[Asterisk-Users] Voicetronix

tim tim at cns-online.net
Tue Jul 20 06:07:12 MST 2004


 Hi.

 With voicetronix Openswitch12, I have installed the latest drivers, 
Asterisk 1.0-RC1 and so far so good. I want the simplest of all cases to 
begin, all 12 channels FXS. But, it doesn't work.
 When I pick up the phone, and dial "1", which in my extensions.conf 
should make my sip phone ring, asterisk doesn't register that I've 
pushed the "1" on the analoge phone. But Asterisk has registered that 
I've picked up, and is sending the dialtone. If I try to ring to it from 
my sip phone, the analouge phone rings for an instance then a hangup is 
done.
 If I stop Asterisk, rmmod vpbhp, and insmod vpbhp, and try again, 
sometimes it works, and I can ring, but if I make the call from analouge 
to sip, then I can hear nothing in the sip phone, in the analouge 
perfect. If I call from the sip phone, bouth parties hear perfect.
 Even worse if I make a call from analouge to analouge, I hear perfect 
in bouth phones. But when I hang up, it is not registered, and there is 
a bridged call left in Asterisk and only way to get rid of it is to 
close Asterisk.

 My conf files are below:
extensions.conf
[vpb-fxs]
exten => s,1,Wait,4
exten => s,2,Answer
exten => s,3,Hangup
; call to sip, dial 1
exten => 1,1,Wait,2
exten => 1,2,Dial(SIP/116,30,t)
; to make call analouge to analouge (line 3) dial 2
exten => 2,1,Wait,2
exten => 2,2,Dial(vpb/1-3/,30,t)
[from-sip]
exten => _41,1,Dial(vpb/1-1/,30,t)
exten => _42,1,Dial(vpb/1-2/,30,t)
exten => _43,1,Dial(vpb/1-3/,30,t)

vpb.conf
[general]
cards = 1
type = v12pci
[interfaces]
board = 1
context = vpb-fxs
mode = dialtone
channel = 1
channel = 2
channel = 3
channel = 4
channel = 5
channel = 6
channel = 7
channel = 8
channel = 9
channel = 10
channel = 11
channel = 12




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