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r292412 | pabelanger | 2010-10-20 20:05:45 -0400 (Wed, 20 Oct 2010) | 17 lines
Merged revisions 292411 via svnmerge from
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r292411 | pabelanger | 2010-10-20 20:00:51 -0400 (Wed, 20 Oct 2010) | 10 lines
Record priv-recordintro as sln, not gsm
This removes the gsm->sln step when transcoding
priv-recordintro.
(closes issue #18176)
Reported by: pabelanger
Patches:
chan_sip.diff uploaded by pabelanger (license 224)
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r292226 | jpeeler | 2010-10-18 16:54:38 -0500 (Mon, 18 Oct 2010) | 18 lines
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r292223 | jpeeler | 2010-10-18 16:50:30 -0500 (Mon, 18 Oct 2010) | 11 lines
Fix improper operator key acceptance and clean up temp recording files.
This is a fix for when pressing the operator key after recording an unavailable,
busy, name, or temporary message in mailbox options. The operator key should not
be accepted here, but should be allowed during the message recording. If the
operator key is pressed during ensure the file is saved or deleted as
apporopriate. Also, ensure removal of temporary recorded files after an early
hang up or when message acceptance confirmation times out.
ABE-2518
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r290613 | rmudgett | 2010-10-06 13:42:41 -0500 (Wed, 06 Oct 2010) | 5 lines
Eliminate a redundant test for AST_CONTROL_REDIRECTING.
Eliminate redundant test for AST_CONTROL_REDIRECTING that prevents running
the redirecting interception macro if it is defined.
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r290375 | dvossel | 2010-10-05 14:54:50 -0500 (Tue, 05 Oct 2010) | 10 lines
Fixes PickupChan() not working with full channel name.
(closes issue #18011)
Reported by: schern
Patches:
app_directed_pickup.c.2.patch uploaded by schern (license 995)
app_directed_pickup.c.trunk.patch uploaded by schern (license 995)
Tested by: schern, dvossel
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r289874 | tilghman | 2010-10-01 23:45:49 -0500 (Fri, 01 Oct 2010) | 15 lines
Merged revisions 289873 via svnmerge from
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r289873 | tilghman | 2010-10-01 23:42:08 -0500 (Fri, 01 Oct 2010) | 8 lines
When forwarding a message, a prepend means that the filesystem will always have a better copy.
(closes issue #17803)
Reported by: dpetersen
Patches:
20100923__issue17803.diff.txt uploaded by tilghman (license 14)
Tested by: dpetersen
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r289425 | russell | 2010-09-30 10:37:29 -0500 (Thu, 30 Sep 2010) | 15 lines
Merged revisions 289424 via svnmerge from
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r289424 | russell | 2010-09-30 10:34:29 -0500 (Thu, 30 Sep 2010) | 8 lines
Fix a crash in app_sms.
Since the data being passed to the generator callback is on the stack of the
SMS() application, we must ensure that the generator is stopped before the
application exits.
ABE-2587
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Review: https://reviewboard.asterisk.org/r/942/
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Simplified the locking code by using a local copy of the redirecting party
information in app_dial.c:do_forward() and app_queue.c:wait_for_answer()
for launching the REDIRECTING interception macro when a call is forwarded.
Reduced the lock time of the 'o->chan' and 'in' channels.
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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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r287387 | tilghman | 2010-09-17 16:08:00 -0500 (Fri, 17 Sep 2010) | 14 lines
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r287386 | tilghman | 2010-09-17 16:06:03 -0500 (Fri, 17 Sep 2010) | 7 lines
Blank columns should get set on reload, not ignored.
(closes issue #16893)
Reported by: haakon
Patches:
20100818__issue16893.diff.txt uploaded by tilghman (license 14)
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Review: https://reviewboard.asterisk.org/r/922/
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r286998 | jpeeler | 2010-09-15 15:28:02 -0500 (Wed, 15 Sep 2010) | 14 lines
Merged revisions 286941 via svnmerge from
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r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15 Sep 2010) | 7 lines
Ensure mailbox is not filled to capacity before doing message forwarding.
Specifically, before prompting to record a prepended message the capacity is
checked first. If the mailbox is full the extension will be reprompted.
ABE-2517
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r285710 | bbryant | 2010-09-09 14:50:13 -0400 (Thu, 09 Sep 2010) | 8 lines
Fixes an issue with dialplan pattern matching where the specificity for pattern ranges and pattern special characters was inconsistent.
