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r221085 | seanbright | 2009-09-30 10:47:58 -0400 (Wed, 30 Sep 2009) | 9 lines
Clarify documentation for VoiceMailMain()'s a() option.
We require box numbers, not names as the documentation implies.
(issue #14740)
Reported by: pj
Patches:
__20090729-app_voicemail-documentation.patch uploaded by lmadsen (license 10)
Tested by: seanbright, lmadsen
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chanspy_ds_chan_fixup() is called with the channel locked.
(closes issue #15965)
Reported by: atis
Patches:
chanspy-deadlock-fix1.diff uploaded by mnicholson (license 96)
Tested by: atis
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r220833 | jpeeler | 2009-09-29 11:58:29 -0500 (Tue, 29 Sep 2009) | 12 lines
Make deletion of temporary greetings work properly with IMAP_STORAGE
When imapgreetings was set to yes, the message was being deleted but wasn't
actually being expunged. When imapgreetings was set to no, the file based
message was not being deleted at all. All good now!
(closes issue #14949)
Reported by: noahisaac
Patches:
vm_tempgreeting_removal.patch uploaded by noahisaac (license 748),
modified by me
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r220289 | tilghman | 2009-09-24 14:41:02 -0500 (Thu, 24 Sep 2009) | 13 lines
Merged revisions 220288 via svnmerge from
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r220288 | tilghman | 2009-09-24 14:39:41 -0500 (Thu, 24 Sep 2009) | 6 lines
Implicitly sending a progress signal breaks some applications.
Call Progress() in your dialplan if you explicitly want progress to be sent.
(Reverts change 216430, closes issue #15957)
Reported by: Pavel Troller on the Asterisk-Dev mailing list
http://lists.digium.com/pipermail/asterisk-dev/2009-September/039897.html
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r219818 | tilghman | 2009-09-22 16:43:22 -0500 (Tue, 22 Sep 2009) | 17 lines
Merged revisions 219816 via svnmerge from
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r219816 | tilghman | 2009-09-22 16:37:03 -0500 (Tue, 22 Sep 2009) | 10 lines
When IMAP variables were changed during a reload, Voicemail did not use the new values.
This change introduces a configuration version variable, which ensures that
connections with the old values are not reused but are allowed to expire
normally.
(closes issue #15934)
Reported by: viniciusfontes
Patches:
20090922__issue15934.diff.txt uploaded by tilghman (license 14)
Tested by: viniciusfontes
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r219412 | tilghman | 2009-09-18 08:54:51 -0500 (Fri, 18 Sep 2009) | 6 lines
Missing value setting line for maxsecs/maxmessage
(closes issue #15696)
Reported by: fhackenberger
Patches:
maxsecs.patch uploaded by fhackenberger (license 592)
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r218731 | tilghman | 2009-09-15 17:33:10 -0500 (Tue, 15 Sep 2009) | 13 lines
Merged revisions 218730 via svnmerge from
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r218730 | tilghman | 2009-09-15 17:27:41 -0500 (Tue, 15 Sep 2009) | 6 lines
If the user enters the same password as before, don't signal an error when the change does nothing.
(closes issue #15492)
Reported by: cbbs70a
Patches:
20090713__issue15492.diff.txt uploaded by tilghman (license 14)
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r218579 | tilghman | 2009-09-15 11:04:41 -0500 (Tue, 15 Sep 2009) | 16 lines
Merged revisions 218577 via svnmerge from
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r218577 | tilghman | 2009-09-15 11:01:17 -0500 (Tue, 15 Sep 2009) | 9 lines
Ensure FollowMe sets language in channels it creates.
Also, not in the original bug report, but related fields are accountcode and
musicclass, and the inheritance of datastores.
(closes issue #15372)
Reported by: Romik
Patches:
20090828__issue15372.diff.txt uploaded by tilghman (license 14)
Tested by: cervajs
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r218361 | tilghman | 2009-09-14 14:29:48 -0500 (Mon, 14 Sep 2009) | 11 lines
Recorded merge of revisions 218331 via svnmerge from
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r218331 | tilghman | 2009-09-14 14:16:35 -0500 (Mon, 14 Sep 2009) | 4 lines
Don't say "Please try again" if we don't give the user another chance to try again.
