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r310142 | tilghman | 2011-03-09 23:53:29 -0600 (Wed, 09 Mar 2011) | 19 lines
Merged revisions 310141 via svnmerge from
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r310141 | tilghman | 2011-03-09 23:51:37 -0600 (Wed, 09 Mar 2011) | 12 lines
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r310140 | tilghman | 2011-03-09 23:38:44 -0600 (Wed, 09 Mar 2011) | 5 lines
Initialize column size to 0 to deal with a potential UnixODBC bug on 64-bit systems.
(closes issue #18295)
Reported by: pruiz
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r309858 | jrose | 2011-03-07 16:07:25 -0600 (Mon, 07 Mar 2011) | 22 lines
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r309857 | jrose | 2011-03-07 16:04:44 -0600 (Mon, 07 Mar 2011) | 15 lines
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r309856 | jrose | 2011-03-07 16:02:12 -0600 (Mon, 07 Mar 2011) | 8 lines
Bug fix for MixMonitor involving filenames with '.' not in the extension
Closes issue #18391)
Reported by: pabelanger
Patches:
bugfix.patch uploaded by jrose (license 1225)
Tested by: jrose
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r309403 | diruggles | 2011-03-03 20:50:44 -0500 (Thu, 03 Mar 2011) | 23 lines
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r309356 | diruggles | 2011-03-03 19:42:28 -0500 (Thu, 03 Mar 2011) | 16 lines
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r309355 | diruggles | 2011-03-03 19:34:13 -0500 (Thu, 03 Mar 2011) | 9 lines
fix small memory leak
fix small memory leak caused by a string allocation that wasn't freed
(closes issue #18907)
Reported by: andy11
Patches:
asterisk_trunk-app_externalivr-leak.patch uploaded by andy11 (license 1224)
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audio ConfBridge, and other stuff
-Functional changes
1. Dynamic global format list build by codecs defined in codecs.conf
2. SILK 8khz, 12khz, 16khz, and 24khz with custom attributes defined in codecs.conf
3. Negotiation of SILK attributes in chan_sip.
4. SPEEX 32khz with translation
5. SLINEAR 8khz, 12khz, 24khz, 32khz, 44.1khz, 48khz, 96khz, 192khz with translation
using codec_resample.c
6. Various changes to RTP code required to properly handle the dynamic format list
and formats with attributes.
7. ConfBridge now dynamically jumps to the best possible sample rate. This allows
for conferences to take advantage of HD audio (Which sounds awesome)
8. Audiohooks are no longer limited to 8khz audio, and most effects have been
updated to take advantage of this such as Volume, DENOISE, PITCH_SHIFT.
9. codec_resample now uses its own code rather than depending on libresample.
-Organizational changes
Global format list is moved from frame.c to format.c
Various format specific functions moved from frame.c to format.c
Review: https://reviewboard.asterisk.org/r/1104/
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r308010 | qwell | 2011-02-15 17:34:03 -0600 (Tue, 15 Feb 2011) | 24 lines
Merged revisions 308007 via svnmerge from
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r308007 | qwell | 2011-02-15 17:33:24 -0600 (Tue, 15 Feb 2011) | 17 lines
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r308002 | qwell | 2011-02-15 17:32:20 -0600 (Tue, 15 Feb 2011) | 10 lines
Fix regression that changed behavior of queues when ringing a queue member.
This reverts r298596, which was to fix a highly bizarre and contrived issue
with a queue member that called into his own queue being transferred back
into his own queue. I couldn't reproduce that issue in any way. I think one
of the other recent transfer fixes actually fixed this.
(closes issue #18747)
Reported by: vrban
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r307962 | rmudgett | 2011-02-15 13:52:45 -0600 (Tue, 15 Feb 2011) | 1 line
Don't crash when forcing caller id.
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r307750 | tilghman | 2011-02-14 00:50:23 -0600 (Mon, 14 Feb 2011) | 23 lines
Calling a gosub routine defined in AEL from Dial/Queue ceased to work.
A bug in AEL did not distinguish between the "s" extension generated by
AEL and an "s" extension that was required to exist by the chan_dahdi
(or another channel) that was not supplied with a starting extension.
