A new xml element was created to manage the AMI actions documentation,
using AstXML.
To register a manager action using XML documentation it is now possible
using ast_manager_register_xml().
The CLI command 'manager show command' can be used to show the parsed
documentation.
Example manager xml documentation:
<manager name="ami action name" language="en_US">
<synopsis>
AMI action synopsis.
</synopsis>
<syntax>
<xi:include xpointer="xpointer(...)" /> <-- for ActionID
<parameter name="header1" required="true">
<para>Description</para>
</parameter>
...
</syntax>
<description>
<para>AMI action description</para>
</description>
<see-also>
...
</see-also>
</manager>
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes:
- CLI command handlers
- CLI command handler arguments
- AGI command handlers
- AGI command handler arguments
- Dialplan application handler arguments
- Speech engine API function arguments
In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing.
Review: https://reviewboard.asterisk.org/r/251/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The channel drivers which have been most heavily tested with these enhancements are
chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be
introduced in a later commit. chan_skinny has code added to it here, but according
to user pj, the support on chan_skinny is not working as of now. This will be fixed in
a later commit.
A special thanks goes out to bugtracker user gareth for getting the ball rolling and
providing the initial support for this work. Without his initial work on this, this would
not have been nearly as painless as it was.
This functionality has been tested by Digium's product quality department, as well as a
customer site running thousands of calls every day. In addition, many many many many bugtracker
users have tested this, too.
(closes issue #8824)
Reported by: gareth
Review: http://reviewboard.digium.com/r/201
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This API provides a generic way for multiple RTP stacks to be
integrated into Asterisk. Right now there is only one present, res_rtp_asterisk,
which is the existing Asterisk RTP stack. Functionality wise this commit
performs the same as previously. API documentation can be viewed in the
rtp_engine.h header file.
Review: http://reviewboard.digium.com/r/209/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@186078 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines
Fix cases where the internal poll() was not being used when it needed to be.
We have seen a number of problems caused by poll() not working properly on
Mac OSX. If you search around, you'll find a number of references to using
select() instead of poll() to work around these issues. In Asterisk, we've
had poll.c which implements poll() using select() internally. However, we
were still getting reports of problems.
vadim investigated a bit and realized that at least on his system, even
though we were compiling in poll.o, the system poll() was still being used.
So, the primary purpose of this patch is to ensure that we're using the
internal poll() when we want it to be used.
The changes are:
1) Remove logic for when internal poll should be used from the Makefile.
Instead, put it in the configure script. The logic in the configure
script is the same as it was in the Makefile. Ideally, we would have
a functionality test for the problem, but that's not actually possible,
since we would have to be able to run an application on the _target_
system to test poll() behavior.
2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT
is not defined.
3) Change uses of poll() throughout the source tree to ast_poll(). I feel
that it is good practice to give the API call a new name when we are
changing its behavior and not using the system version directly in all cases.
So, normally, ast_poll() is just redefined to poll(). On systems where
AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll().
4) Change poll() in main/poll.c to be ast_internal_poll().
It's worth noting that any code that still uses poll() directly will work fine
(if they worked fine before). So, for example, out of tree modules that are
using poll() will not stop working or anything. However, for modules to work
properly on Mac OSX, ast_poll() needs to be used.
(closes issue #13404)
Reported by: agalbraith
Tested by: russell, vadim
http://reviewboard.digium.com/r/198/
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Special thanks goes to DEA who had to redo this patch twice
because we first put unload/load support in and later redid the way
we configure devices and lines.
(closes issue #10297)
Reported by: DEA
Patches:
skinny-reload-trunkv2.diff uploaded by wedhorn (license 30)
skinny-reload-trunk-v4.txt uploaded by DEA (license 3)
With mods by me based on feedback from wedhorn and Russell and seanbright
Tested by: DEA, mvanbaak, pj
Review: http://reviewboard.digium.com/r/130/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179122 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch includes a number of changes to the indications API. The primary
motivation for this work was to improve stability. The object management
in this API was significantly flawed, and a number of trivial situations could
cause crashes.
