* Analog call waiting Caller ID spills could get stuck resulting in one
way audio until the waiting call is answered. This only happens on the
second (and later) call waiting call if the active call is not the first
call.
* The CLI/AMI "dahdi show channel" command could report the wrong channel
information.
Must keep the struct analog_pvt.owner and struct dahdi_pvt.owner pointer
in sync.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@295747 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r295710 | russell | 2010-11-19 18:45:51 -0600 (Fri, 19 Nov 2010) | 29 lines
Fix cache of device state changes for multiple servers.
This patch addresses a regression where device states across multiple servers
were not being processing completely correctly. The code works to determine
the overall state by looking at the last known state of a device on each
server. However, there was a regression due to some invasive rewrites of how
the cache works that led to the cache only storing the last device state change
for a device, regardless of which server it was on.
The code is set up to cache device state change events by ensuring that each
event in the cache has a unique device name + entity ID (server ID). The code
that was responsible for comparing raw information elements (which EID is)
always returned a match due to a memcmp() with a length of 0.
There isn't much code to fix the actual bug. This patch also introduces a new
CLI command that was very useful for debugging this problem. The command
allows you to dump the contents of the event cache.
(closes issue #18284)
Reported by: klaus3000
Patches:
issue18284.rev1.txt uploaded by russell (license 2)
Tested by: russell, klaus3000
(closes issue #18280)
Reported by: klaus3000
Review: https://reviewboard.asterisk.org/r/1012/
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* Restore SMDI support.
* Fixed initial value of struct analog_pvt.use_callerid. It may get
forced on depending upon other config options.
* Call analog_dnd() instead of manual inlined code.
* Removed unused struct analog_pvt.usedistinctiveringdetection.
* Removed the struct analog_pvt.unknown_alarm flag. It was really the
struct analog_pvt.inalarm flag.
* Use ast_debug() instead of ast_log(LOG_DEBUG).
* Rename several function's index variable to idx.
* Some formatting tweaks.
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r294904 | jpeeler | 2010-11-12 14:51:15 -0600 (Fri, 12 Nov 2010) | 23 lines
Merged revisions 294903 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r294903 | jpeeler | 2010-11-12 14:49:09 -0600 (Fri, 12 Nov 2010) | 16 lines
Fix regression causing abort in voicemail after opening a mailbox with no mesgs.
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
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r294822 | rmudgett | 2010-11-11 20:44:12 -0600 (Thu, 11 Nov 2010) | 18 lines
Merged revisions 294821 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r294821 | rmudgett | 2010-11-11 20:41:13 -0600 (Thu, 11 Nov 2010) | 11 lines
Asterisk is getting a "No D-channels available!" warning message every 4 seconds.
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
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This merge does the following things:
* Removes most of the contents from the doc/ directory in favor
of the wiki - http://wiki.asterisk.org/
* Updates the build_tools/prep_tarball script to know how to export
the contents of the wiki in both PDF and plain text formats so that
the documentation is still included in Asterisk release tarballs.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294740 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r294733 | jpeeler | 2010-11-11 15:57:22 -0600 (Thu, 11 Nov 2010) | 25 lines
Merged revisions 294688 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r294688 | jpeeler | 2010-11-11 15:12:27 -0600 (Thu, 11 Nov 2010) | 18 lines
Fix problem with qualify option packets for realtime peers never stopping.
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)
(closes issue #17779)
Reported by: lftsy
Patches:
bug16382-3.patch uploaded by jpeeler (license 325)
Tested by: zerohalo
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r294639 | jpeeler | 2010-11-11 13:31:00 -0600 (Thu, 11 Nov 2010) | 53 lines
Merged revisions 294384 via svnmerge from
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r294384 | jpeeler | 2010-11-09 11:37:59 -0600 (Tue, 09 Nov 2010) | 47 lines
Fix a deadlock in device state change processing.
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
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Add connected line update for sig_analog transfers and simplify the
corresponding sig_pri and chan_misdn transfer code.
Note that if you create a three-way call in sig_analog before transferring
the call, the distinction of the caller/callee interception macros make
little sense. The interception macro writer needs to be prepared for
either caller/callee macro to be executed. The current implementation
swaps which caller/callee interception macro is executed after a three-way
call is created.
Review: https://reviewboard.asterisk.org/r/996/
JIRA ABE-2589
JIRA SWP-2372
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r294277 | jpeeler | 2010-11-08 15:58:13 -0600 (Mon, 08 Nov 2010) | 16 lines
Fix playback failure when using IAX with the timerfd module.
To fix this issue the alert pipe will now be used when the timerfd module is
in use. There appeared to be a race that was not solved by adding locking in the
timerfd module, but needed to be there anyway. The race was between the timer
being put in non-continuous mode in ast_read on the channel thread and the IAX
frame scheduler queuing a frame which would enable continuous mode before the
non-continuous mode event was read. This race for now is simply avoided.
(closes issue #18110)
Reported by: tpanton
Tested by: tpanton
I put tested by tpanton because it was tested on his hardware. Thanks for the
remote access to debug this issue!
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@294278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Bad things have been happening in chan_misdn because the chan_misdn
channel private struct chan_list is not protected from reentrancy. Hangup
collisions have be causing read and write accesses to freed memory.
Converted chan_misdn struct chan_list to an ao2 object for its reference
counting feature.
**********
Removed an impediment to converting chan_list to an ao2 object.
The use of the other_ch member in chan_list is shaky at best. It is set
if the incoming and outgoing call legs are mISDN. The use of the other_ch
member goes against the Asterisk architecture and can even cause problems.
1) It is used to disable echo cancellation. This could be bad if the call
is forked and the winning call leg is not mISDN or the winning call leg is
not the last mISDN channel called by the fork. The other_ch would become
a dangling pointer.
2) It is used when the far end is alerting to hear the far end's inband
audio instead of Asterisk's generated ringback tone. This is bad if the
call is forked. You would only hear the last forked mISDN channel and it
may not be ringing yet.
The other_ch would become a dangling pointer if the call is later
transferred.
**********
JIRA SWP-2423
JIRA ABE-2614
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r293969 | sruffell | 2010-11-04 19:06:02 -0500 (Thu, 04 Nov 2010) | 25 lines
Merged revisions 293968 via svnmerge from
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r293968 | sruffell | 2010-11-04 19:02:53 -0500 (Thu, 04 Nov 2010) | 17 lines
codecs/codec_dahdi: Prevent "choppy" audio when receiving unexpected frame sizes.
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.
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r293806 | rmudgett | 2010-11-03 13:31:57 -0500 (Wed, 03 Nov 2010) | 27 lines
Merged revisions 293805 via svnmerge from
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r293805 | rmudgett | 2010-11-03 13:23:04 -0500 (Wed, 03 Nov 2010) | 20 lines
Party A in an analog 3-way call would continue to hear ringback after party C answers.
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.
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The documentation for ast_rtp_instance_get_(local/remote)_address stated that
they returned 0 for success and -1 on failure. Instead, they returned 0 if the
address structure passed in was already equivalent to the address instance
local/remote address or 1 otherwise. 90% of the calls to these functions
completely ignored the return address and passed in an uninitialized struct,
which would make valgrind complain even though the operation was technically
safe.
This patch fixes the documentation and converts the get_xxx_address functions
to void since all they really do is copy the address and cannot fail.
Additionally two new functions
(ast_rtp_instance_get_and_cmp_(local/remote)_address) are created for the 3
times where the return value was actually checked. The
get_and_cmp_local_address function is currently unused, but exists for the sake
of symmetry.
The only functional change as a result of this change is that we will not do an
ast_sockaddr_cmp() on (mostly uninitialized) addresses before doing the
ast_sockaddr_copy() in the get_*_address functions. So, even though it is an
API change, it shouldn't have a noticeable change in behavior.
Review: https://reviewboard.asterisk.org/r/995/
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r293723 | jpeeler | 2010-11-02 18:07:13 -0500 (Tue, 02 Nov 2010) | 15 lines
Merged revisions 293722 via svnmerge from
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r293722 | jpeeler | 2010-11-02 18:02:51 -0500 (Tue, 02 Nov 2010) | 8 lines
Add enabled/disabled information for rtautoclear sip show settings output.
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
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r293647 | rmudgett | 2010-11-02 16:26:30 -0500 (Tue, 02 Nov 2010) | 13 lines
Merged revisions 293639 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r293639 | rmudgett | 2010-11-02 16:24:13 -0500 (Tue, 02 Nov 2010) | 6 lines
Make warning message have more useful information in it.
Change "Unable to get index, and nullok is not asserted" to "Unable to get
index for '<channel-name>' on channel <number> (<function>(), line
<number>)".
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Also the "dahdi show channel" would not show the correct 3-way call
status.
* Synchronized the inthreeway flag between chan_dahdi and sig_analog.
* Fixed a my_set_linear_mode() sign error and made take an analog sub
channel enum.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When using directmedia, Asterisk needs to limit the codecs offered to just
the ones that both sides recognize, otherwise they may end up sending audio
that the other side doesn't understand.
(closes issue #17403)
Reported by: one47
Patches:
sip_codecs_simplified4 uploaded by one47 (license 23)
Tested by: one47, falves11
Review: https://reviewboard.asterisk.org/r/967/
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@293493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
