[Asterisk-Users] Voicetronix Openswitch

[tim]

  Hi.

  We are trying to use an Openswitch12 with all channels set to fxs, and 
have these problems.

1) When calling from sip phone to analouge phone, all works well.
2) When calling from analouge phone to sip phone, the analouge phone 
hears all well, but the sip phone hears nothing, according to Asterisk, 
the call is up and bridged.
3) When calling from an analouge phone to analouge phone, all works 
well, except that it is impossible to terminate (hangup) the call.

vpb.conf:
[general]
cards = 1
type = v12pci
[interfaces]
board = 1
;echocancel = on
rxhwgain = 10
txhwgain = 10
txgain = 10
rxgain = 10
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


extension.conf:
[vpb-fxs]
exten => s,1,Wait
exten => s,2,Answer
exten => s,3,Hangup
exten => 1,1,Wait
exten => 1,2,Setvar(VPBID=VPB:${CHANNEL:6})
exten => 1,3,SetCIDName(${VPBID})
exten => 1,4,Dial(SIP/116,30,t)
exten => 1,5,Hangup
exten => 2,1,Wait,2
exten => 2,2,Dial(vpb/1-3/,30,tH)
exten => t,1,Playback(vm-isunavail)
exten => t,2,Hangup

[from-sip]
exten => s,1,Answer
exten => s,2,Hangup
exten => _2.,1,Dial(SIP/${EXTEN:1},20,t)
exten => _2.,2,Playback(pbx-invalid)
exten => t,1,Playback(vm-isunavail)
exten => t,2,Hangup
exten => _4X,1,Dial(vpb/1-${EXTEN:1}/,30,t)
exten => _4X,2,Playback(pbx-invalid)