similar to: Integrated management tool?

-----Original Message----- From: Chad Brown Sent: Tuesday, December 21, 2004 8:02 PM To: 'el_flynn@lanvik-icu.com' Subject: RE: [Asterisk-Users] Dialplan help - Can dial any user but not thePSTN Flynn, Yes, that makes sense. However, in my case I have incoming calls arriving on an IAX channel from a PSTN gateway. I think the concept is the same. That said, if incoming calls have access

Hi all, Just wondering if anyone's come across this issue, and what might be a fix for it: We've got several BT-101's deployed, and upgraded to firmware v.1.0.5.16. The phone can do proper supervised transfer, but _only_ once. If the user attempts to transfer a second time, it won't work. any suggestions/hints/tips are welcome.. Flynn

Hi all, I've got some questions to post in regard to running asterisk in a production-grade environment, specifically targeting high-density IVR applications. No VoIP involved, just straight PSTN -> * and perhaps the occasional outdials or agent-based predictive dialing. 1) Which user would you run * under? 2) What other security-related issues do you have to resolve? 3) How do you handle

I just started poking around with writing a python module to interface to the Manager API, and it suddenly hit me... how the heck are you supposed to program this thing? All the events seem to be dumped to all the open connections. If I send a command, such as a login, there seems to me to be no way to determine which response are intended for me, and which may be intended for another open

hi there, i've been able to successfully run asterisk with the Voicetronix OpenLine4 card, it can accept calls and function normally. The only problem I'm experiencing so far is getting the card to outdial to a third party. What I'm trying to achieve is basically call bridging, where the caller dials in to asterisk, some IVR plays and then attempts to perform a "transfer"

I am currently developing a asterisk ncurses interface using the manager API. The project is currently awaiting sourceforge's approval but I have a beta online at http://sig.lange.googlepages.com/assman . The projects real home will be assman.sf.net. This project really consists of two parts, libassman is a C manager API and assman is the ncurses portion. It's still beta but I have been

Dear Faiz, Bootstrapping R^2 using Boot() is straightforward: Simply write a function that returns R^2, possibly in a vector with the regression coefficients, and use it as the f argument to Boot(). That will get you, e.g., bootstrapped confidence intervals for R^2. (Why you want that is another question.) See the example in ?Boot that shows how to bootstrap the estimated error variance (without

Switchboard is a simple, event-observing framework for ActiveRecord. It''s designed to make it easy to add observers for all models in your app, and to easily turn them on and off selectively. Intallation gem sources -a http://gems.github.com sudo gem install zilkey-switchboard Usage First, require switchboard above your rails initializer: # environment.rb require

Hi all, I used to have an OpenLine4 card, but decided against using it due to some problems with hangup detect. Does anyone on the list actively use Voicetronix's OpenSwitch12? What are your opinions on the card? Cheers, Flynn

Hi all, I'm just wondering about these VoIP services -- do you have to sign up one account -per- client that will be using the service? I've got multiple extensions behind my Asterisk box, and I want to be able to allow all my staff to place calls via the provider. So if I sign up for one account, will multiple users behind my Asterisk box be able to make calls, using that same

I've been trying to setup AddQueueMember() as a replacement for the deprecated AgentCallbackLogin(), but I get _tree_ Queue()'s. Massaged extensions.conf (can provide the original if need be): ----- s n i p ----- [default] include => agent-loginout include => local ; ---------- [agent-loginout] exten => _100.,n,Macro(queue-addremove,I${EXTEN:3},dispatch,10)

hi all, i've got a proposed setup that i was wondering if you guys could comment on. the client wants * and a couple of SIP phones to be on a separate network than the rest of the office, so that in case their primary network crashes for some reason the PBX won't be affected. one other factor: the client may at some later point set up SIP UAs sitting on the primary network that will

Hi there, I've got an installation where there's 12 POTS line incoming into *, and am trying to get some insight as to which VoIP hard phone would be most suitable for this scenario. Most of the VoIP phones I've looked at only have 4-6 line presentations; is anyone aware of one that has more? I tried to get some info about Snom's Keypad 220 since it has loads of programmable

Residuals are stored as a numeric vector. The R software comes with a document "Introduction to R" that discusses basic math functions and logical operators that can create logical vectors: abs( stdresiduals ) > 2.5 It also discusses indexing using logical vectors: stdresiduals[ abs( stdresiduals ) > 2.5 ] Note that in most cases it is worth going the extra step of making your

Hi there, I'm looking into developing an in-house switchboard application. Does anyone here know of a way to control a hard-phone from such an application. For example, the attendant forwards a call with another one in queue. Once the first call has been forwarded (by keyboard shortcuts or dragging-n-dropping) - she presses a button (on the computer) to answer the waiting call. Now, if the

Hi, I'm running two boxes side by side, identical specs and setup but with differing dialplans. Both are on ast/zap/libpri versions 1.0.9. Both boxes share the same folder for voicemail, exported via NFS from another file server. Everything was working fine for an extended period of time, until just recently when someone rebooted Box A. Now when I dial an extension associated with a SIP

Hello, I am very new to asterisk and voip in general and so far have managed to get the FXO card and a few sip phones working fine. My question is where does the Sipura SPA 2000 come in the picture? Can it be used as an extension (i.e FXS) ? Or is it to be used as a line (i.e FXO)? Or it can be used as both? My understanding is that its just like another ATA186. Is that true? I guess what I

Hello, we are looking for someone to write a simple switchboard application for asterisk which can park, place on hold, and transfer to extensions. Standard switchboard/operator console features but we also need the following ability. We need to have a screen/panel which monitor the operators/main extension and displays a DID's owner. For instance, let's say we forward all of our

Hi all , I'm from Germany and need your help. I use a telephone switchboard from the German Telekom. The Software is only for Windows but I like to use it with Ubuntu 10.04 and wine. With Windows you have to install the Software this way. 1. install the capi-driver ( is for the switchboard) 2. install the Software( to handle the switchboard) 3. unplug the usb-cable and restart the Computer

I can't check the voicemail for the switchboard. Asterisk hangs up for some unknown reason... ----- s n i p ----- -- Executing [*500 at default:1] Wait("SIP/597-00f0c410", "1") in new stack -- Executing [*500 at default:2] VMAuthenticate("SIP/597-00f0c410", "500 at default|s") in new stack -- <SIP/597-00f0c410> Playing