https://origsvn.digium.com/svn/asterisk/branches/1.8
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r327106 | mnicholson | 2011-07-08 14:52:51 -0500 (Fri, 08 Jul 2011) | 11 lines
Reset our ast_str before passing it on to dialplan function backends.
It is possible for a dialplan backend to not modify the given buffer or ast_str
and still return success. This causes any previous value stored in the buffer
to be used as if the new function call provided it. Some functions also append
to the given buffer assuming it is empty.
The test_substitution unit test has also been modified to detect this problem.
(closes issue ASTERISK-17878)
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r326985 | rmudgett | 2011-07-07 20:08:05 -0500 (Thu, 07 Jul 2011) | 12 lines
Some code cleanup in pbx.c
* Mostly comment and format changes.
* ast_context_remove_extension_callerid() and ast_add_extension_nolock()
will write lock the found specific context.
* ast_context_find() will now tolerate a NULL name.
* Eliminated some inlined versions of find_context() and
find_context_locked().
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r324364 | dvossel | 2011-06-21 15:11:52 -0500 (Tue, 21 Jun 2011) | 10 lines
Fixes locking inversion issue in ast_async_goto()
During this function we can not hold the "chan" lock while
doing the masquerade, the explicit goto on the tmp chan, or
the channel alloc. Instead we need to get the channel lock,
store off information about the channel that we need, and
then let the channel lock go for the remainder of the function.
Review: https://reviewboard.asterisk.org/r/1275/
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r316265 | russell | 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) | 5 lines
Fix a bunch of compiler warnings generated by gcc 4.6.0.
Most of these are -Wunused-but-set-variable, but there were a few others
mixed in here, as well.
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r315645 | rmudgett | 2011-04-26 17:14:31 -0500 (Tue, 26 Apr 2011) | 21 lines
The 'e' special extension fails to trigger in at least two cases.
The 'e' extension is a fall back for the 'i', 't', or 'T' extensions if
any of them do not exist. Many of the places the 'e' extension was
supposed to be invoked fail because the priority was set wrong. There
were two places where the 'e' extension was not even checked for fall
back.
* Made invoke the 'e' extension similarly to the previous 'i', 't', or 'T'
extension check and added the 'e' extension as a fall back to the two
missing locations.
* Prioritized and optimized some hangup tests associated with the 'e'
extension.
(closes issue #19136)
Reported by: kshumard
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1196/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@315649 f38db490-d61c-443f-a65b-d21fe96a405b
This patch is the foundation of an entire new way of looking at media in Asterisk.
The code present in this patch is everything required to complete phase1 of my
Media Architecture proposal. For more information about this project visit the link below.
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal
The primary function of this patch is to convert all the usages of format
bitfields in Asterisk to use the new format and format_cap APIs. Functionally
no change in behavior should be present in this patch. Thanks to twilson
and russell for all the time they spent reviewing these changes.
Review: https://reviewboard.asterisk.org/r/1083/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306010 f38db490-d61c-443f-a65b-d21fe96a405b
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r303549 | russell | 2011-01-24 14:51:37 -0600 (Mon, 24 Jan 2011) | 45 lines
Merged revisions 303548 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r303548 | russell | 2011-01-24 14:49:53 -0600 (Mon, 24 Jan 2011) | 38 lines
Merged revisions 303546 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines
Fix channel redirect out of MeetMe() and other issues with channel softhangup.
Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped
working properly. This issue includes a patch that resolves the issue by
removing a call to ast_check_hangup() from app_meetme.c. I left that in my
patch, as it doesn't need to be there. However, the rest of the patch fixes
this problem with or without the change to app_meetme.
The key difference between what happens before and after this patch is the
effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(),
ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme
sees this which causes it to exit as intended. Checking ast_check_hangup()
caused app_meetme to exit earlier in the process, and the target of the
redirect saw the condition where ast_read() returned NULL.
