similar to: Opinions needed on call quality vs network latency

Hi all, I am looking for a function for following calculation. start.month = "July" end.month = "January" months = f(start.month, end.month, by=1) * f is the function that I am looking for. Actually I want to get months = c("July", "August",.............."January") If start.month = 6 and end.month = 1 then I could use (not properly) seq()

Dear R-helpers, I need to read some data from output of garchFit in fGarch. my model is garch(1,1) and i want to read coefficients(omega,alpha,beta) and timeseries(x) and conditional SD(s). because i need them to use in other formula. for example :omega+x[1]+s[3] and maybe i have several simulation then i need a general way to read them, not to read with my eyes for example the quantity of

Hello everyone! I work for a company that uses OpenSSH to remotely support systems we've sold. Since some of our clients are US Dept. of Defense hospitals, our access to these servers needs to comply with a whole range of requirements and standards. At this point it's looking like the SSH daemon needs to be FIPS 140-2 compliant, and the only package that is certified is F-Secure.

okay, I started to look into winbind and the /etc/nsswitch.conf (and smb.conf)... and it wreaked havoc... I was using the guide at https://wiki.samba.org/index.php/Setup_a_Samba_AD_Member_Server - right guide? the guide is talking about "samba member server that is part of an active directory" Is that correct? I mean, THE ad domain controller is member of the ad, but it sounds like

Thanks Moises. I was kind of hoping that, at least if I hosted my Asterisk server somewhere where there was no NAT for the * box that the SIP phones wouldn't create any issues. How do you people with Hosted PBX handle the deployment of SIP phones behind NAT firewalls? Is it just elbow grease and configuring every single phone for the customer, or is there a way? Mike you can redirect

You cant go by pings. ICMP traffic is given lowest priority on internet routers, where voip rtp or iax might be given much higher priority. Plus I have 2 providers, the provider with the 90ms ICMP ping time is way better than the provider with the 15ms ping time. It depends on so many factors, including their equipment. I have a continuing problem with the voice dropping out for 1 second or

We are doing this with the latest spandsp, iaxmodem and hylafax. Seems to work very well for us so far. -Jonathan > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Michael Gaudette > Sent: Tuesday, March 21, 2006 3:34 PM > To: 'Asterisk Users Mailing List - Non-Commercial

Hello, The issue I'm having is when I try to run Wow in a standard user account vs root. When I run as just a user I reach latency of up to 2600ms. Along with that I'm unable to enable all my addons due to the fact that my frame rate will drop to at extremely low levels. But when I run as root (I know this is a bad idea..that's why I'm trying to fix it) I'm able to enable all

Yes I did. Fair question. I think it`s working, but is there anyway to know for sure? Show modules show app_cdr.so as existing... Mike On Thursday 26 Jan 2006 16:50, Micha?l Gaudette wrote: > Hi, > > I've just reinstalled Asterisk 1.2.3 on a fresh system and since I've > noticed that the CDR logging in MySQL (on a different computer) has > stopped. I thought it

That "program" will be detected by your ISP within a day or so, determined to be a virus, and your service will get disconnected...which n turn will not help your latency or jitter at all. VoIP can tolerate a fair amount of latency; latency over about 100ms is heard as a perceptible delay resulting in a connection that appears to be half duplex. Jitter, on the other had, is the real

[Moved from R-help] > From: fharrell@virginia.edu [mailto:fharrell@virginia.edu] > I store variable labels as "label" attributes of vectors > and use then in various plotting functions as well as the > describe() function. I would like to see general support for label attributes in the R plotting and modeling functions. One possible way of implementing this is to create a

Hi, I'm looking to implement Fax reception on a SIP line. I`ve been looking at the Wiki and some other web pages and it`s far from clear what I need to do, or if it`s even possible. 1) Is it possible, or does it only work on Zap channels? (as I`ve read somewhere) 2) Is there a good reference on the web to do so? Thanks, Michael

Hello, I'm a bit new to SIP, and I've set up a SIP line with Asterisk and my wholesale provider. That worked, fine. I ahd to open up the ports on my router, forward them to the correct box, again fine. Now, if I get one of my customers to connect his SIP phone to my Asterisk box, and HE'S behind a NAT firewall, does he have to go through the same process, or is it just the Asterisk

Dear List, I have the following function: coca <- function(x, ...) { if(is.null(class(x))) class(x) <- data.class(x) UseMethod("coca", x) } and a default method coca.default <- function(x, y, method = c("predictive", "symmetric"), reg.method = c("simpls", "eigen"), weights = NULL,

Hi, I've asked this question in the past, but I didn't get a precise answer. Hopefully somebody will take note of my question. Before I forget, I am using Asterisk 1.2.4. I've been using the Voicemail app with success (i.e. it works) except for one single thing: the ONLY message that it ever played back to the caller is the temporary message. If I delete the temporary message

Hi, I've had a bit of a problem with one way audio, and it happens exactly when I believe it shouldn't (and works perfectly when I would guess I could have issues. Setup: GrandStream GXP2000-------Linksys Router-----------Internet------Asterisk box (hosted somewhere, fixed IP, no NAT) ----------- VoIP provider -------PSTN When a call comes in from the PSTN, the call goes all the way

The ffmpeg2theora chapter[1] of the Theora Cookbook says that the only advantage of two-pass encoding is that you can hit a specific target size. But isn't it so that you can reserve bits for difficult parts of a video, which improves the perceived quality? It may be true that Theora does quite alright without two passes, but it seems like a no-brainer to me that two passes will always be

Hi, I`ve recently upgraded a server from 1.8 to Asterisk 13. While everything is under control, I have one issue with the way CDRs are kept for queues. And I don`t mean ?I don`t like it?. I mean it crashes the server. I realize there are multiple CDRs per queue call ? one per ring/per phone, basically. The issue is that whenever the number of CDRs ?to be recorded? for a call exceeds 5000,

Hi, I'm having an issue where some phones behind a sonicwall are auto-congesting. The status on "sip show peer" shows ping times anywhere from 80ms all the way up to 1100ms. PCs behind the same firewall have a ping time of about 30ms to the PBX itself. Does anyone know if the sonicwall is inserting delay into the SIP signaling path and lagging the OPTIONS messages for qualify?

Here's the problem: > 2) Encoding with rate control as in single pass "Bitrate > control" will not > lead to better quality than fixed quant (with the right value > of the fixed > quant). Ratecontrol doens't know anything about "quality". It > will try to > reach more-or-less CBR. > > But somehow this is not a fair comparison, because