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without a path
use '-n' instead of '! -z' for tests
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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.
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Since we already sent the BEGIN frame for that digit, it makes sense to send
the END as well.
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source file
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the 'm' option functions properly. (issue #8969 reported by christianbee)
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distro's binary c-client library package (at least not at this time)
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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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ASTCFLAGS, instead of at the end. This way, we ensure that we find the local
headers first before accidentally trying to use headers that exist in
locations specified in the ASTCFLAGS passed from the main Makefile.
(issue #8637, ovi)
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because we need to make sure that its configure script gets executed again,
because the CFLAGS we want to pass to editline may have changed.
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if we need to listen for DTMF frames. (issue #8962 reported by caio1982)
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take the low 16 bits of the integer part, and the high 16 bits of the
fractional part. However, the code here was erroneously taking the low 16 bits
of the fractional part. It then shifted the result 16 bits down, so the result
was always zero. This fix makes it grab the appropriate high 16 bits, instead.
(issue #8991, pointed out by andre_abrantes)
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function. (issue #8946 reported by junky)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r53357 | file | 2007-02-07 10:38:48 -0500 (Wed, 07 Feb 2007) | 2 lines
Fix a few potential memory leaks with realtime users and peers. (issue #8999 reported by bsmithurst)
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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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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r52843 | crichter | 2007-01-30 15:38:08 +0100 (Di, 30 Jan 2007) | 1 line
fixed some possible segfaults. also fixed an very important bug which occurs on high load (when calls are very fast generated)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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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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junky)
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#8946 reported by junky)
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SIP channel
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fix an RTP stack issue when the auto option was used. (issue #8972 reported by mdu113)
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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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r53134 | russell | 2007-02-03 14:39:45 -0600 (Sat, 03 Feb 2007) | 2 lines
Revert some changes that accidentally got committed as a part of another fix.
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CS1000 reported at www.voip-info.org
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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.
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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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(issue #8968 reported by blitzrage)
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but it is now
considered experimental until we solve the AST_CONTROL_ANSWER with payload and videocaps
stuff.
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r53107 | qwell | 2007-02-01 17:14:09 -0600 (Thu, 01 Feb 2007) | 2 lines
Fix a small typo. Synopsis lines shouldn't have a newline
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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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required both dev mode and no optimizations to creep up.
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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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on IRC)
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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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