https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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r48273 dealt with the comments and such, this deals with the code itself.
(This couldn't have been easily done if it weren't for 48273 - thanks again for that merbanan)
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implementing T.140 support in RTP.
T.140/RFC 4351 is TDD over IP - text telephony for hearing impaired.
It defines a realtime text chat, much like the old "talk" application
in Unix.
T.140 is character by character in real time. It's not
the same as our current MESSAGE format - that is more like IM, but
can be gatewayed to MESSAGE with a text "codec" if needed.
More patches will follow, as soon as we've separated this code from
the video capabilities functions in the videocaps branch.
Code by John Martin, Aupix (disclaimer on file)
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- Encapsulate RTP timers to the RTP structure, so we have one set for video and one for audio
- Document RTP keepalive configuration option
- Cleanup and document the monitor support function to hangup on RTP timeouts
- Add RTP keepalive to SIP show settings
Imported from 1.4 with modifications for trunk.
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- Add some more documentation for the ast_dynamic_str_............() function
to document the behavior of the function in the case of a partial write.
Also, document the return value and note that the function should never need
to be called directly.
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as well as what the motivation is for using it.
- Add a comment by the declaration of ast_inet_ntoa() noting that this function
is not reentrant, and the result of a previous call to the function is no
longer valid after calling it again.
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- Document cause codes
- Document a bit more on channel variables - global, predefined and local
- Fix some doxygen in channel.h. Adding one comment for two definitions does not
work. They won't be copied to each.
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without duplicating the macro or the code:
/*!
* In many cases we need to print singular or plural
* words depending on a count. This macro helps us e.g.
* printf("we have %d object%s", n, ESS(n));
*/
#define ESS(x) ((x) == 1 ? "" : "s")
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Now you can specify a cli command as
"console autoanswer [on|off]"
which means the on|off argument is optional, or
"console {mute|unmute}"
which means the mute|unmute argument is mandatory.
The blocks in [] or {} do not necessarily need to be at the
end of the string.
Completions for the variant parts are generated automatically.
This should significantly simplify the implementation of
the various handlers.
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the NEW_CLI macro now supports extra arguments (to deprecate other commands).
use this to implement unload and reload, and remove some unused functions.
usual completion fixes (as these function accept multiple arguments).
The summary is still a bit inconsistent.
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This change basically simplifies the interface of the
new-style handler removing almost all the tricks used in
the previous implementation to achieve backward compatibility
(which is still present and guaranteed.)
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WATCH OUT: this changes the binary interface (ABI) for modules,
so e.g. users of g729 codecs need a rebuilt module (but read below).
The new way to write CLI handlers is described in detail in cli.h,
and there are a few converted handlers in cli.c, look for NEW_CLI.
After converting a couple of commands i am convinced that
it is reasonably convenient to use, and it makes it easier to fix the
pending CLI issues.
On passing, note a bug with the current 'complete' architecture:
if a command is a prefix of multiple CLI entries, we miss some
of the possible options. As an example, "core set debug" can
continue with "channel" from one CLI entry, and "off" or "atleast"
from another one.
We address this problem in a separate commit
(when i have figured out a fix, that is).
ABI issues:
I asked Kevin if it was ok to make this change and he said yes.
While it would have been possible to make the change without breaking
the module ABI, the code would have been more convoluted.
I am happy to restore the old ABI (while still being able
to use the "new style" handlers) if there is demand.
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debugging code in the manager.
At the moment the debugging code is very lightweight, if the option
is enabled manager messages also carry a sequence number and
the info where they have been generated e.g.
SequenceNumber: 10
File: chan_sip.c
Line: 11927
Func: handle_response_register
It is not worthwhile having this as a compile time option
right now, because the extra work involved at runtime is
just checking one variable.
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various PBX installations and checks if a module is loaded before using
it.
example IFMODULE(chan_sip3.so)
issue #6671 in the bug tracker, finally gone. Thanks to mithraen for keeping
it updated.
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r46082 | kpfleming | 2006-10-23 22:45:42 -0500 (Mon, 23 Oct 2006) | 2 lines
add an API call to allow channel drivers to determine which media formats are compatible (passthrough or transcode) with the format an existing channel is already using
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r46083 | kpfleming | 2006-10-23 22:53:32 -0500 (Mon, 23 Oct 2006) | 2 lines
ensure that the translation matrix is properly lock-protected every place it is used
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r46152 | kpfleming | 2006-10-24 18:45:19 -0500 (Tue, 24 Oct 2006) | 2 lines
if multiple translators are registered for the same source/dest combination, ensure that the lowest-cost one is always inserted earlier in the list
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r46153 | kpfleming | 2006-10-24 19:10:54 -0500 (Tue, 24 Oct 2006) | 2 lines
code zone experiment: don't offer formats in the outbound INVITE that aren't either passthrough or translatable
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allow custom threadstorage init functions to return failure
use a custom init function for chan_sip's temp_pvt, to improve performance a bit
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be called for each thread specific object after they are allocated. Note that
there was already the ability to define a custom cleanup function. Also, if
the custom cleanup function is used, it *MUST* call free on the thread
specific object at the end. There is no way to have this magically done that
I can think of because the cleanup function registered with the pthread
implementation will only call the function back with a pointer to the
thread specific object, not the parent ast_threadstorage object.
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r45408 | kpfleming | 2006-10-17 17:24:10 -0500 (Tue, 17 Oct 2006) | 3 lines
optimize the 'quick response' code a bit more... no more malloc() or memset() for each response
expand stringfields API a bit to allow reusing the stringfield pool on a structure when needed, and remove some unnecessary code when the structure was being freed
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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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r43861 | pcadach | 2006-09-28 18:47:23 +0600 (Чтв, 28 Сен 2006) | 1 line
Put attribute tag at correct place
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r43862 | pcadach | 2006-09-28 18:58:22 +0600 (Чтв, 28 Сен 2006) | 1 line
Force remote side to start media on outgoing PROGRESS message
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
use "attribute_XXX" instead of *__attribute__ ((XXX)) so we
can handle compiler/os dependencies in our compiler.h
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Russell Bryant
Joshua Colp
Olle Johansson
Kevin P. Fleming
Luigi Rizzo
Tilghman Lesher
Steve Murphy
Jason Parker
Paul Cadach
TransNexus OSP Development
Matt O'Gorman
Matthew Fredrickson