Age | Commit message (Collapse) | Author |
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The logic got reversed, oops. Works properly now when multiple blackfilters are
present.
(closes issue #18283)
Reported by: telecos82
Patches:
ast_managereventfilter.patch uploaded by telecos82 (license 687)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306432 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #18556)
Reported by: kkm
Review: https://reviewboard.asterisk.org/r/1071/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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/
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r305923 | rmudgett | 2011-02-02 18:24:40 -0600 (Wed, 02 Feb 2011) | 24 lines
Merged revisions 305889 via svnmerge from
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r305889 | rmudgett | 2011-02-02 18:15:07 -0600 (Wed, 02 Feb 2011) | 17 lines
Merged revisions 305888 via svnmerge from
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r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011) | 8 lines
Minor AST_FRAME_TEXT related issues.
* Include the null terminator in the buffer length. When the frame is
queued it is copied. If the null terminator is not part of the frame
buffer length, the receiver could see garbage appended onto it.
* Add channel lock protection with ast_sendtext().
* Fixed AMI SendText action ast_sendtext() return value check.
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r302837 | russell | 2011-01-19 17:56:48 -0600 (Wed, 19 Jan 2011) | 2 lines
Only check container count if it exists.
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r302789 | russell | 2011-01-19 17:06:46 -0600 (Wed, 19 Jan 2011) | 11 lines
Merged revisions 302788 via svnmerge from
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r302788 | russell | 2011-01-19 17:06:14 -0600 (Wed, 19 Jan 2011) | 4 lines
Turn a noisy verbose message into a debug message.
This can drown your console if you're using the AMI over HTTP.
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r302785 | russell | 2011-01-19 16:35:15 -0600 (Wed, 19 Jan 2011) | 15 lines
Resolve a memory leak with the manager interface is disabled.
The intent of this check as it stands in previous versions of Asterisk was to
check if there are any active sessions. If there were no sessions, then the
function would return immediately and not bother with queueing up the manager
event to be processed. Since the conversion of storing sessions in an astobj2
container, this check will always pass. I changed it to go back to checking
what was intended.
The side effect of this was that if the AMI is disabled, the manager event
queue is populated anyway, but the code that runs to clear out the queue
never runs. A producer with no consumer is a bad thing.
Reported internally by kmorgan.
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r301595 | mnicholson | 2011-01-12 12:51:37 -0600 (Wed, 12 Jan 2011) | 22 lines
Merged revisions 301594 via svnmerge from
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r301594 | mnicholson | 2011-01-12 12:50:31 -0600 (Wed, 12 Jan 2011) | 15 lines
Removed a usleep(1) that shouldn't be necessary in session_do, and removed the
ms_t member from the mansession_session structure.
Merged revisions 301591 via svnmerge from
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r301591 | mnicholson | 2011-01-12 12:39:03 -0600 (Wed, 12 Jan 2011) | 5 lines
Don't store the thread id for the manager session in the structure we pass to
the thread for the manager session.
ABE-2543
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r293611 | pabelanger | 2010-11-02 16:45:09 -0400 (Tue, 02 Nov 2010) | 2 lines
If manager and tls are disabled, do not display TCP/TLS Bindaddress.
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r292595 | dvossel | 2010-10-21 11:14:33 -0500 (Thu, 21 Oct 2010) | 14 lines
Fixes recursive lock problem in manager.c
It is possible for a AMI session to freeze because of invalid
use of recursive locks during the EVENT processing. This
patch removes the unnecessary locks.
(closes issue #18167)
Reported by: sustav
Patches:
manager_locking_v1.diff uploaded by dvossel (license 671)
Tested by: sustav
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r291227 | dvossel | 2010-10-12 10:58:56 -0500 (Tue, 12 Oct 2010) | 16 lines
Fixes manager.c crash.
This issue was caused by improper use of the mansession lock and
manession_session lock. These two structures are confusing to begin
with so I'm not surprised this occurred. I fixed this by consistently
making sure we use each of these locks only to protect the data
in the corresponding structure. We had mismatched usage of these
locks which resulted in no mutual exclusivity occurring at all.
