[asterisk-commits] tilghman: branch group/ast_storage r53282 - in
/team/group/ast_storage: ./ ap...
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Tue Feb 6 06:47:16 MST 2007
Author: tilghman
Date: Tue Feb 6 07:47:14 2007
New Revision: 53282
URL: http://svn.digium.com/view/asterisk?view=rev&rev=53282
Log:
Merged revisions 53036,53038,53041,53043,53047,53051,53053-53054,53058-53059,53061,53063,53065-53067,53071,53073,53076,53078,53080,53082-53083,53087,53089,53092,53094,53098,53100,53105,53110-53113,53115,53119,53122,53125-53127,53132,53137,53139-53142,53144,53151,53153,53200,53247 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r53036 | russell | 2007-01-31 11:35:14 -0600 (Wed, 31 Jan 2007) | 12 lines
Merged revisions 53035 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53035 | russell | 2007-01-31 11:34:22 -0600 (Wed, 31 Jan 2007) | 4 lines
Instead of always creating a realtime queue member as unpaused, read the
"paused" column and use that value for the paused status of the member.
(issue #8949, jmls)
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r53038 | russell | 2007-01-31 11:39:55 -0600 (Wed, 31 Jan 2007) | 11 lines
Merged revisions 53037 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53037 | russell | 2007-01-31 11:39:28 -0600 (Wed, 31 Jan 2007) | 3 lines
Only changed the paused status in an existing queue member if the paused
column exists.
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r53041 | russell | 2007-01-31 11:45:43 -0600 (Wed, 31 Jan 2007) | 19 lines
Merged revisions 53040 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53040 | russell | 2007-01-31 11:45:05 -0600 (Wed, 31 Jan 2007) | 11 lines
Merged revisions 53039 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r53039 | russell | 2007-01-31 11:41:51 -0600 (Wed, 31 Jan 2007) | 3 lines
Use the proper format string to print unsigned values in the rtp debug output.
(issue #8954, wmis)
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r53043 | russell | 2007-01-31 12:18:58 -0600 (Wed, 31 Jan 2007) | 10 lines
Merged revisions 53042 via svnmerge from
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r53042 | russell | 2007-01-31 12:18:25 -0600 (Wed, 31 Jan 2007) | 2 lines
Remove an extra \r\n from manager user events. (issue #8955, mnicholson)
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r53047 | russell | 2007-01-31 15:35:15 -0600 (Wed, 31 Jan 2007) | 19 lines
Merged revisions 53046 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53046 | russell | 2007-01-31 15:32:08 -0600 (Wed, 31 Jan 2007) | 11 lines
Merged revisions 53045 via svnmerge from
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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r53051 | file | 2007-01-31 18:23:19 -0600 (Wed, 31 Jan 2007) | 10 lines
Merged revisions 53050 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53050 | file | 2007-01-31 18:19:48 -0600 (Wed, 31 Jan 2007) | 2 lines
Add more frame types to forward in the RTP bridge loops.
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r53053 | file | 2007-01-31 18:24:50 -0600 (Wed, 31 Jan 2007) | 10 lines
Merged revisions 53052 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53052 | file | 2007-01-31 18:24:20 -0600 (Wed, 31 Jan 2007) | 2 lines
When going on hold have the side that was put on hold reinvite back to Asterisk. When going off hold have the side that was taken off hold reinvited back to the other party.
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r53054 | oej | 2007-01-31 18:38:43 -0600 (Wed, 31 Jan 2007) | 2 lines
Formatting changes
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r53058 | pcadach | 2007-02-01 05:10:35 -0600 (Thu, 01 Feb 2007) | 8 lines
Blocked revisions 53057 via svnmerge
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r53057 | pcadach | 2007-02-01 03:07:41 -0800 (?\208?\167?\209?\130?\208?\178, 01 ?\208?\164?\208?\181?\208?\178 2007) | 1 line
chan_h323 is very stable, so let it built by default
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r53059 | pcadach | 2007-02-01 05:16:00 -0600 (Thu, 01 Feb 2007) | 9 lines
Oops -- Merged revisions 53057 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53057 | pcadach | 2007-02-01 03:07:41 -0800 (?\208?\167?\209?\130?\208?\178, 01 ?\208?\164?\208?\181?\208?\178 2007) | 1 line
chan_h323 is very stable, so let it built by default
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r53061 | russell | 2007-02-01 08:43:44 -0600 (Thu, 01 Feb 2007) | 3 lines
Remove duplicate calls to pthread_attr_destroy() that I put in yesterday
by accident.
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r53063 | oej | 2007-02-01 10:42:24 -0600 (Thu, 01 Feb 2007) | 10 lines
Merged revisions 53062 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53062 | oej | 2007-02-01 17:35:12 +0100 (Thu, 01 Feb 2007) | 2 lines
Add explanation of port= in combination with defaultip= (thanks jsmith)
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r53065 | file | 2007-02-01 11:41:02 -0600 (Thu, 01 Feb 2007) | 2 lines
Make trunk compile under dev mode.
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r53066 | file | 2007-02-01 11:42:08 -0600 (Thu, 01 Feb 2007) | 10 lines
Merged revisions 53064 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53064 | file | 2007-02-01 11:37:44 -0600 (Thu, 01 Feb 2007) | 2 lines
Fix silly logic. We really want to write UDPTL frames out when the call is up.
