https://origsvn.digium.com/svn/asterisk/branches/1.4
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r207945 | tilghman | 2009-07-21 17:38:54 -0500 (Tue, 21 Jul 2009) | 8 lines
Force an error if a blank is passed to QUOTE (because the documentation states the argument is not optional).
This change makes URIENCODE and QUOTE behave similarly, since the documentation
states that the argument is not optional, for both.
(closes issue #15439)
Reported by: pkempgen
Patches:
20090706__issue15439.diff.txt uploaded by tilghman (license 14)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@207946 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r207647 | kpfleming | 2009-07-21 08:04:44 -0500 (Tue, 21 Jul 2009) | 12 lines
Ensure that user-provided CFLAGS and LDFLAGS are honored.
This commit changes the build system so that user-provided flags (in ASTCFLAGS
and ASTLDFLAGS) are supplied to the compiler/linker *after* all flags provided
by the build system itself, so that the user can effectively override the
build system's flags if desired. In addition, ASTCFLAGS and ASTLDFLAGS can now
be provided *either* in the environment before running 'make', or as variable
assignments on the 'make' command line. As a result, the use of COPTS and LDOPTS
is no longer necessary, so they are no longer documented, but are still supported
so as not to break existing build systems that supply them when building Asterisk.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@207680 f38db490-d61c-443f-a65b-d21fe96a405b
This is a continuation of revision 885 to LibPRI (Capture and expose the Reverse
Charging Indication IE on ISDN PRI) which added the ability to get/set Reverse
Charging Indication in LibPRI. This patch adds the ability to specify RCI on
the outbound leg of a PRI call from within Asterisk, by prefixing the dialed
number with a capital 'C' like:
...,Dial(DAHDI/g1/C4445556666)
And to read it off an inbound channel:
exten => s,1,Set(RCI=${CHANNEL(reversecharge)})
Thanks again to rmudgett for the thorough review.
(closes issue #13760)
Reported by: mrgabu
Review: https://reviewboard.asterisk.org/r/303/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@204749 f38db490-d61c-443f-a65b-d21fe96a405b
CEL is the new system for logging channel events. This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records. For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.
Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code. Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.
Review: https://reviewboard.asterisk.org/r/239/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@203638 f38db490-d61c-443f-a65b-d21fe96a405b
Move function SYSINFO static documentation to the new AstXML form.
(issue #15245)
Reported by: eliel
Patches:
func_sysinfo_static_conversion.txt uploaded by lmadsen (license 10)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@199374 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r197194 | tilghman | 2009-05-27 14:09:42 -0500 (Wed, 27 May 2009) | 5 lines
Use a different determinator on whether to print the delimiter, since leading fields may be blank.
(closes issue #15208)
Reported by: ramonpeek
Patch by me, though inspired in part by a patch from ramonpeek
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197209 f38db490-d61c-443f-a65b-d21fe96a405b
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes:
- CLI command handlers
- CLI command handler arguments
- AGI command handlers
- AGI command handler arguments
- Dialplan application handler arguments
- Speech engine API function arguments
In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing.
Review: https://reviewboard.asterisk.org/r/251/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@196072 f38db490-d61c-443f-a65b-d21fe96a405b
This branch adds additional methods to dialplan functions, whereby the result
buffers are now dynamic buffers, which can be expanded to the size of any
result. No longer are variable substitutions limited to 4095 bytes of data.
In addition, the common case of needing buffers much smaller than that will
enable substitution to only take up the amount of memory actually needed.
The existing variable substitution routines are still available, but users
of those API calls should transition to using the dynamic-buffer APIs.
Reviewboard: http://reviewboard.digium.com/r/174/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@191140 f38db490-d61c-443f-a65b-d21fe96a405b
* Wait for a DivertingLegInformation3 message after receiving a
DivertingLegInformation1 message to complete the redirecting-to information
before queuing a redirecting update to the other channel.
* A DivertingLegInformation2 message should be responded to with a
DivertingLegInformation3 when the COLR is determined. If the call
could or does experience another redirection, you should manually
determine the COLR to send to the switch by setting REDIRECTING(to-pres)
to the COLR and setting REDIRECTING(to-num) = ${EXTEN}.
