similar to: Line sharing

Hello all, I hope this is not off-topic, if it is please let me know. I'm currently playing with an Asterisk at home, in order to get to know it's ins and outs. Very very impressive indeed. I've got it hooked up to my home phone line via a Wildcard clone board (Intel modem with Ambient chipset), and it works like a charm. Zaptel picks up the card as a generic clone, and works with it

One of the suggestions I got for dealing with my analog FXO woes was to look into using a Voicetronix card instead of the Digium. They're more expensive but if it worked better it may be reasonable. They indicated they were compatable with the CVS HEAD version of *. Can someone share with me their experiences with Voicetronix OpenLine and/or OpenSwitch cards with *? I'm specifically

Ron, VoiceTronix worked with me last week and backported the new chan_vpb that's in the current HEAD CVS tree to the 1.0 stable branch. So if you are using 0.7.2 or 0.9.0 you can go to the VoiceTronix website now (note that this is very new) and download an Asterisk package. When you download it there is an new chan_vpb.c that needs to be put in your asterisk source code tree. Also there

Dear all, I need to generate a list of lists as required by the bayesm-package. This means in my application that I have to generate a list which consists of 2000 elements, which are lists themselves: list(list(y1,X1),...,list(y2000,X2000)). The y are vectors and the X are matrices of different dimensions. I tried to solve this problem iteratively by the following code, but received an error

Hi All Can anybody help me for dial plan to barge or Spy(ExtenSpy) specificor selective extemsions among 20 extension in my office. lets say my office extension range is 301-320 & i want to barge only 3 extension say 320, 302,314. is this possible to barge specific extension? . Plz help me for this.I am using Asterisk 1.4.9 & SIP channels. Regards Amit -------------- next

Skipped content of type multipart/alternative-------------- next part -------------- [root@LINUXVM root]# asterisk -vvvc == Parsing '/etc/asterisk/asterisk.conf': Found Asterisk 0.4.0, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer <markster@linux-support.net> ========================================================================= == Parsing

Hello all - I'm trying to get Asterisk working with an Openswitch/6 card with all ports set to FXO for PSTN in/out, and 10 SIP phones on the LAN with the Asterisk server. The SIP phones are working properly to dial one another, but I can't seem to get a call to go out to the PSTN through the voicetronix, and I similarly can't get incoming calls to the lines hooked up to the

HI folks, I have set a barging facility with our production box.Client able to barge a agent but client raise a requirement, they want talk to barge agent but that communication is not listen by customer. It is possible with asterisk or not. thanks in advance. Regards Akhilesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, OS OpenSuSE 10.3 * ver 1.4.26.2 zaptel ver. 1.12 Digium TE122 I have a request for remote users to be able to dial through the system so that the sales managers can barge/chanspy on the sales force. I have the DISA part working with authentication(rather straight forward) but what I can not figure out is how to enable the supervisors to be able to barge on these calls. Is there a

[Apologies in advance for the train of thought prose, but it is brainstorming after all…] I'm going to focus on self-descriptive names rather than literary or fantasy references… Advanced Compiler Kit, affectionately known as “ACK!”? It has an ill- deserved nod to NeXT, even. (Completely the wrong language, after all.) Core Compiler? Heh, I don't think that'd get past certain

Hi, Firstly it appears you are not passing the extension to be dialed to the Dial command, so the driver is dialing badly. It should be something like dial(vpb/1-2/102) since this is a line port. Admittedly the driver needs to check better that the number it is trying to dial is legit but it doesn't :) P. --- You Wrote --- From: "Jacky" <jacky.taiwan@msa.hinet.net>

Hi, I am using a voicetronix OpenLine4. I downloaded a recent asterisk CVS from voicetronix webpage but have had no luck to reduce echo on outgoing calls and for it not to crach on incoming calls. I dont think both problems are related though. Here is an output of what happens when a new call comes in and my voicetronix tries to pick it up and crashes asterisk: > vpb/1-1: Event

Hi, I''ve created snapper packages for Ubuntu, available on https://launchpad.net/~snapper/+archive/stable. For those new to snapper, it''s a tool for managing btrfs snapshots (http://en.opensuse.org/Portal:Snapper). It depends on libblocxx available from https://launchpad.net/~bjoern-esser-n/+archive/blocxx , and currently uses git source up to commit 50dec40. I''ve done

I've seen a few but can't get them to work. I need one where I can drop a call into a conference without them knowing it to us it as a live monitor and barge function, anyone doing this are know of a gui client for windows I can use? Thanks,

Hi we've got Asterisk CVS-HEAD 18-Aug-04 (modified by Voicetronix as available on their site for use with the vpb driver) and an OpenLine4 (4xFXO). The same server also has two X100P. Calls on the Voicetronix card drop instantly when the called party picks up. The vpb driver reports that it detected a hangup (loop drop) yet there is no hangup when connecting the X100Ps or analog phones to

I'm working on putting together some Ideas about using Asterisk in our environment, one of the things I want to consider is DID trunks (analog), what hardware do I need to terminate these trunks? I'm looking at the voicetronix openswitch6 or openswitch12. On the openswitch, I'd like to use some of the lines for analog sets for the breakroom, kitchen, etc where they don't need all

Hi all, Trying to get fax reception going using a voicetronix openline4 card, however, there are two issues (as far as I can see) (1) Currently the voicetronix world (cards, firmware, drivers, channel prog etc.) does not do fax tone detection. I have spoken to the voicetronix guys , understand what has to happen there and may do some hacking if I can resolve the second issue. Meanwhile, I

Hi All I'm using dovecot 2.0.16 with the pigeonhole plugin 0.25. I'm trying to use the notifiy mechanism from sieve to send notifications when a mail arrives in the mailbox. The message is checked to be a 8bit message, otherwise it is replaced by the default message "Notification of new message." How can I create a 8bit message body within the sieve script that is accepted

Hellow, i need the following requirements with asterisk : 1) Can ACD (Automatic Call Distribution) service work with asterisk, and how to set up ACD in asterisk ? 2) How call barging can set up in asterisk ? 3) How call recording can set up in asterisk? Thanks mahboob System Engr SSL -------------- next part -------------- An HTML attachment was scrubbed... URL: