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(closes issue #12084)
Reported by: mmickan
Patches:
chan_vpb.cc.diff uploaded by mmickan (license 400)
module.h.diff uploaded by mmickan (license 400)
vpb.conf.sample uploaded by mmickan (license 400)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104502 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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automatically generated file like it used to be. This still needs to be there
for modules that have to check it to compile against multiple asterisk versions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104244 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #11170)
Reported by: kratzers
Patches:
ResetCDR.1.diff uploaded by kratzers (license 307)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@104215 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r104119 | russell | 2008-02-25 18:25:29 -0600 (Mon, 25 Feb 2008) | 33 lines
Merge changes from team/russell/smdi-1.4
This commit brings in a significant set of changes to the SMDI support in Asterisk.
There were a number of bugs in the current implementation, most notably being that
it was very likely on busy systems to pop off the wrong message from the SMDI message
queue. So, this set of changes fixes the issues discovered as well as introducing
some new ways to use the SMDI support which are required to avoid the bugs with
grabbing the wrong message off of the queue.
This code introduces a new interface to SMDI, with two dialplan functions. First,
you get an SMDI message in the dialplan using SMDI_MSG_RETRIEVE() and then you access
details in the message using the SMDI_MSG() function. A side benefit of this is that
it now supports more than just chan_zap.
For example, with this implementation, you can have some FXO lines being terminated
on a SIP gateway, but the SMDI link in Asterisk.
Another issue with the current implementation is that it is quite common that the
station ID that comes in on the SMDI link is not necessarily the same as the Asterisk
voicemail box. There are now additional directives in the smdi.conf configuration
file which let you map SMDI station IDs to Asterisk voicemail boxes.
Yet another issue with the current SMDI support was related to MWI reporting over
the SMDI link. The current code could only report a MWI change when the change
was made by someone calling into voicemail. If the change was made by some other
entity (such as with IMAP storage, or with a web interface of some kind), then the
MWI change would never be sent. The SMDI module can now poll for MWI changes if
configured to do so.
This work was inspired by and primarily done for the University of Pennsylvania.
(also related to issue #9260)
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privilege to call out to a subshell.
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code. I will continue to work on it in a separate branch to make it stable for a release and test it against the following issues.
(closes issue #11253)
Reported by: falves11
(closes issue #11657)
Reported by: davevg
(closes issue #11033)
Reported by: falves11
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Separate fax detection from digit detected.
Added CED (called) tone detection for fax (previously, only CNG (calling) was supported).
Separate DTMF/MF code paths where appropriate.
Allow detection of arbitary tones.
(closes issue #11796)
Reported by: dimas
Patches:
v6-dsp-faxtones.patch uploaded by dimas (license 88)
Tested by: dimas, IgorG, Cache
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channel. This uses control frames (so they can even pass across IAX2) to negotiate T38 and provided a way of getting the current status of T38 using queryoption. This should by no means cause any issues and if it does I will take responsibility for it.
(closes issue #11873)
Reported by: dimas
Patches:
v4-t38-api.patch uploaded by dimas (license 88)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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does an async goto instead of an explicit goto.
(closes issue #11753)
Reported by: johan
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(closes issue #11268)
Reported by: moy
Patches:
chantrace-datastored-encapsulated-rev94934.patch uploaded by moy (license 222)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103754 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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1024 was used instead of 128 when using AES from OpenSSL. Many thanks
to d1mas for figuring this one out!
(closes issue #11946)
Reported by: bbhoss
Patches:
v1-11946.patch uploaded by dimas (license 88)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103739 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #11996)
Patches:
bug_11996_doxygen.diff uploaded by snuffy (license 35)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103723 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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found on Solaris (it is incompatible).
Also removes second check for awk, which causes Solaris to find an incompatible version of awk.
(closes issue #11829)
Reported by: snuffy
Patches:
bug-11829.diff uploaded by snuffy (license 35)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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integer. Modify calls to various ast_*_flag macros accordingly.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103341 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Added ability to retrieve list of categories in a config file.
