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r297245 | russell | 2010-12-02 07:20:19 -0600 (Thu, 02 Dec 2010) | 20 lines
Merged revisions 297229 via svnmerge from
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r297229 | russell | 2010-12-02 07:16:47 -0600 (Thu, 02 Dec 2010) | 13 lines
Merged revisions 297228 via svnmerge from
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r297228 | russell | 2010-12-02 07:16:15 -0600 (Thu, 02 Dec 2010) | 6 lines
Add "DAHDI" to a couple of app_meetme error messages.
This is in response to some questions on IRC. To the user, there was nothing
that made it obvious that this error had anything to do with DAHDI not being
loaded.
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r296787 | tilghman | 2010-11-30 13:12:48 -0600 (Tue, 30 Nov 2010) | 2 lines
DOC: Conference number can be omitted; if omitted, all users in a meetme are listed.
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r296467 | tilghman | 2010-11-27 04:40:22 -0600 (Sat, 27 Nov 2010) | 12 lines
Merged revisions 296466 via svnmerge from
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r296466 | tilghman | 2010-11-27 04:39:01 -0600 (Sat, 27 Nov 2010) | 5 lines
18 characters is too short for most date/times (20 is the usual, but we add more in case of greater precision).
(closes issue #18369)
Reported by: tnakonz
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Added option v(mailbox@[context]) which tells MeetMe where to look for a users greet file. If one does not exist it clears the v option and defers to the functionality of i/I as/if set by the MeetMe() command.
Review: https://reviewboard.asterisk.org/r/1009/
(closes issue #18297)
Reported by: parisioa
Patches:
meetme_final_patch_v.diff uploaded by parisioa (license 1153)
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r287760 | bbryant | 2010-09-20 20:00:23 -0400 (Mon, 20 Sep 2010) | 30 lines
Merged revisions 287759 via svnmerge from
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r287759 | bbryant | 2010-09-20 19:58:26 -0400 (Mon, 20 Sep 2010) | 23 lines
Merged revisions 287758 via svnmerge from
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r287758 | bbryant | 2010-09-20 19:57:08 -0400 (Mon, 20 Sep 2010) | 16 lines
Fix misvalidation of meetme pins in conjunction with the 'a' MeetMe flag.
When using the 'a' MeetMe flag and having a user and admin pin setup for your
conference, using the user pin would gain you admin priviledges. Also, when no
user pin was set, an admin pin was, the 'a' MeetMe flag wasn't used, and the
user tried to enter a conference then they were still prompted for a pin and
forced to hit #.
(closes issue #17908)
Reported by: kuj
Patches:
pins_2.patch uploaded by kuj (license 1111)
Tested by: kuj
Review: [full review board URL with trailing slash]
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r285533 | bbryant | 2010-09-08 16:58:43 -0400 (Wed, 08 Sep 2010) | 15 lines
Merged revisions 285532 via svnmerge from
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r285532 | bbryant | 2010-09-08 16:56:12 -0400 (Wed, 08 Sep 2010) | 8 lines
Fixes a bug with MeetMe where after announcing the amount of time left in a conference, if music on hold was playing, it doesn't restart.
(closes issue #17408)
Reported by: sysreq
Patches:
asterisk-issue-17408_fixed.patch uploaded by sysreq (license 1009)
Tested by: sysreq
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r280346 | jeang | 2010-07-29 11:07:16 -0500 (Thu, 29 Jul 2010) | 17 lines
Merged revisions 280345 via svnmerge from
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r280345 | jeang | 2010-07-29 11:01:35 -0500 (Thu, 29 Jul 2010) | 10 lines
Merged revisions 280341 via svnmerge from
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r280341 | jeang | 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) | 2 lines
Fix a dsp structure leak occuring when a local channel is put into a meetme
conference, then masquaraded away.
ABE-2422
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trying to find an empty one.
Also, parse the useropts properly, when retrieving from realtime, and add them
to the existing flags.
(closes issue #17502)
Reported by: kenji
Patches:
20100720__issue17502.diff.txt uploaded by tilghman (license 14)
Tested by: kenji
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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/
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r275773 | jpeeler | 2010-07-12 15:34:51 -0500 (Mon, 12 Jul 2010) | 12 lines
Make user removals and traversals thread safe in meetme.
Race conditions present in meetme involving the user list where a lack of
locking has the potential for a user to be removed during a traversal or as in
the case of the reporter after checking if the list is empty could cause a
crash. Fixing this was done by convering the userlist to an ao2 container.
(closes issue #17390)
Reported by: Vince
Review: https://reviewboard.asterisk.org/r/746/
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dahdi pseudo channel, so if we fail doing it, continue creating the conference.
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tracking down the source.
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midterm evaluation.
Review: https://reviewboard.asterisk.org/r/757/
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r273474 | jpeeler | 2010-07-01 15:19:16 -0500 (Thu, 01 Jul 2010) | 14 lines
Allow admin user to join conference without using admin mode and no user pin.
Configuring the conference in meetme.conf like the following:
conf => 2345,,6666
did not prompt for pin when used without admin mode. This meant that the
conference could not be joined as an admin even if the user knew the correct
pin. The original bug report was submitted claiming that the blank user pin
should deny entry into the conference. I think a better way to handle this
would be with a feature enhancement that used the following syntax:
conf => 2345,X,6666 - where X denotes no acceptable pin allowed
(closes issue #15704)
Reported by: modelnine
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r273354 | jpeeler | 2010-07-01 10:05:43 -0500 (Thu, 01 Jul 2010) | 12 lines
Ensure channel placed in meetme in ringing state is properly hung up.
An outgoing channel placed in meetme while still ringing which was then hung up
would not exit meetme and the channel was not properly destroyed. Specifically
checking for this scenario by looking at the appropriate control frames resolves
the issue.
