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Author	SHA1	Message	Date
Russell Bryant	dea36a06bb	app_page: Fix a memory leak on every Page(). ... dial_list is a dynamically allocated array that is allocated at the beginning of Page() based on how many devices will be dialed. This was never being freed. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360363 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	093121050f	app_jack: fix datastore memory leak in error handling path. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360360 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	2b2c5f27fc	Rebuild parsers. ... This is needed to include the last fix to main/ast_expr2.y. The changes look much bigger as this regeneration of the code was done with newer versions of flex and bison. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360357 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	17cd5abb22	expression parser: Fix (theoretical) memory leak. ... Fix a memory leak that is very unlikely to actually happen. If a malloc() succeeded, but the following strdup() failed, the memory from the original malloc() would be leaked. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360356 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	c04afdbc70	Make number not available presentation also set screening to network provided. ... Q.951 indicates that when the presentation indicator is "Number not available due to interworking" for a number then the screening indicator field should be "Network provided". * Made ast_party_id_presentation() return AST_PRES_NUMBER_NOT_AVAILABLE when the presentation is "Number not available due to interworking". This fix makes Asterisk consistent and it also makes it consistent with earlier branches as far as this presentation value is concerned. * Made pri_to_ast_presentation() and ast_to_pri_presentation() conversions handle the "Number not available due to interworking" case better in sig_pri.c. This change is possible because the minimum required libpri version (v1.4.11) has the necessary defines in libpri.h. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360309 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	f9246f83c9	Add missing initialization of update_redirecting in chan_sip.c ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	94b6251ea2	Update install_prereq script to include missing GSM library for debian amd move SQLite3. ... (closes issue ASTERISK-19367) Reported by: Andrew Latham Patches: debian_install_prereq.diff uploaded by Andrew Latham (license 5985) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360138 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tzafrir Cohen	00dc354511	Also detect gmime 2.6 ... Also detect gmime version 2.6 (Michael Biebl) Signed-off-by: Tzafrir Cohen (License #5035) <tzafrir.cohen@xorcom.com> git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360087 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	cfdc12387b	Ensure Asterisk sends a BYE when pending on the final response to a re-INVITE ... When Asterisk detects a hangup and cannot send a BYE due to a pending INVITE, it sets the pendingbye flag and waits for the final response to that INVITE. When the response is received, it transmits the BYE. If, however, that INVITE request is a pending re-INVITE, it needs to first send a CANCEL request to terminate the pending re-INVITE. In that circumstance, Asterisk was, in some scenarios, clearing the pendingbye flag after processing the CANCEL request and not checking for a pending BYE when receiving the final 487 response to the INVITE. This patch ensures that if the pendingbye flag is set, it is honored regardless of the nature of the INVITE request currently in flight. (closes issue ASTERISK-19365) Reported by: Thomas Arimont Tested by: Thomas Arimont Patches: bugASTERISK-19365_2012_03_08.patch uploaded by mjordan (license 6283) Review: https://reviewboard.asterisk.org/r/1807 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360086 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	a25e2ad59c	Prevent Echo() from relaying control, null, and modem frames ... Echo()'s description states that it echoes audio, video, and DTMF except for # while it actually echoes any frame that it receives other than DTMF #. This was causing frame storms in the test suite in some circumstances where Echo() was attached to both ends of a pair of local channels and control frames were being periodically generated. Echo()'s behavior and description have been modifed so that it only echoes media and non-# DTMF frames. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@360033 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	8dc6345d0f	Allow AMI action callback to be reentrant. ... Fix AMI module reload deadlock regression from ASTERISK-18479 when it tried to fix the race between calling an AMI action callback and unregistering that action. Refixes ASTERISK-13784 broken by ASTERISK-17785 change. Locking the ao2 object guaranteed that there were no active callbacks that mattered when ast_manager_unregister() was called. Unfortunately, this causes the deadlock situation. The patch stops locking the ao2 object to allow multiple threads to invoke the callback re-entrantly. There is no way to guarantee a module unload will not crash because of an active callback. The code attempts to minimize the chance with the registered flag and the maximum 5 second delay before ast_manager_unregister() returns. The trunk version of the patch changes the API to fix the race condition correctly to prevent the module code from unloading from memory while an action callback is active. * Don't hold the lock while calling the AMI action callback. (closes issue ASTERISK-19487) Reported by: Philippe Lindheimer Review: https://reviewboard.asterisk.org/r/1818/ Review: https://reviewboard.asterisk.org/r/1820/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359979 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	698cd736b6	Prevent chanspy from binding to zombie channels ... This patch addresses a bug with chanspy on local channels which roughly 50% of the time would create a situation where chanspy can