Cisco CP-7911G-CH1 Administration Guide - Page 120

Where to Go Next Setting Up Cisco Unified IP Phones using SIP Command or Action RTP dynamic payload type values: NTE = 101 Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121, fax-relay=122 CAS=123, ClearChan=125, PCM switch over u-law=0,A-law=8 RTP comfort noise payload type = 19 fax rate = voice, payload size = 20 bytes fax protocol = system fax-relay ecm enable fax NSF = 0xAD0051 (default) codec = g729r8, payload size = 20 bytes, Media Setting = flow-through (global) Expect factor = 0, Icpif = 20, Playout Mode is set to adaptive, Initial 60 ms, Max 300 ms Playout-delay Minimum mode is set to default, value 40 ms Fax nominal 300 ms Max Redirects = 1, signaling-type = cas, VAD = enabled, Poor QOV Trap = disabled, Source Interface = NONE voice class sip url = system, voice class sip rel1xx = system, monitor probe method: icmp-ping ip address: 10.2.161.187, Monitored destination reachable voice class perm tag = `' Time elapsed since last clearing of voice call statistics never Connect Time = 0, Charged Units = 0, Successful Calls = 0, Failed Calls = 0, Incomplete Calls = 0 Accepted Calls = 0, Refused Calls = 0, Last Disconnect Cause is "", Last Disconnect Text is "", Last Setup Time = 0. Purpose Where to Go Next The next step is configuring incoming and outgoing calls for Cisco Unified SRST. For more information, see the "Configuring Call Handling" section on page 123. For additional information, see the "Additional References" section on page 26 in the "Cisco Unified SCCP and SIP SRST Feature Overview" chapter. 120 Cisco Unified SCCP and SIP SRST System Administrator Guide OL-13143-04