search for: ledges

> From: Frank Varnavas <varnavas@ny.ubs.com> > Subject: RE: NT server dropping APPS due to samba > > I sent the following to the first guy reporting the "NT server > dropping apps due to samba" problem. I haven't had any feedback so I > don't know if it helped him or not. > > The synopsis is that when samba does his 'man in the

I am seeing lots of lost packets when running the command sip show channelstats at the CLI. There are issues across multiple Asterisk servers I am trying to diagnose but everything I read seems to point to this command being pretty unreliable. Can I trust the info this command shows? I am showing lots of lost packets in sip show channelstats but I can't see any packet loss when pinging the

...even to truncate to 1000 characters, per 282[12]). However, I kinda thing dovecot shouldn't choke on the results of truncation, which is what I think is the problem. Amy! -- Amelia A. Lewis amyzing {at} talsever.com I have spent nights with matches and knives, leaning over ledges, only two flights up. Cutting my heart, burning my soul. Nothing left to hold. Nothing left, but blood and fire. -- Indigo Girls

Hello all, this is my first message to the list. I have wine 20041201, installed on my slack 10.1, and it was working fine, until I installed a software and then all fonts in menus and messages became weird symbols. I am sending a screenshot to show how it is today. Also I will let the image in http://www.igc.usp.br/pessoais/guano/geocalc_wine.jpg Does anyone know what is going on and how to

I've seen something similar with Adtran SIP gateways. When a re-invite happens the Adtran gets all confused about call stats and marks the pre-reinvite leg of the call as losing large numbers of packets. BTW, IIRC reinvites happen when a codec changes or the channel switches to T.38. Also Adtran SIP gateways appear not to support OPTIONS packets when running in SIP proxy mode, which is

...; is supposed to be doing. (Confession: I'm such a newbie that I've spent much of the day typing "?apply" and then "?par" etc, working through the program to try to figure it out!) In any case, my worry is visible in the QQ plot attached. Why do I have get so few "ledges" from the segments() calls? It's not clear to me how the scale of the data-model deviations come into this problem. Q: should I first be standardizing the residuals from the data-model comparison? I would have guessed that something in this 'QQ envelope' would be doing that....

Hi! I don't know much about network programming, or about terminal handling. I wondered about why the command "scp file user at host:path" didn't work. It gave me the following error messages: "stdin: is not a tty" "Write failed flushing stdout buffer." "write stdout: Broken pipe" I tried it with different versions of ssh, first the newest debian

You could use MTR command. Its a trace route improved. Marlon Araujo > On Jan 20, 2015, at 08:59, asterisk-users-request at lists.digium.com wrote: > > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or,

I would recommend capturing traffic outside your Asterisk server with Wireshark, then running the Telephony/Rtp/Analysize Streams option to determine if you have packet loss at that point in the network. On Mon, Jan 19, 2015 at 1:00 PM, Todd R. <tjrlist at live.com> wrote: > Thanks but no Adtran here. > > I do think these stats are indicating an issue, I just don't know how to

Additional info: At the moment I am running 1.8.x but the other day I was getting the same results on 11.x Here is a sample from show channelstats. I do think this command is showing that there is trouble between specific IP's and my Asterisk box but I don't know if the numbers are accurate and reliable. Peer Call ID Duration Recv: Pack Lost ( %)

Thanks but no Adtran here. I do think these stats are indicating an issue, I just don't know how to prove it outside Asterisk. From: EWieling at nyigc.com To: tjrlist at live.com; asterisk-users at lists.digium.com Date: Mon, 19 Jan 2015 13:55:33 -0500 Subject: RE: [asterisk-users] sip show channelstats reliable? I?ve seen something similar with Adtran SIP gateways. When a re-invite

On Tue, Jan 20, 2015 at 12:43 AM, Scott Griepentrog <sgriepentrog at digium.com > wrote: > I would recommend capturing traffic outside your Asterisk server with > Wireshark, then running the Telephony/Rtp/Analysize Streams option to > determine if you have packet loss at that point in the network. > > On Mon, Jan 19, 2015 at 1:00 PM, Todd R. <tjrlist at live.com> wrote:

hello list!! I am still having some file sharing issues with this puppet server and I was hoping someone would have an opinion they could share on this situation ## puppet errors err: //basefiles/File[/etc/yum/pluginconf.d/priorities.conf]: Failed to retrieve current state of resource: Error 400 on SERVER: Not authorized to call find on /file_metadata/acadaca/yum/pluginconf.d/priorities.conf

Happy holidays! I am experiencing an issue when trying to check my mail using IMAP. with Dovecot I have tried checking my mail using a full GUI client (Thunderbird) and telnet. Both times I get disconnected before all of my messages can be downloaded and I see an error in my mail log. Here are the details: [root at cust19-1-prod-domain userqa]# dovecot --version 2.0.9 [root at

Hello all In a recent spate of paranoia we set our server (SuSE Linux 7.0, kernel 2.2.16) to use SSH version 2 and not SSH1. With openssh 2.3.0p1-5 running as client and server, we find that stdout output is occasionally dropped: ssh server echo "JJJ" usually emits JJJ, but sometimes returns nothing -- although the command is apparently performed. In the happy case the server logs