(closes issue #17994)
Reported by: vrban
Patches:
mansession_locking_fix.diff uploaded by dvossel (license 671)
Tested by: vrban
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r289951 | oej | 2010-10-02 10:56:08 +0200 (Lör, 02 Okt 2010) | 16 lines
Merged revisions 289950 via svnmerge from
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r289950 | oej | 2010-10-02 10:52:03 +0200 (Lör, 02 Okt 2010) | 9 lines
Merged revisions 289949 via svnmerge from
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r289949 | oej | 2010-10-02 10:50:05 +0200 (Lör, 02 Okt 2010) | 2 lines
Add documentation for undocumented option to AMI action originate
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r288572 | twilson | 2010-09-23 13:05:16 -0500 (Thu, 23 Sep 2010) | 2 lines
Make AMI honor enabled=no
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r287471 | oej | 2010-09-19 18:09:28 +0200 (Sön, 19 Sep 2010) | 21 lines
Merged revisions 287470 via svnmerge from
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r287470 | oej | 2010-09-19 18:06:10 +0200 (Sön, 19 Sep 2010) | 14 lines
Merged revisions 287469 via svnmerge from
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r287469 | oej | 2010-09-19 17:56:50 +0200 (Sön, 19 Sep 2010) | 7 lines
Make sure we always free variables properly in manager originate.
(closes issue #17891)
reported, solved and tested by oej
Review: https://reviewboard.asterisk.org/r/869/
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r284065 | russell | 2010-08-28 16:29:45 -0500 (Sat, 28 Aug 2010) | 13 lines
Be more flexible with whitespace on AMI action headers.
Previously, this code required exactly one space to be after the ':' in headers
for an AMI action. This now makes whitespace optional, and allows whitespace that
is there to vary in amount.
(closes issue #17862)
Reported by: cmoye
Patches:
manager.c.patch_trunk uploaded by cmoye (license 858)
manager.c.patch_1.8 uploaded by cmoye (license 858)
Tested by: cmoye
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r279273 | pabelanger | 2010-07-24 13:36:42 -0400 (Sat, 24 Jul 2010) | 6 lines
Default sin_family to AF_INET for TCP / TLS Bindaddress.
Otherwise, 'manager show settings' will generate errors
if manager is not enabled.
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r278023 | tilghman | 2010-07-20 11:37:18 -0500 (Tue, 20 Jul 2010) | 7 lines
Off-by-one error
(closes issue #16506)
Reported by: nik600
Patches:
20100629__issue16506.diff.txt uploaded by tilghman (license 14)
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ACLs can now be configured to match IPv6 networks. This is only
relevant for ACLs in chan_sip for now since other channel drivers
do not support IPv6 addressing. However, once those channel drivers
are outfitted to support IPv6 addressing, the ACLs will already be
ready for IPv6 support.
https://reviewboard.asterisk.org/r/791
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r277261 | tilghman | 2010-07-16 13:04:11 -0500 (Fri, 16 Jul 2010) | 5 lines
If variable gotten is not set, will segfault on Solaris.
(closes issue #17636)
Reported by: bklang
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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: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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This adds a generic API for accommodating IPv6 and IPv4 addresses
within Asterisk. While many files have been updated to make use of the
API, chan_sip and the RTP code are the files which actually support
IPv6 addresses at the time of this commit. The way has been paved for
easier upgrading for other files in the near future, though.
Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne
for their hard work on this.
(closes issue #17565)
Reported by: russell
Patches:
asteriskv6-test-report.pdf uploaded by russell (license 2)
Review: https://reviewboard.asterisk.org/r/743
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packets. This regression was introduced in r48338.
