22011 Commits (fd9f2351cdbe1be6f4c47ab2fdbd6c3634afb91b)

 

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Author	SHA1	Message	Date
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
Kinsey Moore	fea6466555	Ensure Asterisk acknowledges ACKs to 4xx on Replaces errors ... Asterisk was not setting pendinginvite in the upper half of handle_request_invite such that the 4xx was retransmitted repeatedly even though an ack was received for every retransmission. (closes issue ASTERISK-19303) Reported by: Jon Tsiros Patches: fix-19303.patch uploaded by Jeremiah Gowdy (license 6358) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358115 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	d9961b2768	Fix unused-but-set-variable warnings ... All of these were pretty obviously unused. Some were unused because the code that used them was #if 0'd. In those cases, I just commented out the unused-but-set variables. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	7495f69c95	Correct some set-but-unused variable warnings in the mISDN library. ... (from kpfleming's commit to trunk r356292) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358011 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	b686d4785f	Make chan_usbradio compile under dev mode ... x=++x and x=x=1? Really? git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357986 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	f1277cc0e0	Fix case-sensitivity for device-specific event subscriptions and CCSS ... This change fixes case-sensitivity for device-specific subscriptions such that the technology identifier is case-insensitive while the remainder of the device string is still case-sensitive. This should also preserve the original case of the device string as passed in to the event system. CCSS is the only feature affected as it is the only consumer of device-specific event subscriptions. The second part of this patch addresses similar case-sensitivity issues within CCSS itself that prevented it from functioning correctly after the fix to the events system. This adds a unit test to verify that the event system works as expected. (closes issue ASTERISK-19422) Review: https://reviewboard.asterisk.org/r/1780/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357940 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	a36c4234a4	Remove ISDN hold restriction for non-bridged calls. ... The check if an ISDN call is bridged before it could be placed on hold is not necessary and is overly restrictive. The check was originally done to prevent problems with call transfers in case a user tried to transfer a call connected to an application to another call connected to an application. The ISDN transfer code has not required this restriction for quite some time because ECT could transfer any two active calls to each other. * Remove ISDN hold restriction for calls connected to applications. * Made ast_waitfordigit_full() ignore AST_CONTROL_HOLD and AST_CONTROL_UNHOLD instead of generating a warning message. (closes issue ASTERISK-19388) Reported by: Birger Harzenetter Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	b3fb9153dd	The default value for mohinterpret is the empty string, so when resetting to ... default values don't explicitly set the value to "default." git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357811 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	2c3804e06e	Fix channel reference leak in ChanSpy. ... * Fix next_channel() channel reference leak in ChanSpy. (closes issue ASTERISK-19461) Reported by: Irontec Patches: app_chanspy_iteartor_next_unref.patch (license #6213) patch uploaded by Irontec (issue ASTERISK-17515) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	ed7570328a	Fix race condition that can cause important control frames (such as a hangup) to be missed. ... This takes two actions. 1. Move the reading of the alertpipe in __ast_read() to immediately before the removal of frames from the readq. This means we won't do something silly like read from the alertpipe, then ignore the fact that there's a frame to get from the readq since channel's fdno is the AST_TIMING_FD. 2. When ast_settimeout() sets the rate to 0 and the timingfunc to NULL, if the channel's fdno is the AST_TIMING_FD, then set the fdno to -1. This is because if the rate is 0 and the timingfunc is NULL, it means that the channel's timing fd is being invalidated, so any pending reads should not occur. This may actually solve more issues than the referenced one below, but it's not known at this time for sure. (closes issue ASTERISK-19223) reported by Frank-Michael Wittig Review: https://reviewboard.asterisk.org/r/1779 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357761 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	496d1009dd	Second attempt to get optimal translation paths when codec_resample is used. ... This borrows code heavily from changes made in translation code in Asterisk 10. This uses the quality and sample rate change of translation in order to pick paths rather than the computational cost of translations. Computational cost is used solely in determining if a single translation step from a specific translator is better than the same translation step provided by a different translator. (closes issue ASTERISK-16821) reported by Andrew Lindh Review: https://reviewboard.asterisk.org/r/1772 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357760 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	8417ceb751	Prevent outbound SIP NOTIFY packets from displaying a port of 0 ... In the change from 1.6.2 to 1.8, ast_sockaddr was introduced which changed