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2011-01-17Importing release summary for 1.4.39.1 release.v1.4.39.1lmadsen2-0/+150
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.39.1@302152 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-17AST-2011-001lmadsen5-526/+20
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.39.1@302145 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-17Create 1.4.39.1 from 1.4.39.lmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.39.1@302089 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-12Importing release summary for 1.4.39 release.v1.4.39lmadsen2-0/+511
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.39@301497 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-12Update ChangeLog, .version file, and remove summary files.lmadsen4-512/+5
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.39@301496 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-12Create Asaterisk 1.4.39 from 1.4.39-rc1lmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.39@301463 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-13Use autotagged externalsv1.4.39-rc1lmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.39-rc1@298183 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-13Importing release summary for 1.4.39-rc1 release.lmadsen2-0/+511
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.39-rc1@298182 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-13Importing files for 1.4.39-rc1 release.lmadsen3-0/+30913
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.39-rc1@298181 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-13Creating tag for the release of asterisk-1.4.39-rc1lmadsen5-31424/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.39-rc1@298180 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-13Use autotagged externalslmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.39-rc1@298178 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-13Importing release summary for 1.4.39-rc1 release.lmadsen2-0/+511
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.39-rc1@298177 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-13Importing files for 1.4.39-rc1 release.lmadsen3-0/+30913
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.39-rc1@298176 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-13Creating tag for the release of asterisk-1.4.39-rc1lmadsen0-0/+0
git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.39-rc1@298175 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-09Ignore spurious REGISTER requeststwilson1-0/+7
If a REGISTER request with a Call-ID matching an existing transaction is received it was possible that the REGISTER request would overwrite the initreq of the private structure. This info is used to generate messages for other responses in the transaction. This patch ignores REGISTER requests that match non-REGISTER transactions. (closes issue #18051) Reported by: eeman Tested by: twilson Review: https://reviewboard.asterisk.org/r/1050/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@297959 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-07Revert code that changed SSRC for DTMF.jpeeler1-2/+2
Some previous behavior was attempted to be restored, but mistakingly I did not realize that the previous behavior was incorrect. This fixes DTMF not being detected since DTMF shouldn't cause the SSRC to change. (related to issue #17404) (closes issue #18189) (closes issue #18352) Reported by: marcbou Tested by: cmbaker82 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@297823 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-07Use non-deprecated APIs for CoreAudiotilghman3-18/+100
Review: https://reviewboard.asterisk.org/r/1040/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@297818 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-07Avoid a crash if we don't pass an argument to 'astobj2 test.'seanbright1-0/+4
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@297775 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-07Don't create a Local channel if the target extension does not exist.tilghman1-4/+10
(closes issue #18126) Reported by: junky Patches: followme.diff uploaded by junky (license 177) (partially restructured by me to avoid a possible memory leak) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@297689 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-06Improve handling of REGISTER requests with multiple contact headers.jpeeler1-6/+30
The changes here attempt to more strictly follow RFC 3261 section 10.3. Basically the following will now cause a 400 Bad Response to be returned, if: - multiple Contact headers are present with one set to expire all bindings ("*") - wildcard parameter is specified for Contact without Expires header or Expires header is not set to zero. ABE-2442 ABE-2443 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@297603 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-02Resolve compile error under FreeBSDpabelanger1-0/+3
We now set _ASTCFLAGS+=-march=i686 for i386 processors, still allowing ASTCFLAGS to override the setting. Review: https://reviewboard.asterisk.org/r/1043/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@297404 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-02Initialize offset for adaptive jitter buffertwilson1-0/+5
When the adaptive jitter buffer is enabled in sip.conf, the first frame placed in the jitter buffer fails with something like: jb_warning_output: Resyncing the jb. last_delay 0, this delay -215886466, threshold 1000, new offset 215886466 This happens because the offset is not initialized before calling jb_put(). This patch modifies jb_put_first_adaptive() to set the offset to the frame's timestamp. Review: https://reviewboard.asterisk.org/r/1041/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@297310 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-02Add "DAHDI" to a couple of app_meetme error messages.russell1-2/+2
This is in response to some questions on IRC. To the user, there was nothing that made it obvious that this error had anything to do with DAHDI not being loaded. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@297228 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-02If we get a NOTIFY from a non-existing subscription we should answer with ↵oej1-1/+1
