constantly being parked, announced, timed out, parked, announced, timed out... while fun, especially for those callers you wish to drive to the end of madness, this was not the intent of the application. (closes issue ASTERISK-20113) Reported by: serginuez patches: app_parkandannounce.diff uploaded by serginuez (License 6405) ........ Merged revisions 381916 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	2d451c64f7	Fix FastAGI To Properly Check For A Connection When IPv6 support was added to FastAGI, the intent was to have the ability to check all addresses resolved for a host since we might receive an IPv4 address and an IPv6 address. The problem with the current code, is that, since we are doing O_NONBLOCK, we get EINPROGRESS when calling ast_connect() but are ignoring this instead of handling it. We break out of the loop and continue on. When we later call ast_poll(), it succeeds but we never check if we have a connection or not on the socket level. We then attempt to send data to the host address that we think is setup and it fails. We then check the errno and see that we have "connection refused" and then return with agi failed. This patch does the following: * Handles EINPROGRESS by creating the function handle_connection() - ast_poll() was moved into this function - This function checks the results of the connection on the socket level after calling ast_poll() * Continues to the next address if the above fails to create a connection * Once all addresses resolved are tried and we still are unable to establish a connection, then we return that the FastAGI call failed (closes issue ASTERISK-21065) Reported by: Jeremy Kister Tested by: Jeremy Kister, Michael L. Young Patches: asterisk-21065_poll_correctly_v4.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2330/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381893 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	2ae927b930	app_dial: Honor the 'c' flag when the calling party hangs up Apparently this feature became broken in 11, probably as a result of the Hangup Cause project. (closes issue ASTERISK-21113) Reprted by: Heiko Wundram Patches: app_dial.patch uploaded by Heiko Wundram (license 5822) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	a5159f3b9a	Properly detect launchd Asterisk was a little too pro-active in claiming that it found launchd. On systems without launchd - such as FreeBSD - this resulted in certain items in Asterisk that conflict with launchd to not be selectable, such as res_timing_kqueue. (closes issue ASTERISK-20749) Reported by: Oleg Baranov ........ Merged revisions 381847 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381848 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	c157d591f5	Let vm_mailbox_snapshot_create's combine option apply to "Urgent" as well The vm_mailbox_snapshot_create function has an option that combines the contents of INBOX and Old into a single snapshot. The intent of this is that both 'new' messages and 'deleted' messages are given in a single snapshot, as some applications prefer this view of the voicemail world. Unfortunately, the initial implementation ignored the "Urgent" folder. The "Urgent" folder is a pseudo-INBOX, in that new messages left with the 'U' flag will be placed in that folder as opposed to INBOX. Thus, the option failed the intent with which it was added. This patch makes it so that the "Urgent" folder is included in the snapshot when that option is used. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381835 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	cf533947c7	Write the correct callid to the data1 field in queue_log for transfer events. The incorrect callid was being written to the "data1" field in queue_log table for transfer events. The callid of the queue was being written instead of the transfer target's callid. This now gets the correct "transfer to" number and places that in the "data1" field of the queue_log table when a transfer event is triggered. (closes issue ASTERISK-19960) Reported by: vladimir shmagin ........ Merged revisions 381770 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381791 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	5d8ddec0e2	Confbridge channels staying active when all participants leave. If you started/stopped recording of a conference multiple times channels would remain active even when all participants left the conference. This was due to the fact that a reference to the confbridge was being added every time a start record command was issued, but when the recording was stopped there was no matching de-reference thus keeping the conference alive. Made sure only a single reference is added for the record thread no matter how many times recording is started/stopped. A de-reference is issued upon thread ending. Note, this issue is being fixed under AST-1088 since it relates to it and should have been corrected along with those modifications. (issue AST-1088) Reported by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381737 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	2be1724539	Fixed Confbridge file recording deadlock and appending. A deadlock occurred after starting/stopping and then restarting a confbridge recording. Upon starting a recording a record thread is created that