* A DivertingLegInformation2 message must have an original called number
if the redirection count is greater than one. Since Asterisk does
not keep track of this information, we can only indicate that the
number is not available due to interworking.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@190735 f38db490-d61c-443f-a65b-d21fe96a405b
The messages sent by a technology when a connected line update is received
are best determined by the current call state of the channel. The struct
ast_party_connected_line.source value is really only useful as a possible
tracing aid.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@190517 f38db490-d61c-443f-a65b-d21fe96a405b
There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@190423 f38db490-d61c-443f-a65b-d21fe96a405b
There seems to be a bug with old versions of g++ that doesn't allow a structure
member to use the name list. Rename list member to group_list in ast_group_info
and change the few places it is used.
(closes issue #14790)
Reported by: stuarth
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@190057 f38db490-d61c-443f-a65b-d21fe96a405b
Previously, FILE() returned one less character than specified, due to the
terminating NULL. Both the offset and length parameters now behave
identically to the way variable substitution offsets and lengths also work.
(closes issue #14670)
Reported by: BMC
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@182278 f38db490-d61c-443f-a65b-d21fe96a405b
This patch includes a number of changes to the indications API. The primary
motivation for this work was to improve stability. The object management
in this API was significantly flawed, and a number of trivial situations could
cause crashes.
The changes included are:
1) Remove the module res_indications. This included the critical functionality
that actually loaded the indications configuration. I have seen many people
have Asterisk problems because they accidentally did not have an
indications.conf present and loaded. Now, this code is in the core,
and Asterisk will fail to start without indications configuration.
There was one part of res_indications, the dialplan applications, which did
belong in a module, and have been moved to a new module, app_playtones.
2) Object management has been significantly changed. Tone zones are now
managed using astobj2, and it is no longer possible to crash Asterisk by
issuing a reload that destroys tone zones while they are in use.
3) The API documentation has been filled out.
4) The API has been updated to follow our naming conventions.
5) Various bits of code throughout the tree have been updated to account
for the API update.
6) Configuration parsing has been mostly re-written.
7) "Code cleanup"
The code is from svn/asterisk/team/russell/indications/.
Review: http://reviewboard.digium.com/r/149/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@176627 f38db490-d61c-443f-a65b-d21fe96a405b
along the way fix some minor coding style issues in strings.h and add some attribute_pure annotations to functions in the ast_str API
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@169438 f38db490-d61c-443f-a65b-d21fe96a405b
this stops modules from being linked against both sets of libraries on systems that have both installed
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@168734 f38db490-d61c-443f-a65b-d21fe96a405b
A crash was brought up on the bugtracker. The first run through valgrind
was full of legitimate complaints of uninitialized values in func_timeout when
setting a response timeout. These were fixed but the crash persisted.
A second run through showed the real problem. The reference counting used
for filestreams was incorrect because there were some missing increments
when a frame was read from a format module.
(closes issue #14118)
Reported by: blitzrage
Patches:
14118v2.patch uploaded by putnopvut (license 60)
Tested by: blitzrage
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@166267 f38db490-d61c-443f-a65b-d21fe96a405b
This function is being added as a method to allow for
an audiohook to move to a new channel during a channel
masquerade. The most obvious use for such a facility is
for MixMonitor when a transfer is performed. Prior to
the addition of this functionality, if a channel
running MixMonitor was transferred by another party, then
the recording would stop once the transfer had completed.
By using AUDIOHOOK_INHERIT, you can make MixMonitor
continue recording the call even after the transfer
has completed.
It has also been determined that since this is seen
by most as a bug fix and is not an invasive change,
this functionality will also be backported to 1.4 and
merged into the 1.6.0 branches, even though they are
feature-frozen.
(closes issue #13538)
Reported by: mbit
Patches:
13538.patch uploaded by putnopvut (license 60)
Tested by: putnopvut
Review: http://reviewboard.digium.com/r/102/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@166092 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r163253 | russell | 2008-12-11 15:46:29 -0600 (Thu, 11 Dec 2008) | 8 lines
Fix some observed slowdowns in dialplan processing.
The change is to remove autoservice usage from dialplan functions that do not
need it because they do not perform operations that potentially block.
(closes issue #13940)
Reported by: tbelder
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@163254 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r160703 | murf | 2008-12-03 13:41:42 -0700 (Wed, 03 Dec 2008) | 11 lines
(closes issue #13597)
Reported by: john8675309
Patches:
patch.13597 uploaded by murf (license 17)
Tested by: murf, john8675309
This patch causes the setcid func to update the CDR
clid after setting the channel field.
