Russell Bryant
ad720f24bb
The IMAP storage code uses the same code to build the email that is used when
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voicemail is sent via email using something like sendmail. In the patch from
bug 8033 to fix various IMAP storage problems, the line endings in the email
file were changed in the code from "\n" to "\r\n". However, this breaks
sending regular voicemail to email. So, this change conditionally sets line
endings to "\r\n" only if IMAP_STORAGE is enabled.
(issue #9128 , patch by jarjarbinks, modified by me to not break IMAP storage)
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56341 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-23 17:58:57 +00:00
Russell Bryant
f314685447
Merge changes from team/russell/sla_updates.
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This batch of changes to the SLA code does a few different things.
* I made the SLA code event driven instead of having to act in a lot of busy
loops while dialing things to wait for state changes. This makes the code
more efficient and readable at the same time.
* I have implemented a couple of new features. The first is inbound trunk
ringing timeouts. This is an option that defines how long to let an incoming
call on a trunk to ring.
* I have also implemented ring timeouts for stations. They may be specified
for the entire station, meaning it is how long to let the station ring before
giving up. You can also specify a ring timeout for a specific trunk on a
station. So, you can say that you only want a specific station to ring 5
seconds if it is line1 ringing, but otherwise, there is no timeout.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@56277 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-22 23:08:36 +00:00
Joshua Colp
fee3b74617
Merged revisions 55956 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r55956 | file | 2007-02-21 15:32:16 -0500 (Wed, 21 Feb 2007) | 2 lines
Change naughty warning message to provide useful information. If a write now fails on a channel in meetme it will tell you the channel name instead of spitting out the wrong error message.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55957 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-21 20:35:40 +00:00
Russell Bryant
bb3aa2ee41
Simplify the last change to app_meetme, and move the call to dispose_conf()
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up into the block where we know a conf exists.
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2007-02-21 20:22:33 +00:00
Joshua Colp
54f59f4dad
Only dispose of the conference if one was created.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55949 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-21 20:16:34 +00:00
Joshua Colp
10864c42ea
Only start playing the next file if we have not been quieted.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55947 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-21 20:03:38 +00:00
Russell Bryant
39026c9f96
Improve the reference counting to fix bugs where people report seeing
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conferences listed that have no members.
(issue #9073 )
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55758 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-21 01:03:25 +00:00
Joshua Colp
51f20d54e7
Better handle dropped IMAP connections. (issue #9054 reported by bsmithurst)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55741 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-21 00:11:20 +00:00
Tilghman Lesher
f0bf09466a
Merged revisions 55434 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r55434 | tilghman | 2007-02-19 12:09:09 -0600 (Mon, 19 Feb 2007) | 2 lines
forcename and forcegreetings options should check to see if the recording already exists
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2007-02-19 18:11:48 +00:00
Olle Johansson
476d1c932e
Merged revisions 55277 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r55277 | oej | 2007-02-18 13:32:13 +0100 (Sun, 18 Feb 2007) | 2 lines
Documentation update (#9053 , jsmith)
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2007-02-18 12:35:54 +00:00
Joshua Colp
7cec18d933
Add missing membername option to AddQueueMember documentation. (issue #9088 reported by seanbright)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55219 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-17 17:39:32 +00:00
Joshua Colp
ec6e4f6887
Merged revisions 55153 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r55153 | file | 2007-02-16 22:53:45 -0500 (Fri, 16 Feb 2007) | 2 lines
Answer the channel before recording privacy information. (issue #8926 reported by lmamane)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55154 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-17 03:55:30 +00:00
Joshua Colp
fccaadad94
Make the 'i' option of Queue actually work. (issue #8986 reported by utis)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55129 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-17 02:59:50 +00:00
Russell Bryant
960b4de2de
Merged revisions 55005 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r55005 | russell | 2007-02-16 16:48:22 -0600 (Fri, 16 Feb 2007) | 9 lines
Revert the change I did in revisions 54955, 54969, and 54970, in 1.2, 1.4,
and trunk. I decided that once a conference is created from meetme.conf,
it is acceptable behavior that the pin can not be changed until the
conference goes away. I also added a note in meetme.conf to describe this
behavior.
