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r81776 | file | 2007-09-06 16:40:37 -0300 (Thu, 06 Sep 2007) | 7 lines
(closes issue #10122)
Reported by: stevefeinstein
Patches:
meetme-unmute-manager.diff uploaded by qwell (license 4)
Tested by: stevefeinstein
After looking over the code I agree with Qwell. Setting the file descriptor to conference each time just causes a fight back and forth.
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This makes it so it doesn't. Thanks to file for pointing out where the problem was and showing
a similar function in app_dial as an example of how to fix it.
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r81416 | mmichelson | 2007-08-31 14:48:55 -0500 (Fri, 31 Aug 2007) | 6 lines
Fixed broken behavior of a reload on realtime queues. Prior to this patch, if a reload was issued and
a realtime queue had callers waiting in it, then the queue would be removed from the queue list, but it would
not actually be freed (in fact, a debug message warning about a memory leak would come up). With this patch,
reloads do not touch realtime queues at all.
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r81397 | mmichelson | 2007-08-30 17:05:56 -0500 (Thu, 30 Aug 2007) | 7 lines
Removing an extraneous (and possibly misleading) log message. Firstly, if the announce file isn't found, the
streaming functions will report it. Secondly, not all non-zero returns from play_file mean that the announce file
wasn't found. Positive return values simply mean that a digit was pressed (most likely to skip through the announcement).
(closes issue #10612, reported and patched by dimas)
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Reported by: junky
Patches:
minivm_output2.diff uploaded by junky (license 177)
Change console output of minivm show stats to be more simple for external parsing.
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r81349 | mmichelson | 2007-08-29 11:35:29 -0500 (Wed, 29 Aug 2007) | 12 lines
This patch, in essence, will correctly pause a realtime queue member and reflect those
changes in the realtime engine.
(issue #10424, reported by irroot, patch by me)
This patch creates a new function called update_realtime_member_field, which is a generic
function which will allow any one field of a realtime queue member to be updated. This patch
only uses this function to update the paused status of a queue member, but it lays the foundation
for persisting the state of a realtime member the same way that static members' state is maintained
when using the persistentmembers setting
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r81340 | mmichelson | 2007-08-29 10:52:42 -0500 (Wed, 29 Aug 2007) | 8 lines
This fix creates a more accurate way of detecting whether realtime members were deleted.
(closes issue 10541, reported by Alric, patched by me)
The REALLY nice things about this patch is that queue members now have a "realtime" field
which will be true if the member is a realtime member. This means we can check this value
prior to certain processing if it should ONLY be done for realtime members.
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r81158 | mmichelson | 2007-08-27 17:40:19 -0500 (Mon, 27 Aug 2007) | 5 lines
Resolve a potential deadlock. In this case, a single queue is locked, then the queue list. In changethread(), the queue list is
locked, and then each individual queue is locked. Under the right circumstances, this could deadlock. As such, I have unlocked
the individual queue before locking the queue list, and then locked the queue back after the queue list is unlocked.
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r79906 | mmichelson | 2007-08-17 14:14:05 -0500 (Fri, 17 Aug 2007) | 6 lines
Patch allows for more seamless transition from file storage voicemail to ODBC storage voicemail.
If a retrieval of a greeting from the database fails, but the file is found on the file system, then
we go ahead an insert the greeting into the database. The result of this is that people who
switch from file storage to ODBC storage do not need to rerecord their voicemail greetings.
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The way a device state change propagates is kind of silly, in my opinion. A
device state provider calls a function that indicates that the state of a
device has changed. Then, another thread goes back and calls a callback for
the device state provider to find out what the new state is before it can go
send it off to whoever cares.
I have changed it so that you can include the state that the device has changed
to in the first function call from the device state provider. This removes the
need to have to call the callback, which locks up critical containers to go find
out what the state changed to.
This change set changes the "simple" device state providers to use the new method.
This includes parking, meetme, and SLA.
I have also mostly converted chan_agent in my branch, but still have some more
things to think through before presenting the plan for converting channel drivers
to ensure all of the right events get generated ...
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r78907 | mmichelson | 2007-08-09 18:47:00 -0500 (Thu, 09 Aug 2007) | 4 lines
Improved a bit of logic regarding comma-separated mailboxes in has_voicemail. Also added some braces to some compound if statements
since unbraced if statements scare me in general.
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r78859 | mmichelson | 2007-08-09 16:51:17 -0500 (Thu, 09 Aug 2007) | 9 lines
Quite a few changes regarding IMAP storage.
1. instead of using inboxcount as the core message counting function, we use messagecount instead. This makes it possible to count messages in folders besides just INBOX and Old.
2. inboxcount and hasvoicemail now use messagecount as their means of determining return values.
3. Added a copy_message function for IMAP storage. Unfortunately I don't have the means to test it, but it seems like a pretty straightforward function.
4. Removed a #ifndef IMAP_STORAGE and matching #endif from leave_voicemail for a couple of reasons. One, we want to support copying mail to multiple IMAP boxes, and two, IMAP was
broken because a STORE macro had been moved into this section of code.
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r78749 | russell | 2007-08-09 12:24:40 -0500 (Thu, 09 Aug 2007) | 9 lines
Fix subscriptions to multiple mailboxes for ODBC_STORAGE. Also, leave a
comment for this to be fixed for IMAP_STORAGE, as well. I left IMAP alone
since I know MarkM was working on this code right now for another reason.
This is broken even worse in trunk, but for a different reason. The fact
that the mailbox option supported multiple mailboxes is completely not obvious
from the code in the channel drivers. Anyway, I will fix that in another
commit ...
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r78717 | russell | 2007-08-09 11:12:57 -0500 (Thu, 09 Aug 2007) | 7 lines
Fix a problem with the combination of the 'F' option to pass DTMF through a
conference and options that use DTMF to activate various features. The problem
was that the BEGIN frame would be passed through, but the END frame would get
intercepted to activate a feature. Then, the other conference members would hear
DTMF for forever, which they didn't seem to like very much.
(closes issue #10400, reported by stevefeinstein, fixed by me)
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r78575 | mmichelson | 2007-08-08 09:26:36 -0500 (Wed, 08 Aug 2007) | 4 lines
Changing a bit of logic so that someone will NEVER exit the queue on timeout unless they have enabled the 'n' option.
This commit relates to issue #10320. Thanks to jfitzgibbon for detailing the idea behind this code change.
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Reported by: yehavi
Use the filename we parsed using the standard parsing when launching the application specified to ExternalIVR.
