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Cisco IOS Voice Command Reference - S commands
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sccp through service-type call-check
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session through sgcp tse payload
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show aal2 profile through show call filter match-list
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show call history fax through show debug condition
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show dial-peer through show gatekeeper zone prefix
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show gateway through show modem relay statistics
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show mrcp client session active through show sip dhcp
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show sip service through show trunk hdlc
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show vdev through show voice statistics memory-usage
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show voice trace through shutdown (voice-port)
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signal through srv version
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ss7 mtp2-variant through switchover method
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Index
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Feedback
ss7 mtp2-variant through switchover method
- ss7 mtp2-variant
- ss7 mtp2-variant bellcore
- ss7 mtp2-variant itu
- ss7 mtp2-variant ntt
- ss7 mtp2-variant ttc
- ss7 mtp2-variant itu-white
- ss7 session
- ss7 session cumack_t
- ss7 session kp_t
- ss7 session m_cumack
- ss7 session m_outseq
- ss7 session m_rcvnum
- ss7 session m_retrans
- ss7 session retrans_t
- ss7 set
- ss7 set failover-timer
- station-id name
- station-id number
- stats
- stcapp
- stcapp call-control mode
- stcapp feature callback
- stcapp ccm-group
- stcapp feature access-code
- stcapp feature callback
- stcapp feature speed-dial
- stcapp register capability
- stcapp security mode
- stcapp security trustpoint
- stcapp supplementary-services
- stcapp timer
- stun
- stun flowdata agent-id
- stun flowdata catlife
- stun flowdata keepalive
- stun flowdata shared-secret
- stun usage firewall-traversal flowdata
- subaddress
- subcell-mux
- subscription asnl session history
- subscription maximum
- supervisory answer dualtone
- supervisory custom-cptone
- supervisory disconnect
- supervisory disconnect anytone
- supervisory disconnect dualtone
- supervisory disconnect dualtone voice-class
- supervisory disconnect lcfo
- supervisory dualtone-detect-params
- supervisory sit us
- supplementary-service h225-notify cid-update (dal peer)
- supplementary-service h225-notify cid-update (voice-service)
- supplementary-service h450.2 (dial peer)
- supplementary-service h450.2 (voice-service)
- supplementary-service h450.3 (dial peer)
- supplementary-service h450.3 (voice-service)
- supplementary-service h450.7
- supplementary-service h450.12 (dial peer)
- supplementary-service h450.12 (voice-service)
- supplementary-service media-renegotiate
- supplementary-service qsig call-forward
- supplementary-service sip
- supported language
- suppress
- suspend-resume (SIP)
- switchback interval
- switchback method
- switchover method
ss7 mtp2-variant
To configure a Signaling System 7 (SS7) signaling link, use the ss7 mtp2-variant command in global configuration mode. To restore the designated default, use the no form of this command.
ss7 mtp2-variant [ bellcore channel | itu-white channel | ntt channel | ttc channel ] [parameters]
no ss7 mtp2-variant
Syntax Description
Usage Guidelines
The MTP2 variant has timers and parameters that can be configured using the values listed in the following tables. To restore the designated default, use the no or the default form of the command (see the "Examples" section below).
NoteWhen the bellcore or itu white variant is selected, this command enters a new configuration mode for setting MTP2 parameters: ITU configuration mode. See the error correctioncommand reference for information about setting MTP2 parameters from this mode.
Table 1 Bellcore (Telcordia Technologies) Parameters and Values Parameter
Description
Default
Range
T1
Aligned/ready timer duration (milliseconds)
13000
1000 to 65535
T2
Not aligned timer (milliseconds)
11500
1000 to 65535
T3
Aligned timer (milliseconds)
11500
1000 to 65535
T4 Emergency Proving
Emergency proving timer (milliseconds)
1600
1000 to 65535
T4 Normal Proving
Normal proving period (milliseconds)
2300
1000 to 65535
T5
Sending status indication busy (SIB) timer (milliseconds)
100
80 to 65535
T6
Remote congestion timer (milliseconds)
6000
1000 to 65535
T7
Excessive delay timer (milliseconds)
1000
500 to 65535
lssu len
1- or 2-byte link status signal unit (LSSU) format
1
1 to 2
unacked MSUs
Maximum number of message signal units (MSUs) awaiting acknowledgment (ACK)
127
16 to 127
proving attempts
Maximum number of attempts to prove alignment
5
3 to 8
SUERM threshold
Signal Unit Error Rate Monitor (SUERM) error-rate threshold
64
32 to 128
SUERM number octets
SUERM octet-counting mode
16
8 to 32
SUERM number signal units
Signal units (good or bad) needed to decrement Error Rate Monitor (ERM)
256
128 to 512
Tie AERM Emergency
Alignment Error Rate Monitor (AERM) emergency error-rate threshold
1
1 to 8
Tie AERM Normal
AERM normal error-rate threshold
4
1 to 8
Table 2 ITU-white Parameters and Values Parameter
Description
Default
Range
T1
Aligned/ready timer duration (milliseconds)
40000
1000 to 65535
T2
Not aligned timer (milliseconds)
5000
1000 to 65535
T3
Aligned timer (milliseconds)
1000
1000 to 65535
T4 Emergency Proving
Emergency proving timer (milliseconds)
500
1000 to 65535
T4 Normal Proving
Normal proving timer (milliseconds)
8200
1000 to 65535
T5
Sending SIB timer (milliseconds)
100
80 to 65535
T6
Remote congestion timer (milliseconds)
6000
1000 to 65535
T7
Excessive delay timer (milliseconds)
1000
1000 to 65535
lssu len
1- or 2-byte link status signal unit (LSSU) format
1
1 to 2
msu len
message signal unit (MSU) length
1
1 to 2
unacked MSUs
Maximum number of MSUs awaiting acknowledgment (ACK)
127
16 to 127
proving attempts
Maximum number of attempts to prove alignment
5
3 to 8
SUERM threshold
Signal Unit Error Rate Monitor (SUERM) error-rate threshold
64
32 to 128
SUERM number octets
SUERM octet counting mode
16
8 to 32
SUERM - number - signal - units
Signal units (good or bad) needed to decrement Error Rate Monitor (ERM)
256
128 to 512
Tie AERM Emergency
Alignment Error Rate Monitor (AERM) emergency error-rate threshold
1
1 to 8
Tin AERM Normal
AERM normal error-rate threshold
4
1 to 8
Table 3 NTT Parameters and Values Parameter
Description
Default
Range
T1
Aligned/ready timer duration (milliseconds)
15000
1000 to 65535
T2
Not aligned timer (milliseconds)
5000
1000 to 65535
T3
Aligned timer (milliseconds)
3000
1000 to 65535
T4 Emergency Proving
Emergency proving timer (milliseconds)
3000
1000 to 65535
T5
Sending SIB timer (milliseconds)
200
80 to 65535
T6
Remote congestion timer (milliseconds)
2000
1000 to 65535
T7
Excessive delay timer (milliseconds)
3000
1000 to 65535
TA
SIE interval timer (milliseconds)
20
10 to 500
TF
Fill-in Signal Unit (FISU) interval timer (milliseconds)
20
10 to 500
TO
SIO interval timer (milliseconds)
20
10 to 500
TS
SIOS interval timer (milliseconds)
20
10 to 500
unacked MSUs
Maximum number of message signal units (MSUs) awaiting acknowledgment (ACK)
40
16 to 40
proving attempts
Maximum number of attempts to prove alignment
5
3 to 8
SUERM threshold
Signal Unit Error Rate Monitor (SUERM) e error-rate threshold
64
32 to 128
SUE RM - number - octets
SUERM octet counting mode
16
8 to 32
SUERM - number - signal - units
Signal Unit Error Rate Monitor (SUERM) units (good or bad) needed to decrement Error Rate Monitor (ERM)
256
128 to 512
Tie - AERM - Emergency
Alignment Error Rate Monitor (AERM) emergency error-rate threshold
1
1 to 8
Table 4 TTC Parameters and Values Parameter
Description
Default
Range
T1
Aligned/ready timer duration (milliseconds)
15000
1000 to 65535
T2
Not aligned timer (milliseconds)
5000
1000 to 65535
T3
Aligned timer (milliseconds)
3000
1000 to 65535
T4 Emergency Proving
Emergency proving timer (milliseconds)
3000
1000 to 65535
T5
Sending SIB timer (milliseconds)
200
80 to 65535
T6
Remote congestion timer (milliseconds)
2000
1000 to 65535
T7
Excessive delay timer (milliseconds)
3000
1000 to 65535
TA
SIE interval timer (milliseconds)
20
10 to 500
TF
FISU interval timer (milliseconds)
20
10 to 500
TO
SIO interval timer (milliseconds)
20
10 to 500
TS
SIOS interval timer (milliseconds)
20
10 to 500
unacked MSUs
Maximum number of message signal units (MSUs) awaiting acknowledgment (ACK)
40
16 to 40
proving attempts
Maximum number of attempts to prove alignment
5
3 to 8
SUERM threshold
Signal Unit Error Rate Monitor (SUERM) error - rate threshold
64
32 to 128
SUERM number octets
SUERM octet counting mode
16
8 to 32
SUERM number signal units
Signal units (good or bad) needed to decrement ERM
256
128 to 512
Tie AERM Emergency
AERM emergency error-rate threshold
1
1 to 8
Examples
The following example configures an SS7 channel (link) for Preventive Cyclic Retransmission (PCR) with forced retransmission initiated. In this example, SS7 channel 0 is configured with the ITU-white protocol variant using the PCR error correction method.
Router# configure terminal Router(config)# ss7 mtp2-variant itu-white 0 Router(config-ITU)# error-correction pcr forced-retransmission enabled N2 1000 Router(config-ITU)# endThe following example disables error-correction:
Router(config-ITU)# no error-correctionss7 mtp2-variant bellcore
To configure the router for Telcordia Technologies (formerly Bellcore) standards, use the ss7 mtp2-variant bellcore command in global configuration mode.
