Fix devicestate problems for "always-on" agent channels
A revision to chan_agent attempted to "inherit" the device
state of the underlying channel in order to report the
device state of an agent channel more accurately.
The problem with the logic here is that it makes no sense to
use this for always-on agents. If the agent is logged in, then
to the underlying channel, the agent will always appear to be
"in use," no matter if the agent is on a call or not. The reason
is that to the underlying channel, the channel is currently in use
on a call to the AgentLogin application.
The most common cause that I found for this issue to occur was for
a SIP channel to be the underlying channel type for an Agent channel.
If the SIP phone re-registers, then the registration will cause the
device state core to query the device state of the SIP channel. Since the
SIP channel is in use, the Agent channel would also inherit this status.
Once the agent channel was set to "in use" there was no way that the device
state could change on that channel unless the agent logged out.
The solution for this problem is a bit different in 1.4 than it is in the
other branches. In 1.4, there will be a one-line fix to make sure that only
callback agents will inherit device state from their underlying channel type.
For the other branches of Asterisk, since callback support has been removed, there
is also no need for device state inheritance in chan_agent, so I will simply be
removing it from the code.
In addition, the 1.4 source is getting a new comment to help the next person who
edits chan_agent.c. I'm adding a comment that a agent_pvt's loginchan field may be
used to determine if the agent is a callback agent or not.
Prevent a crash from occurring when a jitter buffer interpolated frame is
removed from a slinfactory
slinfactory used the "samples" field of an ast_frame in order to determine
the amount of data contained within the frame. In certain cases, such as
jitter buffer interpolated frames, the frame would have a non-zero value for
"samples" but have NULL "data"
This caused a problem when a memcpy call in ast_slinfactory_read would attempt
to access invalid memory. The solution in use here is to never feed frames into
the slinfactory if they have NULL "data"
Mark Michelson [Tue, 27 Jan 2009 19:30:54 +0000 (19:30 +0000)]
Fix queue crashes that would occur after the calling channel was masqueraded.
The data passed to the end_bridge_callback was assumed to be data which was
still stack'd. The problem was that with some call features, attended transfers
in particular, a new bridge thread is started once the feature completes, meaning
that when the end_bridge_callback is called, the end_bridge_callback_data was
invalid.
To fix this problem, there are two measures taken
1. Instead of pointing to stacked data, we now used heap-allocated data for
passing to the end_bridge_callback in app_queue
2. Since bridges can end multiple times on a single logical call, we wait until
the final bridge is broken to actually set any queue variables. This is accomplished
through reference-counting and the use of an end_bridge_callback_data_fixup function
in app_queue.c
Originally putnopvut implemented some changes in revision 142079 that according to the bug report seemed to have worked then, but somehow fails now.
I guess code, as humans, get old and forget stuff. Anyway, this bug caused CANCEL not to work with picky systems.
Thanks Fredrik for pointing out where the bug in the SIP messaging was.
Russell Bryant [Mon, 26 Jan 2009 21:32:52 +0000 (21:32 +0000)]
Blocked revisions 171452 via svnmerge
........
r171452 | russell | 2009-01-26 15:31:59 -0600 (Mon, 26 Jan 2009) | 13 lines
Resolve some synchronization issues in chan_iax2 scheduler handling.
The important changes here are related to the synchronization between threads
adding items into the scheduler and the scheduler handling thread. By adjusting
the lock and condition handling, we ensure that the scheduler thread sleeps no
longer and no less than it is supposed to. We also ensure that it does not
wake up more often than it has to.
There is no bug report associated with this. It is just something that I found
while putting scheduler thread handling into a reusable form (review 129).
Olle Johansson [Mon, 26 Jan 2009 15:51:00 +0000 (15:51 +0000)]
Continue to move variables into the sip_cfg structure to make them easier to handle in the future as a group of settings for a group of devices.
At some point, I want one sip_cfg per domain handled, so we can have "group" settings.
Olle Johansson [Mon, 26 Jan 2009 15:11:39 +0000 (15:11 +0000)]
Just moving around variable declarations so that we have all globals in the same place.
Default setting is set before we activate the channel or at reloads, not where we declare the variable.
Russell Bryant [Sun, 25 Jan 2009 02:49:30 +0000 (02:49 +0000)]
Change ARRAY_LEN() to be more C++ safe.