(issue #15055, SWP-129)
Reported by: jthurman
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r218224 | mnicholson | 2009-09-14 09:57:23 -0500 (Mon, 14 Sep 2009) | 14 lines
Merged revisions 218223 via svnmerge from
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r218223 | mnicholson | 2009-09-14 09:53:57 -0500 (Mon, 14 Sep 2009) | 8 lines
Ensure we don't pickup ourselves when doing pickup by exten.
(closes issue #15100)
Reported by: lmsteffan
Patches:
(modified) pickup.patch uploaded by lmsteffan (license 779)
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r217990 | tilghman | 2009-09-10 18:54:51 -0500 (Thu, 10 Sep 2009) | 10 lines
Merged revisions 217989 via svnmerge from
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r217989 | tilghman | 2009-09-10 18:52:22 -0500 (Thu, 10 Sep 2009) | 3 lines
Don't ring another channel, if there's not enough time for a queue member to answer.
(Fixes AST-228)
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r217199 | tilghman | 2009-09-08 15:28:41 -0500 (Tue, 08 Sep 2009) | 14 lines
Merged revisions 217156 via svnmerge from
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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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r216438 | oej | 2009-09-04 16:02:34 +0200 (Fre, 04 Sep 2009) | 35 lines
Merged revisions 216430 via svnmerge from
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r216430 | oej | 2009-09-04 15:45:48 +0200 (Fre, 04 Sep 2009) | 27 lines
Make apps send PROGRESS control frame for early media and fix too early media issue in SIP
The issue at hand is that some legacy (dying) PBX systems send empty media frames on PRI
links *before* any call progress. The SIP channel receives these frames and by default
signals 183 Session progress and starts sending media. This will cause phones to
play silence and ignore the later 180 ringing message. A bad user experience.
The fix is twofold:
- We discovered that asterisk apps that support early media ("noanswer") did not send
any PROGRESS frame to indicate early media. Fixed.
- We introduce a setting in chan_sip so that users can disable any relay of media frames
before the outbound channel actually indicates any sort of call progress.
In 1.4, 1.6.0 and 1.6.1, this will be disabled for backward compatibility. In later versions
of Asterisk, this will be enabled. We don't assume that it will change your Asterisk
phone experience - only for the better.
We encourage third-party application developers to make sure that if they have applications
that wants to send early media, add a PROGRESS control frame transmission to make sure that
all channel drivers actually will start sending early media. This has not been the default
in Asterisk previous to this patch, so if you got inspiration from our code, you need to
update accordingly. Sorry for the trouble and thanks for your support.
This code has been running for a few months in a large scale installation (over 250
servers with PRI and/or BRI links to old PBX systems).
That's no proof that this is an excellent patch, but, well, it's tested :-)
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r216593 | seanbright | 2009-09-04 15:29:02 -0400 (Fri, 04 Sep 2009) | 1 line
Use ast_free() instead of free().
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r215338 | dhubbard | 2009-09-01 20:16:59 -0500 (Tue, 01 Sep 2009) | 18 lines
Merged revisions 215270 via svnmerge from
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r215270 | dhubbard | 2009-09-01 18:04:52 -0500 (Tue, 01 Sep 2009) | 12 lines
Use strrchr() so SoftHangup will correctly truncate multi-hyphen channel names
In general channel names are in the form Foo/Bar-Z, but the channel name
could have multiple hyphens and look like Foo/B-a-r-Z. Use strrchr to
truncate the channel name at the last hyphen.
(closes issue #15810)
Reported by: dhubbard
Patches:
dw-softhangup-1.4.patch uploaded by dhubbard (license 733)
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r213833 | jpeeler | 2009-08-24 11:43:57 -0500 (Mon, 24 Aug 2009) | 14 lines
Fix storage of greetings when using IMAP_STORAGE
Fix checking if the imapgreetings option is turned on to store the greeting
in IMAP.
(closes issue #14950)
Reported by: noahisaac
Patches:
14950.patch uploaded by mmichelson (license 60)
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r213404 | jpeeler | 2009-08-20 16:33:11 -0500 (Thu, 20 Aug 2009) | 12 lines
Fix greeting retrieval from IMAP
Properly check for the current voicemail state and if it doesn't exist,
create it.
