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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r78995 | russell | 2007-08-10 10:20:09 -0500 (Fri, 10 Aug 2007) | 4 lines
The last set of changes that I made to "core show locks" made it not able to
track mutexes unless they were declared using AST_MUTEX_DEFINE_STATIC. Locks
initialized with ast_mutex_init() were not tracked. It should work now.
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r78103 | mmichelson | 2007-08-03 15:25:22 -0500 (Fri, 03 Aug 2007) | 7 lines
Changed the behavior of sip's realtime_peer function to match the corresponding way of matching for non-realtime peers.
Now matches are made on both the IP address and port number, or if the insecure setting is set to "port" then just match on the
IP address.
In order to accomplish this, I also added a new API call, ast_category_root, which returns the first variable of an ast_category struct
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r78184 | russell | 2007-08-06 11:50:54 -0500 (Mon, 06 Aug 2007) | 5 lines
Fix the return value of AST_LIST_REMOVE(). This shouldn't be causing any
problems, though, because the only code that uses the return value only checks
to see if it is NULL.
(closes issue #10390, pointed out by mihai)
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The problem was that res_crypto now has a RWLIST named "keys". The macro
for defining this list defines a function used as a constructor for the list
called "init_keys". However, there was another function called init_keys in
this module for a CLI command. The fix is just to prepend the generated
functions with underscores.
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r78095 | russell | 2007-08-03 14:39:49 -0500 (Fri, 03 Aug 2007) | 28 lines
Add some improvements to lock debugging. These changes take effect
with DEBUG_THREADS enabled and provide the following:
* This will keep track of which locks are held by which thread as well as
which lock a thread is waiting for in a thread-local data structure. A
reference to this structure is available on the stack in the dummy_start()
function, which is the common entry point for all threads. This information
can be easily retrieved using gdb if you switch to the dummy_start() stack
frame of any thread and print the contents of the lock_info variable.
* All of the thread-local structures for keeping track of this lock information
are also stored in a list so that the information can be dumped to the CLI
using the "core show locks" CLI command. This introduces a little bit of a
performance hit as it requires additional underlying locking operations
inside of every lock/unlock on an ast_mutex. However, the benefits of
having this information available at the CLI is huge, especially considering
this is only done in DEBUG_THREADS mode. It means that in most cases where
we debug deadlocks, we no longer have to request access to the machine to
analyze the contents of ast_mutex_t structures. We can now just ask them
to get the output of "core show locks", which gives us all of the information
we needed in most cases.
I also had to make some additional changes to astmm.c to make this work when
both MALLOC_DEBUG and DEBUG_THREADS are enabled. I disabled tracking of one
of the locks in astmm.c because it gets used inside the replacement memory
allocation routines, and the lock tracking code allocates memory. This caused
infinite recursion.
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r77788 | russell | 2007-07-30 14:13:31 -0500 (Mon, 30 Jul 2007) | 10 lines
(closes issue #10279)
Reported by: seanbright
Patches:
res_agi.carefulwrite.1.4.07252007.patch uploaded by seanbright (license 71)
res_agi.carefulwrite.trunk.07252007.patch uploaded by seanbright (license 71)
Allow the "agi_network: yes" line to be printed out in the AGI debug output.
Also, allow partial writes to be handled when writing out this line just like
it is for all of the others.
........
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- Makes the structures handling external AGI commands a bit more thread-safe
- Makes AGI transparently work with both live and hungup channels
- DeadAGI is hence no longer necessary and is deprecated
- CLI bug fixes
- Commands will refuse to run if the channel is dead and the command is nonsensical
for dead channels.
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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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Reported by: snuffy
Patches:
doxygen-updates.diff uploaded by snuffy (license 35)
Another big batch of doxygen documentation updates
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Reported by: bbryant
Patches:
20070720__core_debug_by_file.patch uploaded by bbryant (license 36)
(with some modifications by me)
Tested by: russell, bbryant
This set of changes introduces the ability to set the core debug or verbose
levels on a per-file basis. Interestingly enough, in 1.4, you have the ability
to set core debug for a single file, but that functionality was accidentally
lost in the conversion of the CLI commands to the new format.
This patch improves upon what was in 1.4 by letting you set it for more than 1
file, and by also supporting verbose.