Removing ast_check_hangup() works around the issue in app_meetme, but doesn't
solve the issue if another application did the same thing. There are also
other edge cases where if an application finishes at the same time that a
redirect happens, the target of the redirect will think that the channel hung
up. So, I made some changes in pbx.c to resolve it at a deeper level. There
are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to
abort the hangup process. My patch extends this to remove the END_OF_Q frame
from the channel's read queue, making the "abort hangup" more complete. This
same technique was used in every place where a softhangup flag was cleared.
(closes issue #18585)
Reported by: oej
Tested by: oej, wedhorn, russell
Review: https://reviewboard.asterisk.org/r/1082/
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r302266 | jpeeler | 2011-01-18 14:19:57 -0600 (Tue, 18 Jan 2011) | 34 lines
Merged revisions 302265 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r302265 | jpeeler | 2011-01-18 14:13:52 -0600 (Tue, 18 Jan 2011) | 27 lines
Convert device state callbacks to ao2 objects to fix a deadlock in chan_sip.
Lock scenario presented here:
Thread 1
holds ast_rdlock_contexts &conlock
holds handle_statechange hints
holds handle_statechange hint
waiting for cb_extensionstate
Locked Here: chan_sip.c line 7428 (find_call)
Thread 2
holds handle_request_do &netlock
holds find_call sip_pvt_ptr
waiting for ast_rdlock_contexts &conlock
Locked Here: pbx.c line 9911 (ast_rdlock_contexts)
Chan_sip has an established locking order of locking the sip_pvt and then
getting the context lock. So the as stated by the summary, the operations in
thread 2 have been modified to no longer require the context lock.
(closes issue #18310)
Reported by: one47
Patches:
statecbs_ao2.mk2.patch uploaded by one47 (license 23),
modified by me
Review: https://reviewboard.asterisk.org/r/1072/
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@302270 f38db490-d61c-443f-a65b-d21fe96a405b
by splitting up devices from hints into an own ao2_container the callback to
get these devices for statechange handling is faster.
with this changes the length of a device used in a hint isnt longer restricted
to 80 characters.
Tests showed that calling handle_statechange is 40 times faster if no hints
are used and 25 times faster if there are any hints.
(closes issue #17928)
Reported by: mdu113
Tested by: schmidts
Review: https://reviewboard.asterisk.org/r/1003/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@296752 f38db490-d61c-443f-a65b-d21fe96a405b
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r295866 | rmudgett | 2010-11-22 13:36:10 -0600 (Mon, 22 Nov 2010) | 60 lines
Merged revisions 295843 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r295843 | rmudgett | 2010-11-22 13:28:23 -0600 (Mon, 22 Nov 2010) | 53 lines
Merged revisions 295790 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010) | 46 lines
The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.
To recreate the problem:
1) Party A calls Party B
2) Invoke CLI "channel redirect" command to redirect channel call leg
associated with A.
3) All associated channels are hung up.
Note that if the CLI command were done on the channel call leg associated
with B it works.
This regression was a result of the fix for issue #16946
(https://reviewboard.asterisk.org/r/740/).
The regression affects all features that use an async goto to execute the
dialplan because of an external event: Channel redirect, AMI redirect, SIP
REFER, and FAX detection.
The struct ast_channel._softhangup code is a mess. The variable is used
for several purposes that do not necessarily result in the call being hung
up. I have added doxygen comments to describe how the various _softhangup
bits are used. I have corrected all the places where the variable was
tested in a non-bit oriented manner.
The primary fix is the new AST_CONTROL_END_OF_Q frame. It acts as a weak
hangup request so the soft hangup requests that do not normally result in
a hangup do not hangup.
JIRA SWP-2470
JIRA SWP-2489
(closes issue #18171)
Reported by: SantaFox
(closes issue #18185)
Reported by: kwemheuer
(closes issue #18211)
Reported by: zahir_koradia
(closes issue #18230)
Reported by: vmarrone
(closes issue #18299)
Reported by: mbrevda
(closes issue #18322)
Reported by: nerbos
Review: https://reviewboard.asterisk.org/r/1013/
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r290255 | tilghman | 2010-10-04 18:23:11 -0500 (Mon, 04 Oct 2010) | 18 lines
Merged revisions 290254 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2
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r290254 | tilghman | 2010-10-04 18:14:59 -0500 (Mon, 04 Oct 2010) | 11 lines
Change new pattern matcher to regard dashes the same as the old pattern matcher -- as visual candy to be ignored.