Therefore, AEL made incorrect assumptions about what commands were
permissable in the context. This was fixed by making AEL generate a
different extension name. However, Dial and Queue make additional
assumptions about the name of the default gosub extension. Therefore,
they needed to be brought into line with a "macro" rendered by AEL (as
a gosub), without breaking traditional dialplans written without the
aid of AEL.
Related to (issue #18480)
Reported by: nivek
(closes issue #18729)
Reported by: kkm
Patches:
20110209__issue18729.diff.txt uploaded by tilghman (license 14)
018729-dial-queue-gosub-try3.patch uploaded by kkm (license 888)
Tested by: kkm
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From the submitter:
I've added a new manager action to list only the active conferences on an
Asterisk system. It shows the same data displayed when you run a 'meetme list'
on the Asterisk CLI.
(closes issue #17905)
Reported by: rcasas
Patches:
app_meetme.c.patch uploaded by rcasas (license 641)
Review: https://reviewboard.asterisk.org/r/874/
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r306967 | jpeeler | 2011-02-08 13:41:42 -0600 (Tue, 08 Feb 2011) | 16 lines
Merged revisions 306966 via svnmerge from
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r306966 | jpeeler | 2011-02-08 13:41:21 -0600 (Tue, 08 Feb 2011) | 9 lines
Merged revisions 306965 via svnmerge from
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r306965 | jpeeler | 2011-02-08 13:40:58 -0600 (Tue, 08 Feb 2011) | 1 line
fix this line again
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r306962 | jpeeler | 2011-02-08 13:25:38 -0600 (Tue, 08 Feb 2011) | 22 lines
Merged revisions 306961 via svnmerge from
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r306961 | jpeeler | 2011-02-08 13:25:10 -0600 (Tue, 08 Feb 2011) | 15 lines
Merged revisions 306960 via svnmerge from
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r306960 | jpeeler | 2011-02-08 13:18:50 -0600 (Tue, 08 Feb 2011) | 9 lines
Backup file storing message duration is not used with IMAP_STORAGE, remove code.
The message duration is stored in the body of the email when using IMAP_STORAGE,
so nothing needs to happen with the backup file.
(closes issue #18718)
Reported by: kerframil
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r306866 | jpeeler | 2011-02-08 10:21:45 -0600 (Tue, 08 Feb 2011) | 16 lines
Merged revisions 306865 via svnmerge from
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r306865 | jpeeler | 2011-02-08 10:21:25 -0600 (Tue, 08 Feb 2011) | 9 lines
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r306864 | jpeeler | 2011-02-08 10:19:17 -0600 (Tue, 08 Feb 2011) | 1 line
make this safer and fully correct, pointed out by Steve Davis
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The display ie handling can be controlled independently in the send and
receive directions with the following options:
* Block display text data.
* Use display text in SETUP/CONNECT messages for name.
* Use display text for COLP name updates (FACILITY/NOTIFY as appropriate).
* Pass arbitrary display text during a call. Sent in INFORMATION
messages. Received from any message that the display text was not used as
a name.
If the display options are not set then the options default to legacy
behavior.
The arbitrary display text is exchanged between bridged channels using the
AST_FRAME_TEXT frame type.
To send display text from the dialplan use the SendText() application when
the arbitrary display text option is enabled.
JIRA SWP-2688
JIRA ABE-2693
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r306356 | qwell | 2011-02-04 13:24:29 -0600 (Fri, 04 Feb 2011) | 16 lines
Merged revisions 306346 via svnmerge from
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r306346 | qwell | 2011-02-04 13:21:43 -0600 (Fri, 04 Feb 2011) | 9 lines
Don't fallthrough to 'unknown' in the 'ringing' case.
This could cause improper exits from the queue.
(closes issue #18499)
Reported by: zaltar
Patches:
app_queue.patch uploaded by zaltar (license 1148)
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r306324 | rmudgett | 2011-02-04 12:53:06 -0600 (Fri, 04 Feb 2011) | 9 lines
Don't send redirecting updates to the caller if the dialplan forked the call.
Each fork in the dial could be redirected and confuse the caller. For
ISDN the DivLeg1 and DivLeg3 messages would get confused because ISDN
redirects calls in sequence not in parallel.
