they do similar things.
Add a small form on top of the html output so request like
http://foo:8088/asterisk/manager will suggest you what to do.
Note: i suspect there is still a bug somewhere in the session matching
code, as sometimes you have to login twice in order for the following
commands to be recognised.
Apart from this, the cli now is basically usable from a web form!
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45475 65c4cc65-6c06-0410-ace0-fbb531ad65f3
minor optimizations to avoid extra calls of strlen(),
and some variable localization.
One feature worth backporting is the move of ast_variables_destroy()
to a different place in handle_uri() to avoid leaking memory
in case a uri is not found.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45463 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r45441 | russell | 2006-10-17 22:41:36 -0400 (Tue, 17 Oct 2006) | 7 lines
Don't attempt to access private data members of the pthread_mutex_t object,
because this does not work on all linux systems. Instead, just access
the reentrancy field in the ast_mutex_info struct when DEBUG_THREADS is
enabled. If DEBUG_CHANNEL_LOCKS is enabled, the developer probably has
DEBUG_THREADS on as well.
(issue #8139, me)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45409 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Normal responses are sequences of lines of the form "Name: value",
with \r\n as line terminators and an empty line as a response
terminator.
Generi CLI commands, however, do not have such a clean formatting,
and the existing code failed to generate valid XML for them.
Obviously we can only use heuristics here, and we do the following:
- accept either \r or \n as a line terminator, trimming trailing whitespace;
- if a line does not have a ":" in it, assume that from this point on
we have unformatted data, and use "Opaque-data:" as a name;
- if a line does have a ":" in it, the Name field is not always
a legal identifier, so replace non-alphanum characters with underscores;
All the above is to be refined as we improve the formatting of
responses from the CLI.
And, all the above ought to go as a comment in the code rather than
just in a commit message...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45334 65c4cc65-6c06-0410-ace0-fbb531ad65f3
invoked through the http interface.
It is not terribly efficient but better than no output at all.
Todo: use a configurable /tmp directory instead of a hardwired one.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45330 65c4cc65-6c06-0410-ace0-fbb531ad65f3
replace non-alphanum chars with underscore.
This is useful when building field names in xml formatting.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
+ let some commands (Challenge, Login) be processed even if
already authenticated, as it doesn't harm and prevents some
incorrect error messages
+ remove custom code for Logoff - the existing handler was ok.
Some indentation fixes may be necessary
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
+ remove the need for an snprintf in astman_get_header()
+ fix comment for manager list eventq
+ localize one variable and minor code simplifications.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45185 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Mark with XXX one place (during command execution) where
navigation should be protected with actionlock, but is not
because it would block requests for a long time.
To solve this properly we need to put reference counts in
the struct manager_action.
A suboptimal fix is to copy the record on a search and then
unlock the list while we work on the copy.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45177 65c4cc65-6c06-0410-ace0-fbb531ad65f3
On passing, small rearrangement of the code to reduce indentation.
There is a bit more cleanup planned for this file, so a merge to 1.4
will be done when it is all done.
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r45125 | kpfleming | 2006-10-13 18:02:48 -0500 (Fri, 13 Oct 2006) | 7 lines
------------------------------------------------------------------------
r45119 | kpfleming | 2006-10-13 17:57:42 -0500 (Fri, 13 Oct 2006) | 2 lines
don't drop the entire permit/deny list when an attempt is made to add an invalid entry (BE-92)
------------------------------------------------------------------------
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@45126 65c4cc65-6c06-0410-ace0-fbb531ad65f3
simplifications. On passing, use a single exit point.
(once done with the cleanup i will merge the changes into 1.4,
if applicable)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44379 65c4cc65-6c06-0410-ace0-fbb531ad65f3
revert Mark's change that caused a memory leak (cap_set_proc() does not free the capability structure so we always need to call cap_free())
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44152 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- with AST_DEVMODE, building codecs/lpc10 fails because of lots
of warnings, and the configure step in editline fails as well.
Fix this by removing the -Werror in these steps.
- on FreeBSD (but probably on other platforms as well), the final
link of asterisk fails because AST_LIBS was not exported to the
subdirs Makefiles. Add a proper fix in the top-level Makefile
(a possible alternative way is to add "export AST_LIBS" near
the beginning of the file).