Added ability to retrieve the content of a particular category.
Added ability to empty a context.
Created new action to create a new file.
Updated delete action to allow deletion by line number with respect to category.
Added new action insert to add new variable to category at specified line.
Updated action newcat to allow new category to be inserted in file above another existing category.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r102323 | tilghman | 2008-02-04 15:06:09 -0600 (Mon, 04 Feb 2008) | 7 lines
Cross-platform fix: OS X now deprecates the use of the daemon(3) API.
(closes issue #11908)
Reported by: oej
Patches:
20080204__bug11908.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
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Fix it up!
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r101894 | tilghman | 2008-02-01 13:36:12 -0600 (Fri, 01 Feb 2008) | 2 lines
Change detection of getifaddrs to use AST_C_COMPILE_CHECK, backported from trunk (as suggested by kpfleming)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r101772 | tilghman | 2008-02-01 09:55:58 -0600 (Fri, 01 Feb 2008) | 2 lines
Compatibility fix for OpenWRT (reported by Brian Capouch via the mailing list)
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- change function to void because it always returned the same value and no one read it.
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suggestion from (const char *) Kevin. Thanks!
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with libexecinfo will generate inline stack backtraces correctly.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r100581 | russell | 2008-01-28 11:15:41 -0600 (Mon, 28 Jan 2008) | 9 lines
Make some deadlock related fixes. These bugs were discovered and reported
internally at Digium by Steve Pitts.
- Fix up chan_local to ensure that the channel lock is held before the local
pvt lock.
- Don't hold the channel lock when executing the timing function, as it can
cause a deadlock when using chan_local. This actually changes the code back
to be how it was before the change for issue #10765. But, I added some other
locking that I think will prevent the problem reported there, as well.
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that was just merged from 1.4, so this is a changeover to those APIs to use the
macro versions, so that we properly detect errors from ast_sched_del, instead
of simply ignoring the return values.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r100465 | tilghman | 2008-01-27 15:59:53 -0600 (Sun, 27 Jan 2008) | 11 lines
When deleting a task from the scheduler, ignoring the return value could
possibly cause memory to be accessed after it is freed, which causes all
sorts of random memory corruption. Instead, if a deletion fails, wait a
bit and try again (noting that another thread could change our taskid
value).
(closes issue #11386)
Reported by: flujan
Patches:
20080124__bug11386.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, flujan, stuarth`
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the dialing structure does not hang it up.
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would be useful to be able to pass it between functions
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r100264 | kpfleming | 2008-01-24 15:57:41 -0600 (Thu, 24 Jan 2008) | 2 lines
make these macros not assume that the only other field in the structure is 'argc'... this is true when someone uses AST_DECLARE_APP_ARGS, but it's perfectly reasonable to define your own structure as long as it has the right fields
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(closes issue #5768)
Reported by: mguesdon
Patches:
res_config_ldap-v0.7.tar.gz uploaded by mguesdon (license 121)
res_ldap.conf.sample uploaded by suretec (license 70)
asterisk-v3.1.4.ldif uploaded by suretec (license 70)
asterisk-v3.1.4.schema uploaded by suretec (license 70)
Tested by: oej, mguesdon, suretec, cthorner
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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as a channel variable BRIDGEPVTCALLID
This is important for call tracing in log files and CDRs, so that
the SIP callID can be traced along servers.
The CHANNEL dialplan function won't work here, since the outbound
channel is gone when we need the Call-ID.
Other channel drivers may now implement the same function :-),
but this patch only supports chan_sip.so.
Inspired by (issue #11816)
Reported by: ctooley
Patch by oej
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r99341 | tilghman | 2008-01-21 12:11:07 -0600 (Mon, 21 Jan 2008) | 8 lines
Permit the user to specify number of seconds that a connection may remain idle,
which fixes a crash on reconnect with the MyODBC driver.