(closes issue #14597)
Reported by: wtca
Patches:
14597_v2.patch uploaded by mmichelson (license 60)
Tested by: jpeeler
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r213113 | dvossel | 2009-08-19 16:21:00 -0500 (Wed, 19 Aug 2009) | 14 lines
Merged revisions 213103 via svnmerge from
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r213103 | dvossel | 2009-08-19 16:18:37 -0500 (Wed, 19 Aug 2009) | 8 lines
Fixes memory leak caused by incorrectly freeing mixmonitor
(closes issue #15699)
Reported by: edantie
Patches:
mixmonitor.patch uploaded by edantie (license 862)
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r212627 | tilghman | 2009-08-17 14:57:42 -0500 (Mon, 17 Aug 2009) | 4 lines
Check the return value of opendir(3), or we may crash.
(closes issue #15720)
Reported by: tobias_e
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r211957 | mnicholson | 2009-08-12 18:14:36 -0500 (Wed, 12 Aug 2009) | 17 lines
Merged revisions 211953 via svnmerge from
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r211953 | mnicholson | 2009-08-12 18:04:02 -0500 (Wed, 12 Aug 2009) | 10 lines
This patch adds additional checking when generating queue log TRANSFER events.
The additional checks prevent generation of false TRANSFER events in certain situations.
(closes issue #14536)
Reported by: aragon
Patches:
queue-log-xfer-fix1.diff uploaded by mnicholson (license 96)
Tested by: aragon, mnicholson
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r211232 | tilghman | 2009-08-09 02:11:22 -0500 (Sun, 09 Aug 2009) | 4 lines
Check for NULL frame, before dereferencing pointer.
(closes issue #15617)
Reported by: rain
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r211113 | russell | 2009-08-07 15:12:21 -0500 (Fri, 07 Aug 2009) | 11 lines
Recorded merge of revisions 211112 via svnmerge from
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r211112 | russell | 2009-08-07 15:11:31 -0500 (Fri, 07 Aug 2009) | 4 lines
Resolve a deadlock involving app_chanspy and masquerades.
(ABE-1936)
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r211040 | tilghman | 2009-08-07 13:17:41 -0500 (Fri, 07 Aug 2009) | 21 lines
Merged revisions 211038 via svnmerge from
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r211038 | tilghman | 2009-08-07 13:16:28 -0500 (Fri, 07 Aug 2009) | 14 lines
QUEUE_MEMBER_LIST _really_ wants the interface name, not the membername.
This is a partial revert of revision 82590, which was an attempted cleanup,
but in reality, it broke QUEUE_MEMBER_LIST, which has always been intended
as a method by which component interfaces could be queried from the queue.
Membername isn't useful here, because that field cannot be used to obtain
further information about the member. See the documentation on
QUEUE_MEMBER_LIST, RemoveQueueMember, QUEUE_MEMBER_PENALTY, and the various
AMI commands which take a member argument for further justification.
(closes issue #15664)
Reported by: rain
Patches:
app_queue-queue_member_list.diff uploaded by rain (license 327)
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r210908 | tilghman | 2009-08-06 16:29:26 -0500 (Thu, 06 Aug 2009) | 9 lines
Allow Gosub to recognize quote delimiters without consuming them.
(closes issue #15557)
Reported by: rain
Patches:
20090723__issue15557.diff.txt uploaded by tilghman (license 14)
Tested by: rain
Review: https://reviewboard.asterisk.org/r/316/
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r209839 | russell | 2009-08-01 06:02:07 -0500 (Sat, 01 Aug 2009) | 20 lines
Merged revisions 209838 via svnmerge from
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r209838 | russell | 2009-08-01 05:59:05 -0500 (Sat, 01 Aug 2009) | 13 lines
Modify how Playtones() is used in Milliwatt() to resolve gain issue.
When Milliwatt() was changed internally to use Playtones() so that the proper
tone was used, it introduced a drop in gain in the output signal. So, use
the playtones API directly and specify a volume argument such that the output
matches the gain of the original Milliwatt() code.
(closes issue #15386)
Reported by: rue_mohr
Patches:
issue_15386.rev2.diff uploaded by russell (license 2)
Tested by: rue_mohr
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r209554 | dbrooks | 2009-07-30 11:07:05 -0500 (Thu, 30 Jul 2009) | 6 lines
Fixes numerous spelling errors. Patch submitted by alecdavis.