I also notice that in trunk, the num/number of 1.4 is
left out; I decided to include the option to use
either in trunk, so as not to have 1.4 upgraders
not to have problems.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@160760 f38db490-d61c-443f-a65b-d21fe96a405b
because SPRINTF() use non-literal format strings (which cannot be checked), move it into its own module so the rest of func_strings can benefit from format string checking
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@159774 f38db490-d61c-443f-a65b-d21fe96a405b
it would be best to maintain API compatibility. Instead, this commit introduces
ao2_callback_data() which is functionally identical to ao2_callback() except
that it allows you to pass arbitrary data to the callback.
Reviewed by Mark Michelson via ReviewBoard:
http://reviewboard.digium.com/r/64
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@158959 f38db490-d61c-443f-a65b-d21fe96a405b
querying realtime from the dialplan a little more consistent and easy to use.
The original REALTIME function is preserved, for those who are already
accustomed to that interface.
(closes issue #13651)
Reported by: Corydon76
Patches:
20081119__bug13651__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage, Corydon76
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@157870 f38db490-d61c-443f-a65b-d21fe96a405b
- Add CLI aliases module to asterisk.
- Remove all deprecated CLI commands from the code
Initial work done by file.
Junk-Y and lmadsen did a lot of work and testing to
get the list of deprecated commands into the configuration file.
Deprecated CLI commands are now handled by this new module,
see cli_aliases.conf for more info about that.
ok russellb@ via reviewboard
(closes issue #13735)
Reported by: mvanbaak
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@156120 f38db490-d61c-443f-a65b-d21fe96a405b
ao2_callback and ao2_find). Currently, passing OBJ_POINTER to either
of these mandates that the passed 'arg' is a hashable object, making
searching for an ao2 object based on outside criteria difficult.
Reviewed by Russell and Mark M. via ReviewBoard:
http://reviewboard.digium.com/r/36/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@155401 f38db490-d61c-443f-a65b-d21fe96a405b
along the way, change tags used in configure script, menuselect-deps and code for various dependencies to be consistently named
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@154151 f38db490-d61c-443f-a65b-d21fe96a405b
This commit introduces the first phase of an effort to manage documentation of the
interfaces in Asterisk in an XML format. Currently, a new format is available for
applications and dialplan functions. A good number of conversions to the new format
are also included.
For more information, see the following message to asterisk-dev:
http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153365 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r152059 | seanbright | 2008-10-26 16:23:36 -0400 (Sun, 26 Oct 2008) | 7 lines
Since passing \0 as the second argument to strchr is valid (and will
match the trailing \0 of a string) we need to check that first, otherwise
we end up with incorrect results. Fix suggested by reporter.
(closes issue #13787)
Reported by: meitinger
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@152060 f38db490-d61c-443f-a65b-d21fe96a405b
This allows for the ODBC parts to work on OpenBSD as well.
99.99% of the work is done by seanbright (bow, bow) and I actually
did nothing but test and yell at him that it still didn't work :)
Thanks for helping out !
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@146925 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r146799 | tilghman | 2008-10-06 15:52:04 -0500 (Mon, 06 Oct 2008) | 8 lines
Dialplan functions should not actually return 0, unless they have modified the
workspace. To signal an error (and no change to the workspace), -1 should be
returned instead.
(closes issue #13340)
Reported by: kryptolus
Patches:
20080827__bug13340__2.diff.txt uploaded by Corydon76 (license 14)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@146802 f38db490-d61c-443f-a65b-d21fe96a405b
problem within strptime(3), which we are correcting here with ast_strptime().
(closes issue #11040)
Reported by: DEA
Patches:
20080910__bug11040.diff.txt uploaded by Corydon76 (license 14)
Tested by: DEA
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@145649 f38db490-d61c-443f-a65b-d21fe96a405b
Reported by: nickpeirson
The user attached a patch, but the license is not yet
recorded. I took the liberty of finding and replacing
ALL index() calls with strchr() calls, and that
involves more than just main/pbx.c;
chan_oss, app_playback, func_cut also had calls
to index(), and I changed them out. 1.4 had no
references to index() at all.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@144569 f38db490-d61c-443f-a65b-d21fe96a405b
passed directly into the HASH() function. Requested via the -users list, and
committed at Astricon in the Code Zone.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@144199 f38db490-d61c-443f-a65b-d21fe96a405b
when a file is invalid from when a file is missing. This is most important when
we have two configuration files. Consider the following example:
Old system:
sip.conf users.conf Old result New result
======== ========== ========== ==========
Missing Missing SIP doesn't load SIP doesn't load
Missing OK SIP doesn't load SIP doesn't load
Missing Invalid SIP doesn't load SIP doesn't load
OK Missing SIP loads SIP loads
OK OK SIP loads SIP loads
OK Invalid SIP loads incompletely SIP doesn't load
Invalid Missing SIP doesn't load SIP doesn't load
Invalid OK SIP doesn't load SIP doesn't load
Invalid Invalid SIP doesn't load SIP doesn't load
So in the case when users.conf doesn't load because there's a typo that
disrupts the syntax, we may only partially load users, instead of failing with
an error, which may cause some calls not to get processed. Worse yet, the old
system would do this with no indication that anything was even wrong.