(closes issue #15871)
Reported by: Ivan
Patches:
meetme_congestion_trunk_v2.patch uploaded by Ivan (license 229)
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r272255 | pabelanger | 2010-06-23 16:57:01 -0400 (Wed, 23 Jun 2010) | 12 lines
First caller into a dynamic conference now enter pin once.
If MeetMe is configured to use dynamic conference
numbers, then the first caller (which creates the
conference) had to enter the PIN number twice.
(closes issue #15878)
Reported by: shawkris
Patches:
issue15878.patch uploaded by pabelanger (license 224)
Tested by: pabelanger
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Even if there are no stations or trunks defined, we need to start the sla
thread to make sure we get the reload event. Also, when doing a reload we need
to remove the existing trunks and stations or they end up hanging around.
(closes issue #16818)
Reported by: mbonin
Patches:
sla_reload.patch uploaded by twilson (license 396)
Tested by: twilson
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(closes issue #17516)
Reported by: karlfife
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(closes issue #17336)
Reported by: snuffy
Patches:
doxygen-fixes1.diff uploaded by snuffy (license 35)
Tested by: russell
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r262662 | dvossel | 2010-05-12 12:00:04 -0500 (Wed, 12 May 2010) | 11 lines
fixes app_meetme dsp error
We attempted to detect silence after translating a frame
from signed linear. This caused a flooding of errors. To
resolve this the code to detect silence was moved before the
translation.
(closes issue #17133)
Reported by: jsdyer
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in-conf
This patch adds the following in-conference admin DTMF features:
*81 - Roll call (or simply user count if INTROUSER isn't enabled)
*82 - Eject all non-admins
*83 - Mute/unmute all non-admins
*84 - Start recording the conference on the fly
FWIW, this code uses newly recorded prompts.
(closes issue #16379)
Reported by: rfinnie
Patches:
meetme-enhancements-232771-v1.patch uploaded by rfinnie (license 940)
modified slightly by me
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(closes issue #16866)
Reported by: DEA
Patches:
rt-meetme-options.txt uploaded by DEA (license 3)
Tested by: DEA
Review: https://reviewboard.asterisk.org/r/582/
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Reported by philipp64 in #asterisk-dev.
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(closes issue #15725)
Reported by: shanermn
Patches:
v1-15725.patch uploaded by dimas (license 88)
Tested by: shanermn
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r236509 | seanbright | 2009-12-28 07:43:36 -0500 (Mon, 28 Dec 2009) | 12 lines
Avoid a crash with large numbers of MeetMe conferences.
Similar to changes made to Queue(), when we have large numbers of conferences in
meetme.conf (1000s) and we use alloca()/strdupa(), we can blow out the stack and
crash, so instead just use a single fixed buffer.
(closes issue #16509)
Reported by: Kashif Raza
Patches:
20091223_16509.patch uploaded by seanbright (license 71)
Tested by: seanbright
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r234379 | jpeeler | 2009-12-11 16:37:21 -0600 (Fri, 11 Dec 2009) | 11 lines
Fix talking detection status after conference user is muted.
This patch ensures that when a conference user is muted that the accompanying
AMI Meetme talking off event is sent. Also, the meetme list output is updated
to show the muted user as unmonitored.
(closes issue #16247)
Reported by: dimas
Patches:
v3-16247.patch uploaded by dimas (license 88)
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As described in the CHANGES file:
* MeetMe has a new option 'G' to play an announcement before joining a
conference.
* Page has a new option 'A(x)' which will playback an announcement
simultaneously to all paged phones (and optionally excluding the caller's one
using the new option 'n') before the call is bridged.
To add the new option to meetme, the conference flag options had to be extended
to 64 bits.
(closes issue #14365)
Reported by: dferrer
Patches:
page_announce.patch uploaded by dferrer (license 525)
modified by me
Review: https://reviewboard.asterisk.org/r/188/
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(Closes AST-33)
Reviewboard: https://reviewboard.asterisk.org/r/368/
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(closes issue #15450)
Reported by: JimVanM
Patches:
meetmeinfopatch.diff.txt uploaded by dbrooks (license 790)
Tested by: JimVanM
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Reviewboard: https://reviewboard.asterisk.org/r/416/
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r225105 | tilghman | 2009-10-21 11:02:12 -0500 (Wed, 21 Oct 2009) | 4 lines
Fix documentation for ast_softhangup() and correct the misuse thereof.
(closes issue #16103)
Reported by: majorbloodnok
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(Fixes SWP-238)
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Reported by ebroad in #asterisk-bugs
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r217156 | tilghman | 2009-09-08 15:01:45 -0500 (Tue, 08 Sep 2009) | 7 lines
When MOH is playing on the channel, announcements sent through the conference are not heard.
(closes issue #14588)
Reported by: voipas
Patches:
20090716__issue14588__2.diff.txt uploaded by tilghman (license 14)
Tested by: lmadsen, twisted, tilghman
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lengths more consistent.
(closes issue #15493)
Reported by: lasko
Patches:
meetme.diff uploaded by lasko (license 833)
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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/
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r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines
Improve support for media paths that can generate multiple frames at once.
There are various media paths in Asterisk (codec translators and UDPTL, primarily)
that can generate more than one frame to be generated when the application calling
them expects only a single frame. This patch addresses a number of those cases,
at least the primary ones to solve the known problems. In addition it removes the
broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API
functions, and cleans up various code paths affected by these changes.
https://reviewboard.asterisk.org/r/175/
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201056 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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Move function MEETME_INFO static documentation to the new AstXML form.
(issue #15245)
Reported by: eliel
Patches:
app_meetme_static_conversion.txt uploaded by lmadsen (license 10)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199409 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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