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https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r286115 | twilson | 2010-09-10 15:35:25 -0500 (Fri, 10 Sep 2010) | 23 lines
Merged revisions 286059 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010) | 16 lines
Inherit CHANNEL() writes to both sides of a Local channel
Having Local (/n) channels as queue members and setting the language in the
extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2
channel. Hold time report playbacks happen on the Local/...,1 channel and
therefor do not play in the specified language.
This patch modifies func_channel_write to call the setoption callback and pass
the CHANNEL() write info to the callback. chan_local uses this information to
look up the other side of the channel and apply the same changes to it.
(closes issue #17673)
Reported by: Guggemand
Review: https://reviewboard.asterisk.org/r/903/
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https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010) | 8 lines
Fixes an issue with dialplan pattern matching where the specificity for pattern ranges and pattern special characters was inconsistent.
(closes issue #16903)
Reported by: Nick_Lewis
Patches:
pbx.c-specificity.patch uploaded by Nick Lewis (license 657)
Tested by: Nick_Lewis
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@285711 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17336)
Reported by: snuffy
Patches:
doxygen-fixes1.diff uploaded by snuffy (license 35)
Tested by: russell
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@268969 f38db490-d61c-443f-a65b-d21fe96a405b
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function can't be written to.
This patch removes code that was duplicated from pbx.c to manager.c
in order to prevent API change in released versions of Asterisk.
There are propably also other places that would benefit from reading the
return code and react if a function returns error codes on writing a value into it.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@242919 f38db490-d61c-443f-a65b-d21fe96a405b
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Specifically, by setting TESTTIME() to a particular date and time, you
can test whether a dialplan correctly branches as was intended. This was
developed after recent questions on the -users list on how to test their
holiday dialplan logic.
(closes issue #16464)
Reported by: tilghman
Patches:
20100112__issue16464.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/458/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@239957 f38db490-d61c-443f-a65b-d21fe96a405b
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should be resolved with a simple include of frame_defs.h
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227645 f38db490-d61c-443f-a65b-d21fe96a405b
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Reviewboard: https://reviewboard.asterisk.org/r/416/
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r205409 | dvossel | 2009-07-08 16:35:12 -0500 (Wed, 08 Jul 2009) | 6 lines
moving ast_devstate_to_extenstate to pbx.c from devicestate.c
ast_devstate_to_extenstate belongs in pbx.c. This change
fixes a compile time error with chan_vpb as well.
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'channel originate ... application <app>' CLI command.
(And yeah, I cleaned up some whitespace in res_clioriginate.c... big whoop,
wanna fight about it!?)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@196758 f38db490-d61c-443f-a65b-d21fe96a405b
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A new xml element was created to manage the AMI actions documentation,
using AstXML.
To register a manager action using XML documentation it is now possible
using ast_manager_register_xml().
The CLI command 'manager show command' can be used to show the parsed
documentation.
Example manager xml documentation:
<manager name="ami action name" language="en_US">
<synopsis>
AMI action synopsis.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(...)" /> <-- for ActionID
<parameter name="header1" required="true">
<para>Description</para>
</parameter>
...
</syntax>
<description>
<para>AMI action description</para>
</description>
<see-also>
...
</see-also>
</manager>
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@196308 f38db490-d61c-443f-a65b-d21fe96a405b
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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
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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
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The F option to app_dial has been modified to accept no parameters and perform
the above functionality. I don't see anywhere else that is doing function
overloading, but this really is the best place for this operation because:
- It makes it close to the 'g' option in the argument list which provides
similar functionality.
- The existing code to support the current F option provides a very
convienient location to add this new feature.
(closes issue #12381)
Reported by: michael-fig
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@187491 f38db490-d61c-443f-a65b-d21fe96a405b
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Included is a small bugfix to an ast_str helper, but most of these changes
are simply doxygen fixes.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@185912 f38db490-d61c-443f-a65b-d21fe96a405b
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Copied from my review board description:
This is a continuation of the API changes documentation started for describing
changes between releases. Most of the API changes were pretty simple needing
only to be brought to attention via the new "Asterisk API Changes" list.
However, if you see anything that needs further explanation feel free to
supplement what is there. The current method of documenting is to add (in the
header file): \version <ver number> <description of changes> and then to add
the function to the change list in doxyref.h on the AstAPIChanges page. I also
made sure all the functions that were newly added were tagged with \since
1.6.1. I think this is a good habit to start both for the historical aspect as
well as for the future ability to easily add a "New Asterisk API" page.