holds a lock until just before exiting. Stopping the recording does not stop/exit the thread or release the lock. The thread waits until recording begins again. Starting a stopped recording signals the thread to continue and start recording again. However restarting the recording also created another record thread resulting in a deadlock. The fix was to make sure the record thread was only created once. Also it was noted that filenames for the recordings were being concatenated for each start/stop. This was fixed by creating a new file for each conference session and appending the actual recorded data within the file (e.g. passing the 'a' option to MixMonitor). (issue AST-1088) Reported by: John Bigelow Review: http://reviewboard.digium.internal/r/374/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381702 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Walter Doekes	e17ec25d47	Remove "registertrying" and add "rtp_engine" from/to sip.conf.sample The "registertrying" option was removed in r343220. The "rtp_engine" option was added in r186078 but erroneously named "engine" in the sample. Note that there is no global sip setting for a different engine. ........ Merged revisions 381668 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381669 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	7a353badc5	PRESENCE_STATE: Provide better documentation for the 'e' option. Notes that the 'e' option actually decodes data when used as a write function such as with the SET application while it encodes data when used to read. Review: https://reviewboard.asterisk.org/r/2335/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	cb623e7ad6	Don't send presencestate information if the state is invalid Previously, presencestate information was sent whenever the state was not NOT_SET. When r381594 actually returned INVALID presence state in all the places it was supposed to, it caused chan_sip to start adding presence state information to NOTIFY requests that it previously would not have added. chan_sip shouldn't be adding presence state information when the provider is in an invalid state; users can't set the state to invalid and an invalid state always implies that the provider is in an error condition. (issue AST-1084) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381613 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	31d1bd4cd8	Fix crash in PresenceState AMI action when specifying an invalid provider This patch fixes a crash in Asterisk that could be caused by using the PresenceState AMI action while providing an invalid provider. This patch also adds some additional warnings when a user attempts to provide the PresenceState action with invalid data, and removes some NOTICE statements that were still lurking in the code from testing. (closes issue AST-1084) Reported by: John Bigelow Tested by: John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381594 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	a70075ce10	Fix a crash that occurred when a BYE was received on a replaced dialog. Reference counting for the channel and its tech_pvt got messed up at some point between 1.8 and 11. The result was that if a BYE for a dialog that had been replaced (via an INVITE with Replaces) was received, Asterisk would crash due to trying to access data on a channel that was no longer there. The fix I introduced is to remove code that both unrefs the sip_pvt and sets the channel's tech_pvt to NULL when an INVITE with Replaces is handled. This way when a BYE is received, the tech_pvt will be non-NULL and so the BYE can be processed and not cause a crash. (closes issue ASTERISK-20929) reported by Kristopher Lalletti patches: ASTERISK-20929.patch uploaded by Mark Michelson (License #5049) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381566 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	1595101bb6	Stopped spamming of debug messages during attended transfer. While autoservice is running and servicing a channel the callid is being stored and removed in the thread's local storage for each iteration of the thread loop. If debug was set to a sufficient level the log file would be spammed with callid thread local storage debug messages. Added a new function that checks to see if the callid to be stored is different than what is already contained (if anything). If it is different then store/replace and log, otherwise just leave as is. Also made it so all logging of debug messages pertaining to the callid thread storage outputs only when TEST_FRAMEWORK is defined. (issue ASTERISK-21014) (closes issue ASTERISK-21014) Report by: Rusty Newton Review: https://reviewboard.asterisk.org/r/2324/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	120a7cbc03	chan_sip: Use video and text crypto attributes to append RTP profiles to SDP Some bad copy/pasting resulted in using the audio crypto attribute for both text and video RTP. Also the audio crypto isn't set until after these, so it was really just bad all around. (closes ASTERISK-20905) Reported by: Kristopher Lalletti patches: rtp_crypto_video_text.diff uploaded by Jonathan Rose (license 6182) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381553 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago

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