* Also fixed a formatting inconsistency in app_dial.c and make a warning
message more useful about what frame type could not be written.
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(closes issue #18556)
Reported by: kkm
Review: https://reviewboard.asterisk.org/r/1071/
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This patch is the foundation of an entire new way of looking at media in Asterisk.
The code present in this patch is everything required to complete phase1 of my
Media Architecture proposal. For more information about this project visit the link below.
https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal
The primary function of this patch is to convert all the usages of format
bitfields in Asterisk to use the new format and format_cap APIs. Functionally
no change in behavior should be present in this patch. Thanks to twilson
and russell for all the time they spent reviewing these changes.
Review: https://reviewboard.asterisk.org/r/1083/
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r305923 | rmudgett | 2011-02-02 18:24:40 -0600 (Wed, 02 Feb 2011) | 24 lines
Merged revisions 305889 via svnmerge from
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r305889 | rmudgett | 2011-02-02 18:15:07 -0600 (Wed, 02 Feb 2011) | 17 lines
Merged revisions 305888 via svnmerge from
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r305888 | rmudgett | 2011-02-02 18:02:43 -0600 (Wed, 02 Feb 2011) | 8 lines
Minor AST_FRAME_TEXT related issues.
* Include the null terminator in the buffer length. When the frame is
queued it is copied. If the null terminator is not part of the frame
buffer length, the receiver could see garbage appended onto it.
* Add channel lock protection with ast_sendtext().
* Fixed AMI SendText action ast_sendtext() return value check.
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Adding links to http(s)://wiki.asterisk.org
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(closes issue #14389)
(closes issue #18007)
Reported by: jcollie
Patches:
0001-Fix-up-bridging-module-so-that-menuselect-works.patch uploaded by jcollie (license 412)
0002-Add-confbridge-list-and-confbridge-kick-CLI-comm.patch uploaded by jcollie (license 412)
Tested by: file
Review: https://reviewboard.asterisk.org/r/1084/
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r305254 | qwell | 2011-01-31 17:07:00 -0600 (Mon, 31 Jan 2011) | 24 lines
Merged revisions 305253 via svnmerge from
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r305253 | qwell | 2011-01-31 16:59:34 -0600 (Mon, 31 Jan 2011) | 17 lines
Merged revisions 305252 via svnmerge from
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r305252 | qwell | 2011-01-31 16:56:54 -0600 (Mon, 31 Jan 2011) | 10 lines
Prevent a crash when dialing a technology with no destination (ex: Dial(SIP/))
chan_iax2 and other channel drivers already had code to prevent this. The
attempt that app_dial was making to prevent it was not correct, so I fixed that.
(closes issue #18371)
Reported by: gbour
Patches:
18371.patch uploaded by gbour (license 1162)
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r304985 | tilghman | 2011-01-31 01:27:13 -0600 (Mon, 31 Jan 2011) | 16 lines
Merged revisions 304978 via svnmerge from
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r304978 | tilghman | 2011-01-31 01:25:14 -0600 (Mon, 31 Jan 2011) | 9 lines
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r304952 | tilghman | 2011-01-31 00:54:45 -0600 (Mon, 31 Jan 2011) | 2 lines
Fix compilation when ODBC_STORAGE is defined.
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Add and extend the see-also sections to the documentation for applications
and functions in an effort to expand the online documentation of the wiki.
Also check for and update any links to moved documentation in the doc folder.
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r304777 | seanbright | 2011-01-29 13:09:37 -0500 (Sat, 29 Jan 2011) | 22 lines
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r304776 | seanbright | 2011-01-29 13:08:14 -0500 (Sat, 29 Jan 2011) | 15 lines
If we fail to allocate our announcement objects, make sure we don't leak objects.
The majority of this patch was committed already in r304726 and r304729.
(issue #18225)
Reported by: kenji
(issue #18444)
Reported by: junky
(closes issue #18343)
Reported by: kobaz
Patches:
meetme-refs.diff uploaded by kobaz (license 834)
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r304774 | seanbright | 2011-01-29 12:54:43 -0500 (Sat, 29 Jan 2011) | 16 lines
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r304773 | seanbright | 2011-01-29 12:51:28 -0500 (Sat, 29 Jan 2011) | 9 lines
When we pass the S() or L() options to MeetMe, make sure that we honor C as well.