Review: http://reviewboard.digium.com/r/190/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@180719 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
In order to merge this 1.4 patch into trunk,
I had to resolve some conflicts and wait for
Russell to make some changes to res_agi.
I re-ran all the tests; 39 calls in all, and
made fairly careful notes and comparisons: I
don't want this to blow up some aspect of
asterisk; I completely removed the KEEPALIVE
from the pbx.h decls. The first 3 scenarios
involving feature park; feature xfer to 700;
hookflash park to Park() app call all behave
the same, don't appear to leave hung channels,
and no crashes.
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r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | 131 lines
This merges the masqpark branch into 1.4
These changes eliminate the need for (and use of)
the KEEPALIVE return code in res_features.c;
There are other places that use this result code
for similar purposes at a higher level, these appear
to be left alone in 1.4, but attacked in trunk.
The reason these changes are being made in 1.4, is
that parking ends a channel's life, in some situations,
and the code in the bridge (and some other places),
was not checking the result code properly, and dereferencing
the channel pointer, which could lead to memory corruption
and crashes.
Calling the masq_park function eliminates this danger
in higher levels.
A series of previous commits have replaced some parking calls
with masq_park, but this patch puts them ALL to rest,
(except one, purposely left alone because a masquerade
is done anyway), and gets rid of the code that tests
the KEEPALIVE result, and the NOHANGUP_PEER result codes.
While bug 13820 inspired this work, this patch does
not solve all the problems mentioned there.
I have tested this patch (again) to make sure I have
not introduced regressions.
Crashes that occurred when a parked party hung up
while the parking party was listening to the numbers
of the parking stall being assigned, is eliminated.
These are the cases where parking code may be activated:
1. Feature one touch (eg. *3)
2. Feature blind xfer to parking lot (eg ##700)
3. Run Park() app from dialplan (eg sip xfer to 700)
(eg. dahdi hookflash xfer to 700)
4. Run Park via manager.
The interesting testing cases for parking are:
I. A calls B, A parks B
a. B hangs up while A is getting the numbers announced.
b. B hangs up after A gets the announcement, but
before the parking time expires
c. B waits, time expires, A is redialed,
A answers, B and A are connected, after
which, B hangs up.
d. C picks up B while still in parking lot.
II. A calls B, B parks A
a. A hangs up while B is getting the numbers announced.
b. A hangs up after B gets the announcement, but
before the parking time expires
c. A waits, time expires, B is redialed,
B answers, A and B are connected, after
which, A hangs up.
d. C picks up A while still in parking lot.
Testing this throroughly involves acting all the permutations
of I and II, in situations 1,2,3, and 4.
Since I added a few more changes (ALL references to KEEPALIVE in the bridge
code eliimated (I missed one earlier), I retested
most of the above cases, and no crashes.
H-extension weirdness.
Current h-extension execution is not completely
correct for several of the cases.
For the case where A calls B, and A parks B, the
'h' exten is run on A's channel as soon as the park
is accomplished. This is expected behavior.
But when A calls B, and B parks A, this will be
current behavior:
After B parks A, B is hung up by the system, and
the 'h' (hangup) exten gets run, but the channel
mentioned will be a derivative of A's...
Thus, if A is DAHDI/1, and B is DAHDI/2,
the h-extension will be run on channel
Parked/DAHDI/1-1<ZOMBIE>, and the
start/answer/end info will be those
relating to Channel A.
And, in the case where A is reconnected to
B after the park time expires, when both parties
hang up after the joyful reunion, no h-exten
will be run at all.
In the case where C picks up A from the
parking lot, when either A or C hang up,
the h-exten will be run for the C channel.
CDR's are a separate issue, and not addressed
here.
As to WHY this strange behavior occurs,
the answer lies in the procedure followed
to accomplish handing over the channel
to the parking manager thread. This procedure
is called masquerading. In the process,
a duplicate copy of the channel is created,
and most of the active data is given to the
new copy. The original channel gets its name
changed to XXX<ZOMBIE> and keeps the PBX
information for the sake of the original
thread (preserving its role as a call
originator, if it had this role to begin
with), while the new channel is without
this info and becomes a call target (a
"peer").
In this case, the parking lot manager
thread is handed the new (masqueraded)
channel. It will not run an h-exten
on the channel if it hangs up while
in the parking lot. The h exten will
be run on the original channel instead,
in the original thread, after the bridge
completes.
See bug 13820 for our intentions as
to how to clean up the h exten behavior.