481, not bad event. If we answer 481 the subscription that we don't want will be cancelled. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@297185 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-01Fix not stopping MOH when transfered local channel queue member is answered.jpeeler1-0/+3
The problem here is only present when local channels are used with the MOH passthru option as well as no optimization (/nm). I will describe the slightly bizarre scenario that was used to test, where phones B and C are queue members: Phone A dials into a queue with two members using local channels and the above options. Phone B answers. Phone A blind transfers phone B into the same queue. Phone A hangs up. Phone C answers, but phone B didn't stop playing MOH. In this scenario, the unhold frame that should have gotten to phone B never arrived due to the masquerade from the blind transfer. This is usually fine since app_queue manages the starting and stopping of MOH. However, with the passthrough option enabled when app_queue attempts to stop MOH it tries to do so on the local channel rather than the real channel. The easiest solution was to just make sure to send an unhold frame during the transfer since it wouldn't make sense to have MOH playing after a transfer anyway. This only modifies SIP transfers, but the other transfers did not seem to be a problem. If DTMF based transfers were a problem it might be okay to add ast_moh_stop to finishup, but I didn't want to have to add that unless required. ABE-2624 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@297072 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-01Clarify documentation on how we store codec preference lists.tilghman1-1/+9
(closes issue #18397) Reported by: birgita git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@296990 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-01Properly restore backup information file when hanging up during message ↵jpeeler1-0/+10
prepending. ABE-2654 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@296868 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-01Get rid of the annoying startup and shutdown errors on OS X.tilghman2-1/+25
This mainly deals with the problem of constructors on platforms where an explicit constructor order cannot be specified (any system with gcc 4.2 or less). However, this is only a problem on those systems where we need to initialize mutexes with a constructor, because we have other code that also relies upon constructors, and we cannot specify that mutexes are initialized first (and destroyed last). There are two approaches to dealing with this issue, related to whether the code exists in the core Asterisk binary or in a separate code module. In the core case, constructors are run immediately upon load, and the file_versions list mutex needs to be already initialized, as it is referenced in the first constructor within each core source file. In this case, we use pthread_once to ensure that the mutex is initialized immediately before it is used for the first time. The only caveat is that the mutex is not ever destroyed, but because this is the core, it makes no real difference; the only time when destruction is safe would be just prior to process destruction, which takes care of that anyway. And due to using pthread_once, the mutex will never be reinitialized, which means only one structure has leaked at the end of the process. Hence, it is not a problematic leak. The second approach is to use the load_module and unload_module routines, which, for obvious reasons, exist only in loadable modules. In this second case, we don't have a problem with the constructors, but only with destructor order, because mutexes can be destroyed before their final usage is employed. However, we need the mutexes to still be destroyed, in certain scenarios: if the module is unloaded prior to the process ending, it should be clean, with no allocations by the module hanging around after that point in time. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@296867 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-29Make sure nothing else is needed before destroying the scheduler.pabelanger1-2/+2
(closes issue #18398) Reported by: pabelanger git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@296670 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-26Fix bugs in saying numbers using the Swedish language syntaxoej1-38/+45
(closes issue #18355) Reported by: oej Patch by: oej Much help from Peter Lindahl. Testing by the ClearIT team during a coffee break. Review: https://reviewboard.asterisk.org/r/1033/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@296309 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-24Make Asterisk less crashy.russell1-1/+3
Since we might not put a new translation path on the channel, go ahead and set it to NULL right after destroying the old one to ensure we don't try to free an invalid translation path later on. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@296213 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-24Oneway audio to SIP phone from FXS port after FXS port gets a CallWaiting pip.rmudgett1-108/+202
The FXS connected phone has to have CW/CID support to fail, as it will send back a DTMF 'A' or 'D' when it's ready to receive CallerID. A normal phone with no CID never fails. Also the SIP phone does not hear MOH when the CW call is answered. The DTMF end frame is suppressed when the phone acknowledges the CW signal for CID. The problem is the DTMF begin frame needs to be suppressed as well. The DTMF begin frame is causing SIP to start sending the DTMF RTP frames. Since the DTMF end frame is suppressed, SIP will not stop sending those DTMF RTP packets. * Suppress the DTMF begin and end frames when the channel driver is looking for DTMF digits. * Fixed a couple issues caused by not cleaning up the CID spill if you answer the CW call while it is sending the CID spill. * Fixed not sending CW/CID spill to the phone when the call is natively bridged. (Fixed by not using native bridge if CW/CID is possible.) * Suppress received audio when sending CW/CID spills. The other parties involved do not need to hear the CW/CID spills and may be confused if the CW call is for them. (closes issue #18129) Reported by: alecdavis Patches: issue_18129_v1.8_v3.patch uploaded by rmudgett (license 664) Tested by: alecdavis, rmudgett NOTE: * v1.4 does not have the main problem fixed by suppressing the DTMF start frames. The other three items fixed are relevant. * If you really must restore native bridging between analog ports, you need to disable CW/CID either by configuring chan_dahdi.conf callwaitingcallerid=no or dialing *70 before dialing the number to temporarily disable CW. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@296165 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-24Fix false reporting of an error by set_format().russell1-7/+17
