The issue that was reported was about a case where a RINGING channel got
redirected to an extension to pick up a call from parking. Once the parked
call got taken out of parking, it heard silence until the other side answered.
Ideally, the caller that was parked would get a ringing indication. This patch
fixes this case so that the caller receives ringback once it comes out of
parking until the other side answers.
The fixes are:
- Make sure we remember that a channel was an outgoing channel when doing
a masquerade. This prevents an erroneous ast_answer() call on the channel,
which causes a bogus 200 OK to be sent in the case of SIP.
- Add some additional comments to explain related parts of code.
- Update the handling of the ast_channel visible_indication field. Storing
values that are not stateful is pointless. Control frames that are events
or commands should be ignored.
- When a bridge first starts, check to see if the peer channel needs to be
given ringing indication because the calling side is still ringing.
- Rework ast_indicate_data() a bit for the sake of readability.
(closes issue #13747)
Reported by: davidw
Tested by: russell
Review: http://reviewboard.digium.com/r/90/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@164201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
pointer inside editline to look back to asterisk.c, so others don't spend
as much time as I did looking (in the wrong place) for the appropriate
function.
Reported by: ZX81, via the #asterisk-users channel
Fixed by: me (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@163761 65c4cc65-6c06-0410-ace0-fbb531ad65f3
These changes come from team/russell/issue_12658
1) Change autoservice to put digits on the head of the channel's frame readq
instead of the tail. If there were frames on the readq that autoservice
had not yet read, the previous code would have resulted in out of order
processing. This required a new API call to queue a frame to the head
of the queue instead of the tail.
2) Change up the processing of DTMF in ast_read(). Some of the problems
were the result of having two sources of pending DTMF frames. There
was the dtmfq and the more generic readq. Both were used for pending
DTMF in various scenarios. Simplifying things to only use the frame
readq avoids some of the problems.
3) Fix a bug where a DTMF END frame could get passed through when it
shouldn't have. If code set END_DTMF_ONLY in the middle of digit emulation,
and a digit arrived before emulation was complete, digits would get
processed out of order.
(closes issue #12658)
Reported by: dimas
Tested by: russell, file
Review: http://reviewboard.digium.com/r/85/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@163448 65c4cc65-6c06-0410-ace0-fbb531ad65f3
is messed up. By intercepting those events with a signal handler in the remote
console, we can avoid those issues.
(closes issue #13464)
Reported by: tzafrir
Patches:
20081110__bug13464.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@163383 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The test is not valid. Besides, if we actually suspected that recursive
mutexes were not working, we would get a ton of LOG_ERROR messages when
DEBUG_THREADS is turned on.
(inspired by a discussion on the asterisk-dev list)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@162413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The previous code carried over group settings from the old channel to the new
one. However, it did nothing with the group settings that were already on the
new channel. This patch removes all group settings that already existed on the
new channel.
I have a more complicated version of this patch which addresses only the most
blatant problem with this, which is that a channel can end up with multiple
group settings in the same category. However, I could not think of a use case
for keeping any of the group settings from the old channel, so I went this route
for now.
(closes AST-152)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@161948 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: matt_b
Tested by: jpeeler
This mirrors a check that was present in ast_rtp_read to also be in ast_rtp_raw_write to not schedule sending the receiver report if the remote RTCP endpoint address isn't present in the RTCP structure.
Closes AST-142.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@161013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
following: "name <number>" (including the quotation marks), then the parts
would be parsed as
name: "name
number: number
This is because the closing quotation mark was not discovered since the number
and everything after was parsed out of the string earlier. Now, there is a check
to see if the closing quote occurs after the number, so that we can know if we
should strip off the opening quote on the name.
Closes AST-158
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@160943 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@159808 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@159316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@158600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@158126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@157859 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@156297 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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: https://origsvn.digium.com/svn/asterisk/branches/1.4@156164 65c4cc65-6c06-0410-ace0-fbb531ad65f3
deadlock when someone tries to initiate a module reload from the AMI just
as Asterisk is starting.
(closes issue #13778)
Reported by: hotsblanc
Fix suggested by hotsblanc
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@151905 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A comment was made in bug #13726
"3. The same mistake as in (2) is done in a few other places in the code that check for: #if defined(HAVE_ZAPTEL) || defined(HAVE_DAHDI)
Harmless, but still incorrect."
In the case of main/asterisk.c, this is not incorrect because without HAVE_ZAPTEL defined, we're missing
the include for ioctl and the namespace that defines DAHDI_TIMERCONFIG which is still required when
using Zaptel with the 1.4 branch.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@151167 65c4cc65-6c06-0410-ace0-fbb531ad65f3
We'll create a bogus channel for the function call and destroy it when we're done. If we have trouble allocating the bogus channel then we're not going to try executing the function call at all and run the risk of crashing.
