The following is to help with the connection of Cisco CUBE or CallManager Express to our environment.

Dial peers will still need to be adjusted based on your own particular needs.

 

voice service voip
 ip address trusted list
 ipv4 203.30.19.164 255.255.255.255
 ipv4 203.32.124.160 255.255.255.224
 ipv4 202.74.176.160 255.255.255.240
 ipv4 202.74.176.176 255.255.255.248
 ipv4 203.7.224.128 255.255.255.240
 allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer

voice class codec 1
 codec preference 1 g711alaw
 codec preference 2 g711ulaw

! apply calling number or CallerID
voice translation-rule 1111
 rule 1 /.*/ /61894883344/

! remove 0 prefix for Outbound calls
voice translation-rule 1112
 rule 1 /^0/ //

voice translation-profile Remove_0Prefix_ApplyCLID
 translate calling 1111
 translate called 1112

sip
 registrar server expires max 3600 min 3600
 localhost dns:sip.simtex.com.au
 no update-callerid
 sip-profiles 1000

voice class sip-profiles 1000
 request ANY sdp-header Connection-Info remove
 response ANY sdp-header Connection-Info remove

sip-ua
 srv version 2
 credentials username 7xxxxxxxx password 0 xxxxxxxx realm sip.simtex.com.au
 keepalive target dns:sip.simtex.com.au
 authentication username 7xxxxxxxx password 0 xxxxxxxx 
 retry invite 2
 retry register 10
 timers connect 100
 timers keepalive active 100
 registrar dns:sip.simtex.com.au expires 1200
 sip-server dns:sip.simtex.com.au
 connection-reuse
 host-registrar
 g729-annexb override

dial-peer voice 1000 voip
 permission term
 description ** Incoming call from SIP trunk (Demonstration peer from Simtex) **
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target ras
 incoming called-number .%
 dtmf-relay rtp-nte
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad

dial-peer voice 2000 voip
 description ** Outgoing 8-digit Local with 0 prefix (Demonstration peer from Simtex) **
 translation-profile outgoing Remove_0Prefix_ApplyCLID
 destination-pattern 0[2-9].......
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad
dial-peer voice 2001 voip
 description ** Outgoing 10-digit Local with 0 prefix (Demonstration peer from Simtex) **
 translation-profile outgoing Remove_0Prefix_ApplyCLID
 destination-pattern 00[2-9]........
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad

dial-peer voice 2002 voip
 description ** Outgoing 13 Local with 0 prefix (Demonstration peer from Simtex) **
 translation-profile outgoing Remove_0Prefix_ApplyCLID
 destination-pattern 013[1-9]...
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad

dial-peer voice 2003 voip
 description ** Outgoing 13/18 Local with 0 prefix (Demonstration peer from Simtex) **
 translation-profile outgoing Remove_0Prefix_ApplyCLID
 destination-pattern 01[38]00......
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 ip qos dscp cs5 media
 ip qos dscp cs4 signaling
 no vad

 

 

If you have any particular questions in regards to the configuration please do not hesitate to get in contact with us on 1300 888 519 or pop us an email to [email protected]