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Hitting tab without typing any part of a channel name resulted in no results.
This now results in getting a full list of active channels, just as it did
in previous versions of Asterisk.
Review: https://reviewboard.asterisk.org/r/818/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@281529 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@280020 f38db490-d61c-443f-a65b-d21fe96a405b
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The version of libedit that is bundled with asterisk is old and has some bugs.
This patch updates the bundled version of libedit within asterisk, and also
updates asterisk to use the system libedit instead if one is available (and
pkg-config is available). This review integrates several patches from other
users specifically kkm and tzafrir.
(closes issue #15929)
Reported by: kkm
Patches:
015929-astcli-editrc-trunk.240324.diff uploaded by kkm (license 888)
(issue #16858)
Reported by: jw-asterisk
(closes issue #17039)
Reported by: tzafrir
Patches:
0001-allow-using-system-copy-of-libedit.patch uploaded by tzafrir (license 46)
Review: https://reviewboard.asterisk.org/r/807/
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.8@280019 f38db490-d61c-443f-a65b-d21fe96a405b
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The purpose of this patch is to eliminate struct ast_callerid since it has
turned into a miscellaneous collection of various party information.
Eliminate struct ast_callerid and replace it with the following struct
organization:
struct ast_party_name {
char *str;
int char_set;
int presentation;
unsigned char valid;
};
struct ast_party_number {
char *str;
int plan;
int presentation;
unsigned char valid;
};
struct ast_party_subaddress {
char *str;
int type;
unsigned char odd_even_indicator;
unsigned char valid;
};
struct ast_party_id {
struct ast_party_name name;
struct ast_party_number number;
struct ast_party_subaddress subaddress;
char *tag;
};
struct ast_party_dialed {
struct {
char *str;
int plan;
} number;
struct ast_party_subaddress subaddress;
int transit_network_select;
};
struct ast_party_caller {
struct ast_party_id id;
char *ani;
int ani2;
};
The new organization adds some new information as well.
* The party name and number now have their own presentation value that can
be manipulated independently. ISDN supplies the presentation value for
the name and number at different times with the possibility that they
could be different.
* The party name and number now have a valid flag. Before this change the
name or number string could be empty if the presentation were restricted.
Most channel drivers assume that the name or number is then simply not
available instead of indicating that the name or number was restricted.
* The party name now has a character set value. SIP and Q.SIG have the
ability to indicate what character set a name string is using so it could
be presented properly.
* The dialed party now has a numbering plan value that could be useful to
have available.
The various channel drivers will need to be updated to support the new
core features as needed. They have simply been converted to supply
current functionality at this time.
The following items of note were either corrected or enhanced:
* The CONNECTEDLINE() and REDIRECTING() dialplan functions were
consolidated into func_callerid.c to share party id handling code.
* CALLERPRES() is now deprecated because the name and number have their
own presentation values.
* Fixed app_alarmreceiver.c write_metadata(). The workstring[] could
contain garbage. It also can only contain the caller id number so using
ast_callerid_parse() on it is silly. There was also a typo in the
CALLERNAME if test.
* Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id
number string. ast_callerid_parse() alters the given buffer which in this
case is the channel's caller id number string. Then using
ast_shrink_phone_number() could alter it even more.
* Fixed caller ID name and number memory leak in chan_usbradio.c.
* Fixed uninitialized char arrays cid_num[] and cid_name[] in
sig_analog.c.
* Protected access to a caller channel with lock in chan_sip.c.
* Clarified intent of code in app_meetme.c sla_ring_station() and
dial_trunk(). Also made save all caller ID data instead of just the name
and number strings.
* Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge()
function.
* Corrected some weirdness with app_privacy.c's use of caller
presentation.
Review: https://reviewboard.asterisk.org/r/702/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
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Review: https://reviewboard.asterisk.org/r/684/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267138 f38db490-d61c-443f-a65b-d21fe96a405b
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(issue #16009)
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Added the keyword 'all' to the 'channel hangup request' CLI command
so that you can request all channels to be hungup without having to
restart Asterisk.
(closes issue #16009)
Reported by: moy
Patches:
hangup-all-rev-221688.patch uploaded by moy (license 222)
Tested by: moy, russell
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@264117 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17308)
Reported by: pabelanger
Patches:
cli.modules.patch uploaded by pabelanger (license 224)
Tested by: pabelanger, russell
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@262800 f38db490-d61c-443f-a65b-d21fe96a405b
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users expect them to work.
'core set debug' and 'core set verbose' can optionally change the
level for a specific filename; however, this is actually for a
specific source file name, not the module that source file is included
in. With examples like chan_sip, chan_iax2, chan_misdn and others
consisting of multiple source files, this will not lead to the
behavior that users expect. If they want to set the debug level for
chan_sip, they want it set for all of chan_sip, and not to have to
also set it for reqresp_parser and other files that comprise the
chan_sip module.