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r53067 | oej | 2007-02-01 13:04:47 -0600 (Thu, 01 Feb 2007) | 2 lines
Signal HOLD status to phones that subscribe for status.
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r53071 | tilghman | 2007-02-01 13:27:22 -0600 (Thu, 01 Feb 2007) | 18 lines
Merged revisions 53070 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53070 | tilghman | 2007-02-01 13:21:20 -0600 (Thu, 01 Feb 2007) | 10 lines
Merged revisions 53069 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r53069 | tilghman | 2007-02-01 13:13:53 -0600 (Thu, 01 Feb 2007) | 2 lines
No wonder FIELDQTY doesn't work with functions... the documentation in pbx.c was wrong
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r53073 | file | 2007-02-01 13:34:54 -0600 (Thu, 01 Feb 2007) | 10 lines
Merged revisions 53072 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53072 | file | 2007-02-01 13:33:33 -0600 (Thu, 01 Feb 2007) | 2 lines
Add missing 'F' letter to getopt so it magically becomes a valid option. (issue #8960 reported by tzafrir)
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r53076 | tilghman | 2007-02-01 14:12:56 -0600 (Thu, 01 Feb 2007) | 18 lines
Merged revisions 53075 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53075 | tilghman | 2007-02-01 14:09:52 -0600 (Thu, 01 Feb 2007) | 10 lines
Merged revisions 53074 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r53074 | tilghman | 2007-02-01 14:07:35 -0600 (Thu, 01 Feb 2007) | 2 lines
Bug 8965
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r53078 | tilghman | 2007-02-01 14:14:26 -0600 (Thu, 01 Feb 2007) | 9 lines
Blocked revisions 53077 via svnmerge
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r53077 | tilghman | 2007-02-01 14:13:40 -0600 (Thu, 01 Feb 2007) | 2 lines
Oops.
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r53080 | oej | 2007-02-01 14:33:59 -0600 (Thu, 01 Feb 2007) | 10 lines
Merged revisions 53079 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53079 | oej | 2007-02-01 21:28:54 +0100 (Thu, 01 Feb 2007) | 2 lines
Cleaning up the devicestate callback function
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r53082 | oej | 2007-02-01 14:43:49 -0600 (Thu, 01 Feb 2007) | 11 lines
Implementing "busy-limit".
If you set call limit and busy limit, chan_sip will indicate BUSY for a device
that has reached the busy limit and allow calls up to the call limit, allowing
for call transfers (that generate a new call).
If you only set call limit, chan_sip will not indicate BUSY until that limit
is filled.
This affects SIP subscriptions, call queues and manager applications.
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r53083 | oej | 2007-02-01 14:44:49 -0600 (Thu, 01 Feb 2007) | 10 lines
Merged revisions 53081 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53081 | oej | 2007-02-01 21:38:58 +0100 (Thu, 01 Feb 2007) | 2 lines
Change debug level for state change message that is not really informative when debugging app_queue
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r53087 | file | 2007-02-01 15:10:03 -0600 (Thu, 01 Feb 2007) | 9 lines
Blocked revisions 53086 via svnmerge
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r53086 | file | 2007-02-01 15:06:02 -0600 (Thu, 01 Feb 2007) | 2 lines
Make func_strings build under dev mode. Didn't I do this today already in the berkeley DB?
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r53089 | file | 2007-02-01 15:12:26 -0600 (Thu, 01 Feb 2007) | 18 lines
Merged revisions 53088 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53088 | file | 2007-02-01 15:11:28 -0600 (Thu, 01 Feb 2007) | 10 lines
Merged revisions 53084 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r53084 | file | 2007-02-01 15:03:10 -0600 (Thu, 01 Feb 2007) | 2 lines
Return previous behavior of having MOH pick up where it was left off. (issue #8672 reported by sinistermidget)
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r53092 | oej | 2007-02-01 15:17:08 -0600 (Thu, 01 Feb 2007) | 12 lines
Merged revisions 53085 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53085 | oej | 2007-02-01 22:05:34 +0100 (Thu, 01 Feb 2007) | 4 lines
- Clean INC_COUNT flag when we decrement call counter
- If it's still set at time of dialog destruction, make sure we decrement the device call counter properly
before we destroy the dialog
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r53094 | russell | 2007-02-01 15:27:22 -0600 (Thu, 01 Feb 2007) | 10 lines
Merged revisions 53093 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53093 | russell | 2007-02-01 15:24:52 -0600 (Thu, 01 Feb 2007) | 2 lines
Fix the FIELDQTY function to not crash. (reported by blitzrage and Corydon on IRC)
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r53098 | file | 2007-02-01 15:56:23 -0600 (Thu, 01 Feb 2007) | 18 lines
Merged revisions 53097 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53097 | file | 2007-02-01 15:54:28 -0600 (Thu, 01 Feb 2007) | 10 lines
Merged revisions 53095 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r53095 | file | 2007-02-01 15:47:11 -0600 (Thu, 01 Feb 2007) | 2 lines
Don't negotiate RFC2833 when not configured to do so. (issue #8799 reported by mdu113)
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r53100 | file | 2007-02-01 16:05:34 -0600 (Thu, 01 Feb 2007) | 9 lines
Blocked revisions 53099 via svnmerge
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r53099 | file | 2007-02-01 16:04:58 -0600 (Thu, 01 Feb 2007) | 2 lines
Huh... fix the berkeley DB to compile here as well, but it apparently required both dev mode and no optimizations to creep up.
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r53105 | file | 2007-02-01 16:26:11 -0600 (Thu, 01 Feb 2007) | 18 lines
Merged revisions 53104 via svnmerge from
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r53104 | file | 2007-02-01 16:24:32 -0600 (Thu, 01 Feb 2007) | 10 lines
Merged revisions 53103 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r53103 | file | 2007-02-01 16:21:56 -0600 (Thu, 01 Feb 2007) | 2 lines
Copy noncodeccapability over to the joint variable so that telephone-event will get transmitted in the sent INVITE.
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r53110 | oej | 2007-02-01 18:26:25 -0600 (Thu, 01 Feb 2007) | 14 lines
Patch based on this patch with small changes for trunk...
Merged revisions 53109 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53109 | oej | 2007-02-02 01:24:03 +0100 (Fri, 02 Feb 2007) | 4 lines
Disable the direct p2p RTP call setup in SIP. You can enable it in sip.conf, but it is now
considered experimental until we solve the AST_CONTROL_ANSWER with payload and videocaps
stuff.
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r53111 | file | 2007-02-01 18:39:01 -0600 (Thu, 01 Feb 2007) | 2 lines
Read/write lockify the devicestate stuff a bit.
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r53112 | file | 2007-02-01 18:49:14 -0600 (Thu, 01 Feb 2007) | 2 lines
Switch the devicestate thread to operate the same way as the logging thread. Pops all entries off the list to be processed, resets the list back to a clean state, and processes each entry. The thread won't have to acquire the list lock again until it checks to see if there are more to process.
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r53113 | file | 2007-02-01 18:58:09 -0600 (Thu, 01 Feb 2007) | 2 lines
Clean up ast_device_state. It's pretty now!
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r53115 | file | 2007-02-02 09:30:12 -0600 (Fri, 02 Feb 2007) | 10 lines
Merged revisions 53114 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53114 | file | 2007-02-02 09:29:35 -0600 (Fri, 02 Feb 2007) | 2 lines
Add systemname to asterisk.conf generation per recent discussions about it. (issue #8968 reported by blitzrage)
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r53119 | file | 2007-02-02 11:01:04 -0600 (Fri, 02 Feb 2007) | 18 lines
Merged revisions 53118 via svnmerge from
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r53118 | file | 2007-02-02 10:59:53 -0600 (Fri, 02 Feb 2007) | 10 lines
Merged revisions 53117 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r53117 | file | 2007-02-02 10:58:09 -0600 (Fri, 02 Feb 2007) | 2 lines
Pass the glob expanded filename to process_text_line so that error messages contain the actual filename, not the original include one. (issue #8959 reported by tzafrir)
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r53122 | file | 2007-02-02 11:16:05 -0600 (Fri, 02 Feb 2007) | 10 lines
Merged revisions 53120 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53120 | file | 2007-02-02 11:15:22 -0600 (Fri, 02 Feb 2007) | 2 lines
Correct a copy/pasted error message line for RTCP.
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r53125 | file | 2007-02-02 12:21:46 -0600 (Fri, 02 Feb 2007) | 2 lines
Add onHold value to sip show inuse as well.
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r53126 | oej | 2007-02-02 14:02:49 -0600 (Fri, 02 Feb 2007) | 7 lines
Adding a template for documentation on call queues. Please help us add
to this!
Thanks
/OEJ and BJ
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r53127 | oej | 2007-02-02 14:05:52 -0600 (Fri, 02 Feb 2007) | 2 lines
Update with info about SIP channels and queues
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r53132 | pcadach | 2007-02-03 04:12:20 -0600 (Sat, 03 Feb 2007) | 10 lines
Merged revisions 53131 via svnmerge from
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r53131 | pcadach | 2007-02-03 02:02:55 -0800 (?\208?\161?\208?\177?\209?\130, 03 ?\208?\164?\208?\181?\208?\178 2007) | 1 line
Remove quote from H.323 vendor string because due to compatibilities with
Nortel Meridian CS1000 reported at www.voip-info.org
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r53137 | russell | 2007-02-03 14:46:36 -0600 (Sat, 03 Feb 2007) | 20 lines
Merged revisions 53136 via svnmerge from
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r53136 | russell | 2007-02-03 14:44:20 -0600 (Sat, 03 Feb 2007) | 12 lines
Merged revisions 53133 via svnmerge from
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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r53139 | file | 2007-02-03 15:06:36 -0600 (Sat, 03 Feb 2007) | 10 lines
Merged revisions 53138 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53138 | file | 2007-02-03 15:05:02 -0600 (Sat, 03 Feb 2007) | 2 lines
Make SIPDtmfMode application work with recent capability changes, and also fix an RTP stack issue when the auto option was used. (issue #8972 reported by mdu113)
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r53140 | tilghman | 2007-02-03 16:04:09 -0600 (Sat, 03 Feb 2007) | 2 lines
Fix compiler warnings
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r53141 | tilghman | 2007-02-03 16:05:02 -0600 (Sat, 03 Feb 2007) | 2 lines
