Ensure that we can get an address even when we don't have a default route.
(closes issue #9225)
Reported by: junky
Patches:
20080122__bug9225.diff.txt uploaded by Corydon76 (license 14)
Tested by: oej, loloski, sergee
Fixing an issue wherein monitoring local channels was not possible. During a channel
masquerade, the monitors on the two channels involved are swapped. In 99% of the cases
this results in the desired effect. However, if monitoring a local channel, this caused
the monitor which was on the local channel to get moved onto a channel which is immediately
hung up after the masquerade has completed. By swapping the monitors prior to the masquerade,
we avoid the problem by tricking the masquerade into placing the monitor back onto the channel
where we want it.
During the investigation of the issue, the channel's monitor was the only thing that was swapped
in such a manner which did not make sense to have done. All other variable swapping made sense.
Permit the user to specify number of seconds that a connection may remain idle,
which fixes a crash on reconnect with the MyODBC driver.
(closes issue #11798)
Reported by: Corydon76
Patches:
20080119__res_odbc__idlecheck.diff.txt uploaded by Corydon76 (license 14)
Tested by: mvanbaak
Russell Bryant [Sun, 20 Jan 2008 07:28:23 +0000 (07:28 +0000)]
Add a "console active" CLI command, which lets you find out which console device
is currently active for the Asterisk CLI, or to set it. Also, knock multiple device
support off of the to-do list.
Russell Bryant [Sun, 20 Jan 2008 06:11:49 +0000 (06:11 +0000)]
Merge changes from team/russell/console_devices
- Add support for multiple devices. All devices are configured in console.conf.
- Add "console list devices" CLI command to show configured devices. Also, changed
the old "list devices" to be "list available", which queries PortAudio for all
audio devices that are available for use.
Russell Bryant [Sat, 19 Jan 2008 10:06:02 +0000 (10:06 +0000)]
Merged revisions 99187 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r99187 | russell | 2008-01-19 04:05:27 -0600 (Sat, 19 Jan 2008) | 4 lines
Fix a couple of memory leaks with frame handling. Specifically,
ast_frame_free() needed to be called on the frame that came from the translator
to signed linear.
Russell Bryant [Fri, 18 Jan 2008 22:04:33 +0000 (22:04 +0000)]
Merge changes from team/group/sip-tcptls
This set of changes introduces TCP and TLS support for chan_sip. There are various
new options in configs/sip.conf.sample that are used to enable these features. Also,
there is a document, doc/siptls.txt that describes some things in more detail.
This code was implemented by Brett Bryant and James Golovich. It was reviewed
by Joshua Colp and myself. A number of other people participated in the testing
of this code, but since it was done outside of the bug tracker, I do not have their
names. If you were one of them, thanks a lot for the help!
(closes issue #4903, but with completely different code that what exists there.)
Russell Bryant [Fri, 18 Jan 2008 21:38:01 +0000 (21:38 +0000)]
Merged revisions 99081 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r99081 | russell | 2008-01-18 15:37:21 -0600 (Fri, 18 Jan 2008) | 9 lines
Revert adding the packed attribute, as it really doesn't make sense why that
would do any good. Fix the real bug, which is to do the check to see if the
frame came from a translator at the beginning of ast_frame_free(), instead of
at the end. This ensures that it always gets checked, even if none of the
parts of the frame are malloc'd, and also ensures that we aren't looking at
free'd memory in the case that it is a malloc'd frame.
(closes issue #11792, reported by explidous, patched by me)
Russell Bryant [Fri, 18 Jan 2008 21:24:05 +0000 (21:24 +0000)]
Merged revisions 99079 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r99079 | russell | 2008-01-18 15:22:21 -0600 (Fri, 18 Jan 2008) | 4 lines
Since we're relying on the offset between the frame and the beginning of the translator
pvt struct, set the packed attribute to make sure we get to the right place.
