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2009-11-06Use autotagged externalsv1.4.27-rc3lmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.27-rc3@228495 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-06Importing files for 1.4.27-rc3 release.lmadsen3-0/+26709
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.27-rc3@228494 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-06Creating tag for the release of asterisk-1.4.27-rc3lmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.27-rc3@228493 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-06fixes segfault in iLBCdvossel1-0/+5
For reasons not yet known, it appears possible for an ast_frame to have a datalen greater than zero while the actual data is NULL during Packet Loss Concealment. Most codecs don't support PLC so this doesn't affect them. This patch catches the malformed frame and prevents the crash from occuring. Additional efforts to determine why it is possible for a frame to look like this are still being investigated. (issue #16979) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@228418 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-06Fix a bug caused by a partially invalid frame (from the jitterbuffer) ↵file1-5/+1
passing through the Asterisk core. (closes issue #15560) Reported by: jvandal (closes issue #15709) Reported by: covici git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@228409 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-06Properly handle '=' while decoding base64 messages and null terminate ↵mnicholson2-2/+9
strings returned from BASE64_DECODE. (closes issue #15271) Reported by: chappell Patches: base64_fix.patch uploaded by chappell (license 8) Tested by: kobaz git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@228378 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-06fixes crash in astfd.cdvossel1-4/+9
(closes issue #15981) Reported by: slavon git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@228338 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-06fixes memory leak in func_audiohookinherit.cdvossel1-0/+1
(closes issue 0015394) Reported by: boroda Patches: bug15394_memoryleak_diff2.txt uploaded by dbrooks (license 790) Tested by: dbrooks, boroda git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@228272 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-05Fix crash on VPB exception when no hardware is present.qwell1-1/+1
(closes issue #14970) Reported by: tzafrir Patches: vpb_exception.diff uploaded by tzafrir (license 46) Tested by: markwaters git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@228079 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-05chan_misdn Asterisk 1.4.27-rc2 crashdbrooks1-0/+1
Crash related to chan_misdn connection. Patch submitted by gknispel_proformatique, tested by francesco_r. "I have many crash since i have upgraded to Asterisk 1.4.27-rc2. Attached a full bt." This patch zeros out an ast_frame. (closes issue #16041) Reported by: francesco_r git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@228078 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04Fix incorrect filename comparsion after monitor file changejpeeler1-4/+40
The logic to detect if a requested file is indeed a different file from the current file was incorrect. The main issue being confusion of the use of filename_base which was previously set without pathing information and then compared to another full path. Robust file comparison logic has been added to properly check if two files are the same even if symlinks are used. (closes issue #15313) Reported by: caspy Patches: 20091103__issue15313__1.4.diff.txt uploaded by jpeeler (license 325) but mostly tilghman's work git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@227944 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04This patch modifies the Dial application to monitor the calling channel for ↵mnicholson1-13/+62
hangups while playing back announcements. (closes issue #16005) Reported by: falves11 Patches: dial-announce-hangup-fix1.diff uploaded by mnicholson (license 96) Tested by: mnicholson, falves11 Review: https://reviewboard.asterisk.org/r/407/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@227827 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04Modify the SDP parsing code to parse session and media level items separately.mnicholson1-379/+490
With the new code, media level proprieties should no longer be confused with session level proprieties. This change also reorganizes some of the SDP parsing code which should make it easier to manage in the future. (closes issue #14994) Reported by: frawd Tested by: frawd, mnicholson, file Review: https://reviewboard.asterisk.org/r/385/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@227758 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04Fix a security issue where it may be possible for someone to execute a ↵file1-417/+1922
cross-site AJAX request exploit. (AST-2009-009) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@227735 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04Fix a security issue where sending a REGISTER with a differing username in ↵file1-2/+0
the From URI and Authorization header would reveal whether it was valid or not. (AST-2009-008) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@227700 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-03Make sure the outgoing flag is cleared if a new channel fails to get created ↵rmudgett1-0/+3
for outgoing calls. This is the relevant portion of asterisk/trunk -r226648 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@227275 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-03Fix a bug where an RPID header could be generated with a blank username in ↵file1-1/+1
the URI. (closes issue #15909) Reported by: kobaz git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@227166 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-03Fixing bug before someone reports it...oej1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@227090 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-03Adding IP address in Contact ACL log message and removing redundant messageoej1-2/+1
(based on kpfleming's feedback) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@227089 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-03Use proper response code when violating Contact ACL's.oej1-2/+15