AST-359
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(closes issue #17550)
Reported by: kenner
Patches:
manager.c.diff uploaded by kenner (license 1040)
Tested by: kenner
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(closes issue #17520)
Reported by: kobaz
Patches:
manager.patch uploaded by kobaz (license 834)
Tested by: lmadsen
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This fixes a ref count leak in event filters and checks for
a filter container allocation failure during session creation.
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This patch as documented in the sample config allows one to optionally apply
white, black, or both types of filtering to manager events. The new
'eventfilter' option is set per user.
(closes issue #14861)
Reported by: fnordian
Patches:
eventfilter3.patch uploaded by fnordian (license 110),
modified by me
Review: https://reviewboard.asterisk.org/r/673/
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(issue #17234)
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(closes issue #17234)
Reported by: mav3rick
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Asterisk Generic AOC Representation
- Generic AOC encode/decode routines.
(Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame)
- AST_CONTROL_AOC frame type to represent generic encoded AOC data
- Manager events for AOC-S, AOC-D, and AOC-E messages
Asterisk App Support
- app_dial AOC-S pass-through support on call setup
- app_queue AOC-S pass-through support on call setup
AOC Unit Tests
- AOC Unit Tests for encode/decode routines
- AOC Unit Test for manager event representation.
SIP AOC Support
- Pass-through of generic AOC-D and AOC-E messages to snom phones via the
snom AOC specification.
- Creation of chan_sip page3 flags for the addition of the new
'snom_aoc_enabled' sip.conf option.
IAX AOC Support
- Natively supports AOC pass-through through the use of the new
AST_CONTROL_AOC frame type
DAHDI AOC Support
- ETSI PRI full AOC Pass-through support
- 'aoc_enable' chan_dahdi.conf option for independently enabling
pass-through of AOC-S, AOC-D, AOC-E.
- 'aoce_delayhangup' option for retrieving AOC-E on disconnect.
- DAHDI A() dial string option for requesting AOC services.
example usage:
;requests AOC-S, AOC-D, and AOC-E on call setup
exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e))
Review: https://reviewboard.asterisk.org/r/552/
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This feature generates AMI events in the new aoc event class from the
events passed up by libpri.
Review: https://reviewboard.asterisk.org/r/537/
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don't query for them.
Actions (or failures to act) by external clients should not cause memory leaks
in Asterisk, especially when those continued leaks could cause Asterisk to
misbehave later.
(closes issue #17234)
Reported by: mav3rick
Patches:
20100510__issue17234.diff.txt uploaded by tilghman (license 14)
20100517__issue17234__trunk.diff.txt uploaded by tilghman (license 14)
Tested by: mav3rick, davidw
(closes issue #17365)
Reported by: davidw
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Sean Bright pointed out that we lost a set of newline characters in commit
190349 on a line I had recently changed. Yay for code review on commits.
(issue #17231, #10961)
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r263456 | lmadsen | 2010-05-17 09:35:18 -0500 (Mon, 17 May 2010) | 11 lines
Manager cookies are not compatible with RFC2109.
The Version field in the cookies we're setting contain quotes around the version
number which is not compatible with RFC2109 and breaks some implementations.
(closes issue #17231)
Reported by: ecarruda
Patches:
manager_rfc2109-trunk-v1.patch uploaded by ecarruda (license 559)
manager_rfc2109-1.6.2-v1.patch uploaded by ecarruda (license 559)
Tested by: ecarruda, russell
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See the CHANGES file for more details.
(closes issue #16343)
Reported by: pabelanger
Patches:
issue16343.patch.v5 uploaded by pabelanger (license 224)
Tested by: pabelanger, tilghman, lmadsen
Review: https://reviewboard.asterisk.org/r/630/
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r257070 | mnicholson | 2010-04-13 11:46:30 -0500 (Tue, 13 Apr 2010) | 9 lines
Add an option to restore past broken behavor of the Events manager action
Before r238915, certain values for the EventMask parameter of the Events action would result in no response being returned. This patch adds an option to restore that broken behavior. Also while fixing this bug I discovered that passing an empty EventMasks parameter would also result in no response being returned, this has been fixed as well while being preserved when the broken behavior is requested.
