Check for errors from fseek() when loading config file, properly abort on errors from fread(), and supply a traceback for errors generated when loading the config file.
Also, prepend a newline to traceback output so that the main error message is on it's own line.
Jason Parker [Wed, 2 Mar 2011 19:53:47 +0000 (19:53 +0000)]
Fix usage of "hasvoicemail=yes" and "mailbox=" in users.conf for SIP.
Since it's a duplicate, nothing is going to be done, so delme doesn't need to
be set at all. Strangely, when this was added, this was being set to 1 in 1.6,
and 0 in trunk.
Tilghman Lesher [Wed, 2 Mar 2011 01:06:02 +0000 (01:06 +0000)]
Revert previous 2 commits, and instead conditionally redefine the same macro used in flex 2.5.35 that clashed with our workaround.
Not surprisingly, the workaround was exactly the same code as was provided by
the Flex maintainers, albeit in two different places, in different macros.
This should fix the FreeBSD builds, which have an older version of Flex.
Don't broadcast FullyBooted to every AMI connection
The FullyBooted event should not be sent to every AMI connection every
time someone connects via AMI. It should only be sent to the user who
just connected.
Terry Wilson [Thu, 24 Feb 2011 03:38:22 +0000 (03:38 +0000)]
Use remotesecret to authenticate with a remote party
The remotesecret option was only being used for outbound registration
and not for placing calls. This patch uses remotesecret on outbound
calls if it is set, otherwise secret is still used.
Cleaning up a few things in detect disconnect patch
Initialized ast_call_feature in detect_disconnect to avoid accessing uninitialized memory. Cleaned up /param tags in features.h. No longer send dynamic features in ast_feature_detect.
feature.conf has a disconnect option. By default this option is set to '*', but it could be anything. If a user wishes to disconnect a call before the other side answers, only '*' will work, regardless if the disconnect option is set to something else. This is because features are unavailable until bridging takes place. The default disconnect option, '*', was hardcoded in app_dial, which doesn't make any sense from a user perspective since they may expect it to be something different. This patch allows features to be detected from outside of the bridge, but not operated on. In this case, the disconnect feature can be detected before briding and handled outside of features.c.
Properly check the bounds of arrays when decoding UDPTL packets. Also, remove broken support for receiving UDPTL packets larger than 16k. That shouldn't ever happen anyway.
Fix regression that changed behavior of queues when ringing a queue member.
This reverts r298596, which was to fix a highly bizarre and contrived issue
with a queue member that called into his own queue being transferred back
into his own queue. I couldn't reproduce that issue in any way. I think one
of the other recent transfer fixes actually fixed this.
Reentrancy problem if outgoing call gets different B channel than requested.
The chan_dahdi pri_fixup_principle() routine needs to protect the Asterisk
channel with the channel lock when it changes the technology private
pointer to a new private structure.
* Added lock protection while pri_fixup_principle() moves a call from one
private structure to another.
* Made some pri_fixup_principle() messages more meaningful.
Don't try to pickup a call in the middle of a masquerade
If A calls B which doesn't answer and C & D both try to do a call pickup, it is
possible for ast_pickup_call to answer the call, then fail to masquerade one of
the calls because the other one is already in the process of masquerading. This
patch checks to see if the channel is in the process of masquerading before
call before selecting it for a pickup.
Don't allow a REFER w/replaces to replace its own dialog
Asterisk currently accepts a REFER with a Refer-To with an embedded Replaces
header that matches the dialog of the REFER. This would be a situation like A
calls B, A calls C, A transfers B to A, which is just silly. This patch makes
the transfer fail instead of making Asterisk freak out and forget to hang other
channels up.
When a call involves a local channel (like SIP -> Local -> SIP), the hangup
cause was not being set. This resulted in SIP channels sometimes getting a
503 error instead of a 486 when the far side sent a busy. In Asterisk 1.8+
this also can cause issues with CCSS that involve a local channel. This patch
sets the hangupcause for one side of the local channel to the other in
local_hangup for outbound calls.
........
