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20 authors
  1. Dec 19, 2013
    • Joshua Colp's avatar
      res_pjsip_session: Fix SDP negotiation when resending an INVITE with authentication. · 8402cd4c
      Joshua Colp authored 
      The process for resending an INVITE with authentication involves restarting the UAC
session. We were incorrectly passing in that a new offer is being sent, causing the
SDP negotiation to get into a (technically speaking) funky state.
........

Merged revisions 404369 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      8402cd4c
    • Mark Michelson's avatar
      Fix a deadlock that occurred due to a conflict of masquerades. · 1b91ee6c
      Mark Michelson authored 
      For the explanation, here is a copy-paste of the review board explanation:

Initially, it was discovered that performing an attended transfer of a
multiparty bridge with a PJSIP channel would cause a deadlock. A PBX thread
started a masquerade and reached the point where it was calling the fixup()
callback on the "original" channel. For chan_pjsip, this involves pushing a
synchronous task to the session's serializer. The problem was that a task ahead
of the fixup task was also attempting to perform a channel masquerade. However,
since masquerades are designed in a way to only allow for one to occur at a
time, the task ahead of the fixup could not continue until the masquerade
already in progress had completed. And of course, the masquerade in progress
could not complete until the task ahead of the fixup task had completed.
Deadlock.

The initial fix was to change the fixup task to be asynchronous. While this
prevented the deadlock from occurring, it had the frightful side effect of
potentially allowing for tasks in the session's serializer to operate on a
zombie channel.

Taking a step back from this particular deadlock, it became clear that the
problem was not really this one particular issue but that masquerades
themselves needed to be addressed. A PJSIP attended transfer operation calls
ast_channel_move(), which attempts to both set up and execute a masquerade. The
problem was that after it had set up the masquerade, the PBX thread had swooped
in and tried to actually perform the masquerade. Looking at changes that had
been made to Asterisk 12, it became clear that there never is any time now that
anyone ever wants to set up a masquerade and allow for the channel thread to
actually perform the masquerade. Everyone always is calling ast_channel_move(),
performs the masquerade itself before returning.

In this patch, I have removed all blocks of code from channel.c that will
attempt to perform a masquerade if ast_channel_masq() returns true. Now, there
is no distinction between setting up a masquerade and performing the
masquerade. It is one operation. The only remaining checks for
ast_channel_masq() and ast_channel_masqr() are in ast_hangup() since we do not
want to interrupt a masquerade by hanging up the channel. Instead, now
ast_hangup() will wait for a masquerade to complete before moving forward with
its operation.

The ast_channel_move() function has been modified to basically in-line the
logic that used to be in ast_channel_masquerade(). ast_channel_masquerade() has
been killed off for real. ast_channel_move() now has a lock associated with it
that is used to prevent any simultaneous moves from occurring at once. This
means there is no need to make sure that ast_channel_masq() or
ast_channel_masqr() are already set on a channel when ast_channel_move() is
called. It also means the channel container lock is not pulling double duty by
both keeping the container locked and preventing multiple masquerades from
occurring simultaneously.

The ast_do_masquerade() function has been renamed to do_channel_masquerade()
and is now internal to channel.c. The function now takes explicit arguments of
which channels are involved in the masquerade instead of a single channel.
While it probably is possible to do some further refactoring of this method, I
feel that I would be treading dangerously. Instead, all I did was change some
comments that no longer are true after this changeset.

The other more minor change introduced in this patch is to res_pjsip.c to make
ast_sip_push_task_synchronous() run the task in-place if we are already a SIP
servant thread. This is related to this patch because even when we isolate the
channel masquerade to only running in the SIP servant thread, we would still
deadlock when the fixup() callback is reached since we would essentially be
waiting forever for ourselves to finish before actually running the fixup. This
makes it so the fixup is run without having to push a task into a serializer at
all.

(closes issue ASTERISK-22936)
Reported by Jonathan Rose

Review: https://reviewboard.asterisk.org/r/3069
........

Merged revisions 404356 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      1b91ee6c
    • Richard Mudgett's avatar
      udptl: Dead code elimination. ast_udptl_bridge was not used. · 3ccd5dee
      Richard Mudgett authored 
      Removing dead code starting with ast_udptl_bridge() eliminated the code in
this change.

Note: This code has actually been dead since Asterisk v1.4 when it was
first put in.

Review: https://reviewboard.asterisk.org/r/3079/
........

Merged revisions 404354 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      3ccd5dee
    • Scott Griepentrog's avatar
      res_fax.c: crash on framehook with no dsp in fax detect · c061f634
      Scott Griepentrog authored 
      In fax_detect_framehook() a null pointer reference can occur where a
voice frame is processed but no dsp is attached to the fax detection
structure.  The code block that rejects frames that detection cannot
be processed on is checking for dsp but falls through when it should
instead return, as this change implements.

(closes issue ASTERISK-22942)
Reported by: adomjan
Review: https://reviewboard.asterisk.org/r/3076/
........

Merged revisions 404351 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 404352 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404353 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      c061f634
    • Richard Mudgett's avatar
      Voicemail: Remove mailbox identifier format (box@context) assumptions in the system. · e4803bbd
      Richard Mudgett authored 
      This change is in preparation for external MWI support.

