INFO, NOTIFY, OPTIONS, REFER, and MESSAGE requests
were not checking the topmost Via to determine where
to send the response. Adding check_via calls to those
request handlers solves this.
Tilghman Lesher [Thu, 8 Jan 2009 21:32:45 +0000 (21:32 +0000)]
Textual changes, consistency in status variable naming, and other minor bugs.
(closes issue #13943)
Reported by: Marquis
Patches:
minivm_trunk_fixes3.patch uploaded by Marquis (license 32)
Mark Michelson [Thu, 8 Jan 2009 19:48:42 +0000 (19:48 +0000)]
Add the average talk time for a queue
This patch adds the functionality to app_queue of calculating
the average amount of time that channels are bridged for a
queue. The algorithm used to calculate the average is the same
exponential average currently used to calculate the average holdtime.
See the CHANGES file to see the methods you may use to view this
information.
When a SIP request or response arrives for a dialog with an associated Asterisk channel, and the lock on that channel cannot be obtained because it is held by another thread, instead of dropping the request/response, queue it for later processing when the channel lock becomes available.
BJ Weschke [Wed, 7 Jan 2009 18:20:31 +0000 (18:20 +0000)]
Answer the channel if it has not already been answered and we've already found a valid profile for followme.
(closes issue #14140)
Reported by: dimas
Patches:
14140.patch uploaded by dimas
Leif Madsen [Wed, 7 Jan 2009 18:18:45 +0000 (18:18 +0000)]
Update queues.conf.sample documentation.
Update the queues.conf.sample documentation to mention that you need to preload chan_local.so as well if you plan on using Local channels for queue members, and you're preloading pbx_config.so.
Doug Bailey [Wed, 7 Jan 2009 17:05:37 +0000 (17:05 +0000)]
Cleanup fsk spill if off hook is detected during mwi spill.
Correct logic error in handling events when sending mwi spill
(closes issue #14143)
Reported by: alecdavis
Patches:
chan_dahdi.handle_init_event2.diff.txt uploaded by dbailey
Leif Madsen [Wed, 7 Jan 2009 14:26:19 +0000 (14:26 +0000)]
Update the sip-friends.sql file to use the non-deprecated 'defaultname' instead of 'username' and remove an extra comma that would cause the script to fail as-is
A couple of changes to T.38 SDP attribute handling
There are some boolean attributes for T.38 such
as T38FaxFillBitRemoval, T38FaxTranscodingMMR, and
T38FaxTranscodingJBIG. By simply being present, we
should treat these as a "true" value. The current
code, however, was requiring a 1 or 0 as the value
of the attribute in order to parse it. This is due
to the fact that there are some T.38 endpoints and
gateways that also transmit this information
incorrectly. This patch follows the "be liberal in
what you accept and strict in what you send"
philosophy by accepting both the correctly- and
incorrectly-formatted attributes, but only sending
information as it is supposed to be sent.
It was also discovered that a particular type of
T.38 gateway sends some non-standard T.38 SDP
attributes. Instead of using T38FaxMaxDatagram
and T38MaxBitRate, it used T38MaxDatagram and
T38FaxMaxRate respectively. We now will properly
accept these attributes as well.
Note that there are a lot of patches cited in
the below commit message template. This is
because the person who submitted these patches is
an awesome person and wrote 1.4, 1.6.0, and 1.6.1
variants.
(closes issue #13976)
Reported by: linulin
Patches:
chan_sip.c.1.4-update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.0-update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.1-update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.4-relaxedT38_update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.0-relaxedT38_update1.diff uploaded by arcivanov (license 648)
chan_sip.c.1.6.1-relaxedT38_update1.diff uploaded by arcivanov (license 648)
Tested by: arcivanov
Tilghman Lesher [Mon, 5 Jan 2009 16:44:47 +0000 (16:44 +0000)]
More clearly explain that quote marks are no longer necessary.
(closes issue #13718)
Reported by: davidw
Patches:
20081020__bug13718.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage
Jeff Peeler [Sat, 3 Jan 2009 20:29:54 +0000 (20:29 +0000)]
When parsing environment variable ASTERISK_PROMPT, make sure to proceed to the next character when a non format specifier is used (no %). Otherwise, the while loop looking for the null byte will never exit.
Mark Michelson [Mon, 29 Dec 2008 18:04:52 +0000 (18:04 +0000)]
Update app_queue to deal with the removal of AST_PBX_KEEPALIVE
When placing a call to a queue which ran a gosub on the member's
channel, Asterisk would crash every time, stemming from the fact
that the member's channel was being hung up unexpectedly when the
Gosub completed. The necessary change was pretty much copied and
pasted from app_dial's similar changes made last week.
