24218 Commits (bd4aee4b7bd10f793ae0bbd7db8fcdb9eff9c65c)

 

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Author	SHA1	Message	Date
Richard Mudgett	7a29f3603b	queue_log: Post QUEUESTART entry when Asterisk fully boots. ... The QUEUESTART log entry has historically acted like a fully booted event for the queue_log file. When the QUEUESTART entry was posted to the log was broken by the change made by ASTERISK-15863. * Made post the QUEUESTART queue_log entry when Asterisk fully boots. This restores the intent of that log entry and happens after realtime has had a chance to load. AST-1444 #close Reported by: Denis Martinez Review: https://reviewboard.asterisk.org/r/4282/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@430009 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	5c7b830492	chan_sip: Send CANCEL via original INVITE destination even after UPDATE request ... Given the following scenario: * Three SIP phones (A, B, C), all communicating via a proxy with Asterisk * A call is established between A and B. B performs a SIP attended transfer of A to C. B sets the call on hold (A is hearing MOH) and dials the extension of C. While phone C is ringing, B transfers the call (that is, what we typically call a 'blond transfer'). * When the transfer completes, A hears the ringing of phone C, while B is idle. In the SIP messaging for the above scenario, a REFER request is sent to transfer the call. When "sendrpid=yes" is set in sip.conf, Asterisk may send an UPDATE request to phone C to update party information. This update is sent directly to phone C, not through the intervening proxy. This has the unfortunate side effect of providing route information, which is then set on the sip_pvt structure for C. If someone (e.g. B) is trying to get the call back (through a directed pickup), Asterisk will send a CANCEL request to C. However, since we have now updated the route set, the CANCEL request will be sent directly to C and not through the proxy. The phone ignores this CANCEL according to RFC3261 (Section 9.1). This patch updates reqprep such that the route is not updated if an UPDATE request is being sent while the INVITE state is INV_PROCEEDING or INV_EARLY_MEDIA. This ensures that a subsequent CANCEL request is still sent to the correct location. Review: https://reviewboard.asterisk.org/r/4279 ASTERISK-24628 #close Reported by: Karsten Wemheuer patches: issue.patch uploaded by Karsten Wemheuer (License 5930) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429982 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	a77d50cf2e	acl: Fix reloading of configuration if configuration file does not exist at startup. ... The named ACL code incorrectly destroyed the config options information if loading of the configuration file failed at startup. This would result in reloading also failing even if a valid configuration file was put in place. ASTERISK-23733 #close Reported by: Richard Kenner git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429893 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	d7e4eab27c	res_http_websocket.c: Fix incorrect use of sizeof in ast_websocket_write(). ... This won't fix the reported issue but it is an incorrect use of sizeof. ASTERISK-24566 Reported by: Badalian Vyacheslav git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429867 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	375b08b673	chan_dahdi: Don't ignore setvar when using configuration section scheme. ... When the configuration section scheme of chan_dahdi.conf is used (keyword dahdichan instead of channel) all setvar= options are completely ignored. No variable defined this way appears in the created DAHDI channels. * Move the clearing of setvar values to after the deferred processing of dahdichan. AST-1378 #close Reported by: Guenther Kelleter Patch by: Guenther Kelleter git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429825 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	c0c395514c	chan_dahdi.c, res_rtp_asterisk.c: Change some spammy debug messages to level 5. ... ASTERISK-24337 #close Reported by: Rusty Newton git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429804 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	da6805c58a	chan_dahdi: Populate CALLERID(ani2) for incoming calls in featdmf signaling mode. ... For the featdmf signaling mode the incoming MF Caller-ID information is formatted as follows: *${CALLERID(ani2)}${CALLERID(ani)}#*${EXTEN}# Rather than discarding the ani2 digits, populate the CALLERID(ani2) value with what is received instead. AST-1368 #close Reported by: Denis Martinez Patches: extract_ani2_for_featdmf_v11.patch (license #5621) patch uploaded by Richard Mudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	822abf9e9b	Fix printf problems with high ascii characters after r413586 (1.8). ... In r413586 (1.8) various casts were added to silence gcc 4.10 warnings. Those fixes included things like: -out += sprintf(out, "%%%02X", (unsigned char) *ptr); +out += sprintf(out, "%%%02X", (unsigned) *ptr); That works for low ascii characters, but for the high range that yields e.g. FFFFFFC3 when C3 is expected. This changeset: - fixes those casts to use the 'hh' unsigned char modifier instead - consistently uses %02x instead of %2.2x (or other non-standard usage) - adds a few 'h' modifiers in various places - fixes a 'replcaes' typo - dev/urandon typo (in 13+ patch) Review: https://reviewboard.asterisk.org/r/4263/ ASTERISK-24619 #close Reported by: Stefan27 (on IRC) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429673 