Removed code from the system for normal mailbox handling that appends
@default to the mailbox identifier if it does not have a context.  The
only exception is the legacy hasvoicemail users.conf option.  The legacy
option will only work for app_voicemail mailboxes.  The system cannot make
any assumptions about the format of the mailbox identifer used by
app_voicemail.

chan_sip and chan_dahdi/sig_pri had the most changes because they both
tried to interpret the mailbox identifier.  chan_sip just stored and
compared the two components.  chan_dahdi actually used the box
information.

The ISDN MWI support configuration options had to be reworked because
chan_dahdi was parsing the box@context format to get the box number.  As a
result the mwi_vm_boxes chan_dahdi.conf option was added and is documented
in the chan_dahdi.conf.sample file.

Review: https://reviewboard.asterisk.org/r/3072/
........

Merged revisions 404348 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404350 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      e4803bbd
    • Scott Griepentrog's avatar
      astdb: crash in sqlite3 during shutdown · 2882c5f9
      Scott Griepentrog authored 
      When Asterisk is shut down, the astdb_atexit() function releases
(finalize) the previously initiated (prepared) SQL statements in
sqlite3.  Another thread making a subsequent request can cause a
crash in sqlite3.  This patch eliminates that issue by resetting
the statement pointer after it is released/cleared.  The sqlite3
code detects the null pointer, and aborts the operation cleanly.

(closes issue AST-1265)
Reported by: Alexander Hömig
(closes issue ASTERISK-22350)
Reported by: Birger "WIMPy" Harzenetter
Review: https://reviewboard.asterisk.org/r/3078/
........

Merged revisions 404344 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 404345 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      2882c5f9
    • Joshua Colp's avatar
      channel: Add a missing ast_channel_unlock when allocating a Surrogate channel. · eb235ad0
      Joshua Colp authored 
      ........

Merged revisions 404332 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      eb235ad0
    • Alexandr Anikin's avatar
      Handle temporary failures on gk registration · 1370d405
      Alexandr Anikin authored 
      Introduce new 'stopped' state for gk client and restart gk client
on failures
Remove ooh323 stack command lock as it is not need now.
(closes issue ASTERISK-21960)
Reported by: Dmitry Melekhov
Patches:
	ASTERISK-21960.patch
	ASTERISK-21960-stacklockup-2.patch
Tested by: Dmitry Melekhov
........

Merged revisions 404318 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 404320 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404321 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      1370d405
    • Damien Wedhorn's avatar
      Fixup some skinny bugs causing Fracks and ao2 cleanup issues. · c2fd229f
      Damien Wedhorn authored 
      Moved channel locking into setsubstate so that a process can complete
working on a sub before another starts changing it. The existing code
was causing some Fracks with schedule deletion.

Removed multiple rtp cleanup. Now only cleansup up once, fixing ao2 
object cleanup issues.
........

Merged revisions 404306 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404307 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      c2fd229f
    • Matthew Jordan's avatar
      app_cdr,app_forkcdr,func_cdr: Synchronize with engine when manipulating state · 7e9febbf
      Matthew Jordan authored 
      When doing the rework of the CDR engine that pushed all of the logic into cdr.c
and made it respond to changes in channel state over Stasis, we knew that
accessing the CDR engine from the dialplan would be "slightly"
non-deterministic. Dialplan threads would be accessing CDRs while Stasis
threads would be updating the state of said CDRs - whereas in the past,
everything happened on the dialplan threads. Tests have shown that "slightly"
is in reality "very".

This patch synchronizes things by making the dialplan applications/functions
that manipulate CDRs do so over Stasis. ForkCDR, NoCDR, ResetCDR, CDR, and
CDR_PROP now all use Stasis to send their requests over to the CDR engine,
and synchronize on the channel Stasis topic via a subscription so that they
return their values/control to the dialplan at the appropriate time.

While going through this, the following changes were also made:
 * DISA, which can reset the CDR when a user successfully authenticates, now
   just uses the ResetCDR app to do this. This prevents having to duplicate
   the same Stasis synchronization logic in that application.
 * Answer no longer disables CDRs. It actually didn't work anyway - calling
   DISABLE on the channel's CDR doesn't stop the CDR from getting the Answer
   time - it just kills all CDRs on that channel, which isn't what the caller
   would intend.

(closes issue ASTERISK-22884)
(closes issue ASTERISK-22886)

Review: https://reviewboard.asterisk.org/r/3057/
........

Merged revisions 404294 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404295 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      7e9febbf
    • Damien Wedhorn's avatar
      Fixup skinny registration following network issues. · af723c65
      Damien Wedhorn authored 
      On session registration, if device is already reporting that it is
connected to a device, an innocuous packet (update time) is sent to
the already connected device. If the tcp connection is down, the
device will be unregistered and the new connection allowed.

Without this patch, network issues can see a situation where a device
can not reregister until after 3*timeout.
........

Merged revisions 404292 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404293 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      af723c65
  3. Dec 18, 2013
  5. Dec 17, 2013
  7. Dec 16, 2013
  9. Dec 15, 2013
  11. Dec 14, 2013
  13. Dec 13, 2013