Under FreeBSD, the filename_completion used in the above commands does
not work. Not sure why, but on passing i note that the function is
part of readline and is not reentrant, so it needs to be fixed one way
or another.
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code for the deprecated handler.
On passing fix a long standing bug in find_cli() which would not
find the longest match - this part (trivial, basically
just update matchlen on a match) must go in 1.4 and possibly 1.2 as well.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47662 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47654 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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original one instead with appropriate changes in argc/argv.
This is not always applicable, but in some simple cases it is.
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and fix the simple case where a command can have multiple completions,
the first ones coming from keywords in previous CLI entries.
There is no solution yet for the complex case of N1 completions
from a CLI entry, followed by N2 from the next one, and so on,
because the _complete() handlers do not tell us how many matches
it generates, so we don't know how many to skip when interrogating
the other handlers.
The only solution is to avoid, as much as possible, multiple
CLI entries with the same prefix.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47607 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47132 65c4cc65-6c06-0410-ace0-fbb531ad65f3
allocated on the stack using alloca() instead of using malloc() since
they are only used locally to these functions.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46885 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46780 | file | 2006-11-01 13:39:47 -0500 (Wed, 01 Nov 2006) | 2 lines
Force poll() emulation for Darwin to always be on. It's too broken to consider being used. This resolves the console issue OSX users have been seeing. I would have liked to autoconf this but I haven't been able to come up with a test case that works. Que sera.
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- Check the channel name string length to be zero, not non-zero
- Check the message string length to be zero, not non-zero
- unlock the channel *after* calling sendtext
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46661 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Might need better locking support, giving up if we can't get the lock. Right now,
using existing locking in find_resource
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46554 | russell | 2006-10-31 00:55:07 -0500 (Tue, 31 Oct 2006) | 5 lines
Add a small tweak to the code that checks to see whether destination formats
are translatable based on the source format. If we have already determined
that there is no translation path in one direction, don't bother checking the
other direction.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46526 | kpfleming | 2006-10-30 16:19:55 -0600 (Mon, 30 Oct 2006) | 3 lines
when unregistering a translator, don't rebuild the translation matrix unless needed
when filtering formats out of an offer, ensure we check for translation ability in both directions
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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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I currently don't see this as a bug that needs to be fixed in 1.4/1.2 too,
but feel free to backport if you see it that way. RTCP now binds to
ALL IP addresses on the host, RTP to a specific address.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46363 | russell | 2006-10-27 12:39:31 -0500 (Fri, 27 Oct 2006) | 5 lines
We should always be using _exit() after a fork() or vfork() instead of exit().
This is because exit() does some extra cleanup which in some implementations
of vfork(), for example, can actually modify the state of the parent process,
causing very weird bugs or crashes. (issue #7971, Nick Gavrikov)
........
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r46329 | russell | 2006-10-26 11:31:05 -0500 (Thu, 26 Oct 2006) | 11 lines
- If the source has no audio or no video portion, do not call powerof() to
get the format index.
- Don't run through the audio and video loops if there is no audio or video
portion of the source
If 0 is passed to powerof, it will return -1. This value of -1 was then being
used as an array index in these loops, which caused a crash on some systems.
Other than this issue, this code works as we expected it to. If a format is
not in the source, and we have to translation path to it, it is not offered in
the list of acceptable destination formats.
(fixes issue #8231)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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simplifies the manager queue handling as described in
the comment, and will make a lot easier to make further
work on this code.
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and support linux as well (using fopencookie(), which should
be available in glibc).
Update configure.ac to check for funopen (BSD) and fopencookie(glibc),
and while we are at it also for gethostbyname_r
(the generated files need to be updated, or you need
to run bootstrap.sh yourself).
Document the new options in http.conf.sample
(names are only tentative, better ones are welcome).
At this point we can safely enable the option.
Anyone willing to try this on Sun and Apple platforms ?
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The changes are not large. Most of the diff comes from putting the
global variables describing an accept session into a structure, so
we can reuse the existing code for running multiple accept threads
on different ports.