Jeff Peeler [Thu, 3 Feb 2011 20:49:48 +0000 (20:49 +0000)]
Set exception on channel in parking thread when POLLPRI event detected.
This is done just to make the code be equivalent to the old select code. As
noted in 303106 the same issue was already fixed in this branch, but the
exception was not set on the channel in the case of POLLPRI. The reason that
this did not cause a problem here is because in 122923 the check in __ast_read
to check the exception flag was removed.
* Include the null terminator in the buffer length. When the frame is
queued it is copied. If the null terminator is not part of the frame
buffer length, the receiver could see garbage appended onto it.
* Add channel lock protection with ast_sendtext().
* Fixed AMI SendText action ast_sendtext() return value check.
........
Prevent a crash when dialing a technology with no destination (ex: Dial(SIP/))
chan_iax2 and other channel drivers already had code to prevent this. The
attempt that app_dial was making to prevent it was not correct, so I fixed that.
Sean Bright [Sat, 29 Jan 2011 17:51:28 +0000 (17:51 +0000)]
When we pass the S() or L() options to MeetMe, make sure that we honor C as well.
Without this patch, if the user was kicked from the conference via the S() or L()
mechanism, we would just hang up on them even if we also passed C (continue in
dialplan when kicked). With this patch we honor the C flag in those cases.
Sean Bright [Sat, 29 Jan 2011 17:01:51 +0000 (17:01 +0000)]
Make sure that we unref the correct object when ejecting the most recent caller.
Currently, when we kick the last user to enter, we decrement our own reference
count which results in a crash when we kick another user or when we exit the
conference ourselves.
This will fix #18225 in 1.8 and trunk, but that particular bug does not exist in
1.6.2.
Sean Bright [Fri, 28 Jan 2011 21:22:09 +0000 (21:22 +0000)]
Don't leak references if we can't create a pseudo channel for mixing in MeetMe.
If there was a problem allocating a pseudo channel when building our meetme, we
weren't destroying our user container or destroying the mutexes that we created.
Fix default prefix=/usr regression on non-Linux systems.
This partially reverts a change made in branches/1.4/ r267759, which will
cause issue #17013 to be reopened. This issue was pointed out by a user
on #asterisk, who helpfully discovered that paths were being set incorrectly.
To truly understand what was wrong, one should run:
svn diff --force -c<this revision> configure
........
This patch modifies chan_sip to route responses to the address the request came from. It also modifies chan_sip to respect the maddr parameter in the Via header.
Sean Bright [Wed, 26 Jan 2011 01:24:58 +0000 (01:24 +0000)]
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.
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.
........
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.
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.
(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
........
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.
Russell Bryant [Mon, 24 Jan 2011 20:49:53 +0000 (20:49 +0000)]
Merged revisions 303546 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
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.
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".
Leif Madsen [Fri, 21 Jan 2011 16:12:54 +0000 (16:12 +0000)]
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 :)
Shaun Ruffell [Thu, 20 Jan 2011 19:56:34 +0000 (19:56 +0000)]
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.
Leif Madsen [Thu, 20 Jan 2011 15:42:05 +0000 (15:42 +0000)]
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)
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.
Tilghman Lesher [Wed, 19 Jan 2011 20:13:24 +0000 (20:13 +0000)]
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.
Jeff Peeler [Tue, 18 Jan 2011 20:13:52 +0000 (20:13 +0000)]
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
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.
**********
Only offer codecs both sides support for directmedia
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.
Fix CPU spike when pressing DTMF after agent login.
The problem here is that DTMF was being continuously deferred and requeued
since ast_safe_sleep is called in a loop. There are serveral other places in the
code that sleeps and then loops in a similar fashion. Because of this fact I
opted to not defer DTMF any more, which will not affect the original fix:
Paul Belanger [Sun, 9 Jan 2011 21:38:24 +0000 (21:38 +0000)]
SOUND_CACHE_DIR now defaults to empty
Sounds files included in the Asterisk tarball were being ignored and
re-downloaded. Users wanting to cache the files can still override the setting
using the --with-sounds-cache option.
projects / thirdparty / asterisk.git / log
