Age | Commit message (Collapse) | Author | Files | Lines |
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"WARNING[28406]: chan_dahdi.c:6873 ss_thread: CallerID feed failed: Success"
Changed the warning to "Failed to decode CallerID on channel 'name'". The
message before it is likely more specific about why the CallerID decode
failed.
SWP-501
AST-283
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@259531 f38db490-d61c-443f-a65b-d21fe96a405b
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* Add additional sounds prompts for say_enumeration
* Update the English conference sounds prompts so they are better
quality and all sound more consistent
* Clean up the core-sounds-XX.txt and extra-sounds-XX.txt files to
include all present sound files
Both core (en, fr, es) and extra (en, fr) sounds files have been updated.
(closes issue #16200)
Reported by: murf
(closes issue #17137)
Reported by: lmadsen
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Since AC_PATH_TOOL is equiv to AC_CHECK_TOOL when path isn't specified, and
AC_PATH_TOOLS doesn't exist, we'll just switch to AC_CHECK_TOOLS.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@259352 f38db490-d61c-443f-a65b-d21fe96a405b
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Issue #7321 implements a new chan_dahdi configuration option. However, a
change mentioned in the issue was never implemented. This is the change
that will allow the feature to work.
I added a note to chan_dahdi.conf.sample about the feature.
(closes issue #17143)
Reported by: djensen99
Patches:
diff.txt uploaded by djensen99 (license NA) (One line change)
Tested by: djensen99
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No Newchannel manager event will be fired for channels that are
allocated to not match a registered technology type. Thus bogus
channels allocated solely for variable substitution or CDR
operations do not result in a Newchannel event.
(closes issue #16957)
Reported by: atis
Review: https://reviewboard.asterisk.org/r/601
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@259018 f38db490-d61c-443f-a65b-d21fe96a405b
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channel event.
(closes issue #16590)
Reported by: kkm
Patches:
resmonitor-16590-trunk.239289.diff uploaded by kkm (license 888)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@258775 f38db490-d61c-443f-a65b-d21fe96a405b
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This change allows a CDR record previously marked with disposition ANSWERED to be set as BUSY or NO ANSWER.
Additionally this change partially reverts r235635 and does not set the AST_CDR_FLAG_ORIGINATED flag on CDRs generated from ast_call(). To preserve proper CDR behavior, the AST_CDR_FLAG_DIALED flag is now cleared from all brige CDRs in ast_bridge_call().
(closes issue #16797)
Reported by: VarnishedOtter
Tested by: mnicholson
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Specifically, prompting for an extension (when leaving or forwarding a message)
or when prompting for a digit (when saving a message or changing folders).
ABE-2122
SWP-1268
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If a user's mailbox was full and a message was attempted to be forwarded to
said box, warnings on the console would indicate failure. However, the played
prompt was that of success (vm-msgsaved). Now storage failure is taken into
account and the correct prompt (vm-mailboxfull) is played when appropriate.
ABE-2123
SWP-1262
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Mantis issue 17078 reports MixMonitor recordings have shorter durations than
the call duration. This was because the mixmonitor thread was not processing
frames from the audiohook fast enough. The mixmonitor thread would slowly fall
behind the most recent audio frame and when the channel hangs up, the mixmonitor
thread would exit without processing the same number of frames as the channel;
leaving the mixmonitor recording shorter than actual call duration.
This revision fixes this issue by moving the ast_audiohook_trigger_wait() and
the subsequent audiohook.status check into the block where the
ast_audiohook_read_frame() function returns NULL.
(closes issue #17078)
Reported by: geoff2010
Patches:
dw-M17078.patch uploaded by dhubbard (license 733)
Tested by: dhubbard, geoff2010
Review: https://reviewboard.asterisk.org/r/611/
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variety of escaping techniques.
Fixes SWP-1194 (ABE-2143).