Review: https://reviewboard.asterisk.org/r/415/ Thanks kpfleming for a quick review. (EDVX-003) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@227088 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-02SIP channel name uniquenessdbrooks1-1/+3
SIP channel names were supposed to be unique by way of a name suffix derived from the pointer to the channel's private data. Uniqueness was preserved on 32-bit systems, but not on 64-bit systems. This patch, as suggested by kpfleming, replaces this suffix with a simple incremented unsigned int. (closes issue #15152) Reported by: palbrecht Review: https://reviewboard.asterisk.org/r/420/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@226972 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-02Fix a bug where the recorded privacy introduction file would not get removed ↵file1-14/+31
if the caller hung up while the called party had not yet answered. This was fixed by introducing an argument to the 'n' option which, when enabled, removes the introduction file under all scenarios. This was done to preserve the behavior that has existed for quite some time. (closes issue #14674) Reported by: ulogic Patches: bug14674.patch uploaded by jpeeler (license 325) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@226889 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-02Don't allow two separate instances of safe_asterisk when restarting from the ↵tilghman1-1/+7
init script. (closes issue #14562) Reported by: davidw Patches: Initially 20091022__issue14562.diff.txt uploaded by tilghman (license 14) Modified to 20091030__Issue14562_diff.txt uploaded by davidw (license 780) Tested by: davidw git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@226811 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-02fixes crash on iterator_destroy on uninitialized iteratordvossel1-1/+0
(closes issue #16162) Reported by: krn git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@226736 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-02changes calltoken debug messages from LOG_NOTICE to LOG_DEBUG like they are ↵dvossel1-6/+6
supposed to be (closes issue #16144) Reported by: aragon git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@226688 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-29Add an option to enabling passing music on hold start and stop requests ↵file2-3/+8
through instead of acting on them in chan_local. (closes issue #14709) Reported by: dimas git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@226531 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-28Update documentation in sip.conf.sample.lmadsen1-0/+7
Update the documentation in sip.conf.sample in order to make it more clear that directmedia/canreinvite do not cause Asterisk to ignore reINVITEs. It is only used to stop Asterisk from generating a reINVITE, but does not stop it from accepting them if necessary. (closes issue #15644) Reported by: lmadsen git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@226382 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-28Update CALLINGSUBADDR channel variable documentation.lmadsen1-1/+1
(closes issue #15734) Reported by: alecdavis Patches: channelvariables.tex.diff.txt uploaded by alecdavis (license 585) Tested by: alecdavis git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@226377 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-28Fix documentation (pointed out by TheDavidFactor on #-dev)tilghman1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@226304 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-27Manager output is not always NULL-terminated, so force a NULL at the end of ↵tilghman1-2/+6
the filestream. (closes issue #15495) Reported by: pdf Patches: 20090916__issue15495.diff.txt uploaded by tilghman (license 14) Tested by: pdf git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@226138 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-26detect ARM Linux EABI OSARCH as linux-gnu instead of linux-gnueabitzafrir3-14/+14
* Set OSARCH to linux-gnu even if host_os is linux-gnueabi * When checking if we are Linux, check OSARCH rather than host_os The newer ARM ABI ("EABI") shows the OS name 'linux-gnueabi' rather than 'linux-gnu' . This patch sets OSARCH to be 'linux-gnu' even in such a case. OSARCH is tested for the value of 'linux-gnu' in one or two places in the tree. This patch also fixes the check libcap to check for $OSARCH rather than $host_os . See also: http://wiki.debian.org/ArmEabiPort git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@225957 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-23Don't force menuselect.makeopts to be rebuilt on every build.kpfleming1-3/+26
For some reason the menuselect.makeopts file was listed as PHONY in the Makefile, resulting in 'make' needing to rebuild it for every build. This then resulted in the embedded module rules being rebuilt on every build, which can be slow and is unnecessary. This patch fixes the problem by properly allowing 'make' to know when the menuselect.makeopts file needs to be rebuilt (defining the proper dependencies). git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@225581 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-22Clean valgrind output by suppressing false errors.lmadsen2-10/+45
Update valgrind.txt documentation and add valgrind.supp file in order to allow those who are creating valgrind output to have less false errors in the logfile. (closes issue #16007) Reported by: atis Patches: valgrind.txt.diff uploaded by atis (license 242) asterisk2.supp uploaded by atis (license 242) Tested by: atis, amorsen git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@225484 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21IAX2: VNAK loop caused by signaling frames with no destination call numberdvossel1-10/+77
It is possible for the PBX thread to queue up signaling frames before a destination call number is received. This can result in signaling frames being sent out with no destination call number. Since recent versions of Asterisk require accurate destination callnumbers for all Full Frames, this can cause a VNAK loop to occur. To resolve this no signaling frames are sent until a destination callnumber is received, and destination call numbers are now only required for iax_pvt matching when the frame is an ACK. Review: https://reviewboard.asterisk.org/r/413/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@225243 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21Revert 225169, as this doesn't account for the possibility of a list of frames.russell1-6/+2
