mirror of https://github.com/asterisk/asterisk
master
22
23
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${ noResults }
14314 Commits (8964c844bd5b266f2cb0d9842f403e5529ad1565)
| Author | SHA1 | Message | Date |
|---|---|---|---|
|
|
8964c844bd |
Fix issue with cross-compiling failing
(closes issue #18301) Reported by: abelbeck Patches: asterisk-1.4-bugid18301.patch.txt uploaded by abelbeck (license 946) Tested by: abelbeck, russellb git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@301869 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
4717c625e9 |
Don't store the thread id for the manager session in the structure we pass to
the thread for the manager session. ABE-2543 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@301591 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
|
|
4b11743a18 |
Fix CPU spike when pressing DTMF after agent login.
The problem here is that DTMF was being continuously deferred and requeued since ast_safe_sleep is called in a loop. There are serveral other places in the code that sleeps and then loops in a similar fashion. Because of this fact I opted to not defer DTMF any more, which will not affect the original fix: https://reviewboard.asterisk.org/r/674 (closes issue #18130) Reported by: rgj git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@301502 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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08fcb05bc6 |
Prevent buffer overflows in ast_uri_encode()
ABE-2705 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@301305 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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25181e5bdb |
change variable name to the same used in later branches
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@300924 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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93864bd03d |
Ensure good bye prompt in voicemail is played at the correct time.
Specifically in the case of timing out but not leaving voicemail nothing should be heard. And when leaving voicemail it should be heard. ABE-2647 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@300918 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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93782b4eb2 |
Use the sanity check in place of the disconnect/connect cycle.
The disconnect/connect cycle has the potential to cause random crashes. (closes issue #18243) Reported by: ks3 Patches: res_odbc.patch uploaded by ks3 (license 1147) Tested by: ks3 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@300621 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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94cb9ce1ac |
Update the autosupport script from Digium support.
(closes AST-395) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@300428 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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a106ce66a4 |
Don't authenticate SUBSCRIBE re-transmissions
This only skips authentication on retransmissions that are already authenticated. A similar method is already used for INVITES. This is the kind of thing we end up having to do when we don't have a transaction layer... (closes issue #18075) Reported by: mdu113 Patches: diff.txt uploaded by twilson (license 396) Tested by: twilson, mdu113 Review: https://reviewboard.asterisk.org/r/1005/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@300216 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e913a29969 |
Move check for extension existence below variable inheritance, due to the possible use of an eswitch.
(closes issue #16228) Reported by: jlaguilar git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@299624 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ab674cd470 |
Use ast_free() instead of free()
ABE-2656 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@299220 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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7c10869584 |
Remove changes to via processing that were not supposed to go into the last commit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@299198 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b3edf004d2 |
Respond as soon as possible with a 202 Accepted to refer requests.
This change also plugs a few memory leaks that can occur when parking sip calls. ABE-2656 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@299194 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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24118f5839 |
Typos: recieved => received
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@299002 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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78febf3647 |
Let Asterisk find better backtrace information with libbfd.
The menuselect option BETTER_BACKTRACES, if enabled, will use libbfd to search for better symbol information within both the Asterisk binary, as well as loaded modules, to assist when using inline backtraces to track down problems. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@298905 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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f47ff91be3 |
Document the -F option.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@298820 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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ccfbb0a30a |
After recording only silence for a voicemail prepending, restore backup files.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@298683 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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71115232da |
Fix improper hangup when doing an attended transfer to queue.
Had to indicate ringing in wait_for_answer so the attended transfer code would not try and hang up the local channel it created, which would kill the call. ABE-2624 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@298596 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3d26dbf19d |
Only increment the pointer once per loop, otherwise we corrupt the value.
(closes issue #18251) Reported by: bcnit Patches: 20101110__issue18251.diff.txt uploaded by tilghman (license 14) Tested by: trev, jthurman, elguero (closes issue #18279) Reported by: zerohalo Patches: 20101109__issue18279.diff.txt uploaded by tilghman (license 14) Tested by: zerohalo git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@298480 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d8f9e4b1d8 |
Unregister before shutting down the connection, to avoid a race.