With this fix, i believe that some of the platform-specific
conditionals in main/Makefile are redundant (because they should
be already dealt with in the top level Makefile) but i don't
have a platform to check.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@44081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43635 | pcadach | 2006-09-26 03:26:12 +0600 (Втр, 26 Сен 2006) | 1 line
Fix ASN1 description of non-standard Cisco extensions
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r43843 | pcadach | 2006-09-28 12:01:37 +0600 (Чтв, 28 Сен 2006) | 1 line
Don't treat unknown control frames as voice
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r43844 | pcadach | 2006-09-28 12:02:45 +0600 (Чтв, 28 Сен 2006) | 1 line
Don't warn on HOLD/UNHOLD control frames
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r43846 | pcadach | 2006-09-28 16:51:21 +0600 (Чтв, 28 Сен 2006) | 1 line
Do not open transmit channel until TCS is received
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r43803 | qwell | 2006-09-27 12:44:02 -0700 (Wed, 27 Sep 2006) | 4 lines
Fix an issue with PLAYBACKSTATUS not being set under certain circumstances.
Fix a minor issue, to make it use the filenames that were parsed, instead of the entire argument string.
Fix Background() to return -1 like Playback(), if no args are specified.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43804 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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r43779 | russell | 2006-09-27 12:55:49 -0400 (Wed, 27 Sep 2006) | 50 lines
Merged revisions 43778 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43778 | russell | 2006-09-27 12:54:30 -0400 (Wed, 27 Sep 2006) | 42 lines
Fix a problem that occurred if a user entered a digit that matched a bridge
feature that was configured using multiple digits, and the digit that was
pressed timed out in the feature digit timeout period. For example, if blind
transfer is configured as '##', and a user presses just '#'. In this situation,
the call would lock up and no longer pass any frames.
(issue #7977 reported by festr, and issue #7982 reported by michaels and
valuable input provided by mneuhauser and kuj. Fixed by me, with testing help
and peer review from Joshua Colp).
There are a couple of issues involved in this fix:
1) When ast_generic_bridge determines that there has been a timeout, it returned
AST_BRIDGE_RETRY. Then, when ast_channel_bridge gets this result, it calls
ast_generic_bridge over again with the same timestamp for the next event.
This results in an endless loop of nothing until the call is terminated.
This is resolved by simply changing ast_generic_bridge to return
AST_BRIDGE_COMPLETE when it sees a timeout.
2) I also changed ast_channel_bridge such that if in the process of calculating
the time until the next event, it knows a timeout has already occured, to
immediately return AST_BRIDGE_COMPLETE instead of attempting to bridge the
channels anyway.
3) In the process of testing the previous two changes, I ran into a problem in
res_features where ast_channel_bridge would return because it determined
that there was a timeout. However, ast_bridge_call in res_features would
then determine by its own calculation that there was still 1 ms before the
timeout really occurs. It would then proceed, and since the bridge broke
out and did *not* return a frame, it interpreted this as the call was over
and hung up the channels.
The reason for this was because ast_bridge_call in res_features and
ast_channel_bridge in channel.c were using different times for their
calculations. channel.c uses the start_time on the bridge config, which
is the time that the feature digit was recieved. However, res_features
had another time, 'start', which was set right before calling
ast_channel_bridge. 'start' will always be slightly after start_time in the
bridge config, and sometimes enough to round up to one ms.
This is fixed by making ast_bridge_call use the same time as
ast_channel_bridge for the timeout calculation.
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r43710 | russell | 2006-09-26 16:56:42 -0400 (Tue, 26 Sep 2006) | 17 lines
(This was actually BE-65)
Merged revisions 43708 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r43708 | russell | 2006-09-26 16:49:21 -0400 (Tue, 26 Sep 2006) | 7 lines
Back in revision 4798, this message was changed from using ast_cli() to directly
calling write(). During this change, checking if this was a remote console was
removed. This caused this message about using "exit" or "quit" to exit an
Asterisk console to come up in times where it did not make sense. This change
restores the check to see if this is a remote console before printing the
message. (fixes BE-4)
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r43695 | file | 2006-09-26 16:09:41 -0400 (Tue, 26 Sep 2006) | 2 lines
Slight overhaul of the whisper support. 1. We need to duplicate the frame from ast_translate 2. We need to ensure we always have signed linear coming in for signed linear combining. 3. We need to ensure we are always feeding signed linear out. 4. Properly store and restore write format when beeping on the channel we are whispering on. 5. Properly discontinue the stream on the channel for the beep. (issue #8019 reported by timkelly1980)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Original patch by junky, modified by anthonyl, modified again by jcolp (with minor modifications by me)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43300 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Luigi Rizzo
Joshua Colp
Russell Bryant
Kevin P. Fleming
Nadi Sarrar
Jason Parker
Paul Cadach
Steve Murphy
Matt O'Gorman
Mark Spencer
Tilghman Lesher
Anthony LaMantia
Matthew Fredrickson