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git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.23-rc2@160146 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.23-rc2@160145 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.23-rc2@160143 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.23-rc2@160136 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/tags/1.4.23-rc2@160134 f38db490-d61c-443f-a65b-d21fe96a405b
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they
both have the potential to send control frames in the middle of call setup. We
have to wait until we have received a message back from the remote end before
we try to send any more frames. Otherwise, the remote end will consider it
invalid, and we'll get stuck in an INVAL/VNAK storm.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@160003 f38db490-d61c-443f-a65b-d21fe96a405b
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channelname.
Noted by kpfleming and name Bogus/manager suggested by eliel
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The last (10th) argument to ast_channel_alloc here should be a pointer
and NULL is not really a pointer.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@159897 f38db490-d61c-443f-a65b-d21fe96a405b
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Intrepid, so all developers will see the same warnings and errors
since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them
format attributes in a consistent way
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@159808 f38db490-d61c-443f-a65b-d21fe96a405b
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copies and have unversioned files already in place
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@159571 f38db490-d61c-443f-a65b-d21fe96a405b
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mode
add channels/busy.h and channels/ringtone.h to the repository instead of generating them repeatedtly; most users do not change the settings to build them, but the Makefile rules are still there if they wish to do so
ensure that 'make clean' removes dependency files for .i files that are created in COMPILE_DOUBLE mode
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@159476 f38db490-d61c-443f-a65b-d21fe96a405b
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Reported by: yraber
Patches:
12694.2nd.diff uploaded by murf (license 17)
Tested by: murf, laurav
Thanks to file (Joshua Colp) for his IAX fix.
the change to cdr.c allows no-answer to percolate
up into CDR's, and feels like the right place to
locate this fix; if BUSY is done here, no-answer
should be, too.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@159316 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #13919)
Reported by: barthpbx
Patches:
chan_iax2.c.patch uploaded by eliel (license 64)
Tested by: barthpbx
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@159269 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r159245 | tilghman | 2008-11-25 15:37:06 -0600 (Tue, 25 Nov 2008) | 7 lines
Regression fix for last security fix. Set the iseqno correctly.
(closes issue #13918)
Reported by: ffloimair
Patches:
20081119__bug13918.diff.txt uploaded by Corydon76 (license 14)
Tested by: ffloimair
........
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(Closes issue #13863)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@159025 f38db490-d61c-443f-a65b-d21fe96a405b
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Reported by: tzafrir
When compiling against Zaptel dahdi_compat will now only define all the DAHDI defines if the Zaptel define is present. Also, there is no such thing as DAHDI_PRI.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@158629 f38db490-d61c-443f-a65b-d21fe96a405b
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merging the fix into 1.6.0 and realized I missed
the code in the h-exten block, and didn't catch it
because my test case had the h-exten commented out.
So, this corrects the code I missed, as a
preventative against another crash report.
Tested with the h-exten defined, all is well.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@158603 f38db490-d61c-443f-a65b-d21fe96a405b
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because we strip spaces when copying the extension into the structure.
Therefore, use the copied item to place the item into the list.
(found by lmadsen on -dev, fixed by me)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@158600 f38db490-d61c-443f-a65b-d21fe96a405b
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ao2_lock/ao2_unlock
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Reported by: mdu113
This one is totally my fault. The code doesn't even
create a bridge if the channel CDR has POST_DISABLED.
I didn't check for that at the end of the bridge.
Fixed with a few small insertions. Tested. Looks
good. No cdr generated, no crash, no unnecc. data
objects created either.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@158483 f38db490-d61c-443f-a65b-d21fe96a405b
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completely unrelated commit. Thanks to Theo Belder
on the Asterisk-dev list for pointing this out.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@158306 f38db490-d61c-443f-a65b-d21fe96a405b
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frame in a freed ast_filestream. This patch makes use of the
ao2 functions to make sure that we do not free an ast_filestream
structure until the embedded ast_frame has been "freed" as well.
(closes issue #13496)
Reported by: fst-onge
Patches:
filestream_frame_1_4.diff uploaded by putnopvut (license 60)
Tested by: putnopvut
Closes AST-89
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@158126 f38db490-d61c-443f-a65b-d21fe96a405b
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section 15 of RFC 3261, we should terminate a dialog if we
receive a 481 or 408 in response to our BYE. Since I am aware
of at least one phone manufacturer who may sometimes send a
404 as well, I am being liberal and saying that any 4XX response
to a BYE should result in a terminated dialog.
(closes issue #12994)
Reported by: pabelanger
Patches:
12994.patch uploaded by putnopvut (license 60)
Closes AST-129
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@158071 f38db490-d61c-443f-a65b-d21fe96a405b
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that ast_call() fails. For incoming SIP channels, this was causing
us to send a 603 instead of a 486 when the call-limit was reached on
the destination channel.
(closes issue #13867)
Reported by: still_nsk
Patches:
13867.diff uploaded by putnopvut (license 60)
Tested by: blitzrage
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@158053 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/team/group/issue8824
........
Fixes JIRA ABE-1726
The dial extension could be empty if you are using MISDN_KEYPAD
to control ISDN provider features.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@158010 f38db490-d61c-443f-a65b-d21fe96a405b
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variables, unreachable code, etc.), which is good. however, developers usually compile with the optimizer turned off, because if they need to debug the resulting code, optimized code makes that process very difficult. this means that we get code changes committed that weren't adequately checked over for these sorts of problems.
with this build system change, if (and only if) --enable-dev-mode was used and DONT_OPTIMIZE is turned on, when a source file is compiled it will actually be preprocessed (into a .i or .ii file), then compiled once with optimization (with the result sent to /dev/null) and again without optimization (but only if the first compile succeeded, of course).
while making these changes, i did some cleanup work in Makefile.rules to move commonly-used combinations of flag variables into their own variables, to make the file easier to read and maintain
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@157859 f38db490-d61c-443f-a65b-d21fe96a405b
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function. Not setting the invite state correctly on the call was
resulting in the Record application leaving empty files. I also
have updated the doxygen comment next to the declaration of the
INV_EARLY_MEDIA constant to reflect that we also use this state
when we *send* a 18X response to an INVITE.