Command Default
Bellcore is the default variant if no other is configured. See the table below for default parameters.
Usage Guidelines
This MTP2 variant has timers and parameters that can be configured using the values listed in the table below. To restore the designated default, use the no or the default form of the command (see example below).
NoteTimer durations are converted to 10-millisecond units. For example, a T1 value of 1005 is converted to 100, which results in an actual timeout duration of 1000 ms. This is true for all timers and all variants.
Table 5 Bellcore (Telcordia Technologies) Parameters and Values Parameter
Description
Default
Range
T1
Aligned/ready timer duration (milliseconds)
13000
1000 to 65535
T2
Not aligned timer (milliseconds)
11500
1000 to 65535
T3
Aligned timer (milliseconds)
11500
1000 to 65535
T4 -Emergency-Proving
Emergency proving timer (milliseconds)
600
1000 to 65535
T4 -Normal-Proving
Normal proving period (milliseconds)
2300
1000 to 65535
T5
Sending SIB timer (milliseconds)
100
80 to 65535
T6
Remote congestion timer (milliseconds)
6000
1000 to 65535
T7
Excessive delay timer (milliseconds)
1000
500 to 65535
lssu -len
1- or 2-byte LSSU format
1
1 to 2
unacked -MSUs
Maximum number of MSUs waiting ACK
127
16 to 127
proving -attempts
Maximum number of attempts to prove alignment
5
3 to 8
SUERM -threshold
SUERM error-rate threshold
64
32 to 128
SUERM -number-octets
SUERM octet-counting mode
16
8 to 32
SUERM -number-signal units
Signal units (good or bad) needed to dec ERM
256
128 to 512
Tie -AERM-Emergency
AERM emergency error-rate threshold
1
1 to 8
Tie -AERM-Normal
AERM normal error-rate threshold
4
1 to 8
ss7 mtp2-variant itu
To configure the router for ITU (International Telecom United) standards, use the ss7 mtp2-variant itu command in global configuration mode.
Command Default
Bellcore is the default variant if no other is configured. See the table below for ITU default parameters.
Usage Guidelines
The ITU MTP2 variant has timers and parameters that can be configured using the values listed in the table below. To restore the designated default, use the no or the default form of the command (see the example below).
Table 6 ITU (White) Parameters and Values Parameter
Description
Default
Range
T1
Aligned/ready timer duration (milliseconds)
40000
1000 to 65535
T2
Not aligned timer (milliseconds)
5000
1000 to 65535
T3
Aligned timer (milliseconds)
1000
1000 to 65535
T4 -Emergency-Proving
Emergency proving timer (milliseconds)
500
1000 to 65535
T4 -Normal-Proving
Normal proving timer (milliseconds)
8200
1000 to 65535
T5
Sending SIB timer (milliseconds)
100
80 to 65535
T6
Remote congestion timer (milliseconds)
6000
1000 to 65535
T7
Excessive delay timer (milliseconds)
1000
1000 to 65535
lssu -len
1- or 2-byte LSSU format
1
1 to 2
msu -len
unacked -MSUs
Maximum number of MSUs waiting ACK
127
16 to 127
proving -attempts
Maximum number of attempts to prove alignment
5
3 to 8
SUERM -threshold
SUERM error rate threshold
64
32 to 128
SUERM -number-octets
SUERM octet counting mode
16
8 to 32
SUERM -number-signal units
Signal units (good or bad) needed to dec ERM
256
128 to 512
Tie -AERM-Emergency
AERM emergency error-rate threshold
1
1 to 8
Tin -AERM-Normal
AERM normal error-rate threshold
4
1 to 8
ss7 mtp2-variant ntt
To configure the router for NTT (Japan) standards, use the ss7 mtp2-variant ntt command in global configuration mode.
Command Default
Bellcore is the default variant if no other is configured. See the table below for NTT default parameters.
Usage Guidelines
The NTT MTP2 variant has timers and parameters that can be configured using the values listed in the table below. To restore the designated default, use the no or the default form of the command (see the example below).
Table 7 NTT Parameters and Values Parameter
Description
Default
Range
T1
Aligned/ready timer duration (milliseconds)
15000
1000 to 65535
T2
Not aligned timer (milliseconds)
5000
1000 to 65535
T3
Aligned timer (milliseconds)
3000
1000 to 65535
T4 -Emergency-Proving
Emergency proving timer (milliseconds)
3000
1000 to 65535
T5
Sending SIB timer (milliseconds)
200
80 to 65535
T6
Remote congestion timer (milliseconds)
2000
1000 to 65535
T7
Excessive delay timer (milliseconds)
3000
1000 to 65535
TA
SIE interval timer (milliseconds)
20
10 to 500
TF
FISU interval timer (milliseconds)
20
10 to 500
TO
SIO interval timer (milliseconds)
20
10 to 500
TS
SIOS interval timer (milliseconds)
20
10 to 500
unacked -MSUs
Maximum number of MSUs waiting ACK
40
16 to 40
proving -attempts
Maximum number of attempts to prove alignment
5
3 to 8
SUERM -threshold
SUERM error rate threshold
64
32 to 128
SUERM -number-octets
SUERM octet counting mode
16
8 to 32
SUERM -number-signal units
Signal units (good or bad) needed to dec ERM
256
128 to 512
Tie -AERM-Emergency
AERM emergency error-rate threshold
1
1 to 8
ss7 mtp2-variant ttc
To configure the router for TTC (Japan Telecom) standards, use the ss7 mtp2-variant ttc command in global configuration mode.
Command Default
Bellcore is the default variant if no other is configured. See the table below for TTC default parameters.
Usage Guidelines
The TTC MTP2 variant has timers and parameters that can be configured using the values listed in the table below. To restore the designated default, use the no or the default form of the command (see the example below).
Table 8 TTC Parameters and Values Parameter
Description
Default
Range
T1
Aligned/ready timer duration (milliseconds)
15000
1000 to 65535
T2
Not aligned timer (milliseconds)
5000
1000 to 65535
T3
Aligned timer (milliseconds)
3000
1000 to 65535
T4 -Emergency-Proving
Emergency proving timer (milliseconds)
3000
1000 to 65535
T5
Sending SIB timer (milliseconds)
200
80 to 65535
T6
Remote congestion timer (milliseconds)
2000
1000 to 65535
T7
Excessive delay timer (milliseconds)
3000
1000 to 65535
TA
SIE interval timer (milliseconds)
20
10 to 500
TF
FISU interval timer (milliseconds)
20
10 to 500
TO
SIO interval timer (milliseconds)
20
10 to 500
TS
SIOS interval timer (milliseconds)
20
10 to 500
unacked -MSUs
Maximum number of MSUs waiting ACK
40
16 to 40
proving -attempts
Maximum number of attempts to prove alignment
5
3 to 8
SUERM -threshold
SUERM error rate threshold
64
32 to 128
SUERM -number-octets
SUERM octet counting mode
16
8 to 32
SUERM -number-signal-units
Signal units (good or bad) needed to dec ERM
256
128 to 512
Tie -AERM-Emergency
AERM emergency error-rate threshold
1
1 to 8
ss7 mtp2-variant itu-white
To configure the router for International Telecommunications Union (ITU) standards, use the ss7 mtp2-variant itu-white command in global configuration mode.
Command Default
Bellcore is the default variant if no other is configured. See the table below for ITU default parameters.
Usage Guidelines
The ITU MTP2 variant has timers and parameters that can be configured using the values listed in the table below. To restore the designated default, use the no or the default form of the command.
Table 9 ITU (White) Parameters and Values Parameter
Description
Default
Range
T1
Aligned/ready timer duration (milliseconds [ms])
40000
1000 to 65535
T2
Not aligned timer (ms)
5000
1000 to 65535
T3
Aligned timer (ms)
1000
1000 to 65535
T4-Emergency-Proving
Emergency proving timer (ms)
500
1000 to 65535
T4-Normal-Proving
Normal proving timer (ms)
8200
1000 to 65535
T5
Sending SIB timer (ms)
100
80 to 65535
T6
Remote congestion timer (ms)
6000
1000 to 65535
T7
Excessive delay timer (ms)
1000
1000 to 65535
lssu-len
1- or 2-byte Links Status Signal Unit (LSSU) format
1
1 to 2
msu-len
--
--
--
unacked-MSUs
Maximum number of Message Signal Units (MSUs) waiting acknowledgement
127
16 to 127
proving-attempts
Maximum number of attempts to prove alignment
5
3 to 8
SUERM-threshold
Signal unit error monitor (SUERM) error rate threshold
64
32 to 128
SUERM-number-octets
SUERM octet counting mode
16
8 to 32
SUERM-number-signal- units
Signal units (good or bad) needed to dec Embedded Resource Manager (ERM)
256
128 to 512
Tie-AERM-Emergency
Alignment Unit Error Rate Monitor (AERM) emergency error-rate threshold
1
1 to 8
Tin-AERM-Normal
AERM normal error-rate threshold
4
1 to 8
Examples
The following example shows how to set the emergency proving period on channel 1 to 10,000 ms:
Router(config)# ss7 mtp2-variant itu-white 1 Router(config-ITU)# t4-Emergency-Proving 10000The following example shows how to restore the emergency proving period default value of 5000 ms:
Router(config)# ss7 mtp2-variant itu-white 1 Router(config-ITU)# default t4-Emergency-Proving 5000ss7 session
To create a Reliable User Datagram Protocol (RUDP) session and explicitly add an RUDP session to a Signaling System 7 (SS7) session set, use the ss7 session command in global configuration mode. To delete the session, use the no form of this command.
ss7 session session-id address destination-address destinaion-port local-address local-port [ session-set session-number ]
no ss7 session session-id
Syntax Description
session -id
SS7 session number. Valid values are 0 and 1. You must enter a hyphen with no space following it after the session keyword.
address destination -address
Specifies the SS7 session IP address.
destination -address
The local IP address of the router in four-part dotted-decimal format.