Review: https://reviewboard.asterisk.org/r/604/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@257544 f38db490-d61c-443f-a65b-d21fe96a405b
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When a reply to a deregistration is lost in transmit, the client retries the
deregistration. Previously, this would cause a realtime/autocreate peer to be
loaded back into memory, after it had already been correctly purged. Instead,
we just want to resend the reply without loading the peer.
(closes issue #16908)
Reported by: kkm
Patches:
20100412__issue16908.diff.txt uploaded by tilghman (license 14)
Tested by: kkm
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@257467 f38db490-d61c-443f-a65b-d21fe96a405b
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Update the backtrace.txt documentation so it conforms to the same layout as
other documents we've been working on recently. Additionally, add a bunch of
new information about gathering backtraces for crashes and deadlocks, along
with ways of verifying your file before uploading it. Create a couple of one
line commands for people to generate the files we need.
(closes issue #17190)
Reported by: lmadsen
Patches:
backtrace.txt.patch-2 uploaded by lmadsen (license 10)
Tested by: lmadsen, pabelanger
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This is a regression in 1.4, only.
(closes issue #17103)
Reported by: mglazer
Patches:
20100408__issue17103.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@257266 f38db490-d61c-443f-a65b-d21fe96a405b
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Before r238915, certain values for the EventMask parameter of the Events action would result in no response being returned. This patch adds an option to restore that broken behavior. Also while fixing this bug I discovered that passing an empty EventMasks parameter would also result in no response being returned, this has been fixed as well while being preserved when the broken behavior is requested.
(closes issue #17023)
Reported by: nblasgen
Review: https://reviewboard.asterisk.org/r/602/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@257070 f38db490-d61c-443f-a65b-d21fe96a405b
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Paul Belanger has been helping a lot with bug tracking recently and created
this document that we can now point to when additional debugging information
is required. This document will help those filing issues to know how to get
the information required when filing their issues. This will make things
easier on the developers.
Initial text and changes by pabelanger. Tweaks and editing by myself.
(closes issue #17159)
Reported by: pabelanger
Patches:
HOWTO_collect_debug_information.txt.patch uploaded by lmadsen (license 10)
Tested by: tzafrir, pabelanger, lmadsen
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@256900 f38db490-d61c-443f-a65b-d21fe96a405b
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SWP-1231
ABE-2163
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(closes issue #16840)
Reported by: bzing2
Patches:
patch.txt uploaded by bzing2 (license 902)
issue_16840.rev1.diff uploaded by russell (license 2)
Tested by: bzing2, russell
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Fixes an issue with certain Mail Transport Agents, where attachments are not
interpreted correctly.
(closes issue #16557)
Reported by: jcovert
Patches:
20100308__issue16557__1.4.diff.txt uploaded by tilghman (license 14)
20100308__issue16557__1.6.0.diff.txt uploaded by tilghman (license 14)
20100308__issue16557__trunk.diff.txt uploaded by tilghman (license 14)
Tested by: ebroad, zktech
Reviewboard: https://reviewboard.asterisk.org/r/544/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@255591 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17021)
Reported by: kovzol
Tested by: lmadsen, kovzol, davidw, ebroad
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Take 2, without ABI breakage this time.