*** Janitor Project ***
This patch also introduces a new macro, ast_verb(), which is similar
to ast_debug(). Setting the per file verbose value only works for messages that
use this macro. Converting existing uses of ast_verbose() can be done like:
if (option_debug > 2)
ast_verbose(VERBOSE_PREFIX_3 "Something useful\n");
...
ast_verb(3, "Something useful\n");
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(e.g. chan_sip.c in a subsequent commit).
Obviously exposing the internals of a data structure is far from ideal
(especially in a case like this where the implementation is very
inefficient and will need to be changed at some point).
On the other hand, it was also unclear what additional APIs should
we provide instead, and because exposing the stucture has no impact
on source and binary compatibility, this seemed to me the best option at
this time.
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in a consistent way. This is meant to replace the custom code
which is repeated all over the place in the various files when
parsing config files, CLI entries and other string information.
Right now the code supports parsing int32, uint32 and sockaddr_in with
optional default values and bound checks. It contains minimal error
checking, but that can be easily extended as the need arises.
Being a new API i am introducing this only in trunk, though I believe
that once the interface has been ironed out it might become a
worthwhile addition to 1.4 as well - basically, the first time
we will need to fix a piece of argument parsing code, we might as
well bring in this change and use the new API instead.
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use the \retval tag for documenting return values, fixing various warnings
when generating the documentation, and various other things.
(closes issue #10203, snuffy)
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sockets other than RTP ones.
The main change is a new API function in main/rtp.c (see there
for a description)
int ast_stun_request(int s, struct sockaddr_in *dst,
const char *username, struct sockaddr_in *answer)
which can be used to send an STUN request on a socket, and
optionally wait for a reply and store the STUN_MAPPED_ADDRESS
into the 'answer' argument (obviously, the version that
waits for a reply is blocking, but this is no different
from DNS resolutions).
Internally there are minor modifications to let stun_handle_packet()
be somewhat configurable on how to parse the body of responses.
At the moment i am not committing any change to the clients,
but adding STUN client support is extremely simple, e.g. chan_sip.c
could do something like this:
+ add a variable to store the stun server address;
static struct sockaddr_in stunaddr = { 0, }; /*!< stun server address */
+ add code to parse a config file of the form "stunaddr=my.stun.server.org:3478"
(not shown for brevity);
+ right after binding the main sip socket, talk to the stun server to
determine the externally visible address
if (stunaddr.sin_addr.s_addr != 0)
ast_stun_request(sipsock, &stunaddr, NULL, &externip);
so now 'externip' is set with the externally visible address.
so it is really trivial.
Similarly ast_stun_request could be called when creating the RTP
socket (possibly adding a struct sockaddr_in field in the struct
ast_rtp to store the externalip).
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #10133)
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r74374 | qwell | 2007-07-10 13:39:30 -0500 (Tue, 10 Jul 2007) | 13 lines
Merged revisions 74373 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r74373 | qwell | 2007-07-10 13:37:23 -0500 (Tue, 10 Jul 2007) | 5 lines
Use res_ndestroy on systems that have it. Otherwise, use res_nclose.
This prevents a memleak on NetBSD - and possibly others.
Issue 10133, patch by me, reported and tested by scw
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Reported by: julien23
Patches submitted by: julien23
Add the ability to disable recording the input or output streams in res_monitor.
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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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*must* write the file to the FILE *, and not the raw fd. Otherwise, it breaks
TLS support.
Thanks to rizzo for catching this!
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Handle transferring large files from the built-in http server. Previously, the
code attempted to malloc a block as large as the file itself. Now it uses the
sendfile() system call so that the file isn't copied into userspace at all if
it is available. Otherwise, it just uses a read/write of small chunks at a time.
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Also adds a new "reporting" permission for manager, since it can be incredibly spammy.
This permission was discussed on the -dev mailing list some months back.
Issue 8613, patch by johann8384, with some minor changes by me.
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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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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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r67993 | oej | 2007-06-07 11:00:44 +0200 (Thu, 07 Jun 2007) | 6 lines
Issue #9738 - Make sure we can unload res_jabber. Patch by phsultan - thanks!
Due to a bug in the iksemel library, this will not work if you are using GTLS
in the connection. That's being investigated. If you figure out a way to handle
that without us having to patch iksemel, let us know in the bug report. Thanks.