(closes issue #15595)
Reported by: alecdavis
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r209279 | kpfleming | 2009-07-27 16:43:36 -0500 (Mon, 27 Jul 2009) | 7 lines
Cleanup T.38 negotiation changes.
Convert LOG_NOTICE messages about T.38 negotiation in debug level 1 messages,
clean up some looping logic, and correct an improper use of ast_free() for
freeing an ast_frame.
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r209256 | kpfleming | 2009-07-27 16:21:43 -0500 (Mon, 27 Jul 2009) | 10 lines
Make T.38 switchover in ReceiveFAX synchronous.
In receive mode, if the channel that ReceiveFAX is running on supports T.38,
we should *always* attempt to switch T.38, rather than listening for an incoming
CNG tone and only triggering on that. The channel may be using a low-bitrate
codec that distorts the CNG tone, the sending FAX endpoint may not send CNG
at all, or there could be a variety of other reasons that we don't detect it,
but in all those cases if T.38 is available we certainly want to use it.
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r208622 | mmichelson | 2009-07-24 14:24:28 -0500 (Fri, 24 Jul 2009) | 16 lines
Don't impose an arbitrary limit on member lines in queues.conf
I know what some of you are thinking: "UGH! Mark, why are you using
ast_strdup and ast_free for the string when you can just use ast_strdupa
and let the memory free itself?! Have the bats been chewing on your brain
again?"
Based on past experiences, I don't like using ast_strdupa inside a loop.
It's a good way to potentially exhaust stack space. Also, since this only
happens when reloading queues, I don't think that heap allocations and
frees are going to be a huge problem.
(closes issue #15559)
Reported by: amorsen
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r208593 | russell | 2009-07-24 13:42:32 -0500 (Fri, 24 Jul 2009) | 14 lines
Merged revisions 208592 via svnmerge from
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r208592 | russell | 2009-07-24 13:38:24 -0500 (Fri, 24 Jul 2009) | 7 lines
Do not log an ERROR if autoservice_stop() returns -1.
This does not indicate an error. A return of -1 just means that the channel
has been hung up.
(reported in #asterisk-dev)
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r208464 | kpfleming | 2009-07-23 16:57:24 -0500 (Thu, 23 Jul 2009) | 46 lines
Rework of T.38 negotiation and UDPTL API to address interoperability problems
Over the past couple of months, a number of issues with Asterisk
negotiating (and successfully completing) T.38 sessions with various
endpoints have been found. This patch attempts to address many of
them, primarily focused around ensuring that the endpoints'
MaxDatagram size is honored, and in addition by ensuring that T.38
session parameter negotiation is performed correctly according to the
ITU T.38 Recommendation.
The major changes here are:
1) T.38 applications in Asterisk (app_fax) only generate/receive IFP
packets, they do not ever work with UDPTL packets. As a result of
this, they cannot be allowed to generate packets that would overflow
the other endpoints' MaxDatagram size after the UDPTL stack adds any
error correction information. With this patch, the application is told
the maximum *IFP* size it can generate, based on a calculation using
the far end MaxDatagram size and the active error correction mode on
the T.38 session. The same is true for sending *our* MaxDatagram size
to the remote endpoint; it is computed from the value that the
application says it can accept (for a single IFP packet) combined with
the active error correction mode.
2) All treatment of T.38 session parameters as 'capabilities' in
chan_sip has been removed; these parameters are not at all like
audio/video stream capabilities. There are strict rules to follow for
computing an answer to a T.38 offer, and chan_sip now follows those
rules, using the desired parameters from the application (or channel)
that wants to accept the T.38 negotiation.
3) chan_sip now stores and forwards ast_control_t38_parameters
structures for tracking 'our' and 'their' T.38 session parameters;
this greatly simplifies negotiation, especially for pass-through
calls.
4) Since T.38 negotiation without specifying parameters or receiving
the final negotiated parameters is not very worthwhile, the
AST_CONTROL_T38 control frame has been removed. A note has been added
to UPGRADE.txt about this removal, since any out-of-tree applications
that use it will no longer function properly until they are upgraded
to use AST_CONTROL_T38_PARAMETERS.