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r78101 | russell | 2007-08-03 15:14:06 -0500 (Fri, 03 Aug 2007) | 10 lines
(closes issue #10194)
Reported by: blitzrage
Patches:
bug0010194 uploaded by vovochka
Tested by: blitzrage
Fix a problem when you call Voicemail() with multiple mailboxes specified and
ODBC_STORAGE is in use. The audio part of the message was only given to the
first mailbox specified.
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r77854 | mmichelson | 2007-08-01 09:08:57 -0500 (Wed, 01 Aug 2007) | 8 lines
Fixes an issue I introduced to queues wherein a queue with joinempty=yes would kick people out of the queue because of erroneously
thinking the 'n' option was in use.
(closes issue #10320, reported by jfitzgibbon, patched by me, tested by blitzrage and me)
Thank you blitzrage for all the testing you've done lately with queues! It's much appreciated!
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r77852 | mmichelson | 2007-08-01 08:59:59 -0500 (Wed, 01 Aug 2007) | 7 lines
If a queue uses dynamic realtime members, then the member list should be updated after each attempt to call the queue.
This fixes an issue where if a caller calls into a queue where no one is logged in, they would wait forever even if a member
logged in at some point.
(closes issue #10346, reported by and tested by blitzrage, patched by me)
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r77191 | murf | 2007-07-25 16:39:27 -0600 (Wed, 25 Jul 2007) | 1 line
This fix solves problem with intense squelch noise when someone joins conf in bug 9430; We repro'd the problem with meetme opts of 'CciMo'; Josh Colp supplied this patch, and I'm applying it. It looks like playing the recorded username will louse up the next thing played into the channel. Josh rearranged the code so as to start things over before playing data directly into the conference.
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r77176 | file | 2007-07-25 19:16:10 -0300 (Wed, 25 Jul 2007) | 4 lines
(closes issue #10303)
Reported by: jtodd
Add SPEECH_DTMF_TERMINATOR variable so the user can specify the digit to terminate a DTMF string with. If none is specified then no terminator will be used.
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r76801 | mmichelson | 2007-07-24 11:26:58 -0500 (Tue, 24 Jul 2007) | 13 lines
Added a membercount variable to call_queue struct which keeps track of the number of logged in members in a particular queue.
This makes it so that the 'n' option for Queue() can act properly depending on which strategy is used. If the strategy is
roundrobin, rrmemory, or ringall, we want to ring each phone once before moving on in the dialplan. However, if any other strategy is
used, we will only ring one phone since it cannot be guaranteed that a different phone will ring on subsequent attempts to ring a phone.
As a side effect of this, the QUEUE_MEMBER_COUNT dialplan function now just reads the membercount variable instead of traversing through
the member list to figure out how many members there are.
Special thanks to blitzrage for helping to test this out.
(closes issue #10127, reported by bcnit, patched by me, tested by blitzrage)
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r76708 | tilghman | 2007-07-23 17:38:06 -0500 (Mon, 23 Jul 2007) | 4 lines
It was our stated intention for 1.4 that files created in app_voicemail should
depend upon the umask. Unfortunately, mkstemp() creates files with mode 0600,
regardless of the umask. This corrects that deficiency.
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using old methods of parsing arguments to using the standard macros. However, the big
change is that the really old way of specifying application and arguments separated by
a comma will no longer work (e.g. NoOp,foo|bar). Instead, the way that has been
recommended since long before 1.0 will become the only method available (e.g. NoOp(foo,bar).
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r76139 | mmichelson | 2007-07-20 13:42:27 -0500 (Fri, 20 Jul 2007) | 6 lines
When using users.conf for the entries in the directory, if multiple users had the same last name, only the first user listed would be available
in the directory.
(closes issue #10200, reported by mrskippy, patched by me)
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r75969 | mmichelson | 2007-07-19 11:26:10 -0500 (Thu, 19 Jul 2007) | 10 lines
Changes in handling return values of several functions in app_queue. This all started as a fix for issue #10008
but now includes all of the following changes:
1. Simplifying the code to handle positive return values from ast API calls.
2. Removing the background_file function.
3. The fix for issue #10008
(closes issue #10008, reported and patched by dimas)
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(and/or the optimization level) may think it is used uninitialized.
The code was indeed correct, but unfortunately the result of
some compiler checks such as -Wunused and -Wuninitialized depends
heavily on the optimization level.
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r75253 | mmichelson | 2007-07-16 13:16:15 -0500 (Mon, 16 Jul 2007) | 8 lines
Restoring functionality from 1.2 wherein Retrydial will not exit if there is no announce file specified.
This change makes it so that if there is no announce file specified, the application will continue until finished (or caller hangs up).
If a bogus announce file is specified, then a warning message will be printed saying that the file could not be found, but execution will
still continue.
(closes issue #10186, reported by jon, patched by me)
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r75078 | mmichelson | 2007-07-13 15:15:30 -0500 (Fri, 13 Jul 2007) | 13 lines
Merged revisions 75066 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r75066 | mmichelson | 2007-07-13 15:10:39 -0500 (Fri, 13 Jul 2007) | 5 lines
Fixed an issue where chanspy flags were uninitialized if no options were passed.
What triggered this investigation was an IRC chat where some people's quiet flags were
set while others' weren't even though none of them had specified the q option.
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(closes issue #10158)
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r74428 | qwell | 2007-07-10 14:58:53 -0500 (Tue, 10 Jul 2007) | 14 lines
Merged revisions 74427 via svnmerge from
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r74427 | qwell | 2007-07-10 14:57:20 -0500 (Tue, 10 Jul 2007) | 6 lines
Fix an issue where it was possible to have a service level of over 100%
Between the time recalc_holdtime and update_queue was called, it was possible that the call could have been hungup.
Move both additions to the same place, so this won't happen.
Issue 10158, initial patch by makoto, modified by me.
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r74120 | mmichelson | 2007-07-09 13:32:50 -0500 (Mon, 09 Jul 2007) | 6 lines
The n option for Queue should make the queue exit immediately after failure to reach any members and should not
be dependent on the timeout value passed to Queue
(closes issue #10127, reported by bcnit, repaired by me)
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If the call is answered by another phone, other phones won't display the call as "missed".
You can also add an option to the dial command so that you can have a "followme"
scenario and not count the calls as "missed" when you cancel the call.
Thanks to Ramon and Frank for feedback on this feature.
