similar to: DID Trunk

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

Seems strange to me that I should be having this problem but I'm stumped. I'm still playing with a Voicetronix OpenSwitch6 card and Asterisk (development CVS). I using a dialplan that's derived from one on another running installation that's using Digium FXS/FXO interfaces instead of the OpenSwitch card. It seems that as soon as a user on an FXS interface dials 7 digits, a call

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, 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>

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, 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

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

Anyone using the UIP200 with *? I am having difficulty getting the phone to register and *stay* registered for more than 4 seconds. The * console shows the UIP200 registering and records the user agent. The UIP200 displays station name and time on its LCD, then 4 seconds pass and it shows #1 DISCONNECTED. The * console reports no problem and believes the device to still be registered. A call

Hello We recently moved to Asterisk 1.2.4 (from 1.0.x) and our 10 Uniden UIP200 have stopped working ever since. We can make a call with the UIP200 to any other extensions, but it can not receive a call. In fact the UIP200 always appears offline: It does show up in asterisk a few seconds after the UIP200 reboot: -- Saved useragent "Uniden SIP Phone p2 Ver BS4.70" for peer uip200 but

I am trying to get a new Asterisk installation running using a Zultys ZIP2 phone and a Uniden 200 phone. I have the system working reasonably well (although probably not optimal) except for a couple of items. First, I can't get the voice mail message light to work on the Zultys phone but it works just fine on the Uniden phone. Second, the time presented on the Uniden UIP200 phone is 1 hour

I got a Uniden UIP200 and started to configure it and I am lost.... I have a tftp server setup on my * server and have the files unidencom.txt and uniden<mac>.txt there. But it doesn't quite work yet. It registers as a sip phone (sip show peers), but I cann't dial it and the display shows #1 disconnected all the time. It has firmware version BS4.59a in it. I have no idea if I

Hi, We've recently deployed 6 Uniden UIP200 phones (running firmware 4.54). We've been having some serious problems: 1) All the phones randomly reboot themselves. Typically when trying to answer or initiate a call. 2) All the phones will disconnect from a calls with the PSTN after 2-3 minutes. 3) The phones are unable to interact with a remote IVR (digit presses are not received at

Hi, all. I've got my Uniden UIP200 configured via TFTP (had to get DHCP 3.0.1 -- Debian's latest is 2.0.x!), and all seems well... except for the minor detail that it doesn't work. It registers fine with Asterisk, but when I copied my Grandstream's sip.conf info and plugged in the Uniden stuff, no dice. Any ideas? Thanks... -Ken unidencom.txt: OverwriteLocalSettings

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

Hi, all. Got my Uniden UIP200 today (ordered from thetwistergroup.com), and was very excited to set it up... until I came to the realization that there were no docs with it whatsoever. There was, however, a sheet of paper with the stock warnings (don't use the phone in the tub, etc.), AND a URL -- for documentation. Score! Well, no. The URL it gave me, bcs.uniden.com, does indeed have

Hello all, Uncertain if I have a configuration problem or a buggy phone, I'm requesting that Uniden uip200 users attempt the following and share their results: - dial into voicemailmain - press 2 to change folders - listen to each of the folder prompts. Is the audio for the folder number clipped at the beginning? This audio clipping problem will also appear in other applications, such as

hello, yesterday the uniden uip200 phone was recommended to someone. i am looking for an alternative to grandstream bt-100 because i can not do a supervised tranfer with it. here my questions: 1) does the uip200 support supervised transfers? 2) can i buy the phones in europe, especially in germany? thanks in advance, jan goericke

My users are reporting that some voicemail messages are being cut off in the middle of being played back. The recordings are OK (they play fine when forwarded to e-mail, and they can often be accessed OK during a later call to voicemail). I found nothing in the archives on this -- ideas anyone? RH9, P4, CVS-HEAD-09/02/04-08:44:34, aggressive echo suppression turned on. Jim Shilliday IT

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:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I normally don't write emails like this, but the last couple of days have been very frustrating. We built a system for a customer with the following hardware: Asus Vintage (sis chipset) P4 2.8 512M Ram LSI SATA Raid card (2x80g drives) Digium TDM400 we had originally purchased a tdm2400 with echo cancellation but couldn't fit it in the