Add CALLERPRES dialplan function and deprecate SetCallerPres application
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r53142 | tilghman | 2007-02-03 16:06:46 -0600 (Sat, 03 Feb 2007) | 2 lines
Deprecate SetCallerPres application
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r53144 | oej | 2007-02-04 18:30:03 -0600 (Sun, 04 Feb 2007) | 11 lines
Merged revisions 53143 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53143 | oej | 2007-02-05 01:18:34 +0100 (Mon, 05 Feb 2007) | 3 lines
Add some comments on queue system behaviour and how it affects the
SIP channel
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r53151 | file | 2007-02-05 10:03:23 -0600 (Mon, 05 Feb 2007) | 10 lines
Merged revisions 53150 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53150 | file | 2007-02-05 10:02:00 -0600 (Mon, 05 Feb 2007) | 2 lines
Unregister Playback CLI commands as well as dialplan application. (issue #8946 reported by junky)
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r53153 | file | 2007-02-05 11:06:56 -0600 (Mon, 05 Feb 2007) | 10 lines
Merged revisions 53152 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53152 | file | 2007-02-05 11:06:18 -0600 (Mon, 05 Feb 2007) | 2 lines
Ensure say_cfg is NULL when the module is loaded. (issue #8946 reported by junky)
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r53200 | oej | 2007-02-05 15:55:01 -0600 (Mon, 05 Feb 2007) | 2 lines
Doxygen formatting changes
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r53247 | tilghman | 2007-02-06 01:07:22 -0600 (Tue, 06 Feb 2007) | 18 lines
Merged revisions 53246 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53246 | tilghman | 2007-02-06 01:00:52 -0600 (Tue, 06 Feb 2007) | 10 lines
Merged revisions 53245 via svnmerge from
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r53245 | tilghman | 2007-02-06 00:58:28 -0600 (Tue, 06 Feb 2007) | 2 lines
Issue 8987 - Status could return two responses (mnicholson)
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Added:
team/group/ast_storage/doc/queue.txt
- copied unchanged from r53247, trunk/doc/queue.txt
Modified:
team/group/ast_storage/ (props changed)
team/group/ast_storage/Makefile
team/group/ast_storage/UPGRADE.txt
team/group/ast_storage/apps/app_dial.c
team/group/ast_storage/apps/app_meetme.c
team/group/ast_storage/apps/app_playback.c
team/group/ast_storage/apps/app_queue.c
team/group/ast_storage/apps/app_setcallerid.c
team/group/ast_storage/apps/app_userevent.c
team/group/ast_storage/channels/chan_h323.c
team/group/ast_storage/channels/chan_iax2.c
team/group/ast_storage/channels/chan_mgcp.c
team/group/ast_storage/channels/chan_sip.c
team/group/ast_storage/channels/chan_skinny.c
team/group/ast_storage/channels/chan_zap.c
team/group/ast_storage/channels/h323/ast_h323.cxx
team/group/ast_storage/configs/sip.conf.sample
team/group/ast_storage/funcs/func_callerid.c
team/group/ast_storage/funcs/func_odbc.c
team/group/ast_storage/funcs/func_strings.c
team/group/ast_storage/main/asterisk.c
team/group/ast_storage/main/cdr.c
team/group/ast_storage/main/config.c
team/group/ast_storage/main/db1-ast/hash/hash.c
team/group/ast_storage/main/devicestate.c
team/group/ast_storage/main/http.c
team/group/ast_storage/main/io.c
team/group/ast_storage/main/manager.c
team/group/ast_storage/main/pbx.c
team/group/ast_storage/main/rtp.c
team/group/ast_storage/pbx/pbx_dundi.c
team/group/ast_storage/pbx/pbx_spool.c
team/group/ast_storage/res/res_features.c
team/group/ast_storage/res/res_musiconhold.c
Propchange: team/group/ast_storage/
------------------------------------------------------------------------------
automerge = *
Propchange: team/group/ast_storage/
------------------------------------------------------------------------------
Binary property 'branch-1.4-blocked' - no diff available.
Propchange: team/group/ast_storage/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Propchange: team/group/ast_storage/
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--- svnmerge-integrated (original)
+++ svnmerge-integrated Tue Feb 6 07:47:14 2007
@@ -1,1 +1,1 @@
-/trunk:1-53007
+/trunk:1-53247
Modified: team/group/ast_storage/Makefile
URL: http://svn.digium.com/view/asterisk/team/group/ast_storage/Makefile?view=diff&rev=53282&r1=53281&r2=53282
==============================================================================
--- team/group/ast_storage/Makefile (original)
+++ team/group/ast_storage/Makefile Tue Feb 6 07:47:14 2007
@@ -540,6 +540,7 @@
echo "" ; \
echo ";[options]" ; \
echo ";internal_timing = yes" ; \
+ echo ";systemname = my_system_name ; prefix uniqueid with a system name for global uniqueness issues" ; \
echo "; Changing the following lines may compromise your security." ; \
echo ";[files]" ; \
echo ";astctlpermissions = 0660" ; \
Modified: team/group/ast_storage/UPGRADE.txt
URL: http://svn.digium.com/view/asterisk/team/group/ast_storage/UPGRADE.txt?view=diff&rev=53282&r1=53281&r2=53282
==============================================================================
--- team/group/ast_storage/UPGRADE.txt (original)
+++ team/group/ast_storage/UPGRADE.txt Tue Feb 6 07:47:14 2007
@@ -34,3 +34,5 @@
* ChanIsAvail() now has a 't' option, which allows the specified device
to be queried for state without consulting the channel drivers. This
performs mostly a 'ChanExists' sort of function.