The local IP address for both sessions, 0 and 1, must be the same.
destination -port
The number of the local UDP port on which the router expects to receive messages from the media gateway controller (MGC) . Specify any UDP port that is not used by another protocol as defined in RFC 1700 and that is not otherwise used in your network.
The local UDP port must be different for session 0 and session 1.
Valid port ranges are from 1024 to 9999.
local -address
The remote IP address of the MGC in four-part dotted-decimal format.
local -port
The number of the remote UDP port on which the MGC is configured to listen. This UDP port cannot be used by another protocol as defined in RFC 1700 and cannot be otherwise used in the network. Valid port ranges are from 1024 to 9999.
session -setsession - number
(Optional) Assigns an SS7 session to an SS7 session set.
Usage Guidelines
For the Cisco 2600-based SLT, you can configure a maximum of four sessions, two for each Cisco SLT. In a redundant VSC configuration, session 0 and session 2 are configured to one VSC, and session 1 and session 3 are configured to the other. Session 0/1 and session 2/3 run to the Cisco SLT.
The VSC must be configured to send messages to the local port, and it must be configured to listen on the remote port. You must also reload the router whenever you remove a session or change the parameters of a session.
This command replaces the ss7 session-0 address and ss7 session-1 address commands, which contain hard-coded session numbers. The new command is used for the new dual Ethernet capability.
The new CLI supports both single and dual Ethernet configuration by being backward compatible with the previous session-0 and session-1 commands so that you can configure a single Ethernet instead of two, if needed.
For the Cisco AS5350 and Cisco AS5400-based SLT, you can configure a maximum of two sessions, one for each signaling link. In a redundant MGC configuration, session 0 is configured to one MGC and session 1 is configured to the other.
The MGC must be configured to send messages to the local port, and the MGC must be configured to listen on the remote port.
You must reload the router whenever you remove a session or change the parameters of a session.
By default, each RUDP session must belong to SS7 session set 0. This allows backward compatibility with existing SS7 configurations.
If the session-set keyword is omitted, the session is added to the default SS7 session set 0. This allows backward compatibility with older configurations. Entering the no form of the command is still sufficient to remove the session ID for that RUDP session.
If you want to change the SS7 session set to which a session belongs, you have to remove the entire session first. This is intended to preserve connection and recovery logic.
Examples
The following example sets up two sessions on a Cisco 2611 and creates session set 2:
ss7 session-0 address 172.16.1.0 7000 172.16.0.0 7000 session-set 2 ss7 session-1 address 172.17.1.0 7002 172.16.0.0 7001 session-set 2
NoteThe example above shows how the local IP addresses in session-0 and session-1 must be the same.
Related Commands
Command
Description
ss7 session cumack_t
Sets the cumulative acknowledgment timer.
ss7 session k_pt
Sets the null segment (keepalive) timer.
ss7 session m_cumack
Sets the maximum number of segments that can be received before the RUDP sends an acknowledgment.
ss7 session m_outseq
Sets the maximum number of out-of-sequence segments that can be received before the RUDP sends an extended acknowledgment.
ss7 session m_rcvnum
Sets the maximum number of segments that the remote end can send before receiving an acknowledgment.
ss7 session m_retrans
Sets the maximum number of times that the RUDP attempts to resend a segment before declaring the connection invalid.
ss7 session retrans_t
Sets the retransmission timer.
ss7 session cumack_t
To set the Reliable User Datagram Protocol (RUDP) cumulative acknowledgment timer for a specific SS7 signaling link session, use the ss7 session cumack_tcommand in global configuration mode. To reset to the default, use the no form of this command.
Syntax Description
session -number
SS7 session number. Valid values are 0 and 1. You must enter the hyphen, with no space following it, after the session keyword.
milliseconds
Interval, in milliseconds, that the RUDP waits before it sends an acknowledgment after receiving a segment. Range is from 100 to 65535. The value should be less than the value configured for the retransmission timer by using the ss7 session-session number retrans_tcommand.
Usage Guidelines
The cumulative acknowledgment timer determines when the receiver sends an acknowledgment. If the timer is not already running, it is initialized when a valid data, null, or reset segment is received. When the cumulative acknowledgment timer expires, the last in-sequence segment is acknowledged. The RUDP typically tries to "piggyback" acknowledgments on data segments being sent. However, if no data segment is sent in this period of time, it sends a standalone acknowledgment.
Caution
Use the default setting. Do not change session timers unless instructed to do so by Cisco technical support. Changing timers may result in service interruption or outage.
Examples
The following example sets up two sessions and sets the cumulative acknowledgment timer to 320 ms for each one:
ss7 session-0 address 255.255.255.251 7000 255.255.255.254 7000 ss7 session-0 cumack_t 320 ss7 session-1 address 255.255.255.253 7002 255.255.255.254 7001 ss7 session-1 cumack_t 320Related Commands
Command
Description
show ss7
Displays the SS7 configuration.
ss7 session k_pt
Sets the null segment (keepalive) timer.
ss7 session m_cumack
Sets the maximum number of segments that can be received before the RUDP sends an acknowledgment.
ss7 session m_outseq
Sets the maximum number of out-of-sequence segments that can be received before the RUDP sends an extended acknowledgment.
ss7 session m_rcvnum
Sets the maximum number of segments that the remote end can send before receiving an acknowledgment.
ss7 session m_retrans
Sets the maximum number of times that the RUDP attempts to resend a segment before declaring the connection invalid.
ss7 session retrans_t
Sets the retransmission timer.
ss7 session kp_t
To set the null segment (keepalive) timer for a specific SS7 signaling link session, use the ss7 session kp_tcommand in global configuration mode. To reset to the default, use the no form of this command.
Syntax Description
session -number
SS7 session number. Valid values are 0 and 1. You must enter the hyphen, with no space following it, after the session keyword.
milliseconds
Interval, in milliseconds, that the Reliable User Datagram Protocol (RUDP) waits before sending a keepalive to verify that the connection is still active. Valid values are 0 and from100 to 65535. Default is 2000.
Usage Guidelines
The null segment timer determines when a null segment (keepalive) is sent by the client Cisco 2600 series router. On the client, the timer starts when the connection is established and is reset each time a data segment is sent. If the null segment timer expires, the client sends a keepalive to the server to verify that the connection is still functional. On the server, the timer restarts each time a data or null segment is received from the client.
Caution
Use the default setting. Do not change session timers unless instructed to do so by Cisco technical support. Changing timers may result in service interruption or outage.
The value of the server’s null segment timer is twice the value configured for the client. If no segments are received by the server in this period of time, the connection is no longer valid.
To disable keepalive, set this parameter to 0.
Examples
The following example sets up two sessions and sets a keepalive of 1,800 ms for each one:
ss7 session-0 address 255.255.255.251 7000 255.255.255.254 7000 ss7 session-0 kp_t 1800 ss7 session-1 address 255.255.255.253 7002 255.255.255.254 7001 ss7 session-1 kp_t 1800Related Commands
Command
Description
show ss7
Displays the SS7 configuration.
ss7 session cumack_t
Sets the cumulative acknowledgment timer.
ss7 session m_cumack
Sets the maximum number of segments that can be received before the RUDP sends an acknowledgment.
ss7 session m_outseq
Sets the maximum number of out-of-sequence segments that can be received before the RUDP sends an extended acknowledgment.
ss7 session m_rcvnum
Sets the maximum number of segments that the remote end can send before receiving an acknowledgment.
ss7 session m_retrans
Sets the maximum number of times that the RUDP attempts to resend a segment before declaring the connection invalid.
ss7 session retrans_t
Sets the retransmission timer.
ss7 session m_cumack
To set the maximum number of segments that can be received before the Reliable User Datagram Protocol (RUDP) sends an acknowledgment in a specific SS7 signaling link session, use the ss7 session m_cumackcommand in global configuration mode. To reset to the default, use the no form of this command.
Syntax Description
session -number
SS7 session number. Valid values are 0 and 1. You must enter the hyphen, with no space following it, after the session keyword.
segments
Maximum number of segments that can be received before the Reliable User Datagram Protocol (RUDP) sends an acknowledgment. Range is from 0 to 255. Default is 3.
Usage Guidelines
Caution
Use the default setting. Do not change session timers unless instructed to do so by Cisco technical support. Changing timers may result in service interruption or outage.
The cumulative acknowledgment counter records the number of unacknowledged, in-sequence data, null, or reset segments received without a data, null, or reset segment being sent to the transmitter. If this counter reaches the configured maximum, the receiver sends a standalone acknowledgment (a standalone acknowledgment is a segment that contains only acknowledgment information). The standalone acknowledgment contains the sequence number of the last data, null, or reset segment received.
If you set this parameter to 0, an acknowledgment is sent immediately after a data, null, or reset segment is received.
Examples
The following example sets up two sessions and in each session sets a maximum of two segments for receipt before acknowledgment:
ss7 session-0 address 255.255.255.251 7000 255.255.255.254 7001 ss7 session-0 m_cumack 2 ss7 session-1 address 255.255.255.253 7002 255.255.255.254 7000 ss7 session-1 m_cumack 2Related Commands
Command
Description
show ss7
Displays the SS7 configuration.
ss7 session cumack_t
Sets the cumulative acknowledgment timer.
ss7 session k_pt
Sets the null segment (keepalive) timer.
ss7 session m_outseq
Sets the maximum number of out-of-sequence segments that can be received before the RUDP sends an extended acknowledgment.
ss7 session m_rcvnum
Sets the maximum number of segments that the remote end can send before receiving an acknowledgment.
ss7 session m_retrans
Sets the maximum number of times that the RUDP attempts to resend a segment before declaring the connection invalid.
ss7 session retrans_t
Sets the retransmission timer.
ss7 session m_outseq
To set the maximum number of out-of-sequence segments that can be received before the Reliable User Datagram Protocol (RUDP) sends an extended acknowledgment in a specific SS7 signaling link session, use the ss7 session m_outseqcommand in global configuration mode. To reset to the default, use the no form of this command.