the behavior of ast_find_ourip such that port number was wiped out. This caused the port in internip (which is used for Contact and Call-ID on NOTIFYs) to be 0. This change causes ast_find_ourip to be port-preserving again. (closes issue ASTERISK-19430) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357665 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	99a080671d	Fix copying of CDR(accountcode) to local channels. ... In r203638, during the addition of the Channel Event Logging, in mid-2009, this got broken in trunk and ended up in asterisk 1.8 and higher. This fixes so the CDR(accountcode) from the calling channel is available to dialed channels again as well as showing up properly in the CDR's. (closes issue ASTERISK-19384) Reported by: jamicque Patches: accountcode.patch (License #6033) by jamicque Review: https://reviewboard.asterisk.org/r/1775/ Reviewed by: Richard Mudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357575 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	cc470930ba	Adding transport=udp to sample sip.conf - Also changes version of Asterisk 1.8 in UPGRADE ... (issue ASTERISK-19352) Reported by: jamicque Patches: asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	27ce64075f	Add additional character type types to supported data types for cdr_adaptive_odbc ... The reporter was uable to use varchar utf8_unicode_ci with cdr_adaptive_odbc, so this patch adds those along with some other character types to the list of types cdr_adaptive_odbc will work using the varchar conditions. The problem wasn't really UTF8 characters as much as it was a failure to respond to the exact type that was declared/in use on that database. (closes issue ASTERISK-19334) Reported By: Igor Nikolaev Patches: cdr_adaptive_odbc.patch uploaded by Igor Nikolaev (license 6236) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357455 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	e289e9caf9	Correctly reset the dialplan priority. ... When the stack frame is allocated, we save the address to which we should return, when the Gosub returns. However, if we just want to restore the priority, then we need to subtract 1 before setting it. Otherwise, when a Gosub goes to a nonexistent address, it will skip a priority in the dialplan. This is because when we return from an application, the PBX increments the priority for us. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	ec57a80169	Use more reasonable cause code when rejecting incoming call waiting calls. ... (closes issue ASTERISK-19397) Reported by: Birger Harzenetter Patches: nochannel-cause.patch (license #5870) patch uploaded by Birger Harzenetter git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357407 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	0472ec7ba7	Moves UPGRADE.txt notes from r357356 to a new section specific to 1.8.12 ... (issue ASTERISK-19352) reported by: jamicque git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357386 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	60907f7df1	Adds UPGRADE.txt notes to r357266 indicating changes to transport option ... (issue ASTERISK-19352) Reported by: jamicque git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357356 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	00fc360507	Remove dupliate 'i' option table entry in app_page.c. ... (closes issue ASTERISK-19310) Reported by: Makoto Dei Patches: app_page-duplicate-i-option.patch (license #5027) patch uploaded by Makoto Dei git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357352 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	52c50e4da7	Changes transport option in sip.conf so that using multiple instances doesn't stack. ... Prior to this patch, Using "transport=" multiple times would cause them to add to one another like allow/deny. This patch changes that behavior to simply use the transport option specified last. Also, if no transport option is applied now, the default will automatically be UDP. (closes ASTERISK-19352) Reported by: jamicque Patches: asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026) issueA19352_no_transport_is_udp.patch uploaded by Walter Doekes (license 5674) Review: https://reviewboard.asterisk.org/r/1745/diff/#index_header git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	f3fbe7d88f	Make COMPILE_DOUBLE magic actually work. ... The build system has some special magic to ensure that if Asterisk is built with --enable-dev-mode *and* DONT_OPTIMIZE, that all the source is still compiled with the optimizer enabled (even though the result will be thrown away), because the compiler is able to find a great deal of coding errors and bugs as a result of running its optimizers. Unfortunately at some point this mode got broken, and the 'throwaway' compile of the code was no longer done with the optimizer enabled. This patch corrects that problem. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357212 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	100721d217	Fix callerid of Originated calls. ... Thanks to Matt Riddell for tracking this down. (closes issue ASTERISK-19385) Reported by: ornix git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@357093 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	c026fb96b0	Copy CDR variables when set during a bridge ... This patch makes sure amaflags, accountcode, and userfield get copied to the bridge CDR when set during a bridge (like via a custom feature). (closes issue ASTERISK-16990) Review: https://reviewboard.asterisk.org/r/1721/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@356963 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago