Joshua Colp [Thu, 12 Mar 2009 16:29:19 +0000 (16:29 +0000)]
Fix issue with streaming MOH failing if nobody is listening.
When a music class is setup to actually provide music on hold
from a stream we need to constantly read audio from it since it
will constantly be providing audio. This is now done despite there
being no channels listening to it.
Russell Bryant [Wed, 11 Mar 2009 21:42:58 +0000 (21:42 +0000)]
Make code that updates BRIDGEPEER variable thread-safe.
It is not safe to read the name field of an ast_channel without the channel
locked. This patch fixes some places in channel.c where this was being done,
and lead to crashes related to masquerades.
David Vossel [Wed, 11 Mar 2009 17:25:31 +0000 (17:25 +0000)]
encrypted IAX2 during packet loss causes decryption to fail on retransmitted frames
If an iax channel is encrypted, and a retransmit frame is sent, that packet's iseqno is updated while it is encrypted. This causes the entire frame to be corrupted. When the corrupted frame is sent, the other side decrypts it and sends a VNAK back because the decrypted frame doesn't make any sense. When we get the VNAK, we look through the sent queue and send the same corrupted frame causing a loop. To fix this, encrypted frames requiring retransmission are decrypted, updated, then re-encrypted. Since key-rotation may change the key held by the pvt struct, the keys used for encryption/decryption are held within the iax_frame to guarantee they remain correct.
Joshua Colp [Wed, 11 Mar 2009 17:22:52 +0000 (17:22 +0000)]
Fix issue where an attended transfer could not be completed under a rare scenario.
When completing an attended transfer chan_sip does a check to make sure the extension
in the URI portion of the Refer-To header is a local valid extension. We don't actually
need to check this since we know for sure the other channel is already up and talking to
the extension. Some devices do not put the extension in the Refer-To header either, which
can cause the extension check to fail. We now no longer do this check if it is an attended
transfer.
Joshua Colp [Wed, 11 Mar 2009 16:36:50 +0000 (16:36 +0000)]
Fix a problem with inband DTMF detection on outgoing SIP calls when dtmfmode=auto.
When dtmfmode was set to auto the inband DTMF detector was not setup
on outgoing SIP calls. This caused inband DTMF detection to fail.
The inband DTMF detector is now setup for both dtmfmode inband and auto.
Jeff Peeler [Wed, 11 Mar 2009 03:25:04 +0000 (03:25 +0000)]
Fix malloc debug macros to work properly with h323.
The main problem here was that cstdlib was undefining free thereby causing the
proper debug macros to not be used. ast_h323.cxx has been changed to call
ast_free instead to avoid the issue. Because using the ast prefix calls are
a better choice, ast_free_ptr is the new wrapper for free to pass to functions.
Also, a little bit of clean up was done to avoid the debug macros intentionally
being redefined.
David Vossel [Fri, 6 Mar 2009 17:19:55 +0000 (17:19 +0000)]
Fix handling of backreferences for ENUM lookups
enum.c did not handle regex backtraces correctly. The '\1' in the regex is a backreference that requires a pattern match to be inserted. The way the code used to work is that it would find the backreference and insert the entire input string minus the '+'. This is incorrect. The regexec() function takes in a variable called pmatch which is an array of structs containing the start and end indexes for each backreference substring. The original code actually passed the pmatch array pointer into regexec but never did anything with it. Now when a backtrace is found, the backtrace number is looked up in the pmatch array and the correct substring is inserted.
Mark Michelson [Thu, 5 Mar 2009 23:26:11 +0000 (23:26 +0000)]
[IMAP] Fix message retrieval issues when identical mailbox names were defined in separate contexts.
There was a fix put in a while back so that an X-Asterisk-VM-Context message header was
added to stored IMAP voicemails. This would allow for us to differentiate if the same
mailbox name was used in multiple contexts. The problem still left was that not all places
where messages were retrieved actually attempted to use this header for information when
retrieving messages. This commit fixes that so that MWI and message retrieval from VoiceMailMain
work as expected.
Mark Michelson [Thu, 5 Mar 2009 18:58:48 +0000 (18:58 +0000)]
Fix broken mailbox parsing when searchcontexts option is enabled.
When using the searchcontexts option in voicemail.conf, the code
made the assumption that all mailbox names defined were unique across
all contexts. However, the code did nothing to actually enforce this
assumption, nor did it do anything to alert a user that he may have
created an ambiguity in his voicemail.conf file by defining the same
mailbox name in multiple contexts.
With this change, we now will issue a nice long warning if searchcontexts
is on and we encounter the same mailbox name in multiple contexts and ignore
any duplicates after the first box. Whether searchcontexts is enabled or not,
if we come across a duplicate mailbox in the same context, then we will issue
a warning and ignore the duplicated mailbox. I have also added a small note
to voicemail.conf.sample in the explanation for searchcontexts explaining
that you cannot define the same mailbox in multiple contexts if you have
enabled the option.
Fix problems when RTP packet frame size is changed
During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good.
This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes.
Mark Michelson [Tue, 3 Mar 2009 22:48:18 +0000 (22:48 +0000)]
Clarify some documentation of queues.conf.sample
It had always been possible to explicitly specify a "blank"
value for a sound file in queues.conf and have no sound played
back. The problem with this is that it would result in some ugly
CLI warnings from file.c.
This commit introduces a check when playing a file in app_queue
to see if the name of the file is zero-length and return early if
that is the case. Also, the ability to specify the blank sound
files in queues.conf is now mentioned more clearly in queues.conf.sample
Joshua Colp [Tue, 3 Mar 2009 18:27:09 +0000 (18:27 +0000)]
Do not assume that the bridge_cdr is still attached to the channel when the 'h' exten is finished executing.
It is possible for a masquerade operation to occur when the 'h' exten is operating. This operation moves
the CDR records around causing the bridge_cdr to no longer exist on the channel where it is expected to.
We can not safely modify it afterwards because of this, so don't even try.
Steve Murphy [Tue, 3 Mar 2009 18:11:34 +0000 (18:11 +0000)]
These changes allow AEL to better check ${} constructs within $[...], that are concatenated with text.
I modified and added rules in ast_expr2.fl to better handle
the concatenations.
I added some default routines to ast_expr2.y so the standalone would
compile. It also looks like I haven't run this thru bison since 2.1, so
it's good to get this updated.
The Makefile has comments added now for check_expr2 and check_expr to
explain what they are for, and how to run them.
The testexpr2s stuff has been removed, in favor of check_expr2.
expr2.testinput has been updated to include the two expressions
that inspired these changes (from mcnobody on #asterisk this morning)
The regression has been run and all looks well.
Russell Bryant [Tue, 3 Mar 2009 16:45:46 +0000 (16:45 +0000)]
Ensure chan->fdno always gets reset to -1 after handling a channel fd event.
Since setting fdno to -1 had to be moved, a couple of other code paths that
do process an fd event return early and do not pass through the code path
where it was moved to. So, set it to -1 in a few other places, too.
Joshua Colp [Tue, 3 Mar 2009 14:38:09 +0000 (14:38 +0000)]
Move where fdno is set to the default value to *after* the read callback of the channel driver is called.
We have to do this as the underlying channel driver may need the fdno value to determine what to read.
Russell Bryant [Tue, 3 Mar 2009 13:53:52 +0000 (13:53 +0000)]
Make it easier to detect an improper call to ast_read().
When you call ast_waitfor() on a channel, the index into the channel fds array
that holds the file descriptor that poll() determines has input available is
stored in fdno. This patch clears out this value after a call to ast_read()
and also reports errors if ast_read() is called without an fdno set.
Jeff Peeler [Mon, 2 Mar 2009 23:54:39 +0000 (23:54 +0000)]
Fix bridging regression from commit 176701
This fixes a bad regression where the bridge would exit after an attended
transfer was made. The problem was due to nexteventts getting set after the
masquerade which caused the bridge to return AST_BRIDGE_COMPLETE.
Russell Bryant [Mon, 2 Mar 2009 23:34:13 +0000 (23:34 +0000)]
Move ast_waitfor() down to avoid the results of the API call becoming stale.
This call to ast_waitfor() was being done way too soon in this section of code.
Specifically, there was code in between the call to waitfor and the code that
uses the result that puts the channel in autoservice. By putting the channel
in autoservice, the previous results of ast_waitfor() become meaningless,
as the autoservice thread will do it's own ast_waitfor() and ast_read()
on the channel.
So, when we came back out of autoservice and eventually hit the block of code
that calls ast_read() on the channel, there may not actually be any input on
the channel available. Even though the previous call to ast_waitfor() in
app_meetme said there was input, the autoservice thread has since serviced
the channel for some period of time.
