Displaying 8 results from an estimated 8 matches for "freezon".
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freezone
2004 Jan 30
1
Reciprocal Update
If I run "rsync -ruv <local> <remote>" twice, no files are copied the
second time - as expected. However, if I follow "rsync -ruv <local>
<remote>" with "rsync -ruv <remote> <local>", all files are copied from
<remote> to <local>.
I suspect this is because the timestamps of <remote> files are of when
they were
2005 Aug 26
1
lvm initrd -> initramfs
I converted my lvm root initrd to an initramfs by putting glibc, lvm,
pivot_root, my linuxrc, etc. in my initramfs source file. I use ash
compiled against klibc to run my linuxrc
Unfortunately -
pivot_root . initrd
- complains -
pivot_root: Invalid argument
I suspect this may be because you can't pivot_root using a cpio
initramfs root?
If so, what should I do instead? Should I
2007 Jul 14
1
DO NOT REPLY [Bug 4786] New: deletes files in parent directory when destination directory doesn't exist
...directory doesn't exist
Product: rsync
Version: 2.6.9
Platform: Other
OS/Version: Linux
Status: NEW
Severity: normal
Priority: P3
Component: core
AssignedTo: wayned@samba.org
ReportedBy: ms419@freezone.co.uk
QAContact: rsync-qa@samba.org
I'm using the PHP Symfony framework's rsync application publication feature to
publish an application from my development box to a ~archives/symfony directory
on our production box. This invokes the following rsync command:
rsync --progress -...
2010 Jan 26
0
StopPlayTones() after first digit?
I configured our SIP gateway to automatically dial extension "s" when a
phone is picked up. I want Asterisk to play a dial tone, wait for an
extension to be dialled, and hangup on timeout
This works great, but I also want Asterisk to *stop* playing the dial
tone after the first digit is pressed
So far my extensions.conf contains,
[internal]
exten => s,1,Answer
exten =>
2010 Feb 19
1
directmedia/canreinvite/native bridging question
I've got several SIP clients with dynamic IP addresses
Asterisk has one public and one private IP address
SIP clients might connect to Asterisk from either the internet or the
private network (192.168.1.255) - they're portable
By default, directmedia/canreinvite is enabled and Asterisk sets up
direct media connections between clients. In this case clients on the
internet can make calls
2010 May 04
1
Channel failover
How do you configure Asterisk to dial, in order, each channel from a
group of channels until it either finds an available channel, or runs
out of channels?
We recently got VoIP, so when we make a call, Asterisk should first try
to make the call with VoIP, but in case either our VoIP or our internet
service are down, Asterisk should then try to make the call with our old
school analog phone line
2004 Oct 26
0
Policing
My attempts to configure policing are stopping incoming traffic all
together.
From the LARTC HOWTO, I gather that the following lines should limit
incoming traffic on eth0 to 32kbit by dropping packets above this
threshold:
tc qdisc add dev eth0 ingress
tc filter add dev eth0 parent ffff: protocol ip u32 \
match u8 0x0 0x0 \
police rate 32kbit burst 10k drop \
classid :1
Instead,
2008 Oct 17
1
anoyingly answers already in use pstn line
I am using Asterisk and an X101P card as a glorified answering machine.
We have a residential PSTN line with about six phones connected to it.
Like an answering machine, I want Asterisk answer the line *only* when
an incoming call is not answered after four rings.
This mostly works. My extensions.conf is at the end of this message.
The problem is that Asterisk will sometimes answer the line when