Review: https://reviewboard.asterisk.org/r/310/
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r205770 | kpfleming | 2009-07-10 10:28:11 -0500 (Fri, 10 Jul 2009) | 12 lines
Fix some remaining T.38 negotiation problems in app_fax.
Revision 205696 did not quite fix all the issues with the T.38 negotiation
changes and app_fax; this patch corrects them, along with a couple of other
minor issues.
(closes issue #15480)
Reported by: dimas
Patches:
test2-15480.patch uploaded by dimas (license 88)
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r205696 | kpfleming | 2009-07-09 16:20:23 -0500 (Thu, 09 Jul 2009) | 16 lines
Repair ability of SendFAX/ReceiveFAX to respond to T.38 switchover.
Recent changes in T.38 negotiation in Asterisk caused these applications to
not respond when the other endpoint initiated a switchover to T.38; this
resulted in the T.38 switchover failing, and the FAX attempt to be made
using an audio connection, instead of T.38 (which would usually cause the
FAX to fail completely).
This patch corrects this problem, and the applications will now correctly
respond to the T.38 switchover request. In addition, the response will include
the appopriate T.38 session parameters based on what the other end offered
and what our end is capable of.
(closes issue #14849)
Reported by: afosorio
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r205350 | mmichelson | 2009-07-08 14:26:55 -0500 (Wed, 08 Jul 2009) | 20 lines
Merged revisions 205349 via svnmerge from
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r205349 | mmichelson | 2009-07-08 14:26:13 -0500 (Wed, 08 Jul 2009) | 14 lines
Prevent phantom calls to queue members.
If a caller were to hang up while a periodic announcement or position
were being said, the return value for those functions would incorrectly
indicate that the caller was still in the queue. With these changes,
the problem does not occur.
(closes issue #14631)
Reported by: latinsud
Patches:
queue_announce_ghost_call2.diff uploaded by latinsud (license 745)
(with small modification from me)
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r204470 | tilghman | 2009-06-30 13:36:24 -0500 (Tue, 30 Jun 2009) | 18 lines
Recorded merge of revisions 204469 via svnmerge from
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r204469 | tilghman | 2009-06-30 13:23:35 -0500 (Tue, 30 Jun 2009) | 11 lines
"tw" is the language specification for Twi (from Ghana) not Taiwanese.
(closes issue #15346)
Reported by: volivier
Patches:
20090617__issue15346__1.4.diff.txt uploaded by tilghman (license 14)
20090617__issue15346__trunk.diff.txt uploaded by tilghman (license 14)
20090617__issue15346__1.6.0.diff.txt uploaded by tilghman (license 14)
20090617__issue15346__1.6.1.diff.txt uploaded by tilghman (license 14)
20090617__issue15346__1.6.2.diff.txt uploaded by tilghman (license 14)
Tested by: volivier
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r203721 | dbrooks | 2009-06-26 15:13:51 -0500 (Fri, 26 Jun 2009) | 16 lines
Fixing voicemail's error in checking max silence vs min message length
Max silence was represented in milliseconds, yet vmminsecs (minmessage) was represented
as seconds.
Also, the inequality was reversed. The warning, if triggered, was "Max silence should
be less than minmessage or you may get empty messages", which should have been logged
if max silence was greater than minmessage, but the check was for less than.
Also, conforming if statement to coding guidelines.
closes issue #15331)
Reported by: markd
Review: https://reviewboard.asterisk.org/r/293/
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r203699 | file | 2009-06-26 16:27:24 -0300 (Fri, 26 Jun 2009) | 2 lines
Improve T.38 negotiation by exchanging session parameters between application and channel.
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r202183 | seanbright | 2009-06-20 15:09:47 -0400 (Sat, 20 Jun 2009) | 5 lines
Fix version detection for API changes in spandsp.
(closes issue #15355)
Reported by: deuffy
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r201783 | tilghman | 2009-06-18 15:52:36 -0500 (Thu, 18 Jun 2009) | 6 lines
One of the changes in 1.6.1 was to allow app_directory to use functionality
within app_voicemail for directory functions. It is therefore no longer
necessary for app_directory to be linked against the ODBC libraries (and it
never was necessary for app_directory to be linked against IMAP, though it
was).
