in 1.6.1 (as compared to 1.6.0 and versions of 1.4), this change also
deprecates the use of Asterisk with FreeBSD 4, given the central use of va_copy
in core functions. va_copy() is C99, anyway, and we already require C99 for
other purposes, so this isn't really a big change anyway. This change also
simplifies some of the core ast_str_* functions.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@157639 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r157305 | mmichelson | 2008-11-18 12:25:55 -0600 (Tue, 18 Nov 2008) | 12 lines
Fix a crash in the end_bridge_callback of app_dial and
app_followme which would occur at the end of an attended
transfer. The error occurred because we initially stored
a pointer to an ast_channel which then was hung up due
to a masquerade.
This commit adds a "fixup" callback to the bridge_config
structure to allow for end_bridge_callback_data to be
changed in the case that a new channel pointer is needed
for the end_bridge_callback.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@157306 f38db490-d61c-443f-a65b-d21fe96a405b
Reported by: alex70
Patches:
13420.13539.patch uploaded by murf (license 17)
Tested by: murf, awk
This fixes two problems: a spurious linefeed insertion
probably left over from pre-precomment times. Only
generated when category had no previous comments.
The other problem: Insertions could get the line-numbering
out of whack and generate negative line numbers, causing
chunks of line numbers to be emitted, on the scale of the
number of lines up to that point in the file. In such cases,
abort the looping, and all is well.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@157302 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r156688 | tilghman | 2008-11-13 15:24:00 -0600 (Thu, 13 Nov 2008) | 7 lines
Provide more space for all the data which can appear in an originating
channel name.
(closes issue #13398)
Reported by: bamby
Patches:
manager.c.diff uploaded by bamby (license 430)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@156690 f38db490-d61c-443f-a65b-d21fe96a405b
Reported by: smurfix
This reverts a change I made in 116297. At the time it seemed the change was required to solve an issue with attempting a transfer but then letting it timeout without dialing any digits. However, I didn't realize that having an empty extension was possible. I'm removing the immediate return that was added in pbx_find_extension if the extension is null.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@156649 f38db490-d61c-443f-a65b-d21fe96a405b
- MeetMe()
- MeetMeCount()
- MeetMeChannelAdmin()
- MeetMeAdmin()
- SLAStation()
- SLATrunk()
- Add an attribute to optionlist 'hasparams' with the same functionality as the one
we have in <parameter> and <argument> (the DTD was updated)
- Fix a leak when getting an attribute while parsing an <optionlist>.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@156575 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r156297 | murf | 2008-11-12 12:36:16 -0700 (Wed, 12 Nov 2008) | 18 lines
It turns out that the 0x0XX00 codes being returned for
N, X, and Z are off by one, as per conversation with
jsmith on #asterisk-dev; he was teaching a class
and disconcerted that this published rule was not
being followed, with patterns _NXX, _[1-8]22 and
_[2-9]22... and NXX was winning, but [1-8] should
have been.
This change, tested on these 3 patterns now
picks the proper one.
However, this change may surprise users who
set up dialplans based on previous behavior,
which has been there for what, 2 and half
years or so now.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@156299 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r156164 | russell | 2008-11-12 11:29:52 -0600 (Wed, 12 Nov 2008) | 7 lines
Move the sanity check that makes sure "always fork" is not set along with the
console option to be after the code that reads options from asterisk.conf.
This resolves a situation where Asterisk can start taking up 100% when
misconfigured.
(Thanks to Bryce Porter (x86 on IRC) for letting me log in to his system to
figure out what was causing the 100% CPU problem.)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@156166 f38db490-d61c-443f-a65b-d21fe96a405b
- 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: http://svn.digium.com/svn/asterisk/trunk@156120 f38db490-d61c-443f-a65b-d21fe96a405b
A new <agi> element is used to describe the XML documentation.
We have the usual synopsis,syntax,description and seealso for AGI commands.
The CLI 'agi show commands' command was changed to show all the documentation se
ctions.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@156051 f38db490-d61c-443f-a65b-d21fe96a405b
ast_channel_search_locked need to be made. Specifically, the caller needs to be able to
pass arbitrary data which in turn is passed to the callback. This patch addresses all
of the nested functions currently in asterisk trunk.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@155590 f38db490-d61c-443f-a65b-d21fe96a405b
ao2_callback and ao2_find). Currently, passing OBJ_POINTER to either
of these mandates that the passed 'arg' is a hashable object, making
searching for an ao2 object based on outside criteria difficult.
Reviewed by Russell and Mark M. via ReviewBoard:
http://reviewboard.digium.com/r/36/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@155401 f38db490-d61c-443f-a65b-d21fe96a405b
container. The OBJ_MULTIPLE flag must be provided here. Otherwise, this would
only remove a single object.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@155241 f38db490-d61c-443f-a65b-d21fe96a405b
Functions are printed without parenthesis like: FUNTION
Applications are printed with parenthesis like: AppName()
Cli commands are printed like: 'core show application'
The other type of references are printed as they are inside the <ref> tag.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@154967 f38db490-d61c-443f-a65b-d21fe96a405b
channel list calling a caller-defined callback. The callback returns non-zero
if a match is found. This should speed up some of the code that I committed
earlier today in chan_sip (which is also updated by this commit).
Reviewed by russellb and kpfleming via ReviewBoard:
http://reviewboard.digium.com/r/28/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@154429 f38db490-d61c-443f-a65b-d21fe96a405b
This commit introduces the first phase of an effort to manage documentation of the
interfaces in Asterisk in an XML format. Currently, a new format is available for
applications and dialplan functions. A good number of conversions to the new format
are also included.
For more information, see the following message to asterisk-dev:
http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153365 f38db490-d61c-443f-a65b-d21fe96a405b
had no effect
* Updated dialing API documentation to indicate that timeouts
are specified in milliseconds
* Added a new timeout argument to the Page application. If time
expires, any endpoints which have not answered will be hung up.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153223 f38db490-d61c-443f-a65b-d21fe96a405b
the low default value of featuredigittimeout, I decided it
was high time to change it. I have changed the default to
2000 ms based on a suggestion from Leif Madsen.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@152807 f38db490-d61c-443f-a65b-d21fe96a405b
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines
The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.
If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.
If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.
Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden
(in trunk).
All the places that previously tested for
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.
I tested this against the 4 common parking
scenarios:
1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.
2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.
3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.
4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.
No crash.
I also ran the scenarios above against valgrind, and accesses looked good.
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@152536 f38db490-d61c-443f-a65b-d21fe96a405b
tilghman