Where To Go Next - Cisco CP-7911G-CH1 System Administrator Manual

Unified sccp and sip srst
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Where to Go Next

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.
Where to Go Next
The next step is configuring incoming and outgoing calls for Cisco Unified SRST. For more information,
see the
For additional information, see the
SCCP and SIP SRST Feature Overview"
Cisco Unified SCCP and SIP SRST System Administrator Guide
120
"Configuring Call Handling" section on page
Purpose
123.
"Additional References" section on page 26
chapter.
Setting Up Cisco Unified IP Phones using SIP
in the
"Cisco Unified
OL-13143-04

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