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r201445 | dvossel | 2009-06-17 14:45:35 -0500 (Wed, 17 Jun 2009) | 25 lines
Merged revisions 201423 via svnmerge from
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r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17 Jun 2009) | 19 lines
StopMixMonitor race condition (not giving up file immediately)
StopMixMonitor only indicates to the MixMonitor thread to stop
writing to the file. It does not guarantee that the recording's
file handle is available to the dialplan immediately after execution.
This results in a race condition. To resolve this, the filestream
pointer is placed in a datastore on the channel. When StopMixMonitor
is called, the datastore is retrieved from the channel and the
filestream is closed immediately before returning to the dialplan.
Documentation indicating the use of StopMixMonitor to free files
has been updated as well.
(closes issue #15259)
Reported by: travisghansen
Tested by: dvossel
Review: https://reviewboard.asterisk.org/r/283/
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r201056 | kpfleming | 2009-06-16 13:54:30 -0500 (Tue, 16 Jun 2009) | 18 lines
Merged revisions 200991 via svnmerge from
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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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r201090 | kpfleming | 2009-06-16 14:27:12 -0500 (Tue, 16 Jun 2009) | 5 lines
Another minor fix to compiler attribute checking.
Defaulting to 'static' for the function scope was bad... so remove it.
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r200943 | mvanbaak | 2009-06-16 17:51:36 +0200 (Tue, 16 Jun 2009) | 9 lines
add FILE_STORAGE to Voicemail Build Options
Voicemail can only use one storage module at the moment.
Because it's unclear that selecting one of the storage modules
in menuselect will disable filesystem storage we now have
a FILE_STORAGE option that conflicts with the other modules.
(closes issue #15333)
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r198285 | seanbright | 2009-05-29 23:26:06 -0400 (Fri, 29 May 2009) | 15 lines
Merged revisions 198251 via svnmerge from
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r198251 | seanbright | 2009-05-29 22:46:41 -0400 (Fri, 29 May 2009) | 8 lines
Treat an empty FORWARD_CONTEXT the same way we treat a missing one.
(closes issue #15056)
Reported by: p_lindheimer
Patches:
05292009_bug15056.diff uploaded by seanbright (license 71)
Tested by: p_lindheimer
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Updated the MixMonitor documentation for the 'b' option so that
it is more obvious that you must not optimize awat the Local
channel when using this option.
(issue #14829)
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r197543 | mmichelson | 2009-05-28 09:58:06 -0500 (Thu, 28 May 2009) | 27 lines
Merged revisions 197537 via svnmerge from
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r197537 | mmichelson | 2009-05-28 09:49:13 -0500 (Thu, 28 May 2009) | 21 lines
Add flags to chanspy audiohook so that audio stays in sync.
There are two flags being added to the chanspy audiohook here. One
is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set,
we ensure that the read and write slinfactories on the audiohook do
not skew beyond a certain tolerance.
In addition, there is a new audiohook flag added here,
AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for
a slinfactory to build up a substantial amount of audio before
flushing it. For this particular issue, this means that the person
spying on the call will hear the conversations in real time with very
little delay in the audio.
(closes issue #13745)
Reported by: geoffs
Patches:
13745.patch uploaded by mmichelson (license 60)
Tested by: snblitz
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r195839 | tilghman | 2009-05-20 18:30:05 -0500 (Wed, 20 May 2009) | 3 lines
If a variable had a blank value upon the initial setting, then it would do nothing.
Identified by Dmitry Andrianov via private email, fixed by me.
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r195636 | file | 2009-05-20 14:14:42 -0300 (Wed, 20 May 2009) | 12 lines
Merged revisions 195635 via svnmerge from
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r195635 | file | 2009-05-20 14:14:00 -0300 (Wed, 20 May 2009) | 5 lines
Fix a bug where the MeetMe option 'D' did not actually prompt for the pin.
(closes issue #15050)
Reported by: pmhaddad
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r195521 | tilghman | 2009-05-19 15:16:01 -0500 (Tue, 19 May 2009) | 14 lines
Merged revisions 195520 via svnmerge from
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r195520 | tilghman | 2009-05-19 15:12:20 -0500 (Tue, 19 May 2009) | 7 lines
Ensure thread keys are initialized before attempting to access them.
(closes issue #14889)
Reported by: jaroth
Patches:
app_voicemail.c.patch uploaded by msirota (license 758)
Tested by: msirota, BlargMaN
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