similar to: Codec G729 Registration problem

Hi i have buy two license of G729 codec, but after run Registration program i notice this error Registration unsuccessful... Error code: 110 ERROR! Your Internet connection is probably behind a proxy and the Registration program can't communicate with our server, however it has created the file: /var/lib/va-infoclient which contains your machine signature and

So if I first create the value in an entry BB and then build a CopyToReg but then I have to read it in a BB that loops back to itself, with it's own CopyToReg at the end, then I have two CopyToReg nodes for the same value. In this case, I need to create 3 virt regs, 1 for each CopyToReg and a third for the CopyFromReg in the beginning of the loop BB, right? And then I need to build a PHI

Hello all, I am developing an out-of-tree backend for a unique simd processor and I'm looking for a bit of help. In my current design, it's possible for a case to arise where I decide during instruction legalization that a fixed-sized array of vectors which has been initially allocated on the stack needs to be lowered into the vector register file. I have this design working for the case

Classification: UNCLASSIFIED Caveats: NONE A similar question has been posted in the past but never answered. My question is this: for probit analysis, how do you program a 95% confidence interval for the LD50 (or LC50, ec50, etc.), including a heterogeneity factor as written about in "Probit Analysis" by Finney(1971)? The heterogeneity factor comes into play through the chi-squared

So if I create a value with a DAG.getUndef(myVT); call during instruction legalization, how can I access that value as input in another BB/DAG (also during instruction legalization) without worrying about live-ins and/or phi nodes? Can I create a single virtual register and build both a CopyToReg and a CopyFromReg node with it? I assumed that would break SSA. Perhaps I should have said that what

Hello, I'm trying to find out if Asterisk will support plain G729 or G729b. I've read all over that it supports G729, but I can't seem to find any explicit remarks regarding the specific versions of the codec Asterisk will support. I noticed that Digium allows Asterisk users to register and download G729a, but refers to it as G729 on it's pages. I also noticed that on

I own a G729b codec from voiceage which I had from Digium a couple of months ago , I friend of mine had the new Digium G729 codec which registers in the asterisk as a Annex A/B codec, the problem that we saw is that the call goes thru find but we cannot here any sound. Asterisk is showing this : -- SIP/10.10.1.1-babc is ringing -- SIP/10.10.1.1-babc answered

Hi, after reading about it on the list I decided to set up a Free World Dialup account. For those of you who don't know, that is a sip proxy where you and your friends can singn up free and then you can just connect to it with any sip client and call anybody that is registered for free. Pretty much like iaxtel (I belive that was the name of it) for the iax protocol. It even supports clients

Original Post ---------------- I have an Asterisk related problem with mutualphone. I can connect to any number with any softphone that I am using (iaxcomm, SJPhone, and a few others). Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to mutualphone destinations. Other destinations go fine. A working phone call (e.g. from iaxcomm) gives the following on the console: --

<div class="socmaildefaultfont" dir="ltr" style="font-family:Arial;font-size:10.5pt" ><div dir="ltr" style="font-family:Arial;font-size:10.5pt" ><div dir="ltr" >Matthias,</div> <div dir="ltr" > </div> <div dir="ltr" >Thanks for the response.</div> <div

Hi all, I''m looking for a function, gem, etc. that will help to unify dates as they come in different formats. If the date comes in (1/2/2012,2012-01-02,01-02-2012,20120102,01022012, etc.) it would be nice to have it come out 2012-01-02. I''ve searched for a few days and only find ways to convert this format to that format. I might be using the wrong search terms and this

Seeing that many people here hit problems with activating their G.729 licenses, I decided to post my opinion. I have purchased two G.729 licenses, for my private use. I did this even though VoiceAge makes G.729 free for private use, as Windows libraries. I guess a sufficiently motivated person could take the COFF libraries, run them through objcopy on cygwin (producing ELF .o files) and link them

According to digium's site, "Note: Please do not attempt to use the G.729 code in a SCSI-only system. We are currently working with VoiceAge to correct this issue." (found at http://www.digium.com/index.php?menu=asterisk_g729). Does anyone know what these issues are? Can anyone define SCSI-only system? I know this sounds kinda dumb, but I have a server with SCSI and IDE interfaces,

I've just setup a new asterisk server, and I need to have G.729 working on this system. The problem is I don't have any IDE drives (and therefore no /dev/hda etc), but only /dev/sda.   Is there really *no* way to license G.729 on a SCSI-only system? IMHO it's really stupid to replace an entire server because of a licensing issue. There *must* be a solution.   Anyone, please? Or at

I have multiple voiceage g729 licenses installed on a RH9 box, and have a remote C7960 configured to use it (low bandwidth). In calls like: Remote C7960 -> g729 -> asterisk -> g711 -> C7960 the audio is oftentimes rather choppy. Changing the remote 7960 to use g711 seems to eliminate/reduce the choppyness. Any ideas on what might be behind this?

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<div class="socmaildefaultfont" dir="ltr" style="font-family:Arial;font-size:10.5pt" ><div class="socmaildefaultfont" dir="ltr" style="font-family:Arial;font-size:10.5pt" ><div dir="ltr" > </div> <div dir="ltr" >Hi all,</div> <div dir="ltr" > </div> <div

This one is for the archives. I got a call today that the * at one of my clients was not working. The switchboard is set up to ring for a while and then the rest of the phones start up if the switchboard doesn't pick up. This was not happening. Instead the mobile phone of one of the people there was ringing and after the delay the internals started ringing. When I connected to the web

I'm having trouble with the standard deviation distribution as shown on http://mathworld.wolfram.com/StandardDeviationDistribution.html . (Eric Weisstein references Kenney and Keeping 1951, which I can't check.) I believe the graphs they show, but when I code the function in R, according to the listed formula, I get very different graphs. Would someone please point out my error or tell

Hello all, We have the following problem: When calling via iax, the sound is off after a while - most often after about 5 minutes (sometimes later or earlier) - at one end or at both ends. While the channel is up, and packages are still being transmitted, you just can't hear anything. Sometimes you can hear something just a little, but with the voice greatly distorted, sounding like a