When the second part of this macro is written as 0[a] instead of a[0], it will
force a failure if the macro is used on a C++ object that overloads the []
operator.
When a channel is answered make sure any indications currently playing stop. Usually the phone would do this but if the channel was already answered then they are being generated by Asterisk and we darn well need to stop them.
(closes issue #14249)
Reported by: RadicAlish
........
When a call is forwarded stop any active indications. The new channel will provide an indication, if need be, itself.
(closes issue #14310)
Reported by: RadicAlish
........
Use the on hold flag to see if the call is on hold or not. It is possible that our address for them will still be valid even though they are on hold.
(closes issue #14295)
Reported by: klaus3000
........
Joshua Colp [Fri, 23 Jan 2009 17:32:26 +0000 (17:32 +0000)]
Reset the ast_str used for escape substitution. We need to do this since it is a thread local variable that may contain the value of a previous substitution.
(closes issue #14312)
Reported by: pj
Doug Bailey [Fri, 23 Jan 2009 16:35:30 +0000 (16:35 +0000)]
MWI messages included in CID spill was not being properly handled and prevented the call from being processed
(issue #14313)
Reported by: seandarcy
Tested by: dbailey
There was a subtle change in ast_do_masquerade which
resulted in failed attempts to pickup calls. The problem
was that the value of the AST_FLAG_OUTGOING flag was
copied from the clone to the original channel. In the case
of call pickup, this meant that the AST_FLAG_OUTGOING flag
ended up being cleared on the channel that was attempting
to execute the pickup.
Because this flag was not set, when ast_read came across
an answer frame, it ignored it. The result of this was that
the calling channel was never properly answered.
This fix changes the behavior in ast_do_masquerade to set
the flags on the original channel to the union of the flags
on the clone channel. This way, if the AST_FLAG_OUTGOING
flag is set on either of the two channels involved in the
masquerade, the resulting channel will have the flag set
as well.
Don't crash if RTCP is not enabled on an RTP structure but statistics are output.
(closes issue #14234)
Reported by: jcovert
Patches:
rtp.c.patch-1.6.0.3 uploaded by jcovert (license 551)
rtp.c.patch-svn-165599 uploaded by jcovert (license 551)
........
Allow global variables after substitution to be as long as other variables.
(closes issue #14263)
Reported by: markd
Patches:
20090120__bug14263.diff.txt uploaded by Corydon76 (license 14)
........
If we are unable to request a DAHDI pseudo channel and we are using the user introduction without review option make sure it gets unset so other code does not blindly assume a DAHDI pseudo channel exists.
(closes issue #14282)
Reported by: cheesegrits
........
Doug Bailey [Thu, 22 Jan 2009 15:49:24 +0000 (15:49 +0000)]
change VMWI to use new DAHDI_VMWI ioctl call.
Change configure script to detect the new ioctl call data structure.
(issue #14104)
Reported by: alecdavis
Patches:
mwiioctl_structure_asterisk.diff4.txt uploaded by dbailey (license )
Tested by: alecdavis, dbailey
Do a string comparison instead of pointer comparison since some people specify the context they are actually in as an argument to get around some funkiness.
(closes issue #14011)
Reported by: dveiga
Patches:
pbx.c.patch uploaded by dveiga (license 665)
........
Joshua Colp [Thu, 22 Jan 2009 15:01:54 +0000 (15:01 +0000)]
Clear the autoloop flag when parsing and setting the context/extension/priority to go back to. When the channel executes a PBX again we want it to start out at the point we explicitly say and at that point it will not yet be doing autoloop.
(closes issue #14304)
Reported by: jcovert
AST_RWLOCK_INIT_VALUE is always defined. What we really wanted to ask is
whether autoconf detected a static initializer value. This fixes rwlocks
on all such platforms (mainly, Mac OS X).
(closes issue #13767)
Reported by: jcovert
Patches:
20090121__bug13767.diff.txt uploaded by Corydon76 (license 14)
Tested by: jcovert, Corydon76
........
Read lock the contexts to maintain the locking order when we are notified that the state of a device has changed.
(closes issue #13839)
Reported by: mcallist
........