(closes issue #11798)
Reported by: Corydon76
Patches:
20080119__res_odbc__idlecheck.diff.txt uploaded by Corydon76 (license 14)
Tested by: mvanbaak
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This set of changes introduces TCP and TLS support for chan_sip. There are various
new options in configs/sip.conf.sample that are used to enable these features. Also,
there is a document, doc/siptls.txt that describes some things in more detail.
This code was implemented by Brett Bryant and James Golovich. It was reviewed
by Joshua Colp and myself. A number of other people participated in the testing
of this code, but since it was done outside of the bug tracker, I do not have their
names. If you were one of them, thanks a lot for the help!
(closes issue #4903, but with completely different code that what exists there.)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r99081 | russell | 2008-01-18 15:37:21 -0600 (Fri, 18 Jan 2008) | 9 lines
Revert adding the packed attribute, as it really doesn't make sense why that
would do any good. Fix the real bug, which is to do the check to see if the
frame came from a translator at the beginning of ast_frame_free(), instead of
at the end. This ensures that it always gets checked, even if none of the
parts of the frame are malloc'd, and also ensures that we aren't looking at
free'd memory in the case that it is a malloc'd frame.
(closes issue #11792, reported by explidous, patched by me)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r99079 | russell | 2008-01-18 15:22:21 -0600 (Fri, 18 Jan 2008) | 4 lines
Since we're relying on the offset between the frame and the beginning of the translator
pvt struct, set the packed attribute to make sure we get to the right place.
(potential fix for issue #11792)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r99004 | russell | 2008-01-17 16:37:22 -0600 (Thu, 17 Jan 2008) | 10 lines
Have IAX2 optimize the codec translation path just like chan_sip does it. If
the caller's codec is in our codec list, move it to the top to avoid transcoding.
(closes issue #10500)
Reported by: stevedavies
Patches:
iax-prefer-current-codec.patch uploaded by stevedavies (license 184)
iax-prefer-current-codec.1.4.patch uploaded by stevedavies (license 184)
Tested by: stevedavies, pj, sheldonh
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recommendations from Russell.
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rizzo, do you want to discuss so we can rethink this, or do you have another way?
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98951 | file | 2008-01-15 21:13:27 -0400 (Tue, 15 Jan 2008) | 4 lines
Add autoconf logic for speexdsp. Later versions use a separate library for some things so we need to use it if present in codec_speex.
(closes issue #11693)
Reported by: yzg
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r98943 | russell | 2008-01-15 17:26:52 -0600 (Tue, 15 Jan 2008) | 25 lines
Commit a fix for some memory access errors pointed out by the valgrind2.txt
output on issue #11698.
The issue here is that it is possible for an instance of a translator to get
destroyed while the frame allocated as a part of the translator is still being
processed. Specifically, this is possible anywhere between a call to ast_read()
and ast_frame_free(), which is _a lot_ of places in the code. The reason this
happens is that the channel might get masqueraded during this time. During a
masquerade, existing translation paths get destroyed.
So, this patch fixes the issue in an API and ABI compatible way. (This one is
for you, paravoid!)
It changes an int in ast_frame to be used as flag bits. The 1 bit is still used
to indicate that the frame contains timing information. Also, a second flag has
been added to indicate that the frame came from a translator. When a frame with
this flag gets released and has this flag, a function is called in translate.c to
let it know that this frame is doing being processed. At this point, the flag gets
cleared. Also, if the translator was requested to be destroyed while its internal
frame still had this flag set, its destruction has been deffered until it finds out
that the frame is no longer being processed.
Admittedly, this feels like a hack. But, it does fix the issue, and I was not able
to think of a better solution ...
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r97847 | qwell | 2008-01-10 14:12:37 -0600 (Thu, 10 Jan 2008) | 1 line
Fix a comment that is no longer true.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r97734 | russell | 2008-01-10 10:10:09 -0600 (Thu, 10 Jan 2008) | 4 lines
Remove pbx_kdeconsole from the tree. It hasn't worked in ages, and nobody has
complained.
(closes issue #11706, reported by caio1982)
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