(potential fix for issue #11792)
Terry Wilson [Fri, 18 Jan 2008 16:58:50 +0000 (16:58 +0000)]
This should at least temporarily fix a problem where the 't' Dial
option is incorrectly passed to the transferee when built-in
attended transfers are used. There is still a problem with 'T',
but better to fix some problems than no problems while we work
on it.
Tilghman Lesher [Fri, 18 Jan 2008 06:52:18 +0000 (06:52 +0000)]
Permit username and password to be NULL (which enables pass-through from the layer above).
Reported by: lurcher
Patch by: tilghman
(Closes issue #11739)
Russell Bryant [Thu, 17 Jan 2008 22:50:13 +0000 (22:50 +0000)]
Merged revisions 99004 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r99004 | russell | 2008-01-17 16:37:22 -0600 (Thu, 17 Jan 2008) | 10 lines
Have IAX2 optimize the codec translation path just like chan_sip does it. If
the caller's codec is in our codec list, move it to the top to avoid transcoding.
Mark Michelson [Thu, 17 Jan 2008 16:26:41 +0000 (16:26 +0000)]
Get the device state of the state interface instead of the interface when creating a new queue member.
Thanks to Atis Lezdins for bringing this up on the Asterisk-Dev mailing list.
Terry Wilson [Thu, 17 Jan 2008 03:09:32 +0000 (03:09 +0000)]
Update res_phoneprov to default to setting the SERVER variable to the IP
the HTTP request for the config came in on and the SERVER_PORT to the
bindport setting in sip.conf. I've left in the ability to override these
options, because I can't always guess how someone might decide to do something
weird with what is available to them--although needing to is pretty unlikely.
Documentation was updated to reflect preference for not setting serveraddr,
serveriface, or serverport. Tested on Linux and OS X.
Tilghman Lesher [Thu, 17 Jan 2008 00:13:32 +0000 (00:13 +0000)]
Change the way the new filter feature works, by allowing it to be a column NOT
logged into the database. This will allow more granularity of a decision
evaluated in the dialplan, then takes effect when posting the CDR.
Russell Bryant [Thu, 17 Jan 2008 00:05:13 +0000 (00:05 +0000)]
Add support for an easy way to automatically execute some Asterisk CLI commands
immediately at startup. Any commands in the startup_commands file in the Asterisk
config diretory will get executed.
(closes issue #11781)
Reported by: jamesgolovich
Patches:
asterisk-startupcmds.diff.txt uploaded by jamesgolovich (license 176)
-- With some changes by me.
Russell Bryant [Wed, 16 Jan 2008 22:36:47 +0000 (22:36 +0000)]
Blocked revisions 98982 via svnmerge
........
r98982 | russell | 2008-01-16 16:36:24 -0600 (Wed, 16 Jan 2008) | 5 lines
Add an unused pointer to the ast_channel struct. This makes the ast_channel structure
retain the same size as it had in previous 1.4 releases. Also, all of the offsets for
members in the structure are still the same (except for the two pointers that got replaced
for the new spy/whisper architecture.)
Russell Bryant [Wed, 16 Jan 2008 21:53:10 +0000 (21:53 +0000)]
Merge the changes from issue #10665 from the team/group/sip_session_timers branch.
This set of changes introduces SIP session timers support (RFC 4028). In short,
this prevents stuck SIP sessions that were not properly torn down due to network
or endpoint failures during an established SIP session.
To quote some of the documentation supplied with the patch:
"The SIP Session-Timers is an extension of the SIP protocol that allows end-points and proxies to
refresh a session periodically. The sessions are kept alive by sending a RE-INVITE or UPDATE
request at a negotiated interval. If a session refresh fails then all the entities that support Session-
Timers clear their internal session state. In addition, UAs generate a BYE request in order to clear
the state in the proxies and the remote UA (this is done for the benefit of SIP entities in the path
that do not support Session-Timers)."