Also change the AEL parser to not generate dashes within extensions, as those
dashes would be ignored. Update the AEL tests to match this behavior.
(closes issue #17366)
Reported by: murf
Patches:
20100727__issue17366.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
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r281982 | russell | 2010-08-12 11:33:30 -0500 (Thu, 12 Aug 2010) | 5 lines
Remove debugging output from verbose messages.
Pointer values to internal objects is not terribly useful to users in the
verbose messages about adding extensions and contexts.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@281983 f38db490-d61c-443f-a65b-d21fe96a405b
The purpose of this patch is to eliminate struct ast_callerid since it has
turned into a miscellaneous collection of various party information.
Eliminate struct ast_callerid and replace it with the following struct
organization:
struct ast_party_name {
char *str;
int char_set;
int presentation;
unsigned char valid;
};
struct ast_party_number {
char *str;
int plan;
int presentation;
unsigned char valid;
};
struct ast_party_subaddress {
char *str;
int type;
unsigned char odd_even_indicator;
unsigned char valid;
};
struct ast_party_id {
struct ast_party_name name;
struct ast_party_number number;
struct ast_party_subaddress subaddress;
char *tag;
};
struct ast_party_dialed {
struct {
char *str;
int plan;
} number;
struct ast_party_subaddress subaddress;
int transit_network_select;
};
struct ast_party_caller {
struct ast_party_id id;
char *ani;
int ani2;
};
The new organization adds some new information as well.
* The party name and number now have their own presentation value that can
be manipulated independently. ISDN supplies the presentation value for
the name and number at different times with the possibility that they
could be different.
* The party name and number now have a valid flag. Before this change the
name or number string could be empty if the presentation were restricted.
Most channel drivers assume that the name or number is then simply not
available instead of indicating that the name or number was restricted.
* The party name now has a character set value. SIP and Q.SIG have the
ability to indicate what character set a name string is using so it could
be presented properly.
* The dialed party now has a numbering plan value that could be useful to
have available.
The various channel drivers will need to be updated to support the new
core features as needed. They have simply been converted to supply
current functionality at this time.
The following items of note were either corrected or enhanced:
* The CONNECTEDLINE() and REDIRECTING() dialplan functions were
consolidated into func_callerid.c to share party id handling code.
* CALLERPRES() is now deprecated because the name and number have their
own presentation values.
* Fixed app_alarmreceiver.c write_metadata(). The workstring[] could
contain garbage. It also can only contain the caller id number so using
ast_callerid_parse() on it is silly. There was also a typo in the
CALLERNAME if test.
* Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id
number string. ast_callerid_parse() alters the given buffer which in this
case is the channel's caller id number string. Then using
ast_shrink_phone_number() could alter it even more.
* Fixed caller ID name and number memory leak in chan_usbradio.c.
* Fixed uninitialized char arrays cid_num[] and cid_name[] in
sig_analog.c.
* Protected access to a caller channel with lock in chan_sip.c.
* Clarified intent of code in app_meetme.c sla_ring_station() and
dial_trunk(). Also made save all caller ID data instead of just the name
and number strings.
* Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge()
function.
* Corrected some weirdness with app_privacy.c's use of caller
presentation.
Review: https://reviewboard.asterisk.org/r/702/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
Otherwise, it goes to all manager sessions and may exclude the current session,
if the Events mask excludes it.
(closes issue #17504)
Reported by: rrb3942
Patches:
showdialplan_patch.diff uploaded by rrb3942 (license 1003)
Tested by: rrb3942
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273054 f38db490-d61c-443f-a65b-d21fe96a405b
Here is a cut and paste of my review request for this change:
This past weekend, Russell ran our current suite of unit tests for Asterisk
under valgrind. The PBX pattern match test caused valgrind to spew forth two
invalid read errors. This patch contains two changes that shut valgrind up and
do not cause any new memory leaks.