(closes issue #13715)
reported by: makoto
patch by: bweschke
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@150816 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Don't always define HAVE_ZAPTEL_CHANALARMS (since we check if it's defined..)
Minor cleanup to make things clear.
(closes issue #13726)
Reported by: tzafrir
Patches:
dahdi_def.diff uploaded by tzafrir (license 46)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@150557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
I was mistakenly under the assumption that dialplan functions
*always* required that a channel be present. I need to go
home earlier, I think :)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@150304 65c4cc65-6c06-0410-ace0-fbb531ad65f3
the manager's GetVar handler if an invalid channel has
been specified. Several dialplan functions, including
CHANNEL and SIP_HEADER, do not check for NULL-ness of
the channel being passed in.
(closes issue #13715)
Reported by: makoto
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@150298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
so that if packetization of audio is close (but not equal)
we don't end up flushing the audiohooks over small
inconsistencies in synchronization.
Related to issue #13005, and solves the issue
for most people who were experiencing the problem.
However, a small number of people are still experiencing
the problem on long calls, so I am not closing
the issue yet
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@149204 65c4cc65-6c06-0410-ace0-fbb531ad65f3
channels/misdn/isdn_lib.c
* Miscellaneous other fixes from trunk to make merging easier later.
........
r145200 | rmudgett | 2008-09-30 16:00:54 -0500 (Tue, 30 Sep 2008) | 7 lines
* Miscellaneous formatting changes to make v1.4 and trunk
more merge compatible in the mISDN area.
channels/chan_misdn.c
* Eliminated redundant code in cb_events() EVENT_SETUP
........
r144257 | crichter | 2008-09-24 03:42:55 -0500 (Wed, 24 Sep 2008) | 9 lines
improved helptext of misdn_set_opt.
........
r142181 | rmudgett | 2008-09-09 12:30:52 -0500 (Tue, 09 Sep 2008) | 1 line
Cleaned up comment
........
r138738 | rmudgett | 2008-08-18 16:07:28 -0500 (Mon, 18 Aug 2008) | 30 lines
channels/chan_misdn.c
* Made bearer2str() use allowed_bearers_array[]
* Made use the causes.h defines instead of hardcoded numbers.
* Made use Asterisk presentation indicator values if either of the
mISDN presentation or screen options are negative.
* Updated the misdn_set_opt application option descriptions.
* Renamed the awkward Caller ID presentation misdn_set_opt
application option value not_screened to restricted.
Deprecated the not_screened option value.
channels/misdn/isdn_lib.c
* Made use the causes.h defines instead of hardcoded numbers.
* Fixed some spelling errors and typos.
* Added all defined facility code strings to fac2str().
channels/misdn/isdn_lib.h
* Added doxygen comments to struct misdn_bchannel.
channels/misdn/isdn_lib_intern.h
* Added doxygen comments to struct misdn_stack.
channels/misdn_config.c
configs/misdn.conf.sample
* Updated the mISDN presentation and screen parameter descriptions.
doc/misdn.txt (doc/tex/misdn.tex)
* Updated the misdn_set_opt application option descriptions.
* Fixed some spelling errors and typos.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@145293 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- it is no longer necessary to forcibly include asterisk/autoconfig.h; every module already includes asterisk.h as its first header (even before system headers), which serves the same purpose
- astmm.h is now included by asterisk.h when needed, instead of being forced by the Makefile; this means external modules will build properly against installed headers with MALLOC_DEBUG enabled
- simplify the usage of some of these headers in the AEL-related stuff in the utils directory
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@144924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
does not apply since the "found" pointer is not
passed in to this function. If this is going to
be backported, it needs to be done differently or
a deeper backport needs to be done.
Edit: This commit reverts commit number 144677.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@144758 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: mnicholson
Patches:
found1.diff uploaded by mnicholson (license 96)
This patch was mainly meant to apply to trunk and 1.6.x,
but I'm applying it to 1.4 also, which should be a perfectly
harmless fix to the vast majority of users who are not using
external switches, but the few who might be affected
will not have to go to the pain of filing a bug report.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@144677 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Steve Murphy
Russell Bryant
Tilghman Lesher
Joshua Colp
Mark Michelson
Sean Bright
Jeff Peeler
Michiel van Baak
Kevin P. Fleming
BJ Weschke
Jason Parker
Richard Mudgett