similar to: Number of calls

I'm a bit confused how scp works... could someone please explain the local/remote external calls that happen when scp is started... in particular how it relates to ssh on the remote site? To be more specific... I use Kerberos for authentication and I've been working on an ssh wrapper script that checks my Kerberos credentials before running the ssh command. If the credentials are

--delete option is not working for me. The following is a minimal example. $ mkdir box1 box2 $ touch box1/letter box2/extra $ rsync -a -vv box1/ box2/ sending incremental file list delta-transmission disabled for local transfer or --whole-file letter total: matches=0 hash_hits=0 false_alarms=0 data=0 sent 92 bytes received 34 bytes 252.00 bytes/sec total size is 0 speedup is 0.00 $ tree

Hello everybody and Timo. I have the next problem. With LDAP tree like ou=mail | - dc=example1.com,ou=Mail | - mail=box1 at example1.com,dc=example1.com,ou=Mail | - mail=box2 at example1.com,dc=example1.com,ou=Mail ? | - dc=example2.com,ou=Mail | - mail=box1 at example2.com,dc=example2.com,ou=Mail | - mail=box2 at example2.com,dc=example2.com,ou=Mail ? and settings in dovecot

Hi, I get this message on the latest Debian potato for i386. --> ssh -f localhost xterm sb at localhost's password: --> xterm Xt error: Can't open display: That happens also when connecting to another linux box: --> ssh -f box2 /usr/X11R6/bin/xterm --> /usr/X11R6/bin/xterm Xt error: Can't open display: The $DISPLAY is :0.0 for box1 undefined for box2 after

Hi. My goal is to use rsync to syncronize box1 and box2. On box2, I have a tapedrive which I write the data I syncronize from box1 every night. On box1, my /etc/rsyncd.conf looks like this: root@box1# cat /etc/rsyncd.conf max connections = 1 syslog facility = local6 [tmp] path = /tmp read only = yes comment = export of /tmp hosts allow = box2 auth users

Hello everyone, this is my first post to the list, so hello again. We're a small company in Romania and we're trying to set up a really small version of "call center". That is, we want to get a few land-lines from our telco in different countys and "bridge" all calls to our HQ, in order to make it cheeper for our clients to call us. Unfortunatelly there's no ISP

I'm trying to determine the capacity of my host running Asterisk 11.8.1 on CentOS 6.5. The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet. The primary application will be bridging groups of users using meetme(). I'm using 2 boxes -- 1 to initiate calls using call files (box1), and 1 behaving a bit more like a production box -- bridging calls (box2). The call

> Message: 13 > Date: Fri, 9 Jul 2004 11:42:01 +1200 (NZST) > From: =?iso-8859-1?q?Eugen=20Cristea?= <tecristea@yahoo.co.nz> > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] asterisk to asterisk config > Reply-To: asterisk-users@lists.digium.com > > Hi, > > I would like to set two separate asterisks to talk to > each other. > Any

I have 2 linux boxes connected to a switch: box1: eth0 192.168.0.200/255.255.255.0 eth1 public address from ISP dhcp box2: eth0 192.168.0.35/255.255.255.0 box3: eth0 192.168.1.3/255.255.255.0 I want box1 to act as a gateway to the internet (it is doing this now for box2), but also want to communicate from box2 to box3 through box1, and that box3 can use the internet through box1. how can I do

Hi! I'm trying to set up a SIP trunk so that I can test calls, etc., between a new Asterisk box, and an old 1.4 box. --------------------------------------------------------------------------- New box: root at asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf siptrunk.conf: [box1] ; All box1 extensions; see extensions.conf type=peer context=adhearsion host=172.17.0.17 ; IP

Hi All; I have the following configuration in my iax.conf files at asterisk box1 and box2 (two asterisk): At box1: [user1] disallow=all codec=g729 codec=GSM At box2: [user2] disallow=all codec=g729 codec=GSM If G729 is no more available at box1, so how can I let user1 to select GSM codec instead of G729 to communicate with user2 at box 2? Any help? Regards Bilal

Hi, I have two * box. One is box1 and the other is box2. And I have two iax clients A and B. A is registed with box1 and B is registed with box2. If I make a call from A to B using the following method: IAX/[<user>[:<secret>]@]<peer>[:<portno>][/<exten>[@<context>][/<options>]] The CalledID that B got is <user> but not A's CallerID. I

Hello All, I am using the rsync-2.5.1 on Solaris (Sparc) 2.6. I encountered the following while trying to get a directory on two machine in sync. 1) copy the files /var/tmp/test-rsync/* from box1 to box2:/var/tmp/test-rsync/ 2) touch a file AAAA in box1:/var/tmp/test-rsync/ 3) in box1, type in the following /usr/local/bin/rsync -nculaR --stats --partial /var/tmp/test-rsync/

Hello everyone, I'm fairly new to asterisk and this list. Currently I'm working on IAX trunks to send/receive calls between 2 asterisk boxes with asterisk 1.6.1.1+asterisk gui 2.0. After some work in the gui, two boxes can send/receive calls to/from the other just fine, the only problem I have is the caller id. Here is my setup: 1. on both boxes, I added an IAX user in the gui, say the

Hi all, We currently have 2 Asterisk boxes and we pass calls to a fro. All works great except we now need to pass variables between them. For example now on box 1 we have: exten => _23XX,1,SetVar(Foo=1234) exten => _23XX,2,Dial(SIP/${EXTEN:0}@Box2) When the call dials into Box 2 the variable Foo does not get passed... Does anyone have any clever ideas? -------------- next part

HI, i''m trying to write a helper that replaces link_to_remote that accepts a block but it gets a compile error in the template: I don''t know what I''m doing wrong... TIA # application_helper.rb def link_block_to_remote( options = {}, html_options = {}, &block) concat(link_to_remote(capture(&block), options, html_options), block.binding) end # in

Hi, I know some of you are very experienced as to the working of networks. I wondered whether there is some accepted way of determining bandwidth needs based on the network traffic over time. For example, looking at the figures for the network traffic through the server interface, we have hourly, daily and monthly figures. If everything were linear, taking the hourly figure and dividing it by

Hi. I'm having trouble figuring out why I'm not able to make many concurrent VoIP calls on my system. I'm not aiming for a huge number, because I have purposely bought a low powered system, but I would think that I could get more. Here are the details: I have a small-form-factor Asterisk server with an Intel Atom 230 CPU (1.6 GHz, 533 MHz FSB) and 512 MB DDR2 533. It is running Ubuntu

Greetings all, I've setup a new gretap device as per Herbert Xu's new patches and it works as expected for smaller MTU. I'm wondering if gretap tunnels handle fragmentation at all or if this might be a bridge related issue. The same setup works quite well if I use vtun with a tap device however, so while this isn't entirely bridge-related, Maybe someone here might have

hi, alll,,, there two asterisk box share one realtime database... and all the client is IAX2.. and registery dynamic... there have some question need to confirm.. 1, when i run iax2 show peers,,,there no show the peers that registed with real time... the same as run iax2 show users..there not show any real time users.. 2, if user1 have registed with box1,, how user2 on box1 and user3 on box2