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r304096 | seanbright | 2011-01-25 20:24:58 -0500 (Tue, 25 Jan 2011) | 12 lines
Per the man page, setvbuf() must be called before any other operation on an open file.
We use setvbuf() to associate a buffer with a stream, but we have already written
to the open file. This works (by chance) on Linux, but fails on other platforms,
such as OpenSolaris.
(closes issue #16610)
Reported by: bklang
Patches:
setvbuf.patch uploaded by crjw (license 963)
Tested by: bklang, asgaroth, efutch
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r304006 | rmudgett | 2011-01-25 17:25:32 -0600 (Tue, 25 Jan 2011) | 15 lines
Merged revisions 304005 via svnmerge from
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r304005 | rmudgett | 2011-01-25 17:21:09 -0600 (Tue, 25 Jan 2011) | 8 lines
DTMF attended transfers sometimes fail for no apparent reason.
The loop in feature_request_and_dial() can exit when Party C has answered
without processing an AST_CONTROL_ANSWER. Also sometimes an
AST_CONTROL_ANSWER never happens even though Party C has answered.
Don't hangup Party C if he is up or we receive an AST_CONTROL_ANSWER.
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r303960 | twilson | 2011-01-25 16:02:42 -0600 (Tue, 25 Jan 2011) | 23 lines
Merged revisions 303906 via svnmerge from
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r303906 | twilson | 2011-01-25 14:50:59 -0600 (Tue, 25 Jan 2011) | 16 lines
Guard against retransmitting BYEs indefinitely
In the case of an attended transfer (A calls B, A atxfers to C) where
A becomes unreachable before replying to Asterisk's BYE, Asterisk can
sometimes retransmit the BYE indefinitely. This is because
__sip_autodestruct tests p->refer && !ast_test_flag(&p->flags[0],
SIP_ALREADYGONE and will then transmit a BYE. When this BYE times out,
it will not ever be marked as ALREADYGONE, so when __sip_autodestruct
is called again, we end up starting the cycle over.
This patch adds a call to sip_alreadygone(pkt->owner) in retrans_pkt
in the case of a BYE that has timed out. This should prevent Asterisk
from trying to transmit new BYE messages in the future.
Review: https://reviewboard.asterisk.org/r/1077/
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r303769 | rmudgett | 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) | 47 lines
Merged revisions 303765 via svnmerge from
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r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines
Sending out unnecessary PROCEEDING messages breaks overlap dialing.
Issue #16789 was a good idea. Unfortunately, it breaks overlap dialing
through Asterisk. There is not enough information available at this point
to know if dialing is complete. The ast_exists_extension(),
ast_matchmore_extension(), and ast_canmatch_extension() calls are not
adequate to detect a dial through extension pattern of "_9!".
Workaround is to use the dialplan Proceeding() application early in
non-dial through extensions.
* Effectively revert issue #16789.
* Allow outgoing overlap dialing to hear dialtone and other early media.
A PROGRESS "inband-information is now available" message is now sent after
the SETUP_ACKNOWLEDGE message for non-digital calls. An
AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE
messages for non-digital calls.
* Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was
inconsistent with the cause codes.
* Added better protection from sending out of sequence messages by
combining several flags into a single enum value representing call
progress level.
* Added diagnostic messages for deferred overlap digits handling corner
cases.
(closes issue #17085)
Reported by: shawkris
(closes issue #18509)
Reported by: wimpy
Patches:
issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664)
Expanded upon issue18509_early_media_v1.8_v3.patch to include analog
and SS7 because of backporting requirements.
Tested by: wimpy, rmudgett
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r303677 | jpeeler | 2011-01-25 10:59:28 -0600 (Tue, 25 Jan 2011) | 26 lines
Merged revisions 303676 via svnmerge from
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r303676 | jpeeler | 2011-01-25 10:58:29 -0600 (Tue, 25 Jan 2011) | 20 lines
Fix voicemail sequencing for file based storage.
A previous change was made to account for when the number of voicemail messages
exceeds the max limit to be handled properly, but it caused gaps in the messages
to not be properly handled. This has now been resolved.
In later non 1.4 branches, it appears that resequencing wasn't even occurring
due from what appears and accidental code removal.
(closes issue #18498)
Reported by: JJCinAZ
Patches:
bug18498v2.patch uploaded by jpeeler (license 325)
(closes issue #18486)
Reported by: bluefox
Patches:
bug18486.patch uploaded by jpeeler (license 325)
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r303548 | russell | 2011-01-24 14:49:53 -0600 (Mon, 24 Jan 2011) | 38 lines
Merged revisions 303546 via svnmerge from
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r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines
Fix channel redirect out of MeetMe() and other issues with channel softhangup.
Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped
working properly. This issue includes a patch that resolves the issue by
removing a call to ast_check_hangup() from app_meetme.c. I left that in my
patch, as it doesn't need to be there. However, the rest of the patch fixes
this problem with or without the change to app_meetme.
The key difference between what happens before and after this patch is the
effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(),
ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme
sees this which causes it to exit as intended. Checking ast_check_hangup()
caused app_meetme to exit earlier in the process, and the target of the
redirect saw the condition where ast_read() returned NULL.
Removing ast_check_hangup() works around the issue in app_meetme, but doesn't
solve the issue if another application did the same thing. There are also
other edge cases where if an application finishes at the same time that a
redirect happens, the target of the redirect will think that the channel hung
up. So, I made some changes in pbx.c to resolve it at a deeper level. There
are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to
abort the hangup process. My patch extends this to remove the END_OF_Q frame
from the channel's read queue, making the "abort hangup" more complete. This
same technique was used in every place where a softhangup flag was cleared.
(closes issue #18585)
Reported by: oej
Tested by: oej, wedhorn, russell
Review: https://reviewboard.asterisk.org/r/1082/
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r303285 | qwell | 2011-01-21 15:48:09 -0600 (Fri, 21 Jan 2011) | 15 lines
Merged revisions 303284 via svnmerge from
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r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) | 8 lines
Reset configuration before parsing users.conf.
Some values configured in chan_dahdi.conf were able to leak in to users.conf
configuration. This was surprising users, and potentially setting non-sane
"defaults".
ASTNOW-125
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r303285 | qwell | 2011-01-21 15:48:09 -0600 (Fri, 21 Jan 2011) | 15 lines
Merged revisions 303284 via svnmerge from
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r303284 | qwell | 2011-01-21 15:45:34 -0600 (Fri, 21 Jan 2011) | 8 lines
Reset configuration before parsing users.conf.
Some values configured in chan_dahdi.conf were able to leak in to users.conf
configuration. This was surprising users, and potentially setting non-sane
"defaults".
ASTNOW-125
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r303273 | lmadsen | 2011-01-21 10:12:54 -0600 (Fri, 21 Jan 2011) | 9 lines
Fix changes to L() flag in Dial().
Tony Mountifield pointed out an error I had in my patch. I was a bit too aggressive
on changing 'seconds' to 'milliseconds'. So I decided to do some additioanl testing
and have no changed just the appropriate lines. One line says milliseconds, and the
other says seconds. Probably should change this to be either just seconds or
milliseconds, but I've spent too much time on this already :)
(issue #18264)
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r303098 | rmudgett | 2011-01-20 12:11:45 -0600 (Thu, 20 Jan 2011) | 15 lines
CC_INTERFACES does not get built correctly with local channels.
If local channels are used with CCSS, CC_INTERFACES gets garbage prepended
to it so the CC recall fails. Also CC_INTERFACES gets "&(null)" appended
to it.
* Initialize the buffer to eliminate the prepended garbage.
* Filter out the empty interface strings to eliminate the latter.
* Added a diagnostic message if the CC_INTERFACES is ever empty.
JIRA ABE-2740
JIRA SWP-2848
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r303106 | sruffell | 2011-01-20 13:56:34 -0600 (Thu, 20 Jan 2011) | 15 lines
main/features: Use POLLPRI when waiting for events on parked channels.
This change resolves a regression in the 1.6.2 when converting from
select to poll. The DAHDI timers use POLLPRI to indicate that the timer
fired, but features was not waiting for that flag. The result was no
audio for MOH when a call was parked and res_timing_dahdi was in use.
This patch is slightly modified from the one on the mantis issue. It does
not set an exception on the channel if the POLLPRI flag is set.
(closes issue #18262)
Reported by: francesco_r
Patches:
patch_park_moh-trunk-2.txt uploaded by cjacobsen (license 1029)
Tested by: francesco_r, rfrantik, one47
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r303008 | jpeeler | 2011-01-20 11:07:44 -0600 (Thu, 20 Jan 2011) | 14 lines
Merged revisions 303007 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r303007 | jpeeler | 2011-01-20 11:04:08 -0600 (Thu, 20 Jan 2011) | 8 lines
Add new queue strategy to preserve behavior for when queue members moved to ao2.
Add queue strategy called "rrordered" to mimic old behavior from when queue
members were stored in a linked list.
ABE-2707
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r302917 | lmadsen | 2011-01-20 09:42:05 -0600 (Thu, 20 Jan 2011) | 8 lines
Option L() is milliseconds, not seconds.
> Change the verbose output of option L() to say milliseconds and not seconds
> as the value is in milliseconds.
>
> (closes issue #18264)
> Reported by: jacco
> Patches:
> app_dial_patch.txt uploaded by lmadsen (license 10)
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The intent of this check as it stands in previous versions of Asterisk was to
check if there are any active sessions. If there were no sessions, then the
function would return immediately and not bother with queueing up the manager
event to be processed. Since the conversion of storing sessions in an astobj2
container, this check will always pass. I changed it to go back to checking
what was intended.
The side effect of this was that if the AMI is disabled, the manager event
queue is populated anyway, but the code that runs to clear out the queue
never runs. A producer with no consumer is a bad thing.
Reported internally by kmorgan.
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r302693 | rmudgett | 2011-01-19 15:25:41 -0600 (Wed, 19 Jan 2011) | 22 lines
Merged revisions 302671 via svnmerge from
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r302671 | rmudgett | 2011-01-19 15:21:56 -0600 (Wed, 19 Jan 2011) | 15 lines
DTMF transfer plays the wrong sounds for wrong number or other call failure.
* Set the default for features.conf.sample xferfailsound option to "beeperr"
as documented instead of "pbx-invalid" and corrected the use of it in DTMF
blind transfer (#1).
* Improved DTMF blind transfer handling of wrong numbers.
Most of the concerns in this issue were taken care of by the patch for
issue 17999: Issues with DTMF triggered attended transfers.
(closes issue #18379)
Reported by: gincantalupo
Tested by: rmudgett
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r302599 | tilghman | 2011-01-19 14:13:24 -0600 (Wed, 19 Jan 2011) | 15 lines
Kill zombies.
When we ast_safe_fork() with a non-zero argument, we're expected to reap our
own zombies. On a zero argument, however, the zombies are only reaped when
there aren't any non-zero forked children alive. At other times, we
accumulate zombies. This code is forward ported from res_agi in 1.4, so that
forked children are always reaped, thus preventing an accumulation of zombie
processes.
(closes issue #18515)
Reported by: ernied
Patches:
20101221__issue18515.diff.txt uploaded by tilghman (license 14)
Tested by: ernied
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r302548 | seanbright | 2011-01-19 13:37:09 -0500 (Wed, 19 Jan 2011) | 10 lines
Properly handle partial reads from fgets() when handling AGIs.
When fgets() failed with EAGAIN, we were continually decrementing the available
space left in our buffer, resulting in botched command handling.
(closes issue #16032)
Reported by: notahat
Patches:
agi_buffer_patch2.diff uploaded by fnordian (license 110)
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We were passing and storing the requested format as an int instead of format_t
resulting in truncation.
(closes issue #18238)
Reported by: whizemen
Patches:
0018238_speex16.patch uploaded by whizemen (license 1143)
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r302265 | jpeeler | 2011-01-18 14:13:52 -0600 (Tue, 18 Jan 2011) | 27 lines
Convert device state callbacks to ao2 objects to fix a deadlock in chan_sip.
Lock scenario presented here:
Thread 1
holds ast_rdlock_contexts &conlock
holds handle_statechange hints
holds handle_statechange hint
waiting for cb_extensionstate
Locked Here: chan_sip.c line 7428 (find_call)
Thread 2
holds handle_request_do &netlock
holds find_call sip_pvt_ptr
waiting for ast_rdlock_contexts &conlock
Locked Here: pbx.c line 9911 (ast_rdlock_contexts)
Chan_sip has an established locking order of locking the sip_pvt and then
getting the context lock. So the as stated by the summary, the operations in
thread 2 have been modified to no longer require the context lock.
(closes issue #18310)
Reported by: one47
Patches:
statecbs_ao2.mk2.patch uploaded by one47 (license 23),
modified by me
Review: https://reviewboard.asterisk.org/r/1072/
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r302173 | rmudgett | 2011-01-18 12:07:15 -0600 (Tue, 18 Jan 2011) | 95 lines
Merged revisions 302172 via svnmerge from
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r302172 | rmudgett | 2011-01-18 12:04:36 -0600 (Tue, 18 Jan 2011) | 88 lines
Issues with DTMF triggered attended transfers.
Issue #17999
1) A calls B. B answers.
2) B using DTMF dial *2 (code in features.conf for attended transfer).
