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.
Tilghman Lesher [Wed, 13 Feb 2008 00:26:57 +0000 (00:26 +0000)]
Refuse to load app_voicemail if res_adsi is not loaded (which is a symbol dependency)
(closes issue #11760)
Reported by: non-poster
Patches:
20080114__bug11760.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, non-poster, jamesgolovich
Joshua Colp [Mon, 11 Feb 2008 22:09:07 +0000 (22:09 +0000)]
If entering a conference with the 'w' option ensure that we can't listen or speak until the marked user appears.
(closes issue #11835)
Reported by: alanmcmillan
Tilghman Lesher [Sat, 9 Feb 2008 06:23:49 +0000 (06:23 +0000)]
Commit fix for being unable to send voicemail from VoiceMailMain
Reported by: William F Acker (via the -users mailing list)
Patch by: Corydon76 (license 14)
ensure that components of chan_misdn.so are built using any special build options that the configure script generated (reported by Philipp Kempgen on asterisk-dev)
Tilghman Lesher [Mon, 4 Feb 2008 21:06:09 +0000 (21:06 +0000)]
Cross-platform fix: OS X now deprecates the use of the daemon(3) API.
(closes issue #11908)
Reported by: oej
Patches:
20080204__bug11908.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76
Russell Bryant [Fri, 1 Feb 2008 23:06:32 +0000 (23:06 +0000)]
Change the SDP_SAMPLE_RATE macro. It turns out that even though G.722 is 16 kHz,
it is supposed to specified as 8 kHz in the RTP, and RTP timestamps are supposed
to be calculated based on 8 kHz. (Apparently this is due to a bug in a spec, but
people follow it anyway, because it's the spec ...)
Tilghman Lesher [Fri, 1 Feb 2008 21:54:28 +0000 (21:54 +0000)]
Fix the VM_DUR variable for forwarded voicemail, and fixed several other bugs
while I'm in the area.
(closes issue #11615)
Reported by: jamessan
Patches:
20071226__bug11615__2.diff.txt uploaded by Corydon76 (license 14)
Tested by: Corydon76, jamessan
Jason Parker [Fri, 1 Feb 2008 17:41:40 +0000 (17:41 +0000)]
Remove a needless (and incorrect) call to feof() after fgets().
This would have exited the loop early if you had an authentication file with no newline at the end.
Russell Bryant [Fri, 1 Feb 2008 17:23:47 +0000 (17:23 +0000)]
Don't overwrite the last character of a line if it's not a newline. This would
happen if the last line in the file doesn't have a newline.
(pointed out by Qwell)
Russell Bryant [Fri, 1 Feb 2008 00:32:49 +0000 (00:32 +0000)]
Add some more sanity checking on IAX2 dial strings for the case that no peer
or hostname was provided, which is the one part of the dial string that is
absolutely required. If it's not there, bail out.
(closes issue #11897)
Reported by sokhapkin
Patch by me
Russell Bryant [Thu, 31 Jan 2008 23:10:06 +0000 (23:10 +0000)]
Fix a couple of places where ast_frfree() was not called on a frame that came
from a translator. This showed itself by g729 decoders not getting released.
Since the flag inside the translator frame never got unset by freeing the frame
to indicate it was no longer in use, the translators never got destroyed, and
thus the g729 licenses were not released.
(closes issue #11892)
Reported by: xrg
Patches:
11892.diff uploaded by russell (license 2)
Tested by: xrg, russell
Mark Michelson [Thu, 31 Jan 2008 21:00:24 +0000 (21:00 +0000)]
1. Prevent the addition of an extra '/' to the beginning of an absolute pathname.
2. If ast_monitor_change_fname is called and the new filename is the same as the old, then exit early and don't set the
filename_changed field in the monitor structure. Setting it in this case was causing ast_monitor_stop to erroneously
delete them.
Joshua Colp [Wed, 30 Jan 2008 15:41:04 +0000 (15:41 +0000)]
Fix an issue where if a frame of higher sample size preceeded a frame of lower sample size and ast_slinfactory_read was called with a sample size of the combined values or higher a crash would happen.
(closes issue #11878)
Reported by: stuarth
Jason Parker [Wed, 30 Jan 2008 15:34:37 +0000 (15:34 +0000)]
Change default config to use descending channel order of groups, rather than ascending.
Fixes a potential source of confusion in glare-type situations.
Mark Michelson [Wed, 30 Jan 2008 15:23:00 +0000 (15:23 +0000)]
Fix a logic error with regards to autofill. Prior to this change, it was possible
for a caller to go out of turn if autofill were enabled and callers ahead in the queue were attempting
to call a member. This change fixes this.
projects / thirdparty / asterisk.git / log