Mark Michelson [Wed, 21 Jan 2009 21:53:55 +0000 (21:53 +0000)]
Further fix some oddities in sip show users and sip show peers logic
ccesario on IRC pointed out that his sip peers were not displayed
properly when he would issue the command "sip show peers." The problem
was that the onlymatchonip field was used to determine if the endpoint
was a "peer" or "user." The tricky part is that a "friend" is supposed
to be treated as both a "user" and a "peer" but the logic would not allow
"friends" to show up as "peers" since onlymatchonip was set to FALSE
for friends.
I have modified the sip_peer structure to more explicitly keep track of
what type endpoint it is so that the various manager and CLI commands
will display the expected information
Extra NULLs in the output cause some terminal types to abort in the middle of
a color code, causing terminal weirdness.
(closes issue #14130)
Reported by: coolmig
Patches:
20090121__bug14130.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, coolmig
........
Mark Michelson [Wed, 21 Jan 2009 00:33:32 +0000 (00:33 +0000)]
Fix device state parsing issues for channel names with multiple slashes
The fix being applied is a bit different for trunk and the 1.6.X branches.
For trunk, we only wish to strip off the characters beyond the second slash
if the channel is a Local channel (i.e. we are removing the /n from the device
name). Other channel technologies with multiple slashes (e.g. DAHDI) need the
information after the second slash in order to get the proper device state
information.
In addition to this fix, the 1.6.X branches are receiving a much more important
fix as well. The problem in 1.6.X is that the member's device name was being directly
changed instead of having a copy changed. This meant that we would strip off the
second slash and trailing characters and then leave the member's device name like
that permanently thereafter.
Mark Michelson [Tue, 20 Jan 2009 20:10:31 +0000 (20:10 +0000)]
Convert the character pointers in a sip_request to be pointer offsets
When an ast_str expands to hold more data, any pointers that were pointing
to the data prior to the expansion will be pointing at invalid memory. This
change makes such pointers used in chan_sip.c instead be offsets from the
beginning of the string so that the same math may be applied no matter where
in memory the string resides.
To help ease this transition, a macro called REQ_OFFSET_TO_STR has been added
to chan_sip.c so that given a sip_request and an offset, the string at that
offset is returned.
Terry Wilson [Tue, 20 Jan 2009 19:22:24 +0000 (19:22 +0000)]
Make a proper builtin attended transfer to parking work
This is an ugly hack from 1.4 that allows the timeout callback from a parked
call to use the right channel name for the callback when the park is done with
a builtin attended transfer (that isn't completed early). This hasn't ever
worked in trunk and no one has complained yet, so eh.
Kevin P. Fleming [Mon, 19 Jan 2009 21:42:46 +0000 (21:42 +0000)]
ast_str_SQLGetData is *not* part of the ast_str API, it's part of the ast_odbc API and just happens to use an ast_str as the buffer; move all of it to res_odbc.c and res_odbc.h, renaming appropriately
along the way fix some minor coding style issues in strings.h and add some attribute_pure annotations to functions in the ast_str API
Michiel van Baak [Mon, 19 Jan 2009 20:09:11 +0000 (20:09 +0000)]
Redo the event-based MWI in chan_skinny.
Dan saw regular segfaults with the old implementation and
rewrote it to make it really eventbased.
I altered it to be trunk compatible and wedhorn gave some feedback
and ideas how to make it even better.
(closes issue #13821)
Reported by: DEA
Patches:
chan_skinny-mwi-events.txt uploaded by DEA (license 3)
Tested by: mvanbaak, DEA
Prevent a crash in chan_local due to a potential NULL pointer dereference
Move the check for if both channels on a local_pvt have generators to below
where p->chan is checked for NULLity (NULLness?). This prevents a crash from
occurring if p->chan is NULL.
Doug Bailey [Sat, 17 Jan 2009 18:26:44 +0000 (18:26 +0000)]
Add discriminator for when ring pulse alert signal is used to preface MWI spills
This prevents the situation when MWI messages are added to caller ID spills causing the channel to be hung up
Account for possible NULL pointer when we receive a 408 in response to a REGISTER
It may be that by the time we receive a reply to a REGISTER request, the attempt has
timed out and thus the registry structure pointed to by the corresponding sip_pvt has
gone away. This situation was handled properly for a 200 OK response, but the 408
case assumed that the sip_registry struct was non-NULL, thus potentially causing a crash
This commit fixes this assumption and prints out a message to the console if we should
receive a late 408 response to a REGISTER
Convert call to park_call_full to masq_park_call_announce
Since we removed the AST_PBX_KEEPALIVE return value, we need to use masqueraded
parking, otherwise we will try to call ast_hangup() in __pbx_run() and in
do_parking_thread() and then promptly crash.