Review: https://reviewboard.asterisk.org/r/588/
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Review: https://reviewboard.asterisk.org/r/590/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@254639 f38db490-d61c-443f-a65b-d21fe96a405b
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Review: https://reviewboard.asterisk.org/r/528
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@254552 f38db490-d61c-443f-a65b-d21fe96a405b
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Here is a copy and paste of the details from my request on
reviewboard that dealt with these changes:
Fix 1. The first change in place is to fix Mantis issue 15811, which deals with a situation where Asterisk will incorrectly interpret out of order RFC2833 frames as duplicate DTMF digits. For instance, we would receive a sequence like:
seqno 1: DTMF 1
seqno 2: DTMF 1
seqno 3: DTMF 1
seqno 4: DTMF 1
seqno 6: DTMF 1 (end)
seqno 5: DTMF 1
seqno 7: DTMF 1 (end)
seqno 8: DTMF 1 (end)
Prior to this patch when we received the frame with seqno 5, we would interpret this as a new DTMF 1. With this patch, we will check the seqno of the incoming digit and not process the frame if the seqno is lower than the last recorded seqno. Note that we do not record the seqno of the dropped DTMF frame for future processing. While the above situation is what was designed to be fixed, the patch is written in such a way that the following would also be fixed too:
seqno 9: DTMF 1
seqno 10: DTMF 1 (end)
seqno 11: DTMF 1 (end)
seqno 13: DTMF 2
seqno 12: DTMF 1 (end)
seqno 14: DTMF 2
seqno 15: DTMF 2 (end)
seqno 16: DTMF 2 (end)
seqno 17: DTMF 2 (end)
In this second situation, the beginning of the DTMF 2 arrives before the final end frame of the DTMF 1. With the patch, seqno 12 is no processed and thus we properly interpret the DTMF.
Fix 2. The second change in place is to fix an issue like the following:
seqno 1: DTMF 1
seqno 2: DTMF 1
seqno 3: DTMF 1 (end) *packet lost*
seqno 4: DTMF 1 (end) *packet lost*
seqno 5: DTMF 1 (end) *packet lost*
seqno 6: DTMF 2
When we receive seqno 6, we had code in place that was supposed to properly end the previously unended DTMF 1. The problem was that the code was essentially a no-op. The code would set up an end frame for the DTMF 1 but would immediately overwrite the frame with the begin for DTMF 2. I changed process_dtmf_rfc2833() so that instead of returning a single frame, it is given as an output parameter a list of frames. Each frame that needs to be returned is appended to this list.
Fix 3. The final change is a minor one where an AST_CONTROL_SRCCHANGE frame could get lost. If we process a cisco DTMF or an RFC 3389 frame and no frame was returned, then we would return &ast_null_frame. The problem is that earlier in the function, we may have generated an AST_CONTROL_SRCCHANGE frame and put it in the list of frames we wish to return. This frame would be lost in such a case. The patch fixes this problem
Review: https://reviewboard.asterisk.org/r/558
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This is an extension to 248860. As such the dialplan test has been extended:
; non absolute path, not combined
exten => 5040, 1, monitor(wav,tmp/jeff/monitor_test)
exten => 5040, n, dial(sip/5001)
; absolute path, not combined
exten => 5041, 1, monitor(wav,/tmp/jeff/monitor_test2)
exten => 5041, n, dial(sip/5001)
; no path, not combined
exten => 5042, 1, monitor(wav,monitor_test3)
exten => 5042, n, dial(sip/5001)
; combined: changemonitor from non absolute to no path (leaves tmp/jeff)
exten => 5043, 1, monitor(wav,tmp/jeff/monitor_test4,m)
exten => 5043, n, changemonitor(monitor_test5)
exten => 5043, n, dial(sip/5001)
; combined: changemonitor from no path to non absolute path
exten => 5044, 1, monitor(wav,monitor_test6,m)
exten => 5044, n, changemonitor(tmp/jeff/monitor_test7) ; this wasn't possible before
exten => 5044, n, dial(sip/5001)
; non absolute path, combined
exten => 5045, 1, monitor(wav,tmp/jeff/monitor_test8,m)
exten => 5045, n, dial(sip/5001)
; absolute path, combined
exten => 5046, 1, monitor(wav,/tmp/jeff/monitor_test9,m)
exten => 5046, n, dial(sip/5001)
; no path, combined
exten => 5047, 1, monitor(wav,monitor_test10,m)