This patch changes this functionality to be module-name based instead
of file-name based.
To make this work, some Makefile modifications were required to ensure
that the AST_MODULE definition is present in each object file produced
for each module as well.
Review: https://reviewboard.asterisk.org/r/574/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@253917 f38db490-d61c-443f-a65b-d21fe96a405b
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Modified handle_verbose to be LOW_MEMORY aware, removed old RTP related code
in chan_sip.
(closes issue #16381)
Reported by: michael_iedema
Patches:
ast_complete_source_filename.patch uploaded by michael iedema (license 942)
modified by me
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@236893 f38db490-d61c-443f-a65b-d21fe96a405b
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ast_channel_iterator to use it.
This patch finishes the implementation of OBJ_MULTIPLE in astobj2 (the
case where multiple results need to be returned; OBJ_NODATA mode
already was supported). In addition, it converts ast_channel_iterators
(only the targeted versions, not the ones that iterate over all
channels) to use this method.
During this work, I removed the 'ao2_flags' arguments to the
ast_channel_iterator constructor functions; there were no uses of that
argument yet, there is only one possible flag to pass, and it made the
iterators less 'opaque'. If at some point in the future someone really
needs an ast_channel_iterator that does not lock the container, we can
provide constructor(s) for that purpose.
Review: https://reviewboard.asterisk.org/r/379/
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Requiring 'module reload' to reload everything, including
core etc makes russell very unhappy.
The default configuration already loads the 'friendly' aliases template.
Added 'reload=module reload' to that template.
Also removed the comment in main/cli.c that reload should come back.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@208813 f38db490-d61c-443f-a65b-d21fe96a405b
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I keep getting annoyed at having to type "module reload" to reload everything,
so I'm adding a note that we need to add "reload" back. "module reload" doesn't
really make sense as the command to reload everything, including the core.
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CEL is the new system for logging channel events. This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records. For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.
Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code. Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.
Review: https://reviewboard.asterisk.org/r/239/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@203638 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r203380 | twilson | 2009-06-25 16:13:10 -0500 (Thu, 25 Jun 2009) | 4 lines
I didn't see that Mark already fixed the underlying issue!
Yay for removing useless code.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r203311 | twilson | 2009-06-25 15:09:15 -0500 (Thu, 25 Jun 2009) | 2 lines
Don't try to free NULL
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The 'pglobal' tool is quite handy indeed :-)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@200620 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #14833)
Reported by: contactmayankjain
Patches:
all_changes.patch uploaded by contactmayankjain (license 740)
slightly modified by me
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@197616 f38db490-d61c-443f-a65b-d21fe96a405b
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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: http://svn.digium.com/svn/asterisk/trunk@196072 f38db490-d61c-443f-a65b-d21fe96a405b
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There is a lot that could be said about this, but the patch is a big
improvement for performance, stability, code maintainability,
and ease of future code development.
The channel list is no longer an unsorted linked list. The main container
for channels is an astobj2 hash table. All of the code related to searching
for channels or iterating active channels has been rewritten. Let n be
the number of active channels. Iterating the channel list has gone from
O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1).
Searching for a channel by extension is still O(n), but uses a new method
for doing so, which is more efficient.
The ast_channel object is now a reference counted object. The benefits
here are plentiful. Some benefits directly related to issues in the
previous code include:
1) When threads other than the channel thread owning a channel wanted
access to a channel, it had to hold the lock on it to ensure that it didn't
go away. This is no longer a requirement. Holding a reference is
sufficient.
2) There are places that now require less dealing with channel locks.
3) There are places where channel locks are held for much shorter periods
of time.
4) There are places where dealing with more than one channel at a time becomes
_MUCH_ easier. ChanSpy is a great example of this. Writing code in the
future that deals with multiple channels will be much easier.
Some additional information regarding channel locking and reference count
handling can be found in channel.h, where a new section has been added that
discusses some of the rules associated with it.
Mark Michelson also assisted with the development of this patch. He did the
conversion of ChanSpy and introduced a new API, ast_autochan, which makes it
much easier to deal with holding on to a channel pointer for an extended period
of time and having it get automatically updated if the channel gets masqueraded.
Mark was also a huge help in the code review process.
Thanks to David Vossel for his assistance with this branch, as well. David
did the conversion of the DAHDIScan application by making it become a wrapper
for ChanSpy internally.
The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch.
Review: http://reviewboard.digium.com/r/203/
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@190423 f38db490-d61c-443f-a65b-d21fe96a405b
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There seems to be a bug with old versions of g++ that doesn't allow a structure
member to use the name list. Rename list member to group_list in ast_group_info
and change the few places it is used.
