similar to: Two asterisk servers, two different service providers

Hi, I'm trying to automatically have the dialplan assign an extension to a roaming phone on my network. I tried the following without success: exten => 3001,1(readop),BackGround(beep) exten => 3001,n,Read(digito,vm-youhave,3) exten => 3001,n,SayDigits(${digito}) exten => 3001,n,Set(ROAM=${digito}) exten => 3001,n,Set(DB(roam/ext)=${digito}) exten =>

Hi, The following seems to be a bug introduced with version 1.8.0 (it worked without error on my installation of version 1.7.0): > test <- data.frame(time = I(c("2003-08-19:22:55:57"))) > test[["time"]] <- strptime(test[["time"]], format = "%Y-%m-%d:%H:%M:%S") Error in "[[<-.data.frame"(`*tmp*`, "time", value =

Dear all, I tried to install rgl package on my linux machine fc5, I got an error. Here I run R as user, $ R > options(repos=c(CRAN="http://cran.at.r-project.org/")) > install.packages("rgl", lib="/home/subianto/local/lib/R/library/site-packages", dependencies=TRUE) trying URL 'http://cran.at.r-project.org/src/contrib/rgl_0.67-2.tar.gz' Content type

We generate xEXT nodes in many cases. Unlike GCC which vectorizes inner loops, we vectorize the implicit outermost loop of data-parallel workloads (also called whole function vectorization). We vectorize code even if the user uses xEXT instructions, uses mixed types, etc. We choose a vectorization factor which is likely to generate more legal vector types, but if the user mixes types then we

"Rotem, Nadav" <nadav.rotem at intel.com> writes: > We generate xEXT nodes in many cases. Unlike GCC which vectorizes > inner loops, we vectorize the implicit outermost loop of data-parallel > workloads (also called whole function vectorization). We vectorize > code even if the user uses xEXT instructions, uses mixed types, etc. > We choose a vectorization factor

All: My company has an existing ESI IVX E-class system with 45 phones. I can add one more card, to expand it another 6 phones, but it's $8000, and then the system will have to be replaced. I have the Asterisk server up and running, with 2 sip lines from the local phone service. (Thanks to you guys, it is working great!). I'm pretty sure this is the way the company will move, and

All: Getting ready to put the system in production. Any suggestions on "stress testing" the system? I'd like to initiate say 10 sip phone calls to make sure the provider has the bandwidth. Can you do that in CLI? I've called 4 numbers simultaneously with the hard phones I currently have and am thinking of adding 6 or so soft-phones to various pc's to make a total of

Hi All, Have completely moved off the old ESI system, and things have been going pretty good with the new server. I have one issue, which has been reported by several of our customers. I've tested it, and it does indeed seem to be a problem. When the customer is asked to dial in the first three letters of the person they are trying to reach, they will be routed to the wrong extension.

All: Yesterday I discovered something interesting. I dialed 1800ANCESTRY from the asterisk system I am testing and got the number doesn't exist message. I then dialed the same number from our old system and it went through. I realized that the "Y" in ancestry made the number too long, and went back to my dialplan. How do I ignore numbers that are too long? Obviously,

Jim, and all: Thanks for the response. If I can repeat what you are saying: you don't have to define the multiple lines in sip.conf? For comparison, with my current esi setup, we have 10 outgoing lines. If one line is busy, then the service just rolls to the next number. This is set up with the phone service. That doesn't have to done with outgoing sip lines? Does the dialstatus

Hi Everyone, I have been searching the web and I don't know if SNMP is just that complex to setup or that not many people use SNMP to monitor Asterisk but the information is scattered all over. I have got to the point to configure SNMP with Asterisk and then it's all confusing from there on to actually see the graphs in Cacti. I would appreciate it if you can post your steps or point me

Greetings, A fresh install of 7 followed by a cvsup to 7.2-PRE on the 26th results in an inability to build Xorg on the system. A cvsup only an hour ago provides no solution. An attempt at the following: cd /usr/ports/x11/xorg-minimal make produces the following error: ... checking pkg-config files for X11 are available... yes checking for LIBDRM... yes checking for DRI2PROTO... yes checking

Hi Nadav, > I had a few thoughts regarding our short discussion yesterday. > > I am not sure how we can lower SEXT into the vpmovsx family of instructions. I propose the following strategy for the ZEXT and ANYEXT family of functions. what I would like to understand first is why there are any vector xEXT nodes at all! As I tried to explain on IRC, I don't think you ever get these

Duncan, I had a few thoughts regarding our short discussion yesterday. I am not sure how we can lower SEXT into the vpmovsx family of instructions. I propose the following strategy for the ZEXT and ANYEXT family of functions. At first, we let the Type Legalizer/VectorOpLegalizer scalarize the code. Next, we allow the dag-combiner to convert the BUILD_VECTOR node into a shuffle. This is

I'm having a problem with IAX running through an intermediate asterisk box. Perhaps a small diagram will explain the situation better: *A ------- [cloud (public internet)] ------- *B --------[cloud (private network)]----------- *C Asterisk server's A, B, and C, are all connected together with IAX All asterisk servers are 1.6.0.6 Server A and B are geographically close, but connected over

All: Some of the people who dial into to our system will press the pound key when entering an extension for the directory key. When waitexten gets that, I get an error messages as, for example 123# doesn't match any extension. I was going to use ${EXTEN} to just use the first three numbers, but I'm not sure how to use this with WaitExten. so I have exten =>

All: Still experimenting with the asterisk server for the company. My local phone company has given me two sip numbers to experiment with, say 444-456-1234 & 444-456-5678 Calling in and out works, and I've spread a couple of the phones out with some co-workers. My question is this: Do I have to define multiple sip lines in either the sip.conf or the extensions.conf? Now when I

All, The users in our organization are well, quite frankly, sick of phone service that is being provided. The choppy phone calls, and drop outs are detrimental to our sales force. I've tried about everything I can think of. Moved the asterisk server from VM machine to dedicated machine More than enough bandwidth Setting 802.1p = 7 Set Dedicated voice traffic 35% of bandwidth. Not sure

Hi Everyone, I have a SIP turnk which works fine with both inbound and outbound calling. However, the only issue is that there is no Ring Tone if someone calls us. The phones used are Aastra and Polycom connected to the PBX via VPN (SIP). I do get an outbound ring tone, so it's not that there is any media loss between the phones and the PBX. But when the DID is called there is dead silence

We are running our * server as a virtual PBX for 6 companies. I am having all of the Allison prompts plus our own custom IVR prompts being re-recorded for each company, in a different voice (marketing thing) with a different personality (perky, corporate, earthy) . I'm curious if someone could point out a dirty trick to get the voice to play right, for internal and external callers,