(closes issue #14215)
Reported by: waverly360
Tested by: otherwiseguy
(closes issue #14228)
Reported by: kobaz
Tested by: otherwiseguy
........
The reason was that the timing module was written under the assumption
that timerfd_timer_set_rate would not be called between enabling and
disabling continuous mode. What happened in this situation was that
timerfd_timer_enable_continuous saved off our previously set timer (in this
situation a 0 timer, meaning it never runs out). Then timerfd_timer_disable_continuous
would restore this 0 timer, even though it logically should set the timer to be whatever
was set in timerfd_timer_set_rate.
Now the behavior in timerfd_timer_set_rate is to overwrite the saved timer that may
or may not have been set in timerfd_timer_enable_continuous. Even if
timerfd_timer_enable_continuous has not been previously called, this will not harm the
operation.
Thanks to Terry Wilson for discovering the problem and giving me a really great debug
capture that pointed out the problem clearly
Fix the conjugation of Russian and Ukrainian languages.
(related to issue #12475)
Reported by: chappell
Patches:
vm_multilang.patch uploaded by chappell (license 8)
........
This patch fixes a problem where a goto (or jump, in this case)
fails a consistency check because it can't find a matching
extension. The problem was a missing instruction to end
the range notation in the code where it converts the pattern
into a regex and uses the regex code to determine the match.
I tested using the AEL code the user supplied, and now,
the consistency check passes.
Kevin P. Fleming [Thu, 15 Jan 2009 20:18:53 +0000 (20:18 +0000)]
remove the PBX_ODBC logic from the configure script, and add GENERIC_ODCB logic that includes copying the relevant LIB and INCLUDE data from either UnixODBC or iODBC, based on which was found; if both were found, prefer UnixODBC
this stops modules from being linked against both sets of libraries on systems that have both installed
Mark Michelson [Thu, 15 Jan 2009 19:00:06 +0000 (19:00 +0000)]
Remove an unneeded condition for line addition to a SIP request/response
In Asterisk 1.4 and 1.6.0, the sip_request structure had a statically
allocated buffer to hold the text of the request. There was a check in the
add_line function to not attempt to write the line into the buffer if we
did not have room for it.
In trunk and Asterisk versions starting with 1.6.1, an expandable ast_str
structure is used to hold the text. Since it may grow to fit an arbitrarily
sized string, this check in add_line is no longer valid.
I found this oddity while attempting to fix issue #14220; however, I do not
believe that this is the fix for that issue since the output supplied by the
reporter did not contain the warning message that would be printed had this
condition been satisfied.
Olle Johansson [Thu, 15 Jan 2009 18:08:59 +0000 (18:08 +0000)]
Make sure that we have the same terminology in sip.conf.sample and the source code warning.
Thanks Nick Lewis for pointing this out in the bug tracker.
Sean Bright [Thu, 15 Jan 2009 15:33:18 +0000 (15:33 +0000)]
Add a missing unlock and properly handle the 'maxusers' setting on MeetMe
conferences. We were using the 'user number' field to compare against the
maximum allowed users, which works assuming users with lower user numbers
didn't leave the conference.
Fix some crashes from bad datastore handling in app_queue.c
* The queue_transfer_fixup function was searching for and removing
the datastore from the incorrect channel, so this was fixed.
* Most datastore operations regarding the queue_transfer datastore
were being done without the channel locked, so proper channel locking
was added, too.
* Fixed create_process() allocation of process ID values.
The allocated process IDs could overflow their respective
NT and TE fields. Affects outgoing calls.
........
Doug Bailey [Wed, 14 Jan 2009 21:19:45 +0000 (21:19 +0000)]
This fixes a problem where MWI FSK spills were being injected onto off hook fxs lines.
(closes issue #14143)
Reported by: alecdavis
Patches:
chan_dahdi-14143.patch.txt uploaded by dbailey (license )
Tested by: alecdavis
projects / thirdparty / asterisk.git / log