Syntax Description
session -number
SS7 session number. Valid values are 0 and 1. You must enter the hyphen, with no space following it, after the session keyword.
segments
Maximum number of out-of-sequence segments that can be received before the RUDP sends an extended acknowledgment. If the specified number of segments are received out of sequence, an Extended Acknowledgment segment is sent to inform the sender which segments are missing. Range is from 0 to 255. Default is 3.
Usage Guidelines
Caution
Use the default setting. Do not change session timers unless instructed to do so by Cisco technical support. Changing timers may result in service interruption or outage.
The out-of-sequence acknowledgment counter records the number of data segments that have arrived out of sequence. If this counter reaches the configured maximum, the receiver sends an extended acknowledgment segment that contains the sequence numbers of the out-of-sequence data, null, and reset segments received. When the transmitter receives the extended acknowledgment segment, it retransmits the missing data segments.
If you set this parameter to 0, an acknowledgment is sent immediately after an out-of-sequence segment is received.
Examples
The following example sets up two sessions and sets a maximum number of four out-of-sequence segments for each session:
ss7 session-0 address 255.255.255.251 7000 255.255.255.254 7001 ss7 session-0 m_outseq 4 ss7 session-1 address 255.255.255.253 7002 255.255.255.254 7000 ss7 session-1 m_outseq 4Related Commands
Command
Description
show ss7
Displays the SS7 configuration.
ss7 session cumack_t
Sets the cumulative acknowledgment timer.
ss7 session k_pt
Sets the null segment (keepalive) timer.
ss7 session m_cumack
Sets the maximum number of segments that can be received before the RUDP sends an acknowledgment.
ss7 session m_rcvnum
Sets the maximum number of segments that the remote end can send before receiving an acknowledgment.
ss7 session m_retrans
Sets the maximum number of times that the RUDP attempts to resend a segment before declaring the connection invalid.
ss7 session retrans_t
Sets the retransmission timer.
ss7 session m_rcvnum
To set the maximum number of segments that the remote end can send before receiving an acknowledgment in a specific SS7 signaling link session, use the ss7 session m_rcvnumcommand in global configuration mode. To reset to the default, use the no form of this command.
Syntax Description
session -number
SS7 session number. Valid values are 0 and 1. You must enter the hyphen, with no space following it, after the session keyword.
segments
Maximum number of segments that the remote (Cisco IOS software) end can send before receiving an acknowledgment. Range is from 1 to 64. Default is 32.
Usage Guidelines
The outstanding segments counter is the maximum number of segments that the Cisco IOS software end of the connection can send without getting an acknowledgment from the receiver. The receiver uses the counter for flow control.
Caution
Use the default setting. Do not change session timers unless instructed to do so by Cisco technical support. Changing timers may result in service interruption or outage.
Examples
The following example sets up two sessions and for each session sets a maximum of 36 segments for receipt before an acknowledgment:
ss7 session-0 address 255.255.255.251 7000 255.255.255.254 7001 ss7 session-0 m_rcvnum 36 ss7 session-1 address 255.255.255.253 7002 255.255.255.254 7000 ss7 session-1 m_rcvnum 36Related Commands
Command
Description
show ss7
Displays the SS7 configuration.
ss7 session cumack_t
Sets the cumulative acknowledgment timer.
ss7 session k_pt
Sets the null segment (keepalive) timer.
ss7 session m_cumack
Sets the maximum number of segments that can be received before the RUDP sends an acknowledgment.
ss7 session m_outseq
Sets the maximum number of out-of-sequence segments that can be received before the RUDP sends an extended acknowledgment.
ss7 session m_retrans
Sets the maximum number of times that the Reliable User Datagram Protocol (RUDP) attempts to resend a segment before declaring the connection invalid.
ss7 session retrans_t
Sets the retransmission timer.
ss7 session m_retrans
To set the maximum number of times that the Reliable User Datagram Protocol (RUDP) attempts to resend a segment before declaring the connection invalid in a specific SS7 signaling link session, use the ss7 session m_retrans command in global configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
Caution
Use the default setting. Do not change session timers unless instructed to do so by Cisco technical support. Changing timers may result in service interruption or outage.
The retransmission counter is the number of times a segment has been retransmitted. If this counter reaches the configured maximum, the transmitter resets the connection and informs the upper-layer protocol.
If you set this parameter to 0, the RUDP attempts to resend the segment continuously.
Examples
The following example sets up two sessions and for each session sets a maximum number of three times to resend before a session becomes invalid:
ss7 session-0 address 255.255.255.251 7000 255.255.255.254 7001 ss7 session-0 m_retrans 3 ss7 session-1 address 255.255.255.253 7002 255.255.255.254 7000 ss7 session-1 m_retrans 3Related Commands
Command
Description
show ss7
Displays the SS7 configuration.
ss7 session cumack_t
Sets the cumulative acknowledgment timer.
ss7 session k_pt
Sets the null segment (keepalive) timer.
ss7 session m_cumack
Sets the maximum number of segments that can be received before the RUDP sends an acknowledgment.
ss7 session m_outseq
Sets the maximum number of out-of-sequence segments that can be received before the RUDP sends an extended acknowledgment.
ss7 session m_rcvnum
Sets the maximum number of segments that the remote end can send before receiving an acknowledgment.
ss7 session retrans_t
Sets the retransmission timer.
ss7 session retrans_t
To set the amount of time that the Reliable User Datagram Protocol (RUDP) waits to receive an acknowledgment for a segment in a specific SS7 signaling link session, use the ss7 session retrans_tcommand in global configuration mode. If the RUDP does not receive the acknowledgment in this time period, the RUDP retransmits the segment. To reset to the default, use the no form of this command.
Usage Guidelines
Caution
Use the default setting. Do not change session timers unless instructed to do so by Cisco technical support. Changing timers may result in service interruption or outage.
The retransmission timer is used to determine whether a packet must be retransmitted and is initialized each time a data, null, or reset segment is sent. If an acknowledgment for the segment is not received by the time the retransmission timer expires, all segments that have been transmitted--but not acknowledged--are retransmitted.
This value should be greater than the value configured for the cumulative acknowledgment timer by using the ss7 session cumack_tcommand.
Examples
The following example sets up two sessions and specifies 550 ms as the time to wait for an acknowledgment for each session:
ss7 session-0 address 255.255.255.251 7000 255.255.255.254 7001 ss7 session-0 retrans_t 550 ss7 session-1 address 255.255.255.253 7002 255.255.255.254 7000 ss7 session-1 retrans_t 550Related Commands
Command
Description
show ss7
Displays the SS7 configuration.
ss7 session cumack_t
Sets the cumulative acknowledgment timer.
ss7 session k_pt
Sets the null segment (keepalive) timer.
ss7 session m_cumack
Sets the maximum number of segments that can be received before the RUDP sends an acknowledgment.
ss7 session m_outseq
Sets the maximum number of out-of-sequence segments that can be received before the RUDP sends an extended acknowledgment.
ss7 session m_rcvnum
Sets the maximum number of segments that the remote end can send before receiving an acknowledgment.
ss7 session m_retrans
Sets the maximum number of times that the RUDP attempts to resend a segment before declaring the connection invalid.
ss7 set
NoteEffective with Cisco IOS Release 12.2(15)T, the ss7 set command replaces the ss7 set failover-timer command.
To independently select failover-timer values for each session set and to specify the amount of time that the SS7 Session Manager waits for the active session to recover or for the standby media gateway controller (MGC) to indicate that the Cisco Signaling Link Terminal (SLT) should switch traffic to the standby session, use the ss7 setcommand in global configuration mode. To restore the failover timer to its default value of 5, use the no form of this command.
ss7 set [ session-set session-id ] failover-timer ft-value
no ss7 set [ session-set session-id ] failover-timer
Usage Guidelines
The failover-timer keyword and the ft-value argument specify the number of seconds that the Session Manager waits for the active session to recover or for the standby MGC to indicate that the SLT should switch traffic to the standby session and to make that session the active session. If the failover timer expires without recovery of the original session or if the system fails to get an active message from the standby MGC, the signaling links are taken out of service.
The no form of this command restores the failover timer to its default value of 5. Omitting the optional session-set keyword implicitly selects SS7 session set 0, which is the default.
Examples
The following example sets the failover timer to four seconds without using the session-set option:
ss7 set failover-timer 4The following example sets the failover timer to 10 seconds and sets the SS7 session set value to 5:
ss7 set session-set 5 failover-timer 10Related Commands
Command
Description
ss7 session
Creates a Reliable User Datagram Protocol (RUDP) session and explicitly adds an RUDP session to a Signaling System 7 (SS7) session set.
ss7 set failover timer
Specifies the amount of time that the Session Manager waits for the session to recover before declaring the session inactive.
ss7 set failover-timer
To specify the amount of time that the SS7 Session Manager waits for the active session to recover or for the standby Media Gateway Controller to indicate that the SLT should switch traffic to the standby session, use the ss7 set failover-timercommand in global configuration mode. To reset ti the default, use the no form of this command.
Usage Guidelines
This command specifies the number of seconds that the session manager waits for the active session to recover or for the standby media gateway controller to indicate that the SLT should switch traffic to the standby session and to make that session the active session. If the timer expires without a recovery of the original session or an active message from the standby media gateway controller, the signaling links are taken out of service.
station-id name
To specify the name that is to be sent as caller ID information and to enable caller ID, use the station-id name command in voice-port configuration mode at the sending Foreign Exchange Station (FXS) voice port or at a Foreign Exchange Office (FXO) port through which routed caller ID calls pass. To remove the name, use the no form of this command.
