similar to: trunk to trunk forwarding

Say I have agents using a softphone like eyebeam that has 6 lines. They log in to the queue. Say there are 3 agents in my queue. 3 calls come in and all three agents are on a call. Now a fourth call comes in. Is it possible to have it setup so that the 4 call rings on line 2 of one of my agents, if they don't get it within the time limit it rings on line 2 of another agent and so on. An

I'm curious what asterisk solutions there are out there for inbound call centers with multiple DIDs. I'm looking for solutions for a setup where single system may have 1000 DIDs going to it, one for each account. Each account may not get that many calls. Solutions that will all reporting on calls coming into different accounts, call routing for queues based on distribution groups. Like

There is an issue with removal of parallels disk paravirt driver. Without the patch, the VM being migrated may not boot on the desination with BSOD: 0x7b. The patch is a workaround to prevent the BSOD. Signed-off-by: Denis Plotnikov <dplotnikov@virtuozzo.com> --- v2v/convert_windows.ml | 2 ++ 1 file changed, 2 insertions(+) diff --git a/v2v/convert_windows.ml b/v2v/convert_windows.ml

Hello, I have two FXS port on my TDM card. channel 4 is attached with a telco line that I use frequently. And channel 3 have another telco line. but I dont publish that number to my friends. If I receive a call through channel 4, how can I handover that call to channel 3 ..so that I can keep channel 4 open for incoming call. Thanks,

Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem calls, then route them out the Internet. The Telco they were buying the trunks to discovered this configuration and restricted them due to legal conventions, and stated that in order to continue doing this, they would have to talk

I'm having some confusing problems transferring approximately 16GB, in about 541,000 individual files. What is happening is that I am starting with a blank destination directory on a gentoo system running samba 3.0.1, and pulling from one of two windows systems. The problems come in that each time I use rsync (version 2.5.7) over samba to keep the data on the gentoo system's side in sync

Hi , I would like to use ggplot2 to plot a matrix as an image. You can copy paste the following Data<-matrix(data=rnorm(900,80,20),nrow=30,ncol=30) lengthOut<-5 Lengths<- 15 library(reshape2) library(ggplot2) tdm <- melt(Data) ggplot(tdm, aes(x = Var2, y = Var1, fill = factor(value)),levels=seq(0,1,by=0.1)) +                   labs(x = "MHz", y =

Hi: I'm using a TDM-400 to terminate PSTN lines at my Asterisk server with kewlstart signalling. When an outside caller calls the server, the TDM-400 goes off-hook and provides a ringing tone to the caller. If the caller hangs up before the receiving party answers the phone, Asterisk fails to detect the hang-up. The TDM-400 stays off-hook, hogging the line, while Asterisk rings the

I am still having unacceptable echo on my X101P and twidling with the rx/tx gain levels and echo settings appears to have no discernable effect. Some questions for those who may have more significant electrical engineering background than I. 1. This impedance match thing ... will it affect this solution having other phones in parallel with the X101P? This is done so that I can test while not

With our current pbx system, a call comes in from the PSTN to the receptionist. She then hits flash, which puts the caller on hold, calls my extension, says "so and so is on the phone for you", I say "ok put him through", she hangs up and I am connected to the caller. With asterisk@home I can it # then the extension to transfer to and it will ring there. But is there a

Hi All, I am looking for a way to troubleshoot issues with TDM (E1) trunks with a provider. Currently with SIP trunks I am using tcpdump to perform packet captures between our gateways and the SIP providers IPs, capturing traffic on all ports, to include both the SIP messages and the RTP stream. How can I achieve a similar result on TDM links connected to TDM cards in Asterisk servers, where by

We have implemented * in one of our branch offices and recently ran up against a very strange issue. On random occasions, when we would dial out using our trunk lines, we would get a message stating "you do not have to dial a 1 or 0 when calling this number" even if we didn't dial a 1 or 0 in the dial sequence at all. After much troubleshooting, we found users with similar issues

On Feb 23, 2009, at 1:46 PM, Chris Lattner wrote: > In my opinion, the proper direction for shuffles is: > > 1. Back out your patch. > 2. Move the functionality of "is splat" etc to method somewhere, e.g. > on SDNode. > 3. Introduce a new ShuffleVectorSDNode that only has two SDValue > operands (the two input vectors), but that also contains an array of > ints in

Hi all- My TDM long distance is provided by MCI. We use account codes where MCI sends a challenge tone after receiving 1NXXNXXXXXX. Anyone have any suggestions of how to accomplish this? I can't get the soft phones to send the DTMF (the other digits go down the d-channel of our PRI). I also have not bee able to get the dial or the outgoing queue command to work. Anyone run into this?

On a 55 station install onto a Cox PRI with a TE110P (Polycom 501 phones) a few users are complaiining about echo. According to the users, the echo seems to be phone number dependant. They claim that certain phone numbers have echo while others dont. Are there any tuning parametes like there is for a TDM400 card? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's

Folks, I'm experimenting with bringing multiple (8) analog lines from our local telco into a Carrier Access ADIT 600 channel bank with an FXO module, then having this talk to Asterisk via the T1 TDM controller on the ADIT and a TE405P card. I don't know if this will work well (ie: give me decent echo cancellation, call disconnect supervision, caller ID etc) but I haven't had much

Hi, I read in the Asterisk Whitepaper, that you can run two cloned servers, one as a primary, one as a backup, and have them automatically failover to the other unit when it crashes, or when you need to restart it. The primary application of course, would be ensuring calls can be made when frequent updates are being handled, or when an update must be restarted on a busy network. The term

Does anyone have any suggestions for connecting analog DID trunks? I have some small locations that will have 2 analog DID trunks each, the only solution that I can see will work will be using a channel bank and T1 card, but it will be close to $1500 to terminate these DID trunks. Was hoping someone had some experience using an ATA or TDM card and analog DID trunks. Rhino Channel Bank - $750 4

I've been replacing an old Toshiba DK switch with an Asterisk solution. I'm needing a solution for fax machines that works as well as a POTS line from my carrier. If the POTS line is the solution, I'll keep it, but I'd rather move away from that. Here's what I'm thinking...will it work? I would use a dual-port Digium T1 card. In one port, I'd terminate a telco PRI

On Mon, Feb 23, 2009 at 2:19 PM, Nate Begeman <natebegeman at me.com> wrote: > > On Feb 23, 2009, at 1:46 PM, Chris Lattner wrote: > > In my opinion, the proper direction for shuffles is: >> >> 1. Back out your patch. >> 2. Move the functionality of "is splat" etc to method somewhere, e.g. >> on SDNode. >> 3. Introduce a new