In the case that the native format was able to be changed to match the new requested format, the code proceeded to attempt to build a translation path, anyway. The result would be NULL, since no translation path is necessary and resulted in this function thinking an error has occurred. This case is now specifically caught and no attempt to build a translation path is attempted. Thanks to our automated tests and bamboo.asterisk.org for catching this problem and making a whole lot of noise when things started failing. :-) git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@296082 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-24Handle failures building translation paths more effectively.russell2-4/+13
The problem scenario occurred on a heavily loaded system that was using the codec_dahdi module and exceeded the hardware transcoding capacity. The failure mode at that point was not good. The report came in to us as an Asterisk lock-up. The "core show locks" shows a ton of threads locked up (but no obvious deadlock). Upon deeper investigation, when the system is in this state, the CPU was maxed out. The CPU was being consumed by the Asterisk logger spewing messages on every audio frame for calls set up after transcoder capacity was reached. The purpose of this patch is to make Asterisk handle failures to create a translation path in a more graceful manner. If we can't translate, then the call just needs to be dropped, as it's not going to work. These are the changes: 1) In set_format() of channel.c (which is called by set_read_format() and set_write_format()), it was ignoring if ast_translator_build_path() failed and returned NULL. It now pays attention to that case and returns a result reflecting failure. With this change in place, the bridging code will immediately detect a failure and end the bridge instead of proceeding to try to bridge frames that can't be translated and making channel drivers freak out by sending them frames in a format they weren't expecting. 2) In ast_indicate_data() of channel.c, failure of ast_playtones_start() was ignored. It is now reflected in the return value of the function. This didn't turn out to have any affect on the bug, but seemed like a good change to leave in. 3) In app_dial(), when only sending a call to a single endpoint, it will attempt to do some bridging of its own of early audio. It uses make_compatible() when it's going to do this. However, it ignored failure from make compatible. So, even with the fix from #1, if there was early audio going through app_dial, there would still be a period of invalid frames passing through. After detecting failure here, Dial() exits. ABE-2658 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@296000 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-23Fix support of saynumber(1,n) in the Swedish languageoej1-3/+3
(closes issue #18353) Reported by: oej Review: https://reviewboard.asterisk.org/r/1031/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@295906 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-22The channel redirect function (CLI or AMI) hangs up the call instead of ↵rmudgett5-55/+132
redirecting the call. To recreate the problem: 1) Party A calls Party B 2) Invoke CLI "channel redirect" command to redirect channel call leg associated with A. 3) All associated channels are hung up. Note that if the CLI command were done on the channel call leg associated with B it works. This regression was a result of the fix for issue #16946 (https://reviewboard.asterisk.org/r/740/). The regression affects all features that use an async goto to execute the dialplan because of an external event: Channel redirect, AMI redirect, SIP REFER, and FAX detection. The struct ast_channel._softhangup code is a mess. The variable is used for several purposes that do not necessarily result in the call being hung up. I have added doxygen comments to describe how the various _softhangup bits are used. I have corrected all the places where the variable was tested in a non-bit oriented manner. The primary fix is the new AST_CONTROL_END_OF_Q frame. It acts as a weak hangup request so the soft hangup requests that do not normally result in a hangup do not hangup. JIRA SWP-2470 JIRA SWP-2489 (closes issue #18171) Reported by: SantaFox (closes issue #18185) Reported by: kwemheuer (closes issue #18211) Reported by: zahir_koradia (closes issue #18230) Reported by: vmarrone (closes issue #18299) Reported by: mbrevda (closes issue #18322) Reported by: nerbos Review: https://reviewboard.asterisk.org/r/1013/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@295790 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-19Discard responses with more than one Viatwilson1-3/+19
This is not a perfect solution as headers that are joined via commas are not detected. This is a parsing issue that to fix "correctly" would necessitate a new SIP parser. Review: https://reviewboard.asterisk.org/r/1019/ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@295628 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-19Revert a new feature which should have gone into trunk.espiceland1-74/+3
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@295553 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-19Add extra functionality to AGI command WAIT FOR DIGIT.espiceland1-3/+74