This bug manifested itself while dvossel was doing some testing of MeetMe in
Asterisk trunk. He was using the timerfd timing module. When the code hit
ast_read() erroneously, it determined that it must have been called because of
input on the timer fd, as chan->fdno was set to AST_TIMING_FD, since that was
the cause of the last legitimate call to ast_read() done by autoservice.
In this test, an IAX2 channel was calling into the MeetMe conference. It was
_much_ more likely to be seen with an IAX2 channel because of the way audio
is handled. Every audio frame that comes in results in a call to
ast_queue_frame(), which then uses ast_timer_enable_continuous() to notify
the channel thread that a frame is waiting to be handled. So, the chances
of ast_waitfor() indicating that a channel needs servicing due to a timer
event on an IAX2 event is very high.
Finally, it is interesting to note that if a different timing interface was
being used, this bug would probably not be noticed. When ast_read() is called
and erroneously thinks that there is a timer event to handle, it calls the
ast_timer_ack() function. The pthread and dahdi timing modules handle the
ack() function being called when there is no event by simply ignoring it.
In the case of the timerfd module, it results in a read() on the timer fd
that will block forever, as there is no data to read. This caused Asterisk
to lock up very quickly.
Thanks to dvossel and mmichelson for the fun debugging session. :-)
Tilghman Lesher [Mon, 2 Mar 2009 23:09:01 +0000 (23:09 +0000)]
When ending a recording with silence detection, remember to reduce the duration.
The end of the recording is correspondingly trimmed, but the duration was not
trimmed by the number of seconds trimmed, so the saved duration was necessarily
longer than the actual soundfile duration.
(closes issue #14406)
Reported by: sasargen
Patches:
20090226__bug14406.diff.txt uploaded by tilghman (license 14)
Tested by: sasargen
Russell Bryant [Mon, 2 Mar 2009 22:58:18 +0000 (22:58 +0000)]
Ensure that only one thread is calling ast_settimeout() on a channel at a time.
For example, with an IAX2 channel, you can have both the channel thread and the
chan_iax2 processing threads calling this function, and doing so twice at the
same time is a bad thing.
(Found in a debugging session with dvossel and mmichelson)
Steve Murphy [Thu, 26 Feb 2009 21:27:32 +0000 (21:27 +0000)]
This change moves the default feature digit timeout to 1000 ms from the previous default of 500.
As per bug 14515, a dev discussion arrived at a "mediated concensus"
of a default feature digit timeout of 1.0 sec. Some voted for 1300;
ctooley thought 1500 for distracted phone users in phone booths;
kpfleming put his foot down at 1.0 sec.
Users who found the previous default max delay of 250 msec perfect,
are welcome to override the new default. Notice that I said that
250 msec was the default; wait a minute, you might say, the config
file said it was 500 msec!; well, because of the bug fix for 14515,
we found that 500 msec was actually enforcing a max of 250. The bug
fix would restore 500 msec, but we felt even that was a bit tight
for most users... 2000 msec was pushed earlier by mmichelson, so
that reduces to 1000 msec after the bug fix. Enjoy!
David Vossel [Thu, 26 Feb 2009 17:24:02 +0000 (17:24 +0000)]
IAX2 prune realtime fix
Now prune_users() and prune_peers() are called instead of reload_config() to prune all users/peers that are realtime. These functions remove all users/peers with the rtfriend and delme flags set. iax2_prune_realtime() also lacked the code to properly delete a single friend. For example. if iax2 prune realtime <friend> was called, only the peer instance would be removed. The user would still remain.
Steve Murphy [Thu, 26 Feb 2009 17:09:03 +0000 (17:09 +0000)]
This patch prevents the feature detection timeout from being cut in half.
Because the ast_channel_bridge() call will return 0 and pass
a frame pointer for both DTMF_BEGIN and DTMF_END, the feature_timer
field in hte config struct is getting decremented twice, which
effectively cuts the digittimeout in half. I added conditions
to the if statement to only let DTMF_END frames to flow thru,
which solved the problem. Also, when the frame pointer is null,
let control flow thru-- this usually happens on timeouts. I added
a comment to the code to explain what's going on and why.
Many thanks to sodom for reporting this problem. Personnally, it always seemed
like something was wrong with the featuredigittimeout, but I never
could quite decide what... and was too busy to investigate.
