The Cisco Call Manager Express product can be used as the telephony SIP trunk gateway between the local IP telephony network and the IP Telephony Service Provider. Calls from and to PSTN will be handled by a SIP PROXY server located in the Service Provider network.
A sample Call Manager Express configuration for SIP trunking is shown below (a snippet of the complete configuration is shown):
voice service voip
allow-connections sip to sip
sip
registrar server expires max 3600 min 3600
localhost dns:mycompany.test.com
voice class codec 1
codec preference 1 g711ulaw
!— Inbound Translation Rule
!— for Auto Attendant pilot number “500″
voice translation-rule 1
rule 1 /5552222100/ /500/
voice translation-profile AutoAttendant
!— Applied to the inbound dial-peers for AA
translate called 1
!— SIP Trunk Configuration —
dial-peer voice 1 voip
description **Incoming Call from SIP Trunk**
translation-profile incoming AutoAttendant
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number .%
dtmf-relay rtp-nte
no vad
dial-peer voice 2 voip
description **Outgoing Call to SIP Trunk**
destination-pattern 9……….
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
no vad
dial-peer voice 3 voip
description **International Outgoing Call to SIP Trunk**
destination-pattern 9011T
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
no vad
!— SIP UA Configuration —
sip-ua
authentication username 5552222100 password 075A701E1D5E415447425B
no remote-party-id
retry invite 2
retry register 10
retry options 0
timers connect 100
registrar dns: mycompany.test.com expires 3600
sip-server dns: mycompany.test.com
host-registrar
!
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