similar to: ip route - TOS argument does not allways work

Dear All, I need to setup different route with different tos value. I can use the flowing command to add a tos route to routing table. ip route 192.168.0.2/32 via 192.168.0.1 tos 0x1c and ip route 192.168.0.2/32 via 192.168.0.1 tos 0x40 I used "ping -Q" to test it with different tos, the output packet is marked correctly. The problem that is only the tos value defined at

Hi All: I have a Cisco3620 with a proper T1/PRI card installed with asterisk running on the same LAN. Since I have lit up the line, I can dial out and make calls to regular lands lines. However when a call comes back in it rings the destination phone once and disconnects. Here is an error from my router 15:40:45: ISDN Se1/0:23 SERROR: L3_GetUser_NLCB: EVENT 0X45 No NLCB 2 15:40:45: ISDN

Which files do I need to edit in order to initialize a route after a reboot. I have a 802.1q tagged VLAN interface which I need to create a rule, and a route for. I see on my CentOs 4.2 box that in /etc/iproute2 there are the following files. I realize that I can add my tables to /etc/iproute2/rt_tables, but when I want to create a rule I am not sure where to put the information for the

Hi, Thank you Guus, I think the /16 solution is the easier to apply so I modified my tinc-up in host A to be like #!/bin/sh ifconfig $INTERFACE 192.168.10.1 netmask 255.255.0.0 a route -a from HOST A shows: Kernel IP routing table Destination Gateway Genmask Flags Metric Ref Use Iface 0.0.0.0 178.62.128.1 0.0.0.0 UG 0 0 0 eth0 10.129.0.0

Hello all, I might make a fool out of me, but I think the prio qdisc doesn''t work as advertised in any document I could lay my hands on. The following tests and observations were made with Linux kernel 2.6.20.21. This makes it up-to-date in the 2.6.20.x kernel family. My problem was that the link quality reported by the olsr.org olsrd degraded depending on the amount of payload

Hi, after trying for days I ended up with a working tinc configurazion of 2 subnets, now my goal is to add 2 more subnets and comunicate. I might seem dumb at this point but honestly I don't work in IT or Networking stuff, and so I dont have that deep knowledge. A little explanation of my configuration is HOST A (VPN server) Public IP: 1.2.3.4 tun0 Subnet = 192.168.10.0/24 tun0 IP =

I have two phones (111 and 112) on a LAN, and I have on a users site a phone 333. phone 111 uses sip.conf, while 112 uses real-time set-up. 111 can call 333 AND the audio is working 112 can call 333 but audio is just white noise. 333 can call 111 or 112 and audio is working. The phones are identically set-up (just user name = phone number and password are different) sip.conf (for 111 - all

Hello guys I don''t know if this thing has been posted before (if it was , please forgive me). I have 7 computers at home and I want all of them to have access to the internet. In order to do that , I set up a linux router (2 network cards) as a usual router (eth0 : 82.77.69.75 - internet connection ; eth1 : 192.168.10.1 - local network) . The other computers have ips ranging from

Hi all, i am working on an networking product which has a 32 MB flash memory and a 64MB RAM. i need to sync a file named appl.gz which is of size 3.1MB from a unit with ip 192.168.10.1 to a unit with ip 192.168.10.2. the rsync is called from a C code which does a fork-execve operation. the command line options being passed to rsync is as below: command: rsync --stats -aI --temp-dir=/tmp/

Hi, After a few extra days of hard work, debugging, and many coffees, I am pround to announce that Asterisk::LCR has been released on CPAN. Asterisk::LCR is an open-source, Perl-based collection of tools to help you manage efficiently multiple VoIP providers with your Asterisk installation. It is capable of importing providers rates from multiple providers, comparing these rates, and

I have some Asterisk boxes in my SIP provider's colo, and the SIP connections are done without using any kind of authentication credentials. It's a secure network, so I guess that's OK. I now want different sets of calls from one box billed to different customers, by the providers billing system, and the provider says that the distinction can only be made by having the differnet sets

Hello @all, thanks for all the helpfull answers in this mailing list. They helped me compiling a new kernel 2.6.10.11 with Xen 2.0.6 (my first one), setting up a dom0 with RAID, LVM and loop devices for domU, integrating a second network as xen-br1 (I''m calling the /etc/xen/scripts/network script from within /etc/init.d/xend to bring up / down the second bridge.) and bootstrapping

Hello All Anybody had used ooH323 for asterisk i using ooH323-0.7.2 and asterisk CVS may 2005. OpenH323 1.17.1 and pwlib 1.9.0 and GNUGK 2.0.2 audio is very good, better than SIP and IAX, but i have problem. how to router call from openh323 to outside PSTN. my h323.conf setting ; Objective System's H323 Configuration example for tvcti ; ooh323c driver configuration ; ; [general]

Hi all, are there a way to define a associative array/Hash? example: $host = [{ :name => ''host1'', :ip => ''192.168.10.1'' }, { :name => ''host2'', :ip => ''192.168.10.2'' }] Regards --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups

http://bugzilla.mindrot.org/show_bug.cgi?id=1101 Summary: terminal allocation hangs ssh at a blocking tcsetattr (- t option) Product: Portable OpenSSH Version: 4.2p1 Platform: ix86 OS/Version: Linux Status: NEW Severity: normal Priority: P2 Component: ssh AssignedTo: bitbucket

Hi, I am trying to use Asterisk cmd ZapRAS (http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ZapRAS), I pathed the ppp daemon ftp://ftp.digium.com/pub/zaptel/misc/, but when I try to use it, I obtain the following error: Connected to Asterisk 1.2.4 currently running on TSU-R1 (pid = 7242) Verbosity was 0 and is now 44 -- Accepting call from '123456789' to

For the benefit of everyone, having H323 Configuration problem due to H245 Tunnel, check the h323 Config embeded at the end. Comment the offending line as under: ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=yes ; -----Original Message----- From: Tola Ogunsan [mailto:tolaniye@hotmail.com] Sent: Wednesday, May 25, 2005 1:03 PM To: Kanuri, Seshu (Company IT) Subject: RE: oh323 problems

OOH323 Debugging Enabled -- Executing Answer("SIP/3513-090f7d40", "") in new stack -- Executing Wait("SIP/3513-090f7d40", "1") in new stack -- Executing DeadAGI("SIP/3513-090f7d40", "a2billing.php|1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php a2billing.php|1: line:58 - IDCONFIG : 1

Hello, I have the need to connect 2 remote site with vpn, the windows pc of the 2 site it can share the HD and printer. This is my configuration : LOCAL NETWORK A : ip from 192.168.10.2 to 192.168.10.99 | | | | eth0: 192.168.10.1 FIREWALL A : ( with debian ; openvpn ver. 2.0.beta15 ; shorewall ver 2.0.11 ) eth1 : xxx.xxx.xxx.xxx ( pubblic ip address ) | | | | INTERNET | | | eth1 :

I am trying to work out IAX <--> IAX communications with my * box. I have a registration with iaxtel and I thought I would start there for my learning. I am able to call the number for Digium's support line (700-428-6000), but the sound is horribly chopping. Some reading revealed the jitterbuffer settings, so I enabled them in iax.conf. I have the following now: ; Inter-Asterisk