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r73727 | mmichelson | 2007-07-06 11:36:17 -0500 (Fri, 06 Jul 2007) | 8 lines
Fixing a rare case which causes voicemail to crash when compiled with IMAP storage.
inboxcount has the possibility of finding an "interactive" vm_state when no persistent "non-interactive"
vm_state exists for that mailbox. If this should happen when someone attempts to leave a message, it results in
a crash. This patch, along with my commit in revision 72670 fix issue 10053, reported by jaroth.
closes issue #10053
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r73400 | mmichelson | 2007-07-05 10:59:41 -0500 (Thu, 05 Jul 2007) | 5 lines
Correcting a minor CLI bug I found. When issuing the queue show command, if you type
queue show and then press tab, you can continue pressing tab and it will keep auto-completing
queue names even though only 1 queue can be used as an argument.
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This does not break existing configs - the arguments to p are optional.
Issue 8827, initial patch by junky, mostly rewritten by fw to re-use option p, further modified by me.
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possible for there to be entries in the queue and the thread is just sleeping
(Thanks to mmichelson for bringing the problem to my attention)
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The imapuser being passed in was never getting compared to imapusers of any of the vm_states
in the vmstates list.
I also found some places in the code where I used my typical brace style and changed it to match
the typical Asterisk brace style.
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This feature may be turned on by adding imapgreetings=yes to the general section of voicemail.conf
voicemail.conf.sample has details on the options added.
As a result, IMAP storage now has RETRIEVE and DISPOSE macros defined.
In addition to the IMAP greeting changes, I also have added an enum for the voicemail folders
and so now the code should be easier to understand and maintain when it comes to this area.
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r71953 | mmichelson | 2007-06-26 18:02:09 -0500 (Tue, 26 Jun 2007) | 4 lines
Removing a pointless line. This variable was already set earlier and between then and this
line, there is no way that the values on the right side of the assignment could have changed.
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r71877 | mmichelson | 2007-06-26 14:00:05 -0500 (Tue, 26 Jun 2007) | 11 lines
A few changes, the ultimate goal of which is to keep better track of the number of messages
that a mailbox currently has. A description of the changes:
1. Changed the "updated" field of the vm_state struct to act more as a binary semaphore than a
counting semaphore, since its current implementation made the inboxcount function not work properly.
This change falls in line with a change made by UPenn with their IMAP setup and helps to sync our changes with theirs.
2. Eliminated some redundant calls to get_vm_state_by_mailbox inside leave_voicemail
3. Use the play_folder variable to keep track of the number of old and new messages in a mailbox as the messages are deleted
4. Added an increment to the number of new messages that was not there previously in the leave_voicemail function
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you had 0 messages when using IMAP storage.
Secondary fixes: adding locks to list access in several places
Big thanks to Russell Bryant for helping out with this.
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r69702 | russell | 2007-06-18 11:35:02 -0500 (Mon, 18 Jun 2007) | 6 lines
To prevent 92138749238754 more reports of "I have unixodbc installed, but
still can't build *_odbc.so!", check for ltdl directly, instead of just listing
it as another library to include in the unixodbc check in the configure script.
This also makes ltdl show up as a dependency in menuselect so people know what
to go install. (related to issue #9989, patch by me)
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r69518 | russell | 2007-06-15 10:27:34 -0500 (Fri, 15 Jun 2007) | 5 lines
The SLATRUNK_STATUS variable indicated "SUCCESS" for both an answer of the
incoming call on the trunk, or if the trunk reached its ring timeout.
This patch changes the variable to say "RINGTIMEOUT" in that case.
(issue #9973, reported by n00dle, patch by me)
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r69181 | mmichelson | 2007-06-13 14:41:13 -0500 (Wed, 13 Jun 2007) | 5 lines
Contains a patch for fixing an encoding problem when using Outlook to view voicemail emails and attachments.
This fix has also been tested on Thunderbird, Evolution, Pine, and Mutt.
(Issue 9336, reported by marwick, patched by mutterc)
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the beginning of the file. Also, add a channel variable that indicates
the location in the file where the Playback was stopped.
(closes issue #7655, patch from sharkey)
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r68198 | mmichelson | 2007-06-07 14:47:42 -0500 (Thu, 07 Jun 2007) | 5 lines
Submitting a fix for Issue 8016. Added a check to make sure that greetings get stored properly.
(Issue 8016, reported by edhorton, patched by alamantia with modification by me. Thanks to Jason Parker
for the advice on this).
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r67804 | mmichelson | 2007-06-06 14:26:55 -0500 (Wed, 06 Jun 2007) | 10 lines
Fix for Issue 9810. There was a segfault under a specific set of circumstances:
1. VoiceMailMain was configured in the dialplan with an extension as its argument
2. A message was left for this mailbox
3. Tried to call VoiceMailMain but hung up before entering password.
This was fixed by checking that a pointer was non-null prior to trying to dereference it.
(Issue 9810, reported by xmarksthespot, patched by Corydon76 with modifications by me).
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was enabled. Though no bug was reported to the bugtracker, there was mention of this made as a note on
bug 9810 by edhorton.
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r67558 | russell | 2007-06-05 18:01:44 -0500 (Tue, 05 Jun 2007) | 5 lines
Fix some crashes related to the use of the "meetme" CLI command. The code for
this command was not locking the conference list at all.
(issue #9351, reported by and patch submitted by Junk-Y, committed patch
is different and by me)
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r67424 | mmichelson | 2007-06-05 13:32:50 -0500 (Tue, 05 Jun 2007) | 5 lines
Fix for bug number 9786, wherein voicemails saved to IMAP storage using extensions other than gsm were
unable to be played over the phone. (Issue 9786, reporter: xmarksthespot, Patched by xmarksthe spot with revisions by me,
reviewed by Russell Bryant).
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places that cared about device states were app_queue and the hint code in pbx.c.
The changes include converting it to use the Asterisk event system, as well as
other efficiency improvements.
* app_queue: This module used to register a callback into devicestate.c to
monitor device state changes. Now, it is just a subscriber to Asterisk
events with the type, device state.
* pbx.c hints: Previously, the device state processing thread in devicestate.c
would call ast_hint_state_changed() each time the state of a device changed.
Then, that code would go looking for all the hints that monitor that device,
and call their callbacks. All of this blocked the device state processing
thread. Now, the hint code is a subscriber of Asterisk events with the
type, device state. Furthermore, when this code receives a device state
change event, it queues it up to be processed by another thread so that it
doesn't block one of the event processing threads.
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r66897 | mmichelson | 2007-06-01 16:09:30 -0500 (Fri, 01 Jun 2007) | 3 lines
Submitting a fix for voicemail with IMAP storage. Attachments with format specified as gsm were duplicated (i.e. two attachments) were left.