(closes issue #14790)
Reported by: stuarth
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(closes issue #14219)
Reported by: jamesgolovich
Patches:
asterisk-setverbosecrash.diff.txt uploaded by jamesgolovich (license 176)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@168626 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #13969)
Reported by: jtodd
Patches:
20081205__bug13969.diff.txt uploaded by Corydon76 (license 14)
Tested by: mvanbaak, eliel
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(closes issue #13990)
Reported by: eliel
Patches:
array_len.diff uploaded by eliel (license 64)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@161218 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r160207 | tilghman | 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008) | 3 lines
Ensure that Asterisk builds with --enable-dev-mode, even on the latest gcc
and glibc.
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Based on cli_permissions.conf configuration file, we are able to permit or deny
cli commands based on some patterns and the local user and group running rasterisk.
(Sorry if I missed some of the testers).
Reviewboard: http://reviewboard.digium.com/r/11/
(closes issue #11123)
Reported by: eliel
Tested by: eliel, IgorG, Laureano, otherwiseguy, mvanbaak
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- 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
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Prodded on IRC by Russell and fixed by eliel
(closes issue #13730)
Reported by: eliel
Patches:
main_cli.patch uploaded by eliel (license 64)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@150664 f38db490-d61c-443f-a65b-d21fe96a405b
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Reported by: nickpeirson
Patches:
pbx.c.patch uploaded by nickpeirson (license 579)
replace_bzero+bcopy.patch uploaded by nickpeirson (license 579)
Tested by: nickpeirson, murf
1. replaced all refs to bzero and bcopy to memset and memmove instead.
2. added a note to the CODING-GUIDELINES
3. add two macros to asterisk.h to prevent bzero, bcopy from creeping
back into the source
4. removed bzero from configure, configure.ac, autoconfig.h.in
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the first item in the CLI entry list. This was preventing '!' from
showing up in either 'help' or in tab completion.
(closes issue #13578)
Reported by: mvanbaak
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@146198 f38db490-d61c-443f-a65b-d21fe96a405b
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This work is done by lmadsen, junky and mvanbaak
during AstriDevCon.
This is the second audit the CLI got, and
this time lmadsen made sure he had _ALL_ modules
loaded that have CLI commands in them.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@145121 f38db490-d61c-443f-a65b-d21fe96a405b
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Changes applied by this patch:
- Fix autocompletion in 'sip prune realtime', sip peers where never auto completed. Now we complete this command with:
'sip prune realtime peer' -> all | like | sip peers
Also I have modified the syntax in the usage, was wrong...
- Pass ast_cli_args->argv and ast_cli_args->argc while running autocompletion on CLI commands (CLI_GENERATE).
With this we avoid comparisons on ast_cli_args->line like this:
strcasestr(a->line, " description")
strcasestr(a->line, "descriptions ")
strcasestr(a->line, "realtime peer"), and so on..
Making the code more confusing (check the spaces in description!).
The only thing we must be sure is to first check a->pos or a->argc.
- Fix 'iax2 prune realtime' autocompletion, now we autocomplete this command with 'all' & 'iax2 peers', check a look that iax2 peers where all the peers, now only the ones in the cache..
(closes issue #13133)
Reported by: eliel
Patches:
clichanges.patch uploaded by eliel (license 64)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@141464 f38db490-d61c-443f-a65b-d21fe96a405b
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files from main/
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r135597 | seanbright | 2008-08-05 09:25:00 -0400 (Tue, 05 Aug 2008) | 1 line
Use PATH_MAX for filenames
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(closes issue #13054)
Reported by: pabelanger
Patches:
ARRAY_LEN.patch2 uploaded by pabelanger (license 224)
Tested by: seanbright
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AMI commands can display that a channel is under control of an AGI.
Work inspired by work at customer site, but paid for by Edvina AB
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r118551 | tilghman | 2008-05-27 14:15:27 -0500 (Tue, 27 May 2008) | 6 lines
When showing an error message for a command, don't shorten the command output,
as it tends to confuse the user (it's fine for suggesting other commands,
however).
Reported by: seanbright (on #asterisk-dev)
Fixed by: me
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(closes issue #10540)
Reported by: spendergrass
Patches:
20080417__bug10540.diff.txt uploaded by Corydon76 (license 14)
Tested by: blitzrage
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interpreted as format strings. Most of these changes are solely to make compiling with -Wsecurity and -Wformat=2 happy, and were not
actual problems, per se. I also added format attributes to any printf wrapper functions I found that didn't have them. -Wsecurity and -Wmissing-format-attribute added to --enable-dev-mode.
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(closes issue #11268)
Reported by: moy
Patches:
chantrace-datastored-encapsulated-rev94934.patch uploaded by moy (license 222)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #11718)
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r97622 | qwell | 2008-01-09 14:28:43 -0600 (Wed, 09 Jan 2008) | 5 lines
Correctly display a message if a command could not be found.
Also fix a comment which may have led to this happening.
Issue 11718, reported by kshumard.
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