don't generate D-Channel "up" and "down" messages unless the channel state is actually changing; also, generate the "up" message when an implicit "up" occurs due to reception of a normal event when we thought the channel was "down"
Russell Bryant [Fri, 7 Mar 2008 17:16:58 +0000 (17:16 +0000)]
Change a warning message to a debug message. This is happening quite frequently,
and it is not worth spamming users with these messages unless we are pretty confident
that it should never happen. As it stands today, it _will_ and _does_ happen and
until that gets cleaned up a reasonable amount on the development side, let's not
spam the logs of everyone else.
Mark Michelson [Thu, 6 Mar 2008 22:10:07 +0000 (22:10 +0000)]
Quell an annoying message that is likely to print every single time that
ast_pbx_outgoing_app is called. The reason is that __ast_request_and_dial
allocates the cdr for the channel, so it should be expected that the channel
will have a cdr on it.
Joshua Colp [Wed, 5 Mar 2008 22:32:10 +0000 (22:32 +0000)]
Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things.
(closes issue #12148)
Reported by: jcomellas
document var_metric so no bugreports will come in when it's actually a configuration issue.
(issue #12151)
Reported and patched by: caio1982
1.4 patch by me
when a PRI call must be moved to a different B channel at the request of the other endpoint, ensure that any DSP active on the original channel is moved to the new one
Tilghman Lesher [Wed, 5 Mar 2008 15:17:16 +0000 (15:17 +0000)]
Correctly initialize retransid in SIP, and ensure that the warning when failing to delete a schedule entry can actually hit the log.
(closes issue #12140)
Reported by: slavon
Patches:
sch2.patch uploaded by slavon (license 288)
(Patch slightly modified by me)
Russell Bryant [Wed, 5 Mar 2008 01:52:18 +0000 (01:52 +0000)]
Fix a bug that I just noticed in the RTP code. The calculation for setting the
len field in an ast_frame of audio was wrong when G.722 is in use. The len field
represents the number of ms of audio that the frame contains. It would have
set the value to be twice what it should be.
Joshua Colp [Tue, 4 Mar 2008 18:05:28 +0000 (18:05 +0000)]
When a new source of audio comes in (such as music on hold) make sure the marker bit gets set.
(closes issue #10355)
Reported by: wdecarne
Patches:
10355.diff uploaded by file (license 11)
(closes issue #11491)
Reported by: kanderson
Russell Bryant [Tue, 4 Mar 2008 04:31:29 +0000 (04:31 +0000)]
Backport a minor bug fix from trunk that I found while doing random code
cleanup. Properly break out of the loop when a context isn't found when
verify that includes are valid.
Russell Bryant [Mon, 3 Mar 2008 17:05:16 +0000 (17:05 +0000)]
Fix a potential memory leak of the local_pvt struct when ast_channel allocation
fails. Also, in passing, centralize the code necessary to destroy a local_pvt.
Russell Bryant [Mon, 3 Mar 2008 15:50:43 +0000 (15:50 +0000)]
Merge in some changes from team/russell/autoservice-nochans-1.4
These changes fix up some dubious code that I came across while auditing what
happens in the autoservice thread when there are no channels currently in
autoservice.
1) Change it so that autoservice thread doesn't keep looping around calling
ast_waitfor_n() on 0 channels twice a second. Instead, use a thread condition
so that the thread properly goes to sleep and does not wake up until a
channel is put into autoservice.
This actually fixes an interesting bug, as well. If the autoservice thread
is already running (almost always is the case), then when the thread goes
from having 0 channels to have 1 channel to autoservice, that channel would
have to wait for up to 1/2 of a second to have the first frame read from it.
2) Fix up the code in ast_waitfor_nandfds() for when it gets called with no
channels and no fds to poll() on, such as was the case with the previous code
for the autoservice thread. In this case, the code would call alloca(0), and
pass the result as the first argument to poll(). In this case, the 2nd
argument to poll() specified that there were no fds, so this invalid pointer
shouldn't actually get dereferenced, but, this code makes it explicit and
ensures the pointers are NULL unless we have valid data to put there.
Joshua Colp [Mon, 3 Mar 2008 15:28:59 +0000 (15:28 +0000)]
It is possible for no audio to pass between the current digit and next digit so expand logic that clears emulation to AST_FRAME_NULL.
