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]