(closes issue #10665)
Reported by: rjain
Patches:
chan_sip.c.1.diff uploaded by rjain (license 226)
chan_sip.c.diff uploaded by rjain (license 226)
sip.conf.sample.diff uploaded by rjain (license 226)
proc_422_rsp_comment.diff uploaded by rjain (license 226)
chan_sip.c.cache.diff uploaded by rjain (license 226)
chan_sip.memalloc uploaded by rjain (license 226)
chan_sip.memalloc.bugfix uploaded by rjain (license 226)
Patches tracked in team/group/sip_session_timers, with some additional fixes
by russell and oej.
Fix a deadlock in chan_local in local_hangup. There was contention because
the local_pvt was held and it was attempting to lock a channel, which is the
incorrect locking order.
Introduce a lock into the dialing API that protects it when destroying the structure.
(closes issue #11687)
Reported by: callguy
Patches:
11687.diff uploaded by file (license 11)
Don't drop the old record route information when dealing with packets related to a reinvite.
(closes issue #11545)
Reported by: kebl0155
Patches:
reinvite-patch.txt uploaded by kebl0155 (license 356)
Joshua Colp [Wed, 16 Jan 2008 02:30:13 +0000 (02:30 +0000)]
Remove DNS lookup from sip_devicestate. This seems to come from way back when and I can't think of a reason for it being here, plus it could cause needless DNS lookups.
(closes issue #10983)
Reported by: jtodd
Add autoconf logic for speexdsp. Later versions use a separate library for some things so we need to use it if present in codec_speex.
(closes issue #11693)
Reported by: yzg
Russell Bryant [Tue, 15 Jan 2008 23:53:28 +0000 (23:53 +0000)]
Merged revisions 98946 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r98946 | russell | 2008-01-15 17:50:10 -0600 (Tue, 15 Jan 2008) | 11 lines
Change a buffer in check_auth() to be a thread local dynamically allocated
buffer, instead of a massive buffer on the stack. This fixes a crash reported
by Qwell due to running out of stack space when building with LOW_MEMORY defined.
On a very related note, the usage of BUFSIZ in various places in chan_sip is
arbitrary and careless. BUFSIZ is a system specific define. On my machine,
it is 8192, but by definition (according to google) could be as small as 256.
So, this buffer in check_auth was 16 kB. We don't even support SIP messages
larger than 4 kB! Further usage of this define should be avoided, unless it
is used in the proper context.
Russell Bryant [Tue, 15 Jan 2008 23:31:53 +0000 (23:31 +0000)]
Merged revisions 98943 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r98943 | russell | 2008-01-15 17:26:52 -0600 (Tue, 15 Jan 2008) | 25 lines
Commit a fix for some memory access errors pointed out by the valgrind2.txt
output on issue #11698.
The issue here is that it is possible for an instance of a translator to get
destroyed while the frame allocated as a part of the translator is still being
processed. Specifically, this is possible anywhere between a call to ast_read()
and ast_frame_free(), which is _a lot_ of places in the code. The reason this
happens is that the channel might get masqueraded during this time. During a
masquerade, existing translation paths get destroyed.
So, this patch fixes the issue in an API and ABI compatible way. (This one is
for you, paravoid!)
It changes an int in ast_frame to be used as flag bits. The 1 bit is still used
to indicate that the frame contains timing information. Also, a second flag has
been added to indicate that the frame came from a translator. When a frame with
this flag gets released and has this flag, a function is called in translate.c to
let it know that this frame is doing being processed. At this point, the flag gets
cleared. Also, if the translator was requested to be destroyed while its internal
frame still had this flag set, its destruction has been deffered until it finds out
that the frame is no longer being processed.
Admittedly, this feels like a hack. But, it does fix the issue, and I was not able
to think of a better solution ...
