!
voice service voip
 ip address trusted list
  ipv4 10.0.0.0 255.0.0.0
  ipv4 12.194.41.30
  ipv4 12.194.107.150
  ipv4 12.253.1.102
  ipv4 12.194.107.142
ipv4 10.46.52.1
  ipv4 10.46.52.65
  ipv4 12.253.102.70
  ipv4 12.253.102.182
  ipv4 12.253.102.73
  ipv4 12.253.102.185
  ipv4 12.253.102.86
  ipv4 12.253.103.134
 rtp-port range 16384 32766
 address-hiding
 mode border-element 
 media bulk-stats
 media disable-detailed-stats
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 no supplementary-service sip handle-replaces
 redirect ip2ip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 modem relay nse codec g711ulaw gw-controlled
 h323
  h225 timeout t302 15
no h225 timeout keepalive
  h225 timeout ntf 200
  h225 signal overlap
  h225 display-ie ccm-compatible
  no call service stop
 sip
  bind control source-interface GigabitEthernet0/0/0
  bind media source-interface GigabitEthernet0/0/0
  header-passing
  error-passthru
  asserted-id pai
  no update-callerid
  midcall-signaling passthru
  privacy-policy passthru
!
voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8 bytes 30
!
voice class codec 2
 codec preference 1 g729r8 bytes 30
 codec preference 2 g711ulaw
!
voice class codec 3
 codec preference 1 g711ulaw
!
voice class h323 1
  h225 timeout tcp establish 2
  h225 timeout setup 2
  call preserve 
!
!
voice class sip-profiles 1
 response ANY sip-header Allow-Header modify "UPDATE," "" 
 request INVITE sip-header Diversion modify "<sip:2(....)@" "<sip:845892\1@" 
 request INVITE sip-header Diversion modify "<sip:4(....)@" "<sip:845894\1@" 
 request INVITE sdp-header Audio-Attribute modify "a=ptime:20" "a=ptime:30" 
 response ANY sdp-header Audio-Attribute modify "a=ptime:20" "a=ptime:30" 
 request INVITE sdp-header Audio-Attribute add "a=ptime:30" 
!
!
voice class e164-pattern-map 6001
 url flash:dp1.txt
!
voice class e164-pattern-map 6002
 url flash:dp1.txt

!
voice iec syslog
!
!
voice translation-rule 1
 rule 1 /^911/ /911/
 rule 2 /^9911/ /911/
 rule 3 /^9/ //
!
voice translation-rule 2
 rule 1 /^[2]\(....\)/ /845892\1/
 rule 2 /^[4]\(....\)/ /845894\1/
!
voice translation-rule 3
 rule 1 /^84589/ //
 rule 2 /^84559/ //
!
voice translation-rule 4
 rule 1 /8455926506/ /26506/
!
voice translation-rule 9
 rule 1 /^[2]\(....\)/ /845592\1/

!
voice translation-profile onsemi-pstn-in
 translate called 4
!
voice translation-profile onsemi-pstn-out
 translate calling 9
 translate called 1
!
voice translation-profile pstn-in
 translate called 3
!
voice translation-profile pstn-out
 translate calling 2
 translate called 1
!
!

dspfarm profile 1 transcode  
 codec g729abr8
 codec g729ar8
 codec g711alaw
 codec g711ulaw
 codec g729r8
 codec g729br8
 maximum sessions 8
 associate application CUBE
!
dspfarm profile 2 mtp  
 codec g711ulaw
 maximum sessions hardware 8
 associate application CUBE
!
dial-peer voice 200 voip
 description Outgoing to AT&T -VoIP and PBX Call Leg
 translation-profile outgoing pstn-out
 no modem passthrough
 incoming called-number .
 voice-class codec 1  
  voice-class sip asymmetric payload full
voice-class sip asserted-id pai
 voice-class sip privacy-policy passthru
 voice-class sip profiles 1
 voice-class sip bind control source-interface GigabitEthernet0/0/1
 voice-class sip bind media source-interface GigabitEthernet0/0/1
 fax-relay ecm disable
 fax-relay sg3-to-g3
 fax rate disable
 fax nsf 000000
 fax protocol pass-through g711ulaw
 ip qos dscp cs3 signaling
 no vad

!
dial-peer voice 6000000 voip
 translation-profile outgoing onsemi-pstn-in
 destination-pattern 8455926506
session protocol sipv2
 session target ipv4:10.46.52.65
 voice-class codec 1  
 dtmf-relay rtp-nte
 ip qos dscp cs3 signaling
 no vad
!