(closes issue #10690)
Reported by: dtyoo
Patches:
20080716__bug10690.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@142992 f38db490-d61c-443f-a65b-d21fe96a405b
detecting warnings when we redefine a structure each time it is used, even
though the structure is identical.
Reported by: mvanbaak, via #asterisk-dev
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@141507 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r140488 | mmichelson | 2008-08-29 12:34:17 -0500 (Fri, 29 Aug 2008) | 22 lines
After working on the ao2_containers branch, I noticed
something a bit strange. In all cases where we provide
a callback function to ao2_container_alloc, the callback
function would only return 0 or CMP_MATCH. After inspecting
the ao2_callback() code carefully, I found that if you're
only looking for one specific item, then you should return
CMP_MATCH | CMP_STOP. Otherwise, astobj2 will continue
traversing the current bucket until the end searching for
more matches.
In cases like chan_iax2 where in 1.4, all the peers are
shoved into a single bucket, this makes for potentially
terrible performance since the entire bucket will be
traversed even if the peer is one of the first ones come
across in the bucket.
All the changes I have made were for cases where the
callback function defined was passed to ao2_container_alloc
so that calls to ao2_find could find a unique instance
of whatever object was being stored in the container.
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@140489 f38db490-d61c-443f-a65b-d21fe96a405b
They want (char *)NULL as sentinel.
An example is OpenBSD (confirmed on 4.3) that ships with gcc 3.3.4
This commit introduces a contstant SENTINEL which is declared as:
#define SENTINEL ((char *)NULL)
All places I could test compile on my openbsd system are converted.
Update CODING-GUIDELINES to tell about this constant.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@124127 f38db490-d61c-443f-a65b-d21fe96a405b
function and begin the transition from SIPCHANINFO() to just using CHANNEL().
(closes issue #12856)
Reported by: mostyn
Patches:
iax_and_sip_channel_info.patch uploaded by mostyn (license 398)
(with some additional cleanup by me)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122802 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r122046 | murf | 2008-06-12 07:47:34 -0600 (Thu, 12 Jun 2008) | 37 lines
(closes issue #10668)
Reported by: arkadia
Tested by: murf, arkadia
Options added to forkCDR() app and the CDR() func to
remove some roadblocks for CDR applications.
The "show application ForkCDR" output was upgraded
to more fully explain the inner workings of forkCDR.
The A option was added to forkCDR to force the
CDR system to NOT change the disposition on the
original CDR, after the fork. This involves
ast_cdr_answer, _busy, _failed, and so on.
The T option was added to forkCDR to force
obedience of the cdr LOCKED flag in the
ast_cdr_end, all the disposition changing
funcs (ast_cdr_answer, etc), and in the
ast_cdr_setvar func.
The CHANGES file was updated to explain ALL
the new options added to satisfy this bug report
(and some requests made verbally and via
email, irc, etc, over the past months/year)
The 's' option was added to the CDR() func,
to force it to skip LOCKED cdr's in the
chain.
Again, the new options should be totally transparent
to existing apps! Current behavior of CDR,
forkCDR, and the rest of the CDR system should
not change one little bit. Until you add the
new options, at least!
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@122091 f38db490-d61c-443f-a65b-d21fe96a405b
and normal deviations. Currently this is implemented for chan_sip, but could be added to the func_channel_read callbacks for the CHANNEL function
for any channel that uses RTP.
(closes issue #10590)
Reported by: gasparz
Patches:
chan_sip_c.diff uploaded by gasparz (license 219)
rtp_c.diff uploaded by gasparz (license 219)
rtp_h.diff uploaded by gasparz (license 219)
audioqos-trunk.diff uploaded by snuffy (license 35)
rtpqos-trunk-r119891.diff uploaded by sergee (license 138)
Tested by: jsmith, gasparz, snuffy, marsosa, chappell, sergee
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@120635 f38db490-d61c-443f-a65b-d21fe96a405b
of platform/compiler-dependent warnings when handing
struct timeval fields, both reading and printing them.
