pattern 9T (13) voice-class codec 1 voice-class sip early-offer forced (14) session protocol sipv2 session target sip-server dtmf-relay rtp-nte - (15) no vad dial-peer voice 101 voip description INBOUND G729 Voice SIP calls from VzB voice-class codec 1 session protocol sipv2 session target sip-server incoming called-number 1 dtmf-relay rtp-nte no vad dial-peer voice 200 voip description inbound FAX dial peer from VZ session protocol sipv2 session target sip-server incoming called-number 1018 (17) codec g711ulaw (18) fax rate disable (19) no vad continues 200 SIP Trunking Example 10-2 continued dial-peer voice 201 voip description outbound FAX calls to VZ translation-profile outgoing DIGITSTRIP-8 (20) destination-pattern 8T (21) voice-class sip early-offer forced session protocol sipv2 session target sip-server codec g711ulaw fax rate disable no vad dial-peer voice 300 voip description To/From CUCM subscriber for FAX Calls preference 2 destination-pattern 1018 session protocol sipv2 session target ipv4:192.168.0.4 incoming called-number 8T (22) dial-peer voice 301 voip description To/From CUCM publisher for FAX Calls preference 5 destination-pattern 1018 session protocol sipv2 session target ipv4:192.168.0.6 (23) incoming called-number 8T codec g711ulaw fax rate disable no vad dial-peer Voice 401 voip description To/From CUCM subscriber for Voice preference 2 destination-pattern 1 - . voice-class codec 1 session protocol sipv2 session target ipv4:192.168.0.4 incoming called-number 9T (24) fax rate disable no vad dial-peer Voice 411 voip description To/From CUCM publisher for Voice Chapter 10: Deployment Scenarios 201 preference 5 destination pattern 1... voice-class codec 1 session protocol sipv2 session target ipv4:192.168.0.6 (25) incoming called-number 9T dtmf-relay rtp-nte no vad sip-ua retry invite 2 retry bye 2 retry cancel 2 sip-server dns:sip.server.com g729-annexb override Note the following: These commands define DSP resources for CUCM's use (conferencing). 2. This command enables SIP-to-SIP call process