Thank you very much to xmarksthespot for submitting the patch that fixed this. (Issues 9787 and 8873, Reported by xmarksthespot and jerjer, patched by xmarksthespot)
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- Don't free structures before calling load_config(), because load_config()
already does it
- Use the existing functions for freeing the minivm structures instead of
replicating the code
(issue #9846, patch from eliel)
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is no reason to keep a thread attribute structure on the conference structure.
(Pointed out by Tony Mountifield on the asterisk-dev list)
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places in the code where the same block of code for creating detached threads
was replicated. (patch from bbryant)
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saw this, I couldn't help myself from changing it. Previously, for *every*
device state change, app_queue would spawn a thread to handle it. Now, the
device state callback just puts the state change in a queue and it gets
handled by a single state change processing thread.
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r65200 | murf | 2007-05-18 16:06:27 -0600 (Fri, 18 May 2007) | 9 lines
Merged revisions 65172 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1 line
This update will fix the situation that occurs as described by 9717, where when several targets are specified for a dial, if any one them reports FAIL, the whole call gets FAIL, even though others were ringing OK. I rearranged the priorities, so that a new disposition, NULL, is at the lowest level, and the disposition get init'd to NULL. Then, next up is FAIL, and next up is BUSY, then NOANSWER, then ANSWERED. All the related set routines will only do so if the disposition value to be set to is greater than what's already there. This gives the intended effect. So, if all the targets are busy, you'd get BUSY for the call disposition. If all get BUSY, but one, and that one rings is not answered, you get NOANSWER. If by some freak of nature, the NULL value doesn't get overridden, then the disp2str routine will report NOANSWER as before.
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r64868 | russell | 2007-05-17 21:48:51 -0500 (Thu, 17 May 2007) | 5 lines
Fix a small bug I noticed while working on something else. app_queue did not
unregister its device state monitoring callback in unload_module(). So, this
would make Asterisk crash on the first device state change after you
unload the module.
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except it lets you operate on a channel by name instead of conference member
number. It is very useful in combination with the 'X' option to ChanSpy.
(issue #9671, patch by mnicholson, with some small modifications by me)
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created will now be stored. Then, every channel that joins the conference will
have the MEETMEUNIQUEID channel variable set with this ID. This can be used to
relate callers that come and go from long standing conferences.
(issue #7295, patch by softins)
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entries in the queue log.
(issue #7561, reported and originally patched by fkasumovic, patch slightly
modified and updated to trunk by me)
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This set of changes introduces a new generic event API for use within Asterisk.
I am still working on a way for events to be shared between servers, but this
part is ready and can already be used inside of Asterisk.
This set of changes introduces the first use of the API, as well. I have
restructured the way that MWI (message waiting indication) is handled. It is
now event based instead of polling based. For example, if there are a bunch
of SIP phones subscribed to mailboxes, then chan_sip will not have to
constantly poll the mailboxes for changes. app_voicemail will generate events
when changes occur.
See UPGRADE.txt and CHANGES for some more information on the effects of these
changes from the user perspective. For developer information, see the text in
include/asterisk/event.h.
As always, additional feedback is welcome on the asterisk-dev mailing list.
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minimum amount of time between queue announcements for use when the caller's queue
position changes frequently.
(issue #9604, patch by Matthew Roth)
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blocks instead of one large monolithic app. Supports multiple templates
and is designed mostly for voicemail delivery over e-mail.
There's a todo with a list of ideas in the source code if you want
to contribute. Feedback is appreciated!
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r60521 | russell | 2007-04-06 13:58:46 -0500 (Fri, 06 Apr 2007) | 16 lines
Fix a few problems with SLA. (issue #9459, reported by francesco_r, fixed by me)
* The original behavior was that if one station put a call on hold, another one
picked it up, and then hung up, the code would still consider the call on
hold by the first station, so the trunk would not be hung up. However, to
better comply with what most people seem to expect it to behave, it will now
hang up the trunk.
* Fix a problem with "barge=no". This was only intended to prevent people from
joining calls that are in progress. However, it also prevented other people
from picking up a call that was on hold. This has been fixed.
* When there are no active stations on a trunk and it is on hold, the code now
indicates the HOLD and UNHOLD conditions to the trunk channel. This allows
music on hold to be played to the trunk when it is on hold.
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r60069 | russell | 2007-04-04 11:26:23 -0500 (Wed, 04 Apr 2007) | 4 lines
Fix a problem where if a trunk was hung up while it was on hold, all of the
hints would reflect the line still on hold, even though it should reflect that
it is back to not in use. (issue #9459, reported by francesco_r, fixed by me)
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just having one that can be re-used. There is no functional change here (that
is intentional, anyway!).
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r59361 | file | 2007-03-29 13:38:55 -0400 (Thu, 29 Mar 2007) | 10 lines
Merged revisions 59360 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r59360 | file | 2007-03-29 13:33:58 -0400 (Thu, 29 Mar 2007) | 2 lines
Keep a global array of variables indicating whether certain conference rooms are in use. This ensures that two people going into a new dynamic conference when the 'e' option is set don't go into the same conference room. (issue #8835 reported by eliel)
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r58931 | russell | 2007-03-15 17:25:12 -0500 (Thu, 15 Mar 2007) | 13 lines
Merge changes from svn/asterisk/team/russell/LaTeX_docs.
* Convert most of the doc directory into a single LaTeX formatted document
so that we can generate a PDF, HTML, or other formats from this
information.
* Add a CLI command to dump the application documentation into LaTeX format
which will only be include if the configure script is run with
--enable-dev-mode.
* The PDF turned out to be close to 1 MB, so it is not included. However, you
can simply run "make asterisk.pdf" to generate it yourself. We may include
it in release tarballs or have automatically generated ones on the web site,
but that has yet to be decided.
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r58894 | russell | 2007-03-14 11:33:01 -0500 (Wed, 14 Mar 2007) | 8 lines
By default, don't attempt to do any CallerID handling at all with SLA because
it is known to not work properly in some situations. However, add an option to
enable it for those that would like to use it anyway.
The short story behind this is that to properly handle CallerID with SLA, we
need the ability to change the CallerID on an existing call, and we are not
ready to handle that.
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r58512 | russell | 2007-03-08 16:15:15 -0600 (Thu, 08 Mar 2007) | 5 lines
Hang up the channel that put the call on hold in the event processing thread to
avoid a race condition. Also, if the station originated the call that it is
putting on hold, don't hang up the trunk if it was the only station on the call
and it is hanging up due to hold and not a normal hangup.
