ATCOM AX-2G4A Product Manual page 17

[from-pstn]
exten=>s,1,Answer()
exten=>s,2,Dial(sip/500)
exten=>s,3,hangup()
9. Use a registered sip phone to make an outgoing call, according to the dial-plan above, if you
want to make the outgoing call through the first channel, you have to add a prefix "1" when
you dial out, if you want to make the outgoing call through the second channel, you have to
add a prefix "2". Here, the destination number is 15128731752, so dialing 115128731752,
then you can get the following output from asterisk CLI:
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*CLI> == Using SIP RTP CoS mark 5
--Executing [115128731752@from-internal:1] Dial("SIP/500-00000003",
"dahdi/1/15128731752") in new stack
-- Called 1/15128731752
-- DAHDI/1-1 answered SIP/500-00000003
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The called party rings, and pick up the phone, so the call is successful.
10. Please use a phone to make an incoming call, here, the SIM card number for the first channel
is 15889587467, using a cell phone to dial 15889587467, the sip 500 rings, and you can get
the following output from asterisk CLI:
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*CLI> -- Starting simple switch on 'DAHDI/1-1'
[Jul 1 11:42:53] NOTICE[20678]: chan_dahdi.c:8734 ss_thread: Got event 18 (Ring Begin)...
-- Executing [s@from-pstn:1] Answer("DAHDI/1-1", "") in new stack
-- Executing [s@from-pstn:2] Dial("DAHDI/1-1", "sip/500") in new stack
== Using SIP RTP CoS mark 5
-- Called 500
-- SIP/500-00000004 is ringing
-- SIP/500-00000004 answered DAHDI/1-1
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11. Plugging PSTN line into the FXO ports, customer can dial-out via FXO ports. Connecting
analog phone to FXS ports, customer can use analog phone to dial-out via FXO ports or via
GSM channels.
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