+* SetCallerPres() has been replaced with the CALLERPRES() dialplan function
+ and is now deprecated.
Modified: team/group/ast_storage/apps/app_dial.c
URL: http://svn.digium.com/view/asterisk/team/group/ast_storage/apps/app_dial.c?view=diff&rev=53282&r1=53281&r2=53282
==============================================================================
--- team/group/ast_storage/apps/app_dial.c (original)
+++ team/group/ast_storage/apps/app_dial.c Tue Feb 6 07:47:14 2007
@@ -82,7 +82,7 @@
" ANSWEREDTIME - This is the amount of time for actual call.\n"
" DIALSTATUS - This is the status of the call:\n"
" CHANUNAVAIL | CONGESTION | NOANSWER | BUSY | ANSWER | CANCEL\n"
-" DONTCALL | TORTURE\n"
+" DONTCALL | TORTURE | INVALIDARGS\n"
" For the Privacy and Screening Modes, the DIALSTATUS variable will be set to\n"
"DONTCALL if the called party chooses to send the calling party to the 'Go Away'\n"
"script. The DIALSTATUS variable will be set to TORTURE if the called party\n"
@@ -1183,6 +1183,7 @@
struct privacy_args pa = {
.sentringing = 0,
.privdb_val = 0,
+ .status = "INVALIDARGS",
};
int sentringing = 0, moh = 0;
const char *outbound_group = NULL;
@@ -1201,23 +1202,27 @@
if (ast_strlen_zero(data)) {
ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
+ pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
return -1;
}
u = ast_module_user_add(chan); /* XXX is this the right place ? */
parse = ast_strdupa(data);
-
+
AST_STANDARD_APP_ARGS(args, parse);
memset(&config,0,sizeof(struct ast_bridge_config));
if (!ast_strlen_zero(args.options) &&
- ast_app_parse_options(dial_exec_options, &opts, opt_args, args.options))
+ ast_app_parse_options(dial_exec_options, &opts, opt_args, args.options)) {
+ pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
goto done;
+ }
if (ast_strlen_zero(args.peers)) {
ast_log(LOG_WARNING, "Dial requires an argument (technology/number)\n");
+ pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
goto done;
}
@@ -1231,6 +1236,7 @@
calldurationlimit = atoi(opt_args[OPT_ARG_DURATION_STOP]);
if (!calldurationlimit) {
ast_log(LOG_WARNING, "Dial does not accept S(%s), hanging up.\n", opt_args[OPT_ARG_DURATION_STOP]);
+ pbx_builtin_setvar_helper(chan, "DIALSTATUS", pa.status);
goto done;
}
if (option_verbose > 2)
Modified: team/group/ast_storage/apps/app_meetme.c
URL: http://svn.digium.com/view/asterisk/team/group/ast_storage/apps/app_meetme.c?view=diff&rev=53282&r1=53281&r2=53282
==============================================================================
--- team/group/ast_storage/apps/app_meetme.c (original)
+++ team/group/ast_storage/apps/app_meetme.c Tue Feb 6 07:47:14 2007
@@ -1081,6 +1081,7 @@
pthread_attr_init(&conf->attr);
pthread_attr_setdetachstate(&conf->attr, PTHREAD_CREATE_DETACHED);
ast_pthread_create_background(&conf->recordthread, &conf->attr, recordthread, conf);
+ pthread_attr_destroy(&conf->attr);
}
time(&user->jointime);
Modified: team/group/ast_storage/apps/app_playback.c
URL: http://svn.digium.com/view/asterisk/team/group/ast_storage/apps/app_playback.c?view=diff&rev=53282&r1=53281&r2=53282
==============================================================================
--- team/group/ast_storage/apps/app_playback.c (original)
+++ team/group/ast_storage/apps/app_playback.c Tue Feb 6 07:47:14 2007
@@ -68,7 +68,7 @@
;
-static struct ast_config *say_cfg;
+static struct ast_config *say_cfg = NULL;
/* save the say' api calls.
* The first entry is NULL if we have the standard source,
* otherwise we are sourcing from here.
