Reported by: murf
Tested by: murf
For: J. Geis
The 'data' field in the ast_exten struct was being
'moved' from the current dialplan to the replacement
dialplan. This was not good, as the current dialplan
could have problems in the time between the change
and when the new dialplan is swapped in.
So, I modified the merge_and_delete code to strdup
the 'data' field (the args to the app call), and
then it's freed as normal.
I improved a few messages; I added code to limit
the number of calls to the context_merge_incls_swits_igps_other_registrars()
to one per context. I don't think having it called
multiple times per context was doing anything bad,
but it was inefficient.
I hope this fixes the problems Mr. Geiss was noting in
asterisk-users, see
http://lists.digium.com/pipermail/asterisk-users/2008-July/215634.html
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Reported by: mnicholson
Spent most of the day on this bug, and the
solution was so simple. Just had to find and
understand the problem.
The problem was, that the routine to copy
the existing switches, includes, and ignorepats
from the old context to the new one, wasn't
getting called when the context is already
existent. (In other words, if AEL is adding
a new context to the mix, they get copied,
but if pbx_config already defined a context,
then the copy wasn't happening. This made
no sense, so I moved the call to copy the
includes & etc, no matter the case.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r130959 | tilghman | 2008-07-15 12:19:13 -0500 (Tue, 15 Jul 2008) | 8 lines
astman_send_error does not need a newline appended -- the API takes care of
that for us.
(closes issue #13068)
Reported by: gknispel_proformatique
Patches:
asterisk_1_4_astman_send.patch uploaded by gknispel (license 261)
asterisk_trunk_astman_send.patch uploaded by gknispel (license 261)
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Reported by: eliel
OK, now the context registrar slot is strdup'd. It is freed
on destruction. I don't see the need to do this with all
the structs' registrar fields, but if some wild case proves
they should also be handled this way, then we can
put in the extra work at that time.
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Reported by: eliel
Tested by: murf
(closes issue #12960)
Reported by: mnicholson
In this 'omnibus' fix, I **think** I solved both
the problem in 13041, where unloading pbx_ael.so
caused crashes, or incomplete removal of previous
registrar'ed entries. And I added code to completely
remove all includes, switches, and ignorepats that
had a matching registrar entry, which should
appease 12960.
I also added a lot of seemingly useless brackets
around single statement if's, which helped debug
so much that I'm leaving them there.
I added a routine to check the correlation between
the extension tree lists and the hashtab
tables. It can be amazingly helpful when you have
lots of dialplan stuff, and need to narrow
down where a problem is occurring. It's ifdef'd
out by default.
I cleaned up the code around the new CIDmatch code.
It was leaving hanging extens with bad ptrs, getting confused
over which objects to remove, etc. I tightened
up the code and changed the call to remove_exten
in the merge_and_delete code.
I added more conditions to check for empty context
worthy of deletion. It's not empty if there are
any includes, switches, or ignorepats present.
If I've missed anything, please re-open this bug,
and be prepared to supply example dialplan code.
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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
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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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r127663 | murf | 2008-07-02 18:16:25 -0600 (Wed, 02 Jul 2008) | 30 lines
The CDRfix4/5/6 omnibus cdr fixes.
(closes issue #10927)
Reported by: murf
Tested by: murf, deeperror
(closes issue #12907)
Reported by: falves11
Tested by: murf, falves11
(closes issue #11849)
Reported by: greyvoip
As to 11849, I think these changes fix the core problems
brought up in that bug, but perhaps not the more global
problems created by the limitations of CDR's themselves
not being oriented around transfers.
Reopen if necc, but bug reports are not the best
medium for enhancement discussions. We need to start
a second-generation CDR standardization effort to cover
transfers.
(closes issue #11093)
Reported by: rossbeer
Tested by: greyvoip, murf
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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: https://origsvn.digium.com/svn/asterisk/trunk@123165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit merges in the rest of the code needed to support distributed device
state. There are two main parts to this commit.
Core changes:
- The device state handling in the core has been updated to understand device
state across a cluster of Asterisk servers. Every time the state of a device
changes, it looks at all of the device states on each node, and determines the
aggregate device state. That resulting device state is what is provided to
modules in Asterisk that take actions based on the state of a device.
New module, res_ais:
- A module has been written to facilitate the communication of events between
nodes in a cluster of Asterisk servers. This module uses the SAForum AIS
(Service Availability Forum Application Interface Specification) CLM and EVT
services (Cluster Management and Event) to handle this task. This module
currently supports sharing Voicemail MWI (Message Waiting Indication) and
device state events between servers. It has been tested with openais, though
other implementations of the spec do exist.