I also took the opportunity to change a LOG_DEBUG message in
app_dial to use ast_debug. I am guessing this was due to a direct
merge from 1.4 that was not corrected to use trunk's preferred
syntax.
Terry Wilson [Wed, 24 Dec 2008 15:10:42 +0000 (15:10 +0000)]
There is no section 22.2.2 in rfc 3261. I believe 26.2.2 is what was meant:
Note that in the SIPS URI scheme, transport is independent of TLS,
and thus "sips:alice@atlanta.com;transport=tcp" and
"sips:alice@atlanta.com;transport=sctp" are both valid (although
note that UDP is not a valid transport for SIPS). The use of
"transport=tls" has consequently been deprecated, partly because
it was specific to a single hop of the request. This is a change
since RFC 2543.
Steve Murphy [Tue, 23 Dec 2008 18:13:49 +0000 (18:13 +0000)]
Merged revisions 166093 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
In order to merge this 1.4 patch into trunk,
I had to resolve some conflicts and wait for
Russell to make some changes to res_agi.
I re-ran all the tests; 39 calls in all, and
made fairly careful notes and comparisons: I
don't want this to blow up some aspect of
asterisk; I completely removed the KEEPALIVE
from the pbx.h decls. The first 3 scenarios
involving feature park; feature xfer to 700;
hookflash park to Park() app call all behave
the same, don't appear to leave hung channels,
and no crashes.
These changes eliminate the need for (and use of)
the KEEPALIVE return code in res_features.c;
There are other places that use this result code
for similar purposes at a higher level, these appear
to be left alone in 1.4, but attacked in trunk.
The reason these changes are being made in 1.4, is
that parking ends a channel's life, in some situations,
and the code in the bridge (and some other places),
was not checking the result code properly, and dereferencing
the channel pointer, which could lead to memory corruption
and crashes.
Calling the masq_park function eliminates this danger
in higher levels.
A series of previous commits have replaced some parking calls
with masq_park, but this patch puts them ALL to rest,
(except one, purposely left alone because a masquerade
is done anyway), and gets rid of the code that tests
the KEEPALIVE result, and the NOHANGUP_PEER result codes.
While bug 13820 inspired this work, this patch does
not solve all the problems mentioned there.
I have tested this patch (again) to make sure I have
not introduced regressions.
Crashes that occurred when a parked party hung up
while the parking party was listening to the numbers
of the parking stall being assigned, is eliminated.
These are the cases where parking code may be activated:
1. Feature one touch (eg. *3)
2. Feature blind xfer to parking lot (eg ##700)
3. Run Park() app from dialplan (eg sip xfer to 700)
(eg. dahdi hookflash xfer to 700)
4. Run Park via manager.
The interesting testing cases for parking are:
I. A calls B, A parks B
a. B hangs up while A is getting the numbers announced.
b. B hangs up after A gets the announcement, but
before the parking time expires
c. B waits, time expires, A is redialed,
A answers, B and A are connected, after
which, B hangs up.
d. C picks up B while still in parking lot.
II. A calls B, B parks A
a. A hangs up while B is getting the numbers announced.
b. A hangs up after B gets the announcement, but
before the parking time expires
c. A waits, time expires, B is redialed,
B answers, A and B are connected, after
which, A hangs up.
d. C picks up A while still in parking lot.
Testing this throroughly involves acting all the permutations
of I and II, in situations 1,2,3, and 4.
Since I added a few more changes (ALL references to KEEPALIVE in the bridge
code eliimated (I missed one earlier), I retested
most of the above cases, and no crashes.
H-extension weirdness.
Current h-extension execution is not completely
correct for several of the cases.
For the case where A calls B, and A parks B, the
'h' exten is run on A's channel as soon as the park
is accomplished. This is expected behavior.
But when A calls B, and B parks A, this will be
current behavior:
After B parks A, B is hung up by the system, and
the 'h' (hangup) exten gets run, but the channel
mentioned will be a derivative of A's...
Thus, if A is DAHDI/1, and B is DAHDI/2,
the h-extension will be run on channel
Parked/DAHDI/1-1<ZOMBIE>, and the
start/answer/end info will be those
relating to Channel A.
And, in the case where A is reconnected to
B after the park time expires, when both parties
hang up after the joyful reunion, no h-exten
will be run at all.
In the case where C picks up A from the
parking lot, when either A or C hang up,
the h-exten will be run for the C channel.
CDR's are a separate issue, and not addressed
here.
As to WHY this strange behavior occurs,
the answer lies in the procedure followed
to accomplish handing over the channel
to the parking manager thread. This procedure
is called masquerading. In the process,
a duplicate copy of the channel is created,
and most of the active data is given to the
new copy. The original channel gets its name
changed to XXX<ZOMBIE> and keeps the PBX
information for the sake of the original
thread (preserving its role as a call
originator, if it had this role to begin
with), while the new channel is without
this info and becomes a call target (a
"peer").
