similar to: Voicetronix OpenSwitch6 with 10-digit Dialing

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

In my search for a better analog FXO interface, I'm looking into the Voicetronix cards. Can someone let me know if they've been using the OpenSwitch cards with Asterisk? Also, can anybody comment on the differences in FXO performance between the OpenCall and OpenSwitch cards? Thanks in advance, Paul -- Paul A. Dugas Dugas Enterprises, LLC email:

While setting my first couple IP phones, I set their voicemail buttons to an extension that runs VoicemailMain. exten => 8500,1,Wait(1) ; voicemail exten => 8500,2,VoicemailMain ; exten => 8500,3,Hangup ; I would like to be able to pass the mailbox number allowing each phone to go in directly but I'd rather tno have

New to asterisk so please be gentle. I'm guessing I'm among a number of recent additions to the list after the article in Linux Mag. I gotta say I'm *very* intersted in the project and will be doing lots of reading shortly. A couble quick questions first... How suitable is Asterisk for use as an IVR providing callers with textual data out of a database? Can it be combined easily

Been tinkering and found Zapateller appears to be answering when I didn't expect it to. I have something like so: [incoming] exten => s,1,NoOp ;Zapateller(answer|nocallerid) exten => s,2,PrivacyManager exten => s,3,Dial(${RING},20) ... I have a 1x1 analog * installation with a couple IP phones too. I've got the FXO interface connected to the home phone line. When I get an

In case it helps others, a couple gotchas that I had to dig and beat my head against to get past... - building on FC2, use "make linux26" when building asterisk - also for FC2, a couple links in /usr/src named linux and linux-2.6 and pointing to /lib/modules/2.6.8-1.521/build/ were needed to build - yes, the power connector on the TDM400P needs to be connected to the power

Looking to outfit a 8-person office shortly and am looking for IP phone recomendations. Saw recent threads and the wiki with all kinds of suggestions and notes on configs but little on the "feel" and sound quality. I have a PolyCom SoundPointPro analog desk/speaker phone and the Cisco 7940 here. How to the other business-class phones compare, from a user's point of view, to these?

I've got a single IAXy installed in a little office nearby and got a call from someone on site a finew mintues ago. Apparently they couldn't make a call on that extension. They'd pick up the phone and get nothing; no dial-tone. Running "iax2 show peers" showed the unit still registered and I could ping it too. Didn't try to call it while it was doing this; sorry. The

Hi all, Happy Wednesday! Got a 7940G off eBay to tinker with and, per the wiki, contacted Cisco about a service contract. Being just little ol' me, they won't deal directly and blew me off (I hate dealing with Cisco, Nortel, etc as a small-business person...) Again per the wiki, I hit CDW.com and found CON-SNT-CP7960 for $9.14. Looking for support for the 7940 (not 7960), I just

I've been struggling with a test * install for a couple months now in a small office and am just about ready to give up on it. It's not that the system itself is a problem. I've got everything (attendant, voicemail, FXS extensions, Cisco and Polycom hard-IP phones, and 2 VOIP carriers) working except for the frigging analog FXO interfaces. These things are driving me completely mad.

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

Hello, I would like to know if voicetronix card (specially openswitch6 and 12) can be used with asterisk. Is there any driver for this card? Best regards, Daniel -- Daniel ANDRE (mailto:dandre@iris-tech.fr) IRIS Technologies - http://www.iris-tech.com Serveur kwartz - http://www.kwartz.com

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

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

Hello, Hacker I install VoiceTronix OpenSwitch 12 port PCI Telephone Card, and setting vpb.conf, extensions.conf following My problem is: When i dial to fxo(channel 9-12), it is ok, but when i continue press exten 102, the channel crach with error messages following exception caught: VPBAPI_DIAL_INVALID_LEVEL, file: vpbdial.cpp line:872 Do i ignore some setting for VoiceTronix OpenSwitch12

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"

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>

Good people, Please find attached a first shot at a driver for the Voicetronix OpenSwitch hardware. It does work (ish). After paying $900-odd for it I figured I might try to make it work with Asterisk (could never get enough documentation on Bayonne!) I have problems with the audio quality, and since am new to Asterisk, I could really do with some input. First is that because the

dear Hackers, i have a voicetronix Openswitch card, and i have been finding it very dificult to get it to work with asterisk. i intend to connect 8 ports to the PSTN and 4 as station ports. problem 1: On running asterisk all i get at first i get : event[9=>[11] station OFF hook] on vpb/1-12 even [12=>[11] loop drop on vpb/1-12 event [12=>[11] Tone detect:GRUNT event [2=>[11] Dial

Am I doing something wrong here? GOt a SPC-841 the other day and have it registering properly. Can place and recive calls as expected but when on the phone, a second call is immediately dumped to "busy" voicemail. Does this thing not support call-waiting? Or, have I just got my configs wrong? Paul -- Paul A. Dugas Dugas Enterprises, LLC paul@dugasenterprises.com