This bug forced the issue, and now we know.
Sodom had other issues in 14515, but I couldn't reproduce them. If
he still has problems, and wants to get them solved, he is welcome
to reopen 14515.
Steve Murphy [Wed, 25 Feb 2009 21:00:50 +0000 (21:00 +0000)]
This patch completes the fixes nec. to make 1.4 asterisk dialplan expressions ($[...]) 8-bit transparent
While I was updating ast_expr2.fl, I missed one rule that would allow 8-bit chars to be caught
in tokens; and in so doing, it absorbs the ${ sequence and messes up the
checking of raw exprs by AEL.
Tilghman Lesher [Tue, 24 Feb 2009 23:25:24 +0000 (23:25 +0000)]
Add section about the #exec command in configuration files.
(closes issue #14540)
Reported by: jtodd
Patch by: jtodd, with additional notes by tilghman (license 14)
Joshua Colp [Tue, 24 Feb 2009 15:16:07 +0000 (15:16 +0000)]
Skip check for extension when subscribing for MWI.
Since the remote side is not actually subscribing to a specific extension when
subscribing for MWI just skip the check to see if the extension exists. They can't use it
to specify the mailbox either since we require configuration of that in sip.conf
Russell Bryant [Mon, 23 Feb 2009 23:09:01 +0000 (23:09 +0000)]
Fix infinite DTMF when a BEGIN is received without an END.
This commit is related to rev 175124 of 1.4 where a previous attempt was made
to fix this problem. The problem with the previous patch was that the inserted
code needed to go _before_ setting the lastrxts to the current timestamp.
Because those were the same, the dtmfcount variable was never decremented, and
so the END was never sent.
In passing, I removed the dtmfsamples variable which was completed unused. I
also removed a redundant setting of the lastrxts variable.
Tilghman Lesher [Fri, 20 Feb 2009 21:15:01 +0000 (21:15 +0000)]
This exception does not appear to still be true for Solaris 10, and OpenSolaris definitely needs it to be removed.
Fixed for snuff-home on -dev channel.
David Vossel [Fri, 20 Feb 2009 20:17:37 +0000 (20:17 +0000)]
Fixes issue with undefined audio codecs in chan_iax2
During iax2 call negotiation, supported codecs are passed in an Information Element containing a 2 byte field where each bit correlates to a specific codec. In 1.4 only audio codec bits 0-12 are defined, leaving bits 13-15 undefined. By default all bits are enabled unless specified otherwise. Since its a 2 byte field and 13-15 are not defined, these bits are never turned off. In trunk, bits 13-15 are defined, which means 1.4 is advertising support for codecs it does not have when talking to trunk. I fixed this by adding #define for undefined audio codec bits. These bits are then removed from iax2's full bandwidth capabilities.
Tilghman Lesher [Thu, 19 Feb 2009 22:26:01 +0000 (22:26 +0000)]
Fix up potential crashes, by reducing the sharing between interactive and non-interactive threads.
(closes issue #14253)
Reported by: Skavin
Patches:
20090219__bug14253.diff.txt uploaded by Corydon76 (license 14)
Tested by: Skavin
Steve Murphy [Wed, 18 Feb 2009 22:43:14 +0000 (22:43 +0000)]
This patch fixes a regression of sorts that was introduced in
rev 24425.
It basically fixes AST-190/ABE-1782.
What was wrong: the user has 6000 extensions in one context; and
then 6000 contexts, one per extension. The parser could only handle
about 4893 of the 6000 extens in the single context.
This was due to the regression I mentioned. To get rid of
shift/reduce conflicts, Luigi set up right-recursive lists
for globals, context elements, switch lists, and statements.
Right recursive lists got rid of the warnings, but instead, they
use up a tremendous amount of stack space when the lists are long.
I saw this a few years back, and resolved not to fix it until
someone complained. That day has arrived!
After the changes were made, I ran the regression test suite,
and there were no problems.
I took the test case the user provided, and added 100,000
extensions to the single context, that already had 6,000 extens
in it. (I'll see your 6, and raise you 100!) It takes a few minutes
to read it all in, check it and generate code for it, but no
problems.
So, I think I can say that fundamentally, there are no longer
any limits on the number of items you can place in contexts,
statement blocks, switches, or globals, beyond your virt mem
constraints.