It is a lost battle to handle the different ways struct timeval
is handled on the various platforms and compilers, so try
to be pragmatic and go through int/long which are universally
supported.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@116557 f38db490-d61c-443f-a65b-d21fe96a405b
new paramater. The new options for ENUM* functions include 'u', 's', 'i', and 'd' which return the full uri, trigger isn specific rewriting, look
for branches into an infrastructure enum tree, or do a direct dns lookup of a number respectively. The new paramater for TXCIDNAME adds a
zone-suffix argument for looking up caller id's in DNS that aren't e164.arpa.
This patch is based on the original code from otmar, modified by snuffy, and tested by jtodd, me, and others.
(closes issue #8089)
Reported by: otmar
Patches:
20080508_bug8089-1.diff
- original code by otmar (license 480),
- revised by snuffy (license 35)
Tested by: oej, otmar, jtodd, Corydon76, snuffy, alexnikolov, bbryant
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115584 f38db490-d61c-443f-a65b-d21fe96a405b
and denoising to a channel, AGC() and DENOISE(). Also included, is a change
to the audiohook API to add a new function (ast_audiohook_remove) that can
remove an audiohook from a channel before it is detached.
This code is based on a contribution from Switchvox.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114926 f38db490-d61c-443f-a65b-d21fe96a405b
Reported by: oej
Tested by: jpeeler
This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option.
Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114487 f38db490-d61c-443f-a65b-d21fe96a405b
actual problems, per se. I also added format attributes to any printf wrapper functions I found that didn't have them. -Wsecurity and -Wmissing-format-attribute added to --enable-dev-mode.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@109447 f38db490-d61c-443f-a65b-d21fe96a405b
- Remove some magic in unload_module that isn't needed. Module use counts
already ensure that the function isn't going to be in use at this point.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@108034 f38db490-d61c-443f-a65b-d21fe96a405b
a variable from an Asterisk configuration file in the dialplan, or anywhere
else where dialplan functions can be used.
(Inspired by a discussion with Tilghman and Pari)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@107787 f38db490-d61c-443f-a65b-d21fe96a405b
do some additional code cleanup and improvement in passing.
(closes issue #12106)
Reported by: nizon
Patches:
devstate-patch.txt uploaded by nizon (license 415)
-- Updated to trunk, and tab completion added by me
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105461 f38db490-d61c-443f-a65b-d21fe96a405b
(closes issue #8925)
About a year ago, as Leif Madsen and Jim van Meggelen were going over the CLI
commands in Asterisk 1.4 for the next version of their book, they documented
a lot of inconsistencies. This set of changes addresses all of these issues
and has been reviewed by Leif.
While this does introduce even more changes to the CLI command structure, it
makes everything consistent, which is the most important thing.
Thanks to all that helped with this one!
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103171 f38db490-d61c-443f-a65b-d21fe96a405b
I just added the ability to set multiple fields at once after discussions with
Tilghman and Russell. Currently limited to 30 fields.
(closes issue #11887)
Reported by: sergee
Patches:
rt-func-store-destroy-multivalue.diff uploaded by otherwiseguy (license 396)
Tested by: sergee, otherwiseguy
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@102700 f38db490-d61c-443f-a65b-d21fe96a405b
based on configuration templates that use Asterisk dialplan function and
variable substitution. It should be possible to create phone profiles and
templates that work for the majority of phones provisioned over http. It
is currently only intended to provision a single user account per phone.
An example profile and set of templates for Polycom phones is provided.
NOTE: Polycom firmware is not included, but should be placed in
AST_DATA_DIR/phoneprov/configs to match up with the included templates.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@97634 f38db490-d61c-443f-a65b-d21fe96a405b
revision changed, every module that used the version was getting rebuilt after
every svn update. This severly annoyed me pretty quickly, so I have improved
the situation.
Now, instead of generating version.h, main/version.c is generated. version.c
includes the version information, as well as a couple of API calls for modules
to retrieve the version. So now, only version.c will get rebuilt, and the main
asterisk binary relinked, which is must faster than rebuilding http.c, manager.c,
asterisk.c, relinking the asterisk binary, chan_sip.c, func_version.c, res_agi ...
The only minor change in behavior here is that the version information reported by
chan_sip, for example, is the version of the Asterisk core, and not necessarily the
Asterisk version that the chan_sip module came from.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@96717 f38db490-d61c-443f-a65b-d21fe96a405b
iconv dependency for func_iconv.
This fixes some build issues on CYGWIN and FreeBSD and probably
other platforms where libiconv is not there by default
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@95624 f38db490-d61c-443f-a65b-d21fe96a405b
tilghman