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addition to where it is already sent if either side hangs up.
(issue #9219, rgollent)
In passing, I put this code in a function so it would not be duplicated
a third time.
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r57364 | russell | 2007-03-01 17:42:53 -0600 (Thu, 01 Mar 2007) | 16 lines
Merge changes from svn/asterisk/team/russell/sla_updates
* Originally, I put in the documentation that only Zap interfaces would be
supported on the trunk side. However, after a discussion with Qwell, we came
up with a way to make IP trunks work as well, using some things already in
Asterisk. So, here it is, this now officially supports IP trunks.
* Update the SLA documentation to reflect how to setup IP trunks.
* Add a section in sla.txt that describes how to set up an SLA system with
voicemail.
* Simplify the way DTMF passthrough is handled in MeetMe.
* Fix a bug that exposed itself when using a Local channel on the trunk side
in SLA. The station's channel needs to be passed to the dial API when
dialing the trunk.
* Change a WARNING message to DEBUG in channel.h. This message is of no use
to users.
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r57203 | russell | 2007-02-28 16:07:05 -0600 (Wed, 28 Feb 2007) | 7 lines
Merge more changes from svn/asterisk/team/russell/sla_updates
* Add support for private hold. By setting "hold=private" for a trunk, only
the station that put the call on hold will be able to retrieve it from hold.
Also, by setting "hold=private" for a station, any call that station puts
on hold can only be retrieved by that station.
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r57144 | russell | 2007-02-28 13:56:20 -0600 (Wed, 28 Feb 2007) | 6 lines
Merge changes from svn/asterisk/team/russell/sla_updates
* Add support for the "barge=no" option for trunks. If this option is set,
then stations will not be able to join in on a call that is on progress
on this trunk.
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r57089 | russell | 2007-02-28 12:20:05 -0600 (Wed, 28 Feb 2007) | 8 lines
Merge current set of changes from svn/asterisk/team/russell/sla_updates
* Add support for station ring delays. Ring delays can be set globally for a
station or for specific trunks on the station.
* Fix a few bugs in existing code.
* Restructure and Reorganize code to improve readability and maintainability.
* Improve formatting of the "sla show (trunks|stations)" CLI commands.
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external libraries and URLs to these. Please help me add these
references.
We might want to create a similar macro "\linuxpackage" to list
the needed Linux packages in popular distributions.
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r56341 | russell | 2007-02-23 11:58:57 -0600 (Fri, 23 Feb 2007) | 8 lines
The IMAP storage code uses the same code to build the email that is used when
voicemail is sent via email using something like sendmail. In the patch from
bug 8033 to fix various IMAP storage problems, the line endings in the email
file were changed in the code from "\n" to "\r\n". However, this breaks
sending regular voicemail to email. So, this change conditionally sets line
endings to "\r\n" only if IMAP_STORAGE is enabled.
(issue #9128, patch by jarjarbinks, modified by me to not break IMAP storage)
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r56277 | russell | 2007-02-22 17:08:36 -0600 (Thu, 22 Feb 2007) | 18 lines
Merge changes from team/russell/sla_updates.
This batch of changes to the SLA code does a few different things.
* I made the SLA code event driven instead of having to act in a lot of busy
loops while dialing things to wait for state changes. This makes the code
more efficient and readable at the same time.
* I have implemented a couple of new features. The first is inbound trunk
ringing timeouts. This is an option that defines how long to let an incoming
call on a trunk to ring.
* I have also implemented ring timeouts for stations. They may be specified
for the entire station, meaning it is how long to let the station ring before
giving up. You can also specify a ring timeout for a specific trunk on a
station. So, you can say that you only want a specific station to ring 5
seconds if it is line1 ringing, but otherwise, there is no timeout.
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r55957 | file | 2007-02-21 15:35:40 -0500 (Wed, 21 Feb 2007) | 10 lines
Merged revisions 55956 via svnmerge from
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r55956 | file | 2007-02-21 15:32:16 -0500 (Wed, 21 Feb 2007) | 2 lines
Change naughty warning message to provide useful information. If a write now fails on a channel in meetme it will tell you the channel name instead of spitting out the wrong error message.
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convert various #if expressions to #ifdef for macros that may not be defined (and where the value is not important)
Note: two of these changes are in bison generated files which is going to be inconvenient when they are regenerated
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r55006 | russell | 2007-02-16 16:49:42 -0600 (Fri, 16 Feb 2007) | 17 lines
Merged revisions 55005 via svnmerge from
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r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16 Feb 2007) | 9 lines
Revert the change I did in revisions 54955, 54969, and 54970, in 1.2, 1.4,
and trunk. I decided that once a conference is created from meetme.conf,
it is acceptable behavior that the pin can not be changed until the
conference goes away. I also added a note in meetme.conf to describe this
behavior.
We still have another issue in 1.4 and trunk where some conferences with no
users don't go away. That is the real bug that needs to be addressed here.
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r54969 | russell | 2007-02-16 15:12:18 -0600 (Fri, 16 Feb 2007) | 13 lines
Merged revisions 54955 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r54955 | russell | 2007-02-16 14:56:58 -0600 (Fri, 16 Feb 2007) | 5 lines
For conferences that are configured in meetme.conf, check the configuration
file every time someone joins the conference instead of only when the
conference is first created. This is to ensure that changes to the pin
numbers in the config file are always honored. (issue #9073)
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pretty cool things.
First, you can get the device state of anything in the dialplan:
NoOp(SIP/mypeer has state ${DEVSTATE(SIP/mypeer)})
NoOp(The conference room 1234 has state ${DEVSTATE(MeetMe:1234)})
Most importantly, this allows you to create custom device states so you can
control phone lamps directly from the dialplan.
Set(DEVSTATE(Custom:mycustomlamp)=BUSY)
...
exten => mycustomlamp,hint,Custom:mycustomlamp
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r54066 | russell | 2007-02-12 11:58:43 -0600 (Mon, 12 Feb 2007) | 4 lines
- Add the ability to register a callback to monitor state changes in an
asynchronous dial operation.
- Rename the various references to "status" to "state" in the dial API
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r53810 | russell | 2007-02-09 18:35:09 -0600 (Fri, 09 Feb 2007) | 24 lines
Merge team/russell/sla_rewrite
This is a completely new implementation of the SLA functionality introduced in
Asterisk 1.4. It is now functional and ready for testing. However, I will be
adding some additional features over the next week, as well.
