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extensions.conf
[from-pbx] ;;; LLM added stanza for local SIP phone line
exten => <node#>,1,Answer
exten => <node#>,n,Playback(rpt/node)
exten => <node#>,n,Playback(digits/5)
exten => <node#>,n,Playback(digits/1)
exten => <node#>,n,Playback(digits/8)
exten => <node#>,n,Playback(digits/4)
exten => <node#>,n,Playback(digits/6)
exten => <node#>,n,Playback(rpt/connected)
exten => <node#>,n,rpt(<node#>|P|ad6g-PBX) ; P = Phone exten... *99 to key and # to unkey (but with full control)
;; Alternative examples:
;exten => <node#>,n,rpt(<node#>|S|ad6g-PBX) ; S = Simple exten... single * to key and # (or * again) to unkey (no other control)
;exten => <node#>,n,rpt(<node#>|Pv|ad6g-PBX) ; Pv = Phone VOX instead of DTMF PTT
iax.conf
; LLM incoming from PBX
[pbx-to-node]
type=user
secret=<password>
disallow=all
allow=ulaw
allow=g726aal2
allow=gsm
codecpriority=host
context=from-pbx
transfer=no
Based on:http://lists.hamvoip.org/pipermail/arm-allstar/2017-November/006520.html
[arm-allstar] reverse autopatch and local phone extension
“<node#>” is the local
node number
“ad6g-PBX” will show up as the
Node ID in connection tables.
username=pbx-to-node
is the outgoing peer
username on the pbx trunk
<password> should match
“secret=” on trunk as above
Asterisk PBX IAX trunk into Allstar node
PBX
Trunk: Disallow all asterisk trunk dialing options that enable in-call feature codes (like DTMF call transfer) as they will probably interfere with allstar commands.
Route: Disable MOH.