similar to: Loop Drop on vpb/1-7

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

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

When dialling in and dialling my extension, when answered I get " Read_channel ## vpb/1-3: Setting record mode, bridge = 0 WARNING[20499]: File chan_sip.c, Line 1111 (sip_write): Asked to transmit frame type 8, while native formats is 4 (read/write = 4/4) == Spawn extension (default, 1004, 1) exited non-zero on 'vpb/1-3' -- hangup on vpb (vpb/1-3) -- Hungup on vpb/1-3 complete --

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

So I've got a Voicetronix card and it looks like the kernel driver works. Other than the 0's for ID info. vpb: Driver Version = 4.0 vpb: major = 251 vpb: tmp [0xfc8fec00] dev->res3 [0xfc8fec00] vpb: tmp [0xfc8c0000] dev->res2 [0xfc8c0000] vpb: 1WS Write cycle vpb: Manufactured 00/00/0000 vpb: Card version 00.00 vpb: Serial number 00000000 vpb: Setting up udev... vpb:1 V4PCI's

Hi, I hope some can help me out, as I have a problem with the vpb driver in *, and I am hoping someone has hit these issues and can offer some advice. First off, I'm connected to a British Telecom SystemX exhange with the local Tel Co. I can ring out fine, but when the callee on POTS answers, the vpb driver detects a loop drop and immediantly disconnects the call. I have currently set

HI All I've tried to do Call pickup /pickup group using VPB channel. I configure the same way as in zapata.conf ( callgroup=1 pickupgroup=1). The result is, the phone can only be picked up by other SIP phone under the same group. If we called that SIP phone and tried to pick up using VPB channel, it cannot pick up the call. Is anyone has the solution? Thanks.

I have a Cisco 7940 when you call in from outside and dial the Cisco phone extension I get this Read_channel ## vpb/1-3: Setting record mode, bridge = 0 WARNING[18451]: File chan_sip.c, Line 1111 (sip_write): Asked to transmit frame type 8, while native formats is 4 (read/write = 4/4) == Spawn extension (default, 1004, 1) exited non-zero on 'vpb/1-3' -- hangup on vpb (vpb/1-3)

Hello, list, I am using Asterisk with an OpenLine4 card. It worked well with Asterisk 1.0. Then I upgraded the system, Asterisk 1.4 had some problem to compile chan_vpb, but I managed to compile it manually. Still Asterisk does not work because it refused to load chan_vpb module. I had to revert back to install Asterisk 1.2. Then I get a error: asterisk: hip.cpp:793:

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

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 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

Hi there, I am trying to delay sending out DTMF from Voicetronix OpenLine4 to the CO line. The reason being is that Voicetronix sends out the DTMF too fast even before the line is fully established with the carrier. Usually when dialing an 8 digit number, only 7 digits are actually successfully heard by the carrier. Currently, my dial plan is: exten => _9.,1,Dial(vpb/1-1/${EXTEN:1}) Daniel

Good day all We have a voicetronix openline4 card If someone calls in from the outside the pstn and into the system and hangsup asterisk does not deteck the hangup any Idea why please Help Altus

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

Greetings, I would like to add 3D arrows (i.e. arrow-headed vectors linking X1Y1Z1 to X2,Y2,Z2) to a 3D plot; ideally the sort of plot that can be rotated interactively. Is this possible using plot3d, or another 3d plotter in R? While it is easy to draw segments in plot3d (e.g. below), I haven't figured out how to add arrow heads, or to create 3d arrows from scratch. ##two headless

So, given what I've read in the documentation I would create a text file named document_database.txt that might have the following: remote 192.168.1.10:30000 chert /var/lib/xapian_database/segment1 remote 192.168.1.10:30000 chert /var/lib/xapian_database/segment2 remote 192.168.1.10:30000 chert /var/lib/xapian_database/segment3 etc. I would then in my PHP program open

I am a serious Asterisk newbie: just installed asterisk last week and it is now running with our Voicetronix OpenLine4 hardware. All is working as expected with one exception, in the following sequence (extracted from my extensions.conf file): [GetConfirmation] exten => s,n,SetVar(TimeOut=0) ; if timeout and TimeOut=1 then user already timed out once, so hangup exten =>

Hi, I am fairly new to asterisk. I am currently testing my first setup. I've been able to debug most of the problems to make asterisk work with my hardware setup until this time. Currently I have the following issue: Voicemail is running but when I test to leave a voicemail thru my incoming PSTN channel (voicetronix / vpb), asterisk will not detect sound (according to the log) on that

Hello all, We had an experimental system which works on OpenLine4 telephony card and Asterisk 1.0.9. Customer asked to upgrade Asterisk to 1.4, then we found our problem: At first Asterisk 1.4 does not compile chan_vpb.so. The problem is it tries to compile chan_vpb.cpp to chan_vpb.o and chan_vpb.oo, then try to link them together. I manually compiled it, then make went smoothly. But it