latch onto a zombie channel, keeping the zombie alive forever and causing the channel doing the spying to never be able to hang up. (closes issue ASTERISK-19493) Reported by: lvl Review: https://reviewboard.asterisk.org/r/1819/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359892 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Alec L Davis	1ded276c97	Missed lastinvite CSeq int to uint32_t change ... from Review: https://reviewboard.asterisk.org/r/1699/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	4666efb849	Fix remotely exploitable stack overflow in HTTP manager ... There exists a remotely exploitable stack buffer overflow in HTTP digest authentication handling in Asterisk. The particular method in question is only utilized by HTTP AMI. When parsing the digest information, the length of the string is not checked when it is copied into temporary buffers allocated on the stack. This patch fixes this behavior by parsing out pre-defined key/value pairs and avoiding unnecessary copies to the stack. (closes issue ASTERISK-19542) Reported by: Russell Bryant Tested by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359706 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	2073ca37b6	Fix remotely exploitable stack overrun in Milliwatt ... Milliwatt is vulnerable to a remotely exploitable stack overrun when using the 'o' option. This occurs due to the milliwatt_generate function not accounting for AST_FRIENDLY_OFFSET when calculating the maximum number of samples it can put in the output buffer. This patch resolves this issue by taking into account AST_FRIENDLY_OFFSET when determining the maximum number of samples allowed. Note that at no point is remote code execution possible. The data that is written into the buffer is the pre-defined Milliwatt data, and not custom data. (closes issue ASTERISK-19541) Reported by: Russell Bryant Tested by: Matt Jordan Patches: milliwatt_stack_overrun.rev1.txt by Russell Bryant (license 6283) Note that this patch was written by Russell, even though Matt uploaded it ........ Merged revisions 359645 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359656 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	e446657808	Add missing connected line macro calls to initial dial for Dial and Queue apps. ... The connected line interception macros do not get executed when the outgoing channel is initially created and that channel's caller-id is implicitly imported into the incoming channel's connected line data. If you are using the interception macros, you would expect that they get run for every change to a channel's connected line information outside of normal dialplan execution. Review: https://reviewboard.asterisk.org/r/1817/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359609 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	167dd53f80	chan_iax2: Fix use of uninitialized sockaddr_in in try_transfer(). ... Initialize a struct sockaddr_in in try_transfer() so that the code isn't (potentially) trying to read from it while uninitialized. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359558 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	ba3230cc45	chan_gtalk: Fix use of uninitialized vars in config handling. ... Fix potential use of context, parkinglot, and prefs before they are initialized. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359534 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	e470c22ed6	chan_gtalk: Fix potential use of uninitialized variable. ... Avoid potential use of idroster in gtalk_alloc() before it has been initialized. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359508 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	3627d4c840	app_chanisavail: Fix use of uninitialized variable. ... Ensure that status is set before it is used by resetting it during each loop iteration. This could have resulted in incorrect results from this app. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359486 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	d9c2ea6d15	udptl: Ensure fec[] in udptl_build_packet() is initialized. ... Scan results indicated that this array could be used uninitialized. At a quick look, it looks correct. In any case, initializing it is a Good Thing (tm). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359457 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	d500229f3c	app.h: Always initialize AST_DECLARE_APP_ARGS(). ... This patch ensures that the struct defined by AST_DECLARE_APP_ARGS() is always fully initialized. I'm not sure if this fixes any real bugs, but it silences a bunch of warnings from coverity, and is generally a good thing to do anyway. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359452 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	5b61050209	Fix deadlock potential with some ast_indicate/ast_indicate_data calls. ... Calling ast_indicate()/ast_indicate_data() with the channel lock held can result in a deadlock with a local channel because of how local channels need to avoid deadlock. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359451 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	6625f0a50c	Fix incorrect jitter buffer overflow due to missed resynchronizations ... When a change in time occurs, such that the timestamps associated with frames being placed into an adaptive jitter buffer (implemented in jitterbuf.c) are significantly different then the previously inserted frames, the jitter buffer checks to see if it needs to be resynched to the new time frame. If three consecutive packets break the threshold, the jitter buffer resynchs itself to the new timestamps. This currently only occurs when history is calculated, and hence only on JB_TYPE_VOICE frames. JB_TYPE_CONTROL frames, on the other hand, are never passed to the history calculations. Because of this, if the jump in time is greater then the maximum allowed length of the jitter buffer, the JB_TYPE_CONTROL frames are dropped and no resynchronization occurs. Alterntively, if the overfill logic is not triggered, the JB_TYPE_CONTROL frame will be placed into the buffer, but with a time reference that is not applicable. Subsequent JB_TYPE_VOICE frames will