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r67716 | russell | 2007-06-06 11:55:59 -0500 (Wed, 06 Jun 2007) | 13 lines
Merged revisions 67715 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06 Jun 2007) | 5 lines
We have some bug reports showing crashes due to a double free of a channel.
Add a sanity check to ast_channel_free() to make sure we don't go on trying
to free a channel that wasn't found in the channel list.
(issue #8850, and others...)
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r67492 | russell | 2007-06-05 15:53:28 -0500 (Tue, 05 Jun 2007) | 16 lines
This bug has been hanging over my head ever since I wrote this SLA code.
Every time I tried to go debug it by adding some debug output, the behavior
would change. It turns out I wasn't crazy. I had the following piece of code:
if (remove)
AST_LIST_REMOVE_CURRENT(...);
Well, AST_LIST_REMOVE_CURRENT was not wrapped in braces, so my conditional
statement didn't do much good at all. It always ran at least all of the
macro minus the first statement, so I was seeing list entries magically
disappear when they weren't supposed to.
After many hours of debugging, I have come to this extremely irritating fix. :)
(issues #9581, #9497)
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r67308 | russell | 2007-06-05 10:51:53 -0500 (Tue, 05 Jun 2007) | 5 lines
When shutting down "gracefully", go through and run the unload() callbacks for
all of the modules. "stop now" is considered a non-graceful shutdown and will
not go through this process.
(issue #9804, reported by chrisost, patch by me)
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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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places in the code where the same block of code for creating detached threads
was replicated. (patch from bbryant)
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class in musiconhold.conf, a caller on hold may press this digit to switch
to listening to that music class.
This involved adding a new callback for generators, which allow generators
to get notified of DTMF from the channel they are running on. Then, a callback
was implemented for the music on hold generators.
(patch from bbryant)
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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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allow you to initiate an ENUM query using ENUMQUERY, and then access the
details of all of the results using ENUMRESULT. Previously, if you wanted
to access multiple results, Asterisk would have to do a new DNS lookup every
time. (patch by bbryant)
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- Add missing option to options.h
- Add missing variables to asterisk.h
- Move manager version to manager.h include file
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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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r61781 | russell | 2007-04-24 14:00:06 -0500 (Tue, 24 Apr 2007) | 6 lines
Improve DTMF handling in ast_read() even more in response to a discussion on
the asterisk-dev mailing list. I changed the enforced minimum length of a
digit from 100ms to 80ms. Furthermore, I made it now enforce a gap of 45ms in
between digits. These values are not configurable in a configuration file
right now, but they can be easily changed near the top of main/channel.c.
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I started this for use with SLA but ended up deciding not to use it. However,
there is no reason not to put this part in, anyway.
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r60850 | tilghman | 2007-04-08 22:01:12 -0500 (Sun, 08 Apr 2007) | 10 lines
Merged revisions 60849 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r60849 | tilghman | 2007-04-08 21:49:06 -0500 (Sun, 08 Apr 2007) | 2 lines
Don't check for error when lowering priority (according to the manpage, it should never happen anyway). It might could happen, though, if another thread messed with the priority, so safeguard against that (reported via -dev list).
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r60603 | russell | 2007-04-06 15:58:43 -0500 (Fri, 06 Apr 2007) | 13 lines
To be able to achieve the things that we would like to achieve with the
Asterisk GUI project, we need a fully functional HTTP interface with access
to the Asterisk manager interface. One of the things that was intended to be
a part of this system, but was never actually implemented, was the ability for
the GUI to be able to upload files to Asterisk. So, this commit adds this in
the most minimally invasive way that we could come up with.
A lot of work on minimime was done by Steve Murphy. He fixed a lot of bugs in
the parser, and updated it to be thread-safe. The ability to check
permissions of active manager sessions was added by Dwayne Hubbard. Then,
hacking this all together and do doing the modifications necessary to the HTTP
interface was done by me.
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r59207 | russell | 2007-03-26 12:45:55 -0500 (Mon, 26 Mar 2007) | 7 lines
The AUDIORTPQOS and VIDEORTPQOS variables are not fully functional in some
because they get set in sip_hangup. So, there are common situations where
the variables will not be available in the dialplan at all. So, this patch
provides an alternate method for getting to this information by introducing
AUDIORTPQOS and VIDEORTPQOS dialplan functions.