For information on how to set this up, see configs/sla.conf.sample
and doc/sla.txt.
In addition to the changes in app_meetme.c for the SLA implementation itself,
this merge brings in various other changes:
chan_sip:
- Add the ability to indicate HOLD state in NOTIFY messages.
- Queue HOLD and UNHOLD control frames even if the channel is not bridged to
another channel.
linkedlists.h:
- Add support for rwlock based linked lists.
dial.c:
- Add the ability to run ast_dial_start() without a reference channel to
inherit information from.
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r53783 | russell | 2007-02-09 18:15:50 -0600 (Fri, 09 Feb 2007) | 4 lines
When the Echo() application receives the digit '#', echo that back as well.
Since we already sent the BEGIN frame for that digit, it makes sense to send
the END as well.
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the location of the header files.
On passing, add a cast to insure -Werror clean compilation
on FreeBSD 6.x, where time_t does not match %ld
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r51167 | qwell | 2007-01-16 16:50:19 -0600 (Tue, 16 Jan 2007) | 6 lines
Fix an issue with IMAP storage and realtime voicemail.
Also update the vmdb sql script for IMAP specific options.
Issue 8819, initial patches by bsmithurst (slightly modified by me)
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r50151 | tilghman | 2007-01-09 07:40:45 -0600 (Tue, 09 Jan 2007) | 12 lines
Merged revisions 50150 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r50150 | tilghman | 2007-01-09 07:30:04 -0600 (Tue, 09 Jan 2007) | 4 lines
The advent of realtime has enabled people to use commas in the fullname field.
This could cause an issue with sending voicemails, when the field is unquoted.
(Issue 8595)
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r50098 | qwell | 2007-01-08 17:39:12 -0600 (Mon, 08 Jan 2007) | 4 lines
Fix an issue with voicemail and users.conf, where it wouldn't ever parse a password, since it was using "secret" instead of "password"
Issue 8761, reported by and patch suggestion from ssokol.
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r49355 | mogorman | 2007-01-03 17:32:03 -0600 (Wed, 03 Jan 2007) | 14 lines
Merged revisions 49354 via svnmerge from
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r49354 | mogorman | 2007-01-03 17:22:47 -0600 (Wed, 03 Jan 2007) | 6 lines
When using ODBC_STORAGE VoicemailMain doesn't create the
subdirectories for a mailbox such as the INBOX directory.
this patch solves that problem, was written by anthony
be-125
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2. Rename 'minmessage' to 'minsecs' for parity.
3. Make 'maxsecs' a per-user option, in addition to global.
(Issue # 8624)
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Comments should explain what the code does, not when something was changed
or who changed it. If you have done a larger contribution, make sure
that you are added to the CREDITS file.
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defined in indications.h to ind_tone_zone_sound and ind_tone_zone,
to avoid conflicts with the structs with the same names
defined in tonezone.h
Hope i haven't missed any instance.
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- document existing but undocumented parameters to send a message
(untested but unchanged;
- ad a new option p(N) to set the initial message delay to N ms
so this can be adapted from the dialplan to various countries;
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to relevant documents and comment on timing issues.
Initial merge of the code in
http://bugs.digium.com/view.php?id=8586
by Filippo Grassilli (Hyppo) to support
the SMS Protocol 2.
In this commit i have tried to minimize the diffs, so further
code cleanup will come in subsequent commits.
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connect to a channel.
Before committing to 1.4 i would like some other people to
review and test this fix - thanks.
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r48396 | mogorman | 2006-12-11 16:11:35 -0600 (Mon, 11 Dec 2006) | 12 lines
Merged revisions 48394 via svnmerge from
https://svn.digium.com/svn/asterisk/branches/1.2
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r48394 | mogorman | 2006-12-11 15:55:43 -0600 (Mon, 11 Dec 2006) | 4 lines
app_externalivr needs a real silence file, and additional
changes to add silence files into core instead of extra
patch provided by bug 8177 with minor additions.
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r48391 | file | 2006-12-11 16:31:23 -0500 (Mon, 11 Dec 2006) | 2 lines
Return non-existant callerid handling to that which it was before. In 1.4 and trunk callerid can be allocated but not have any contents so we have to use ast_strlen_zero before passing it to the relevant functions. (issue #8567 reported by pabelanger)
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r48375 | tilghman | 2006-12-10 18:47:21 -0600 (Sun, 10 Dec 2006) | 13 lines
Merged revisions 48374 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48374 | tilghman | 2006-12-10 18:33:59 -0600 (Sun, 10 Dec 2006) | 5 lines
When doing a fork() and exec(), two problems existed (Issue 8086):
1) Ignored signals stayed ignored after the exec().
2) Signals could possibly fire between the fork() and exec(), causing Asterisk
signal handlers within the child to execute, which caused nasty race conditions.
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r48252 | tilghman | 2006-12-04 19:34:34 -0600 (Mon, 04 Dec 2006) | 14 lines
Merged revisions 48251 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r48251 | tilghman | 2006-12-04 19:26:08 -0600 (Mon, 04 Dec 2006) | 6 lines
If the recording in the database is too large, it will fail to retrieve with
an mmap error. Not too sure why this doesn't happen when we put it in the
database, also, but since that doesn't seem to be broken, I'm not going to fix
it (at least until someone reports it). Solution is to ask for the file in
smaller chunks. (Bug 8385)
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In general this code needs a deep revision, because the body of
do_forward() deletes/overwrites the output channel without freeing
the resouce in some cases, and without notifying the caller.
Also, on FreeBSD with MALLOC_OPTIONS set i am seeing various panics
(duplicate freee etc.)
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In the original code this would happen in the case of
o->forwards >= AST_MAX_FORWARDS
Likely an 1.2/1.4 isse as well - please someone have a look,
while I am hunting a few more similar panics now.
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while also still changing the password "internally".
Issue 7371, initial patch by pdunkel, with rewrite/config comments by me.
Additional modifications (yay bitmask) by pdunkel.
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2. Fix the bug that Asterisk generats wrong dial string (no in IAX2/[username[:password]@]peer[:port][/exten[@context]][/options] format) for IAX.