(closes issue #11911)
Reported by: edgreenberg
Patches:
v1-11911.patch uploaded by dimas (license 88)
Tested by: tbsky
Joshua Colp [Mon, 3 Mar 2008 15:15:39 +0000 (15:15 +0000)]
Add a comment to describe some logic.
(closes issue #12120)
Reported by: flefoll
Patches:
chan_sip.c.br14.patch-just-a-comment uploaded by flefoll (license 244)
Russell Bryant [Fri, 29 Feb 2008 23:34:32 +0000 (23:34 +0000)]
Fix a major bug in autoservice. There was a race condition in the handling of
the list of channels in autoservice. The problem was that it was possible for
a channel to get removed from autoservice and destroyed, while the autoservice
thread was still messing with the channel. This led to memory corruption, and
caused crashes. This explains multiple backtraces I have seen that have
references to autoservice, but do to the nature of the issue (memory corruption),
could cause crashes in a number of areas.
(fixes the crash in BE-386)
(closes issue #11694)
(closes issue #11940)
The following issues could be related. If you are the reporter of one of these,
please update to include this fix and try again.
Russell Bryant [Thu, 28 Feb 2008 22:23:05 +0000 (22:23 +0000)]
Fix a bug in the lock tracking code that was discovered by mmichelson. The issue
is that if the lock history array was full, then the functions to mark a lock as
acquired or not would adjust the stats for whatever lock is at the end of the array,
which may not be itself. So, do a sanity check to make sure that we're updating
lock info for the proper lock.
(This explains the bizarre stats on lock #63 in BE-396, thanks Mark!)
Jason Parker [Thu, 28 Feb 2008 19:20:10 +0000 (19:20 +0000)]
Make pbx_exec pass an empty string into applications, if we get NULL.
This protects against possible segfaults in applications that may try
to use data before checking length (ast_strdupa'ing it, for example)
Mark Michelson [Wed, 27 Feb 2008 21:49:20 +0000 (21:49 +0000)]
Two fixes:
1. Make the list of ast_dial_channels a lockable list. This is because in some cases,
the ast_dial may exist in multiple threads due to asynchronous execution of its application, and
I found some cases where race conditions could exist.
2. Check in ast_dial_join to be sure that the channel still exists before attempting to lock it, since
it could have gotten hung up but the is_running_app flag on the ast_dial_channel may not have been
cleared yet.
Russell Bryant [Wed, 27 Feb 2008 17:33:04 +0000 (17:33 +0000)]
Fix a problem in ChanSpy where it could get stuck in an infinite loop without
being able to detect that the calling channel hung up.
(closes issue #12076, reported by junky, patched by me)
Joshua Colp [Wed, 27 Feb 2008 17:07:33 +0000 (17:07 +0000)]
Use the lock (which already existed, it just wasn't used) on the updaters list to protect the contents instead of the overall module list lock.
(closes issue #12080)
Reported by: ChaseVenters
Kevin P. Fleming [Wed, 27 Feb 2008 16:53:06 +0000 (16:53 +0000)]
fallback to standard English prompts properly when using new prompt directory layout
(closes issue #11831)
Reported by: IgorG
Patches:
fallbacken.v1.diff uploaded by IgorG (license 20) (modified by me to improve code and conform rest of function to coding guidelines)
Russell Bryant [Wed, 27 Feb 2008 16:45:00 +0000 (16:45 +0000)]
When we receive a known alarm, make sure that the unknown alarm flag is not still
set to make sure that when we come back out of alarm, it gets reported in the log
and manager interface (after discussion with tzafrir on the -dev list)
Russell Bryant [Wed, 27 Feb 2008 00:54:29 +0000 (00:54 +0000)]
Zaptel 1.4 now exposes FXO battery state as an alarm. However, Asterisk 1.4
does not know what to do with these alarms. Only Asterisk 1.6 cares about it.
So, if we get an unknown alarm in chan_zap, don't generate confusing log messages
about it.
Jason Parker [Tue, 26 Feb 2008 16:08:44 +0000 (16:08 +0000)]
Fix a very bizarre issue we were seeing with our buildbot when using a DESTDIR that
wasn't an absolute path (such as DESTDIR=~/asterisk-1.4).
Apparently what was happening, was that some of the targets were being expanded to
the full path, so $@ ended up being /root/asterisk-1.4/[...]/ rather than ~/asterisk-1.4/[...]/
It appears that this may be a new "feature" in GNU make.
(*cough* http://en.wikipedia.org/wiki/Principle_of_least_surprise *cough*)
Russell Bryant [Tue, 26 Feb 2008 00:25:29 +0000 (00:25 +0000)]
Merge changes from team/russell/smdi-1.4
This commit brings in a significant set of changes to the SMDI support in Asterisk.
There were a number of bugs in the current implementation, most notably being that
it was very likely on busy systems to pop off the wrong message from the SMDI message
queue. So, this set of changes fixes the issues discovered as well as introducing
some new ways to use the SMDI support which are required to avoid the bugs with
grabbing the wrong message off of the queue.
This code introduces a new interface to SMDI, with two dialplan functions. First,
you get an SMDI message in the dialplan using SMDI_MSG_RETRIEVE() and then you access
details in the message using the SMDI_MSG() function. A side benefit of this is that
it now supports more than just chan_zap.
For example, with this implementation, you can have some FXO lines being terminated
on a SIP gateway, but the SMDI link in Asterisk.
Another issue with the current implementation is that it is quite common that the
station ID that comes in on the SMDI link is not necessarily the same as the Asterisk
voicemail box. There are now additional directives in the smdi.conf configuration
file which let you map SMDI station IDs to Asterisk voicemail boxes.
Yet another issue with the current SMDI support was related to MWI reporting over
the SMDI link. The current code could only report a MWI change when the change
was made by someone calling into voicemail. If the change was made by some other
entity (such as with IMAP storage, or with a web interface of some kind), then the
MWI change would never be sent. The SMDI module can now poll for MWI changes if
configured to do so.
This work was inspired by and primarily done for the University of Pennsylvania.
Russell Bryant [Mon, 25 Feb 2008 23:42:42 +0000 (23:42 +0000)]
This patch fixes some pretty significant problems with how app_chanspy handles
pointers to channels that are being spied upon. It was very likely that a
crash would occur if the channel being spied upon hung up. This was because
the current ast_channel handling _requires_ that the object is locked or else
it could disappear at any time (except in the owning channel thread). So, this
patch uses some channel datastore magic on the spied upon channel to be able to
detect if and when the channel goes away.
(closes issue #11877)
(patch written by me, but thanks to kpfleming for the idea, and to file for review)
Russell Bryant [Mon, 25 Feb 2008 23:19:05 +0000 (23:19 +0000)]
Improve the lock tracking code a bit so that a bunch of old locks that threads
failed to lock don't sit around in the history. When a lock is first locked,
this checks to see if the last lock in the list was one that was failed to be
locked. If it is, then that was a lock that we're no longer sitting in a trylock
loop trying to lock, so just remove it.
(inspired by issue #11712)
Joshua Colp [Mon, 25 Feb 2008 21:37:20 +0000 (21:37 +0000)]
Make it so a users.conf user creates both a SIP peer and a SIP user. The user will be used for inbound authentication for the device, and peer will be used for placing calls to the device.
(closes issue #9044)
Reported by: queuetue
Patches:
sip-gui-friend.diff uploaded by qwell (license 4)
Jason Parker [Mon, 25 Feb 2008 20:49:42 +0000 (20:49 +0000)]
Allow the use of #include and #exec in situations where the max include depth was only 1.
Specifically, this fixes using #include and #exec in extconfig.conf.
This was basically caused because the config file itself raises the include level to 1.
I opted not to raise the include limit, because recursion here could cause very bizarre behavior.
Joshua Colp [Mon, 25 Feb 2008 16:16:13 +0000 (16:16 +0000)]
If a resubscription comes in for a dialog we no longer know about tell the remote side that the dialog does not exist so they subscribe again using a new dialog.
(closes issue #10727)
Reported by: s0l4rb03
Patches:
10727-2.diff uploaded by file (license 11)
Joshua Colp [Mon, 25 Feb 2008 15:17:18 +0000 (15:17 +0000)]
Due to recent changes tag will no longer be NULL if not present so we have to use ast_strlen_zero to see if it's actually blank.
(closes issue #12061)
Reported by: flefoll
Patches:
chan_sip.c.br14.patch_pedantic_no_totag uploaded by flefoll (license 244)
Mark Michelson [Thu, 21 Feb 2008 20:12:38 +0000 (20:12 +0000)]
Remove an incorrect debug message. It reported that it had received a specific event and tried to report
which event was received. What actually was happening was that it was reporting the number of bytes returned
from a call to read().
