In order to be more safe, some error handling code was changed to respect more
error conditions including the potential memory allocation failure for deleted
and heard message tracking introduced in 293004. However, last_message_index
returns -1 for zero messages (perhaps as expected) and was triggering the
stricter error checking. Because last_message_index is only called directly
in one place, just return 0 from open_mailbox (for file based storage) when no
messages are detected unless a real error has occurred.
(closes issue #18240)
Reported by: leobrown
Patches:
bug18240.1-6-2.diff.txt uploaded by alecdavis (license 585)
Tested by: pabelanger
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@294903 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Asterisk is just whining too much with this message: "No D-channels
available! Using Primary channel XXX as D-channel anyway!".
Filtered the message so it only comes out once if there is no D channel
available without an intervening D channel available period.
(closes issue #17270)
Reported by: jmls
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@294821 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The option packets not only never stopped, but if a realtime peer was not in
the peer list multiple options dialogs could accumulate over time. This
scenario has the potential to progress to the point of saturating a link just
from options packets. The fix was to ensure that the poke scheduler checks to
see if a peer is in the peer list before continuing to poke. The reason a peer
must be in the peer list to be able to properly manage an options dialog is
because otherwise the call pointer is lost when the peer is regenerated from
the database, which is how existing qualify dialogs are detected.
(closes issue #16382)
Reported by: lftsy
Patches:
bug16382-3.patch uploaded by jpeeler (license 325)
Tested by: zerohalo
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@294688 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Copied from some notes from the original author (Russell):
Deadlock scenario:
Thread 1: device state change thread
Holds - rdlock on contexts
Holds - hints lock
Waiting on channels container lock
Thread 2: SIP monitor thread
Holds the "iflock"
Holds a sip_pvt lock
Holds channel container lock
Waiting for a channel lock
Thread 3: A channel thread (chan_local in this case)
Holds 2 channel locks acquired within app_dial
Holds a 3rd channel lock it got inside of chan_local
Holds a local_pvt lock
Waiting on a rdlock of the contexts lock
A bunch of other threads waiting on a wrlock of the contexts lock
To address this deadlock, some locking order rules must be put in place and
enforced. Existing relevant rules:
1) channel lock before a pvt lock
2) contexts lock before hints lock
3) channels container before a channel
What's missing is some enforcement of the order when you involve more than any
two. To fix this problem, I put in some code that ensures that (at least in the
code paths involved in this bug) the locks in (3) come before the locks in (2).
To change the operation of thread 1 to comply, I converted the storage of hints
to an astobj2 container. This allows processing of hints without holding the
hints container lock. So, in the code path that led to thread 1's state, it no
longer holds either the contexts or hints lock while it attempts to lock the
channels container.
(closes issue #18165)
Reported by: antonio
ABE-2583
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@294384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This prevents a scheduler entry from leaking if we receive a 491 response when one is pending. If a scheduler entry leaks, the pvt it is associated my get destroyed before the scheduler entry fires, and then memory corruption and crashes can occur when the scheduled reinvite attempts to access and modify the memory of the destroyed pvt.
ABE-2543
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@294163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
dahdi-linux 2.4.0 (specifically commit 9034) added the capability for
the wctc4xxp to return more than a single packet of data in response to
a read. However, when decoding packets, codec_dahdi was still assuming
that the default number of samples was in each read.
In other words, each packet your provider sent you, regardless of size,
would result in 20 ms of decoded data (30 ms if decoding G723). If your
provider was sending 60 ms packets then codec_dahdi would end up
stripping 40 ms of data from each transcoded frame resulting in "choppy"
audio.
This would only affect systems where G729 packets are arriving in sizes
greater than 20ms or G723 packets arriving in sizes greater than 30ms.
DAHDI-744.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@293968 65c4cc65-6c06-0410-ace0-fbb531ad65f3
All parties are analog FXS ports.
1) A calls B.
2) A flash hooks to call C.
3) A flash hooks to bring C into 3-way call before C answers. (A and B hear ringback)
4) C answers
5) A continues to hear ringback during the 3-way call. (All parties can hear each other.)
* Fixed use of wrong variable in dahdi_bridge() that stopped ringback on
the wrong subchannel.
* Made several debug messages have more information.
A similar issue happens if B and C are SIP channels. B continues to hear
ringback. For some reason this only affects v1.8 and trunk.
* Don't start ringback on the real and 3-way subchannels when creating the
3-way conference. Removing this code is benign on v1.6.2 and earlier.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@293805 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When setting to zero/"no", the numeric default was shown making it not obvious
the disabled setting was respected.
(closes issue #18123)
Reported by: zerohalo
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@293722 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Change "Unable to get index, and nullok is not asserted" to "Unable to get
index for '<channel-name>' on channel <number> (<function>(), line
<number>)".
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@293639 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The comparison function logic was off, so the number of sessions for a given
mailbox were not being incremented properly. This problem caused the maximum
number of messages per folder to not be respected when simultaneously leaving
multiple voicemails just below the threshold.
These problems should be fixed by the above, but just in case:
Fixed resequence_mailbox to rely on the actual number of detected number of
files in a directory rather than just assuming only 10 messages more than the
maximum had been left. Also if more messages than the maximum are deleted they
are actually removed now.
The second purpose of this commit should have been separated out probably, but
is related to the above. Again, if the number of messages in a given voicemail
folder exceeds the maximum set limit make sure to allocate enough space for the
deleted and heard index tracking array.
A few random fixes:
There was a forgotten decrement of the inprocess count in imap_store_file.
When using IMAP storage, do not look in the directory where file based storage
messages may still reside and influence the message count.
Ensure to use only the first format in sendmail.
ABE-2516
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@293004 65c4cc65-6c06-0410-ace0-fbb531ad65f3
chan_local does some dangerous things involving deadlock avoidance.
tech_pvt functions like hangup and queue_frame are provided with a
locked channel upon entry. Those functions are completely safe as
long as you don't attempt to give up that channel lock, but that is
impossible to guarantee due to the required deadlock avoidance necessary
to lock both the tech_pvt and both channels involved.
In the past, we have tried to account for this by doing things like
setting a "glare" flag that indicates what function should destroy the
pvt. This was used in local_hangup and local_queue_frame to decided
who should destroy the pvt if they collided in separate threads. I
have removed the need to do this by converting all chan_local tech_pvts
to astobj2. This means we can ref a pvt before deadlock avoidance
and not have to worry about that pvt possibly getting destroyed under
us. It also cleans up where we destroy the tech_pvt. The only unlink
from the tech_pvt container occurs in local_hangup now, which is where
it should occur.
Since there still may be thread collisions on some functions like
local_hangup after deadlock avoidance, I have added some checks to detect
those collisions and exit appropriately. I think this patch is going to
solve quite a bit of weirdness we have had with local channels in the past.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@292866 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is a fix for when pressing the operator key after recording an unavailable,
busy, name, or temporary message in mailbox options. The operator key should not
be accepted here, but should be allowed during the message recording. If the
operator key is pressed during ensure the file is saved or deleted as
apporopriate. Also, ensure removal of temporary recorded files after an early
hang up or when message acceptance confirmation times out.
ABE-2518
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@292223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
There is a deadlock between dahdi_exception() and dahdi_indicate() for
analog ports. The call-waiting and three-way-calling feature can
experience deadlock if these features are trying to do something and an
event from the bridged channel happens at the same time.
Deadlock avoidance code added to obtain necessary channel locks before
attemting an operation with call-waiting and three-way-calling.
(closes issue #16847)
Reported by: shin-shoryuken
Patches:
issue_16847_v1.4.patch uploaded by rmudgett (license 664)
issue_16847_v1.6.2.patch uploaded by rmudgett (license 664)
issue_16847_v1.8_v2.patch uploaded by rmudgett (license 664)
Tested by: alecdavis, rmudgett
Review: https://reviewboard.asterisk.org/r/971/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@291643 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When an outgoing call is answered and hung up by the far end *very* quickly, we
may not read any frames and therefor end up with a call that displays the wrong
disposition/DIALSTATUS. The reason is because ast_queue_hangup() immediately
sets the _softhangup flag on the channel and then queues the HANGUP control
frame, but __ast_read refuses to read any frames if ast_check_hangup() indicates
that a hangup request has been made (which it will if _softhangup is set). So,
we end up losing control frames.
This change makes __ast_read continue to read frames even if a soft hangup has
been requested. It queues a hangup frame to make sure that __ast_read() will
still eventually return NULL.
Much thanks to David Vossel for all of the reviews, discussion, and help!
(closes issue #16946)
Reported by: davidw
Review: https://reviewboard.asterisk.org/r/740/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@291577 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A recent change was made to avoid a race condition on shutdown which only called
the end functions from the console thread. However, when pressing Ctrl-C the
quit handler is called from the signal handler thread.
(closes issue #17698)
Reported by: jmls
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@290862 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Use the directories found for the parent when using lib dependencies.
(closes issue #17314)
Reported by: tzafrir
Patches:
17314-withdeps.diff uploaded by qwell (license 4)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@290750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The scenario here is with a non P2P early media session. The reported time
length of DTMF presses are coming up short when sending to the remote side.
Currently the event duration is a running total that is incremented when sending
continuation packets. These continuation packets are only triggered upon
incoming media from the remote side, which means that the running total probably
is not going to end up matching the actual length of time Asterisk received
DTMF. This patch changes the end event duration to be lengthened if it is
detected that the end event is going to come up short.
Review: https://reviewboard.asterisk.org/r/957/
ABE-2476
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@289797 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit takes a simliar approach to 288112 and checks the dialplan to
determine the proper action for an incoming contact header as to whether or not
it should be decoded or not. sip_new was blindly always decoding the extension,
which also caused the outgoing contact header to be incorrect as well as failing
to match the encoded extension in the dialplan.
(closes issue #17892)
Reported by: wdoekes
Patches:
bug17892-1.patch uploaded by jpeeler (license 325)
Tested by: wdoekes
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@289699 65c4cc65-6c06-0410-ace0-fbb531ad65f3
On every incoming subscribe there is a iteration through all dialogs to find old subscribes and delete them. This is slow and not RFC conform. This was only needed in 1.2 cause a subscribe was not deleted when a dialog was destroyed, after 1.4 a subscribe get removed when its dialog is destroyed.
(closes issue #17950)
Reported by: schmidts
Tested by: schmidts
Review: https://reviewboard.asterisk.org/r/901/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@289622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Since the data being passed to the generator callback is on the stack of the
SMS() application, we must ensure that the generator is stopped before the
application exits.
ABE-2587
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@289424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The timeout to wait for an answer was being set to 0 when a device forwarded to another
extension. We don't always need the timeout set like this, so make it an optional
parameter, and don't use it in this case.
ABE-2544
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@289338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