We still have another issue in 1.4 and trunk where some conferences with no
users don't go away. That is the real bug that needs to be addressed here.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@55006 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-16 22:49:42 +00:00
Russell Bryant
cad10249de
Merged revisions 54955 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r54955 | russell | 2007-02-16 14:56:58 -0600 (Fri, 16 Feb 2007) | 5 lines
For conferences that are configured in meetme.conf, check the configuration
file every time someone joins the conference instead of only when the
conference is first created. This is to ensure that changes to the pin
numbers in the config file are always honored. (issue #9073 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54969 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-16 21:12:18 +00:00
Joshua Colp
3a1ceb4ff8
Need to check macro extension as well as macro context for directed pickup.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-16 18:51:15 +00:00
Joshua Colp
7722d96bea
Allow directed pickup to pick up the real context instead of the macro context if a Macro is used. (issue #8984 reported by jamesb63)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54884 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-16 17:02:35 +00:00
Joshua Colp
ae5eec37dd
Don't let dtmf leak over into the engine and let it skew the results... also give DTMF results priority. (issue #9014 reported by surftek)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54714 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-16 00:48:48 +00:00
Joshua Colp
bddfe6fea7
Merged revisions 54622 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r54622 | file | 2007-02-15 11:14:40 -0500 (Thu, 15 Feb 2007) | 2 lines
Use a separate variable to indicate execution should continue instead of the return value. (issue #8842 reported by pluto70)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-15 16:19:39 +00:00
Joshua Colp
dadf652e26
Forward begin DTMF frames as well as end. (issue #9068 reported by mhardeman)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-14 21:07:23 +00:00
Russell Bryant
5bc6ee1714
- Add the ability to register a callback to monitor state changes in an
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asynchronous dial operation.
- Rename the various references to "status" to "state" in the dial API
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@54066 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-12 17:58:43 +00:00
Paul Cadach
f02a11732b
Much simpler than previous one ;-)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53880 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-10 09:08:55 +00:00
Russell Bryant
c16059235e
Add some output for "show application SLAStation/SLATrunk"
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53821 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-10 01:02:00 +00:00
Russell Bryant
7ee02f585d
Merge team/russell/sla_rewrite
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This is a completely new implementation of the SLA functionality introduced in
Asterisk 1.4. It is now functional and ready for testing. However, I will be
adding some additional features over the next week, as well.
For information on how to set this up, see configs/sla.conf.sample
and doc/sla.txt.
In addition to the changes in app_meetme.c for the SLA implementation itself,
this merge brings in various other changes:
chan_sip:
- Add the ability to indicate HOLD state in NOTIFY messages.
- Queue HOLD and UNHOLD control frames even if the channel is not bridged to
another channel.
linkedlists.h:
- Add support for rwlock based linked lists.
dial.c:
- Add the ability to run ast_dial_start() without a reference channel to
inherit information from.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53810 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-10 00:35:09 +00:00
Russell Bryant
2f5eabb51e
When the Echo() application receives the digit '#', echo that back as well.
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Since we already sent the BEGIN frame for that digit, it makes sense to send
the END as well.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53783 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-10 00:15:50 +00:00
Kevin P. Fleming
07462b8d1b
add some inter-module dependencies
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53780 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-09 23:51:41 +00:00
Joshua Colp
1ba2aa702d
Temporarily change musicclass on channel to one specified in Dial so that the 'm' option functions properly. (issue #8969 reported by christianbee)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-09 19:33:31 +00:00
Joshua Colp
dce4b744a1
Fix timeout issue when utterance is longer then timeout itself.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53601 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-08 17:54:32 +00:00
Tilghman Lesher
1612f65f9c
Merged revisions 53529 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r53529 | tilghman | 2007-02-08 07:36:10 -0600 (Thu, 08 Feb 2007) | 2 lines
Issue 9003 - If fullname is empty, quote() passes back "\""
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-08 13:40:02 +00:00
Joshua Colp
670a0ff144
Directly load say.conf in load_module instead of calling the reload function. (issue #8946 reported by junky)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53399 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-07 17:04:44 +00:00
Tilghman Lesher
7f0c317655
Merged revisions 53354 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r53354 | tilghman | 2007-02-07 09:30:02 -0600 (Wed, 07 Feb 2007) | 2 lines
Issue 7440 - Macro called from Macro from the h extension exits prematurely
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53355 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-07 15:33:51 +00:00
Joshua Colp
687b672a09
Ensure say_cfg is NULL when the module is loaded. (issue #8946 reported by junky)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53152 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-05 17:06:18 +00:00
Joshua Colp
e647794710
Unregister Playback CLI commands as well as dialplan application. (issue #8946 reported by junky)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53150 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-05 16:02:00 +00:00
Russell Bryant
8f04a45ee9
Merged revisions 53133 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r53133 | russell | 2007-02-03 14:38:13 -0600 (Sat, 03 Feb 2007) | 4 lines
set the DIALSTATUS variable to contain "INVALIDARGS" when the dial application
exits early because of invalid arguments instead of just leaving it empty.