Usage Guidelines
This optional command is configured on FXS voice ports that are used to originate on-net calls. The information entered is displayed by the telephone attached to the FXS port at the far end of the on-net call. It can also be configured on the FXO port of a router on which caller ID information is expected to be received from the Central Office (CO), to suit situations in which a call is placed from the CO, then goes through the FXO interface, and continues to a far-end FXS port through an on-net call. In this case, if no caller ID information is received from the CO telephone line, the far-end call recipient receives the information configured on the FXO port.
NoteThis feature applies only to caller ID name display provided by an FXS port connection to a telephone device. The station-id name is not passed through telephone trunk connections supporting Automatic Number Identification (ANI) calls. ANI supplies calling number identification only and does not support calling number names.
Do not use this command when the caller ID standard is dual-tone multifrequency (DTMF). DTMF caller ID can carry only the calling number.
If the station-id name, station-id number, or a caller-id alertingcommand is configured on the voice port, caller ID is automatically enabled, and the caller-id enablecommand is not necessary.
station-id number
To specify the telephone or extension number that is to be sent as caller ID information and to enable caller ID, use the station-id number command in voice-port configuration mode at the sending Foreign Exchange Station (FXS) voice port or at a Foreign Exchange Office (FXO) port through which routed caller ID calls pass. To remove the number, use the no form of this command.
Usage Guidelines
This optional command is configured on FXS voice ports that are used to originate on-net calls. The information entered is displayed by the telephone attached to the FXS port at the far end of the on-net call. It can also be configured on the FXO port of a router on which caller ID information is expected to be received from the Central Office (CO), to suit situations in which a call is placed from the CO, then goes through the FXO interface, and continues to a far-end FXS port through an on-net call. In this case, if no caller ID information is received from the CO telephone line, the far-end call recipient receives the information configured on the FXO port.
Within the network, if an originating station-id does not include configured number information, Cisco IOS software determines the number by using reverse dial-peer search.
NoteThis feature applies only to caller ID name display provided by an FXS port connection to a telephone device. The station-id name is not passed through telephone trunk connections supporting Automatic Number Identification (ANI) calls. ANI supplies calling number identification only and does not support calling number names.
If the station-id name, station-id number, or a caller-id alertingcommand is configured on the voice port, caller ID is automatically enabled, and the caller-id enablecommand is not necessary.
stats
To enable statistics collection for voice applications, use the stats command in application configuration monitor mode. To reset to the default, use the no form of this command.
Usage Guidelines
To display the application statistics, use the show call application session-level, show call application app-level, or show call application gateway-levelcommand. To reset the application counters in history to zero, use the clear call application stats command.
Examples
The following example enables statistics collection for voice applications:
application monitor statsRelated Commands
Command
Description
call application interface stats
Enables statistics collection for application interfaces.
call application stats
Enables statistics collection for voice applications.
clear call application stats
Clears application-level statistics in history and subtracts the statistics from the gateway-level statistics.
clear call application stats
Clears application-level statistics in history and subtracts the statistics from the gateway-level statistics.
interface stats
Enables statistics collection for application interfaces.
show call application app-level
Displays application-level statistics for voice applications.
show call application gateway-level
Displays gateway-level statistics for voice application instances.
show call application session-level
Displays event logs and statistics for voice application instances.
stcapp
To enable the SCCP Telephony Control Application (STCAPP), use the stcapp command in global configuration mode. To disable the STCAPP, use the no form of this command.
Command Default
The Cisco CallManager does not control Cisco IOS gateway-connected analog and BRI endpoints.
Usage Guidelines
Use the stcappcommand to enable basic Skinny Client Call Control (SCCP) call control features for BRI and foreign exchange stations (FXS) analog ports within Cisco IOS voice gateways. The stcappcommand enables the Cisco IOS gateway application to support the following features:
- Line-side support for the Multilevel Precedence and Preemption (MLPP) feature
- Cisco CallManager registration of analog and Basic Rate Interface BRI endpoints
- Cisco CallManager endpoint autoconfiguration support
- Modem pass-through support
- Cisco Survivable Remote Site Telephony (SRST) support
Related Commands
Command
Description
ccm-manager config server
Specifies the TFTP server for SCCP gateway downloads.
ccm-manager sccp local
Specifies the SCCP local interface for Cisco CallManager registration.
sccp
Enables the SCCP protocol.
show stcapp device
Displays configuration information about STCAPP) voice ports.
show stcapp statistics
Displays call statistics for STCAPP voice ports.
stcapp ccm-group
Configures the Cisco CallManager group number for use by the STCAPP.
stcapp timer
Enables STCAPP timer configuration.
stcapp call-control mode
To configure call control mode for Skinny Client Control Protocol (SCCP) gateway supplementary features, use the stcapp call-control mode command in global configuration mode. To disable call control mode, use the no form of this command
Usage Guidelines
This command enables feature mode call control, which allows SCCP analog phone users to invoke a feature by dialing a feature access code (FAC). The following table lists the features and FACs that you can use in feature mode.
Examples
The following partial output from the show running-config command shows feature call control mode enabled:
Router# show running-config . . . stcapp call-control mode feature !The following partial output from the show running-config command shows standard call control mode enabled:
Router# show running-config . . . stcapp call-control mode standard ! !stcapp feature callback
To enable CallBack on Busy and enter the STC application feature callback configuration mode, use the stcapp feature callback command in global configuration mode. To disable the feature in the STC application, use the no form of this command.
Usage Guidelines
This command enables CallBack on Busy and enters the STC application feature callback configuration mode for modifying the default values of the callback activation key and timer for CallBack on Busy.
stcapp ccm-group
To configure the Cisco CallManager group number for use by the SCCP Telephony Control Application (STCAPP), use the stcapp ccm-group command in global configuration mode. To disable STCAPP Cisco CallManager group number configuration, use the no form of this command.
Usage Guidelines
The Cisco CallManager group identifier must have been configured for the service provider interface (SPI) using the sccp ccm-group group-idcommand.
Examples
The following example configures the STCAPP to use Cisco CallManager group 2:
Router(config)# stcapp ccm-group 2Related Commands
Command
Description
show stcapp device
Displays configuration information about SCCP Telephony Control Application (STCAPP) voice ports.
show stcapp statistics
Displays call statistics for SCCP Telephony Control Application (STCAPP) voice ports.
stcapp
Enables the SCCP Telephony Control Application (STCAPP).
stcapp timer
Enables SCCP Telephony Control Application (STCAPP) timer configuration.
stcapp feature access-code
To enable feature access codes (FACs) in the STC application and enter the STC application feature access-code configuration mode, use the stcapp feature access-code command in global configuration mode. To disable the use of all STC application feature access codes, use the no form of this command.
Usage Guidelines
This command enables feature access codes (FACs) in the SCCP telephony control (STC) application and enters the STC application feature access-code configuration mode to modify the default values of the prefix and feature codes for FACs.
The no form of this command blocks the use of FACs on all analog ports.
Use the show stcapp feature codes command to display a list of all FACs.
Examples
The following example shows how to enable FACs in the STC application.
Router(config)# stcapp feature access-code Router(stcapp-fac)#Related Commands
Command
Description
call forward all
Defines the feature code in the feature access code (FAC) for forwarding all calls.
call forward cancel
Defines the feature code in the feature access code (FAC) for cancelling Call Forward All.
pickup direct
Defines the feature code in the feature access code (FAC) for Directed Call Pickup.
pickup group
Defines the feature code in the feature access code (FAC) for call pickup from another group.
pickup local
Defines the feature code in the feature access code (FAC) for call pickup from the local group.
prefix (stcapp-fac)
Defines the prefix for feature access codes (FACs).
show stcapp feature codes
Displays all feature access codes (FACs).
stcapp feature callback
To enable CallBack on Busy and enter the STC application feature callback configuration mode, use the stcapp feature callback command in global configuration mode. To disable the feature in the STC application, use the no form of this command.
Usage Guidelines
This command enables CallBack on Busy and enters the STC application feature callback configuration mode for modifying the default values of the callback activation key and timer for CallBack on Busy.
stcapp feature speed-dial
To enable STC application feature speed-dial codes and enter their configuration mode, use the stcapp feature speed-dial command in global configuration mode. To disable the use of all STC application feature speed-dial codes, use the no form of this command.
Usage Guidelines
This command is used with the SCCP telephony control (STC) application, which enables certain features on analog FXS endpoints that use Skinny Client Control Protocol (SCCP) for call control.
Although feature speed-dial (FSD) prefixes and codes for analog FXS ports are configured on the voice gateway that has the FXS ports, the actual numbers that are dialed using these codes are configured on Cisco CallManager or the Cisco CallManager Express system.
The no form of this command blocks the use of FSD codes on all analog ports.
Note that all the STC FSD codes have defaults. To return codes under this configuration mode to their defaults, you must use the no form of the individual commands one at a time.
Examples
The following example sets an FSD prefix of three asterisks (***) and speed-dial codes from 2 to 7. After these values are configured, a phone user presses ***2 on the keypad to speed-dial the telephone number that is stored with speed-dial 1 on the call-control system (Cisco CallManager or Cisco CallManager Express).
Router(config)# stcapp feature speed-dial Router(stcapp-fsd)# prefix *** Router(stcapp-fsd)# speed dial from 2 to 7 Router(stcapp-fsd)# redial 9 Router(stcapp-fsd)# voicemail 8 Router(stcapp-fsd)# exitThe following example shows how the speed-dial range that is set in the example above is mapped to the speed-dial positions on the call-control system. Note that the range from 2 to 7 is mapped to speed-dial 1 to 6.
