similar to: CIFS/ACLs

Sharing my experience with SSO of Linux clients to Active Directory. Over the last 2 years or so, i had a great deal of trouble getting and _keeping_ authentication to our Win2000/Win2003 Active Directory system working from OpenSUSE and CentOS clients. ADS authentication would work until reboot, a few days, a month max. We'll see how long this lasts. Another problem was dealing with the

Darryl Moore <darryl at moores.ca> schrieb: > I'd start by turning on sip debugging in asterisk > >sip set debug ip [your_phone_ip] Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d at 172.16.34.133' Method: OPTIONS Reliably Transmitting (no NAT) to 192.168.200.11:5060: OPTIONS sip:00493512222222 at 192.168.200.11:5060 SIP/2.0 Via: SIP/2.0/UDP

Folks, Ive got two ISP connections that I am using with: --- ip route add 192.168.200.0/24 dev eth2 src 192.168.200.11 table connection1 ip route add default via 192.168.200.1 table connection1 ip route add x.175.244.0/24 dev eth1 src x.175.244.2 table connection2 ip route add default via x.175.244.1 table connection2 ip rule add from 192.168.200.11 table connection1 ip rule add from x.175.244.2

Hi list! I have a problem and I hope someone can help me... I configured an Asterisk on a VM to serve more accounts and act as a proxy to other SIP-providers. The first account running on my phone works without any problem. A second account, running on the phone of my wife, is always UNREACHABLE. I can just see in the log: [May 28 21:48:46] NOTICE[3646]: chan_sip.c:22933 sip_poke_noanswer: Peer

I have a server running Centos4. I found I was seeing some network problems so I tried "route" and found there was no routing. I thought reboot would solve it but it didn't. I had to manually enter a route to the default gateway, there is still no route to the loopback interface. How is this possible? I thought there was default routes built into the kernel, taken from

> Darryl Moore <darryl at moores.ca> schrieb: > > > I'd start by turning on sip debugging in asterisk > > >sip set debug ip [your_phone_ip] > > Really destroying SIP dialog '490d1996593c8e11217828b71aae5c4d at 172. > 16.34.133' Method: OPTIONS > Reliably Transmitting (no NAT) to 192.168.200.11:5060: > OPTIONS sip:00493512222222 at

Hi Luca Y need to check your wifes codec priority list -seems to be GSM on the first place. Luca Bertoncello <lucabert at lucabert.de> wrote: >Hi list! > >I noticed that when the phone of my wife calls the gsm codec will be used, >but if someone calls the phone, alaw will be used: > >00493511111111 calls 00493512222222: >OpenWrt*CLI> sip show channels >Peer

Hi list! I noticed that when the phone of my wife calls the gsm codec will be used, but if someone calls the phone, alaw will be used: 00493511111111 calls 00493512222222: OpenWrt*CLI> sip show channels Peer User/ANR Call ID Format Hold Last Message Expiry Peer 192.168.200.11 00493512222222 5305ad0e07977dd 0x4 (ulaw) No

Hi All, Let me try again, this time with fixed width ! First off the scenario. We have a local and remote site, both of which connect to the internet via a wireless network. The wireless cards themselves have both an ethernet (ixp1) and an "Atheros" (ath0) interface. The devices connect via PPPoE, and so there is also a ppp0 interface on each device. On one side we have : IXP1 IP

Hi Puppet Users, Puppet Version - v2.7.13 Ruby Version - ruby 1.8.7 (2010-08-16 patchlevel 302) [i486-linux] I am trying to use the Puppetlabs DHCP module (https://github.com/puppetlabs/puppet-dhcp) to setup a DHCP server on one of my nodes. I have followed the example in the readme.md file in my node declaration for the server (extracted part below) class { ''dhcp'':

> I have a problem and I hope someone can help me... > I configured an Asterisk on a VM to serve more accounts and act as a proxy to > other SIP-providers. > > The first account running on my phone works without any problem. > A second account, running on the phone of my wife, is always UNREACHABLE. > I can just see in the log: > > [May 28 21:48:46] NOTICE[3646]:

I'd start by turning on sip debugging in asterisk >sip set debug ip [your_phone_ip] and use tcpdump or wireshark to see what the OS sees tcpdump host [your_phone_ip] and udp port 5060 On 15-05-28 03:58 PM, Luca Bertoncello wrote: > Hi list! > > I have a problem and I hope someone can help me... > I configured an Asterisk on a VM to serve more accounts and act as a proxy

Hello, can someone give me a hint, where to find an info, how to setup dovecote replication with sieve? On my active.active setup, it replicates the user mdboxes and sieve works also, on the active destination fileserver. Client changes the sieve, tests work fine. But the sieve file won't get replicated to the replica server. As i've researched, it was introduced in dovecot v2.2.rc3

Hi, I installed samba4 (zentyal 3.4) and I followed the Samba4 AD DC Howto from the wiki to test the linux-side. There all seems OK. Unfortunately, when I try to register a Win7 PC, the domain is not found. So what can I do to test things on the Windows-side ? I did : C:\net view /domain:ace_domain returns : \\ZENTYAL1 which is my DC After manually "mounting" the test1-share : C:\net

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > What kind of phone are we talking about, both yours that works and your > wife's that does not? Right! > Can you ping the unreachable phone and does it respond to a ping? I can ping both phones from the VM > Many phones will have a network test function built in to them to help you > determine if the phone

Hi again! I tried to attach two VoIP-phones to my new Asterisk 13.14.1 on a Banana PI with Armbian/Debian 9. First test was to call a test service that say the time. Works! Second test was to record my voice and play it again. Works! Third test was to call the other VoIP-phone. It does NOT work... :( Then I noticed that, by starting, Asterisk says the following messages: [Feb 15 18:42:54]

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>