The reporter was uable to use varchar utf8_unicode_ci with cdr_adaptive_odbc, so
this patch adds those along with some other character types to the list of types
cdr_adaptive_odbc will work using the varchar conditions. The problem wasn't really
UTF8 characters as much as it was a failure to respond to the exact type that was
declared/in use on that database.
(closes issue ASTERISK-19334)
Reported By: Igor Nikolaev
Patches:
cdr_adaptive_odbc.patch uploaded by Igor Nikolaev (license 6236)
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When the stack frame is allocated, we save the address to which we should
return, when the Gosub returns. However, if we just want to restore the
priority, then we need to subtract 1 before setting it. Otherwise, when
a Gosub goes to a nonexistent address, it will skip a priority in the
dialplan. This is because when we return from an application, the PBX
increments the priority for us.
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Prior to this patch, Using "transport=" multiple times would cause them to add to one
another like allow/deny. This patch changes that behavior to simply use the transport
option specified last. Also, if no transport option is applied now, the default will
automatically be UDP.
(closes ASTERISK-19352)
Reported by: jamicque
Patches:
asterisk-19352-transport-warning-message-v1.patch uploaded by Michael L. Young (license 5026)
issueA19352_no_transport_is_udp.patch uploaded by Walter Doekes (license 5674)
Review: https://reviewboard.asterisk.org/r/1745/diff/#index_header
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Add the ability to specify what kind of locking an ao2 object has when it
is allocated. The locking could be one of: MUTEX, RWLOCK, or none.
New API:
ao2_t_alloc_options()
ao2_alloc_options()
ao2_t_container_alloc_options()
ao2_container_alloc_options()
ao2_rdlock()
ao2_wrlock()
ao2_tryrdlock()
ao2_trywrlock()
The OBJ_NOLOCK and AO2_ITERATOR_DONTLOCK flags have a slight meaning
change. They no longer mean that the object is protected by an external
mechanism. They mean the lock associated with the object has already been
manually obtained by one of the ao2_lock calls. This change is necessary
for RWLOCK support since they are not reentrant. Also an operation on an
ao2 container may require promoting a read lock to a write lock by
releasing the already held read lock to re-acquire as a write lock.
Replaced API calls:
ao2_t_link_nolock()
ao2_link_nolock()
ao2_t_unlink_nolock()
ao2_unlink_nolock()
with the respective
ao2_t_link_flags()
ao2_link_flags()
ao2_t_unlink_flags()
ao2_unlink_flags()
API calls to be more flexible and to allow an anticipated enhancement to
control linking duplicate objects into a container.
The changes to format.c and format_cap.c are taking advantange of the new
ao2 locking options to simplify the use of the format capabilities
containers.
Review: https://reviewboard.asterisk.org/r/1554/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357272 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The build system has some special magic to ensure that if Asterisk is built
with --enable-dev-mode *and* DONT_OPTIMIZE, that all the source is still compiled
with the optimizer enabled (even though the result will be thrown away), because
the compiler is able to find a great deal of coding errors and bugs as a result
of running its optimizers. Unfortunately at some point this mode got broken,
and the 'throwaway' compile of the code was no longer done with the optimizer
enabled. This patch corrects that problem.
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Occasionally there is a need to put all objects in one container also into
another container.
Some reasons you might need to do this:
1) You need to reconfigure a container. You would do this by creating a
new container with the new configuration and ao2_container_dup the old
container into it. Then replace the old container with the new. Then
destroy the old container.
2) You need the contents of a container to remain stable while operating
on all of the objects. You would do this by creating a cloned container
of the original with ao2_container_clone. The cloned container is a
snapshot of the objects at the time of the cloning. When done, just
destroy the cloned container.
Review: https://reviewboard.asterisk.org/r/1746/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357145 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* Fix opaquification conversion error.
(closes issue ASTERISK-19424)
Reported by: Jeremy Pepper
Patches:
asterisk-19424-initialize_priority_regression.diff (license #5026) patch uploaded by Michael L. Young
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357101 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit adds GoSub alternatives to connected line, redirecting, and CCSS
macro hooks so that macro can finally be deprecated. This also adds
deprecation warnings for those features when used and in documentation.
Review: https://reviewboard.asterisk.org/r/1760/
(closes issue SWP-4256)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
chan_iax2 to pass in the correct types.
chan_iax2 is the only consumer for the various ast_netsock_* functions in trunk
at this point, so this feels like a safe change to make.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357005 65c4cc65-6c06-0410-ace0-fbb531ad65f3
send_diversion=no will prevent Diversion headers from being added to SIP
requests. This doesn't prevent Diversion from being added with dialplan
such as with SIPAddHeader.
(closes issue ASTERISK-16862)
Reported by: rsw686
Review: https://reviewboard.asterisk.org/r/1769/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
In r354890, a memory leak in app_voicemail was fixed by properly disposing of
the allocated heard/deleted pointers. However, there are situations,
particularly when no messages are found in a folder, where these pointers are
not allocated and not NULL. In that case, an invalid free would be attempted,
which could crash app_voicemail. As there are a number of code paths where
this could occur, this patch uses the number of messages detected in the folder
before it attempts to free the pointers. This resolves the crash detected in
the Asterisk Test Suite's check_voicemail_nominal test.