Router# show stcapp feature codes . . . stcapp feature speed-dial prefix *** redial ***9 voicemail ***8 speeddial1 ***2 speeddial2 ***3 speeddial3 ***4 speeddial4 ***5 speeddial5 ***6 speeddial6 ***7Related Commands
Command
Description
prefix (stcapp-fsd)
Designates a prefix to precede the dialing of an STC application feature speed-dial code.
redial
Designates an STC application feature speed-dial code to dial again the last number that was dialed.
show stcapp feature codes
Displays configured and default STC application feature codes.
speed dial
Designates a range of STC application feature speed-dial codes.
voicemail (stcapp-fsd)
Designates an STC application feature speed-dial code to dial the voice-mail number.
stcapp register capability
To specify modem capability for SCCP Telephony Control Application (STCAPP) devices, use the stcapp register capabilitycommand in global configuration mode. To disable modem capability, use the no form of this command.
stcapp register capability voice-port [ both | modem-passthrough | modem-relay ]
no stcapp register capability voice-port
Usage Guidelines
Use the stcapp register capability command to specify modem transport methods for STCAPP-controlled devices registering with Cisco Call-Manager. If this command is applied while the voice port is idle, the port automatically reregisters with the Cisco CallManager. If there is an active call on the voice port when this command is applied, the port does not reregister.Although Cisco does not recommend the procedure, to force device reregistration you must either manually shut down the device using the shutdown command in voice-port configuration mode, or reset it from the Cisco CallManager.
Use the voice service configuration command modem passthrough to globally enable modem pass-through capability, thereby providing fallback to voice band data (modem pass-through) when the voice gateway communicates with a Secure Telephone Unit (STU) or nonmodem-relay capable gateway.
stcapp security mode
To enable security for Skinny Client Control Protocol (SCCP) Telephony Control Application (STCAPP) endpoints and specify the security mode to be used for setting up the Transport Layer Security (TLS) connection, use the stcapp security mode command in global configuration mode. To disable security for the endpoint, use the no form of this command.
Syntax Description
mode
Sets the global security mode for all STCAPP endpoints.
authenticated
Sets the security mode to authenticated and enables SCCP signaling between the voice gateway and Cisco Unified CME through the secure TLS connection on TCP port 2443.
encrypted
Sets the security mode to encrypted and enables SCCP signaling between the voice gateway and Cisco Unified CME to take place through Secure Real-Time Transport Protocol (SRTP).
none
Sets the security mode to none (Default).
Usage Guidelines
You must enter both the stcapp security modeand stcapp security trustpoint commands to enable security for the STCAPP end point. The stcapp security trustpoint command must be configured for the STCAPP service to start.
SCCP signaling security mode can be set for each dial peer using the security mode command in dial peer configuration mode. If you use both the stcapp security mode and the security modecommands, the dial-peer level command, security mode, overrides the global setting.
Examples
The following example configures STCAPP security mode with the trustpoint mytrustpoint:
Router(config)# stcapp ccm-group 1 Router(config)# stcapp security mytrustpoint Router(config)# stcapp security mode encrypted Router(config)# stcappRelated Commands
Command
Description
security mode
Sets the security mode for a specific dial peer using STCAPP services in a secure Cisco Unified CME network.
stcapp
Enables the STCAPP.
stcapp ccm-group
Configures the Cisco Unified Communications Manager group number for use by the STCAPP.
stcapp security trustpoint
Enables security for STCAPP endpoints and specifies the trustpoint for setting up the TLS connection.
stcapp security trustpoint
To enable security for Skinny Client Control Protocol (SCCP) Telephony Control Application (STCAPP) endpoints and specify the trustpoint to be used for setting up the Transport Layer Security (TLS) connection, use the stcapp security command in global configuration mode. To disable security for the endpoint and delete all identity information and certificates associated with the trustpoint, use the no form of this command.
Usage Guidelines
You must enter both the stcapp security modeand stcapp security trustpoint commands to enable security for the STCAPP endpoint. The stcapp security trustpoint command must be configured for the STCAPP service to start. The trustpoint configured by this command contains the device certificate and must match the trustpoint configured on the router using the crypto pki trustpoint command. All analog phones use the same certificate. Cisco Unified Communications Manager Express does not require a different certificate for each phone.
Examples
The following example configures STCAPP security mode with the trustpoint mytrustpoint:
Router(config)# stcapp ccm-group 1 Router(config)# stcapp security mytrustpoint Router(config)# stcapp security mode encrypted Router(config)# stcappRelated Commands
Command
Description
crypto pki trustpoint
Declares the trustpoint that your router should use.
stcapp ccm-group
Configures the Cisco Unified Communications Manager group number for use by the STCAPP.
stcapp
Enables the STCAPP.
stcapp security mode
Enables security for STCAPP endpoints and specifies the security mode to be used for setting up the TLS connection.
stcapp supplementary-services
To enter supplementary-service configuration mode for configuring STC application supplementary-service features on an FXS port, use the stcapp supplementary-services command in global configuration mode. To remove the configuration, use the no form of this command.
Usage Guidelines
This command enters the supplementary-service configuration mode for configuring STC application supplementary-service features for analog FXS ports on a Cisco IOS voice gateway, such as a Cisco integrated services router (ISR) or Cisco VG224 Analog Phone Gateway.
Examples
The following example shows how to enable the Hold/Resume STC application supplementary-service feature for analog phones connected to port 2/0 on a Cisco VG224.
Router(config)# stcapp supplementary-services Router(config-stcapp-suppl-serv)# port 2/0 Router(config-stcapp-suppl-serv-port)# hold-resume Router(config-stcapp-suppl-serv-port)# endstcapp timer
To enable SCCP Telephony Control Application (STCAPP) timer configuration, use the stcapp timercommand in global configuration mode. To disable STCAPP timer configuration, use the no form of this command.
Usage Guidelines
Use this command to configure the STCAPP ROH timer for the maximum time that ROH tone is played. ROH tone signals a subscriber that the phone remains off hook when there is no active call.
stun
To enter STUN configuration mode for configuring firewall traversal parameters, use the stun command in voice-service voip configuration mode. To remove stun parameters, use the no form of this command.
Usage Guidelines
Use this command to enter the configuration mode to configure firewall traversal parameters for VoIP communications.
Examples
The following example shows how to enter STUN configuration mode.
Router(config)#voice service voip Router(config-voi-serv)#stunRelated Commands
Command
Description
stun flowdata agent-id
Configures the agent ID.
stun flowdata keepalive
Configures the keepalive interval.
stun flowdata shared-secret
Configures a secret shared between call control agent and firewall.
stun usage firewall-traversal flowdata
Enables firewall traversal using stun.
voice-class stun-usage
Enables firewall traversal for VoIP communications.
stun flowdata agent-id
To configure the stun flowdata agent ID, use the stun flowdata agent-idcommand in STUN configuration mode. To return to the default value for agent ID, use the no form of this command.
Usage Guidelines
Use the stun flowdata agent-idcommand to configure the agent id and the boot count to configure call control agents which authorize the flow of traffic.
Configuring the boot-count keyword helps to prevent anti-replay attacks after the router is reloaded. If you do not configure a value for boot count, the boot-count is initialized to 0 by default. After it is initialized, it is incremented by one automatically upon each reboot and the value saved back to NVRAM. The value of boot count is reflected in show running configuration command.
stun flowdata catlife
To configure the lifetime of the CAT, use the stun flowdata catlife command in STUN configuration mode. To return to the default catlife value, use the no form of this command.
stun flowdata catlife liftetime keepalive interval
no stun flowdata catlife liftetime keepalive interval
Usage Guidelines
Use the stun flowdata catlifecommand to configure call control agents which authorize the flow of traffic.
stun flowdata keepalive
NoteEffective with Cisco IOS Release 15.0(1)M, the stun flowdata keepalive command is replaced by the command stun flowdata catlife.
To configure the keepalive interval, use the stun flowdata keepalive command in STUN configuration mode. To return to the default keepalive value, use the no form of this command.
Usage Guidelines
You can use the stun flowdata keepalive command to decide how often to send keepalives. Keepalives are application mechanisms for maintaining alive the firewall traversal mappings associated with firewalls.
TRP works with a Call Agent which supports firewall traversal. In this mode, the Call Agent sends a request to TRP to open the pinhole. The request contains local, remote IP /Port, token, and other Cisco-flow data parameters.
TRP sends a STUN indication message to the firewall with Cisco-flow data, after processing the request. The message contains the STUN header, STUN username, and Cisco-flow data. The firewall validates the token in Cisco-flow data after receiving the STUN packet, and opens the pinhole if validation is successful.
Keepalives in STUN flow between the UDP peers to ensure that the firewall keeps the pinholes open.
This command is hidden and depreciated in Cisco IOS Release 15.0(1)M release because the keepalive interval is configured along with stun flowdata catlife command. When this command is configured or present in start-up configuration during reload, the following command will be nvgen’ed and displayed in show run command.
In addition, the following message will be printed during the configuration/reload:
Deprecated command. Setting catlife=1270 sec and keepalive=30 sec. Use the following command to configure non-default values: stun flowdata catlife <lifetime> keepalive <interval>Examples
The following example shows how to change the stun flowdata keepalive interval from the default value (10) to 5 seconds.
Router(config)# voice service voip Router(config-voi-serv)#stun Router(config-serv-stun)#stun flowdata agent-id 35 Router(config-serv-stun)#stun flowdata shared-secret 123abc123abc Router(config-serv-stun)#stun flowdata keepalive 5stun flowdata shared-secret
To configure a secret shared on a call control agent, use the stun flowdata shared-secret command in STUN configuration mode. To return the shared secret to the default value, use the no form of this command.
Command Default
The default value of this command sets the shared secret to an empty string. No firewall traversal is performed when the shared-secret has the default value.
Usage Guidelines
A shared secret on a call control agent is a string that is used between a call control agent and the firewall for authentication purposes. The shared secret value on the call control agent and the firewall must be the same. This is a string of 12 to 80 characters. The no form of this command will remove the previously configured shared-secret if any. The default form of this command will set the shared-secret to NULL. The password can be encrypted and validated before it is accepted. Firewall traversal is not performed when the shared-secret is set to default.
stun usage firewall-traversal flowdata
To enable firewall traversal using stun, use the stun usage firewall-traversal flowdata command in voice class stun-usage configuration mode. To disable firewall traversal with stun, use the no form of this command.
subaddress
To configure a subaddress for a POTS port, use the subaddress command in dial-peer voice configuration mode. To disable the subaddress, use the no form of this command.