(closes issue #16903)
Reported by: Nick_Lewis
Patches:
pbx.c-specificity.patch uploaded by Nick Lewis (license 657)
Tested by: Nick_Lewis
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r280671 | tilghman | 2010-08-02 16:26:11 -0500 (Mon, 02 Aug 2010) | 2 lines
Allow the pipe, but also allow the comma
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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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r280160 | seanbright | 2010-07-28 12:51:11 -0400 (Wed, 28 Jul 2010) | 8 lines
Plug a reference leak in app_queue when adding members dynamically.
(closes issue #17738)
Reported by: bobwienholt
Patches:
issue17738.patch uploaded by bobwienholt (license 950)
Tested by: bobwienholt, seanbright
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r279207 | rmudgett | 2010-07-23 17:11:23 -0500 (Fri, 23 Jul 2010) | 14 lines
Merged revisions 279206 via svnmerge from
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r279206 | rmudgett | 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010) | 7 lines
SIP promiscuous redirect could fail to dial the redirect.
The ast_channel was created with one variable to ast_request() but the
call to ast_call() that initiates the outgoing call was using a different
variable. The two variables are not equivalent if the call_forward string
included a channel technology specifier. e.g., SIP/200
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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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r278261 | tilghman | 2010-07-20 17:23:13 -0500 (Tue, 20 Jul 2010) | 7 lines
Delete IMAP messages in reverse order, to ensure reordering after each expunge does not cause deletion of the wrong message.
(closes issue #16350)
Reported by: noahisaac
Patches:
20100623__issue16350.diff.txt uploaded by tilghman (license 14)
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Just say the number of seconds (after minutes) rather than doing some incorrect
calculation with respect to minutes.
(closes issue #17498)
Reported by: corruptor
Patches:
holdesecs_bug.diff uploaded by corruptor (license 253)
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(closes issue #17471)
Reported by: jazzy
Patches:
app_queue.c.diff uploaded by jazzy (license 1056)
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r277182 | pabelanger | 2010-07-16 13:10:36 -0400 (Fri, 16 Jul 2010) | 8 lines
Total analysis time error with SIP and silence suppression
When using app_amd with SIP providers that have silence
suppression on, the iTotalTime count increases exponentially.
(closes issue #17656)
Reported by: juls
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Review: https://reviewboard.asterisk.org/r/777/
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Expand the ani field in ast_party_caller and ast_party_connected_line to
an ast_party_id.
This is an extension to the ast_callerid restructuring patch in review:
https://reviewboard.asterisk.org/r/702/
Review: https://reviewboard.asterisk.org/r/744/
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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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(closes issue #17566)
Reported by: outcast
Patches:
voicemail-rdnis.patch uploaded by outcast (license 1071)
Tested by: outcast
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tracking down the source.
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r275027 | mnicholson | 2010-07-09 11:04:21 -0500 (Fri, 09 Jul 2010) | 8 lines
Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx via the G option in app_dial
(closes issue #17592)
Reported by: jamicque
Patches:
G-flag-cdr-fix1.diff uploaded by mnicholson (license 96)
Tested by: jamicque, mnicholson
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This adds a generic API for accommodating IPv6 and IPv4 addresses
within Asterisk. While many files have been updated to make use of the
API, chan_sip and the RTP code are the files which actually support
IPv6 addresses at the time of this commit. The way has been paved for
easier upgrading for other files in the near future, though.
Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne
for their hard work on this.
(closes issue #17565)
Reported by: russell
Patches:
asteriskv6-test-report.pdf uploaded by russell (license 2)
Review: https://reviewboard.asterisk.org/r/743
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midterm evaluation.
Review: https://reviewboard.asterisk.org/r/757/
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force directly to the default case.
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(Also fix the typos in the comments)
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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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This version of the patch only adds AgentComplete for attended transfers. It was already present for blind transfers.
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r272367 | mnicholson | 2010-06-23 17:33:51 -0500 (Wed, 23 Jun 2010) | 8 lines
Send AgentComplete manager events in the event of blind and attended transfers.
(closes issue #16819)
Reported by: elbriga
Patches:
app_queue.diff uploaded by elbriga (license 482)
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272368 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272259 f38db490-d61c-443f-a65b-d21fe96a405b
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272257 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272146 f38db490-d61c-443f-a65b-d21fe96a405b
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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
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@272109 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #16869)
Reported by: chappell
Patches:
app_say_counted-20100317.c uploaded by chappell (license 8)
Tested by: chappell
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271520 f38db490-d61c-443f-a65b-d21fe96a405b
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(closes issue #17516)
Reported by: karlfife
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@271089 f38db490-d61c-443f-a65b-d21fe96a405b
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