search for: timestamping

Hy Guys, My samba version: 4.0.13 ntp version: 4.2.6p3 Ubuntu 12.04.3 LTS I have a weird problem, I sat up the time service via https://wiki.samba.org/index.php/Configure_NTP On the servers I have no firewall. This port is open when i check with nmap, ntp runs. Looks like all of my Windows 7 clients works fine but w32tm /resync says Permission denied 0x8007005, so not that good :) Windows

I posted this question a while back, and I'm posting again in hopes that someone has some ideas. Sorry if you've already seen this. When dialing out using a SIP or IAX provider (Broadvoice, SimpleTelecom, VoicePulse Connect) I often find that after the call is answered the first few seconds of audio are cut off (i.e. I don't hear the called party). This usually results in the called

hallo, i am really confused how nativ briging is working with asterisk, i use a asterisk server as central server and register another asterisk and an iaxcomm client to the server, all three have public ips on the internet. somtimes, when i call from iaxcomm to my asterisk, the calls go peer to peer (i can see it with tcpdump) but sometimes the get routed through the central asterisk server

The waiting time is about 50s on my CentOS box now. "yum remove openssh* "and "yum install openssh*" can't make it right. "mv ~/.ssh{,.bak}" not works either. Here comes my tcpdump log, I am not an expert on SSH, Can anyone here get me out of this? Thanks Ryan [root at centos5u2 ~]# tcpdump -s 1520 -nn port 22 tcpdump: WARNING: peth0: no IPv4 address assigned

Hi, I am using read.table to get this data that has a timestamp. The data is for many days, but I only want to run the code I wrote only for specific day/days/times. I can´t figure out how to select a timeframe from the list. I have tried using subset() and didn't work I then used: * timestamp[for(timestamp==as.character("2011-03-15 00:00:00" ) |

Hi I've 3 iax connections to my provider , each of them have own DID , PH1<----| | \/ PH2<-->|-----| <---------------------------> |----|<-- DID1 | A1 | <---------------------------> |ISP |<-- DID2 PH3<-->|-----| <---------------------------> |----|<-- DID3 I had iax phone on each of this connection , but now I want to terminate all

Hi, I have two asterisk servers one is connected to the PSTN and the other one is connected to SIP users. The two servers connect with each other using IAX. When I have an incoming call from PSTN to the asterisk servers and have a forward to go back out to the PSTN the two IAX channel bridge together. Now every time I dial a DTMF digit, the asterisk is sending two DTMF digits. I enable

I'm new to this list, and I apologize if this is an already answered question, but my Google-fu was not strong enough to find the answer if it was. I'm having a problem with DTMF on incoming IAX calls. For the first few seconds of the call (between maybe 1 and 15, it varies from call to call) everything works fine. After that I continue get DTMF_E messages from the remote IAX server

What settings are people using? I've seen the ones from dslreports but I'm in that lucky group of people that paid the 1 euro just to have it no longer work. Even after I setup a additional account over the weekend it still doesn't work. And, of course, etherreal only shows encrypted traffic so I can't snag any config settings from it. Any assistance? -----Original

In order to get more information on IO performance problems I created the script below: #!/usr/sbin/dtrace -s #pragma D option flowindent syscall::*write*:entry /pid == $1 && guard++ == 0/ { self -> ts = timestamp; self->traceme = 1; printf("fd: %d", arg0); } fbt::: /self->traceme/ { /* elapsd =timestamp - self -> ts; printf("

Dear List, I'm hoping there is someone that might give me some pointers to solving the following problem. I run Mac OSX Tiger and perform a filecopy using Finder to a samba share on a Gentoo Linux machine running samba version 3.0.14a. Copying a 1GB file to the fileserver takes roughly 2 hours (33,9 MB in 4 min). This means 142 kb/s over a Gigabit network. Now here is the strange thing: When

The deflation request to the target, which isn't unaligned to the guest page size causes endless deflation and inflation actions. For example, we receive the flooding QMP events for the changes on memory balloon's size after a deflation request to the unaligned target is sent for the ARM64 guest, where we have 64KB base page size. /home/gavin/sandbox/qemu.main/build/qemu-system-aarch64

Rsync's "does this file need to be updated" check can conclude "this file does not need updating" even though the "last changed" timestamp differs. This happens when the size and modify timestamp are equal. Why doesn't rsync consider the "last changed" timestamp in the same respect as the modify timestamp? Doesn't changed mean, er, changed?

Hello, In Asterisk 1.4 beta 3, the UPGRADE.txt file says: Variables: * The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM}, ${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP}, ${ACCOUNTCODE}, and ${LANGUAGE} have all been deprecated in favor of their related dialplan functions. You are encouraged to move towards the associated dialplan function, as these

Hi all, I am using Samba 4.0.4 as AD DC on my test environment and realized that all my W2k clients (default installation, no special setups made on the clients) cannot receive the correct time of my samba 4.0.4 AD domain controller. Windows XP and 7 work fine though. The problem occurs at three W2k test clients I tried with. The default behavior of Windows clients is to use the update type

I am having some problems getting TelIax service to work with *. Outbound calls work just fine. When I try an inbound call the phone rings and there is no audio. Upon further investigation "iax2 show channels" indicates that the codec is "unknown" The provider confirmed that they are set for ulaw and so am I. Does anyone have an idea what could be causing the codecs to

It seems that FUSE can invoke flush to make sure the pending changes (e.g. to the attributes) of a file are set. Since a missing flush implementation is handled as if it were returning ENOSYS, this can cause issues later. To overcome this, just provide a stub implementation which does nothing, since we have nothing to do and don't want to have FUSE error out. Furthermore, uncomment the

Hi, My termination with sixtel stopped working, is it something I did or anybody else is having the same problem. I am attaching log: *CLI> -- Executing GotoIf("SIP/300-fbe0", "1?4") in new stack -- Goto (macro-dialout-default,s,4) -- Executing GotoIf("SIP/300-fbe0", "1?6") in new stack -- Goto (macro-dialout-default,s,6) -- Executing

I had a working 1.0.9 asterisk installation and tried to get a Firefly IAX phone to register, but it was failing. I upgraded to asterisk 1.2.5 and the PBX is working fine, but the IAX phone still won't connect. Below is my iax.conf and the output from setting iax2 debug while the phone tries to connect. Could somebody please give me some pointers? This doesn't seem to be a normal

Hi list, I know * generates it's outgoing RTP stream based on the incomming RTP stream, i'm having some audio problems after i recieve an rtp reinvite from my carrier. Situation: Phone -- Asterisk A --- Asterisk B --- Carrier --- PSTN Asterisk A: reinvite = no Asterisk B: reinvite = no If i dial out on phone via asterisk A, Asterisk B relay's the INVITE to the carrier, after the