Usage Guidelines
You can use this command for any dial-peer voice POTS port. You can configure only one subaddress for each of the POTS ports. The latest entered subaddress on the dial-peer voice port is stored. To check the status of the subaddress configuration, use the show running-config command.
Examples
The following examples show that a subaddress of 20 has been set for POTS port 1 and that a subaddress of 10 has been set for POTS port 2:
dial-peer voice 1 pots destination-pattern 5555555 port 1 no call-waiting ring 0 volume 4 caller-number 1111111 ring 3 caller-number 2222222 ring 1 caller-number 3333333 ring 1 subaddress 20 dial-peer voice 2 pots destination-pattern 4444444 port 2 no call-waiting ring 0 volume 2 caller-number 6666666 ring 2 caller-number 7777777 ring 3 subaddress 10subcell-mux
To enable ATM adaption layer 2 (AAL2) common part sublayer (CPS) subcell multiplexing on a Cisco router, use the subcell-mux command in voice-service configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
Use thiscommand to enable ATM adaption layer 2 (AAL2) common part sublayer (CPS) subcell multiplexing when the Cisco router interoperates with other equipment that uses subcell multiplexing.
subscription asnl session history
To specify how long to keep Application Subscribe/Notify Layer (ASNL) subscription history records and how many history records to keep in memory, use the subscription asnl session history command in global configuration mode. To reset to the default, use the no form of this command.
subscription asnl session history { count number | duration minutes }
no subscription asnl session history { count | duration }
Usage Guidelines
The ASNL layer maintains subscription information. Active subscriptions are retained in the active subscription table in system memory. When subscriptions are terminated, they are moved to the subscription table in system memory.
This command controls the ASNL history table. Use this command to specify how many minutes the history record is retained after the subscription is removed, and to specify how many records are retained at any given time.
Examples
The following example specifies that a total of 100 records are to be kept in the RTSP client history:
subscription asnl session history count 100Related Commands
Command
Description
clear subscription
Clears all active subscriptions or a specific subscription.
debug asnl events
Traces event logs in the ASNL.
show subscription
Displays information about ASNL-based and non-ASNL-based SIP subscriptions.
subscription maximum
Specifies the maximum number of outstanding subscriptions to be accepted or originated by a gateway.
subscription maximum
To specify the maximum number of outstanding subscriptions to be accepted or originated by a gateway, use the subscription maximum command in voice service voip sip configuration mode. To remove the maximum number of subscriptions specified, use the no form of this command.
Command Default
The default number of subscriptions is equal to twice the number of dial-peers configured on the platform.
Usage Guidelines
Use this command to configure the maximum number of concurrent SIP subscriptions, up to twice the number of dial-peers configured.
Examples
The following example configures subscription maximums:
Router(config)# voice service voip Router(conf-voi-serv)# sip Router(conf-serv-sip)# subscription maximum originate 10supervisory answer dualtone
To enable answer supervision on a Foreign Exchange Office (FXO) voice port, use the supervisory answer dualtone command invoice-port configuration mode. To disable answer supervision on a voice port, use the no form of this command.
Usage Guidelines
This command configures the FXO voice port to detect voice, fax, and modem traffic when calls are answered. If answer supervision is enabled, calls are not recorded as connected until answer supervision is triggered.
This command enables a ring-no-answer timeout that drops calls after a specified period of ringback. The period of ringback can be configured using the timeouts ringing command.
This command automatically enables disconnect supervision in the preconnect mode on the voice port if disconnect supervision is not already enabled with the supervisory disconnect dualtonecommand.
This command is applicable to analog FXO voice ports with loop-start signaling.
If false answering is detected, decrease the sensitivity setting. If answering detection is failing, increase the sensitivity setting.
Examples
The following example enables answer supervision on voice port 0/1/1:
voice-port 0/1/1 supervisory answer dualtoneRelated Commands
Command
Description
supervisory custom-cptone
Associates a class of custom call-progress tones with a voice port.
supervisory disconnect dualtone
Enables disconnect supervision on an FXO voice port.
timeouts ringing
Specifies the time that the calling voice port allows ringing to continue if a call is not answered.
voice class custom-cptone
Creates a voice class for defining custom call-progress tones.
voice class dualtone-detect-params
Modifies the frequency, power, and cadence tolerances of call-progress tones.
supervisory custom-cptone
To associate a class of custom call-progress tones with a voice port, use the supervisory custom-cptone command invoice-port configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
This command associates a class of custom call-progress tones, defined by the voice class custom-cptone command, with a voice port.
You can associate the same custom call-progress tones to multiple voice ports.
You can associate only one class of custom call-progress tones with a voice port. If you associate a second class of custom call-progress tones with a voice port, the second class of custom tones replaces the one previously assigned.
This command is applicable to analog Foreign Exchange Office (FXO) voice ports with loop-start signaling.
Examples
The following example associates the class of custom call-progress tones named country-x with voice ports 1/4 and 1/5:
voice-port 1/4 supervisory custom-cptone country-x exit voice-port 1/5 supervisory custom-cptone country-x exitRelated Commands
Command
Description
dualtone
Defines a call-progress tone to be detected.
supervisory answer dualtone
Enables answer supervision on an FXO voice port.
supervisory disconnect dualtone
Enables disconnect supervision on an FXO voice port.
voice class custom-cptone
Creates a voice class for defining custom call-progress tones.
supervisory disconnect
To enable a supervisory disconnect signal on Foreign Exchange Office (FXO) ports, use the supervisory disconnect command in voice-port configuration mode. To disable the signal, use the no form of this command.
Usage Guidelines
This command indicates whether supervisory disconnect signaling is available on the FXO port. Supervisory disconnect signaling is a power denial from the switch lasting at least 350 ms. When this condition is detected, the system interprets this as a disconnect indication from the switch and clears the call.
You should configure no supervisory disconnect on the voice port if there is no supervisory disconnect available from the switch.
NoteIf there is no disconnect supervision on the voice port, the interface could be left active if the caller abandons the call before the far end answers. After the router collects the dialed digits but before the called party answers, the router starts a tone detector. Within this time window, the tone detector listens for signals (such as a fast busy signal) that occur if the originating caller hangs up. If this occurs, the router interprets those tones as a disconnect indication and closes the window.
supervisory disconnect anytone
To configure a Foreign Exchange Office (FXO) voice port to go on-hook if the router detects any tone from a PBX or the PSTN before an outgoing call is answered, use the supervisory disconnect anytone command invoice-port configuration mode. To disable the supervisory disconnect function, use the no form of this command.
Usage Guidelines
Use this command to provide disconnect if the PBX or PSTN does not provide a supervisory tone. Examples of tones that trigger a disconnect include busy tone, fast busy tone, and dial tone.
This command is enabled only during call setup (before the call is answered).
You must enable echo cancellation; otherwise, ringback tone from the router can trigger a disconnect.
This command replaces the no supervisory disconnect signalcommand. If you enter thiscommand, the supervisory disconnect anytone feature is enabled, and the message supervisory disconnect anytoneis displayed when show commands are entered.
If you enter either the supervisory disconnect anytonecommand or the no supervisory disconnect signalcommand, answer supervision is automatically disabled.
Examples
The following example configures voice ports 1/4 and 1/5 to go on-hook if any tone from the PBX or PSTN is detected before the call is answered:
voice-port 1/4 supervisory disconnect anytone exit voice-port 1/5 supervisory disconnect anytone exitThe following example disables the disconnect function on voice port 1/5:
voice-port 1/5 no supervisory disconnect anytone exitRelated Commands
Command
Description
supervisory answer dualtone
Enables answer supervision on an FXO voice port.
supervisory disconnect dualtone
Enables disconnect supervision on an FXO voice port.
timeouts call-disconnect
Specifies the timeout value for releasing an FXO voice port when an incoming call is not answered.
supervisory disconnect dualtone
To enable disconnect supervision on a Foreign Exchange Office (FXO) voice port, use the supervisory disconnect dualtone command invoice-port configuration mode. To disable the supervisory disconnect function, use the no form of this command.
Syntax Description
mid -call
Disconnect supervision operates throughout the duration of the call.
pre -connect
Disconnect supervision operates during call setup and stops when the called telephone goes off-hook.
Disconnect supervision is not enabled on voice ports.
Usage Guidelines
This command configures an FXO voice port to disconnect calls when the router detects call-progress tones from a PBX or the PSTN. Disconnection occurs after the wait-release time specified on the voice port.
Disconnect supervision is automatically enabled in the preconnect mode on the voice port if the supervisory answer dualtonecommand is entered.
This feature is applicable to analog FXO voice ports with loop-start signaling.
Examples
The following example specifies tone detection during the entire call duration:
voice-port 1/5 supervisory disconnect dualtone mid-call exitThe following example specifies tone detection only during call setup:
voice-port 0/1/1 supervisory disconnect dualtone pre-connect exitRelated Commands
Command
Description
supervisory answer dualtone
Enables answer supervision on an FXO voice port.
supervisory custom-cptone
Associates a class of custom call-progress tones with a voice port.
timeouts call-disconnect
Specifies the timeout value for releasing an FXO voice port when an incoming call is not answered.
timeouts wait-release
Specifies the timeout value for releasing a voice port when an outgoing call is not answered.
voice class dualtone-detect-params
Modifies the frequency, power, and cadence tolerances of call-progress tones.
supervisory disconnect dualtone voice-class
To assign a previously configured voice class for Foreign Exchange Office (FXO) supervisory disconnect tone to a voice port, use the supervisory disconnect dualtone voice-class command in voice port configuration mode. To remove a voice class from a voice-port, use the no form of this command.
supervisory disconnect dualtone { mid-call | pre-connect } voice-class tag
no supervisory disconnect dualtone voice-class tag
Syntax Description
mid -call
Tone detection operates throughout the duration of a call.
pre -connect
Tone detection operates during call setup and stops when the called telephone goes off-hook.
tag
Unique identification number assigned to one voice class. The tag number maps to the tag number assigned using the voice class dualtone global configuration command. Range is from 1 to 10000.