Jeff Peeler [Wed, 18 Feb 2009 20:06:45 +0000 (20:06 +0000)]
Modify h323 to build against PTLib as well as the older PWLib
Several changes in PTLib have occurred requiring build time detection. Changes
accounted for include the library name change, config option change, install
location change, and a boolean type change which is handled by ast_ptlib.h.
Also, the sed check has been modified to properly work with autoconf >= 2.62.
Shaun Ruffell [Wed, 18 Feb 2009 00:34:38 +0000 (00:34 +0000)]
Several changes to codec_dahdi to play nice with G723.
This commit brings in the changes that were living out on the
svn/asterisk/team/sruffell/asterisk-1.4-transcoder branch. codec_dahdi.c now
always uses signed linear as the simple codec so that a soft g729 codec will
not end up being preferred to the hardware codec. There are also changes to
allow codec_dahdi.c to feed packets to the hardware in the native sample size of
the codec. This solves problems with choppy audio when using G723.
Jeff Peeler [Tue, 17 Feb 2009 21:54:34 +0000 (21:54 +0000)]
Modify bridging to properly evaluate DTMF after first warning is played
The main problem is currently if the Dial flag L is used with a warning sound,
DTMF is not evaluated after the first warning sound. To fix this, a flag has
been added in ast_generic_bridge for playing the warning which ensures that if
a scheduled warning is missed, multiple warrnings are not played back (due to a
feature evaluation or waiting for digits). ast_channel_bridge was modified to
store the nexteventts in the ast_bridge_config structure as that information
was lost every time ast_channel_bridge was reentered, causing a hangup due to
incorrect time calculations.
(closes issue #14315)
Reported by: tim_ringenbach
Reviewed on reviewboard:
http://reviewboard.digium.com/r/163/
Tilghman Lesher [Tue, 17 Feb 2009 21:21:41 +0000 (21:21 +0000)]
Backport change to 1.4:
Prior to masquerade, move the group definitions to the channel performing the
masq, so that the group count lingers past the bridge.
(closes issue #14275)
Reported by: kowalma
Patches:
20090216__bug14275.diff.txt uploaded by Corydon76 (license 14)
Tested by: kowalma
Tilghman Lesher [Tue, 17 Feb 2009 00:49:22 +0000 (00:49 +0000)]
After a 'sip reload', qualifies for realtime peers weren't immediately
restarted, instead waiting until the next registration. We're now
caching the qualify across a reload/restart and starting the qualify
immediately upon loading the peer.
(closes issue #14196)
Reported by: pdf
Patches:
20090120__bug14196_1.4.diff.txt uploaded by pdf (license 663)
Tested by: pdf
David Vossel [Mon, 16 Feb 2009 23:30:52 +0000 (23:30 +0000)]
Fixes issue with AST_CONTROL_SRCUPDATE not being relayed correctly during bridging
This should have been committed with rev176247, but I missed it. srcupdate frames no longer break out of the native bridge, but are not being sent to the other call leg either. This fixs that.
Kevin P. Fleming [Mon, 16 Feb 2009 21:41:46 +0000 (21:41 +0000)]
correct a logic error in the last stringfields commit... don't mark additional space as allocated if the string was built using already-allocated space
Mark Michelson [Mon, 16 Feb 2009 21:34:27 +0000 (21:34 +0000)]
Open the DAHDI pseudo device and set it to be nonblocking atomically
Apparently on FreeBSD, attempting to set the O_NONBLOCKING flag separately
from opening the file was causing an "inappropriate ioctl for device" error.
While I cannot fathom why this would be happening, I certainly am not opposed
to making the code a bit more compact/efficient if it also fixes a bug.
David Vossel [Mon, 16 Feb 2009 21:28:20 +0000 (21:28 +0000)]
Fixes issue with AST_CONTROL_SRCUPDATE breaking out of native bridge
In iax2, when a AST_CONTROL_SRCUPDATE is received it breaks from the native bridge, but since there is no code path to handle srcupdate it just goes to be beginning of the loop. This was causing packet storms of srcupdate frames between servers. Now srcupdate frames do not break the native bridge for processing.
Kevin P. Fleming [Mon, 16 Feb 2009 21:10:38 +0000 (21:10 +0000)]
fix a flaw in the ast_string_field_build() family of API calls; these functions made no attempt to reuse the space already allocated to a field, so every time the field was written it would allocate new space, leading to what appeared to be a memory leak.