Change 1: In ast_context_remove_extension_callerid2, valgrind reported an
invalid read in the for loop close to the function's end. Specifically, one of
the the strcmp calls in the loop control was reading invalid memory. This was
because the caller of ast_context_remove_extension_callerid2 (__ast_context
destroy in this case) passed as a parameter a shallow copy of an ast_exten's
exten field. This same ast_exten was what was destroyed inside the for loop,
thus any iterations of the for loop beyond the destruction of the ast_exten
would result in invalid reads. My fix for this is to make a copy of the
ast_exten's exten field and pass the copy to
ast_context_remove_extension_callerid2. In addition, I have also acted
similarly with the ast_exten's matchcid field. Since in this case a NULL is
handled quite differently than an empty string, I needed to be a bit more
careful with its handling.
Change 2: In __ast_context_destroy, we iterated over a hashtab and called
ast_context_remove_extension_callerid2 on each item. Specifically, the hashtab
over which we were iterating was an ast_exten's peer_table. Inside of
ast_context_remove_extension_callerid2, we could possibly destroy this
ast_exten, which also caused the hashtab to be freed. Attempting to call
ast_hashtab_end_traversal on the hashtab iterator caused an invalid read to
occur when trying to read the iterator->tab->do_locking field since
iterator->tab had already been freed. My handling of this problem is a bit less
straightforward. With each iteration over the hashtab's contents, we set a
variable called "end_traversal" based on the return of
ast_context_remove_extension_callerid2. If 0 is ever returned, then we know
that the extension was found and destroyed. Because of this, we cannot call
ast_hashtab_end_traversal because we will be guaranteeing a read of invalid
memory. In such a case, we forego calling ast_hashtab_end_traversal and instead
call ast_free on the hashtab iterator.
Review: https://reviewboard.asterisk.org/r/585
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254362 f38db490-d61c-443f-a65b-d21fe96a405b
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
Allows CDR variables added in cdr.c:set_one_cid to become visable during the call,
by executing ast_cdr_update() early in __ast_pbx run.
Reverts sig_pri changes in trunk that are specific to isdn technology only.
(closes issue #16638)
Reported by: alecdavis
Patches:
cdr_update.diff3.txt uploaded by alecdavis (license 585)
Tested by: alecdavis
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@241416 f38db490-d61c-443f-a65b-d21fe96a405b
passdata was only being set in pbx_substitue_variables when arguments were
passed.
(closes issue #16406)
(closes issue #16586)
Reported by: DLNoah
Patches:
bug16586v2.patch uploaded by jpeeler (license 325)
Tested by: DLNoah
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@241366 f38db490-d61c-443f-a65b-d21fe96a405b
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
The second is the default state for matching CID in the dialplan (no matching)
while the first matches one particular CallerID. This is a regression.
(fixes AST-314, SWP-611)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@239571 f38db490-d61c-443f-a65b-d21fe96a405b
During a module reload if multiple extension configs are present,
such as both extensions.conf and extensions.ael, watchers for one
config's hints will be lost during the merging of the other config.
This happens because hint watchers are only preserved for the
current config being merged. The old context list is destroyed
after the merging takes place, meaning any watchers that were not
perserved will be removed.
Now all hints are preserved during merging regardless of what config
file is being merged. These hints are only restored if they
are present within the new context list.
(closes issue #16093)
Reported by: jlaroff
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@237839 f38db490-d61c-443f-a65b-d21fe96a405b
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r237405 | tilghman | 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010) | 16 lines
Add a flag to disable the Background behavior, for AGI users.
This is in a section of code that relates to two other issues, namely
issue #14011 and issue #14940), one of which was the behavior of
Background when called with a context argument that matched the current
context. This fix broke FreePBX, however, in a post-Dial situation.
Needless to say, this is an extremely difficult collision of several
different issues. While the use of an exception flag is ugly, fixing all
of the issues linked is rather difficult (although if someone would like
to propose a better solution, we're happy to entertain that suggestion).
(closes issue #16434)
Reported by: rickead2000
Patches:
20091217__issue16434.diff.txt uploaded by tilghman (license 14)
20091222__issue16434__1.6.1.diff.txt uploaded by tilghman (license 14)
Tested by: rickead2000
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mnicholson