similar to: What's wrong with my exclude rules?

I am try to backup entire server using rsync. Can someone help me with a script. The only thing i don't want include are /prc, /tmp and /lost+found I have root access on both servers i create an account on backup server called backup.

I'm using rsync as part of a system to create nightly snapshots of several servers, much like several people are appearently doing. However, I have a bit of a problem with things that I don't want backed up. On most of the systems, things like devfs, procfs, sysfs, etc don't need backed up. In addition, I have directories from one server NFS mounted on several of the other servers -

Am 07.01.2016 um 10:55 schrieb Frank: > On Wed, 2016-01-06 at 17:03 +0100, Juergen Sauer wrote: Thx, 4answer. :) >> with in my sip.conf, I have got for this hardphone: >> [...] >> [hard1] >> username=hard1 >> secret=correct-and-three-times-checked-4-digit-pin > > In most cases, there is no need to set the "username=" option. The

Hi! I wish you all e Happy New Year first! Allthough, I'm relative new to Asterisk, I got our server up and Running, Softphones, ISDN, and a brand new Snom 821 are working flawlessly. :) Platform is Debian 8/Asterisk Packages (11) from Debian Repo. But I am running into problems setting up 2 older Hardphones, Thomson 2030S. :( with in my sip.conf, I have got for this hardphone: [...]

Hello, I'm trying to compile DAHDI on DEBIAN but i have the following error: root at Sangoma-Testing:/usr/src/dahdi-linux-2.1.0.4# make echo "You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed." You do not appear to have the sources for the 2.6.26-2-amd64 kernel installed. exit 1 make: *** [modules] Error 1 What should i do? Thanks! -------------- next

Hi, I have two objects: Contacts Lists I want to be able to add multiple contacts to multiple lists. I''ve created a linking table and a form that allows me to select the contacts using checkboxes, but I''m having some problems with the controller and model. I receive the following error message when I submit the form: Mysql::Error: Cannot add or update a child row: a

Hi Guys, Currently, I have a Sangoma A400 installed with 20 ZAP PSTN analogue lines. The first line is giving me problems due to rain (probably coroded line). My server using FreePBX dials out with g0 (group 0 which includes all 20 lines) and it happens that the bad line is the very first line. Can I simply put ; in zapata.conf like this to seclude the first zap line from getting calls in or

Hello. I don't know if this is a problem, but I was expecting a different behavior. Users, have to dial "0" to get an external line, and afterwords the number they want to dial (exe 12345). The thing is: 1-If user dial "012345" there is an error and the call isn't made and the error is "handle_request_invite: Call from 'XXX' to extension '012345'

Dear Sam, Thanks for your detailed response, however I am using the command line tool. The problem is that exactly the same arguments to CMake on Mac OS X and Linux give different results. On Mac OS X, headers are not installed. On Linux, they are installed. I've tested this on Debian Linux and OS X 10.6.4 both with CMake >= 2.8.0 I am not sure how to debug this problem. Does anyone

Hello. I run samba 3.0.21c on RHEL AS 4 EMT_64 U3 and I've experienced a strange problem where smbd will go (and stay) in the D state. While waiting for IO (possibly), these smbd processes do not respond to client requests, thus samba appears to be down. I could not kill these processes via kill -9 , so I rebooted which worked. The only error I could see out of the ordinary would be

I'm trying to decide if I need to open an issue for this or if it's just a misconfiguration issue of some sort. Here's the situation - yesterday morning, I downloaded asterisk 1.8.19.1 and installed it on a fresh CentOS 5.8 installation and got a shell of a basic asterisk install setup (minimum required configuration files, etc, with no dialplan or sip peers setup yet). In the

Hi, Greetings to All, Im looking for some help on configuring VPN on the Asterisk PBX that I have hosted in US. Im currently in Middle East and as everyone knows some countries here has taboo to VOIP. Im not able to get phy phones registered to my PBX as they are blocking SIP and IAX2. Hence im looking for a VPN solution. For this first i need to setup VPN on my server .. Am i right? Well if

Hi, I'm running asterisk 1.4.22 on a debian server. I have php5 installed and it works correctly command line. When trying to run a php script via AGI, I get messages such as: GI Tx >> I> AGI Rx << #!/usr/bin/php5 -q AGI Tx >> 510 Invalid or unknown command The scripts are completely executable and owned by asterisk -rwxr-xr-x 1 asterisk asterisk Googling is not helping

I have successfully installed IceCast with SSL support : https://relay2.dgnet.be/ Stream is working fine with this URL : https://relay2.dgnet.be:443/different.ogg Nevertheless the listen URL still points to http://relay2.dgnet.be:443/different.ogg ( without https then ) So the links provided/generated by IceCast are wrong . What did I miss ? Is it possible to force the listenurl value in the

Hello, Could you please explain or send an example for decoding a packet that contains multiple Speex frames. According to the API document the speex_decode_int() decodes only one frame of speech. Best regards, Danny Peled R&D Software Engineer Intelligence Solutions NICE Systems Ltd. ISRAEL *+ (972) 9 769 7161 ? + (972) 54 4 602840 *Danny.Peled at nice.com <mailto:*Danny.Peled at

On Fri, 21 Nov 2014, "Thomas B. R?cker" wrote: > On 11/21/2014 02:21 PM, Geoff Shang wrote: >> I would like to request that this be configurable somehow. I agree >> that this is undesirable under most circumstances, but there are >> situations where this is required. > > It has been requested before and the answer is still the same, we have > it on our

Hi gang, We are moving our 1.4 asterisk with DAHDI over to 10.0 with SIP. Everything is going nicely except that I can't get NV_FAXDETECT to compile properly into 10.0. Because of this, I will have to have my receptionist manually transfer incoming faxes. Any suggestions? Thanks in Advance Danny Nicholas -------------- next part -------------- An HTML attachment

Hello, I sent a wav-audio to Asterisk though SIP and ISDN channels and recorded it in wav-audio at the Asterisk server. I found the loudness level of the recorded audio was too high comparing with the orginal audio. How can I ajust it, so that there will be no amplifier used for recording. Thanks a lot. best regards Felix -------------- next part -------------- An HTML attachment was

I have 2 drives in a Linux 7.1 server , 1 primary and 1 secondary. I am trying to clone to the secondary slave drive and be take the primary out and change the slave to primary and boot from it. The problem I am having is that it the copied drive will not boot, It is not getting the lilo boot and config properly. When I try to boot the copied drive , it boots to a Blank screen with an