@@ -468,6 +468,8 @@
res = ast_unregister_application(app);
+ ast_cli_unregister_multiple(cli_playback, sizeof(cli_playback) / sizeof(struct ast_cli_entry));
+
ast_module_user_hangup_all();
if (say_cfg)
Modified: team/group/ast_storage/apps/app_queue.c
URL: http://svn.digium.com/view/asterisk/team/group/ast_storage/apps/app_queue.c?view=diff&rev=53282&r1=53281&r2=53282
==============================================================================
--- team/group/ast_storage/apps/app_queue.c (original)
+++ team/group/ast_storage/apps/app_queue.c Tue Feb 6 07:47:14 2007
@@ -564,7 +564,7 @@
AST_LIST_UNLOCK(&interfaces);
if (!curint) {
- if (option_debug)
+ if (option_debug > 2)
ast_log(LOG_DEBUG, "Device '%s/%s' changed to state '%d' (%s) but we don't care because they're not a member of any queue.\n", technology, loc, sc->state, devstate2str(sc->state));
free(sc);
return NULL;
@@ -631,6 +631,7 @@
ast_log(LOG_WARNING, "Failed to create update thread!\n");
free(sc);
}
+ pthread_attr_destroy(&attr);
return 0;
}
@@ -956,10 +957,11 @@
}
}
-static void rt_handle_member_record(struct call_queue *q, char *interface, const char *membername, const char *penalty_str)
+static void rt_handle_member_record(struct call_queue *q, char *interface, const char *membername, const char *penalty_str, const char *paused_str)
{
struct member *m, *prev_m;
int penalty = 0;
+ int paused = 0;
if (penalty_str) {
penalty = atoi(penalty_str);
@@ -967,6 +969,12 @@
penalty = 0;
}
+ if (paused_str) {
+ paused = atoi(paused_str);
+ if (paused < 0)
+ paused = 0;
+ }
+
/* Find the member, or the place to put a new one. */
for (m = q->members, prev_m = NULL;
m && strcmp(m->interface, interface);
@@ -974,7 +982,7 @@
/* Create a new one if not found, else update penalty */
if (!m) {
- if ((m = create_queue_member(interface, membername, penalty, 0))) {
+ if ((m = create_queue_member(interface, membername, penalty, paused))) {
m->dead = 0;
add_to_interfaces(interface);
if (prev_m) {
@@ -985,6 +993,8 @@
}
} else {
m->dead = 0; /* Do not delete this one. */
+ if (paused_str)
+ m->paused = paused;
m->penalty = penalty;
}
}
@@ -1106,7 +1116,8 @@
while ((interface = ast_category_browse(member_config, interface))) {
rt_handle_member_record(q, interface,
S_OR(ast_variable_retrieve(member_config, interface, "membername"), interface),
- ast_variable_retrieve(member_config, interface, "penalty"));
+ ast_variable_retrieve(member_config, interface, "penalty"),
+ ast_variable_retrieve(member_config, interface, "paused"));
}
/* Delete all realtime members that have been deleted in DB. */
Modified: team/group/ast_storage/apps/app_setcallerid.c
URL: http://svn.digium.com/view/asterisk/team/group/ast_storage/apps/app_setcallerid.c?view=diff&rev=53282&r1=53281&r2=53282
==============================================================================
--- team/group/ast_storage/apps/app_setcallerid.c (original)
+++ team/group/ast_storage/apps/app_setcallerid.c Tue Feb 6 07:47:14 2007
@@ -68,9 +68,14 @@
{
struct ast_module_user *u;
int pres = -1;
+ static int deprecated = 0;
u = ast_module_user_add(chan);
-
+
+ if (!deprecated) {
+ deprecated = 1;
+ ast_log(LOG_WARNING, "SetCallerPres is deprecated. Please use Set(CALLERPRES()=%s) instead.\n", (char *)data);
+ }
pres = ast_parse_caller_presentation(data);
if (pres < 0) {
Modified: team/group/ast_storage/apps/app_userevent.c
URL: http://svn.digium.com/view/asterisk/team/group/ast_storage/apps/app_userevent.c?view=diff&rev=53282&r1=53281&r2=53282
==============================================================================
--- team/group/ast_storage/apps/app_userevent.c (original)
+++ team/group/ast_storage/apps/app_userevent.c Tue Feb 6 07:47:14 2007
@@ -83,7 +83,7 @@
buflen += 2;
}
- manager_event(EVENT_FLAG_USER, "UserEvent", "UserEvent: %s\r\n%s\r\n", args.eventname, buf);
+ manager_event(EVENT_FLAG_USER, "UserEvent", "UserEvent: %s\r\n%s", args.eventname, buf);
ast_module_user_remove(u);
Modified: team/group/ast_storage/channels/chan_h323.c
URL: http://svn.digium.com/view/asterisk/team/group/ast_storage/channels/chan_h323.c?view=diff&rev=53282&r1=53281&r2=53282
==============================================================================
--- team/group/ast_storage/channels/chan_h323.c (original)
+++ team/group/ast_storage/channels/chan_h323.c Tue Feb 6 07:47:14 2007
@@ -35,7 +35,7 @@
/*** MODULEINFO
<depend>openh323</depend>
- <defaultenabled>no</defaultenabled>
+ <defaultenabled>yes</defaultenabled>
***/
#ifdef __cplusplus
@@ -2610,8 +2610,10 @@
monitor_thread = AST_PTHREADT_NULL;
ast_mutex_unlock(&monlock);
ast_log(LOG_ERROR, "Unable to start monitor thread.\n");
+ pthread_attr_destroy(&attr);
return -1;
}
+ pthread_attr_destroy(&attr);
}
ast_mutex_unlock(&monlock);
return 0;
Modified: team/group/ast_storage/channels/chan_iax2.c