(closes issue #18481) Reported by: pabelanger Patches: 20101215__issue18481.diff.txt uploaded by tilghman (license 14) Tested by: pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@298392 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d644f402d0 |
Fix reference and container leaks when running 'astobj2 test.'
We need to make sure that ao2_iterator_destroy is called once for each time that ao2_iterator_init is called. Also make sure to unref a newly allocated object that we've linked into a container. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@298345 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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0543f25d7c |
Outgoing PRI/BRI calls cannot do DTMF triggered transfers.
Outgoing PRI/BRI calls cannot do DTMF triggered transfers if a PROCEEDING message is not received. The debug output shows that the DTMF begin event is seen, but the DTMF end event is missing. When the DTMF begin happens, the call is muted so we now have one way audio (until a DTMF end event is somehow seen). * Made set the proceeding flag when the PRI_EVENT_ANSWER event is received. * Made absorb the DTMF begin and DTMF end events if we are overlap dialing and have not seen a PROCEEDING message. * Added a debug message when absorbing a DTMF event. JIRA SWP-2690 JIRA ABE-2697 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@298193 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d7877331c0 |
Ignore spurious REGISTER requests
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@297959 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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cd535276c1 |
Revert code that changed SSRC for DTMF.
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@297823 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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f614a52b52 |
Use non-deprecated APIs for CoreAudio
Review: https://reviewboard.asterisk.org/r/1040/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@297818 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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0135ff19ed |
Avoid a crash if we don't pass an argument to 'astobj2 test.'
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@297775 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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4369b3c4ae |
Don't create a Local channel if the target extension does not exist.
(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: https://origsvn.digium.com/svn/asterisk/branches/1.4@297689 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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919ad38940 |
Improve handling of REGISTER requests with multiple contact headers.
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@297603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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15 years ago |
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449a5e22fb |
Resolve compile error under FreeBSD
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@297404 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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59db525cc0 |
Initialize offset for adaptive jitter buffer
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@297310 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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92ea2179e5 |
Add "DAHDI" to a couple of app_meetme error messages.
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@297228 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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8b2304108a |
If we get a NOTIFY from a non-existing subscription we should answer with 481, not bad event.
If we answer 481 the subscription that we don't want will be cancelled. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@297185 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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046f56ce1c |
Fix not stopping MOH when transfered local channel queue member is answered.
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@297072 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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415e95df9a |
Clarify documentation on how we store codec preference lists.
(closes issue #18397) Reported by: birgita git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@296990 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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25c2ccaac3 |
Properly restore backup information file when hanging up during message prepending.
ABE-2654 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@296868 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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196032208f |
Get rid of the annoying startup and shutdown errors on OS X.
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@296867 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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728bb5ac67 |
Make sure nothing else is needed before destroying the scheduler.
(closes issue #18398) Reported by: pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@296670 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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0876c9fbe1 |
Fix bugs in saying numbers using the Swedish language syntax
(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: https://origsvn.digium.com/svn/asterisk/branches/1.4@296309 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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f95e212afc |
Make Asterisk less crashy.
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@296213 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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c98862958a |
Oneway audio to SIP phone from FXS port after FXS port gets a CallWaiting pip.
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@296165 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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41656ad0bf |
Fix false reporting of an error by set_format().
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@296082 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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72d3cd8b47 |
Handle failures building translation paths more effectively.
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@296000 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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74270307c9 |
Fix support of saynumber(1,n) in the Swedish language
(closes issue #18353) Reported by: oej Review: https://reviewboard.asterisk.org/r/1031/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@295906 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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d32582d42f |
The channel redirect function (CLI or AMI) hangs up the call instead of 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: https://origsvn.digium.com/svn/asterisk/branches/1.4@295790 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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62d947a168 |
Discard responses with more than one Via
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@295628 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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28922df829 |
Revert a new feature which should have gone into trunk.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@295553 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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fa62a9700a |
Add extra functionality to AGI command WAIT FOR DIGIT.
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@295552 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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b174c8f30e |
Dead code elimination in channel.c:ast_channel_bridge() variable who.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@295280 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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3fc98d5530 |
Ensure original message duration is preserved when prepending a message.
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@295200 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |
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e36e50146a |
Err, oops. Made it const to verify that it wasn't altered, but forgot to revert before commit.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@295031 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
15 years ago |