Commit Graph

22909 Commits

Author SHA1 Message Date
Richard Mudgett 84849cf413 Recorded merge of revisions 372373 from http://svn.asterisk.org/svn/asterisk/branches/11
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Fix compile error.
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Merged revisions 372372 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372374 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05 19:44:32 +00:00
Kinsey Moore edb290ca09 Correct documentation for ModuleLoad AMI action
The documentation incorrectly listed 'rtp' as a reloadable subsystem
and left out many other reloadable subsystems. It is now also
documented that subsystems may only be reloaded, not loaded or
unloaded.

(closes issue AST-977)
Reported-by: John Bigelow
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Merged revisions 372354 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 372358 from http://svn.asterisk.org/svn/asterisk/branches/10
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Merged revisions 372365 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372371 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05 19:26:07 +00:00
Kinsey Moore 8fb1897e89 Ensure counts generated in manager_show_dialplan_helper are correct
When manager_show_dialplan_helper was written, the counter increment
for the total number of contexts was placed with the extensions
increment instead of in the enclosing loop.  This function should
now generate correct context counts.

(closes issue AST-970)
Reported-by: John Bigelow
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Merged revisions 372337 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 372338 from http://svn.asterisk.org/svn/asterisk/branches/10
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05 19:08:15 +00:00
Alec L Davis d39b417074 dsp.c: in ast_mf_detect_init incorrectly sets goertzel samples to 160, should be MF_GSIZE
Remove unused goertzel_state_t member 'samples'.

Related https://reviewboard.asterisk.org/r/2097/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372343 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05 18:56:39 +00:00
Richard Mudgett 6b2183244a Multiple revisions 372327-372328
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  r372327 | rmudgett | 2012-09-05 12:33:11 -0500 (Wed, 05 Sep 2012) | 15 lines
  
  Fix RTP/RTCP read error message confusion.
  
  The RTP/RTCP read error message can report "fail: success" when the
  read failure is because of an ICE failure.
  
  * Changed __rtp_recvfrom() to generate a PJ ICE message when ICE fails.
  
  * Changed RTP/RTCP read error message to indicate an unspecified error
  when errno is zero.
  
  (closes issue ASTERISK-20288)
  Reported by: Joern Krebs
  Patches:
        jira_asterisk_20288_err_msg.patch (license #5621) patch uploaded by rmudgett (modified)
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  r372328 | rmudgett | 2012-09-05 12:35:20 -0500 (Wed, 05 Sep 2012) | 1 line
  
  Fix coding guidelines issue with a recent commit.
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Merged revisions 372327-372328 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05 17:38:22 +00:00
Mark Michelson be500bbafb Re-fix sending unnegotiated payloads during a P2P RTP bridge.
The previous fix still would look in the static_RTP_PT table, which
is inappropriate since we specifically want to find a codec that has
been negotiated.

(closes issue ASTERISK-20296)
reported by NITESH BANSAL
Patches:
	codec_negotiation.patch Uploaded by NITESH BANSAL (License #6418)
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Merged revisions 372311 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05 16:24:19 +00:00
Mark Michelson a7391a37c1 Add fixes and cleanup to app_alarmreceiver.
This work comes courtesy of Pedro Kiefer (License #6407)
The work was posted to review board by Kaloyan Kovachev (License #5506)

(closes issue ASTERISK-16668)
Reported by Grant Crawshay

(closes issue ASTERISK-16694)
Reported by Fred van Lieshout

(closes issue ASTERISK-18417)
Reported by Kostas Liakakis

(closes issue ASTERISK-19435)
Reported by Deon George

(closes issue ASTERISK-20157)
Reported by Pedro Kiefer

(closes issue ASTERISK-20158)
Reported by Pedro Kiefer

(closes issue ASTERISK-20224)
Reported by Pedro Kiefer

Review: https://reviewboard.asterisk.org/r/2075


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372310 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05 15:56:33 +00:00
Matthew Jordan e965020d0c Fix memory leaks in app_voicemail when using IMAP storage or realtime config
This patch fixes two memory leaks:

1. When find_user is called with NULL as its first parameter, the voicemail
   user returned is allocated on the heap.  The inboxcount2 function uses
   find_user in such a fashion when counting new messages, and fails to free
   the resulting voicemail user object.

