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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

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

...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 --pr...

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 =>

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

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

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,

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