Review: http://reviewboard.digium.com/r/29/
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This is part of an effort to completely remove AST_PBX_KEEPALIVE and other
similar return codes from the source. While this usage was perfectly safe,
there are others that are problematic. Since we know ahead of time that
we do not want to PBX to destroy the channel, the PBX API has been changed
so that information can be provided as an argument, instead, thus removing
the need for the KEEPALIVE return value.
Further changes to get rid of KEEPALIVE and related code is being done by
murf. There is a patch up for that on review 29.
Review: http://reviewboard.digium.com/r/98/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@165723 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #14028)
Reported by: mostyn
Patches:
timezone-v2.patch uploaded by mostyn (license 398)
(with additional code guideline fixes and a memory leak fix by me - license 14)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@164976 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r164422 | mmichelson | 2008-12-15 13:53:08 -0600 (Mon, 15 Dec 2008) | 3 lines
Add the deadlock note to ast_spawn_extension as well
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r164416 | mmichelson | 2008-12-15 13:45:07 -0600 (Mon, 15 Dec 2008) | 4 lines
Add notes to autoservice and pbx doxygen regarding a potential
deadlock scenario so that it is avoided in the future
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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
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines
The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.
If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.
If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.
Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden
(in trunk).
All the places that previously tested for
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.
I tested this against the 4 common parking
scenarios:
1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.
2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.
4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.
No crash.
I also ran the scenarios above against valgrind, and accesses looked good.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@129307 f38db490-d61c-443f-a65b-d21fe96a405b
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My mistake, pointed out by Russell.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128378 f38db490-d61c-443f-a65b-d21fe96a405b
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AMI commands can display that a channel is under control of an AGI.
Work inspired by work at customer site, but paid for by Edvina AB
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@128240 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r127973 | tilghman | 2008-07-03 22:30:30 -0500 (Thu, 03 Jul 2008) | 8 lines
Fix the 'dialplan remove extension' logic, so that it a) works with cidmatch,
and b) completes contexts correctly when the extension is ambiguous.
(closes issue #12980)
Reported by: licedey
Patches:
20080703__bug12980.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
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Reported by: ys
Many thanks to ys for doing the research on this problem.
I didn't think it would be best to unlock the contexts
and then relock them after the remove_extension2() call,
so I added an extra arg to remove_extension2() and set it
appropriately in each call. There were not that many.
I considered forcing the code to lock the contexts before
the call to remove_extension2(), but that would require
a slightly greater degree of changes, especially since
the find_context_locked is local to pbx.c
I did a simple sanity test to make sure the code doesn't
mess things up in general.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@123165 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #10540)
Reported by: spendergrass
Patches:
20080417__bug10540.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@115076 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #12351)
Reported by: Corydon76
Patches:
20080402__pbx_incomplete__3.diff.txt uploaded by Corydon76 (license 14)
pbx_incomplete_with_timeout.diff uploaded by fabled (license 448)
Tested by: Corydon76, fabled
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114773 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #12490)
Reported by: bcnit
Patches:
12490-queuevars-3.diff uploaded by qwell (license 4)
Tested by: qwell
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@114540 f38db490-d61c-443f-a65b-d21fe96a405b
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thread-safe.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@111909 f38db490-d61c-443f-a65b-d21fe96a405b
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Reported by: mvanbaak
Tested by: murf, mvanbaak
Due to a bug that occurred when merge_contexts_and_delete scanned the "old" or existing contexts, and found a context
that doesn't exist in the new set, yet owned by a different registrar. The context is created in the new set, with the
old registrar, and and all the priorities and extens that have a different registrar are copied into it. But, not the
includes, ignorepats, and switches. I added code to do this immediately after the context is created.
This still leaves a logical hole in the code. If you define a context in two places, (eg. in extensions.conf and also
in extensions.ael), and they both have includes, but different in composition, no new context will be generated, and
therefore the 'old' includes, switches, and ignorepats will not be copied. I'd have added code to simply add any non-duplicates
into the 'new' context that had a different registrar, but there is one big complication: includes, and switches are definitely
order dependent. (ignorepats I'm not sure about). And we'll have to develop some sort of policy about how we
merge order dependent lists, especially if the intersection of the two sets is empty. (in other words, they do not have any
elements in common). Do the new go first, or the old? I've elected to punt this issue until a user complains. Hopefully,
this is pretty rare thing.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@109169 f38db490-d61c-443f-a65b-d21fe96a405b
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Reported by: ssokol
Patches:
20080304__bug6019.diff.txt uploaded by Corydon76 (license 14)
Tested by: putnopvut
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@107231 f38db490-d61c-443f-a65b-d21fe96a405b
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Reported by: rizzo
Tested by: murf
Proposal of the changes to be made, and then an announcement of how they were accomplished:
http://lists.digium.com/pipermail/asterisk-dev/2008-February/032065.html
and:
http://lists.digium.com/pipermail/asterisk-dev/2008-March/032124.html
Here is a recap, file by file, of what I have done:
pbx/pbx_config.c
pbx/pbx_ael.c
All funcs that were passed a ptr to the context list, now will ALSO be passed a hashtab ptr to the same set.