Once this is done, and if your system has the funopen() library
function (and ssl, of course), it is just a matter of calling
the appropriate functions to set up the ssl connection on the
existing socket, and everything works on the secure channel now.
At the moment, the code is disabled because i have not implemented yet
the autoconf code to detect the presence of funopen(), and add -lssl
to main/Makefile if ssl libraries are present. And a bit of documentation
on the http.conf arguments, too.
If you want to manually enable https support, that is very simple
(step 0 1 2 will be eventually detected by ./configure, the
rest is something you will have to do anyways).
0. make sure your system has funopen(3). FreeBSD does, linux probably
does too, not sure about other systems.
1. uncomment the following line in main/http.c
// #define DO_SSL /* comment in/out if you want to support ssl */
2. add -lssl to AST_LIBS in main/Makefile
3. add the following options to http.conf
sslenable=yes
sslbindport=4433 ; pick one you like
sslcert=/tmp/foo.pem ; path to your certificate file.
4. generate a suitable certificate e.g. (example from mini_httpd's Makefile:
openssl req -new -x509 -days 365 -nodes -out /tmp/foo.pem -keyout /tmp/foo.pem
and here you go:
https://localhost:4433/asterisk/manager
now works.
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reply back to http clients.
Use fprintf/fwrite instead, since we are already using a FILE *
to read the input.
If you wonder why, this is because it makes it trivial to
implement https support (as long as your system has funopen()).
And this is what i am going to put in with the next few commits...
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to minimize diffs with future modifications.
The current implementation is problematic for the following reasons:
+ all insertions are O(N) because the event list does not have a tail
pointer;
+ there is only a single lock protecting both session and users queues.
+ the implementation of the queue itself is not documented.
I think i have figured it out, more or less, but am unclear on
whether there is proper locking in place
The rewrite (which i have working locally) uses a tailq so insertions
are O(1), separate locks for the event and session queues, and has
a documented implementation so hopefully we can figure out if/where
bug exist.
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Of interest:
+ ast_get_manager_by_name_locked() is now without the ast_
prefix as it is a local function;
+ unuse_eventqent() renamed to unref_event(), and returns
the pointer to the next entry.
+ marked with XXX a couple of usages of unref_event()
because i suspect we are addressing the wrong entry.
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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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+ more comments on struct mansession and global variables;
+ small improvements to the session matching code so it supports
multiple sessions from the same IP
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around so that functions belonging to the same group are
close to each other.
At the beginning of each group i have added a bit of documentation
to explain what the group does and what is the typical flow - basically,
all i have learned by code inspection over the past few days should
be documented for you to read.
I have not put many doxygen annotations just because i am not
sure what are the proper ones. Hopefully some doxygen experts will jump in.
Next on the plate: try to figure out how "struct eventqent"
are supposed to work.
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On passing, fix a bug: close the socket if the allocation
of a structure for the new session fails.
(the bugfix is a candidate for 1.4)
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to start manager responses that need further lines.
This removes a lot of duplicate code from the various handlers
that at the moment build an ActionID string themselves.
Once settled, the function should move to manager.h so
it can be used by other files (chan_agent, chan_iax2, chan_sip,
chan_zap, res_jabber and app_queue).
I am not totally clear if there is a preferred position for
the ActionID: line in a message. Some instances put it at
the end, but one would argue that it is preferable to have
it at the beginning.
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rather than use inline code for them.
Things are more readable this way, and also error processing
is more consistent.
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(they are committed a bit at a time so it is easier to
revert them in case we find a bug at a later time).
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of the file, so it can be used from more places;
+ make the declaration of contenttype[] more robust;
+ remove the wrappers around __xml_translate(), since they were
used only in one place, and rename to xml_translate().
This allows for a bit of simplifications.
+ document the output produced by the above function.
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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!
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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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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...
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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.
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replace non-alphanum chars with underscore.
This is useful when building field names in xml formatting.
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+ 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
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+ remove the need for an snprintf in astman_get_header()
+ fix comment for manager list eventq
+ localize one variable and minor code simplifications.
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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.
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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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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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())
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- 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.
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43780 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@43715 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