URL: http://svn.digium.com/view/asterisk/team/group/ast_storage/channels/chan_iax2.c?view=diff&rev=53282&r1=53281&r2=53282
==============================================================================
--- team/group/ast_storage/channels/chan_iax2.c (original)
+++ team/group/ast_storage/channels/chan_iax2.c Tue Feb 6 07:47:14 2007
@@ -6163,6 +6163,8 @@
if (ast_pthread_create(&newthread, &attr, dp_lookup_thread, dpr)) {
ast_log(LOG_WARNING, "Unable to start lookup thread!\n");
}
+
+ pthread_attr_destroy(&attr);
}
struct iax_dual {
@@ -6237,8 +6239,11 @@
d->chan1 = chan1m;
d->chan2 = chan2m;
- if (!ast_pthread_create_background(&th, &attr, iax_park_thread, d))
+ if (!ast_pthread_create_background(&th, &attr, iax_park_thread, d)) {
+ pthread_attr_destroy(&attr);
return 0;
+ }
+ pthread_attr_destroy(&attr);
free(d);
}
return -1;
Modified: team/group/ast_storage/channels/chan_mgcp.c
URL: http://svn.digium.com/view/asterisk/team/group/ast_storage/channels/chan_mgcp.c?view=diff&rev=53282&r1=53281&r2=53282
==============================================================================
--- team/group/ast_storage/channels/chan_mgcp.c (original)
+++ team/group/ast_storage/channels/chan_mgcp.c Tue Feb 6 07:47:14 2007
@@ -3007,6 +3007,7 @@
/*ast_queue_control(sub->owner, AST_CONTROL_ANSWER);*/
}
}
+ pthread_attr_destroy(&attr);
}
static int handle_request(struct mgcp_subchannel *sub, struct mgcp_request *req, struct sockaddr_in *sin)
Modified: team/group/ast_storage/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/group/ast_storage/channels/chan_sip.c?view=diff&rev=53282&r1=53281&r2=53282
==============================================================================
--- team/group/ast_storage/channels/chan_sip.c (original)
+++ team/group/ast_storage/channels/chan_sip.c Tue Feb 6 07:47:14 2007
@@ -526,6 +526,7 @@
static struct ast_codec_pref default_prefs; /*!< Default codec prefs */
/* Global settings only apply to the channel */
+static int global_directrtpsetup; /*!< Enable support for Direct RTP setup (no re-invites) */
static int global_limitonpeers; /*!< Match call limit on peers only */
static int global_rtautoclear;
static int global_notifyringing; /*!< Send notifications on ringing */
@@ -562,7 +563,6 @@
/*! \brief Codecs that we support by default: */
static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
-static int noncodeccapability = AST_RTP_DTMF;
/* Object counters */
static int suserobjs = 0; /*!< Static users */
@@ -944,6 +944,7 @@
int peercapability; /*!< Supported peer capability */
int prefcodec; /*!< Preferred codec (outbound only) */
int noncodeccapability; /*!< DTMF RFC2833 telephony-event */
+ int jointnoncodeccapability; /*!< Joint Non codec capability */
int redircodecs; /*!< Redirect codecs */
int maxcallbitrate; /*!< Maximum Call Bitrate for Video Calls */
struct t38properties t38; /*!< T38 settings */
@@ -1095,6 +1096,7 @@
int inRinging; /*!< Number of calls ringing */
int onHold; /*!< Peer has someone on hold */
int call_limit; /*!< Limit of concurrent calls */
+ int busy_limit; /*!< Limit where we signal busy */
enum transfermodes allowtransfer; /*! SIP Refer restriction scheme */
char vmexten[AST_MAX_EXTENSION]; /*!< Dialplan extension for MWI notify message*/
char mailbox[AST_MAX_EXTENSION]; /*!< Mailbox setting for MWI checks */
@@ -2960,7 +2962,8 @@
p->callingpres = ast->cid.cid_pres;
p->jointcapability = ast_translate_available_formats(p->capability, p->prefcodec);
-
+ p->jointnoncodeccapability = p->noncodeccapability;
+
/* If there are no audio formats left to offer, punt */
if (!(p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
ast_log(LOG_WARNING, "No audio format found to offer. Cancelling call to %s\n", p->username);
@@ -3013,6 +3016,12 @@
if (sip_debug_test_pvt(p) || option_debug > 2)
ast_verbose("Really destroying SIP dialog '%s' Method: %s\n", p->callid, sip_methods[p->method].text);
+
+ if (ast_test_flag(&p->flags[0], SIP_INC_COUNT)) {
+ update_call_counter(p, DEC_CALL_LIMIT);
+ if (option_debug > 1)
+ ast_log(LOG_DEBUG, "This call did not properly clean up call limits. Call ID %s\n", p->callid);
+ }
/* Remove link from peer to subscription of MWI */
if (p->relatedpeer && p->relatedpeer->mwipvt)
@@ -3155,9 +3164,10 @@
/* incoming and outgoing affects the inUse counter */
case DEC_CALL_LIMIT:
/* Decrement inuse count if applicable */
- if (inuse && ast_test_flag(&fup->flags[0], SIP_INC_COUNT))
+ if (inuse && ast_test_flag(&fup->flags[0], SIP_INC_COUNT)) {
ast_atomic_fetchadd_int(inuse, -1);
- else
+ ast_clear_flag(&fup->flags[0], SIP_INC_COUNT);
+ } else
*inuse = 0;
/* Decrement ringing count if applicable */
if (inringing && ast_test_flag(&fup->flags[1], SIP_PAGE2_INC_RINGING)) {
@@ -3642,7 +3652,7 @@
we simply forget the frames if we get modem frames before the bridge is up.