The changes included are:
1) Remove the module res_indications. This included the critical functionality
that actually loaded the indications configuration. I have seen many people
have Asterisk problems because they accidentally did not have an
indications.conf present and loaded. Now, this code is in the core,
and Asterisk will fail to start without indications configuration.
There was one part of res_indications, the dialplan applications, which did
belong in a module, and have been moved to a new module, app_playtones.
2) Object management has been significantly changed. Tone zones are now
managed using astobj2, and it is no longer possible to crash Asterisk by
issuing a reload that destroys tone zones while they are in use.
3) The API documentation has been filled out.
4) The API has been updated to follow our naming conventions.
5) Various bits of code throughout the tree have been updated to account
for the API update.
6) Configuration parsing has been mostly re-written.
7) "Code cleanup"
The code is from svn/asterisk/team/russell/indications/.
Review: http://reviewboard.digium.com/r/149/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Dan saw regular segfaults with the old implementation and
rewrote it to make it really eventbased.
I altered it to be trunk compatible and wedhorn gave some feedback
and ideas how to make it even better.
(closes issue #13821)
Reported by: DEA
Patches:
chan_skinny-mwi-events.txt uploaded by DEA (license 3)
Tested by: mvanbaak, DEA
"no probs by me" from wedhorn
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169367 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Behaviour now is that general codec config flows to default_line and default_device. [devices] stuff amends default_device and similar for [lines]. These are copied to individual device and line as they are created.
Added confcapability and confprefs for the configured stuff which doesn't change as device and so on are connected. prefs are based on line prefs if they exist, else the device prefs are used (prefs identifies codec order).
(closes issue #13806)
Reported by: pj
Patches:
codecs.diff uploaded by wedhorn (license 30)
Tested by: pj and me
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
We dont want to scare users with this, so we added a devmode compile flag
(closes issue #13952)
Reported by: wedhorn
Patches:
packetdebug3.diff uploaded by wedhorn (license 30)
Tested by: mvanbaak, wedhorn
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Without this reorder tone will play after hangup and both wedhorn's and my wife have threatened to use an axe on our asterisk box
(closes issue #13948)
Reported by: wedhorn
Patches:
switch.diff uploaded by wedhorn (license 30)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Add CLI aliases module to asterisk.
- Remove all deprecated CLI commands from the code
Initial work done by file.
Junk-Y and lmadsen did a lot of work and testing to
get the list of deprecated commands into the configuration file.
Deprecated CLI commands are now handled by this new module,
see cli_aliases.conf for more info about that.
ok russellb@ via reviewboard
(closes issue #13735)
Reported by: mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(closes issue #13524)
Reported by: wedhorn
Patches:
unload.diff uploaded by wedhorn (license 30)
With small tweak by me to prevent a crash
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@143799 65c4cc65-6c06-0410-ace0-fbb531ad65f3
when a file is invalid from when a file is missing. This is most important when
we have two configuration files. Consider the following example:
Old system:
sip.conf users.conf Old result New result
======== ========== ========== ==========
Missing Missing SIP doesn't load SIP doesn't load
Missing OK SIP doesn't load SIP doesn't load
Missing Invalid SIP doesn't load SIP doesn't load
OK Missing SIP loads SIP loads
OK OK SIP loads SIP loads
OK Invalid SIP loads incompletely SIP doesn't load
Invalid Missing SIP doesn't load SIP doesn't load
Invalid OK SIP doesn't load SIP doesn't load
Invalid Invalid SIP doesn't load SIP doesn't load
So in the case when users.conf doesn't load because there's a typo that
disrupts the syntax, we may only partially load users, instead of failing with
an error, which may cause some calls not to get processed. Worse yet, the old
system would do this with no indication that anything was even wrong.
(closes issue #10690)
Reported by: dtyoo
Patches:
20080716__bug10690.diff.txt uploaded by Corydon76 (license 14)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This will print the subs and their status for every line (if any).
wedhorn did most of the work with his patch which introduced
'skinny show debug' but a discussion on IRC stated that it should be
added to 'skinny show lines'
Input on the output format by Qwell on IRC.