(closes issue #13878)
Reported by: nahuelgreco
Patches:
sip-early-media-recording-1.4.22.patch uploaded by nahuelgreco (license 162)
Tested by: putnopvut
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@157503 f38db490-d61c-443f-a65b-d21fe96a405b
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Reported by: akkornel
This fix is the result of a bug fix in ast_app_separate_args r124395. If an argument does not exist it should always be set to a null string rather than a null pointer.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@157365 f38db490-d61c-443f-a65b-d21fe96a405b
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app_followme which would occur at the end of an attended
transfer. The error occurred because we initially stored
a pointer to an ast_channel which then was hung up due
to a masquerade.
This commit adds a "fixup" callback to the bridge_config
structure to allow for end_bridge_callback_data to be
changed in the case that a new channel pointer is needed
for the end_bridge_callback.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@157305 f38db490-d61c-443f-a65b-d21fe96a405b
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first, and when running top-level dist-clean, do not run subdirectory clean operations twice
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@157162 f38db490-d61c-443f-a65b-d21fe96a405b
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1) update to doxygen 1.5.x style file, as used in trunk
2) tell doxygen where are header files are, so include-file processing can be done
3) make all macros that are used to define variables/functions be expanded, so that doxygen will properly document the resulting variable/function
4) make all macros that are used to provide the contents of a variable (structure) be expanded, so that doxygen will be able to document the resulting fields
5) suppress compiler attributes (__attribute__(xxx)) from being seen by doxygen, so it will properly match up function definition and usage (for an example of th effect of this, look at the doxygen docs for ast_log() from before and afte this commit)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@157104 f38db490-d61c-443f-a65b-d21fe96a405b
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resulted in the password not being saved, even if the input matched
what you gave when first prompted to enter a new password. This is
because the return value of ast_readstring was checked, but not checked
properly.
This bug was discovered by Jared Smith during an Asterisk training course.
Thanks for reporting it!
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@156816 f38db490-d61c-443f-a65b-d21fe96a405b
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reason. This prevents While/EndWhile from working within the "h"
extension.
Reported by: jgalarneau (for ABE C.2)
Fixed by: me
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@156755 f38db490-d61c-443f-a65b-d21fe96a405b
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channel name.
(closes issue #13398)
Reported by: bamby
Patches:
manager.c.diff uploaded by bamby (license 430)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@156688 f38db490-d61c-443f-a65b-d21fe96a405b
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instead.
(closes issue #13851)
Reported by: ruddy
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@156386 f38db490-d61c-443f-a65b-d21fe96a405b
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N, X, and Z are off by one, as per conversation with
jsmith on #asterisk-dev; he was teaching a class
and disconcerted that this published rule was not
being followed, with patterns _NXX, _[1-8]22 and
_[2-9]22... and NXX was winning, but [1-8] should
have been.
This change, tested on these 3 patterns now
picks the proper one.
However, this change may surprise users who
set up dialplans based on previous behavior,
which has been there for what, 2 and half
years or so now.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@156297 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #13889)
Reported by: eliel
Patches:
app_meetme.c.patch uploaded by eliel (license 64)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@156294 f38db490-d61c-443f-a65b-d21fe96a405b
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uninitialized variable when compiling 1.6.1.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@156289 f38db490-d61c-443f-a65b-d21fe96a405b
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to be sent, and instead, schedule a task to destroy the iax2 pvt structure
10 seconds later. This allows the IAX2 HANGUP packet to be queued,
transmitted, and ACKed before the pvt is destroyed.
(closes issue #13645)
Reported by: dzajro
Patches:
20081111__bug13645__3.diff.txt uploaded by Corydon76 (license 14)
Tested by: vazir
Reviewed: http://reviewboard.digium.com/r/51/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@156229 f38db490-d61c-443f-a65b-d21fe96a405b
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Reported by: pep
This change adds an announce_thread responsible for playing announcements to an existing conference. This allows all announcing to be immediately stopped if necessary but more importantly allows other threads that need to play something to not block. There are multiple benefits to this, but the actual bug is for solving the scenario for a channel to be unusable after hang up for the entire duration of the parting announcement. The parting announcement can be extremely long depending on what the user recorded upon joining the conference.
Reviewed by Russell on Review Board:
http://reviewboard.digium.com/r/25/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@156178 f38db490-d61c-443f-a65b-d21fe96a405b
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if an attended transfer occurred during the call. I traced the cause to
the CDR on one of the channels being NULL. murf suggested a check in
the end bridge callback to be sure the CDR is non-NULL before proceeding,
so that's what I'm adding.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@156167 f38db490-d61c-443f-a65b-d21fe96a405b
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console option to be after the code that reads options from asterisk.conf.
This resolves a situation where Asterisk can start taking up 100% when
misconfigured.
(Thanks to Bryce Porter (x86 on IRC) for letting me log in to his system to
figure out what was causing the 100% CPU problem.)
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@156164 f38db490-d61c-443f-a65b-d21fe96a405b
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is invoked, that the channel on which the callback was called
is locked. This patch corrects an instance in chan_agent where
a channel's indicate callback is called directly without first
locking the channel.
This was leading to some observed locking issues in chan_local,
but considering that all channel drivers operate under the
same expectations, the generic fix in chan_agent is the right
way to go.
AST-126
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