Why? because (for the time being), the dialplan is stored in both, to facilitate a quick, low-cost move to
hash-tables to speed up dialplan processing. If it was deemed necessary to pass the context LIST, well, it
is just as necessary to have the TABLE available. This is because the list/table in question might not be
the global one, but temporary ones we would use to stage the dialplan on, and then swap into the global
position when things are ready.
We now have one external function for apps to use, "ast_context_find_or_create()" instead of the pre-existing
"find" and "create", as all existing usages used both in tandem anyway.
pbx_config, and pbx_ael, will stage the reloaded dialplan into local lists and tables, and
then call merge_contexts_and_delete, which will merge (now) existing contexts and
priorities from other registrars into this local set by copying them. Then, merge_contexts_and_delete will
lock down the contexts, swap the lists and tables, and unlock (real quick), and then
destroy the old dialplan.
chan_sip.c
chan_iax.c
chan_skinny.c
All the channel drivers that would add regcontexts now use the ast_context_find_or_create now.
chan_sip also includes a small fix to get rid of warnings about removing priorities that never got entered.
apps/app_meetme.c
apps/app_dial.c
apps/app_queue.c
All the apps that added a context/exten/priority were also modified to use ast_context_find_or_create instead.
include/asterisk/pbx.h
ast_context_create() is removed. Find_or_create_ is the new method.
ast_context_find_or_create() interface gets the hashtab added.
ast_merge_contexts_and_delete() gets the local hashtab arg added.
ast_wrlock_contexts_version() is added so you can detect if someone else got a writelock between your readlocking and writelocking.
ast_hashtab_compare_contexts was made public for use in pbx_config/pbx_ael
ast_hashtab_hash_contexts was in like fashion make public.
include/asterisk/pval.h
ast_compile_ael2() interface changed to include the local hashtab table ptr.
main/features.c
For the sake of the parking context, we use ast_context_find_or_create().
main/pbx.c
I changed all the "tree" names to "table" instead. That's because the original
implementation was based on binary trees. (had a free library). Then I moved
to hashtabs. Now, the names move forward too.
refcount field added to contexts, so you can keep track of how many modules
wanted this context to exist.
Some log messages that are warnings were inflated from LOG_NOTICE to LOG_WARNING.
Added some calls to ast_verb(3,...) for debug messages
Lots of little mods to ast_context_remove_extension2, which is now excersized in ways
it was not previously; one definite bug fixed.
find_or_create was upgraded to handle both local lists/tables as well as the globals.
context_merge() was added to do the per-context merging of the old/present contexts/extens/prios into the new/proposed local list/tables
ast_merge_contexts_and_delete() was heavily modified.
ast_add_extension2() was also upgraded to handle changes.
the context_destroy() code was re-engineered to handle the new way of doing things,
by exten/prio instead of by context.
res/ael/pval.c
res/ael/ael.tab.c
res/ael/ael.tab.h
res/ael/ael.y
res/ael/ael_lex.c
res/ael/ael.flex
utils/ael_main.c
utils/extconf.c
utils/conf2ael.c
utils/Makefile
Had to change the interface to ast_compile_ael2(), to include the hashtab ptr.
This ended up involving several external apps. The main gotcha was I had to
include lock.h and hashtab.h in several places.
As a side note, I tested this stuff pretty thoroughly, I replicated the problems
originally reported by Luigi, and made triply sure that reloads worked, and everything
worked thru "stop gracefully". I found a and fixed a few bugs as I was merging into
trunk, that did not appear in my tests of bug6002.
How's this for verbose commit messages?
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@106757 f38db490-d61c-443f-a65b-d21fe96a405b
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does an async goto instead of an explicit goto.
(closes issue #11753)
Reported by: johan
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103765 f38db490-d61c-443f-a65b-d21fe96a405b
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Add number of total calls and current calls to SNMP.