Without this patch, if the user was kicked from the conference via the S() or L()
mechanism, we would just hang up on them even if we also passed C (continue in
dialplan when kicked). With this patch we honor the C flag in those cases.
(closes issue #17317)
Reported by: var
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r304730 | seanbright | 2011-01-29 12:15:27 -0500 (Sat, 29 Jan 2011) | 22 lines
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r304729 | seanbright | 2011-01-29 12:01:51 -0500 (Sat, 29 Jan 2011) | 15 lines
Make sure that we unref the correct object when ejecting the most recent caller.
Currently, when we kick the last user to enter, we decrement our own reference
count which results in a crash when we kick another user or when we exit the
conference ourselves.
This will fix #18225 in 1.8 and trunk, but that particular bug does not exist in
1.6.2.
(closes issue #18225)
Reported by: kenji
Patches:
issue18225.patch uploaded by seanbright (license 71)
Tested by: seanbright
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r304727 | seanbright | 2011-01-29 11:28:27 -0500 (Sat, 29 Jan 2011) | 16 lines
Merged revisions 304726 via svnmerge from
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r304726 | seanbright | 2011-01-29 11:26:57 -0500 (Sat, 29 Jan 2011) | 9 lines
Fix user reference leak in MeetMe.
We were unlinking the user from the conferences user container, but not
decrementing the reference count of the user as well, resulting in a leak.
(closes issue #18444)
Reported by: junky
Tested by: seanbright
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r304683 | seanbright | 2011-01-28 17:54:23 -0500 (Fri, 28 Jan 2011) | 16 lines
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r304659 | seanbright | 2011-01-28 16:22:09 -0500 (Fri, 28 Jan 2011) | 5 lines
Don't leak references if we can't create a pseudo channel for mixing in MeetMe.
If there was a problem allocating a pseudo channel when building our meetme, we
weren't destroying our user container or destroying the mutexes that we created.
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r304682 | seanbright | 2011-01-28 17:38:05 -0500 (Fri, 28 Jan 2011) | 2 lines
Revert part of the previous commit that snuck in.
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Followme answers an incoming call if it hasn't already been answered and starts
MOH. Some poorly designed autodialers see the answer and start playing their
message to the hold music. The 'N' option has been added to indicate ringing and
not answer until the call is accepted.
(closes issue #18479)
Reported by: ianc
Patches:
trunk_followme.diff uploaded by ianc (license 998)
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r303678 | jpeeler | 2011-01-25 11:02:38 -0600 (Tue, 25 Jan 2011) | 33 lines
Merged revisions 303677 via svnmerge from
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r303677 | jpeeler | 2011-01-25 10:59:28 -0600 (Tue, 25 Jan 2011) | 26 lines
Merged revisions 303676 via svnmerge from
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r303676 | jpeeler | 2011-01-25 10:58:29 -0600 (Tue, 25 Jan 2011) | 20 lines
Fix voicemail sequencing for file based storage.
A previous change was made to account for when the number of voicemail messages
exceeds the max limit to be handled properly, but it caused gaps in the messages
to not be properly handled. This has now been resolved.
In later non 1.4 branches, it appears that resequencing wasn't even occurring
due from what appears and accidental code removal.
(closes issue #18498)
Reported by: JJCinAZ
Patches:
bug18498v2.patch uploaded by jpeeler (license 325)
(closes issue #18486)
Reported by: bluefox
Patches:
bug18486.patch uploaded by jpeeler (license 325)
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r303549 | russell | 2011-01-24 14:51:37 -0600 (Mon, 24 Jan 2011) | 45 lines
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r303548 | russell | 2011-01-24 14:49:53 -0600 (Mon, 24 Jan 2011) | 38 lines
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r303546 | russell | 2011-01-24 14:32:21 -0600 (Mon, 24 Jan 2011) | 31 lines
Fix channel redirect out of MeetMe() and other issues with channel softhangup.