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	03d831ec94	chan_sip: Allow T.38 switch-over when SRTP is in use. ... Previously when SRTP was enabled on a channel it was not possible to switch to T.38 as no crypto attributes would be present. This change makes it so it is now possible. If a T.38 re-invite comes in SRTP is terminated since in practice you can't encrypt a UDPTL stream. Now... if we were doing T.38 over RTP (which does exist) then we'd have a chance but almost nobody does that so here we are. ASTERISK-24449 #close Reported by: Andreas Steinmetz patches: udptl-ignore-srtp-v2.patch submitted by Andreas Steinmetz (license 6523) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429632 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	b97dbd724e	DEBUG_THREADS: Fix regression and lock tracking initialization problems. ... This patch started with David Lee's patch at https://reviewboard.asterisk.org/r/2826/ and includes a regression fix introduced by the ASTERISK-22455 patch. The initialization of a mutex's lock tracking structure was not protected in a critical section. This is fine for any mutex that is explicitly initialized, but a static mutex may have its lock tracking double initialized if multiple threads attempt the first lock simultaneously. * Added a global mutex to properly serialize initialization of the lock tracking structure. The painful global lock can be mitigated by adding a double checked lock flag as discussed on the original review request. * Defer lock tracking initialization until first use. * Don't be "helpful" and initialize an uninitialized lock when DEBUG_THREADS is enabled. Debug code is not supposed to fix or change normal code behavior. We don't need a lock initialization race that would force a re-setup of lock tracking. Lock tracking already handles initialization on first use. * Properly handle allocation failures of the lock tracking structure. * No need to initialize tracking data in __ast_pthread_mutex_destroy() just to turn around and destroy it. The regression introduced by ASTERISK-22455 is the result of manipulating a pthread_mutex_t struct outside of the pthread library code. The pthread_mutex_t struct seems to have a global linked list pointer member that can get changed by other threads. Therefore, saving and restoring the contents of a pthread_mutex_t struct is a bad thing. Thanks to Thomas Airmont for finding this obscure regression. * Don't overwrite the struct ast_lock_track.reentr_mutex member to restore tracking data in __ast_cond_wait() and __ast_cond_timedwait(). The pthread_mutex_t struct must be treated as a read-only opaque variable. Miscellaneous other items fixed by this patch: * Match ast_suspend_lock_info() with ast_restore_lock_info() in __ast_cond_timedwait(). * Made some uninitialized lock sanity checks return EINVAL and try a DO_THREAD_CRASH. * Fix bad canlog initialization expressions. ASTERISK-24614 #close Reported by: Thomas Airmont Review: https://reviewboard.asterisk.org/r/4247/ Review: https://reviewboard.asterisk.org/r/2826/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429539 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	94c6c279e8	res/res_agi: Make Verbose message for 'stream file' match other playbacks ... The Verbose message displayed when a file is played back via 'stream file' was formatted differently than other playbacks: * It didn't include the channel name * It didn't include the channel language It does, however, include the playback offset as well as any escape digits. That information was kept; however, this patch updates the formatting to more closely match the Verbose messages displayed when a file is played back by 'control stream file', Playback, ControlPlayback, or any other file playback operation. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	3aeed57b3c	res_http_websocket: Fix crash due to double freeing memory when receiving a payload length of zero. ... Frames with a payload length of 0 were incorrectly handled in res_http_websocket. Provided a frame with a payload had been received prior it was possible for a double free to occur. The realloc operation would succeed (thus freeing the payload) but be treated as an error. When the session was then torn down the payload would be freed again causing a crash. The read function now takes this into account. This change also fixes assumptions made by users of res_http_websocket. There is no guarantee that a frame received from it will be NULL terminated. ASTERISK-24472 #close Reported by: Badalian Vyacheslav Review: https://reviewboard.asterisk.org/r/4220/ Review: https://reviewboard.asterisk.org/r/4219/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429270 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	1a934e007f	res/res_monitor: Reset in/out sample counts on Monitor start ... When repeatedly starting/stopping a Monitor on a channel, the accumulated in/out sample counts are never reset to 0. This can cause inadvertent jumps in the recordings, as the code in the channel core will determine incorrectly that a jump in the recorded file position should occur. Setting the sample counts to 0 simply reflects the initial state a Monitor should be in when it is started, as this is the initial count that would be on the channels at that time. ASTERISK-24573 #close Reported by: Nuno Borges patches: 24573.patch uploaded by Nuno Borges (License 6116) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429031 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	c02beb1097	apps/app_meetme: Apply default values on