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@225171 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21Isolate the frame returned from ast_translate().russell1-2/+6
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@225169 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21Fix documentation for ast_softhangup() and correct the misuse thereof.tilghman3-5/+6
(closes issue #16103) Reported by: majorbloodnok git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@225105 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21Suffix is not needed for a matchtilghman1-16/+6
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@225103 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21IAX/SIP shrinkcallerid optiondvossel4-10/+51
The shrinking of caller id removes '(', ' ', ')', non-trailing '.', and '-' from the string. This means values such as 555.5555 and test-test result in 555555 and testtest. There are instances, such as Skype integration, where a specific value is passed via caller id that must be preserved unmodified. This patch makes the shrinking of caller id optional in chan_sip and chan_iax in order to support such cases. By default this option is on to preserve previous expected behavior. (closes issue #15940) Reported by: dimas Patches: v2-15940.patch uploaded by dimas (license 88) 15940_shrinkcallerid_trunk.c uploaded by dvossel (license 671) Tested by: dvossel Review: https://reviewboard.asterisk.org/r/408/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@225032 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-21Isolate frames returned from a DSP instance or codec translator.russell7-123/+12
The reasoning for these changes are the same as what I wrote in the commit message for rev 222878. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@224931 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-20Pay attention to the return value of the manipulate function.tilghman1-4/+12
While this looks like an optimization, it prevents a crash from occurring when used with certain audiohook callbacks (diagnosed with SVN trunk, backported to 1.4 to keep the source consistent across versions). git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@224855 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-20Add support for relaying early media in the features attended transfer option.file1-1/+5
(closes issue #14828) Reported by: licedey git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@224773 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-19Correct timestamp calculations when RTP sample rates over 8kHz are used.kpfleming1-9/+14
While testing some endpoints that support 16kHz and 32kHz sample rates, some log messages were generated due to calc_rxstamp() computing timestamps in a way that produced odd results, so this patch sanitizes the result of the computations. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@224670 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-19Do not attempt early media bridging (ie: direct RTP setup) if options are ↵file1-6/+11
enabled that should prevent it. (closes issue #14763) Reported by: cupotka git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@224565 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-17Fix stale caller id data from being reported in AMI NewChannel eventjpeeler1-2/+9
The problem here is that chan_dahdi is designed in such a way to set certain values in the dahdi_pvt only once. One of those such values is the configured caller id data in chan_dahdi.conf. For PRI, the configured caller id data could be overwritten during a call. Instead of saving the data and restoring, it was decided that for all non-analog channels it was simply best to not set the configured caller id in the first place and also clear it at the end of the call. (closes issue #15883) Reported by: jsmith git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@224330 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-16Never released PRI channels when using Busy() or Congestion() dialplan apps.rmudgett1-4/+12
When the Busy() or Congestion() application is used towards ISDN (an ISDN progress is sent), the responding ISDN Disconnect or Release may contain the ISDN cause user busy or one of the congestion causes. In chan_dahdi.c these causes will only set the needbusy or needcongestion flags and not activate the softhangup procedure. Unfortunately only the latter can interrupt the endless wait loop of Busy()/Congestion(). Result: PRI channels staying in state busy for the rest of asterisk life or until the other end times out and forces the call to clear. (in issue 0014292) Reported by: tomaso Patches: disc_rel_userbusy.patch uploaded by tomaso (license 564) (This patch is unrelated to the issue.) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@224260 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-13Fix PRI timer T309 operationjeang1-1/+1
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@223955 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-12Ensure ringing continues for branched calls after progress is receivedjpeeler1-2/+7
While waiting for an answer, don't send progress for branched calls for which ringing was sent. (closes issue #15028) Reported by: fnordian git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@223804 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-12Remove automatic switching from T.38 to voice mode in chan_sip.kpfleming1-2/+1
chan_sip has some code to automatically switch from T.38 mode to voice mode when a voice frame is written to the channel while it is in T.38 mode; this was intended to handle the situation when a FAX transmission has ended and the channel is not yet hung up, but is causing problems at the beginning of FAX sessions as well when there are still voice frames 'in flight' at the time the T.38 negotiation completes. This patch removes the automatic switchover. (issue #16025) Reported by: jamicque git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@223692 f38db490-d61c-443f-a65b-d21fe96a405b
2009-10-11Remove a duplicate ao2_iterator_destroy().russell1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@223550 f38db490-d61c-443f-a65b-d21fe96a405b
generated by cgit v1.2.3 (git 2.25.1) at 2024-06-07 19:42:45 +0000