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Currently, when using res_srtp, once the SRTP policy has been added to the
current session the policy is locked into place. Any attempt to replace an
existing policy, which would be needed if the remote endpoint negotiated a new
cryptographic key, is instead rejected in res_srtp. This happens in particular
in transfer scenarios, where the endpoint that Asterisk is communicating with
changes but uses the same RTP session.
This patch modifies res_srtp to allow remote and local policies to be reloaded
in the underlying SRTP library. From the perspective of users of the SRTP API,
the only change is that the adding of remote and local policies are now added
in a single method call, whereas they previously were added separately. This
was changed to account for the differences in handling remote and local
policies in libsrtp.
Review: https://reviewboard.asterisk.org/r/1741/
(closes issue ASTERISK-19253)
Reported by: Thomas Arimont
Tested by: Thomas Arimont
Patches:
srtp_renew_keys_2012_02_22.diff uploaded by Matt Jordan (license 6283)
(with some small modifications for this check-in)
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Custom parking extensions may not be coded such that the first and only
extension priority is the Park application. These custom parking
extensions will not be recognized as parking extensions. When a call is
blind transferred to an extension that is not recognized as a parking
extension, the normal blind transfer code causes the transferred channel
to start executing dialplan. Calls that get parked in this manner do not
know the original channel name that parked the call so the original parker
could never be called back if the parked call is not retrieved before the
timeout time. The parking space is also announced to the call being
parked as a side effect of not knowing the original parking channel.
* Fix handling of BLINDTRANSFER channel variable for call parking.
* Fixed SIP blind transfer using the wrong dialplan context variable to
check for the parking extension.
(closes issue ASTERISK-19322)
Reported by: aragon
Tested by: rmudgett, jparker
Review: https://reviewboard.asterisk.org/r/1730/
JIRA AST-766
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When we send an ACK for a 2xx response to an INVITE, we are supposed
to use the learned route set. However, when we receive a non-2xx final
response to an INVITE, we are supposed to send the ACK to the same place
we initially sent the INVITE.
We had been doing this up until the changes went in that would build a route
set from provisional responses. That introduced a regression where we would
use the learned route set under all circumstances.
With this change, we now will set the destination of our ACK based on the
invitestate. If it is INV_COMPLETED then that means that we have received
a non-2xx final response (INV_TERMINATED indicates a 2xx response was received).
If it is INV_CANCELLED, then that means the call is being canceled, which
means that we should be ACKing a 487 response.
The other change introduced here is setting the invitestate to INV_CONFIRMED
when we send an ACK *after* the reqprep instead of before. This way, we can
tell in reqprep more easily what the invitestate is prior to sending the ACK.
(closes issue ASTERISK-19389)
reported by Karsten Wemheuer
patches:
ASTERISK-19389v2.patch uploaded by Mark Michelson (license #5049)
(with some slight modifications prior to commit)
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If the res_calendar module was followed immediately by one of the
calendar tech modules and "core stop gracefully" was run, Asterisk
would crash.
This patch adds use count tracking for res_calendar so that it is
unloaded after the tech modules when shutting down gracefully. It
is now not possible to unload all the of the calendar modules via
"module unload res_calednar.so", but it is still possible to unload
them all via "module unload -h res_calendar.so".
Review: https://reviewboard.asterisk.org/r/1752/
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https://origsvn.digium.com/svn/asterisk/branches/10
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r356215 | mjordan | 2012-02-22 08:53:53 -0600 (Wed, 22 Feb 2012) | 32 lines
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r356214 | mjordan | 2012-02-22 08:50:20 -0600 (Wed, 22 Feb 2012) | 27 lines
Fix potential buffer overrun and memory leak when executing "sip show peers"
The "sip show peers" command uses a fix sized array to sort the current peers
in the peers ao2_container. The size of the array is based on the current
number of peers in the container. However, once the size of the array is
determined, the number of peers in the container can change, as the peers
container is not locked. This could cause a buffer overrun when populating
the array, if peers were added to the container after the array was created.
Additionally, a memory leak of the allocated array would occur if a user
caused the _show_peers method to return CLI_SHOWUSAGE.
We now create a snapshot of the current peers using an ao2_callback with the
OBJ_MULTIPLE flag. This size of the array is set to the number of peers
that the iterator will iterate over; hence, if peers are added or removed
from the peers container it will not affect the execution of the "sip show
peers" command.
Review: https://reviewboard.asterisk.org/r/1738/
(closes issue ASTERISK-19231)
(closes issue ASTERISK-19361)
Reported by: Thomas Arimont, Jamuel Starkey
Tested by: Thomas Arimont, Jamuel Starkey
Patches: sip_show_peers_2012_02_16.diff uploaded by mjordan (license 6283)
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The accessors names for the "emulate_dtmf_digit" field on the ast_channel
are misleading. Change them to ast_channel_dtmf_digit_to_emulate*.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Jonathan Rose
Tilghman Lesher
Richard Mudgett
Sean Bright
Mark Michelson
Kevin P. Fleming
Kinsey Moore
Terry Wilson
Alexandr Anikin
Matthew Jordan
Paul Belanger