exten => 5047, n, dial(sip/5001)
; combined: changemonitor from non absolute to absolute (leaves tmp/jeff)
exten => 5048, 1, monitor(wav,tmp/jeff/monitor_test11,m)
exten => 5048, n, changemonitor(/tmp/jeff/monitor_test12)
exten => 5048, n, dial(sip/5001)
; combined: changemonitor from absolute to non absolute (leaves /tmp/jeff)
exten => 5049, 1, monitor(wav,/tmp/jeff/monitor_test13,m)
exten => 5049, n, changemonitor(tmp/jeff/monitor_test14)
exten => 5049, n, dial(sip/5001)
; combined: changemonitor from no path to absolute
exten => 5050, 1, monitor(wav,monitor_test15,m)
exten => 5050, n, changemonitor(/tmp/jeff/monitor_test16)
exten => 5050, n, dial(sip/5001)
; combined: changemonitor from absolute to no path (leaves /tmp/jeff)
exten => 5051, 1, monitor(wav,/tmp/jeff/monitor_test17,m)
exten => 5051, n, changemonitor(monitor_test18)
exten => 5051, n, dial(sip/5001)
; not combined: changemonitor from non absolute to no path (leaves tmp/jeff)
exten => 5052, 1, monitor(wav,tmp/jeff/monitor_test19)
exten => 5052, n, changemonitor(monitor_test20)
exten => 5052, n, dial(sip/5001)
; not combined: changemonitor from no path to non absolute
exten => 5053, 1, monitor(wav,monitor_test21)
exten => 5053, n, changemonitor(tmp/jeff/monitor_test22)
exten => 5053, n, dial(sip/5001)
; not combined: changemonitor from non absolute to absolute (leaves tmp/jeff)
exten => 5054, 1, monitor(wav,tmp/jeff/monitor_test23)
exten => 5054, n, changemonitor(/tmp/jeff/monitor_test24)
exten => 5054, n, dial(sip/5001)
; not combined: changemonitor from absolute to non absolute (leaves /tmp/jeff)
exten => 5055, 1, monitor(wav,/tmp/jeff/monitor_test24)
exten => 5055, n, changemonitor(tmp/jeff/monitor_test25)
exten => 5055, n, dial(sip/5001)
; not combined: changemonitor from no path to absolute
exten => 5056, 1, monitor(wav,monitor_test26)
exten => 5056, n, changemonitor(/tmp/jeff/monitor_test27)
exten => 5056, n, dial(sip/5001)
; not combined: changemonitor from absolute to no path (leaves /tmp/jeff)
exten => 5057, 1, monitor(wav,/tmp/jeff/monitor_test28)
exten => 5057, n, changemonitor(monitor_test29)
exten => 5057, n, dial(sip/5001)
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DEBUG_THREADS/DEBUG_CHANNEL_LOCKS differences.
This can be guaranteed by forcing the ABI to no longer change when these compiler flags are set.
An unfortunate side-effect to this is that there is an ABI change here. However, there is some
mitigation. Existing modules *will* fail to load since they would require functions that no
longer exist.
Review: https://reviewboard.asterisk.org/r/508/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@254046 f38db490-d61c-443f-a65b-d21fe96a405b
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(related to issue #16331)
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(issue #16963)
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The original reporter, Kobaz, of an issue with a Local channel that inspired the
Local channel documentation provided some tweaks to the documentation after testing
what I had written. Hopefully anything that was vague or unclear has been cleaned
up by these changes.
(closes issue #16963)
Reported by: kobaz
Patches:
localchannel-2.txt uploaded by kobaz (license 834)
Tested by: kobaz, lmadsen
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@253252 f38db490-d61c-443f-a65b-d21fe96a405b
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This re-renames ast_rtp_update_source to ast_rtp_new_source
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Add the french snipset to say.conf.
(Closes issue #15799)
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muted supports OS X, but uses functions marked as deprecated in 10.6. However,
the functions are still supported, so just ignore the warnings for now and
allow the build to proceed.
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Fixing up a couple more overlapping global variable namespaces shared with
extensions.conf.sample. Also noticed a few of the lines that were commented
out didn't have the closing semi-colon so I added that as well.
(issue #17035)
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