3. Add support for oh323 channel driver.
4. Re-formate the code.
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r47693 | kpfleming | 2006-11-15 14:27:38 -0600 (Wed, 15 Nov 2006) | 12 lines
Merged revisions 47677 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r47677 | kpfleming | 2006-11-15 11:56:42 -0600 (Wed, 15 Nov 2006) | 4 lines
ensure that message duration is included in email notifications for forwarded messages (BE-96, fix by me after corydon used his clue-bat on me)
ensure that duration in the message metadata is updated if prepending is done during forwarding (related to BE-96)
remove prototype for API call that does not exist
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r47621 | tilghman | 2006-11-14 12:54:40 -0600 (Tue, 14 Nov 2006) | 3 lines
Conversion of res_odbc API to include ast_ prefix did not completely transition app_voicemail
when ODBC_STORAGE is used (reported on IRC by caio1982, not in bugtracker)
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r47617 | file | 2006-11-14 11:45:57 -0500 (Tue, 14 Nov 2006) | 2 lines
Use LOG_DEBUG to print out the indication that app_amd is using default settings instead of using LOG_NOTICE. This stops needless logging of this information under normal circumstances. (issue #8361 reported by Seb7)
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r47432 | kpfleming | 2006-11-10 10:34:04 -0600 (Fri, 10 Nov 2006) | 2 lines
reflect addition/removal of dynamic queue members in queue_log, so that people using dialplan replacement for AgentCallbackLogin can still track login/logout (issue #7736, reported/patched by whoiswes but this commit was written by me and covers all three paths for AQM/RQM)
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r47391 | russell | 2006-11-09 16:26:27 -0500 (Thu, 09 Nov 2006) | 7 lines
Work around an issue that caused menuselect to display a bogus description for
app_voicemail and chan_zap. These modules use some preprocessor directives to
determine what it will report to Asterisk as its description. However, the way
we extract this information from the source files for menuselect is not smart
enough to figure this out.
(issue #8326, #8328)
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also a better implementation of several parts of the original work.
patch provided by 8033 with major upgrades. minor differences from 1.4 patch do to
changes in app_voicemail
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some possibly missing parts in the privacy screening code.
Now that it is more streamlined it is easier to see differences
in handling the various cases.
Have not tested the code in depth.
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On passing, avoid two null-pointer string dereference
while printing messages (which are
sometimes not fatal in some platforms, but still wrong).
These two lines at least should be merged to 1.4 once i am
done with all the changes here.
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Mark with XXX a possible bug in previous code which used
the wrong source in case of a forwarded call.
the function do_forward() needs to be split further, as the initial
part is replicated in another places (with some minor differences, most likely
forgotten when updating after the copy).
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as usual a bit at a time to ease locating new bugs or fixes
worth merging into other branches.
In this commit, introduce a macro, S_REPLACE, that replaces
a string possibly freeing the previous value.
In one of these places (see the comment marked XXX) the previous
code might leak memory - if so, this ought to be merged in 1.4
The macro might be worth putting in one of the global headers
(e.g. include/asterisk/strings.h) as the construct is used
in a million places in the asterisk code.
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see queues.conf.sample for details.
* Added QUEUE_VARIABLES function to set queue variables added setqueuevar and
setqueueentryvar options for each queue, see queues.conf.sample for details.
(#8216, jmls reported and submitted)
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r46363 | russell | 2006-10-27 12:39:31 -0500 (Fri, 27 Oct 2006) | 5 lines
We should always be using _exit() after a fork() or vfork() instead of exit().
This is because exit() does some extra cleanup which in some implementations
of vfork(), for example, can actually modify the state of the parent process,
causing very weird bugs or crashes. (issue #7971, Nick Gavrikov)
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application by configuring them in voicemail.conf (issue #7415, patch by
fkasumovic, with some fixes and documentation updates by myself)
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r44378 | kpfleming | 2006-10-04 14:47:22 -0500 (Wed, 04 Oct 2006) | 4 lines
update thread creation code a bit
reduce standard thread stack size slightly to allow the pthreads library to allocate the stack+data and not overflow a power-of-2 allocation in the kernel and waste memory/address space
add a new stack size for 'background' threads (those that don't handle PBX calls) when LOW_MEMORY is defined
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r43899 | bweschke | 2006-09-28 12:41:05 -0400 (Thu, 28 Sep 2006) | 11 lines
Merged revisions 43897 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43897 | bweschke | 2006-09-28 12:37:15 -0400 (Thu, 28 Sep 2006) | 3 lines
app_queue is comparing the device names incorrectly while checking their statuses. It's internal list of interfaces includes the dial string, while the argument passed to this function does not have the dial string (/n for a local channel). This causes it to ignore the device state changes because it thinks it belongs to none of its members. (#8040 reported and patch by tim_ringenbach)
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r43803 | qwell | 2006-09-27 12:44:02 -0700 (Wed, 27 Sep 2006) | 4 lines
Fix an issue with PLAYBACKSTATUS not being set under certain circumstances.
Fix a minor issue, to make it use the filenames that were parsed, instead of the entire argument string.
Fix Background() to return -1 like Playback(), if no args are specified.
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In passing, I have cleaned up some formatting to better comply with our
guidelines. I have also changed one place to use S_OR(), and a couple of
places to use ast_strlen_zero() as appropriate.
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Gives "at a glance" information about a single queue, or all queues.
Issue #8035, patch by rgollent, slightly modified (formatting) by me.
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r43700 | russell | 2006-09-26 16:24:39 -0400 (Tue, 26 Sep 2006) | 14 lines
Merged revisions 43699 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43699 | russell | 2006-09-26 16:23:15 -0400 (Tue, 26 Sep 2006) | 6 lines
When parsing the sections of voicemail.conf that contain mailbox definitions,
don't introduce a length limit on the definition by using a 256 byte temporary
storage buffer. Instead, make the temporary buffer just as big as it needs
to be to hold the entire mailbox definition.
(fixes BE-68)
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r43695 | file | 2006-09-26 16:09:41 -0400 (Tue, 26 Sep 2006) | 2 lines
Slight overhaul of the whisper support. 1. We need to duplicate the frame from ast_translate 2. We need to ensure we always have signed linear coming in for signed linear combining. 3. We need to ensure we are always feeding signed linear out. 4. Properly store and restore write format when beeping on the channel we are whispering on. 5. Properly discontinue the stream on the channel for the beep. (issue #8019 reported by timkelly1980)
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Add option to the various methods of adding a queue member, to add the "member name".
Member name is used in (most) queue log records, in place of the interface name.
This makes it consistent, so that you can log in from any device, and still be logged as "member name"
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r42783 | tilghman | 2006-09-11 16:47:23 -0500 (Mon, 11 Sep 2006) | 4 lines
When paging, only wait 5 seconds for the marked user to enter the conference.
After that, assume the paging already completed by the time the channel entered
the conference and drop back out. (Issue 7275)
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in 1.2. This corrected a longstanding confusion about the return value.