Add the ability to play a sound file, listen for more than just one digit, specify escape characters. Backwards compatible (to work with only timeout specified). (closes issue #15531) Reported by: diLLec Patches: asterisk-res_agi-203638-patched.patch uploaded by diLLec (license 839) Tested by: diLLec, espiceland git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@295552 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-16Dead code elimination in channel.c:ast_channel_bridge() variable who.rmudgett1-5/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@295280 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-16Ensure original message duration is preserved when prepending a message.jpeeler1-6/+19
It seems the fix to issue 17103 was a little overzealous and removed the code that backed up the textfile containing the original message duration. This code has now been restored. (related to issue #17103) ABE-2654 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@295200 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-15Err, oops. Made it const to verify that it wasn't altered, but forgot to ↵tilghman1-1/+1
revert before commit. git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@295031 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-15Create test verifying results of expression parsertilghman1-0/+191
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@295026 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-12Fix regression causing abort in voicemail after opening a mailbox with no mesgs.jpeeler1-1/+2
In order to be more safe, some error handling code was changed to respect more error conditions including the potential memory allocation failure for deleted and heard message tracking introduced in 293004. However, last_message_index returns -1 for zero messages (perhaps as expected) and was triggering the stricter error checking. Because last_message_index is only called directly in one place, just return 0 from open_mailbox (for file based storage) when no messages are detected unless a real error has occurred. (closes issue #18240) Reported by: leobrown Patches: bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585) Tested by: pabelanger git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@294903 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-12Asterisk is getting a "No D-channels available!" warning message every 4 ↵rmudgett1-2/+11
seconds. Asterisk is just whining too much with this message: "No D-channels available! Using Primary channel XXX as D-channel anyway!". Filtered the message so it only comes out once if there is no D channel available without an intervening D channel available period. (closes issue #17270) Reported by: jmls git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@294821 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-11I didn't mean to merge this, sorryjpeeler1-3/+0
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@294739 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-11Fix problem with qualify option packets for realtime peers never stopping.jpeeler1-1/+24
The option packets not only never stopped, but if a realtime peer was not in the peer list multiple options dialogs could accumulate over time. This scenario has the potential to progress to the point of saturating a link just from options packets. The fix was to ensure that the poke scheduler checks to see if a peer is in the peer list before continuing to poke. The reason a peer must be in the peer list to be able to properly manage an options dialog is because otherwise the call pointer is lost when the peer is regenerated from the database, which is how existing qualify dialogs are detected. (closes issue #16382) Reported by: lftsy Patches: bug16382-3.patch uploaded by jpeeler (license 325) Tested by: zerohalo git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@294688 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-11One small addition to 294384 found while very carefully merging to 1.6.jpeeler1-0/+6
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@294641 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-09Fix a deadlock in device state change processing.jpeeler3-107/+193
Copied from some notes from the original author (Russell): Deadlock scenario: Thread 1: device state change thread Holds - rdlock on contexts Holds - hints lock Waiting on channels container lock Thread 2: SIP monitor thread Holds the "iflock" Holds a sip_pvt lock Holds channel container lock Waiting for a channel lock Thread 3: A channel thread (chan_local in this case) Holds 2 channel locks acquired within app_dial Holds a 3rd channel lock it got inside of chan_local Holds a local_pvt lock Waiting on a rdlock of the contexts lock A bunch of other threads waiting on a wrlock of the contexts lock To address this deadlock, some locking order rules must be put in place and enforced. Existing relevant rules: 1) channel lock before a pvt lock 2) contexts lock before hints lock 3) channels container before a channel What's missing is some enforcement of the order when you involve more than any two. To fix this problem, I put in some code that ensures that (at least in the code paths involved in this bug) the locks in (3) come before the locks in (2). To change the operation of thread 1 to comply, I converted the storage of hints to an astobj2 container. This allows processing of hints without holding the hints container lock. So, in the code path that led to thread 1's state, it no longer holds either the contexts or hints lock while it attempts to lock the channels container. (closes issue #18165) Reported by: antonio ABE-2583 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@294384 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-08Modify our handling of 491 responses to drop any pending reinvite retry ↵mnicholson1-0/+8
scheduler entries if we get a new 491. This prevents a scheduler entry from leaking if we receive a 491 response when one is pending. If a scheduler entry leaks, the pvt it is associated my get destroyed before the scheduler entry fires, and then memory corruption and crashes can occur when the scheduled reinvite attempts to access and modify the memory of the destroyed pvt. ABE-2543 git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@294163 f38db490-d61c-443f-a65b-d21fe96a405b
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