similar to: MeetMe errorhandling

I have two CentOS5 systems server1 and server2. There is user peter on server1 who can ssh to server2 using public ssh keys and no password is needed. What I noticed is that running remote ssh commands in bash script breaks while loops. ====== #!/bin/sh for i in server2 server2; do echo "--> Start" ssh peter@$i ls echo "--> END" done echo " server2

I have Wine 20041019 and have done all the configuration with Wine tools. Then I Installed Photoshop 7 by executing ~/bin/dos then going to the cdrom drive and running the setup executable. The setup went fine and photoshop was installed. Then I simulated a windows reboot from wine tools. Then i again executed ~/bin/dos and went to the place where i Installed photoshop 7 and executed

Dear Recently,I am interested in libvirt,but I found the libvirt does't support revert and delete the external snapshot which the raw disk based guest.My libvirt and qemu release are both 3.2.0.So I have a question,the community how to solve it,if have any plans about it.Can you tell me more about it?Thank you very much. Best Regards Rambo

I have the problem with rcom 1.2.2 package. It does't work with R 2.2.0. Is there any opportunities to correct the work of this package? Thanks for your help. Yulia

I have a command that reads in some data: x <- read.csv("Sales2007.dat", header=TRUE) Then I try to organize the data: sc <- split(x, list(x$Category, x$SubCategory), drop=TRUE) Then I want to iterate through the data. I was able to get the following to run on the R console: for(i in 1:length(sc)) { sum(sc[[i]]$Quantity) } But notiing is primted on the console. I find

Hi I've installed CentOs 6 successfully, but, Firefox does't lunch. I've downloaded Chrome and when I try to execute "google-chrome &" command, I had command not found. --

Dear All, I bought one IP PHONE from Integrated Networks which was showed to wiki too: http://www.voip-info.org/tiki-index.php?page=Intergrated-Networks I have problems with the Asterisk authentication. It does't want to LOG IN to Asterisk; it always says "LOG ON FAILED". I'm using the IAX2 protocol and all paramters seems to be correct. Does somebody use the same IP PHONE with

Hi all, there is a vector v with several NAs. I want to create a new vector n of the same length as v and the same NAs as in v and tried this: n <- vector(length=length(v), mode="numeric") replace(n, which(is.na(v)), NA) but this does't work, all values in n are 0. What went wrong? Thanks, Sven -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-

I have no speex translation ulaw alaw gsm g726 g726aal2 adpcm slin8 slin12 slin16 slin24 slin32 slin44 slin48 slin96 slin192 lpc10 ilbc g722 testlaw ulaw - 9150 15000 15000 15000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 17250 15000 alaw 9150 - 15000 15000 15000 15000 9000 17000 17000 17000 17000 17000 17000

Hello, I haven't been able to find an example for the second case below -- or perhaps I didn't recognize it when I saw it. Is there a value for x such that svalue(x) will return "bbb", either by itself or as part of an array? Or do I need to do something else entirely? (R2.5.1; Windows XP) > #### gWidgets test > options("guiToolkit"="tcltk") >

I have two questions about R: 1) Does anyone knows why I get the plots but not the summary results of the linear fitting when I run this code? When I run it outside the if condition it works fine. if (op==4) { t=1:length(time(SeaIce)) fitlinear=lm(SeaIce~t) windows() plot.ts(SeaIce) lines(ts(fitlinear$fitted.values,start=1979,frequency=12)) summary(fitlinear)

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi folks. Please, someone confirm if i'm right. if i'm wrong, please forgive-me. - -------------------------------------------------------------------------------------------------------------- I've developed a little tool to stress test tcp connections.( Sending syn and answer ack-syn ). that simuates a real tcp connection. (

The duplicate symbol errors are for __llvm_profile_filename and __llvm_profile_raw_version. IIUC, these are supposed to be weak symbols but Thin LTO seems to break this in some way. This does't happen with gold or no LTO or full LTO. $ cat > a.c extern int foo(); int main() { return foo(); } $ cat > b.c int foo() { return 0; } $ clang a.c -fprofile-generate -flto=thin -c $ clang

Am 13.06.2020 um 13:47 schrieb Michael Keuter: Hi > Try "sip show peer <peername>" for a phone. So: mobile phone: bpi*CLI> sip show peer 0049177xxxxxxx * Name : 0049177xxxxxxx Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Record On feature : automon

I installed a hardware g729 codec card in my asterisk, and I'm getting the following error when calling from a g729 sip extension to a SIP trunk also set to g729. The call goes through just fine, but these error messages keep flying by until I disconnect the call. Any ideas? ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5

Hi all! Asterisk 13.1.0 Ubuntu 16.04, all latest. Can anybody explain this to me - I run Originate command from dialplan: same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum}) and I get crazy sound distortion in the conference, and I see that transcoding takes place here: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin

Again - when Originate is run from dialplan, i get: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin at 8000)->(slin at 192000) ReadTranscode: No When it's made with a call file (no matter how a call file is created), I see NativeFormats: (slin) WriteFormat: slin ReadFormat: slin WriteTranscode: No ReadTranscode: No Please

I need to make a Asterisk 18.0's offline compiling, SVN mp3 support sources downloading does't particulary works cause my asterisk is in an isolated network with NO network access whatsoever, I ve read this thread ( http://lists.digium.com/pipermail/asterisk-users/2013-June/279298.html) but I 'm not understading one thing, because I download the file and run the script but there is no

Greetings everyone, I was wondering if there was a way to change the codec that Asterisk uses when streaming via MulticastRTP. Or perhaps a way to transcode the multicast stream. In the CLI, when I have a multicast stream in progress, I am typing 'core show channel MulticastRTP/0x7f7........' to get lots of helpful information. I have noticed that when I do a MULTICAST page and send data

Hi Users, I was wondering if someone of you have the same thing on CentOS 64x? asterisk01*CLI> core show translation Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 siren7 siren14 slin16 g723