Mark Michelson [Mon, 14 Jan 2008 22:11:50 +0000 (22:11 +0000)]
Big improvement for app_directory. This patch breaks the do_directory function up
so that it is more easily parsed by the human brain. It also fixes some errors. I'll quote
dimas from the original bug description:
"app_directory contained some duplicate code even before addition of 'm' option. Addition of that option doubled amount of that code. Worst of all, there are minor differences between these code block and bugs caused by these differences.
1. There is a memory leak. In the 'menu' mode, result of the convert(pos) function is not freed while it should be.
2. In the 'menu' mode check for OPT_LISTBYFIRSTNAME flag ('f' option) is not negated as result, application works in the mode opposite to what user expect (checking last name when user wants the first nd vice versa).
3. select_item function plays message for user using res = func1() || func2() || func3()... construct. This construct loses the actual value returned by ast_waitstream() for example so at the end, res does not contain digit user dialed while listening to the message.
4. (also in 1.4) application announces entries from voicemail.conf/realtime separately from entries from users.conf. I see no reason why doing so instead of building combined list.
5. Alot of duplicated code as already mentioned."
This was tested by dimas and I (I tested under valgrind). A word of caution: any bug fixes that happen
in app_directory in 1.4 will almost certainly not merge cleanly into trunk as a result of this, but it is
well worth it.
Huge thanks to dimas for this wonderful submission.
Jason Parker [Mon, 14 Jan 2008 17:40:36 +0000 (17:40 +0000)]
Add proper call forwarding (all and busy) support for chan_skinny.
Note: NoAnswer support is currently not implemented, as it would take a
significant amount of work to figure out how to do correctly.
Closes issue #11310, patches, testing, and support by DEA, mvanbaak, and myself.
Adding explicit defaults for missing options to init_queue. This is necessary because
if a user either removes or comments one of these options and reloads their queues, the
option will not reset to its default, instead maintaining the value from prior to the
reload.
Thanks to John Bigelow for pointing this error out to me.
Joshua Colp [Mon, 14 Jan 2008 15:07:30 +0000 (15:07 +0000)]
Print out a warning when spaces are used in the variable name in Set and MSet. It is extremely hard to debug this issue so this should make it easier.
(closes issue #11759)
Reported by: caio1982
Patches:
setvar_space_warning1.diff uploaded by caio1982 (license 22)
Russell Bryant [Mon, 14 Jan 2008 04:53:08 +0000 (04:53 +0000)]
Add another small option for the JACK app and JACK_HOOK function. The 'n'
option tells JACK not to start jackd automatically if it is not already
running. Otherwise, the default is that jackd will get started for you if
it isn't running already.
Russell Bryant [Sun, 13 Jan 2008 19:19:57 +0000 (19:19 +0000)]
Bring in the code from team/russell/jack/.
Add a new module, app_jack, which provides interfaces to JACK, the Jack
Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are
provided; there is a JACK() application, and a JACK_HOOK() function. Both
interfaces create an input and output JACK port. The application makes
these ports the endpoint of the call. The audio coming from the channel
goes out the output port and whatever comes back in on the input port is
what gets sent to the channel. The JACK_HOOK() function turns on a JACK
audiohook on the channel. This lets you run the audio coming from a
channel through JACK, and whatever comes back in is what gets forwarded
on as the channel's audio. This is very useful for building custom
vocoders or doing recording or analysis of the channel's audio in another
application.
In case anyone is curious, the platform that inspired me to write this is
PureData (http://puredata.info/). I wrote these JACK interfaces so that I
could use Pd to do interesting things with the audio of phone calls ...
Russell Bryant [Sun, 13 Jan 2008 00:10:00 +0000 (00:10 +0000)]
Remove a duplicate lock of the audiohook lock when destroying manipulate
audiohooks. This causes an error when we attempt to destroy the lock later
when freeing the audiohook.
Add a connection timeout attribute, as that was what was intended with the
login timeout, but ODBC divides it up into 2 different timeouts.
(Closes issue #11745)
projects / thirdparty / asterisk.git / log