Mantis issue #18585 reports that a channel redirect out of MeetMe() stopped
working properly. This issue includes a patch that resolves the issue by
removing a call to ast_check_hangup() from app_meetme.c. I left that in my
patch, as it doesn't need to be there. However, the rest of the patch fixes
this problem with or without the change to app_meetme.
The key difference between what happens before and after this patch is the
effect of the END_OF_Q control frame. After END_OF_Q is hit in ast_read(),
ast_read() will return NULL. With the ast_check_hangup() removed, app_meetme
sees this which causes it to exit as intended. Checking ast_check_hangup()
caused app_meetme to exit earlier in the process, and the target of the
redirect saw the condition where ast_read() returned NULL.
Removing ast_check_hangup() works around the issue in app_meetme, but doesn't
solve the issue if another application did the same thing. There are also
other edge cases where if an application finishes at the same time that a
redirect happens, the target of the redirect will think that the channel hung
up. So, I made some changes in pbx.c to resolve it at a deeper level. There
are already places that unset the SOFTHANGUP_ASYNCGOTO flag in an attempt to
abort the hangup process. My patch extends this to remove the END_OF_Q frame
from the channel's read queue, making the "abort hangup" more complete. This
same technique was used in every place where a softhangup flag was cleared.
(closes issue #18585)
Reported by: oej
Tested by: oej, wedhorn, russell
Review: https://reviewboard.asterisk.org/r/1082/
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r303009 | jpeeler | 2011-01-20 11:10:32 -0600 (Thu, 20 Jan 2011) | 21 lines
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r303008 | jpeeler | 2011-01-20 11:07:44 -0600 (Thu, 20 Jan 2011) | 14 lines
Merged revisions 303007 via svnmerge from
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r303007 | jpeeler | 2011-01-20 11:04:08 -0600 (Thu, 20 Jan 2011) | 8 lines
Add new queue strategy to preserve behavior for when queue members moved to ao2.
Add queue strategy called "rrordered" to mimic old behavior from when queue
members were stored in a linked list.
ABE-2707
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r302921 | russell | 2011-01-20 10:12:15 -0600 (Thu, 20 Jan 2011) | 9 lines
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r302920 | russell | 2011-01-20 10:11:58 -0600 (Thu, 20 Jan 2011) | 2 lines
Resolve a compiler warning.
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r302918 | lmadsen | 2011-01-20 09:45:39 -0600 (Thu, 20 Jan 2011) | 16 lines
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r302917 | lmadsen | 2011-01-20 09:42:05 -0600 (Thu, 20 Jan 2011) | 8 lines
Option L() is milliseconds, not seconds.
> Change the verbose output of option L() to say milliseconds and not seconds
> as the value is in milliseconds.
>
> (closes issue #18264)
> Reported by: jacco
> Patches:
> app_dial_patch.txt uploaded by lmadsen (license 10)
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r302834 | seanbright | 2011-01-19 18:49:00 -0500 (Wed, 19 Jan 2011) | 14 lines
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r302833 | seanbright | 2011-01-19 18:47:22 -0500 (Wed, 19 Jan 2011) | 7 lines
Support greetingsfolder as documented in voicemail.conf.sample.
(closes issue #17870)
Reported by: edhorton
Patches:
__20100816-app_voicemail-greetingsfolder-support.txt uploaded by lmadsen (license 10)
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r301177 | pabelanger | 2011-01-08 17:00:12 -0500 (Sat, 08 Jan 2011) | 14 lines
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r301176 | pabelanger | 2011-01-08 16:58:24 -0500 (Sat, 08 Jan 2011) | 7 lines
Indicate log level argument for Log() is not optional
(closes issue #18586)
Reported by: kshumard
Patches:
app_verbose.c.patch uploaded by kshumard (license 92)
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r301090 | qwell | 2011-01-07 14:53:02 -0600 (Fri, 07 Jan 2011) | 15 lines
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r301089 | qwell | 2011-01-07 14:52:00 -0600 (Fri, 07 Jan 2011) | 8 lines
Initialize useropts/adminopts in case there is no column in the realtime DB.
(closes issue #18182)
Reported by: dimas
Patches:
v1-18182.patch uploaded by dimas (license 88)
Tested by: dimas
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r301047 | jpeeler | 2011-01-07 13:58:30 -0600 (Fri, 07 Jan 2011) | 15 lines
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r301046 | jpeeler | 2011-01-07 13:57:42 -0600 (Fri, 07 Jan 2011) | 8 lines
Fix regression causing forwarding voicemails to not work with file storage.
I had actually already fixed this in 295200 in 1.4 and thought it wasn't
missing in the other branches for some reason.
(closes issue #18358)
Reported by: cabal95
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r300955 | jpeeler | 2011-01-07 11:24:14 -0600 (Fri, 07 Jan 2011) | 21 lines
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r300951 | jpeeler | 2011-01-07 11:23:37 -0600 (Fri, 07 Jan 2011) | 14 lines
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r300918 | jpeeler | 2011-01-07 11:13:21 -0600 (Fri, 07 Jan 2011) | 7 lines
Ensure good bye prompt in voicemail is played at the correct time.
Specifically in the case of timing out but not leaving voicemail nothing
should be heard. And when leaving voicemail it should be heard.
ABE-2647
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playing_silence was not initialized with the struct
was initialized, it was being set after the fact
which caused problems if something that relied on
playing_silence being set was called too quickly
(closes issue #18430)
Reported by: stevebrandli
Patches:
externalivr.patch uploaded by thedavidfactor (license 903)
Tested by: thedavidfactor, stevebrandli
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r299989 | tilghman | 2010-12-29 16:02:59 -0600 (Wed, 29 Dec 2010) | 4 lines
Quote arguments, just in case there's a space in a pathname.
(Diagnosed by pabelanger on #asterisk-dev, fixed by me.)
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r299865 | pabelanger | 2010-12-28 13:53:37 -0500 (Tue, 28 Dec 2010) | 9 lines
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r299864 | pabelanger | 2010-12-28 13:51:13 -0500 (Tue, 28 Dec 2010) | 2 lines
Documentation typo
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r298685 | jpeeler | 2010-12-16 17:31:50 -0600 (Thu, 16 Dec 2010) | 16 lines
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r298684 | jpeeler | 2010-12-16 17:30:59 -0600 (Thu, 16 Dec 2010) | 9 lines
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r298683 | jpeeler | 2010-12-16 17:29:30 -0600 (Thu, 16 Dec 2010) | 2 lines
After recording only silence for a voicemail prepending, restore backup files.
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r298598 | jpeeler | 2010-12-16 14:51:44 -0600 (Thu, 16 Dec 2010) | 21 lines
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r298597 | jpeeler | 2010-12-16 14:49:33 -0600 (Thu, 16 Dec 2010) | 14 lines
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r298596 | jpeeler | 2010-12-16 14:46:52 -0600 (Thu, 16 Dec 2010) | 7 lines
Fix improper hangup when doing an attended transfer to queue.
Had to indicate ringing in wait_for_answer so the attended transfer code would
not try and hang up the local channel it created, which would kill the call.
ABE-2624
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r297733 | tilghman | 2010-12-06 18:29:26 -0600 (Mon, 06 Dec 2010) | 22 lines
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r297713 | tilghman | 2010-12-06 18:21:50 -0600 (Mon, 06 Dec 2010) | 15 lines
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r297689 | tilghman | 2010-12-06 18:07:37 -0600 (Mon, 06 Dec 2010) | 8 lines
Don't create a Local channel if the target extension does not exist.
(closes issue #18126)
Reported by: junky
Patches:
followme.diff uploaded by junky (license 177)
(partially restructured by me to avoid a possible memory leak)
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r297245 | russell | 2010-12-02 07:20:19 -0600 (Thu, 02 Dec 2010) | 20 lines
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r297229 | russell | 2010-12-02 07:16:47 -0600 (Thu, 02 Dec 2010) | 13 lines
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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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r296870 | jpeeler | 2010-11-30 18:28:16 -0600 (Tue, 30 Nov 2010) | 18 lines
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r296869 | jpeeler | 2010-11-30 18:24:58 -0600 (Tue, 30 Nov 2010) | 11 lines
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r296868 | jpeeler | 2010-11-30 18:23:19 -0600 (Tue, 30 Nov 2010) | 4 lines
Properly restore backup information file when hanging up during message prepending.
ABE-2654
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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
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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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