Joshua Colp [Mon, 16 Feb 2009 15:33:53 +0000 (15:33 +0000)]
Don't have the Via header stored as a stringfield as it can change often during the lifetime of a dialog.
This issue crept up with subscriptions on the AA50. When an outgoing NOTIFY is sent a new branch value
is created and the Via header is changed to reflect it. Since this was a stringfield a new spot in the
pool was used for the value while the old was left untouched/unused. If the current pool was full a new
pool was created. This would cause memory usage to increase steadily.
Mark Michelson [Fri, 13 Feb 2009 19:47:48 +0000 (19:47 +0000)]
Fix a potential crash situation when using IMAP voicemail
If calling into VoiceMailMain when using IMAP storage, it was
possible to crash Asterisk by hanging up the phone when prompted
for a voicemail mailbox. This patch fixes the issue.
While it may appear that this patch is superficial, it allows code
execution to continue to the failure case just below the IMAP_STORAGE
code block where this patch has been applied
Tilghman Lesher [Thu, 12 Feb 2009 21:19:40 +0000 (21:19 +0000)]
Fix crashes when receiving certain T.38 packets. Also, increase the maximum
size of T.38 packets and warn users when they try to set the limits above those
maximums.
(closes issue #13050)
Reported by: schern
Patches:
20090212__bug13050.diff.txt uploaded by Corydon76 (license 14)
Tested by: schern
Jeff Peeler [Thu, 12 Feb 2009 20:34:36 +0000 (20:34 +0000)]
Fix ParkedCall event information for From field in the case of a blind transfer
If the parker information can not be obtained from the peer, try and see if
the BLINDTRANSFER channel variable has been set. Previously, a blind transfer
to the ParkAndAnnounce app would return nothing for the From.
Russell Bryant [Thu, 12 Feb 2009 16:51:13 +0000 (16:51 +0000)]
Don't send DTMF for infinite time if we do not receive an END event.
I thought that this was going to end up being a pretty gnarly fix, but it turns
out that there was actually already a configuration option in rtp.conf,
dtmftimeout, that was intended to handle this situation. However, in between
Asterisk 1.2 and Asterisk 1.4, the code that processed the option got lost.
So, this commit brings it back to life.
The default timeout is 3 seconds. However, it is worth noting that having
this be configurable at all is not really the recommended behavior in RFC 2833.
From Section 3.5 of RFC 2833:
Limiting the time period of extending the tone is necessary
to avoid that a tone "gets stuck". Regardless of the
algorithm used, the tone SHOULD NOT be extended by more than
three packet interarrival times. A slight extension of tone
durations and shortening of pauses is generally harmless.
Three seconds will pretty much _always_ be far more than three packet
interarrival times. However, that behavior is not required, so I'm going to
leave it with our legacy behavior for now.
Philippe Sultan [Thu, 12 Feb 2009 10:16:21 +0000 (10:16 +0000)]
Set the initiator attribute to lowercase in our replies when receiving calls.
This attribute contains a JID that identifies the initiator of the GoogleTalk
voice session. The GoogleTalk client discards Asterisk's replies if the
initiator attribute contains uppercase characters.
Tilghman Lesher [Wed, 11 Feb 2009 20:54:18 +0000 (20:54 +0000)]
Restore a behavior that was recently changed, when we fixed issue #13962 and
issue #13363 (related to issue #6176). When a hangup occurs during a Macro
execution in earlier 1.4, the h extension would execute within the Macro
context, whereas it was always supposed to execute only within the main context
(where Macro was called). So this fix checks for an "h" extension in the
deepest macro context where a hangup occurred; if it exists, that "h" extension
executes, otherwise the main context "h" is executed.
(closes issue #14122)
Reported by: wetwired
Patches:
20090210__bug14122.diff.txt uploaded by Corydon76 (license 14)
Tested by: andrew
Joshua Colp [Tue, 10 Feb 2009 18:50:50 +0000 (18:50 +0000)]
Go off hold when we get an empty reinvite telling us to.
(closes issue #14448)
Reported by: frawd
Patches:
hold_invite_nosdp.patch uploaded by frawd (license 610)
Improve behavior of jitterbuffer when maxjitterbuffer is set.
This change improves the way the jitterbuffer handles maxjitterbuffer and
dramatically reduces the number of frames dropped when maxjitterbuffer is
exceeded. In the previous jitterbuffer, when maxjitterbuffer was exceeded, all
new frames were dropped until the jitterbuffer is empty. This change modifies
the code to only drop frames until maxjitterbuffer is no longer exceeded.
Also, previously when maxjitterbuffer was exceeded, dropped frames were not
tracked causing stats for dropped frames to be incorrect, this change also
addresses that problem.
Joshua Colp [Mon, 9 Feb 2009 14:48:21 +0000 (14:48 +0000)]
Don't overwrite our pointer to the music class when music on hold stops. We will use this if it starts again to see if we can resume the music where it left off.
(closes issue #14407)
Reported by: mostyn
check ast_strlen_zero() before calling ast_strdupa() in sip_uri_headers_cmp()
and sip_uri_params_cmp()
The reporter didn't actually upload a properly-formed patch, instead a
modified chan_sip.c file was uploaded. I created a patch to determine the
changes, then modified the suggested changes to create a proper fix. The
summary above is a complete description of the changes.
Joshua Colp [Fri, 6 Feb 2009 17:14:15 +0000 (17:14 +0000)]
Some clients do not put the call-id for replaces at the beginning, so support it being anywhere in the string.
(closes issue #14350)
Reported by: fhackenberger
Jeff Peeler [Thu, 5 Feb 2009 20:47:51 +0000 (20:47 +0000)]
Add new configuration option to make shared IMAP mailboxes function as expected.
The new option is "imapvmshareid" which is an ID to tag multiple mailboxes
using the same IMAP storage location to function as one mailbox. This allows
all messages to be retrieved for any user in the group. The patch alters the
'X-Asterisk-VM-Extension' header that is responsible for matching voicemails
for a given user.
Mark Michelson [Thu, 5 Feb 2009 17:34:33 +0000 (17:34 +0000)]
Fix a problem where a channel pointer becomes invalid due to masquerading or hanging up.
app_mixmonitor runs its own thread to monitor the channel's activity and write the mixed
audio to a file. Since this thread runs independently of the channel, it is possible that
the mixmonitor thread's channel pointer will point to freed memory when the channel either
is masqueraded or hangs up (technically, both cases are hangups, but we need to handle the
cases slightly differently).
The solution for this is to employ a datastore, which has the nice benefit of allowing us
to hook into channel masquerades and hangups and update our pointer as necessary. If this
looks familiar, this same technique is employed in app_chanspy. app_chanspy is a bit more
involved since it does a lot more operations on the channel that is being spied upon.
app_mixmonitor does have an extra touch that app_chanspy doesn't have, though. Since there
is a thread race between the channel's thread and the mixmonitor thread on a hangup, we em-
ploy a condition-and-boolean combination to ensure that the channel thread finishes with
our structure before the mixmonitor thread attempts to free it. No crashes!
David Vossel [Tue, 3 Feb 2009 23:35:55 +0000 (23:35 +0000)]
Fixes issue with IAX2 transfer not handing off calls.
Fixes issue with IAX2 transfers not taking place. As it was, a call that was being transfered would never be handed off correctly to the call ends because of how call numbers were stored in a hash table. The hash table, "iax_peercallno_pvt", storing all the current call numbers did not take into account the complications associated with transferring a call, so a separate hash table was required. This second hash table "iax_transfercallno_pvt" handles calls being transfered, once the call transfer is complete the call is removed from the transfer hash table and added to the peer hash table resuming normal operations. Addition functions were created to handle storing, removing, and comparing items in the iax_transfercallno_pvt table.
Jeff Peeler [Tue, 3 Feb 2009 21:57:01 +0000 (21:57 +0000)]
Parking attempts made to one end of a bridge no longer will hang up due to a
parking failure.
Parking attempts made using either one-touch, or doing either a blind or
assisted transfer to the parking extension now keep up the bridge instead of
hanging up the attempted parked party. Normal causes for the parking attempt
to fail includes the specific specified extension (via PARKINGEXTEN) not being
available or if all the parking spaces are currently in use. To avoid having
to reverse a masquerade park_space_reserve was made to provide foresight if
a parking attempt will succeed and if so reserve the parking space.
(closes issue #13494)
Reported by: mdu113
Reviewed by Russell: http://reviewboard.digium.com/r/133/
projects / thirdparty / asterisk.git / log