(issue #9370, patch by Corydon76, with some testing by blitzrage)
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r59202 | nadi | 2007-03-26 17:25:53 +0200 (Mo, 26 Mär 2007) | 4 lines
* mISDN >= 1.2 provides a dsp pipeline for i.e. echo cancellation modules, make chan_misdn use it.
* add a check for linux/mISDNdsp.h to configure.ac and update the autogenerated files: 'configure', 'autoconfig.h.in'
(the 'configure' script was not in sync with the latest configure.ac, so the diff is a bit bigger than expected).
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* Add new module, cdr_sqlite3_custom which allows logging custom CDRs into a
SQLite3 database. (issue #7149, alerios)
* Add new module, res_config_sqlite, which adds realtime database configuration
support for SQLite version 2. I decided that this was ok since we didn't have
any realtime support for version 3. If someone ports this to version 3, then
version 2 support can be removed or marked deprecated.
(issue #7790, rbarun_proformatique)
* Mark cdr_sqlite as deprecated in favor of cdr_sqlite3_custom.
Also, note that there were other modules on the bug tracker that did not make
the cut because they provided some duplicated functionality. Those are:
* cdr_sqlite3 (issue #6754, moy)
* cdr_sqlite3 (issue #8694, bsd)
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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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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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T.140/RFC 2793 is a live communication channel, originally
created for IP based text phones for hearing impaired.
Feels very much like the old Unix talk application.
This code is developed and disclaimed by John Martin of Aupix, UK.
Tested for interoperability by myself and Omnitor in Sweden,
the company that wrote most of the specifications.
A big thank you to everyone involved in this.
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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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r52997 | russell | 2007-01-30 17:23:24 -0600 (Tue, 30 Jan 2007) | 5 lines
When we are checking for a system installed version of libgsm, we need to check
for gsm.h as well. Furthermore, when checking for this header, it may be
located in a gsm/ sub directory, so check for that, as well.
(issue #8773)
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r52494 | jdixon | 2007-01-28 22:18:36 -0600 (Sun, 28 Jan 2007) | 4 lines
Fixed problem with jitterbuf, whereas it would not complain about, and
would allow itself to be overfilled (per the max_jitterbuf parameter). Now
it rejects any data over and above that size, and complains about it.
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r52506 | russell | 2007-01-29 10:54:27 -0600 (Mon, 29 Jan 2007) | 5 lines
Clean up a few things in the last commit to the adaptive jitterbuffer code.
- Specifically indicate to the compiler that the "dropem" variable only
needs one but.
- Change formatting to conform to coding guidelines.
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r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
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AST_INLINE_API() is a macro that takes a block of code as an argument.
Using preprocessor #directives in the argument is not supported by all
compilers, and it is a bit of an obfuscation anyways, so avoid it.
As a workaround, define a macro that produces either its argument
or nothing, and use that instead of #ifdef/#endif within the
argument to AST_INLINE_API().
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place, rather than repeating the check on every single file.
Changes to the individual files are coming.
The header file name has been suggested by kevin.
Approved by: kpfleming
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r50867 | kpfleming | 2007-01-15 09:03:06 -0600 (Mon, 15 Jan 2007) | 2 lines
use the ACX_PTHREAD macro from the Autoconf macro archive for setting up compiler pthreads support... should improve portability to platforms with unusual pthreads requirements
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r50466 | file | 2007-01-11 00:19:39 -0500 (Thu, 11 Jan 2007) | 2 lines
Add support to see whether NAT was detected (yay symmetric RTP) and also add a check in chan_sip so that if NAT has been detected and the reinvite behind nat option has been turned off, then just do partial bridge. (issue #8655 reported by mnicholson)
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previously set are erroneously still set (Bug 6701). After discussion,
it was determined this should only be changed in trunk.
