[Asterisk-Users] Voicetronix Openswitch 12
Jeff Gustafson
ncjeffgus at zimage.com
Wed Mar 17 19:21:02 MST 2004
Voicetronix is actively working to improve the vpb driver in Asterisk.
Right now the code they sent me is for the Asterisk CVS code. Right now
the Asterisk CVS code seems to be totally hosed for incoming audio over
an analog line even with a x101p. I'm hoping that things will clean up
soon so I can try the new driver.
...Jeff
On Wed, 2004-03-17 at 15:31, tim_mickelson wrote:
> Hello.
>
> I have a client that has installed Asterisk and an Openswitch 12,
> all seems
> ok, but we can not make any calls. When making a call, the output
> looks ok, but,
> the phone we're calling is not ringing, even thought the asterisk says
> that it
> is ringing, please could we have some help?
>
> (note that channel 9 is the channel connected to the PSTN)
>
> below is the output from Asterisk and below that output are the
> extensions.conf
> and vpb.conf
>
> Read_channel ## vpb/1-9: Setting record mode, bridge = 0
> -- 1-9 requested, got: [vpb/1-9]
> -- Calling 1-9/3487446196 on vpb/1-9
> -- VPB Calling 1-9/3487446196 [t=0] on vpb/1-9 returned 0
> -- Called 1-9/3487446196
> -- vpb/1-9 is ringing
> -- hangup on vpb (vpb/1-9)
> -- Hungup on vpb/1-9 complete
> == Spawn extension (default, 3487446196, 1) exited non-zero on
> 'SIP/110-db94'
>
> the two hangup lines, are after I hangup, on line says ringing, but
> only my
> sip-phone is ringing!!
>
> extensions.conf
> exten => _XXXXXXXXXX,1,Dial(vpb/1-9/3487446196,30,r)
>
> vpb.conf
> [interfaces]
> echocancel = on
> board = 1
> context = default
>
> mode = fxo
>
> channel = 9
> channel = 10
> channel = 11
> channel = 12
>
> mode dialtone
>
> channel = 1
> channel = 2
> channel = 3
> channel = 4
> channel = 5
> channel = 6
> channel = 7
> channel = 8
>
>
>
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