(closes issue #13344)
Reported by: wedhorn
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140938 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r136062 | mmichelson | 2008-08-06 10:58:40 -0500 (Wed, 06 Aug 2008) | 16 lines
Since adding the AST_CONTROL_SRCUPDATE frame type,
there are places where ast_rtp_new_source may be called
where the tech_pvt of a channel may not yet have an
rtp structure allocated. This caused a crash in chan_skinny,
which was fixed earlier, but now the same crash has been
reported against chan_h323 as well. It seems that the best
solution is to modify ast_rtp_new_source to not attempt to
set the marker bit if the rtp structure passed in is NULL.
This change to ast_rtp_new_source also allows the removal
of what is now a redundant pointer check from chan_skinny.
(closes issue #13247)
Reported by: pj
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136063 65c4cc65-6c06-0410-ace0-fbb531ad65f3
A lot of whitespace issues have been resolved in this commit
Also some doc updates, but that's only 6 lines
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@118824 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- make data member of the ast_frame struct a named union instead of a void
Recently the ast_queue_hangup function got a new parameter, the hangupcause
Feedback came in that this is no good and that instead a new function should be created.
This I did.
The hangupcause was stored in the seqno member of the ast_frame struct. This is not very
elegant, and since there's already a data member that one should be used.
Problem is, this member was a void *.
Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone
wants to store another type in there in the future.
This commit is so massive, because all ast_frame.data uses have to be
altered to ast_frame.data.data
Thanks russellb and kpfleming for the feedback.
(closes issue #12674)
Reported by: mvanbaak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
marked "urgent" are considered to be higher priority than other messages
and so they will be played before any other messages in a user's mailbox.
There are two ways to leave an urgent message.
1. send the 'U' option to VoiceMail().
2. Set review=yes in voicemail.conf. This will give instructions for
a caller to mark a message as urgent after the message has been recorded.
I have tested that this works correctly with file and ODBC storage, and James
Rothenberger (who wrote initial support for this feature) has tested its use
with IMAP storage.
(closes issue #11817)
Reported by: jaroth
Based on branch http://svn.digium.com/svn/asterisk/team/jrothenberger/asterisk-urgent
Tested by: putnopvut, jaroth
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@115588 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: oej
Tested by: jpeeler
This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option.
Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
when dialing a channel that does not provide progress (analog ZAP lines)
The phone does handle the double update on calls to channels that do
provide progress and wont insert duplicate items
(closes issue #12239)
Reported by: DEA
Patches:
chan_skinny-call-log.txt uploaded by DEA (license 3)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@109168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: rizzo
Tested by: murf
Proposal of the changes to be made, and then an announcement of how they were accomplished:
http://lists.digium.com/pipermail/asterisk-dev/2008-February/032065.html
and:
http://lists.digium.com/pipermail/asterisk-dev/2008-March/032124.html
Here is a recap, file by file, of what I have done:
pbx/pbx_config.c
pbx/pbx_ael.c
All funcs that were passed a ptr to the context list, now will ALSO be passed a hashtab ptr to the same set.
Why? because (for the time being), the dialplan is stored in both, to facilitate a quick, low-cost move to
hash-tables to speed up dialplan processing. If it was deemed necessary to pass the context LIST, well, it
is just as necessary to have the TABLE available. This is because the list/table in question might not be
the global one, but temporary ones we would use to stage the dialplan on, and then swap into the global
position when things are ready.
We now have one external function for apps to use, "ast_context_find_or_create()" instead of the pre-existing
"find" and "create", as all existing usages used both in tandem anyway.
pbx_config, and pbx_ael, will stage the reloaded dialplan into local lists and tables, and
then call merge_contexts_and_delete, which will merge (now) existing contexts and
priorities from other registrars into this local set by copying them. Then, merge_contexts_and_delete will
lock down the contexts, swap the lists and tables, and unlock (real quick), and then
destroy the old dialplan.
chan_sip.c
chan_iax.c
chan_skinny.c
All the channel drivers that would add regcontexts now use the ast_context_find_or_create now.