(closes issue #17023)
Reported by: nblasgen
Review: https://reviewboard.asterisk.org/r/602/
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From Reviewboard:
CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date
overview of the architecture can be found in the file doc/CCSS_architecture.pdf
in the CCSS branch. Off the top of my head, the big differences between what is
implemented and what is in the document are as follows:
1. We did not end up modifying the Hangup application at all.
2. The document states that a single call completion monitor may be used across
multiple calls to the same device. This proved to not be such a good idea
when implementing protocol-specific monitors, and so we ended up using one
monitor per-device per-call.
3. There are some configuration options which were conceived after the document
was written. These are documented in the ccss.conf.sample that is on this
review request.
For some basic understanding of terminology used throughout this code, see the
ccss.tex document that is on this review.
This implements CCBS and CCNR in several flavors.
First up is a "generic" implementation, which can work over any channel technology
provided that the channel technology can accurately report device state. Call
completion is requested using the dialplan application CallCompletionRequest and can
be canceled using CallCompletionCancel. Device state subscriptions are used in order
to monitor the state of called parties.
Next, there is a SIP-specific implementation of call completion. This method uses the
methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion
using SIP signaling. There are a few things to note here:
* The agent/monitor terminology used throughout Asterisk sometimes is the reverse of
what is defined in the referenced draft.
* Implementation of the draft required support for SIP PUBLISH. I attempted to write
this in a generic-enough fashion such that if someone were to want to write PUBLISH
support for other event packages, such as dialog-state or presence, most of the effort
would be in writing callbacks specific to the event package.
* A subportion of supporting PUBLISH reception was that we had to implement a PIDF
parser. The PIDF support added is a bit minimal. I first wrote a validation
routine to ensure that the PIDF document is formatted properly. The rest of the
PIDF reading is done in-line in the call-completion-specific PUBLISH-handling
code. In other words, while there is PIDF support here, it is not in any state
where it could easily be applied to other event packages as is.
Finally, there are a variety of ISDN-related call completion protocols supported. These
were written by Richard Mudgett, and as such I can't really say much about their
implementation. There are notes in the CHANGES file that indicate the ISDN protocols
over which call completion is supported.
Review: https://reviewboard.asterisk.org/r/523
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253540 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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When potentially sending manager events, return immediately if there are no
sessions or hooks. Also, avoid locking the hooks list if it is empty.
(issue #16455)
Reported by: atis
Patches:
manager_hooks_trunk.patch uploaded by atis (license 242)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243986 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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: https://origsvn.digium.com/svn/asterisk/trunk@242919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r242850 | oej | 2010-01-25 21:03:38 +0100 (Mån, 25 Jan 2010) | 2 lines
Report error when writing to functions returns error in AMI setvar action
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@240368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r238915 | tilghman | 2010-01-08 18:57:58 -0600 (Fri, 08 Jan 2010) | 6 lines
-1 is interpreted as an error, intead of the maximum mask.
(closes issue #16241)
Reported by: vnovy
Patches:
manager.c.patch uploaded by vnovy (license 922)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238916 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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New parameters ExtraContext, ExtraExtension, and ExtraPriority have been added
to redirect the second channel to a different location. Previously, it was only
possible to redirect both channels to the same place.
(closes issue #15853)
Reported by: haakon
Patches:
trunk-manager.c.patch uploaded by haakon (license 880)
Tested by: jpeeler
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235265 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r232581 | jpeeler | 2009-12-02 15:57:42 -0600 (Wed, 02 Dec 2009) | 7 lines
Send ack (response/message) after receiving manager action userevent
(closes issue #16264)
Reported by: dimas
Patches:
event-ack.patch uploaded by dimas (license 88)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #16275)
Reported by: vnovy
Patches:
manager.c.diff uploaded by vnovy (license 922)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@232576 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(Closes AST-33)
Reviewboard: https://reviewboard.asterisk.org/r/368/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@230111 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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