Thanks to Jared Smith for bringing the issue up on IRC
Mark Michelson [Wed, 20 Feb 2008 22:32:22 +0000 (22:32 +0000)]
Clear up confusion when viewing the QUEUE_WAITING_COUNT of a
"dead" realtime queue. Since from the user's perspective, the queue
does exist, we shouldn't tell them we couldn't find the queue. Instead
since it is a dead queue, report a 0 waiting count
Joshua Colp [Wed, 20 Feb 2008 22:06:59 +0000 (22:06 +0000)]
Don't wait for additional digits when overlap dialing is enabled if the setup message contains the sending_complete information element.
(closes issue #11785)
Reported by: klaus3000
Patches:
sending_complete_overlap_asterisk-1.4.17.patch.txt uploaded by klaus3000 (license 65)
Joshua Colp [Mon, 18 Feb 2008 23:56:48 +0000 (23:56 +0000)]
Ensure that emulated DTMFs do not get interrupted by another begin frame.
(closes issue #11740)
Reported by: gserra
Patches:
v1-11740.patch uploaded by dimas (license 88)
(closes issue #11955)
Reported by: tsearle
(closes issue #10530)
Reported by: xmarksthespot
Mark Michelson [Mon, 18 Feb 2008 20:52:09 +0000 (20:52 +0000)]
There was an invalid assumption when calculating the duration of a file that the filestream in question
was created properly. Unfortunately this led to a segfault in the situation where an unknown format was
specified in voicemail.conf and a voicemail was recorded. Now, we first check to be sure that the stream
was written correctly or else assume a zero duration.
Tilghman Lesher [Mon, 18 Feb 2008 17:31:52 +0000 (17:31 +0000)]
When a SIP channel is being auto-destroyed, it's possible for it to still be
in bridge code. When that happens, we crash. Delay the RTP destruction until
the bridge is ended.
(closes issue #11960)
Reported by: norman
Patches:
20080215__bug11960__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: norman
Russell Bryant [Fri, 15 Feb 2008 19:50:11 +0000 (19:50 +0000)]
In the case that you try to directly reload a module has returned
AST_MODULE_LOAD_DECLINE, log a message indicating that the module is not fully
initialized and must be initialized using "module load".
Russell Bryant [Fri, 15 Feb 2008 18:33:29 +0000 (18:33 +0000)]
Don't attempt to execute the reload callback for a module that returned
AST_MODULE_LOAD_DECLINE. This fixes a crash that was reported against
chan_console in trunk.
(closes issue #11953, reported by junky, fixed by me)
Mark Michelson [Fri, 15 Feb 2008 17:26:37 +0000 (17:26 +0000)]
Final round of changes for configure script logic for IMAP
Now if a directory is specified, then we will search that directory for
a source installation of the IMAP toolkit. If none is found, then we will
use that directory as the basis for detecting a package installation of
the IMAP c-client. If that check fails, then configure will fail.
Mark Michelson [Fri, 15 Feb 2008 15:05:49 +0000 (15:05 +0000)]
Fix a bit of wrong logic in the configure script that caused problems when trying to configure
without IMAP. Patch suggestion from phsultan, but I modified it slightly.
Mark Michelson [Fri, 15 Feb 2008 00:50:49 +0000 (00:50 +0000)]
I apparently misunderstood one of the requirements of this configure change.
Now, if a source directory is specified with the --with-imap option, and a valid
source installation is not detected there, then configure will fail and will not
check for a package installation.
Mark Michelson [Thu, 14 Feb 2008 23:30:17 +0000 (23:30 +0000)]
Change to the configure logic regarding IMAP. Prior to this commit, if you wished to configure
Asterisk with IMAP support, you would use the --with-imap configure switch in one of the following
two ways:
--with-imap=/some/directory would look in the directory specified for a UW IMAP source installation
--with-imap would assume that you had imap-2004g installed in .. relative to the Asterisk source
With this set of changes the two above options still work the same, but there are two new behaviors, too.
--with-imap=system will assume that you have -libc-client.so where you store your shared objects and will
attempt to find c-client headers in your include path either in the imap or c-client directory.
If either of the two original methods of specifying the imap option should fail, then the check for --with-imap
=system will be performed in addition. It is only after this "system" check that failure can happen.
projects / thirdparty / asterisk.git / log