In this case, the parking lot manager
thread is handed the new (masqueraded)
channel. It will not run an h-exten
on the channel if it hangs up while
in the parking lot. The h exten will
be run on the original channel instead,
in the original thread, after the bridge
completes.
See bug 13820 for our intentions as
to how to clean up the h exten behavior.
Fix a deadlock relating to channel locks and autoservice
It has been discovered that if a channel is locked prior
to a call to ast_autoservice_stop, then it is likely that
a deadlock will occur. The reason is that the call to
ast_autoservice_stop has a check built into it to be sure
that the thread running autoservice is not currently trying
to manipulate the channel we are about to pull out of
autoservice.
The autoservice thread, however, cannot advance beyond where
it currently is, though, because it is trying to acquire
the lock of the channel for which autoservice is attempting
to be stopped.
The gist of all this is that a channel MUST NOT be locked
when attempting to stop autoservice on the channel.
In this particular case, the channel was locked by a call
to ast_read. A call to ast_exists_extension led to autoservice
being started and stopped due to the existence of dialplan
switches.
It may be that there are future commits which handle the same
symptoms but in a different location, but based on my looks through
the code, it is very rare to see a construct such as this one.
Russell Bryant [Mon, 22 Dec 2008 17:09:36 +0000 (17:09 +0000)]
Introduce ast_careful_fwrite() and use in AMI to prevent partial writes.
This patch introduces a function to do careful writes on a file stream which
will handle timeouts and partial writes. It is currently used in AMI to
address the issue that has been reported. However, there are probably a few
other places where this could be used.
(closes issue #13546)
Reported by: srt
Tested by: russell
http://reviewboard.digium.com/r/104/
Joshua Colp [Mon, 22 Dec 2008 16:08:13 +0000 (16:08 +0000)]
Record the previous port in the temporary address structure so that the comparison does not treat the host as having changed even if it did not. This would have been uninitialized before and would have led to a baddddd port.
(closes issue #13628)
Reported by: pananix
Patches:
bug13628.patch uploaded by jpeeler (license 325)
Tested by: file, blitzrage
Mark Michelson [Mon, 22 Dec 2008 16:07:59 +0000 (16:07 +0000)]
Fix a file playback crash and explicitly initialize values in func_timeout.c
A crash was brought up on the bugtracker. The first run through valgrind
was full of legitimate complaints of uninitialized values in func_timeout when
setting a response timeout. These were fixed but the crash persisted.
A second run through showed the real problem. The reference counting used
for filestreams was incorrect because there were some missing increments
when a frame was read from a format module.
Russell Bryant [Mon, 22 Dec 2008 14:16:54 +0000 (14:16 +0000)]
Remove AST_PBX_KEEPALIVE usage from res_agi.
This patch removes the usage of AST_PBX_KEEPALIVE from res_agi. The only usage
was for the AGI command, "asyncagi break". This patch removes this feature.
Normally, a feature would not be removed like this. However, this code is
broken and usage of it will result in a memory leak.
Usage of this feature will make the AGI code return a result of
AST_PBX_KEEPALIVE. The PBX handler assumes that another thread has assumed
ownership of the channel. The channel thread will exit without destroying the
channel. Unfortunately, _no_ thread has ownership of the channel at this
point. There are a couple of serious problems here:
1) The only way to recover the caller is to issue a channel redirect. This
will work, but this will be done with a masquerade, and the old ast_channel
structure will be lost.
2) Until the channel redirect happens, there is no code servicing the channel.
That means nothing is reading audio or handling events coming from the
channel. This is very bad.
The recommended way to get this same "break" functionality is to issue the
redirect while the channel is still being handled by the AGI code. That way,
there will be no memory leak, and there will be no period of time that the
channel is not being serviced.
Mark Michelson [Fri, 19 Dec 2008 22:26:16 +0000 (22:26 +0000)]
Adding a new dialplan function AUDIOHOOK_INHERIT
This function is being added as a method to allow for
an audiohook to move to a new channel during a channel
masquerade. The most obvious use for such a facility is
for MixMonitor when a transfer is performed. Prior to
the addition of this functionality, if a channel
running MixMonitor was transferred by another party, then
the recording would stop once the transfer had completed.
By using AUDIOHOOK_INHERIT, you can make MixMonitor
continue recording the call even after the transfer
has completed.
It has also been determined that since this is seen
by most as a bug fix and is not an invasive change,
this functionality will also be backported to 1.4 and
merged into the 1.6.0 branches, even though they are
feature-frozen.
Replace a bunch of if defined checks for Zaptel/DAHDI through several new defines in dahdi_compat.h. This removes a lot of code duplication. Example from bug:
Russell Bryant [Fri, 19 Dec 2008 14:42:51 +0000 (14:42 +0000)]
Introduce commit message formatting guidelines.