2. When populate_defaults is called on a voicemail user, it wipes whatever
   flags have been set on the object by copying over the global flags object.
   If the VM_ALLOCED flag was ste on the voicemail user prior to doing so,
   that flag is removed.  This leaks the voicemail user when free_user is later
   called.

(closes issue ASTERISK-19155)
Reported by: Filip Jenicek
patches:
  asterisk.patch2 uploaded by Filip Jenicek (license 6277)

Patch slightly modified for this commit.

Review: https://reviewboard.asterisk.org/r/2096
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Merged revisions 372268 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 372289 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05 14:44:36 +00:00
Darren Sessions 7e46e4d17b LDAP Realtime Peers Cannot Register
Prior to 1.8, it was not necessary for an explicit "type" to be set for an
asterisk LDAP realtime peer. Now the routine find_peer actually checks the
type field during registration and fails to find the peer if it is not set.

The attached patch makes the realtime type equal whatever type is being 
searched for if the type is 0 upon return from routine build_peer. 

(closes issue ASTERISK-17222)
Reported by: John Covert
Patch by: David Vossel
Tested by: Darren Sessions

Review: https://reviewboard.asterisk.org/r/2095/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05 14:12:11 +00:00
Michael L. Young 35ac3b645e Fix breakage caused by last merge. Missing a variable for 11 and trunk.
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Merged revisions 372266 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372267 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05 12:18:47 +00:00
Alec L Davis 7af245a403 dsp.c: Fix multiple issues when no-interdigit delay is present, and fast DTMF 50ms/50ms
Revert DTMF hit/miss detector to original -r349249 method with some changes, remove unnecessary;
  1. reseting of hits=0, when no signal, only need to set it once.
  2. incrementing of hits, when the hit is the same as the current hit.
  3. setting of lasthit, when it's the same as before.

Change HITS_TO_BEGIN to 2, MISSES_TO_END to 3

& 3 spelling mistakes

(closes issue ASTERISK-19610)
alecdavis (license 585)
Reported by: Jean-Philippe Lord
Tested by: alecdavis
 
Review: https://reviewboard.asterisk.org/r/2085/
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Merged revisions 372239 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 372241 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372242 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05 07:43:32 +00:00
Alec L Davis 2f9f9ecffb dsp.c: optimize goerztzel sample loops, in dtmf_detect, mf_detect and tone_detect
use a temporary short int when repeatedly used to call goertzel_sample.

alecdavis (license 585)
Reported by: alecdavis
Tested by: alecdavis
 
Review: https://reviewboard.asterisk.org/r/2093/
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Merged revisions 372212 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372215 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05 06:52:30 +00:00
Michael L. Young aab42a92cb Fix Incrementing Sequence Number For Retransmitted DTMF End Packets
In Asterisk 1.4+, a fix was put in place to increment the sequence number for
retransmitted DTMF end packets.  With the introduction of the RTP engine API in
1.8, the sequence number was no longer being incremented.  This patch fixes this
regression as well as cleans up a few lines that were not doing anything.

(closes issue ASTERISK-20295)
Reported by: Nitesh Bansal
Tested by: Michael L. Young
Patches: 
01_rtp_event_seq_num.patch uploaded by Nitesh Bansal (license 6418)
asterisk-20295-dtmf-fix-cleanup.diff uploaded by Michael L. Young (license 5026)

Review: https://reviewboard.asterisk.org/r/2083/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372200 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05 04:55:07 +00:00
Matthew Jordan d88f91c8d0 Fix memory leak when CEL is successfully written to PostgreSQL database
PQClear is not called when the result object of a call to PQExec has a
status of PGRES_COMMAND_OK.  Interestingly enough, the off nominal case was
handled properly, so this memory leak only occurred when CEL records were
successfully written.

This patch properly clears the result in the nominal code path.

(closes issue ASTERISK-19991)
Reported by: Etienne Lessard
Tested by: Etienne Lessard
patches:
  mem_leak_cel_pgsql.patch uploaded by Etienne Lessard (license #6394)
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Merged revisions 372158 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 372175 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372176 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05 02:26:54 +00:00
Jonathan Rose 4272ec14c5 app_queue: PAUSEALL/UNPAUSEALL logged only if interface is a queue member
Adding UPGRADE.txt entry for r372148

(issue AST-946)
Reported by: John Bigelow


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372149 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-04 19:30:34 +00:00
Jonathan Rose b02c65752c app_queue: Only log PAUSEALL/UNPAUSEALL when 1+ memebers changed.
Prior to this patch, if pause or unpause was issued on an interface
without specifying a specific queue, a PAUSEALL or UNPAUSEALL event
would be logged in the queue log even if that interface wasn't a
member of any queues. This patch changes it so that these events are
only logged when at least one member of any queue exists for that
interface.

(closes issue AST-946)
Reported by: John Bigelow
Review: https://reviewboard.asterisk.org/r/2079/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-04 19:26:02 +00:00
Mark Michelson a40f702aef Fix issue where SIP devices were not notified when custom devices changed to "ringing".
The problem had to do with logic used when checking for what the oldest ringing channel
was. The problem was that if no channel was found, then no notification would be sent.
For custom device states, there is no associated channel, so no notification would get
sent. This fixes the issue by still sending the notification even if no associated
channel can be found for a ringing device state change.

(closes issue ASTERISK-20297)
Reported by Noah Engelberth
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Merged revisions 372137 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-04 15:50:30 +00:00
Mark Michelson 1b6cf69e7b Prevent crash from using app_page with no confbridge.conf file provided.
Also prevents other potential crashes when using aco API
with uninitialized aco_info structs.

(closes issue ASTERISK-20305)
reported by Noah Engelberth
Tested by Noah Engelberth

Review: https://reviewboard.asterisk.org/r/2086
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Merged revisions 372135 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372136 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-04 15:35:02 +00:00
Mark Michelson e7ef469826 Prevent local RTP bridges from sending inappropriate formats to participants.
A change for Asterisk 11 caused a check for failure to incorrectly check the return
value. This resulted in the possibility of transmitting media that a party had not
negotiated. If this media happened to be G.729, then this could potentially result
in one-way audio if no G.729 translators are installed.

(closes issue ASTERISK-20296)
reported by NITESH BANSAL
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Merged revisions 372118 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372119 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-31 21:15:07 +00:00
Mark Michelson 1ab2639cf2 Prevent crash on shutdown due to refcount error on queues container.
When app_queue is unloaded, the queues container has its refcount
decremented, potentially to 0. Then the taskprocessor responsible
for handling device state changes is unreferenced. If the
taskprocessor happens to be just about to run its task, then it
will create and destroy an iterator on the queues container.
This can cause the refcount on the queues container to increase to
1 and then back to 0. Going back to 0 a second time results in
double frees.

This failure was seen periodically in the testsuite when Asterisk
would shut down.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-30 20:54:51 +00:00
Mark Michelson c3b5ec70ac Help prevent ringing queue members from being rung when ringinuse set to no.
Queue member status would not always get updated properly when the member
was called, thus resulting in the member getting multiple calls. With this
change, we update the member's status at the time of calling, and we also
check to make sure the member is still available to take the call before
placing an outbound call.

(closes issue ASTERISK-16115)
reported by nik600
Patches:
	app_queue.c-svn-r370418.patch uploaded by Italo Rossi (license #6409)
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Merged revisions 372048 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-30 18:39:16 +00:00
Matthew Jordan acbe1f90e7 AST-2012-013: Resolve ACL rules being ignored during calls by some IAX2 peers
When an IAX2 call is made using the credentials of a peer defined in a dynamic
Asterisk Realtime Architecture (ARA) backend, the ACL rules for that peer are
not applied to the call attempt. This allows for a remote attacker who is aware
of a peer's credentials to bypass the ACL rules set for that peer.

This patch ensures that the ACLs are applied for all peers, regardless of their
storage mechanism.

(closes issue ASTERISK-20186)
Reported by: Alan Frisch
Tested by: mjordan, Alan Frisch
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Merged revisions 372028 from http://svn.asterisk.org/svn/asterisk/branches/11


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372029 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-30 16:25:34 +00:00
Matthew Jordan d624f2c550 AST-2012-012: Resolve AMI User Unauthorized Shell Access through ExternalIVR
The AMI Originate action can allow a remote user to specify information that can
be used to execute shell commands on the system hosting Asterisk. This can
result in an unwanted escalation of permissions, as the Originate action, which    
requires the "originate" class authorization, can be used to perform actions
that would typically require the "system" class authorization. Previous attempts
to prevent this permission escalation (AST-2011-006, AST-2012-004) have sought
to do so by inspecting the names of applications and functions passed in with
the Originate action and, if those applications/functions matched a predefined
set of values, rejecting the command if the user lacked the "system" class
authorization. As noted by IBM X-Force Research, the "ExternalIVR"
application is not listed in the predefined set of values. The solution for     
this particular vulnerability is to include the "ExternalIVR" application in the
set of defined applications/functions that require "system" class authorization.             
          
Unfortunately, the approach of inspecting fields in the Originate action against
known applications/functions has a significant flaw. The predefined set of
values can be bypassed by creative use of the Originate action or by certain
dialplan configurations, which is beyond the ability of Asterisk to analyze at
run-time. Attempting to work around these scenarios would result in severely         
restricting the applications or functions and prevent their usage for legitimate
means. As such, any additional security vulnerabilities, where an
application/function that would normally require the "system" class
authorization can be executed by users with the "originate" class authorization,
will not be addressed. Instead, the README-SERIOUSLY.bestpractices.txt file has
been updated to reflect that the AMI Originate action can result in commands
requiring the "system" class authorization to be executed. Proper system
configuration can limit the impact of such scenarios.         
          
(closes issue ASTERISK-20132)
Reported by: Zubair Ashraf of IBM X-Force Research
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372001 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-30 16:14:26 +00:00
Matthew Jordan 8018b879a2 Clean up doxygen warnings
This patch fixes numerous doxygen warnings across Asterisk.  It also updates
the makefile to regenerate the doxygen configuration on the local system
before running doxygen to help prevent warnings/errors on the local system.

Much thanks to Andrew for tackling one of the Asterisk janitor projects!

(issue ASTERISK-20259)
Reported by: Andrew Latham
Patches:
  doxygen_partial.diff uploaded by Andrew Latham (license 5985)
  make_progdocs.diff uploaded by Andrew Latham (license 5985)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-30 14:23:28 +00:00
Matthew Jordan 4781be6e93 Restore CODING-GUIDELINES to doc folder
In r294740, the CODING-GUIDELINES was removed from the doc folder in favor
of the content on the Asterisk wiki.  Some folks still look in the doc folder
initially for coding guideline suggestions; as such, this patch adds a
CODING-GUIDELINES file back into the doc folder.  The content of the file
merely points to the correct page on the Asterisk wiki where the coding
guidelines currently live.

(closes issue ASTERISK-20279)
Reported by: Andrew Latham
Patches:
  CODING-GUIDELINES.diff uploaded by Andrew Latham (license 5985)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-30 12:50:03 +00:00
Richard Mudgett f075e7631f Ensure alignment of in[] field in MD5Context struct.
The struct MD5Context character buffer is cast to an int32_t* without
making sure that said buffer is aligned.

Since the buffer follows two uint32_t's, the chance of 'in' being (32
bits) unaligned is nil in practice.  But adding code to ensure that 'in'
stays aligned costs nothing and removes all doubts about the casts being
safe.

(closes issue ASTERISK-20241)
Reported by: Walter Doekes
Patches:
      tmp.diff (license #5674) patch uploaded by Walter Doekes


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371952 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-29 22:48:08 +00:00
Richard Mudgett d7e0b9fd91 Fix compile errors.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-29 22:40:18 +00:00
Jonathan Rose a4d3cb86d8 app_meetme: Adding test events for following activity in MeetMe.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371922 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-29 21:15:24 +00:00
Richard Mudgett 7a6393d8de Fix theoretical compile error with HAVE_EPOLL.
Really shows how much epoll is used since it had not been reported yet.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-29 19:57:24 +00:00
Richard Mudgett 6e2d8c06ea Initialize file descriptors for dummy channels to -1.
Dummy channels usually aren't read from, but functions like SHELL and CURL
use autoservice on the channel.