For more information on testing distributed device state, see the following doc:
- doc/distributed_devstate.txt
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This commit breaks out some logic from pbx.c into a simple API. The hint
processing code had logic for taking the state from multiple devices and
turning that into the state for a single extension. So, I broke this out
and made an API that lets you take multiple device states and determine
the aggregate device state. I needed this for some core device state changes
to support distributed device state.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121501 65c4cc65-6c06-0410-ace0-fbb531ad65f3
DUNDi uses a concept called the Entity ID for unique server identifiers. I have
pulled out the handling of EIDs and made it something available to all of Asterisk.
There is now a global Entity ID that can be used for other purposes as well, such
as code providing distributed device state, which is why I did this. The global
Entity ID is set automatically, just like it was done in DUNDi, but it can also be
set in asterisk.conf. DUNDi will now use this global EID unless one is specified
in dundi.conf.
The current EID for the system can be seen in the "core show settings" CLI command.
It is also available in the dialplan via the ENTITYID variable.
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and off for new installations. This includes the translation from pipes to commas
for pbx_realtime and the EXEC command for AGI, as well as the change to the Set
application not to support multiple variables at once.
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r114579 | file | 2008-04-23 11:54:11 -0300 (Wed, 23 Apr 2008) | 4 lines
Instead of stopping dialplan execution when SayNumber attempts to say a large number that it can not print out a message informing the user and continue on.
(closes issue #12502)
Reported by: bcnit
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Reported by: triccyx
I had a bit a problem reproducing this in my setup (trying not to disturb my other stuff)
but finally, I got it. The problem appears to be that the extension is being added in
replace mode, which kinda assumes that the pattern trie has been formed, when in fact,
in this case, it was not. The checks being done are not nec. when the tree is not yet
formed, as changes like this will be summarized when the trie is formed in the future.
I tested the fix, and the crash no longer happens. Feel free to open the bug again if
this fix doesn't cure the problem.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
a. fix a self-found problem with SPAWN-ing an extension,
where matches were not being found
b. correct some wording in a comment
c. Add some debug for future debugging.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
were handled. In 1.4, if the absolute timeout were reached on a call, no matter what
the return value of ast_spawn_extension was, the pbx would attempt to go to the 'T'
extension or hangup otherwise. The rearrangement of this function in trunk made this check
only happen in the case that ast_spawn_extension returned 0. If ast_spawn_extension returned
1, then the fact that the timeout expired resulted in a no-op, and would cause an infinite
loop to occur in __ast_pbx_run. This change fixes this problem. Now timeouts will
behave as they did in 1.4
(closes issue #11550)
Reported by: pj
Tested by: putnopvut
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113836 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: falves11
Patches:
12298.patch1 uploaded by murf (license 17)
Tested by: murf
I have hopes that the changes made over the last few days will
finalize and solidify this code. While there are bound to be
small tweaks still needed, I feel that the job (at last) is
somewhat completed. Finally, I had a chance to comprehend how
the scoring of extension patterns was done in the previous
version, and I've come very close to using the exact same
criteria in the new pattern matching code. The left-right
sorting is now replicated in the trie structure itself, such
that the first match found will the 'best' match. Compared
the results against 1.4 for several extensions. Replicated
falves11's setup and it works. Used some devious patterns
provided by jsmith, supplemented with a few of my own.
Looks good.
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r111391 | murf | 2008-03-27 07:03:28 -0600 (Thu, 27 Mar 2008) | 9 lines
These small documentation updates made in response to a query in
asterisk-users, where a user was using Playback, but needed the
features of Background, and had no idea that Background existed,
or that it might provide the features he needed. I thought the
best way to avert these kinds of queries was to provide "See Also"
references in all three of "Background", "Playback", "WaitExten".
Perhaps a project to do this with all related apps is in order.
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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.
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r107161 | russell | 2008-03-10 15:17:11 -0500 (Mon, 10 Mar 2008) | 8 lines
Fix another bug specifically related to asynchronous call origination. Once the
PBX is started on the channel using ast_pbx_start(), then the ownership of the
channel has been passed on to another thread. We can no longer access it in this
code. If the channel gets hung up very quickly, it is possible that we could
access a channel that has been free'd.
(inspired by BE-386)
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r107158 | russell | 2008-03-10 15:04:27 -0500 (Mon, 10 Mar 2008) | 9 lines
Fix some bugs related to originating calls. If the code failed to start a PBX
on the channel (such as if you set a call limit based on the system's load
average), then there were cases where a channel that has already been free'd
using ast_hangup() got accessed. This caused weird memory corruption and
crashes to occur.
(fixes issue BE-386)
(much debugging credit goes to twilson, final patch written by me)
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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: https://origsvn.digium.com/svn/asterisk/trunk@106757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Steve Murphy
Tilghman Lesher
Matthew Fredrickson
Brett Bryant
Olle Johansson
Russell Bryant
Joshua Colp
Jeff Peeler
Dwayne M. Hubbard
Jason Parker
Mark Michelson