similar to: Re: please check your mesages thnx

Hi all, I'm running rsync (2.6.2) this way (which do not remove files from the destination side which were removed from the sending side): rsync -vbu --backup-dir=/bkp --stats /dir1/*.fle /dir2 I want to sync all .fle files from dir1 to the dir2 and also remove files from the destination which were removed from the sending side, using no recursion. Already tried this:

Hi, my name is Grant Taylor. I am a subscriber to the LARTC mail list lartc@mailman.ds9a.nl. The LARTC mail lists has been plagued with viral email coming from changing IPs in your one of your subnets. Based on the fact that the WhoIs information below says that the subnet in question is a dial up pool this would explain the changing IPs. In less than 6 days the list has received 14 viral

Hello, does anybody here know the exact meaning of "numopen" in the open/close file log messages? Does it mean the number of open sessions on the file: - at all? - per user? - per client? Sometimes numopen is greater than zero ofter the last user has closed as file. Is this an error? Here an extract of <hostname>.log --------------------------------- [2001/10/31 08:57:24, 2]

Hi, I am getting the following messages for a few days. WARNING[10148]: chan_iax2.c:1219 __send_lagrq: I was supposed to send a LAGRQ with callno 15691, but no such call exists (and I cannot remove lagid, either). I got 2 iax trunks which used by test purposes only. And no iax clients or no calling rules thru this iax channnels so far.. So why do i keep getting that messages.

Is it possible to turn off the colorizing of clang error messages? The reason I ask is that using clang in an XEmacs buffer results in user unfriendly output like this: $ clang -c -Wall -Wno-switch -Wundef -Wsign-compare -Wno-char-subscripts -Wpacked -Wshadow -Wmissing-declarations -Wmissing-prototypes -Wstrict-prototypes -Wdeclaration-after-statement -Wunused-parameter -ggdb -march=native

We are using rsync to push DNS records from one machine (running rsync 2.5.6 as a client process on Debian woody) to a collocation server ( running rsync 2.5.6 as a daemon on RedHat 7.3). To test this process we are using a script that pushes one file from the Debian box to the collocation server. It appears that a file transfers, but upon checking there is no file transfer and an error in the

Guys, this is weird.. Today I started having some problems with calls been dropped. Im suing X100p cards (clones) and I have this setting on my zatala fle: [channels] language=sp signalling=fxs_ks usecallerid=yes cidsignalling=bell cidstart=ring hidecallerid=no callwaiting=yes usecallingpres=yes ;sendcalleridafter=1 callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes

Who ever owns this server can you please correct this issue. Filtering 'spam' based on subject like is extremely stupid, and I'm getting tired of getting this email every other message. - Ben ---------- Forwarded message ---------- Date: Fri, 8 Nov 2002 14:18:13 -0000 From: NAIBTX002SBAYECEX005 at btx002.bt.com To: mouring at etoh.eviladmin.org Subject: ALERT - GroupShield ticket

Hi,all i want to hear the voicemail recorded, but when hear "if you want to play message , press 3", after i press 3 i only hear that that's the time the file recorded. not the content. do you know how to hear content of voicemail fle? debug message: == Parsing '/var/spool/asterisk/voicemail/95040654321/3/INBOX/msg0001.txt': Found -- <SIP/1003-00000058>

hello i have copied some sites from another server to cent os but i have faced with that non latin chars in sites seems like -> " ? " firstly i think it is an sql problem or encoding but i see that file manager see file names like : VBULLET?N4 r?tbe and i decide to create a folder with the same non latin char ? ??????iii??????? it worked so how could i repair this copied fle names

Hello, I have figured out that my audio problem was just how I was converting the sound files. I am trying to convert the Asterisk gsm files to ULAW. I just did a: sox file.gsm file.ul, open it in Audacity. I used: Project, Import Raw, U-law, No endian, 1 channel, start offest 1 byte, sample rate 8000hz. The file sounds fine in Audacity. Now, if I do a record on Asterisk, using pcm, au, or

Im having some problems with connecting a TNT to asterisk. The problem is when the call is sent to asterisk and signaling is done the RTP syncs however no audio is produced. Can someone give me some idea of how to accomplish this? I am using the standard configs and g711 and 729 do the same. No audio. Public IPs on both ends. No nat. Any ideas would be appreciated.

Does anyone in here know how to setup auto negotiation between g729 and g711ulaw on a cisco 5400? I would imagine it would be the same on a 3660. The problem I am having is natively the call is setup for g729 however when the call is transferred to voicemail it uses ULAW so when the cisco tries to connect to the voice mail I get a SIP error that the codec couldn't be negotiated. I need

Hello, i am using wine 1.1.19 under ubuntu 8.10. I have an Geforce 8400 GS with newest nvidia driver 180x installed. I am playing here the game metin2 i have some problems like rendering or what it is i don't know here are a screenshot of the dots ingame. Does anbody knows what is it? thx [Image: http://img245.imageshack.us/img245/4527/fle.jpg ]

Still no progress. Just tried to transfer 1.3GB fle and my box locked up. I am running MD 10 (Community) as a file server on a Shuttle SB61G2. This setup worked very well under Mandrake 9.2 however, everytime I try to copy files larger than say <550 ~650MB using MD 10, my linux box freezes and must be rebooted. I can FTP the same file(s) perfectly fine to other PC 's on my home

I created some scripts to logrotate. I am having a problem. After I do it, I am sending kill -HUP to the process its not using the newly created messages file again. Could someone help me out with how I can rotate asterisk's log's without killing the process? ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL

I had the same problem and I believe it was the payload size of the codec. What code are you using? ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Federico Alves Sent: Friday,

I run a bunch of the Linksys ATA's.. I always use rfc2833 for DTMF. works very well and have never had a problem with it. ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of

Just use a cisco with 5 T1 ports and have everything over IP use ultra monkey to load balance your asterisk boxes. I have found this works very well. .o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com

as far as I know there isn't. I use 80 bytes for G711U that may or may not fix your issue. You can also do a ethereal trace to find out what the actual error is. ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com