https://origsvn.digium.com/svn/asterisk/branches/1.4
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r85533 | russell | 2007-10-13 01:48:10 -0400 (Sat, 13 Oct 2007) | 12 lines
Fix an issue with console verbosity when running asterisk -rx to execute a command
and retrieve its output. The issue was that there was no way for the main Asterisk
process to know that the remote console was connecting in the -rx mode. The way that
James has fixed this is to have all remote consoles muted by default. Then, regular
remote consoles automatically execute a CLI command to unmute themselves when they
first start up.
(closes issue #10847)
Reported by: atis
Patches:
asterisk-consolemute.diff.txt uploaded by jamesgolovich (license 176)
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- Added development section (backtrace.tex)
- Correct filesystem path formating
- Replace all "|" argument separator to ","
- Endless count of spaces at the end of line
- Using astlisting to make listings do not take so much place
- Take back ASTRISKVERSION on first page
- Make localchannel.tex readable by inserting extra end of lines
(closes issue #10962)
Reported by: IgorG
Patches:
texdoc-85177-1.patch uploaded by IgorG (license 20)
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r85397 | file | 2007-10-11 12:26:20 -0300 (Thu, 11 Oct 2007) | 6 lines
When creating a new packet don't try to stop retransmission of it. It was just allocated/created so it's impossible for it to have already been scheduled.
(closes issue #10945)
Reported by: flefoll
Patches:
chan_sip.c.br14.85280.xmit_reliable-patch uploaded by flefoll (license 244)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r85316 | russell | 2007-10-10 10:56:23 -0500 (Wed, 10 Oct 2007) | 6 lines
I introduced a new member to the ast_filestream struct in 1.4.12, but put it
in the middle of the struct, instead of at the end. One of the Debian folks,
paravoid, pointed out that this breaks binary compatability with modules
compiled against older headers. So, I'm moving the new member to the end
of the struct to resolve the situation.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r85242 | file | 2007-10-10 11:14:56 -0300 (Wed, 10 Oct 2007) | 6 lines
Close voicemail message description file if duration did not meet the minimum, or else we will eventually run out of file descriptors.
(closes issue #10918)
Reported by: brak2718
Patches:
vm1.4.12.1.patch uploaded by brak2718 (license 279)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85254 65c4cc65-6c06-0410-ace0-fbb531ad65f3
in the Dial command. The 'j' option _must_ be used in conjunction with the 'n'
option.
This feature will allow you to use the existing jitterbuffer implementation to
put a jitterbuffer on incoming SIP calls connecting to Asterisk applications by
putting a local channel in the middle.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85097 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r84957 | russell | 2007-10-07 22:28:34 -0500 (Sun, 07 Oct 2007) | 6 lines
Enable file dependency tracking for _all_ builds, and not just for builds with
dev-mode enabled. I have seen enough problems caused by this that I don't think
it's worth keeping. I want to continue to encourage anybody that is interested
to continue to run Asterisk from svn. Furthermore, I do not want their systems
to break when we change a structure definition in a header file. :)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r84902 | phsultan | 2007-10-07 18:15:39 +0200 (Sun, 07 Oct 2007) | 5 lines
Presence packets from a client who's connected with our Jabber ID are
valid, therefore, those clients must be considered as buddies. The resource
string helps us make the distinction between clients.
Closes issue #10707, reported by yusufmotiwala.
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r84890 | phsultan | 2007-10-07 17:52:44 +0200 (Sun, 07 Oct 2007) | 5 lines
Prevent Asterisk from crashing when receiving a presence packet
without resource from a buddy that is known to have a resource list.
Revert a change I previously made, where Asterisk could point to a
freed memory location.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r84818 | file | 2007-10-05 15:55:36 -0300 (Fri, 05 Oct 2007) | 4 lines
Update the remembered RTP peer information when putting an endpoint on hold or taking it off hold so that the RTP stack does not initiate a needless reinvite.
(closes issue #10868)
Reported by: mavince
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