chan_sip also includes a small fix to get rid of warnings about removing priorities that never got entered.
apps/app_meetme.c
apps/app_dial.c
apps/app_queue.c
All the apps that added a context/exten/priority were also modified to use ast_context_find_or_create instead.
include/asterisk/pbx.h
ast_context_create() is removed. Find_or_create_ is the new method.
ast_context_find_or_create() interface gets the hashtab added.
ast_merge_contexts_and_delete() gets the local hashtab arg added.
ast_wrlock_contexts_version() is added so you can detect if someone else got a writelock between your readlocking and writelocking.
ast_hashtab_compare_contexts was made public for use in pbx_config/pbx_ael
ast_hashtab_hash_contexts was in like fashion make public.
include/asterisk/pval.h
ast_compile_ael2() interface changed to include the local hashtab table ptr.
main/features.c
For the sake of the parking context, we use ast_context_find_or_create().
main/pbx.c
I changed all the "tree" names to "table" instead. That's because the original
implementation was based on binary trees. (had a free library). Then I moved
to hashtabs. Now, the names move forward too.
refcount field added to contexts, so you can keep track of how many modules
wanted this context to exist.
Some log messages that are warnings were inflated from LOG_NOTICE to LOG_WARNING.
Added some calls to ast_verb(3,...) for debug messages
Lots of little mods to ast_context_remove_extension2, which is now excersized in ways
it was not previously; one definite bug fixed.
find_or_create was upgraded to handle both local lists/tables as well as the globals.
context_merge() was added to do the per-context merging of the old/present contexts/extens/prios into the new/proposed local list/tables
ast_merge_contexts_and_delete() was heavily modified.
ast_add_extension2() was also upgraded to handle changes.
the context_destroy() code was re-engineered to handle the new way of doing things,
by exten/prio instead of by context.
res/ael/pval.c
res/ael/ael.tab.c
res/ael/ael.tab.h
res/ael/ael.y
res/ael/ael_lex.c
res/ael/ael.flex
utils/ael_main.c
utils/extconf.c
utils/conf2ael.c
utils/Makefile
Had to change the interface to ast_compile_ael2(), to include the hashtab ptr.
This ended up involving several external apps. The main gotcha was I had to
include lock.h and hashtab.h in several places.
As a side note, I tested this stuff pretty thoroughly, I replicated the problems
originally reported by Luigi, and made triply sure that reloads worked, and everything
worked thru "stop gracefully". I found a and fixed a few bugs as I was merging into
trunk, that did not appear in my tests of bug6002.
How's this for verbose commit messages?
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106757 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(closes issue #8925)
About a year ago, as Leif Madsen and Jim van Meggelen were going over the CLI
commands in Asterisk 1.4 for the next version of their book, they documented
a lot of inconsistencies. This set of changes addresses all of these issues
and has been reviewed by Leif.
While this does introduce even more changes to the CLI command structure, it
makes everything consistent, which is the most important thing.
Thanks to all that helped with this one!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Note: NoAnswer support is currently not implemented, as it would take a
significant amount of work to figure out how to do correctly.
Closes issue #11310, patches, testing, and support by DEA, mvanbaak, and myself.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98776 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Refer to the proper documentation
- Implement separate signalling/media QoS/CoS in many channels using RTP
- Improve warnings and verbose messages
- Deprecate some old settings
Minor modifications by me, a big effort from IgorG.
Thanks!
Reported by: IgorG
Patches:
qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20)
Tested by: IgorG
(closes issue #11145)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
build times - tested, there is no measureable difference before and
after this commit.
In this change:
use asterisk/compat.h to include a small set of system headers:
inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h,
stdlib.h, alloca.h, stdio.h
Where available, the inclusion is conditional on HAVE_FOO_H as determined
by autoconf.
Normally, source files should not include any of the above system headers,
and instead use either "asterisk.h" or "asterisk/compat.h" which does it
better.
For the time being I have left alone second-level directories
(main/db1-ast, etc.).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This prevents modifying the strings in the stored variables,
and catched a few instances where this was actually done.