Unfortunately, it broke this app in the process. (Issue 7906)
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to function along with an option to keep backward compatible with "old-style" functionality in 1.2.
(#6595 - davetroy reported and patched w/some very minor mods/corrections)
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now reports AST_MODULE_LOAD_DECLINE when loading if config file
is not there, also fixed an error in res_config_pgsql where it
had a non static function when it should.
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There are some situations in Asterisk where ast_frame and/or iax_frame
structures are rapidly allocatted and freed (at least 50 times per second
for one call).
This code significantly improves the performance of ast_frame_header_new(),
ast_frdup(), ast_frfree(), iax_frame_new(), and iax_frame_free() by keeping
a thread-local cache of these structures and using frames from the cache
whenever possible instead of calling malloc/free every time.
This commit also converts the ast_frame and iax_frame structures to use the
linked list macros.
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- restructured build tree and makefiles to eliminate recursion problems
- support for embedded modules
- support for static builds
- simpler cross-compilation support
- simpler module/loader interface (no exported symbols)
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to another user (options 3, 5, 2).
If the context/extension didn't exist in the dialplan (and why should it have to?),
it would fail, saying that it's an "invalid extension".
(issue BE-71)
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fix prototype for a channel walking function to use a const input pointer
use existing channel walk by name prefix instead of reproducing that code in this app
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38389 65c4cc65-6c06-0410-ace0-fbb531ad65f3
move guts of dialplan application into separate function, so it can be shared bythe new application i'm about to add :-)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38374 65c4cc65-6c06-0410-ace0-fbb531ad65f3
use API call for finding channel by name prefix
code formatting to match guidelines (lost about half the of the indenting)
remove useless automatic variable initializations
don't set the spying channel's read format to SLINEAR when we don't do anything with the voice frames we read from it anyway
use proper option argument checking for volume argument
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
inet_ntoa, which uses thread specific data (aka thread local storage) instead
of stack allocatted buffers to store the result.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@38042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r37441 | kpfleming | 2006-07-12 10:46:56 -0500 (Wed, 12 Jul 2006) | 3 lines
fix a case where ast_lock_path() could leave a randomly-named lock file hanging around
make ast_unlock_path actually report when unlocking fails
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r37442 | kpfleming | 2006-07-12 10:53:53 -0500 (Wed, 12 Jul 2006) | 2 lines
fix a weird case where a lock file could be left (but would happen almost never)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@37514 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Remove obsolete modules from modules.conf.sample
(make install will warn if those exist on the machine)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36979 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- use ast_calloc instead of malloc + memset
- return immediately on ast_calloc failure instead of indenting the whole func
- remove a duplicate ast_strdupa
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36700 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r36377 | tilghman | 2006-06-30 09:05:53 -0500 (Fri, 30 Jun 2006) | 5 lines
Bug 7349 - Directory did not work correctly when USE_ODBC_STORAGE was defined.
Note: Russell agreed that this should have worked, which is why this is
classified as a bugfix.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36380 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-----------
- Adding devicestate providers, a new architecture to add non-channel related
device state information, like parking lots, queues, meetmes, vending machines
and Windows 98 reboots (lots of blinking on those lights)
- Adding provider for parking lots, so you can subscribe to the status of a
parking lot
- Adding provider for meetme, so you can have a blinking lamp for a meetme
( Example: exten => edvina,hint,meetme:1234 )
- Adding support for directed parking - set the PARKINGEXTEN before you manually
call Park() and you will be parked on that space. If it's occupied, dialplan
execution will continue.
This work was sponsored by Voop A/S - www.voop.com
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@36055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Don't name internal static functions ast_
- Expand the buffer for variables, since I almost always hit the limit on my channels
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
support the new location for zaptel.h and tonezone.h
use the dependency information output by menuselect to build Makefile rules for each module for header files and libraries
combine the common rules into a top-level Makefile.rules file
remove all (now) unnecessary stuff from subdir Makefiles
change translator API so that the newpvt() callback returns an int instead of a pointer (it no longer allocates memory)
alphabetize --with-<foo> options in configure script
enhance Net-SNMP support in configure script to provide a --with-netsnmp option
fix support for --with-pq so that if pg-config is not found when --with-pq is specified, an error will be generated
add 'optional package' usage to modules now that menuselect can output it
allow res_snmp to build by default, since the new loader changes coming soon will solve the function naming problem (and users can disable it via menuselect anyway)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
sufficient amount of time. Even if they happen to be still present, the main
Makefile will spit out a huge warning telling the user that modules not
installed by that run of "make install" are present in the modules directory.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35605 65c4cc65-6c06-0410-ace0-fbb531ad65f3
subdirectory instead of a for loop
- remove the FORCE target from the main Makefile and add the couple places
I used it to the .PHONY target. .PHONY does the same thing and is a built-in
more efficient way of doing it.
- add a bunch more targets to .PHONY ...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35503 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- add a copyright header to the build_tools Makefile
- remove 'depend' from the 'all' target in agi/ and utils/ since it is handled
by the main Makefile already
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
since they are targets that do not have resulting files and are never listed
as prerequisites to real targets. Using .PHONY in this manner improves make
performance by never having to check for resulting files.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@35415 65c4cc65-6c06-0410-ace0-fbb531ad65f3
reverted per-directory .cleancount support
added ability for 'remove_on_change' to support multiple filenames
add 'remove_on_change' support to members, not just categories
only do 'remove_on_change' removals if the config is actually saved
add a 'remove_on_change' entry for each module found by prep_moduledeps so that if it gets turned off any existing .o/.so files will disappear
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@34577 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r34159 | kpfleming | 2006-06-14 17:17:37 -0500 (Wed, 14 Jun 2006) | 2 lines
use existing dial string parser for strings supplied to iax2_devicestate, because they can be complete dial strings, not just device names
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r34160 | kpfleming | 2006-06-14 17:22:21 -0500 (Wed, 14 Jun 2006) | 2 lines
coding style cleanups on queue interface handling code that was committed for the last release
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@34161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Joshua Colp
Mark Michelson
Dwayne M. Hubbard
Jason Parker
Tilghman Lesher
Steve Murphy
Russell Bryant
BJ Weschke
Kevin P. Fleming
Luigi Rizzo
Olle Johansson
Brett Bryant
Paul Cadach
Matt O'Gorman
TransNexus OSP Development
Matthew Fredrickson
Anthony LaMantia
North Antara
Mark Spencer
Jim Dixon