similar to: DID failover

Dear all, I am still a R apprentice... Apologies for the basic question. I am trying to reshape a dataframe based on the order of two variables (a character variable and a numerical variable). To simplify it, consider the following dataframe > df<-data.frame(id=c("b","b","a","a","a"),ord=c(2,1,1,3,2)) id ord 1 b 2 2 b 1 3 a 1 4

I need clarification as to DID in T1 connection. T1 provides 24 channels for voice/data. Do it assign each channel to particular DID. Or you can have unlimited DID to share the 24 channel as an example. ie. Outgoing/incoming traffic is not bound to particular channel. Whatever is available will be used according to the grouping in zapata.conf. -- David Kwok, CISSP Tel: 612 82315701 ext 1002

Hi all, Occasionally, DIDs from different providers stop working for some reason. I would like to be able to monitor situations like that and react before any of my clients start going ballistic on me. Any ideas? Scripts you know of or wrote and willing to share? Any info?would be greatly appreciated. ? Robert ? -------------- next part -------------- An HTML attachment was scrubbed... URL:

It wouldn't take me more than an hour or two to do the format conversion. It is rather trivial, actually. Just say the word and I'm on to it like a starving bee. I guess I should get used to using the Sphinx layout; I'm simply more familiar with the format I use on my own websites. Have you guys ever considered making a less formal wiki for LLVM documentation - a place where tiny

Has anyone had any experience connecting Asterisk to Cbeyond's SIPconnect service (http://www.sipconnect.info)? Any opinions? Thanks, -Brian

Hello all, I first want to thank everyone for all your contributions. I've building an asterisk system for a month or so now and without everyone in the online asterisk community I wouldn't have made it this far yet. Thanks! ...ok, mushiness out of the way.. :) I am looking for a failover and ultimately a load balancing asterisk solution. I've done a good bit of research and I

Just turned up a new PRI with DID's in the US. I'm receiving 5 digits of the DID numbers as I requested. Assuming I have 100 DID numbers but only define 50 of those in extensions.conf, is there an easy way to send the incoming calls for the 20 undefined numbers to a common resource (ivr, operator, or canned message) without having to define each one?

I have a box running *@Home 1.0 and I noticed today when I went to change some settings to do with the DID's that it is no longer detecting the different lines. I have a Digium 4 port line card and Im pretty sure that the DID's used to work when I used fxs_ks signaling on them. However I changed to fxs_ls signaling because the ks wasn't detecting when people were hanging up properly.

I need 100 DID's in a specific rate center (916-854-xxxx). How do I go about finding who owns the rate center ? If the DID's are available in this rate center ? Thanks Vikas

Hi guys, I posted a somewhat similar question about a month ago and got a thoughtful resonse from Steven Critchfield, but I've got a quick follow up question to it. I'm looking to setup a 16 extension / 10-14 phone line Asterisk install for a customer who would like to have DID numbers for the extensions, since they're currently on Centrex and already have the 1-to-1

Hello, I'm looking for US DID and US50/CA 800# Providers. I found voiceconduits.com 8 month ago, there interface looks good, but there are still not live, I believe they won't be any time soon. I found sixtel, but order take eternities, they probably won't get my orders right any soon. So i'm looking for a good provider for DIDs and 800# from the US and CA, who offer online

Hello, our gateway receive incoming calls from an outside gateway for multiple DIDs. For some of them we want fax detection, for other no. To do so, faxdetect is set to yes, but how to disable the fax detection for a specific incoming DID? For those DIDs, we just want to forward the call to a real fax machine DID which will do the job. Thanks for any hint Regards -- Daniel

I know that Asterisk supports DID, but does anyone have documentation on how to write the configuration for it? I'll be trying to setup a hybrid system where some incoming numbers will be DID enabled and others won't, so I'll need to be able to sort between the two, i.e. directly connect the DID dialed numbers and route the others to an autoattendant for extension dialing.

In speaking to the representative at Verizon, we came to the conclusion that DID numbers were not the correct solution; however we were told by Verizon that they could do something called assign individual numbers to the PRI. What this would in effect do is give us an additional phone number that we would like to route to a specific extension; however unlike the DID number, it would not be

I am a Linux sysadmin who has been tasked with developing the phone system for our nonprofit's new US headquarters building. We cannot bring our legacy phone system with us, so I am building this completely from scratch. I have already read "Asterisk: The Future of Telephony" and done a fair amount of googling. I am completely sold on Asterisk, and the new building's

Anyone have experience with these guys? If so, good, bad, average? http://www.junctionnetworks.com Thanks, Wiley -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050523/4222e3e7/attachment.htm

Hi, I still have the same problem trying to configure ITSP failover in extensions.conf for a connected PRI. Any comments thoughts or direction would be greatly appreciated. I sympathize with wanting inbound DID failover. If we have a client with multiple DIDs we will spread them across two or three ITSPs so that all inbound connectivity will not be lost if one of them has an issue. I

On Sun, Nov 17, 2013 at 3:50 AM, Mikael Lyngvig <mikael at lyngvig.org> wrote: > It wouldn't take me more than an hour or two to do the format conversion. > It is rather trivial, actually. Just say the word and I'm on to it like a > starving bee. I guess I should get used to using the Sphinx layout; I'm > simply more familiar with the format I use on my own

On Sun, Nov 17, 2013 at 5:17 AM, Sean Silva <chisophugis at gmail.com> wrote: > > > > On Sun, Nov 17, 2013 at 3:50 AM, Mikael Lyngvig <mikael at lyngvig.org>wrote: > >> It wouldn't take me more than an hour or two to do the format conversion. >> It is rather trivial, actually. Just say the word and I'm on to it like a >> starving bee. I

Went to their site today. Site claims they are still in biz. What is the story? What really happened to Nufone anyway?