Fax will re-transmit.
*/
- if (p->udptl && ast->_state != AST_STATE_UP)
+ if (p->udptl && ast->_state == AST_STATE_UP)
res = ast_udptl_write(p->udptl, frame);
sip_pvt_unlock(p);
}
@@ -5207,7 +5217,7 @@
newjointcapability = p->capability & (peercapability | vpeercapability);
newpeercapability = (peercapability | vpeercapability);
- newnoncodeccapability = noncodeccapability & peernoncodeccapability;
+ newnoncodeccapability = p->noncodeccapability & peernoncodeccapability;
if (debug) {
@@ -5221,7 +5231,7 @@
ast_getformatname_multiple(s4, BUFSIZ, newjointcapability));
ast_verbose("Non-codec capabilities (dtmf): us - %s, peer - %s, combined - %s\n",
- ast_rtp_lookup_mime_multiple(s1, BUFSIZ, noncodeccapability, 0, 0),
+ ast_rtp_lookup_mime_multiple(s1, BUFSIZ, p->noncodeccapability, 0, 0),
ast_rtp_lookup_mime_multiple(s2, BUFSIZ, peernoncodeccapability, 0, 0),
ast_rtp_lookup_mime_multiple(s3, BUFSIZ, newnoncodeccapability, 0, 0));
}
@@ -5240,9 +5250,9 @@
/* We are now ready to change the sip session and p->rtp and p->vrtp with the offered codecs, since
they are acceptable */
- p->jointcapability = newjointcapability; /* Our joint codec profile for this call */
- p->peercapability = newpeercapability; /* The other sides capability in latest offer */
- p->noncodeccapability = newnoncodeccapability; /* DTMF capabilities */
+ p->jointcapability = newjointcapability; /* Our joint codec profile for this call */
+ p->peercapability = newpeercapability; /* The other sides capability in latest offer */
+ p->jointnoncodeccapability = newnoncodeccapability; /* DTMF capabilities */
ast_rtp_pt_copy(p->rtp, newaudiortp);
if (p->vrtp)
@@ -5253,6 +5263,9 @@
if (newnoncodeccapability & AST_RTP_DTMF) {
/* XXX Would it be reasonable to drop the DSP at this point? XXX */
ast_set_flag(&p->flags[0], SIP_DTMF_RFC2833);
+ /* Since RFC2833 is now negotiated we need to change some properties of the RTP stream */
+ ast_rtp_setdtmf(p->rtp, 1);
+ ast_rtp_setdtmfcompensate(p->rtp, ast_test_flag(&p->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
} else {
ast_set_flag(&p->flags[0], SIP_DTMF_INBAND);
}
@@ -6394,7 +6407,7 @@
/* Now add DTMF RFC2833 telephony-event as a codec */
for (x = 1; x <= AST_RTP_MAX; x <<= 1) {
- if (!(p->noncodeccapability & x))
+ if (!(p->jointnoncodeccapability & x))
continue;
add_noncodec_to_sdp(p, x, 8000,
@@ -6965,6 +6978,10 @@
pidfnote = "Unavailable";
break;
case AST_EXTENSION_ONHOLD:
+ statestring = "confirmed";
+ local_state = NOTIFY_INUSE;
+ pidfstate = "busy";
+ pidfnote = "On hold";
break;
case AST_EXTENSION_NOT_INUSE:
default:
@@ -7060,6 +7077,11 @@
else
ast_build_string(&t, &maxbytes, "<dialog id=\"%s\">\n", p->exten);
ast_build_string(&t, &maxbytes, "<state>%s</state>\n", statestring);
+ if (state == AST_EXTENSION_ONHOLD) {
+ ast_build_string(&t, &maxbytes, "<local>\n<target uri=\"%s\">\n"
+ "<param pname=\"+sip.rendering\" pvalue=\"no\">\n"
+ "</target>\n</local>\n", mto);
+ }
ast_build_string(&t, &maxbytes, "</dialog>\n</dialog-info>\n");
break;
case NONE:
@@ -9534,7 +9556,7 @@
snprintf(ilimits, sizeof(ilimits), "%d", iterator->call_limit);
else
ast_copy_string(ilimits, "N/A", sizeof(ilimits));
- snprintf(iused, sizeof(iused), "%d/%d", iterator->inUse, iterator->inRinging);
+ snprintf(iused, sizeof(iused), "%d/%d/%d", iterator->inUse, iterator->inRinging, iterator->onHold);
if (showall || iterator->call_limit)
ast_cli(fd, FORMAT2, iterator->name, iused, ilimits);
ASTOBJ_UNLOCK(iterator);
@@ -10198,6 +10220,8 @@
ast_cli(fd, " VM Extension : %s\n", peer->vmexten);
ast_cli(fd, " LastMsgsSent : %d/%d\n", (peer->lastmsgssent & 0x7fff0000) >> 16, peer->lastmsgssent & 0xffff);
ast_cli(fd, " Call limit : %d\n", peer->call_limit);
+ if (peer->busy_limit)
[... 1295 lines stripped ...]
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