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r49102 | kpfleming | 2007-01-01 17:34:35 -0600 (Mon, 01 Jan 2007) | 2 lines
check specifically for VLDTMF and transcoding support in the system's Zaptel installation, and make only the modules that need those features dependent on them (this will allow building the other Zaptel-using parts of Asterisk against older versions of Zaptel or those on other platforms that haven't caught up yet to the Linux version)
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r48998 | kpfleming | 2006-12-27 15:08:30 -0600 (Wed, 27 Dec 2006) | 3 lines
move extern declaration for this option to a header file where it belongs
provide an initial value for 'languageprefix' option, instead of relying on randomness to provide a useful value
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r48964 | file | 2006-12-25 23:31:58 -0500 (Mon, 25 Dec 2006) | 2 lines
Add an API call that initializes an RTP structure. We need this because chan_sip is cheeky and uses a temporary RTP structure for codec purposes, and the API calls that are used rely on the lock. (Pointed out on asterisk-dev by Andy Wang)
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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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this, implementing locking of this list to make it thread-safe.
- Add a "redirect" option to http.conf that allows redirecting one URI to
another. I was inspired to do this while playing with the Asterisk GUI. I
got tired of typing this URL to get to the GUI:
http://localhost:8088/asterisk/static/config/cfgadvanced.html
So, now I have the following line in http.conf:
redirect=/=/asterisk/static/config/cfgadvanced.html
Now, I can type the following into my browser and go to the GUI:
http://localhost:8088
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I don't know when the bug was introduced, but with the typical usage
c->fin = FRAMECOUNT_INC(c->fin)
the frame counters stay to 0.
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http://bugs.digium.com/view.php?id=8602
(i am not sure if there is still missing cast in
front of the alloca() call - being a macro this is
probably triggered only when actually used).
Add function ast_str_reset() to reinitialize the
string to an empty string instead of playing with
the internal fields.
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r48521 | kpfleming | 2006-12-16 14:12:41 -0600 (Sat, 16 Dec 2006) | 2 lines
since we really, really have to have autoconfig.h included before all other headers (especially system headers), the Makefile will now force it to happen (this will fix build problems with files like ast_expr2f.c, where we can't control the inclusion order in the file itself)
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renaming them to ast_str ... and putting the
struct ast_threadstorage pointer into the struct ast_str.
This makes the code a lot more readable.
At this point we can use these routines also to
replace ast_build_string().
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While doing this, add a bit of documentation, and slightly
extend the functionality as follows:
+ a max_len of -1 means that we take whatever the current size
is, and never try to extend the buffer;
+ add support for alloca()-ted dynamic strings, which is very
useful for all cases where we do an ast_build_string() now.
Next step is to simplify the interface by using shorter names
(e.g. ast_str as a prefix) and removing the _thread variant
of the functions by saving the threadstorage reference into
the struct ast_str. This can be done by overloading the
'type' field.
Finally, I will do my best to remove the convoluted interface
that results from trying to support platforms without va_copy().
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r48472 | file | 2006-12-14 12:36:12 -0500 (Thu, 14 Dec 2006) | 2 lines
Payload values on the RTP structure can change AFTER a bridge has started. This comes from the packet handling of the SIP response when indication that it was answered has been sent. Therefore we need to protect this data with a lock when we read/write. (issue #8232 reported by tgrman)
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are passed as an argument.
- Update the code in main/http.c to use the new interface
(the diff is large but mostly mechanical, due to the name change of
several variables);
- And since now it is trivial, implement "AMI over TLS", and document
the possible options in manager.conf
- And since the test client (openssl s_client -connect host:port )
does not generate \r\n as a line terminator, make get_input()
also accept just a \n as a line terminator (Mac users: do you
also need the \r-only version ?)
The option parsing in manager.conf is not very efficient, and needs
to be cleaned up and made similar to what we have in http.conf
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and implement services over tcp and/or tcp-tls.
This commit is nothing more than moving structure definitions
(and documentation) from main/http.c to include/asterisk/http.h
(temporary location until we find a better place), and removing the
'static' qualifier from server_root() and server_start().
The name change (adding the ast_ prefix as a minimum, and then
possibly a more meaningful name) is postponed to future commits.
Does not apply to other versions of asterisk.
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Russell Bryant
Joshua Colp
Doug Bailey
Mark Michelson
Tilghman Lesher
Luigi Rizzo
Steve Murphy
Jason Parker
Olle Johansson
Kevin P. Fleming
Dwayne M. Hubbard
Nadi Sarrar