Given the differences between trunk and 1.4 (and the fact that this
is effectively an API change) it is better to fix 1.4 independently.
These are
chan_sip.c::sip_register()
chan_skinny.c:: near line 2847
config.c:: near line 1774
logger.c::make_components()
res_adsi.c:: near line 1049
I may have missed some instances for modules that do not build here.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Remove the AST_FORMAT_MAX_* types, as these are consuming 3 out of our available 32 bits.
- Add a native slin16 type, so that 16kHz codecs can translate without losing resolution.
(This doesn't affect anything immediately, until another codec has wb support.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
a patch for it. It replaces a bunch of simple calls to snprintf with ast_copy_string
(closes issue #10843)
Reported by: Corydon76
Patches:
2007092900_10843.diff uploaded by mvanbaak (license 7)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84173 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r79174 | file | 2007-08-13 11:18:04 -0300 (Mon, 13 Aug 2007) | 4 lines
(closes issue #10437)
Reported by: haklin
Don't set the callerid name and number a second time on a newly created channel. ast_channel_alloc itself already sets it and setting it twice would cause a memory leak.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79175 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #10299)
........
r77947 | qwell | 2007-08-02 13:42:36 -0500 (Thu, 02 Aug 2007) | 5 lines
Make sure we clear the prompt status message on a hangup.
Also rearrange messages to better fit with what a wireshark trace shows it should be.
Issue 10299, initial patch and solution by sbisker, modified by me to fit with wireshark trace.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77948 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #10358)
........
r77894 | qwell | 2007-08-02 10:15:45 -0500 (Thu, 02 Aug 2007) | 5 lines
Make sure that we show the correct extension if dialed from a macro
"From: 5555" rather than "From: s"
Issue 10358, initial patch by DEA, reworked by me to use S_OR, tested by sbisker
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77895 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r77883 | qwell | 2007-08-01 16:08:42 -0500 (Wed, 01 Aug 2007) | 7 lines
Fix an issue that caused one-way audio on some newer devices (specifically the 7921),
due to sending packets in the wrong order during hangup.
Also make sure we clear tones/messages on the correct line/instance.
Issue 10291, patch by DEA, tested by sbisker and myself.
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77884 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This allows you to just Dial(Skinny/line), as long as line isn't ambiguous.
Note that this does not remove or deprecate the "old" syntax, as it's still
quite useful - even moreso if shared lines get implemented.
Initial patch by me, with some changes and suggestions from wedhorn.
(closes issue #10263)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r67156 | qwell | 2007-06-04 18:26:28 -0500 (Mon, 04 Jun 2007) | 6 lines
Fix for skinny keepalives.
If there is no traffic from the phone for (keep_alive * 1100) ms (arbitrarily
adding 10% for network issues, etc), unregister the device.
Issue 8394, patch by DEA.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r66881 | russell | 2007-06-01 14:41:30 -0500 (Fri, 01 Jun 2007) | 6 lines
Changes to the way DTMF is handled in the core broke dialing in chan_skinny.
This patch makes chan_skinny usable again. I did not end up testing this,
but there are multiple positive test reports listed in the bug report.
(issue #9596, reported by pj, testing by pj and mvanbaak, and the fix was
written by DEA)
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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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See skinny.conf.sample for configuration example.
Note: Some devices (seen on 12SP+/30VIP) will lock up if they monitor too many hints.
This seems to be a hardware limitation - there isn't anything we can do about it.
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r56569 | qwell | 2007-02-23 20:02:53 -0600 (Fri, 23 Feb 2007) | 4 lines
Make sure to set a speeddials parent on creation.
Don't crash if hold is pressed when no call is active.
Don't return in places that we shouldn't..
Update softkey map when call is connected
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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Eliel C. Sardanons
Kevin P. Fleming
Joshua Colp
Mark Michelson
Russell Bryant
Michiel van Baak
Tilghman Lesher
Terry Wilson
Sean Bright
Brett Bryant
Jeff Peeler
Jason Parker
Steve Murphy
Olle Johansson
Luigi Rizzo