Closes issue #10057, patch by jcmoore.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91779 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r89893 | russell | 2007-11-27 18:20:13 -0600 (Tue, 27 Nov 2007) | 4 lines
- update documentation for some of the goto functions to note that they
handle locking the channel as needed
- update ast_explicit_goto() to lock the channel as needed
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89915 f38db490-d61c-443f-a65b-d21fe96a405b
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didn't match 2012; There were a small set of problems to fix: 1. I needed NOT to score patterns unless you are at the end of the data string. 2. Capital N,X,Z and small n,x,z are OK in patterns. I canonicalize the patterns in the trie to caps. 3. When a pattern ends with dot or exclamation, CANMATCH/MATCHMORE should always report this pattern, no matter the length. With this commit, I also supplied the wish of Luigi, where the user can select which pattern matching algorithm to use, the old (legacy) pattern matcher, or the new, trie based matcher. The OLD matcher is the default. A new [general] section variable, extenpatternmatchnew, is added to the extensions.conf, and the example config has it set to false. If true, the new matcher is used. In all other respects, the context/exten structs are the same; the tries and hashtabs are formed, but in the new mode the tries are not used. A new CLI command 'dialplan set extenpatternmatch true/false' is provided to allow switching at run time. I beg users that are forced to return to the old matcher to please report the reason in the bug tracker. Measured the speed benefit of the new matcher against an impossibly large context with 10,000 extensions: the new matcher is 374 times faster.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89547 f38db490-d61c-443f-a65b-d21fe96a405b
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In this commit:
- move the ast_register/unregister_app functions to module.h
to avoid the need to include pbx.h for the simpler apps;
- move the ast_group structure to channel.h to remove the
dependency of app.h on linkedlists.h
Note, this is a long process that I am doing in small steps.
The main difficulty is that now for each subsystem we
have a single header (e.g. channel.h) included by the subsystem
provider (usually one file, e.g. channel.c) and by its clients
(dozens of them, e.g. we have some 70+ apps and 30+ functions).
This requires the clients to include all the extra headers
required by the provider (eg. lock.h, linkedlists.h, definitions
of substructures...) even though many of the clients would be
just happy with opaque struct declarations and function prototypes.
The long term plan is to eventually rectify this structure
so that the compilation can become faster, and also APIs
are more stable.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89522 f38db490-d61c-443f-a65b-d21fe96a405b
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Also fix a common typo I kept seeing (arguement) in various files.
Closes issue #11222, patch by snuffy (with arguement > argument by me).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89202 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r88805 | russell | 2007-11-05 16:07:54 -0600 (Mon, 05 Nov 2007) | 12 lines
After seeing crashes related to channel variables, I went looking around at the
ways that channel variables are handled. In general, they were not handled in
a thread-safe way. The channel _must_ be locked when reading or writing from/to
the channel variable list.
What I have done to improve this situation is to make pbx_builtin_setvar_helper()
and friends lock the channel when doing their thing. Asterisk API calls almost
all lock the channel for you as necessary, but this family of functions did not.
(closes issue #10923, reported by atis)
(closes issue #11159, reported by 850t)
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88934 f38db490-d61c-443f-a65b-d21fe96a405b
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Reported by: mnicholson
Patch by: mnicholson
Closes issue #11140
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88250 f38db490-d61c-443f-a65b-d21fe96a405b
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in the pbx, a 25-30% speed boost. The two updates used, are, first, to merge the ast_exists_extension() and the ast_spawn_extension() where they are called sequentially in a loop in the code, into a slightly upgraded version of ast_spawn_extension(), with a few extra args; and, second, I modified the substitute_variables_helper_full, so it zeroes out the byte after the evaluated string instead of demanding you pre-zero the buffer; I also went thru the code and removed the code that zeroed this buffer before every call to the substitute_variables_helper_full. The first fix provides about a 9% speedup, and the second the rest. These figures come from the 'PIPS' benchmark I describe in blogs, conf. reports, etc.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88166 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@84580 f38db490-d61c-443f-a65b-d21fe96a405b
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redistributed/repartitioned to allow code re-use both inside and outside of Asterisk. This commit introduces the utils/conf2ael program, and an external config-file reader, for both normal config files, and for extensions.conf (context, exten, prio); It provides an API for programs outside of asterisk to use to play with the dialplan and config files.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@79595 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75260 f38db490-d61c-443f-a65b-d21fe96a405b
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call ast_module_user_add and ast_module_user_remove. These are now called in the ast_func_read and ast_func_write functions outside of the module.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75255 f38db490-d61c-443f-a65b-d21fe96a405b
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ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75200 f38db490-d61c-443f-a65b-d21fe96a405b
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