(issue #8975 )
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2007-02-03 20:44:20 +00:00
BJ Weschke
c71401e705
I'm baaaaaaaaaack. :)
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Post a warning to the console that things might possibly be misconfigured when queue member's states are still 'Not in Use' when we're about to bridge them with a caller from queue. Also, put some documentation quoted from oej's queues.txt efforts started in /trunk today.
This commit puts #7433 into feedback state for 1.4, and pending no further negative feedback, it will finally be closed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53129 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-02 21:26:45 +00:00
Olle Johansson
10f0da24c2
Change debug level for state change message that is not really informative when debugging app_queue
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-01 20:38:58 +00:00
Russell Bryant
9aab046002
Merged revisions 53045 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 Jan 2007) | 3 lines
Fix a bunch of places where pthread_attr_init() was called, but
pthread_attr_destroy() was not.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53046 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-31 21:32:08 +00:00
Russell Bryant
43e68cac9a
Remove an extra \r\n from manager user events. (issue #8955 , mnicholson)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-31 18:18:25 +00:00
Russell Bryant
ad254da05d
Only changed the paused status in an existing queue member if the paused
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column exists.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@53037 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-31 17:39:28 +00:00
Russell Bryant
8dc0a7778f
Instead of always creating a realtime queue member as unpaused, read the
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"paused" column and use that value for the paused status of the member.
(issue #8949 , jmls)
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2007-01-31 17:34:22 +00:00
Joshua Colp
f92505eec1
Merged revisions 52716 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r52716 | file | 2007-01-29 18:39:39 -0500 (Mon, 29 Jan 2007) | 2 lines
Now that filename is part of the structure and since it comes before postprocess... we have to add it to our postprocess line. (reported on asterisk-dev by Boris Bakchiev)
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2007-01-29 23:43:40 +00:00
Joshua Colp
a4d3a2dbc9
Use ast_calloc instead of malloc.
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2007-01-29 18:59:41 +00:00
Steve Murphy
f2fab700c3
this is for 8778 (pt_BR backport to 1.4). It was committed to trunk via 7663. But it wasn't so much an enhancement as a fix for the bad language output for portuguese in Brazil, so, after a lot of prodding from patient Brazilians, here is the same fix for 1.4
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52535 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-29 17:57:08 +00:00
Joshua Colp
e67ade2403
Set quota information to 0 when creating a vm_state. (issue #8924 reported by neutrino88)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52523 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-29 17:33:19 +00:00
Joshua Colp
47f9dd508a
Merged revisions 52415 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r52415 | file | 2007-01-26 21:09:10 -0500 (Fri, 26 Jan 2007) | 2 lines
Make COMPLETECALLER and COMPLETEAGENT output to queue_log follow documentation. (issue #7677 reported by amilcar)
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2007-01-27 02:13:41 +00:00
Joshua Colp
6124ace110
Merged revisions 52162 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r52162 | file | 2007-01-24 20:48:52 -0500 (Wed, 24 Jan 2007) | 2 lines
Add another note about audio files being played back to each bridged party. (issue #8718 reported by ppyy)
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2007-01-25 01:51:35 +00:00
Russell Bryant
96beb30159
By suggestion from kpfleming last week, change "vmpassword" to "vmsecret".
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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@52160 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-25 01:37:16 +00:00
Joshua Colp
8acccb9254
Merge in dialing API and the app_page that uses it. (issue #BE-118)
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2007-01-24 18:20:05 +00:00
Russell Bryant
d7d0c10c21
Merged revisions 51828 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r51828 | russell | 2007-01-23 18:17:50 -0600 (Tue, 23 Jan 2007) | 4 lines
Don't set a new value for the END_ variable on the channel before using the
old value. If you do, it will lead to accessing a memory address that has
been free()'d. (issue #8895 , arkadia)
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2007-01-24 00:19:55 +00:00
Joshua Colp
c60c2b5ffc
Merged revisions 51406 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r51406 | file | 2007-01-22 14:08:52 -0500 (Mon, 22 Jan 2007) | 2 lines
Move filestream creation to Mixmonitor loop. This will prevent a blank file from being created if no frames ever pass through to be recorded. (issue #7589 reported by steve_mcneil)
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2007-01-22 19:13:44 +00:00