3) A hears MOH. B dial number C
4) C ringing. A hears MOH.
5) B hangup. A still hears MOH. C ringing.
6) A hangup. C still ringing until "atxfernoanswertimeout" expires.
For v1.4 C will ring forever until C answers the dead line. (Issue #17096)
Problem: When A and B hangup, C is still ringing.
Issue #18395
SIP call limit of B is 1
1. A call B, B answered
2. B *2(atxfer) call C
3. B hangup, C ringing
4. Timeout waiting for C to answer
5. Recall to B fails because B has reached its call limit.
Because B reached its call limit, it cannot do anything until the transfer
it started completes.
Issue #17273
Same scenario as issue 18395 but party B is an FXS port. Party B cannot
do anything until the transfer it started completes. If B goes back off
hook before C answers, B hears ringback instead of the expected dialtone.
**********
Note for the issue #17273 and #18395 fix:
DTMF attended transfer works within the channel bridge. Unfortunately,
when either party A or B in the channel bridge hangs up, that channel is
not completely hung up until the transfer completes. This is a real
problem depending upon the channel technology involved.
For chan_dahdi, the channel is crippled until the hangup is complete.
Either the channel is not useable (analog) or the protocol disconnect
messages are held up (PRI/BRI/SS7) and the media is not released.
For chan_sip, a call limit of one is going to block that endpoint from any
further calls until the hangup is complete.
For party A this is a minor problem. The party A channel will only be in
this condition while party B is dialing and when party B and C are
conferring. The conversation between party B and C is expected to be a
short one. Party B is either asking a question of party C or announcing
party A. Also party A does not have much incentive to hangup at this
point.
For party B this can be a major problem during a blonde transfer. (A
blonde transfer is our term for an attended transfer that is converted
into a blind transfer. :)) Party B could be the operator. When party B
hangs up, he assumes that he is out of the original call entirely. The
party B channel will be in this condition while party C is ringing, while
attempting to recall party B, and while waiting between call attempts.
WARNING:
The ATXFER_NULL_TECH conditional is a hack to fix the problem. It will
replace the party B channel technology with a NULL channel driver to
complete hanging up the party B channel technology. The consequences of
this code is that the 'h' extension will not be able to access any channel
technology specific information like SIP statistics for the call.
ATXFER_NULL_TECH is not defined by default.
**********
(closes issue #17999)
Reported by: iskatel
Tested by: rmudgett
JIRA SWP-2246
(closes issue #17096)
Reported by: gelo
Tested by: rmudgett
JIRA SWP-1192
(closes issue #18395)
Reported by: shihchuan
Tested by: rmudgett
(closes issue #17273)
Reported by: grecco
Tested by: rmudgett
Review: https://reviewboard.asterisk.org/r/1047/
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The sig_pri_new_ast_channel() is called with the channel private lock held
when pri_dchannel() calls it and no channel private lock held when
dahdi_request() calls it. The use of pri_grab() in
sig_pri_new_ast_channel() could leave the channel private lock held when
it returns if the lock was not held before calling it.
Make sig_pri_new_ast_channel() just lock the PRI span lock instead of
using pri_grab(). It is safe to do this because dahdi_request() does not
have the channel private lock and the deadlock potential with the PRI span
lock is only between pri_dchannel() and other threads.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@301946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Two fixes:
1) One must always have the private unlocked before calling
pbx_builtin_setvar_helper to not invalidate locking order since it locks the
channel.
2) Unlock the channel before calling pbx_find_extension, which starts and stops
autoservice during the lookup. The problem scenario as illustrated by the
reporter:
Thread: do_monitor
-----------------------
handle_request_do
handle_incoming
handle_request_refer
ast_parking_ext_valid
pbx_find_extension
ast_autoservice_stop
while (chan_list_state == as_chan_list_state) { usleep(1000); }
Thread: autoservice_run
-----------------------
autoservice_run
chan = ast_waitfor_n
ast_waitfor_nandfds
ast_waitfor_nandfds_classic / simple / complex (depending on your system)
ast_channel_lock(c[x]);
handle_request_do and schedule_process_request_queue locks the owner
if it exists. The autoservice thread is waiting for the channel lock, which
wasn't ever released since the do_monitor thread was waiting for autoservice
operations to complete. Solved by unlocking the channel but keeping a reference
to guarantee safety.
(closes issue #18403)
Reported by: jthurman
Patches:
20110103-blind_deadlock.diff uploaded by jthurman (license 614)
issue18403.patch uploaded by jpeeler (license 325)
Tested by: jthurman
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