initial load with no config file ... When the app_meetme module is loaded without its configuration file, the module settings aren't initialized. In particular, this impacts the use of logging realtime members. This patch guarantees that we always set the default module settings on initial load. Review: https://reviewboard.asterisk.org/r/4242/ ASTERISK-24572 #close Reported by: Nuno Borges patches: 24572.patch uploaded by Nuno Borges (License 6116) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429027 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	4a054379de	apps/app_voicemail: Fix crash with IMAP when streams are opened simultaneously ... The UW IMAP library is instrinsically not thread-safe, and relies upon higher level applications to guarantee thread safety. For the most part, this is provided by the vms object, which provides locking for individual streams. Unfortunately, this is not sufficient for calls to mail_open which create the IMAP stream. mail_open can, on some systems, call into a UW IMAP specific function for determining the address of a system based on a hostname, ip_nametoaddr. In the ip6_unix implementation of this function, static variables are used to hold parsing buffers. This can cause a crash if multiple threads attempt to convert a hostname to an address at the same time. Locking on a single mail stream is not sufficient to prevent simultaneous access to these static variables. In the IMAP library, this function can be called from the mail_open and imap_status functions. As the imap_status function is not used by app_voicemail, locking on access to mail_open is sufficient to prevent any mangling of the buffers. Review: https://reviewboard.asterisk.org/r/4188/ ASTERISK-24516 #close Reported by: David Duncan Ross Palmer Tested by: David Duncan Ross Palmer patches: ASTERISK-24516.diff uploaded by David Duncan Ross Palmer (License 6660) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428863 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	a6ed286a9a	pbx/pbx_loopback: Speed up switches by avoiding unneeded lookups ... This patch makes a small rearrangement to only do dialplan lookups during loopback switches if the pattern matches. Prior to this patch, the dialplan lookups were always performed, even when the result would be discarded. Dialplan lookups can be very costly if remote switches - like DUNDi - are present. In those cases extension matching is sped up considerably, making the issue of lost digits more manageable. As collateral damage, 6 trailing spaces were killed. Review: https://reviewboard.asterisk.org/r/4211 ASTERISK-24577 #close Reported by: Birger Harzenetter patches: ast-loopback.patch uploaded by Birger Harzenetter (License 5870) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	d2d6a36bc8	app_record: Fix bug where using the 'k' option and hanging up would trim 1/4 of a second of the recording. ... The Record dialplan function trims 1/4 of a second from the end of recordings in case they are terminated because of DTMF. When hanging up, however, you don't want this to happen. This change makes it so on hangup this does not occur. ASTERISK-24530 #close Reported by: Ben Smithurst patches: app_record_v2.diff submitted by Ben Smithurst (license 6529) Review: https://reviewboard.asterisk.org/r/4201/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428653 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	14d534a73a	manager: Fix could not extend string messages. ... When shutting down Asterisk that has an active AMI connection, you get several "failed to extend from %d to %d" messages because use of the EVENT_FLAG_SHUTDOWN attempts to add all AMI permission strings to the event. * Created MAX_AUTH_PERM_STRING to use when creating stack based struct ast_str variables used with the authority_to_str() and user_authority_to_str() functions instead of a variety of magic numbers that could be too small. * Added a special check for EVENT_FLAG_SHUTDOWN to authority_to_str() so it will not attempt to add all permission level strings. Review: https://reviewboard.asterisk.org/r/4200/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428570 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	4035e91fb6	Fix error with mixed address family ACLs. ... Prior to this commit, the address family of the first item in an ACL was used to compare all incoming traffic. This could lead to traffic of other IP address families bypassing ACLs. ASTERISK-24469 #close Reported by Matt Jordan Patches: ASTERISK-24469-11.diff uploaded by Matt Jordan (License #6283) AST-2014-012 ........ Merged revisions 428402 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	31d0fc0ecc	AST-2014-018 - func_db: DB Dialplan function permission escalation via AMI. ... The DB dialplan function when executed from an external protocol (for instance AMI), could result in a privilege escalation. Asterisk now inhibits the DB function from being executed from an external interface if the live_dangerously option is set to no. ASTERISK-24534 Reported by: Gareth Palmer patches: submitted by Gareth Palmer (license 5169) ........ Merged revisions 428331 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428363 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	060ced4b54	AST-2014-017 - app_confbridge: permission escalation/ class authorization. ... Confbridge dialplan function permission escalation via AMI and inappropriate class authorization on the ConfbridgeStartRecord action. The CONFBRIDGE dialplan function when executed from an external protocol (for instance AMI), could result in a privilege escalation. Also, the AMI action “ConfbridgeStartRecord” could also be used to execute arbitrary system commands without first checking for system access. Asterisk now inhibits the CONFBRIDGE function from being executed from an external interface if the live_dangerously option is set to no. Also, the “ConfbridgeStartRecord” AMI action is now only allowed to execute under a user with system level access. ASTERISK-24490 Reported by: Gareth Palmer git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428332 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	ce20eaa076	AST-2014-014: Fix race condition where channels may get stuck in ConfBridge under load. ... Under load it was possible for the bridging API, and thus ConfBridge, to get channels that may have hung up stuck in it. This is because handling of state transitions for a bridged channel within a bridge was not protected and simply set the new state without regard to the existing state. If the existing state had been hung up this would get overwritten. This change adds locking to protect changing of the state and also takes into consideration the existing state. ASTERISK-24440 #close Reported by: Ben Klang Review: https://reviewboard.asterisk.org/r/4173/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428299 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	094eeade6e	ast_str: Fix improper member access to struct ast_str members. ... Accessing members of struct ast_str outside of the string manipulation API routines is invalid since struct ast_str is supposed to be treated as opaque. Review: https://reviewboard.asterisk.org/r/4194/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428244 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	b3e0d05aad	chan_sip: Fix theoretical leak of p->refer. ... If transmit_refer is called when p->refer is already allocated, it leaks the previous allocation. Updated code to always free previous allocation during a new allocation. Also instead of checking if we have a previous allocation, always create a clean record. ASTERISK-15242 #close Reported by: David Woolley Review: https://reviewboard.asterisk.org/r/4160/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428117 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	f20ddb1285	apps/app_confbridge: Ensure 'normal' users hear message when last marked leaves ... When r428077 was made for ASTERISK-24522, it failed to take into account users who are neither wait_marked nor end_marked. These users are *also* supposed to hear the 'leader has left the conference' message. Granted, this behaviour is a bit odd; however, that is how it used to work... and behaviour changes are not good. This patch ensures that if there are any 'normal' users present when the last marked user leaves the conference, the message will still be played to them. Note that this regression was caught by the Asterisk Test Suite's confbridge_nominal test, which has a quirky combination of users. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428113 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	10d242b728	app_confbridge: Don't play leader leaving prompt if no one will hear it ... Consider the following: - A marked user in a conference - One or more end_marked only users in the conference When the marked users leaves, we will be in the conf_state_multi_marked state. This currently will traverse the users, kicking out any who have the end_marked flags. When they are kicked, a full ast_bridge_remove is immediately called on the channels. At this time, we also unilaterally set the need_prompt flag. When the need_prompt flag is set, we then playback a sound to the bridge informing everyone that the leader has left; however, no one is left in the bridge. This causes some odd behaviour for the end_marked users - they are stuck waiting for the bridge to be unlocked. This results in them waiting for 5 or 6 seconds of dead air before hearing that they've been kicked. Unfortunately, we do have to keep the bridge locked while we're playing back the 'leader-has-left' prompt. If there are any wait_marked users in the conference, this behaviour can't be easily changed - but we do make the case of the end_marked users better with this patch. Review: https://reviewboard.asterisk.org/r/4184/ ASTERISK-24522 #close Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428077 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	2f27faa037	cel/cel_odbc: Provide microsecond precision in 'eventtime' column when possible ... This patch adds microsecond precision when inserting a CEL record into a table with an "eventtime" column of type timestamp, instead of second precision. The documentation (configs/cel_odbc.conf.sample) was already saying that the eventtime column included microseconds precision, but that was not the case. Also, without this patch, if you had a table with an "eventtime" column of type varchar, you had millisecond precision. With this patch, you also get microsecond precision in this case. Review: https://reviewboard.asterisk.org/r/3980 ASTERISK-24283 #close Reported by: Etienne Lessard patches: cel_odbc_time_precision.patch uploaded by Etienne Lessard (License 6394) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	d0495f4139	stun: correct attribute string padding to match rfc ... When sending the USERNAME attribute in an RTP STUN response, the implementation in append_attr_string passed the actual length, instead of padding it up to a multiple of four bytes as required by the RFC 3489. This change adds separate variables for the string and padded attributed lengths, and performs padding correctly. Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/4139/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427874 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	093db340b1	app_confbridge: Play "leader has left" sound even when musiconhold is enabled. ... Currently if the leader of a conference bridge leaves any participant that has musiconhold enabled will not hear the "leader has left" sound. This is because musiconhold is started and THEN the sound is played. This change makes it so that the sound is played and THEN musiconhold is started. This provides a better experience for users as they may not have known previously why they went back to musiconhold. Review: https://reviewboard.asterisk.org/r/4177/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427844 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	329d09913e	pbx: Fix off-nominal case where a freed extension may still be used. ... If during the operation of adding an extension a priority is added but fails it is possible for the extension to be freed but still exist in the PBX core. If this occurs subsequent lookups may try to access the extension and end up in freed memory. This change removes the extension from the PBX core when the priority addition fails and then frees the extension. ASTERISK-24444 #close Reported by: Leandro Dardini Review: https://reviewboard.asterisk.org/r/4162/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427709 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	e55b96ad57	Fix compiler error when using ./configure --enable-dev-mode --enable-coverage ... When DONT_OPTIMIZE is enabled with dev-mode, it causes a shadow compilation to be done with output to /dev/null. This can cause errors with coverage when GCC attempts to write to /dev/null.gcno. This change disables coverage for the shadow compilation. ASTERISK-24502 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4151/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427682 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	0e37018d93	manager: Fix HTTP connection reference leaks. ... Fix reference leak that happens if (session && !blastaway). ASTERISK-24505 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4153/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427641 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	cc46ce38bc	configs/features.conf: Add documentation noting potential chan_agent conflict ... In chan_agent, a '*' is used by default to terminate a bridge with a caller. This can lead to all sorts of problems if '*' is used by a feature in features.conf, as the chan_agent disconnect '*' may be detected first. This patch adds a documentation snippet to features.conf so that users who attempt to use features with agents know of the potential conflict. ASTERISK-20402 #close Reported by: Matt Riddell patches: features.conf.diff uploaded by Matt Riddell (License 5023) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427617 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	938f6d3bc0	channels/chan_mgcp: Fix regression which causes gateways to be skipped ... In r227276, a while loop was turned into a for loop. Unfortunately, a portion of the while loop was left in the code such that, when a static gateway is encountered in the list of MGCP gateways, the next gateway would be skipped. At best, we would simply flip past a gateway; at worst, this could lead to a crash. ASTERISK-24500 #close Reported by: Xavier Hienne patches: chan_mgcp.patch uploaded by Xavier Hienne (License 6657) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427613 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	80439dac04	addons/chan_mobile: Increase buffer size of UCS2 encoded SMS messages ... When UCS2 character encoding is used, one symbol in national language can be expanded to 4 bytes. The current buffer used for receiving message in do_monitor_phone is 256 bytes, which is not large enough for incoming messages. For example: * AT+CMGR phone response prefix '+CMGR: "REC UNREAD","+7**********",,"14/10/29,13:31:39+12"\r\n' - 60 bytes * SMS body with UCS2 encoding (max) - 280 bytes * AT+CMGR phone response suffix '\r\n\r\nOK\r\n' - 8 bytes * Terminating null character - 1 byte This results in a needed buffer size of 349 bytes. Hence, this patch opts for a 350 byte buffer. ASTERISK-24468 #close Reported by: Dmitriy Bubnov patches: chan_mobile-1_8.diff uploaded by Dmitriy Bubnov (License 6651) chan_mobile-trunk.diff uploaded by Dmitry Bubnov (License 6651) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427607 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	ed7dabef46	chan_console: Fix reference leaks to pvt. ... Fix a bunch of calls to get_active_pvt where the reference is never released. ASTERISK-24504 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4152/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	8745e12323	main/file.c: fix possible extra ast_module_unref to format modules. ... fn_wrapper only adds a reference to the format's module if the file was able to be opened. If not this causes an unmatched ast_module_unref in filestream_destructor. Move ast_module_ref to get_stream. ASTERISK-24492 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	e9f3480121	Fix unintential memory retention in stringfields. ... * Fix missing / unreachable calls to __ast_string_field_release_active. * Reset pool->used to zero when the current pool->active reaches zero. ASTERISK-24307 #close Reported by: Etienne Lessard Tested by: ibercom, Etienne Lessard Review: https://reviewboard.asterisk.org/r/4114/ ........ Merged revisions 427380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	48f329bfe8	test_strings: Remove string tests that exercise asserts. ... Since unit tests are run with DO_CRASH, those tests were causing the test to fail. Tested-by: George Joseph git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427354 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	03fcc1ad72	config: Make text_file_save and 'dialplan save' escape semicolons in values. ... When a config file is read, an unescaped semicolon signals comments which are stripped from the value before it's stored. Escaped semicolons are then unescaped and become part of the value. Both of these behaviors are normal and expected. When the config is serialized either by 'dialplan save' or AMI/UpdateConfig however, the now unescaped semicolons are written as-is. If you actually reload the file just saved, the unescaped semicolons are now treated as start of comments. Since true comments are stripped on read, any semicolons in ast_variable.value must have been escaped originally. This patch re-escapes semicolons in ast_variable.values before they're written to file either by 'dialplan save' or config/ast_config_text_file_save which is called by AMI/UpdateConfig. I also fixed a few pre-existing formatting issues nearby in pbx_config.c Tested-by: George Joseph ASTERISK-20127 #close Review: https://reviewboard.asterisk.org/r/4132/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	47ee18acc1	Fix compile error caused by review 4138 ... There is no procedure called ast_closeframe, fix code to use ast_closestream. Reported By: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427087 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	9dc2f92921	Fix ast_writestream leaks ... Fix cleanup in __ast_play_and_record where others[x] may be leaked. This was caught where prepend != NULL && outmsg != NULL, once realfile[x] == NULL any further others[x] would be leaked. A cleanup block was also added for prepend != NULL && outmsg == NULL. 11+: Fix leak of ast_writestream recording_fs in app_voicemail:leave_voicemail. ASTERISK-24476 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4138/ ........ Merged revisions 427023 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427024 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	0b55748232	func_jitterbuffer: fix frame leaks. ... Fix code paths where it is possible for frames to leak. Fix uninitialized variable in jb_get_fixed and jb_get_adaptive. ASTERISK-22409 #related Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4128/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427019 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Tzafrir Cohen	72bf6d5052	Fix syntax from commit r426927 ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426931 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Tzafrir Cohen	4a313981f1	install init.d files on GNU/kFreeBSD ... Review: https://reviewboard.asterisk.org/r/4118/ ........ Merged revisions 426926 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426927 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	9da14f75e0	channels/sip/reqresp_parser: Fix unit tests for r426594 ... When r426594 was made, it did not take into account a unit test that verified that the function properly populated the unsupported buffer. The function would previously memset the buffer if it detected it had any contents; since this function can now be called iteratively on successive headers, the unit tests would now fail. This patch updates the unit tests to reset the buffer themselves between successive calls, and updates the documentation of the function to note that this is now required. ........ Merged revisions 426858 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426860 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	2716e17f51	REF_DEBUG: Install refcounter.py to $(ASTDATADIR)/scripts ... This change ensures refcounter.py is installed to a place where it can be found by the Asterisk testsuite if REF_DEBUG is enabled. ASTERISK-24432 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4094/ ........ Merged revisions 426830 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426831 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	a3ec9d8f1b	app_queue: fix a couple leaks to struct call_queue in set_member_value ... set_member_value has a couple leaks to references in the variable q found through testsuite tests/queues/set_penalty. Also remove the REF_DEBUG_ONLY_QUEUES compiler declaration, this is no longer possible with the updated REF_DEBUG code. ASTERISK-24466 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4125/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426805 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	15f16e3187	app_voicemail: Fix unchecked bounds of myArray in IMAP_STORAGE. ... In update_messages_by_imapuser(), messages were appended to a finite array which resulted in a crash when an IMAP mailbox contained more than 256 entries. This memory is now dynamically increased as needed. Observe that this patch adds a bunch of XXX's to questionable code. See the review (url below) for more information. ASTERISK-24190 #close Reported by: Nick Adams Tested by: Nick Adams Review: https://reviewboard.asterisk.org/r/4126/ ........ Merged revisions 426691 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Igor Goncharovskiy	865aa54aac	Add additional checks for NULL pointers to fix several crashes reported. ...	11 years ago