quickly trigger the overflow logic until reads from the jitter buffer reach the errant JB_TYPE_CONTROL frame. This patch allows JB_TYPE_CONTROL frames to resynch the jitter buffer. As JB_TYPE_CONTROL frames are unlikely to occur in multiples, it perform the resynchronization on any JB_TYPE_CONTROL frame that breaks the resynch threshold. Note that this only impacts chan_iax2, as other consumers of the adaptive jitter buffer use the abstract jitter buffer API, which does not use JB_TYPE_CONTROL frames. Review: https://reviewboard.asterisk.org/r/1814/ (closes issue ASTERISK-18964) Reported by: Kris Shaw Tested by: Kris Shaw, Matt Jordan Patches: jitterbuffer-2012-2-26.diff uploaded by Kris Shaw (license 5722) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359356 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	7141a29e1b	Fix Dial m and r options and forked calls generating warnings for voice frames. ... When connected line support was added, the wait_for_answer() variable single changed its meaning slightly. Unfortunately, the places where single was used did not necessarily get updated to reflect that change. Also audio/video frames were sent to all forked calls when the endpoints were never made compatible. * Don't pass audio/video media frames when the channels have not been made compatible. * Added handling of AST_CONTROL_SRCCHANGE to app_dial.c. * Fixed app_dial.c passing on AST_CONTROL_HOLD because that frame can also pass a requested MOH class. (closes issue ASTERISK-16901) Reported by: Chris Gentle (closes issue ASTERISK-17541) Reported by: clint Review: https://reviewboard.asterisk.org/r/1805/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359344 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	5a87fa5de9	Fix bogus reads/writes of console log levels in asterisk.c ... This patch updates the NUMLOGLEVELS define in logger.h to 32, to match the fact that logger.c implements 32 log levels (because of the custom log level stuff). asterisk.c uses this define to size an array of levels per remote console. This array is modified in ast_console_toggle_loglevel(), which is called by the "logger set level" CLI command. While the documentation for the CLI command doesn't make it terribly obvious, you can use this CLI command to toggle a custom log level on a remote console, as well. However, doing so led to an invalid array index in asterisk.c. This array is read from any time a log message is written to a console. So, all custom log level messages resulted in a bogus read if a remote console was connected. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	08e1d392f4	Fix invalid reads/writes due to incorrect sizeof(). ... These few places in the code used sizeof() on h_addr in struct hostent. This is sizeof(char *). The correct way to get the size of this address is to use h_length. This error would result in reads/writes of 8 bytes instead of 4 on 64-bit machines. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359211 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	a66d14f2ae	Fix inaccurate sizeof() in sched.c. ... This code just needed sizeof(int), not sizeof(int *). git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	a12e5aab1a	Fix incorrect sizeof() in astman. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359116 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	7bff9be851	Fix incorrect usage of sizeof() in res_crypto. ... In this case, just remove the memset(). There was a redundant memset that is done correctly just 2 lines later. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359110 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	30d3e59447	Fix broken usage of sizeof() in res_adsi. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359088 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	48b1c5e5a6	Fix incorrect sizeof() usage in features.c. ... This didn't actually result in a bug anywhere, luckily. The only place where the result of these memcpys was used is in app_dial, and the only field that it read out of ast_call_feature was the first one, which is an int, so these memcpys always copied just enough to avoid a problem. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359069 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	c8f8fcc518	Fix incorrect sizeof() on a pointer in MD5Final(). ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359059 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	f5067300be	Don't use a buffer after it goes out of scope. ... 's' is set to 'workspace'. Make sure 'workspace' doesn't go out of scope while the reference to it via 's' is still used. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359056 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	ac53c0d94e	Dump cache of published events when a node joins the cluster. ... Also use a more reliable method for stopping the poll() thread. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359053 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	0da9d71905	Remove chan_usbradio and app_rpt. ... These modules are being maintained outside of the tree and have been for a long time now, so it doesn't make sense to keep them here. Review: https://reviewboard.asterisk.org/r/1764/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@359050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	c546ff5110	Fix setting CDR variables in the hangup extension ... A previous CDR fix for setting CDR variables during a bridge via custom dialplan features broke setting CDR variables in the hangup extension. This patch fixes the issue. Review: https://reviewboard.asterisk.org/r/1794/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358978 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	7622a34c89	Make hints for invalid SIP devices return Unavail, not idle ... This patch drastically simplifies the device state aggegation code. The old method was not only overly complex, but also made it impossible to return AST_DEVICE_INVALID from the aggregation code. The unit test update is as a result of fixing that bug. The SIP change stems from a bug introduced by removing a DNS lookup for hostname-based SIP channels. (closes issue ASTERISK-16702) Review: https://reviewboard.asterisk.org/r/1808/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358943 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	5ed9d4128d	Requested changes documenting the fixed AEL functionality. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358859 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	2477215dd4	Enable macros in 1.8 to find the next highest "h" extension in a context, like in 1.4. ... This change restores functionality that was present in 1.4, when AEL macros were implemented with the Macro dialplan application. Macros are fraught with functionality issues, because they consume a large portion of the underlying application stack. This limits the ability of AEL users to call many layers of subroutines, an issue which Gosub does not have (originally tested to 100,000 levels deep). Therefore, starting in 1.6.0, AEL macros were implemented with Gosub. However, there were some implicit behaviors of Macro, which were not replicated at the same time as with the transition to Gosub, one of which is documented in the related issue. In particular, the "h" extension is designed to execute not in the Macro context, but in the topmost calling context. Due to legacy issues with a misapplied bugfix many years ago, when a macro exited in 1.4, it looks in all calling contexts, bubbling up from the deepest level until it finds an "h" extension. Since AEL hides the complexity of the underlying dialplan logic from the AEL programmer, it's reasonable to assume that this behavior should not change in the transition from Asterisk 1.4 LTS to Asterisk 1.8 LTS, lest we break working AEL configurations in the transition to Asterisk 1.8 LTS. This fix is the result, which implements a search for the "h" extension in all calling Gosub contexts. Fixes ASTERISK-19336 Patch: 20120308__ael_bugfix_for_trunk__2.diff (License #5003) by Tilghman Lesher (with slight modifications for 1.8) Tested by: Johan Wilfer Review: https://reviewboard.asterisk.org/r/1776/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358810 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	8e2e2bf059	Make transfer not ignore port information with SIP. ... Attempting to transfer with SIP to an address like 1XXXXX@ip.ad.re.ss:5061 would fail because port would be cut from the host string and ignored. This simply keeps chan_sip from cutting off the port number during these kinds of transfers. (closes issue ASTERISK-19321) Reported by: Federico Alves Review: https://reviewboard.asterisk.org/r/1790/diff/#index_header git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358643 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	fff4e1ca50	Change directly setting _softhangup in sig_ss7.c to use ast_softhangup_nolock(). ... Update to: (issue ASTERISK-19372) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358530 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	acd1d86053	Return g729 and g723.1 frames with the number of samples set properly. ... If the wctc4xxp returns more than a single packet, we need to update the number of samples in the returned frame accordingly. Acked-by: Shaun Ruffell <sruffell@digium.com> git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358484 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	fe1d703804	Set snarkiness = 0 in cdr_adaptive_odbc.conf.sample ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358438 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	522372b3df	Add detection for ODBC WCHAR fields ... Without detecting these types, cel_odbc blows up when the character set for the table is utf8. This also wraps cdr_adaptive_odbc's use of those types in the HAVE_ODBC_WCHAR #ifdef seen in other parts of the code. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358435 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	0c315bb90a	Fix ring cadance setup for outgoing calls on FXS ports. ... * Fix referencing the wrong variable in chan_dahdi.c:my_set_cadence(). Thanks to Sean Bright for compiling with -Wshadow and finding this bug. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358377 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	4c9168a4c7	Drop SS7 call if not connected yet when INCOMPLETE/BUSY/CONGESTION. ... SS7 is a trunk protocol and should clear a failed call as soon as possible. * Made SS7 hangup a call immediately if it has not connected yet for INCOMPLETE/BUSY/CONGESTION causes. Otherwise, play an appropriate inband tone. (closes issue ASTERISK-19372) Reported by: Igor Nikolaev git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358278 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	e0c235bd9b	Setup DSP when SS7 call is connected or early media is available. ... Outgoing SS7 calls fail to detect incoming DTMF so any bridged channel that requires out-of-band DTMF will not work. * Added sig_ss7_open_media() calls at appropriate places in sig_ss7.c. The new call converts conditionaled out unconverted code and shows that the code really did something useful. * Improved some chan_dahdi DTMF debug messages to help track DTMF handling. (closes issue ASTERISK-19312) Reported by: Igor Nikolaev git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	47244a11d6	Eliminate double close of file descriptor in manager.c ... The process_output function in manager.c attempted to call fclose and close immediately afterwards. Since fclose implies close, this resulted in a potential double free on file descriptors. This patch changes that behavior and also adds error checking to fclose and close depending on which was deemed necessary. Also error messages. Thanks to Rosen Iliev for pointing out the location of the problem. (closes issue ASTERISK-18453) Reported By: Jaco Kroon Review: https://reviewboard.asterisk.org/r/1793/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358214 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Joshua Colp	ffa247ce6c	Defer sending the connected line reinvite if a reinvite is already in progress. ... (issue ASTERISK-19355) Reported by: tomaso (closes issue AST-825) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358162 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago