asterisk/asterisk.git
7 years agoAdd Asterisk TestSuite event hooks to support ConfBridge testing
Matthew Jordan [Thu, 22 Dec 2011 20:44:53 +0000 (20:44 +0000)]
Add Asterisk TestSuite event hooks to support ConfBridge testing

This patch adds initial testsuite event hooks so that ConfBridge tests
can be executed in the Asterisk TestSuite.

(issue ASTERISK-19059)
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7 years agoAllow packetization vaules > 127
Terry Wilson [Thu, 22 Dec 2011 20:39:48 +0000 (20:39 +0000)]
Allow packetization vaules > 127

According to the RTP packetization documentation, and the maximum values
listed in AST_FORMAT_LIST, we should support values > that the signed
char array that ast_codec_pref makes available to store the value. All
places in the code treat the framing field as though it were an int
array instaead of a char array anyway, so this just fixes the type of
the array.

(closes issue ASTERISK-18876)
Review: https://reviewboard.asterisk.org/r/1639/
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7 years agoMake codecs/speex ignore *.i files also.
Richard Mudgett [Wed, 21 Dec 2011 20:13:37 +0000 (20:13 +0000)]
Make codecs/speex ignore *.i files also.
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7 years agoMake apps/confbridge ignore *.i files also.
Richard Mudgett [Wed, 21 Dec 2011 20:08:36 +0000 (20:08 +0000)]
Make apps/confbridge ignore *.i files also.
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7 years agoFix chan_iax2 to not report an RDNIS number if it is blank.
Richard Mudgett [Tue, 20 Dec 2011 23:11:29 +0000 (23:11 +0000)]
Fix chan_iax2 to not report an RDNIS number if it is blank.

Some ISDN switches complain or block the call if the RDNIS number is
empty.

* Made chan_iax2 not save a RDNIS number into the ast_channel if the
string is blank.  This is what other channel drivers do.

(closes issue ASTERISK-17152)
Reported by: rmudgett
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7 years agoThis adds support for setting several safe_asterisk parameters using
Matthew Nicholson [Tue, 20 Dec 2011 20:06:17 +0000 (20:06 +0000)]
This adds support for setting several safe_asterisk parameters using
environment variables and also enables a custom run directory for asterisk
(instead of defaulting to /tmp).

Patch by: Byron Clark (byronclark)
(closes ASTERISK-17810)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348698 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoFix crashes on other platforms caused by interference from Darwin weak symbol support.
Richard Mudgett [Mon, 19 Dec 2011 21:43:19 +0000 (21:43 +0000)]
Fix crashes on other platforms caused by interference from Darwin weak symbol support.

Support weak symbols on a platform specific basis.  The Mac OS X (Darwin)
support must be isolated from the other platforms because it has caused
other platforms to crash.  Several other platforms including Linux have
GCC versions that define the weak attribute.  However, this attribute is
only setup for use in the code by Darwin.

(closes issue ASTERISK-18728)
Reported by: Ben Klang

Review: https://reviewboard.asterisk.org/r/1617/
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7 years agoUpdate documentation for MESSAGE_SEND_STATUS variable.
Leif Madsen [Mon, 19 Dec 2011 19:55:18 +0000 (19:55 +0000)]
Update documentation for MESSAGE_SEND_STATUS variable.

(Closes issue ASTERISK-19056)
Reported by: Yuri
Patches:
     348360.diff uploaded by Yuri (license #5242)
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7 years agoAdd a separate buffer for SRTCP packets
Terry Wilson [Mon, 19 Dec 2011 01:36:21 +0000 (01:36 +0000)]
Add a separate buffer for SRTCP packets

The function ast_srtp_protect used a common buffer for both SRTP and SRTCP
packets. Since this function can be called from multiple threads for the same
SRTP session (scheduler for SRTCP and channel for SRTP) it was possible for the
packets to become corrupted as the buffer was used by both threads
simultaneously.

This patch adds a separate buffer for SRTCP packets to avoid the problem.

(closes issue ASTERISK-18889, Reported/patch by Daniel Collins)
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7 years agoCorrect two flaws in sip.conf.sample related to AST-2011-013.
Kevin P. Fleming [Sun, 18 Dec 2011 18:29:47 +0000 (18:29 +0000)]
Correct two flaws in sip.conf.sample related to AST-2011-013.

* The sample file listed *two* values for the 'nat' option as being the default.
  Only 'force_rport' is the default.

* The warning about having differing 'nat' settings confusingly referred to both
  peers and users.
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7 years agoClean-up on isle five for __ast_request_and_dial() and ast_call_forward().
Richard Mudgett [Fri, 16 Dec 2011 23:58:44 +0000 (23:58 +0000)]
Clean-up on isle five for __ast_request_and_dial() and ast_call_forward().

* Add locking when a channel inherits variables and datastores in
__ast_request_and_dial() and ast_call_forward().  Note: The involved
channels are not active so there was minimal potential for problems.

* Remove calls to ast_set_callerid() in __ast_request_and_dial() and
ast_call_forward() because the set information is for the wrong direction.

* Don't use C++ keywords for variable names in ast_call_forward().

* Run the redirecting interception macro if defined when forwarding a call
in ast_call_forward().  Note: Currently will never execute because the
only callers that supply a calling channel supply a hungup or zombie
channel.

* Make feature_request_and_dial() put the transferee into autoservice when
it calls ast_call_forward() in case a redirection interception macro is
run.  Note: Currently will never happen because the caller channel (Party
B) is always hungup at this time.

* Make feature_request_and_dial() ignore the AST_CONTROL_PROCEEDING frame
to silence a log message.
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7 years agoVoicemail with the saycid option will now play a caller's name based on cid if available.
Jonathan Rose [Fri, 16 Dec 2011 22:00:37 +0000 (22:00 +0000)]
Voicemail with the saycid option will now play a caller's name based on cid if available.

In order to check the availability of the caller's name, app_voicemail will check for an
audio file in <astspooldir>/recordings/callerids/
This change sets a precedent for where to put recordings of names. Currently the idea is
that recordings here could also be used for applications like confbridge and meetme to
find recorded names in this folder from callerid (when another recording isn't available)

(closes issue ASTERISK-18565)
Reporter: Russell Brown
Patches:
r uploaded by Russel Brown (license 6182)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348416 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoFix cut and past error in ast_call_forward().
Richard Mudgett [Fri, 16 Dec 2011 21:30:35 +0000 (21:30 +0000)]
Fix cut and past error in ast_call_forward().

(issue ASTERISK-18836)
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7 years agoFix crash during CDR update.
Richard Mudgett [Fri, 16 Dec 2011 21:10:19 +0000 (21:10 +0000)]
Fix crash during CDR update.

The ast_cdr_setcid() and ast_cdr_update() were shown in ASTERISK-18836 to
be called by different threads for the same channel.  The channel driver
thread and the PBX thread running dialplan.

* Add lock protection around CDR API calls that access an ast_channel
pointer.

(closes issue ASTERISK-18836)
Reported by: gpluser

Review: https://reviewboard.asterisk.org/r/1628/
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7 years agoFix ParkAndAnnounce to pass the CallerID to the announcing channel.
Richard Mudgett [Fri, 16 Dec 2011 01:29:20 +0000 (01:29 +0000)]
Fix ParkAndAnnounce to pass the CallerID to the announcing channel.

ParkAndAnnounce tried to pass the CallerID to the announcing channel but
the ID was wiped out by the channel masquerade done when parking the call.

* Save the CallerID before parking the channel to pass it to the
announcing channel.

* Fixed a minor memory leak in ParkAndAnnounce.

* Updated some ParkAndAnnounce log messages.
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7 years agoAdded support for all slin formats to app_originate
Matthew Jordan [Wed, 14 Dec 2011 22:36:30 +0000 (22:36 +0000)]
Added support for all slin formats to app_originate

Previously, app_originate could not originate a call into a non-8kHz conference
bridge as the formats for non-8kHz slin codecs were not applied to the created
channel.  This patch adds all of the formats by default, such that if a created
channel has a codec that supports a higher sampling rate, a translation path
can be built between it and other channels.
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7 years agoFixed Asterisk crash when function QUEUE_MEMBER receives invalid input
Matthew Jordan [Wed, 14 Dec 2011 22:08:55 +0000 (22:08 +0000)]
Fixed Asterisk crash when function QUEUE_MEMBER receives invalid input

The function QUEUE_MEMBER has two required parameters (queuename, option).  It
was only checking for the presence of queuename.  The patch checks for the
existence of the option parameter and provides better error logging when
invalid values are provided for the option parameter as well.
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7 years agoDon't clear LOCALSTATIONID before sending or receiving. The user may set that
Matthew Nicholson [Wed, 14 Dec 2011 22:05:57 +0000 (22:05 +0000)]
Don't clear LOCALSTATIONID before sending or receiving. The user may set that
variable.

ASTERISK-18921
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7 years agoAdd and document PARKEDCALL variable set during timeout
Jonathan Rose [Wed, 14 Dec 2011 21:08:20 +0000 (21:08 +0000)]
Add and document PARKEDCALL variable set during timeout

PARKEDCALL variable tracks which parking lot the call was last parked in.  This can be
used afterwards for flow control when returntoorigin is set to off. I went ahead and
documented both this and the existing variable set during timeout (PARKINGSLOT) in
the sample features.conf since there was no prior mention of variables being set during
timeout.

(closes issue ASTERISK-16239)
Reported By: Clod Patry
Patches:
M17503.diff uploaded by Clod Patry (license 5138)

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7 years agoImprove error message in CONFBRIDGE_INFO
Matthew Jordan [Wed, 14 Dec 2011 20:51:39 +0000 (20:51 +0000)]
Improve error message in CONFBRIDGE_INFO

Provided a more descriptive error message when a value supplied for the parameter
type is not one of the acceptable values.

(closes issue ASTERISK-18717)
Reported by: Paul Belanger
Patches:
  __20111103-better-confbridge_info-error-msg.txt (License #4999)

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348160 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoFix accidental use of tabs instead of spaces from previous features.conf.sample change
Jonathan Rose [Wed, 14 Dec 2011 20:37:11 +0000 (20:37 +0000)]
Fix accidental use of tabs instead of spaces from previous features.conf.sample change
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7 years agoDocument PARKINGSLOT variable in features.conf.sample
Jonathan Rose [Wed, 14 Dec 2011 20:32:40 +0000 (20:32 +0000)]
Document PARKINGSLOT variable in features.conf.sample

(issue ASTERISK-16239)
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7 years agoFix FollowMe CallerID on outgoing calls.
Richard Mudgett [Tue, 13 Dec 2011 23:10:42 +0000 (23:10 +0000)]
Fix FollowMe CallerID on outgoing calls.

The addition of the Connected Line support changed how CallerID is passed
to outgoing calls.  The FollowMe application was not updated to pass
CallerID to the outgoing calls.

* Fix FollowMe CallerID on outgoing calls.

* Restructured findmeexec() to fix several memory leaks and eliminate some
duplicated code.

* Made check the return value of create_followme_number().  Putting a NULL
into the numbers list is bad if create_followme_number() fails.

* Fixed a couple uses of ast_strdupa() inside loops.

* The changes to bridge_builtin_features.c fix a similar CallerID issue
with the bridging API attended and blind transfers.  (Not used at this
time.)

(closes issue ASTERISK-17557)
Reported by: hamlet505a
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/1612/
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7 years agoFix possible misshandling of an incoming SIP response as a peer poke response.
Stefan Schmidt [Tue, 13 Dec 2011 15:22:48 +0000 (15:22 +0000)]
Fix possible misshandling of an incoming SIP response as a peer poke response.
Also make sure peer has even qualify enabled when handle a peer poke response.

(closes issue ASTERISK-18940)
Reported by: Vitaliy
Tested by: Vitaliy and UnixDev

Review: https://reviewboard.asterisk.org/r/1620
Reviewed by: David Vossel
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7 years agoBacked out core changes from r346391
Matthew Jordan [Mon, 12 Dec 2011 19:35:08 +0000 (19:35 +0000)]
Backed out core changes from r346391

During testing, it was discovered that there were a number of side effects
introduced by r346391 and subsequent check-ins related to it (r346429,
r346617, and r346655).  This included the /main/stdtime/ test 'hanging',
as well as the remote console option failing to receive the appropriate output
after a period of time.

I only backed out the changes to main/ and utils/, as this was adequate
to reverse the behavior experienced.

(issue ASTERISK-18974)

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7 years agoUpdate sample configs to put incoming calls into context public.
Richard Mudgett [Mon, 12 Dec 2011 17:34:39 +0000 (17:34 +0000)]
Update sample configs to put incoming calls into context public.

* Add warning about the SIP allowguest option in context public.

(closes issue ASTERISK-14122)
Reported by: Alec Davis
Review: https://reviewboard.asterisk.org/r/719/
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7 years agoAdds MixMonitor and StopMixMonitor AMI commands to the manager
Jonathan Rose [Fri, 9 Dec 2011 21:47:28 +0000 (21:47 +0000)]
Adds MixMonitor and StopMixMonitor AMI commands to the manager

These commands work much like the dialplan applications that would otherwise invoke them.
A nice benefit of these is that they can be invoked on a call remotely and at any time
during a call. They work much like the Monitor and StopMonitor ami commands.

(closes issue ASTERISK-17726)
Reported by: Sergio González Martín
Patches:
mixmonitor_actions.diff uploaded by Sergio González Martín (license 5644)
Review: https://reviewboard.asterisk.org/r/1193/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347903 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoRemove autojump extensions from SayUnixTime, make an option to perform automatic...
Jonathan Rose [Fri, 9 Dec 2011 20:27:03 +0000 (20:27 +0000)]
Remove autojump extensions from SayUnixTime, make an option to perform automatic jumps.

When a caller sends DTMF while the SayUnixTime application is saying the time, The call
would jump to the next extension much like it does during Background(). This patch adds
option 'j' to SayUnixTime which when used employs the old behavior. Also, this patch
allows arguments to sayunixtime to not be used as empty strings in the case of something
like 'sayunixtime(,,,j)' or 'sayunixtime(,,pattern).

(closes issue ASTERISK-16675)
Reported by: jlpedrosa
Patches:
patch_SayUnixTime_noJump.patch uploaded by jlpedrosa (license 5959)
Review: https://reviewboard.asterisk.org/r/956/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347866 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoFix some parsing issues in add_exten_to_pattern_tree().
Richard Mudgett [Fri, 9 Dec 2011 01:33:29 +0000 (01:33 +0000)]
Fix some parsing issues in add_exten_to_pattern_tree().

* Simplify compare_char() and avoid potential sign extension issue.

* Fix infinite loop in add_exten_to_pattern_tree() handling of character
set escape handling.

* Added buffer overflow checks in add_exten_to_pattern_tree() character
set collection.

* Made ignore empty character sets.

* Added escape character handling to end-of-range character in character
sets.  This has a slight change in behavior if the end-of-range character
is an escape character.  You must now escape it.

* Fix potential sign extension issue when expanding character set ranges.

* Made remove duplicated characters from character sets.  The duplicate
characters lower extension matching priority and prevent duplicate
extension detection.

* Fix escape character handling when the escape character is trying to
escape the end-of-string.  We could have continued processing characters
after the end of the exten string.  We could have added the previous
character to the pattern matching tree incorrectly.

(closes issue ASTERISK-18909)
Reported by: Luke-Jr
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Merged revisions 347812 from http://svn.asterisk.org/svn/asterisk/branches/10

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7 years agoFix regression when using tcpenable=no and tlsenable=yes.
Walter Doekes [Thu, 8 Dec 2011 21:32:36 +0000 (21:32 +0000)]
Fix regression when using tcpenable=no and tlsenable=yes.

The tlsenable settings are tucked away in main/tcptls.c, so I missed
them when resolving ASTERISK-18837. This should resolve the test suite
breakage of the sip tls tests.

Review: https://reviewboard.asterisk.org/r/1615
Reviewed by: Matt Jordan
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7 years agoFix regressed behavior of queue set penalty to work without specifying 'in <queuename>'
Jonathan Rose [Thu, 8 Dec 2011 20:55:19 +0000 (20:55 +0000)]
Fix regressed behavior of queue set penalty to work without specifying 'in <queuename>'

r325483 caused a regression in Asterisk 10+ that would make Asterisk segfault when
attempting to set penalty on an interface without specifying a queue in the queue set
penalty CLI command. In addition, no attempt would be made whatsoever to perform the
penalty setting on all the queues in the core list with either the cli command or the
non-segfaulting ami equivalent. This patch fixes that and also makes an attempt to
document and rename some functions required by this command to better represent what
they actually do. Oh yeah, and the use of this command without specifying a specific
queue actually works now.

Review: https://reviewboard.asterisk.org/r/1609/
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7 years agoMark channel running the h exten with the soft-hangup flag.
Richard Mudgett [Thu, 8 Dec 2011 17:55:07 +0000 (17:55 +0000)]
Mark channel running the h exten with the soft-hangup flag.

When a bridge is broken, ast_bridge_call() might execute the h exten on
the calling channel.  However, that channel may not have been the channel
that broke the bridge by hanging up.  The channel executing the h exten
must be in a hung up state so things like AGI run in the correct mode.

* Make sure ast_bridge_call() marks the channel it is executing the h
exten on as hung up.  (The AST_SOFTHANGUP_APPUNLOAD flag is used so as to
match the pbx.c main dialplan execution loop when it executes the h
exten.)

(closes issue ASTERISK-18811)
Reported by: David Hajek
Patches:
      jira_asterisk_18811_v1.8.patch (license #5621) patch uploaded by rmudgett
Tested by: David Hajek, rmudgett
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7 years agoDon't crash on INFO automon request with no channel
Terry Wilson [Thu, 8 Dec 2011 16:24:29 +0000 (16:24 +0000)]
Don't crash on INFO automon request with no channel

AST-2011-014. When automon was enabled in features.conf, it was possible
to crash Asterisk by sending an INFO request if no channel had been
created yet.

(closes issue ASTERISK-18805)
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Merged revisions 347530 from http://svn.asterisk.org/svn/asterisk/branches/1.6.2
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Merged revisions 347531 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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Merged revisions 347532 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347533 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoFix segfault on answer.
Damien Wedhorn [Thu, 8 Dec 2011 06:59:01 +0000 (06:59 +0000)]
Fix segfault on answer.

Fix a segfault if an attempt to answer a call is made between when
the inbound call gives up (and the channel is removed) and when the
device is notified and removes the call from the device.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347490 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoUpdate AMI Getvar and Setvar documentation about supplying a channel name.
Richard Mudgett [Wed, 7 Dec 2011 21:42:29 +0000 (21:42 +0000)]
Update AMI Getvar and Setvar documentation about supplying a channel name.

(closes issue ASTERISK-18958)
Reported by: Red
Patches:
      jira_asterisk_18958_v1.8.patch (license #5621) patch uploaded by rmudgett
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7 years agoFix: Meetme recording variables from realtime DB use null entries over channel variables
Jonathan Rose [Wed, 7 Dec 2011 20:34:23 +0000 (20:34 +0000)]
Fix: Meetme recording variables from realtime DB use null entries over channel variables

Meetme would attempt to substitute the realtime values of RECORDING_FILE and
RECORDING_FORMAT from the meetme db entry instead of using the channel variable set
for those variables in spite of those database entries being NULL or even lacking
a column to represent them.

(closes issue ASTERISK-18873)
Reported by: Byron Clark
Patches:
ASTERISK-18873-1.patch uploaded by Byron Clark (license 6157)
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7 years agoAdd ASTSBINDIR to the list of configurable paths
Terry Wilson [Wed, 7 Dec 2011 20:15:29 +0000 (20:15 +0000)]
Add ASTSBINDIR to the list of configurable paths

This patch also makes astdb2sqlite3 and astcanary use the configured
directory instead of relying on $PATH.

(closes issue ASTERISK-18959)
Review: https://reviewboard.asterisk.org/r/1613/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347345 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoMake SIP INFO messages for dtmf-relay signals case insensitive.
Richard Mudgett [Tue, 6 Dec 2011 23:58:44 +0000 (23:58 +0000)]
Make SIP INFO messages for dtmf-relay signals case insensitive.

(closes issue ASTERISK-18924)
Reported by: Kevin Taylor
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347294 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoDocuments CHANNEL(musicclass) taking priority over m([x]) in waitExten
Jonathan Rose [Tue, 6 Dec 2011 22:01:00 +0000 (22:01 +0000)]
Documents CHANNEL(musicclass) taking priority over m([x]) in waitExten

If waitExten specifies a music class to use with its music on hold option, it will use
CHANNEL(musicclass) instead if that channel variable has been set on the initiating
channel.  This documents that behavior in the waitExten app so that this can be known
without checking the documentation of the code in function local_ast_moh_start.

(closes issue ASTERISK-18804)
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Merged revisions 347240 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347241 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoAdd VM_INFO() dialplan function to gather information about a mailbox.
Walter Doekes [Tue, 6 Dec 2011 20:23:13 +0000 (20:23 +0000)]
Add VM_INFO() dialplan function to gather information about a mailbox.

Deprecates MAILBOX_EXISTS. Provides count, email, exists, fullname,
language, locale, pager, password, tz.

(closes issue ASTERISK-18634)
Patch by: Kris Shaw
Review: https://reviewboard.asterisk.org/r/1568
Reviewed by: Walter Doekes

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347192 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoDon't allow transport=tcp when tcpenable=no.
Walter Doekes [Tue, 6 Dec 2011 19:44:27 +0000 (19:44 +0000)]
Don't allow transport=tcp when tcpenable=no.

When tcpenable=no, sending to transport=tcp hosts was still allowed.
Resolving the source address wasn't possible and yielded the string
"(null)" in SIP messages. Fixed that and a couple of not-so-correct
log messages.

(closes issue ASTERISK-18837)
Reported by: Andreas Topp

Review: https://reviewboard.asterisk.org/r/1585
Reviewed by: Matt Jordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347168 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoAdd regression tests for issue ASTERISK-18838.
Walter Doekes [Tue, 6 Dec 2011 19:30:14 +0000 (19:30 +0000)]
Add regression tests for issue ASTERISK-18838.

Review: https://reviewboard.asterisk.org/r/1572
Reviewed by: Matt Jordan
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347163 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoThe voicemail [general] zonetag and locale variables weren't loaded
Walter Doekes [Tue, 6 Dec 2011 19:28:18 +0000 (19:28 +0000)]
The voicemail [general] zonetag and locale variables weren't loaded
until after the mailboxes were initialized. This caused the settings to
be unset for those mailboxes until a reload was performed.

(closes issue ASTERISK-18838)

Review: https://reviewboard.asterisk.org/r/1570
Reviewed by: Matt Jordan
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Merged revisions 347124 from http://svn.asterisk.org/svn/asterisk/branches/10

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7 years agoDoubly linked lists unit test and update to implementation.
Richard Mudgett [Tue, 6 Dec 2011 19:09:56 +0000 (19:09 +0000)]
Doubly linked lists unit test and update to implementation.

Update the doubly linked list implementation.  Now safe traversing can
insert before and after the current node when traversing in either
direction.

Updated the linked lists unit test test_linkedlist to also test doubly
linked lists.  The old test_dlinkedlist requires a manual check of results
and probably should be removed.

Review: https://reviewboard.asterisk.org/r/1569/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347110 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoFixed crash from orphaned MWI subscriptions in chan_sip
Matthew Jordan [Tue, 6 Dec 2011 17:34:35 +0000 (17:34 +0000)]
Fixed crash from orphaned MWI subscriptions in chan_sip

This patch resolves the issue where MWI subscriptions are orphaned
by subsequent SIP SUBSCRIBE messages.  When a peer is removed, either
by pruning realtime SIP peers or by unloading / loading chan_sip, the
MWI subscriptions that were orphaned would still be on the event engine
list of valid subscriptions but have a pointer to a peer that no longer
was valid.  When an MWI event would occur, this would cause a seg fault.

(closes issue ASTERISK-18663)
Reported by: Ross Beer
Tested by: Ross Beer, Matt Jordan
Patches:
  blf_mwi_diff_12_06_11.txt uploaded by Matt Jordan (license 6283)

Review: https://reviewboard.asterisk.org/r/1610/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347069 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoRestore call progress code for analog ports.
Richard Mudgett [Mon, 5 Dec 2011 17:44:15 +0000 (17:44 +0000)]
Restore call progress code for analog ports.

Extracting sig_analog from chan_dahdi lost call progress detection
functionality.

* Fix analog ports from considering a call answered immediately after
dialing has completed if the callprogress option is enabled.

(closes issue ASTERISK-18841)
Reported by: Richard Miller
Patches:
      chan_dahdi.diff (license #5685) patch uploaded by Richard Miller (Modified by me)
      sig_analog.c.diff (license #5685) patch uploaded by Richard Miller (Modified by me)
      sig_analog.h.diff (license #5685) patch uploaded by Richard Miller
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347008 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoResolve duplicate label used in multiple priorities for the same extension.
Jonathan Rose [Mon, 5 Dec 2011 15:04:12 +0000 (15:04 +0000)]
Resolve duplicate label used in multiple priorities for the same extension.

Prior to this patch, if labels with the same name were used for different priorities in
the same extension, the new label would be accepted, but it would be unusable since
attempts to reach that label would just go to the first one. Now pbx.c detects this,
generates a warning in logs, and culls the label before adding it to the dialplan.

(closes issue ASTERISK-18807)
Reported by: Kenneth Shumard
Patches:
pbx.c.patch uploaded by Kenneth Shumard (License 5077)
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7 years agoFix chan_jingle/gtalk load regression introduced in r346087
Kinsey Moore [Mon, 5 Dec 2011 14:47:11 +0000 (14:47 +0000)]
Fix chan_jingle/gtalk load regression introduced in r346087

Add missing symbol exports for ast_aji_client_destroy and ast_aji_buddy_destroy
for usage outside res_jabber.  Testing of these changes focused on res_jabber
itself, so this problem was missed.

Reported-by: Michael Spiceland
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Merged revisions 346951 from http://svn.asterisk.org/svn/asterisk/branches/1.8
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7 years agoFor SIP REGISTER fix domain-only URIs and domain ACL bypass.
Walter Doekes [Sun, 4 Dec 2011 10:08:19 +0000 (10:08 +0000)]
For SIP REGISTER fix domain-only URIs and domain ACL bypass.

The code that allowed admins to create users with domain-only uri's had
stopped to work in 1.8 because of the reqresp parser rewrites. This is
fixed now: if you have a [mydomain.com] sip user, you can register with
useraddr sip:mydomain.com. Note that in that case -- if you're using
domain ACLs (a configured domain list) -- mydomain.com must be in the
allow list as well.

Reviewboard r1606 shows a list of registration combinations and which
SIP response codes are returned.

Review: https://reviewboard.asterisk.org/r/1533/
Reviewed by: Terry Wilson

(closes issue ASTERISK-18389)
(closes issue ASTERISK-18741)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346901 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoUpdate SIP MESSAGE To parsing to correctly handle URI
Matthew Jordan [Fri, 2 Dec 2011 23:30:21 +0000 (23:30 +0000)]
Update SIP MESSAGE To parsing to correctly handle URI

The previous patch (r346040) incorrectly parsed the URI in the presence
of a port, e.g., user@hostname:port would fail as the port would be
double appended to the SIP message.  This patch uses the parse_uri function
to correctly parse the URI into its username and hostname parts, and places
them in the correct fields in the sip_pvt structure.

(issue ASTERISK-18903)
Review: https://reviewboard.asterisk.org/r/1597/
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7 years agoimplement nat option for rtp channels with ooh323
Alexandr Anikin [Fri, 2 Dec 2011 19:40:21 +0000 (19:40 +0000)]
implement nat option for rtp channels with ooh323

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346816 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoMerged revisions 346763 via svnmerge from
Alexandr Anikin [Fri, 2 Dec 2011 18:03:31 +0000 (18:03 +0000)]
Merged revisions 346763 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

................
  r346763 | may | 2011-12-02 20:42:32 +0400 (Fri, 02 Dec 2011) | 14 lines

  Merged revisions 346762 via svnmerge from
  https://origsvn.digium.com/svn/asterisk/branches/1.8

  ........
    r346762 | may | 2011-12-02 20:19:19 +0400 (Fri, 02 Dec 2011) | 7 lines

    process null frame pointer returned by ast_rtp_instance_read correctly

    (closes issue ASTERISK-16697)
    Reported by: under
    Patches:
            segfault.diff (License #5871) patch uploaded by under
  ........
................

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346777 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoRe-resolve the STUN address if a STUN poll fails for res_stun_monitor.
Richard Mudgett [Thu, 1 Dec 2011 21:19:41 +0000 (21:19 +0000)]
Re-resolve the STUN address if a STUN poll fails for res_stun_monitor.

The STUN socket must remain open between polls or the external address
seen by the STUN server is likely to change.  However, if the STUN request
poll fails then the STUN server address needs to be re-resolved and the
STUN socket needs to be closed and reopened.

* Re-resolve the STUN server address and create a new socket if the STUN
request poll fails.

* Fix ast_stun_request() return value consistency.

* Fix ast_stun_request() to check the received packet for expected message
type and transaction ID.

* Fix ast_stun_request() to read packets until timeout or an associated
response packet is found.  The stun_purge_socket() hack is no longer
required.

* Reduce ast_stun_request() error messages to debug output.

* No longer pass in the destination address to ast_stun_request() if the
socket is already bound or connected to the destination.

(closes issue ASTERISK-18327)
Reported by: Wolfram Joost
Tested by: rmudgett

Review: https://reviewboard.asterisk.org/r/1595/
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7 years agoChange 183 Ringing in sipfrag body to 180 ringing. 183 Ringing isn't even a thing.
Jonathan Rose [Thu, 1 Dec 2011 20:46:12 +0000 (20:46 +0000)]
Change 183 Ringing in sipfrag body to 180 ringing. 183 Ringing isn't even a thing.

183 is actually a session progress message.

(closes issue ASTERISK-18925)
Reported by: Sebastian Denz
Tested by: jrose
Patches:
asterisk18-use_180_instead_of_183_in_sipfrag.diff by Sebastian Denz (License #6139)
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7 years agoRemove the few places where we try to ast_verbose() without a newline.
Tilghman Lesher [Wed, 30 Nov 2011 23:38:34 +0000 (23:38 +0000)]
Remove the few places where we try to ast_verbose() without a newline.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346655 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoFix edge case for overflow buffer.
Tilghman Lesher [Wed, 30 Nov 2011 22:40:23 +0000 (22:40 +0000)]
Fix edge case for overflow buffer.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346617 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agor346525 | jrose | 2011-11-30 15:10:38 -0600 (Wed, 30 Nov 2011) | 18 lines
Jonathan Rose [Wed, 30 Nov 2011 22:03:02 +0000 (22:03 +0000)]
r346525 | jrose | 2011-11-30 15:10:38 -0600 (Wed, 30 Nov 2011) | 18 lines

Cleaning up chan_sip/tcptls file descriptor closing.

This patch attempts to eliminate various possible instances of undefined behavior caused
by invoking close/fclose in situations where fclose may have already been issued on a
tcptls_session_instance and/or closing file descriptors that don't have a valid index
for fd (-1). Thanks for more than a little help from wdoekes.

(closes issue ASTERISK-18700)
Reported by: Erik Wallin

(issue ASTERISK-18345)
Reported by: Stephane Cazelas

(issue ASTERISK-18342)
Reported by: Stephane Chazelas

Review: https://reviewboard.asterisk.org/r/1576/
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7 years agoReverting 346525 due to accidental patch against trunk instead of 1.8
Jonathan Rose [Wed, 30 Nov 2011 21:32:23 +0000 (21:32 +0000)]
Reverting 346525 due to accidental patch against trunk instead of 1.8

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346563 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoCleaning up chan_sip/tcptls file descriptor closing.
Jonathan Rose [Wed, 30 Nov 2011 21:10:38 +0000 (21:10 +0000)]
Cleaning up chan_sip/tcptls file descriptor closing.

This patch attempts to eliminate various possible instances of undefined behavior caused
by invoking close/fclose in situations where fclose may have already been issued on a
tcptls_session_instance and/or closing file descriptors that don't have a valid index
for fd (-1). Thanks for more than a little help from wdoekes.

(closes issue ASTERISK-18700)
Reported by: Erik Wallin

(issue ASTERISK-18345)
Reported by: Stephane Cazelas

(issue ASTERISK-18342)
Reported by: Stephane Chazelas

Review: https://reviewboard.asterisk.org/r/1576/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346525 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoUpdate queues.conf.sample documentation.
Leif Madsen [Wed, 30 Nov 2011 19:37:25 +0000 (19:37 +0000)]
Update queues.conf.sample documentation.

Update the documentation surrounding the use of MONITOR_EXEC to make it more clear
that it can be used for both Monitor() and MixMonitor() usage.

(closes issue ASTERISK-17413)
Reported by: David Woolley
Patches:
     issue18817_mixmonitor_queues_doc.diff by Michael L. Young (License #5026)
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7 years agoFix compilation of utilities (caught by Bamboo).
Tilghman Lesher [Tue, 29 Nov 2011 20:32:53 +0000 (20:32 +0000)]
Fix compilation of utilities (caught by Bamboo).

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346429 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoAllow each logging destination and console to have its own notion of the verbosity...
Tilghman Lesher [Tue, 29 Nov 2011 18:43:16 +0000 (18:43 +0000)]
Allow each logging destination and console to have its own notion of the verbosity level.

Review: https://reviewboard.asterisk.org/r/1599

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346391 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoMerged revisions 346349 via svnmerge from
David Vossel [Tue, 29 Nov 2011 00:03:36 +0000 (00:03 +0000)]
Merged revisions 346349 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/10

........
  r346349 | dvossel | 2011-11-28 18:00:11 -0600 (Mon, 28 Nov 2011) | 10 lines

  Fixes memory leak in message API.

  The ast_msg_get_var function did not properly decrement
  the ref count of the var it retrieves.  The way this is
  implemented is a bit tricky, as we must decrement the var and then
  return the var's value.  As long as the documentation for the
  function is followed, this will not result in a dangling pointer as
  the ast_msg structure owns its own reference to the var while it
  exists in the var container.
........

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7 years agoFix regression that 'rtp/rtcp set debup ip' only works when also a port was specified.
Stefan Schmidt [Mon, 28 Nov 2011 14:34:14 +0000 (14:34 +0000)]
Fix regression that 'rtp/rtcp set debup ip' only works when also a port was specified.

(closes issue ASTERISK-18693)
Reported by: Davide Dal Fra

Review: https://reviewboard.asterisk.org/r/1600/
Reviewed by: Walter Doekes
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7 years agoFix calls to ast_get_ip() not initializing the address family.
Richard Mudgett [Wed, 23 Nov 2011 23:03:32 +0000 (23:03 +0000)]
Fix calls to ast_get_ip() not initializing the address family.
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7 years agoMinor cleanup in chan_sip get_msg_text() function.
Walter Doekes [Wed, 23 Nov 2011 20:48:42 +0000 (20:48 +0000)]
Minor cleanup in chan_sip get_msg_text() function.

In r116240, get_msg_text() got an extra parameter to fix the unwanted
addition of trailing newlines to SIP MESSAGE bodies. This caused all
linefeeds to be trimmed, which isn't right either. This is a stop-gap;
the right fix is to return the original SIP request body.

Review: https://reviewboard.asterisk.org/r/1586
Reviewed by: Matt Jordan
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7 years agoFix ast_str_truncate signedness warning and documentation.
Walter Doekes [Wed, 23 Nov 2011 19:58:19 +0000 (19:58 +0000)]
Fix ast_str_truncate signedness warning and documentation.

Review: https://reviewboard.asterisk.org/r/1594
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7 years agoFix res_jabber resource leaks
Kinsey Moore [Wed, 23 Nov 2011 17:16:33 +0000 (17:16 +0000)]
Fix res_jabber resource leaks

This should fix almost all resource leaks in res_jabber that involve
ASTOBJ_CONTAINER_FIND and resolves an ambiguous situation where
ast_aji_get_client would sometimes bump an object's refcount and sometimes not.

Review: https://reviewboard.asterisk.org/r/1553
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7 years agoFixed SendMessage stripping extension from To: header in SIP MESSAGE
Matthew Jordan [Wed, 23 Nov 2011 16:23:34 +0000 (16:23 +0000)]
Fixed SendMessage stripping extension from To: header in SIP MESSAGE

When using the MessageSend application to send a SIP MESSAGE to a non-peer,
chan_sip attempted to validate the hostname or IP Address.  In the process,
it stripped off the extension and failed to add it back to the sip_pvt
structure before transmitting.  This patch adds the full URI passed in
from the message core to the sip_pvt structure.

(closes issue ASTERISK-18903)
Reported by: Shaun Clark
Tested by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1597/
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7 years agoResume playing existing hold music for cached realtime MOH
Terry Wilson [Wed, 23 Nov 2011 16:12:34 +0000 (16:12 +0000)]
Resume playing existing hold music for cached realtime MOH

As a result of the fix for ASTERISK-18039, realtime caching MOH no longer
properly resumes playing back a file between different holds in the same call.
This is because scanning for new files causes the existing file array to be
emptied and we were just comparing that the saved pointer to the filename
matched the pointer to the filename in a particular position in the array. An
easy fix is to save the filename instead of a pointer to it and then do a
strcmp instead of comparing the addresses.

(closes issue ASTERISK-18912)
Review: https://reviewboard.asterisk.org/r/1596/
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7 years agoAdded support level for new modules
Paul Belanger [Wed, 23 Nov 2011 16:10:45 +0000 (16:10 +0000)]
Added support level for new modules
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7 years agoFix dnsmgr entries to ask for the same address family each time.
Richard Mudgett [Tue, 22 Nov 2011 23:06:11 +0000 (23:06 +0000)]
Fix dnsmgr entries to ask for the same address family each time.

The dnsmgr refresh would always get the first address found regardless of
the original address family requested.  So if you asked for only IPv4
addresses originally, you might get an IPv6 address on refresh.

* Saved the original address family requested by ast_dnsmgr_lookup() to be
used when the address is refreshed.
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7 years agoClarify why the AST_LOG_* macros exist next to the LOG_* macros.
Walter Doekes [Tue, 22 Nov 2011 20:32:51 +0000 (20:32 +0000)]
Clarify why the AST_LOG_* macros exist next to the LOG_* macros.

(issue ASTERISK-17973)
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7 years agoAdd missing sound_only_one config variable
Paul Belanger [Tue, 22 Nov 2011 16:41:58 +0000 (16:41 +0000)]
Add missing sound_only_one config variable

(closes issue ASTERISK-18895)
Reported by: zvision
Patches:
        conf_config_parser.diff (license #5755) patch uploaded by zvision
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7 years agoDefault to nat=yes; warn when nat in general and peer differ
Terry Wilson [Mon, 21 Nov 2011 21:09:59 +0000 (21:09 +0000)]
Default to nat=yes; warn when nat in general and peer differ

It is possible to enumerate SIP usernames when the general and user/peer
nat settings differ in whether to respond to the port a request is sent
from or the port listed for responses in the Via header. In 1.4 and 1.6.2,
this would mean if one setting was nat=yes or nat=route and the other was
either nat=no or nat=never. In 1.8 and 10, this would mean when one was
nat=force_rport and the other was nat=no.

In order to address this problem, it was decided to switch the default
behavior to nat=yes/force_rport as it is the most commonly used option
and to strongly discourage setting nat per-peer/user when at all possible.

For more discussion of the issue, please see:
  http://lists.digium.com/pipermail/asterisk-dev/2011-November/052191.html

(closes issue ASTERISK-18862)
Review: https://reviewboard.asterisk.org/r/1591/
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7 years agoAdd #tryinclude statement
Paul Belanger [Mon, 21 Nov 2011 16:40:17 +0000 (16:40 +0000)]
Add #tryinclude statement

This provides the same functionality as #include however an asterisk module will
still load if the filename does not exist.

Review: https://reviewboard.asterisk.org/r/1476/

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345735 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agoUpdate the documentation to better clarify how the existing commands work.
Tilghman Lesher [Sat, 19 Nov 2011 15:11:45 +0000 (15:11 +0000)]
Update the documentation to better clarify how the existing commands work.

Review: https://reviewboard.asterisk.org/r/1593/
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7 years agoFix a change in behavior in 'database show' from 1.8.
Tilghman Lesher [Fri, 18 Nov 2011 22:20:47 +0000 (22:20 +0000)]
Fix a change in behavior in 'database show' from 1.8.

In 1.8 and previous versions, one could use any fullword portion of the key
name, including the full key, to obtain the record.  Until this patch, this
did not work for the full key.

Closes issue ASTERISK-18886

Patch: by tilghman
Review: by twilson (http://pastebin.com/7rtu6bpk) on #asterisk-dev
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7 years agoAccidentally readded sipfriends.sql in r345560. This was removed
Matthew Jordan [Thu, 17 Nov 2011 19:47:29 +0000 (19:47 +0000)]
Accidentally readded sipfriends.sql in r345560.  This was removed
in r342871

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7 years agoAdd admin toggle mute all and participant count menu options to app_confbridge
Matthew Jordan [Thu, 17 Nov 2011 18:09:13 +0000 (18:09 +0000)]
Add admin toggle mute all and participant count menu options to app_confbridge

This patch adds two new menu features to app_confbridge, admin_toggle_menu_
participants and participant_count.  The admin action will globally mute /
unmute all non-admin participants on a converence, while the participant
count simply exposes the existing participant count function to the
conference bridge menu.

This also adds configuration options to change the sound played when the
conference is globally muted / unmuted, as well as the necessary config
hooks to place these functions in the DTMF menus.

(closes issue ASTERISK-18204)
Reported by: Kevin Reeves
Tested by: Matt Jordan
Patches:
  app_confbridge.c.patch.txt, conf_config_parser.c.patch.txt,
  confbridge.h.patch.txt uploaded by Kevin Reeves (license 6281)

Review: https://reviewboard.asterisk.org/r/1518/

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7 years agoRemove dead code since pri_grab() can never fail.
Richard Mudgett [Thu, 17 Nov 2011 17:31:16 +0000 (17:31 +0000)]
Remove dead code since pri_grab() can never fail.

Dead code makes programmers sick.  I am sick of looking at it.
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7 years agoGuarantee messages go into the right folders with multiple recipients
Jonathan Rose [Wed, 16 Nov 2011 14:56:03 +0000 (14:56 +0000)]
Guarantee messages go into the right folders with multiple recipients

Before, using the U flag in Voicemail with multiple recipients would put urgent messages
in the INBOX folder for all users past the first thanks to a bug with the message
copying function. This would also cause messages to fail to be sent if the INBOX
directory hadn't been created for that mailbox yet.

(closes issue ASTERISK-18245)
Reported by: Matt Jordan

(closes issue ASTERISK-18246)
Reported by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1589/
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7 years agoMake FastAGI HANGUP show up in AGI debug output.
Richard Mudgett [Tue, 15 Nov 2011 20:11:06 +0000 (20:11 +0000)]
Make FastAGI HANGUP show up in AGI debug output.

* Change from using send() to ast_agi_send() so the HANGUP shows up in the
AGI debug output.

(closes issue ASTERISK-18723)
Reported by: James Van Vleet
Patches:
      jira_asterisk_18723_v1.8.patch (license #5621) patch uploaded by rmudgett
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7 years agoFix typo in sig_pri using wrong structure name.
Richard Mudgett [Tue, 15 Nov 2011 18:18:11 +0000 (18:18 +0000)]
Fix typo in sig_pri using wrong structure name.

It is fortunate that the typo does not alter generated code since the
e->restart.channel and e->ring.channel members are in the same position.

(closes issue ASTERISK-18868)
Reported by: zvision
Patches:
      sig_pri.c.diff (License #5755) patch uploaded by zvision
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7 years agoMake queue log indicate if ADDMEMBER is paused for AMI and realtime.
Richard Mudgett [Mon, 14 Nov 2011 22:27:42 +0000 (22:27 +0000)]
Make queue log indicate if ADDMEMBER is paused for AMI and realtime.

* Add parameter to queue log ADDMEMBER to indicate if the member is
paused.

(closes issue ASTERISK-18645)
Reported by: garlew
Patches:
      paused.diff (License #5337) patch uploaded by garlew
Tested by: rmudgett, garlew

Review: https://reviewboard.asterisk.org/r/1469/
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7 years agoRestore SIP DTMF overlap dialing method.
Richard Mudgett [Mon, 14 Nov 2011 22:05:39 +0000 (22:05 +0000)]
Restore SIP DTMF overlap dialing method.

The recent fix for ASTERISK-17288 to get RFC3578 SIP overlap support
working correctly removed a long standing ability to do overlap dialing
using DTMF in the early media phase of a call.

See ASTERISK-18702 it has a very good description of the issue.

I started with Pavel Troller's chan_sip.diff patch on issue
ASTERISK-18702.

* Added 'dtmf' enum value to sip.conf allowoverlap config option.  The new
option value causes the Incomplte application to not send anything with
chan_sip so the caller can supply more digits via DTMF.

* Renames SIP_GET_DEST_PICKUP_EXTEN_FOUND to SIP_GET_DEST_EXTEN_MATCHMORE
since that is what it really means.

* Fixed get_destination() inconsistency with the pickup extension
matching.

* Fixed initialization of PAGE3 of global_flags in reload_config().

(closes issue ASTERISK-18702)
Reported by: Pavel Troller

Review: https://reviewboard.asterisk.org/r/1517/

Review: https://reviewboard.asterisk.org/r/1582/
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7 years agoFix Progress spelling error in main/pbx.c.
Richard Mudgett [Mon, 14 Nov 2011 20:48:19 +0000 (20:48 +0000)]
Fix Progress spelling error in main/pbx.c.

(closes issue ASTERISK-18857)
Reported by: David M
Patches:
      mainpbx-trivial.patch (License #6326) patch uploaded by David M
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7 years agoDon't read past end of input when calling write()
Terry Wilson [Mon, 14 Nov 2011 19:12:49 +0000 (19:12 +0000)]
Don't read past end of input when calling write()

int blah = 1;
...
write(chan->alertpipe[1], &blah, new_frames * sizeof(blah)) !=
(new_frames * sizeof(blah)))

is only valid when new_frames == 1. Otherwise we start reading into adjacent
variables declared on the stack. The read end discards what is read, so the
values don't matter but it's not a good idea to read past where we want even
though new_frames is almost always 1 and should never be large. This patch is
basically taken out of kpfleming's eventfd branch, as he mentioned that he
remembered fixing it there when I talked to him about this issue.

Review: https://reviewboard.asterisk.org/r/1583/
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7 years agoUpdate reqresp_parser parse_uri doxygen comments.
Walter Doekes [Mon, 14 Nov 2011 19:03:29 +0000 (19:03 +0000)]
Update reqresp_parser parse_uri doxygen comments.

The issue mentioned in the bug report had been fixed recently by
twilson. The reporter included this documentation fix.

(closes issue ASTERISK-18572)
Reported by: Richard Miller
Patch by: Richard Miller (modified)
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7 years agoMoves voicemail setup password entry to the end of the setup process.
Jonathan Rose [Mon, 14 Nov 2011 16:21:06 +0000 (16:21 +0000)]
Moves voicemail setup password entry to the end of the setup process.

This change was made because forcegreeting and forcename settings in voicemail could be
circumvented by hanging up after entering a password, because the only way voicemail
currently observes whether a mailbox is new or not is by checking to see if the password
is the same as the mailbox number or not.

(closes issue ASTERISK-18282)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/1581/
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7 years agoEnsure that a null vmexten does not cause a segfault
Kinsey Moore [Mon, 14 Nov 2011 15:11:09 +0000 (15:11 +0000)]
Ensure that a null vmexten does not cause a segfault

When sip_send_mwi_to_peer was modified recently to avoid deadlocks, vmexten
was not expected to be null.  This change handles that situation to avoid
a segfault.
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7 years agoIncreased max number of destinations.
TransNexus OSP Development [Mon, 14 Nov 2011 01:25:25 +0000 (01:25 +0000)]
Increased max number of destinations.

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345023 65c4cc65-6c06-0410-ace0-fbb531ad65f3

7 years agomISDN Round Robin break when no channel is available
Gregory Nietsky [Sat, 12 Nov 2011 16:32:45 +0000 (16:32 +0000)]
mISDN Round Robin break when no channel is available

Prevent channels been parsed repetitively.
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7 years agoDon't forget to rescan MOH files for cached realtime classes
Terry Wilson [Sat, 12 Nov 2011 00:36:37 +0000 (00:36 +0000)]
Don't forget to rescan MOH files for cached realtime classes

Realtime MOH class caching was implemented because without it, you would build
a completely new MOH class and would start the music over at the beginning each
time hold was pressed in a conversation. Unfortunately, this broke re-scanning
for file changes for realtime MOH classes. This patch corrects that issue.

(closes issue ASTERISK-18039)
Review: https://reviewboard.asterisk.org/r/1579/
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7 years agoUse __alignof__ instead of sizeof for stringfield length storage.
Walter Doekes [Fri, 11 Nov 2011 22:00:14 +0000 (22:00 +0000)]
Use __alignof__ instead of sizeof for stringfield length storage.

Kevin P Fleming suggested that r343157 should use __alignof__ instead
of sizeof. For most systems this won't be an issue, but better fix it
now while it's still fresh.

Review: https://reviewboard.asterisk.org/r/1573
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7 years agoVideo format was treated as audio when removed from the file playback scheduler
Matthew Jordan [Fri, 11 Nov 2011 21:57:46 +0000 (21:57 +0000)]
Video format was treated as audio when removed from the file playback scheduler

This patch fixes the format type check in ast_closestream and
filestream_destructor.  Previously a comparison operator was used, but since
audio formats are no longer contiguous (and AST_FORMAT_AUDIO_MASK includes
formats that have a value greater than the video formats), a bitwise AND
operation is used instead.  Duplicated code was also moved to filestream_close.

(closes issue ASTERISK-18682)
Reported by: Aldo Bedrij
Tested by: Matt Jordan

Review: https://reviewboard.asterisk.org/r/1580/
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7 years agoRemove unneeded if(params) checks in reqresp_parser.
Walter Doekes [Fri, 11 Nov 2011 21:37:53 +0000 (21:37 +0000)]
Remove unneeded if(params) checks in reqresp_parser.

Nick Lewis added them in https://reviewboard.asterisk.org/r/549/diff/1-2/
for no apparent reason. There is no way that params could become NULL in
that piece of code, so I removed these excess checks again.
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7 years agoFix bad quoting of multiline mxml opaque_data that caused invalid xml.
Walter Doekes [Fri, 11 Nov 2011 21:33:54 +0000 (21:33 +0000)]
Fix bad quoting of multiline mxml opaque_data that caused invalid xml.

The opaque_data was added and enclosed in single quotes, assuming it
would be only a single line. The rest of the lines were appended after
the closing quote.

(closes issue ASTERISK-18852)
Reported by: peep_ on IRC

Review: https://reviewboard.asterisk.org/r/1577
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7 years agoFix regression introduced by SDP fixups
Kinsey Moore [Fri, 11 Nov 2011 20:15:16 +0000 (20:15 +0000)]
Fix regression introduced by SDP fixups

If capability is adjusted when switching to UDPTL during fax transmission, fax
teardown fails.  Make sure capability is only touched if RTP is active.  This
regression was introduced in R344385.
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7 years agoCheck sip.conf maxforwards parameter for range 1 <= x <= 255.
Richard Mudgett [Fri, 11 Nov 2011 18:37:32 +0000 (18:37 +0000)]
Check sip.conf maxforwards parameter for range 1 <= x <= 255.

JIRA AST-710
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