(closes issue ASTERISK-20283)
Reported by: Gareth Palmer
Patches:
      svn-371580.patch (license #5169) patch uploaded by Gareth Palmer (modified)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371892 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-29 19:48:56 +00:00
Jonathan Rose 6c07c904aa chan_sip: Change manager event to confirm SIPqualifypeer into an ack
Matt Jordan  informed me that it was more appropriate to use an
astman_send_ack here instead of making an event response. I've also
used this opportunity to update UPGRADE.txt to mention this change
in behavior.

(issue AST-969)
Reported by: John Bigelow


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371889 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-29 19:38:52 +00:00
Richard Mudgett adefb772c4 Fix hangup cause passthrough regression.
The v1.8 -r369258 change to fix the F and F(x) action logic introduced a
regression in passing the hangup cause from the called channel to the
caller channel.

(closes issue ASTERISK-20287)
Reported by: Konstantin Suvorov
Patches:
      app_dial_hangupcause.patch (license #6421) patch uploaded by Konstantin Suvorov (modified)
Tested by: rmudgett
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2012-08-29 18:40:04 +00:00
Jonathan Rose 3f69a4e34f chan_sip: Send 408 on retransmit timeout instead of 603
(closes issue ASTERISK-20124)
Reported by: Walter Doekes
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2012-08-29 17:35:32 +00:00
Jonathan Rose 504cfd1070 chan_sip: Send a manager event to confirm SIPqualifypeer completes
Prior to this patch, Issuing SIPqualifypeer either resulted in an
error or if it succeeded, a few \r\ns.  This patch adds a
SIPqualifypeerComplete event issued as a response when the command
is successfully executed.

(closes issue AST-969)
Reported by: John Bigelow



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371823 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-29 16:44:48 +00:00
Mark Michelson 44d854938b Fix misleading documentation in agents.conf.sample regarding ackcall usage.
The documentation made it sound as if the DTMF acknowledgment was needed
at the time the agent logs in, rather than when the agent is called. This
is likely a relic from the days when there were multiple ways of logging
in agents.

(closes issue AST-962)
reported by Steve Pitts
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2012-08-27 21:51:47 +00:00
Mark Michelson c81d960ed6 Fix incorrect documentation of the MailboxStatus manager command.
The "Waiting" field was misdocumented as reporting the number of
messages waiting. In reality, it simply indicated the presence or
absence of waiting messages.
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2012-08-27 21:33:02 +00:00
David M. Lee 0312bb7a7e svn:ignore pjproject bin & output for all platforms.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371754 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-27 18:16:28 +00:00
Mark Michelson 4b50e3f1ee Fix incorrectly documented option in queues.conf
sharedlastcall defaults to "no" not "yes"

(closes issue AST-979)
reported by Steve Pitts
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2012-08-27 17:52:16 +00:00
David M. Lee 05fd2ef0a6 Fixes ast_rwlock_timed[rd|wr]lock for BSD and variants.
The original implementations simply wrap pthread functions, which take
absolute time as an argument. The spinlock version for systems without
those functions treated the argument as a delta. This patch fixes the
spinlock version to be consistent with the pthread version.

(closes issue ASTERISK-20240)
Reported by: Egor Gorlin
Patches:
	lock.c.patch uploaded by Egor Gorlin (license 6416)
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2012-08-27 16:56:56 +00:00
Kinsey Moore e13db61695 Implement workaround for BETTER_BACKTRACES crash
When compiling with BETTER_BACKTRACES enabled, Asterisk will sometimes
crash when "core show locks" is run. This happens regularly in the
testsuite since several tests run "core show locks" to help with
debugging. This seems to be a fault with libraries on certain operating
systems (notably CentOS 6.2/6.3) running on virtual machines and
utilizing gcc 4.4.6.

(closes issue ASTERISK-20090)
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2012-08-27 14:13:44 +00:00
Alec L Davis 1295d551f9 mf_detect: incorrectly used DTMF_GSIZE instead of MF_GSIZE
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2012-08-26 23:10:30 +00:00
Mark Michelson 96de9ecf30 I forgot to add the unit tests for scoped locks earlier today.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371633 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-23 04:12:32 +00:00
Joshua Colp 09b121bb50 Add support for call-id logging to chan_motif.
Review: https://reviewboard.asterisk.org/r/2077/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371620 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-22 15:55:26 +00:00
Mark Michelson 6a539ace84 Fix misuses of asprintf throughout the code.
This fixes three main issues

* Change asprintf() uses to ast_asprintf() so that it
pairs properly with ast_free() and no longer causes
MALLOC_DEBUG to freak out.

* When ast_asprintf() fails, set the pointer NULL if
it will be referenced later.

* Fix some memory leaks that were spotted while taking
care of the first two points.

(Closes issue ASTERISK-20135)
reported by Richard Mudgett

Review: https://reviewboard.asterisk.org/r/2071
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2012-08-21 21:01:11 +00:00
Mark Michelson 89a5ff859d Add scoped locks to Asterisk.
With the SCOPED_LOCK macro, you can create a variable
that locks a specific lock and unlocks the lock when the
variable goes out of scope. This is useful for situations
where many breaks, continues, returns, or other interruptions
would require separate unlock statements. With a scoped lock,
these aren't necessary.

There are specializations for mutexes, read locks, write locks,
ao2 locks, ao2 read locks, ao2 write locks, and channel locks.
Each of these is a SCOPED_LOCK at heart though.

Review: https://reviewboard.asterisk.org/r/2060



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-21 19:04:32 +00:00
Mark Michelson db69da3667 Use thread-local storage to store pj_thread_descs.
pj_thread_register() takes a parameter of type pj_thread_desc.
It was assumed that pj_thread_register either used this item
temporarily or made a copy of it. Unfortunately, all it does is
keep a pointer to the structure in thread-local storage. This
means that if our pj_thread_desc goes out of scope, then pjlib
will be referencing bogus data quite often, most commonly on
operations involving a pj_mutex_t.

In our case, our pj_thread_desc was on the stack and went out
of scope very shortly after registering our thread with pjlib.
With this change, the pj_thread_desc is stored in thread-local
storage so the pointer that pjlib keeps in thread-local storage
will reference legitimate memory.

(closes issue ASTERISK-20237)
reported by Jeremy Pepper
Patches:
	ASTERISK-20237.patch uploaded by Mark Michelson (license #5049)
Tested by Jeremy Pepper
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2012-08-20 20:19:52 +00:00
Kinsey Moore a2068c3db6 Ignore recovered zero-length secondary UDPTL packets
In some cases, recovering lost packets using the secondary packet
recovery mechanism with UDPTL/T.38 can result in the recovery of
zero-length packets. These must be ignored or the frame generated from
them can cause segfaults and allocation failures.

(closes issue ASTERISK-19762)
(closes issue ASTERISK-19373)
Reported-by: Benjamin (bulkorok)
Reported-by: Rob Gagnon (rgagnon)
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2012-08-20 15:39:15 +00:00
Kinsey Moore 6c8f4f5fff Fix for commit r371535
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-20 15:01:08 +00:00
Kinsey Moore ca314fe1e2 Apply work-around for BETTER_BACKTRACES crash
When compiling with BETTER_BACKTRACES enabled, Asterisk will sometimes
crash when "core show locks" is run. This happens regularly in the
testsuite since several tests run "core show locks" to help with
debugging. This seems to be a fault with libraries on certain operating
systems (notably CentOS 6.2/6.3) running on virtual machines and
utilizing gcc 4.4.6.

(issue ASTERISK-20090)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-20 14:45:07 +00:00
Matthew Jordan f737698654 Remove old debug code from http configuration loading
(closes issue ASTERISK-20254)
Reported by: Andrew Latham
Patches:
  http.diff uploaded by Andrew Latham (license #5985)
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2012-08-18 02:09:30 +00:00