similar to: Echo on a PRI

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@Michel using virtualenv we can create an isolated environment in the system. Just go through - http://docs.python-guide.org/en/latest/dev/virtualenvs/ @Eric I'm using Ubuntu, and I was able to install and use libvirt using sudo apt-get install python-libvirt but its not available in tg2 virtual environment. Moreover if you try to install in the virtualenv using the sudo apt-get install

Dear useRs, I'm having troubles with using ylim in barplot(): even though I reduce the y-scale using ylim, the bars still extend down to 0into the x-labels. The sample data is below, and here is the code. #This works fine but I would like to plot only from 50 to 70: barplot(t(mydata), beside=TRUE, col=c("orange", "green", "yellow", "purple"),

Does anyone have a working NFAS config for Zapata and zaptel for 2 NFAS trunks? First two DS1s on tg 1 and other two on tg2? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050825/95961694/attachment.htm

Hello, I am having an issue with making two simultaneous outbound calls. When I dial, both phones try to take the same channel and it causes an error. Anyone have any suggestions. My set up is as follows: CO - PRI - ASTERISK - VODAVI(pbx). Thanks, Dave *CLI> -- Starting simple switch on 'Zap/69-1' -- Executing Wait("Zap/69-1", ".1") in new

Hi, Could you guide me to install libvirt in a virtual environment..? OR How python-libvirt could be used within a virtual environment after installing it, in the system. Regards Sijo

Hi, We have 7 Dell 2850 servers with dual xeon 3 gig processors running the APF firewall version 0.9.6 http://rfxnetworks.com/apf.php They run fine for a day or two, then suddenly lock out all incoming connections, other than the backend IP, sometimes restarting the firewall resolves this, but occasionally we may have to leave it 10 mins or so before restarting where it will actually

If you are using a Windows system, you can Export the plot from RStudio and save it as a metafile without using package devEMF and it will crop the bars with xpd=FALSE. When I used devEMF on a Windows machine, the bars were not cropped with barplot() as you indicated, but when I switched to plotrix::barp() they were cropped. The arguments are a bit different, but I did not need xpd=FALSE:

I have Asterisk configured to record calls. Both in and out record ok but SoxMix fails to join the two files. The error from the CLI is as follows: Execute of ( nice -n 19 soxmix /var/spool/asterisk/monitor/Support-in.wav /var/spool/asterisk/monitor/Support-out.wav /var/spool/asterisk/monitor/Support.wav && rm -f /var/spool/asterisk/monitor/Support-* ) & failed. If I run exactly the

This was an interesting post from another list I'm on. I think that with all of the echo cancellation discussion happening, that someone perhaps might be on the lookout for this type of software, so this post was well-timed for Asterisk uses. It is perhaps the case that those of you working on volume/echo/etc stuff might find this useful towards your goals. JT >To: [snip]

Thank you David, That's a nice workaround using plotrix::barp(), but that doesn't explain why ylim doesn't work as intended (or at least, as I expect it to work), or why xpd has no influence when using devEMF::emf()... The problem with saving directly in RStudio is that it requires to manually save the plot, and this becomes troublesome when there are a lot of plot commands in a

Hi, does anyone have the setup for go2call ? I have digium boards and quicknet linejacks and phonejacks. The cards work fine in asterisk without the g729 or g723.1 for the phonejack. I will like to do SIP origination using the codec in the phonejack and linejack g729 or g723 and send the calls to go2call. Anyone has the setup for this ? Or similar setup to a SIP provider using g729 or g723

Hello! After upgrade to version 2.0, now using 2.0.1, I'm experiencing problems with glusterfs stability. I'm running 2 node setup with cliet side afr, and glusterfsd also is running on same servers. Time to time glusterfs just hangs, i can reproduce this running iozone benchmarking tool. I'm using patched Fuse, but same result is with unpatched.

Client: Dual 1GHz G4 OSX 10.2 Gig-Ethernet Server: Dual 2GHz P4 Linux 2.4.18 Raid-5 1TB Gig-Ethernet With netatalk 1.5.5 I get sustained writes of 66MB/s (yes, megabytes) With samba 2.2.7a I get sustained writes of 15MB/s I've tweaked the settings and ended up with these: socket options = TCP_NODELAY SO_RCVBUF=8192 SO_SNDBUF=8192 IPTOS_LOWDELAY SO_KEEPALIVE read raw =

I am planning to buy Dell 2650 server with dual Xeon processors. And I would like to buy two TE410P cards for PCI with 3,3v. This is on Dell site about PCI slots for Dell 2650 server: 3 PCI-X (1x64-bit/133MHz, and 2x64-bit/100MHz) Does that mean I will be able to buy two TE410P cards ? Or I need to buy two TE405P cards ? Thanks for help. bartek

Hi All, This is out of the topic question, but I''m hoping that you can help me. If for example I''m going to install a Fedora 4 64-bit with 8 to 12 Gig of physical memory, how much swap file of directory do I need to create? Thank you very much, Wennie

Hi everyone, I want to fit the following mixed effect model Y_ij = b_0i + b_1i * (t_ij*grp_ij == 1) + b_2i * (t_ij*grp_ij == 2) + v_0i + v_1i*t_ij + e_ij with a different covariance matrix of random effects for each group. (Y is the response t is time grp is the group indicator b 's are fixed effects v 's are random effects) I know that this is possible in SAS

On 03/16/2017 04:22 PM, Kevin Stange wrote: >> I still can't rest assured the NIC issue is fixed, but no 4.4 or 4.9 >> server has yet had a NIC issue, with some being up almost a full month. >> It looks promising! (I'm knocking on all the wood everywhere, though.) > > I'm ready to call this conclusive. The problems I was having across the > board seemed to

Hello, I''m trying to get lossless VoIP traffic over my 3000k/500k ADSL line. Shaping outgoing traffic is no problem: I set total ceil for outgiong device (ppp0) to 450kbit and put VoIP into highest prio class. Even during full upload the voice is clean on the other end. Now I tried to get the same result for incoming data. I attached HTB to eth1 where the incoming voip traffic is

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type"> </head> <body bgcolor="#ffffff" text="#000000"> I have an Asterisk 1.4.18 with a mix of cordless phones connected using Linksys SPA2102 ATAs and Cisco 7940G