This documents the recommended outline to use for commit message. It also
covers information on special tags that can be used in commit messages, as well
as the template functionality that is available on bugs.digium.com.
Russell Bryant [Thu, 18 Dec 2008 21:44:47 +0000 (21:44 +0000)]
Merged revisions 165796 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r165796 | russell | 2008-12-18 15:39:25 -0600 (Thu, 18 Dec 2008) | 11 lines
Make ast_carefulwrite() be more careful.
This patch handles some additional cases that could result in partial writes
to the file description. This was done to address complaints about partial
writes on AMI.
(issue #13546) (more changes needed to address potential problems in 1.6)
Reported by: srt
Tested by: russell
Review: http://reviewboard.digium.com/r/99/
Add mutexes around accesses to the IMAP library interface. This prevents
certain crashes, especially when shared mailboxes are used.
(closes issue #13653)
Reported by: howardwilkinson
Patches:
asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by howardwilkinson (license 590)
Tested by: jpeeler
........
Mark Michelson [Thu, 18 Dec 2008 19:34:33 +0000 (19:34 +0000)]
Fix crashes in res_odbc.
The variable "class" was being set NULL just prior to
being dereferenced in an ao2_link call. I have moved
the setting of the variable to NULL until after the
ao2_link call.
Russell Bryant [Thu, 18 Dec 2008 19:33:42 +0000 (19:33 +0000)]
Remove the need for AST_PBX_KEEPALIVE with the GoSub option from Dial.
This is part of an effort to completely remove AST_PBX_KEEPALIVE and other
similar return codes from the source. While this usage was perfectly safe,
there are others that are problematic. Since we know ahead of time that
we do not want to PBX to destroy the channel, the PBX API has been changed
so that information can be provided as an argument, instead, thus removing
the need for the KEEPALIVE return value.
Further changes to get rid of KEEPALIVE and related code is being done by
murf. There is a patch up for that on review 29.
Only care about a compatible codec for early bridging if we are actually bridging to another channel. If we are not we actually want to bring the audio back to us.
(closes issue #13545)
Reported by: davidw
........
Do not crash if we are not passed in a followme id.
(closes issue #14106)
Reported by: ys
Patches:
app_followme.c.2.diff uploaded by ys (license 281)
........
Remove duplicate code from the ast_str API. We now use __AST_STR_* to
access 'struct ast_str' members, but this must only be used inside the API implementation.
Russell Bryant [Thu, 18 Dec 2008 14:23:22 +0000 (14:23 +0000)]
Add a \todo note for app_originate.
Jared Smith suggested that we add a way to be able to set variables
and functions on the outbound channel. I think that it's a great idea, so I
have added it as a todo so that it gets done at some point.
Reverse the fix from issue #6176 and add proper handling for that issue.
(Closes issue #13962, closes issue #13363)
Fixed by myself (license 14)
........
Terry Wilson [Wed, 17 Dec 2008 19:55:10 +0000 (19:55 +0000)]
Polycom phones close the connection after reading a little bit of the firmware files, we should stop sending in that case. Also, make that case print out a debug statement instead of a scary WARNING.
This patch adds a new 'ignoresdpversion' option to sip.conf. When this is
enabled (either globally or for a specific peer), chan_sip will treat any SDP
data it receives as new data and update the media stream accordingly. By
default, Asterisk will only modify the media stream if the SDP session version
received is different from the current SDP session version. This option is
required to interoperate with devices that have non-standard SDP session
version implementations (observed by toc on the bug tracker with Microsoft OCS
which always uses 0 as the session version).
Mark Michelson [Wed, 17 Dec 2008 17:52:50 +0000 (17:52 +0000)]
Use the create_vm_state_from_user function in a place where
it was not being used before. Also, I've moved the urgent
folder check in messagecount() up a bit so that the flow is
a bit better.
This was something I noticed while taking a look at issue
#13973, although I don't think this is the underlying cause
of the issue.
Steve Murphy [Wed, 17 Dec 2008 05:04:56 +0000 (05:04 +0000)]
A possibly "horrible fix" for a "horribly broken"
situation.
As stuff shifts around in the asterisk code, the
miscellaneous inclusions from the standalone stuff
gets broken. There's no easy fix for this situation.
I made sure that everything in utils builds without
problem ***AND*** that aelparse runs the regressions
correctly with the following make menuselect options
both on and off:
I think from now on, I'm going to #undef
all these features in the various utils native
files; I guess I could do the same for the
copied-in files, surrounded by STANDALONE ifdef.
A standalone isn't going to care about threads,
mutexes, etc.
projects / thirdparty / asterisk.git / log
