OpenVox B200P User Manual page 27

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[from-isdn]
exten => s,1,Dial(SIP/100,,r)
exten => s,n,Hangup()
[from-internal]
exten => _X.,1,Dial(dahdi/1/${EXTEN},,r)
exten => _X.,n,Hangup()
exten => _0XX,1,Dial(dahdi/${EXTEN:1},,r)
exten => _0XX,n,Hangup()
After saving your dialplan, please run "asterisk –r", then execute
"reload" in the CLI. Now you are able to make calls. The above
dialplan achieves that:
 If there is a call from the ISDN line, Asterisk will transfer to SIP
extension 100 directly, then the SIP phone rings.
 If SIP extension 100 dials any phone number, Asterisk will transfer to
the destination phone through the first channel.
 If SIP extension 100 dials a channel number, Asterisk will transfer to
the destination ISDN phone which connects the channel. For example,
if the SIP phone dials 001, Asterisk will get through to the ISDN
phone which connects the first DAHDI channel.
When calls are running, you can check the status of Hardware Echo
Cancellation Module. In time slots which are active, EC module status is
ON, otherwise it's OFF. The status of EC looks like as follows:
OpenVox Communication Co. LTD.
B200P/BE200P on DAHDI User Manual
Figure 8 dial plan
26
URL: www.openvox.cn

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