Usage Guidelines
You can apply an FXO supervisory disconnect tone voice class to multiple voice ports. You can assign only one FXO supervisory disconnect tone voice class to a voice port. If a second voice class is assigned to a voice port, the second voice class replaces the one previously assigned. You cannot assign separate FXO supervisory disconnect tone commands directly to the voice port.
This feature is applicable to analog FXO voice ports with loop-start signaling.
Examples
The following example assigns voice class 70 to FXO voice port 0/1/1 and specifies tone detection during the entire call duration:
voice-port 0/1/1 no echo-cancel enable supervisory disconnect dualtone mid-call voice-class 70The following example assigns voice class 80 to FXO voice port 0/1/1 and specifies tone detection only during call setup:
voice-port 0/1/1 no echo-cancel enable supervisory disconnect dualtone pre-connect voice-class 80supervisory disconnect lcfo
To enable a supervisory disconnect signal on an FXS port, use the supervisory disconnect lcfo command in voice-port configuration mode. To disable the signal, use the no form of this command.
Usage Guidelines
This command enables a disconnect indication by triggering a power denial using a loop current feed open (LCFO) signal on FXS ports with loop-start signaling. Third-party devices, such as an interactive voice response (IVR) system, can detect a disconnect and clear the call when it receives the power denial signal. To disable the power denial during the disconnect stage, use the no supervisory disconnect lcfo command. The duration of the power denial is set with the timeouts power-denial command.
supervisory dualtone-detect-params
To associate a class of modified tone-detection tolerance limits with a voice port, use the supervisory dualtone-detect-params command invoice-port configuration mode. To reset to the default, use the no form of this command.
Usage Guidelines
This command associates a specific set of modified tone-detection tolerance limits, defined by the voice class dualtone-detect-paramscommand, with a voice port.
You can associate the same class of modified tone-detection tolerance limits to multiple voice ports.
You can associate only one class of modified tone-detection tolerance limits to a voice port. If you associate a second class of modified tone-detection tolerance limits with a voice port, the second class replaces the one previously assigned.
This command is applicable to analog Foreign Exchange Office (FXO) voice ports with loop-start signaling.
Examples
The following example associates the class of modified tone-detection tolerance limits that has tag 70 with voice ports 1/5 and 1/6.
voice-port 1/5 supervisory dualtone-detect-params 70 exit voice-port 1/6 supervisory dualtone-detect-params 70 exitThe following example restores the default tone-detection parameters to voice port 1/5.
voice-port 1/5 no supervisory dualtone-detect-params exitsupervisory sit us
To provide detection of eight standard U.S. special information tones (SITs) and certain nonstandard tones (including the AT&T SIT), and to report the detected tone with a preassigned disconnect cause code for disconnect supervision on a Foreign Exchange Office (FXO) voice port, use the supervisory sit uscommand in voice-port configuration mode. To turn off the detection and disconnect activity, use the no form of this command.
Syntax Description
all-tones
(Optional) Disconnects the call when a SIT or a nonstandard tone is detected.
tone-selector
(Optional) Defines a specific response for call-disconnect when a standard SIT or a nonstandard tone is detected on the incoming or outgoing call.
value
Acceptable values are 0, 1, 2, or 3:
- 0--Detection of a standard SIT drops the call, but an AT&T SIT or a nonstandard tone does not cause a disconnect.
- 1--Detection of either a standard SIT or nonstandard tone drops the call, but the AT&T SIT does not cause a disconnect.
- 2--Detection of a standard SIT or an AT&T SIT results in a call disconnect, but any other nonstandard tone does not cause a disconnect.
- 3--Detection of a standard SIT, AT&T SIT, or another nonstandard tone results in a disconnect.
immediate-release
(Optional) Disconnects the call immediately when a SIT is detected on the incoming or outgoing call. Nonstandard tones are ignored.
Command Default
No detection or disconnect occurs for the eight standard U.S. SITs, nonstandard tones, or the AT&T SIT on the FXO voice port for incoming and outgoing calls.
Usage Guidelines
This command configures an FXO voice port to detect and disconnect calls when the router detects call-progress tones from a PBX or the PSTN.
Prior to Cisco IOS Release 12.4(24)T, this command specifically detected eight standard U.S. SITs, but not nonstandard tones or the AT&T SIT. Beginning in Cisco IOS Release 12.4(24)T, the tone-selectorvalue option can be configured to detect nonstandard tones played by the service provider when the called number is invalid.
Disconnection occurs after the wait-release time specified on the voice port. Calls are disconnected immediately after a SIT is detected from the PSTN when the immediate-release keyword is configured. To configure the delay timeout before the system starts the process for releasing voice ports, use the timeouts wait-releasecommand on the voice port.
The SIT reporting complies with standard Q.850 messages in order for fax servers to uniquely identify each condition. This capability is supported for analog FXO trunk and T1/E1 channel-associated signaling (CAS) FXO loop-start.
NoteThe SIT detection and reporting feature enabled by the supervisory sit us command is supported on c5510 and LSI digital signal processors (DSPs). No other DSPs support this feature.
The table below identifies eight standard U.S. SITs and their associated disconnect cause codes.
NoteThese eight tones are referred to as standard tones based on the tone frequencies and durations shown in the table. These tones are defined in the Telcordia Technologies specification GR-1162-CORE (which is specific to North America). There are other nonstandard SITs that can occur. The AT&T SIT is one of the more common examples of the other variations. The nonstandard SITs can have durations and frequencies comparable to the nominal values for the eight tone segments shown in the table below or the nonstandard SITs can deviate significantly from these nominal values. The supervisory sit us command has been modified in Cisco IOS Release 12.4(24)T to provide flexibility in handling these variations.
Table 10 Eight U.S. SITs and Associated Disconnect Cause Codes Name
First Tone (Hz)
ms
Second Tone (Hz)
ms
Third Tone (Hz)
ms
Disconnect Cause Code
IC
913.8
274
1370.6
274
1776.7
380
8
VC
985.2
380
1370.6
274
1776.7
380
1
RO
985.2
274
1370.6
380
1776.7
380
86
RO
913.8
274
1428.5
380
1776.7
380
86
NC
913.8
380
1370.6
380
1776.7
380
34
NC
985.2
380
1428.5
380
1776.7
380
34
#1
913.8
380
1428.5
274
1776.7
274
21
#2
985.2
274
1428.5
274
1776.7
380
21
Examples
The following example shows how to enable SIT detection for the eight standard U.S. tones and provide for immediate disconnect on the voice port:
Router# configure terminal Router(config)# voiceport 1/0/1 Router(config-voiceport)# supervisory sit us immediate-releaseThe following example shows how to enable SIT detection for all eight standard U.S. tones and configure the delay timeout for 10 seconds:
Router# configure terminal Router(config)# voiceport 1/0/1 Router(config-voiceport)# supervisory sit us Router(config-voiceport)# timeouts wait-release 10The following example shows how to enable detection for a standard SIT or the AT&T SIT and to provide for immediate disconnect on the voice port (in this case, a nonstandard SIT does not cause a disconnect):
Router# configure terminal Router(config)# voiceport 1/0/1 Router(config-voiceport)# supervisory sit us tone-selector 2 immediate-releasesupplementary-service h225-notify cid-update (dal peer)
To enable individual dial peers to send H.225 messages with caller-ID updates, use the supplementary-service h225-notify cid-update command in dal peer configuration mode. To disable the sending of H.225 messages with caller-ID updates, use the no form of this command.
Usage Guidelines
This command specifies that an individual dial peer should provide caller ID updates through H.225 notify messages when a call is transferred or forwarded between Cisco CallManager Express and Cisco CallManager systems. The default is that this behavior is enabled. The no form of the command disables caller-ID updates, which is not recommended. Use the supplementary-service h225-notify cid-update command in voice-service configuration mode to specify this capability globally.
If this command is enabled globally and enabled on a dial peer, the functionality is enabled for that dial peer. This is the default.
If this command is enabled globally and disabled on a dial peer, the functionality is disabled for that dial peer.
If this command is disabled globally and either enabled or disabled on a dial peer, the functionality is disabled for that dial peer.
Examples
The following example globally enables the sending of H.225 messages to transmit caller-ID updates and then disables that capability on dial peer 24.
Router(config)# voice service voip Router(config-voi-serv)# supplementary-service h225-notify cid-update Router(config-voi-serv)# exit Router(config)# dial-peer voice 24 voip Router(config-dial-peer)# no supplementary-service h225-notify cid-update Router(config-dial-peer)# exitsupplementary-service h225-notify cid-update (voice-service)
To globally enable the sending of H.225 messages with caller-ID updates, use the supplementary-service h225-notify cid-update command in voice-service configuration mode. To disable the sending of H.225 messages with caller-ID updates, use the no form of this command.
Usage Guidelines
This command globally provides caller ID updates through H.225 notify messages when a call is transferred or forwarded between Cisco CallManager Express and Cisco CallManager systems. The default is that this behavior is enabled. The no form of the command disables caller-ID updates, which is not recommended. Use the supplementary-service h225-notify cid-update command in dial-peer configuration mode to specify this capability for individual dial peers.
If this command is enabled globally and enabled on a dial peer, the functionality is enabled for that dial peer. This is the default.
If this command is enabled globally and disabled on a dial peer, the functionality is disabled for that dial peer.
If this command is disabled globally and either enabled or disabled on a dial peer, the functionality is disabled for that dial peer.
Examples
The following example globally enables the sending of H.225 messages to transmit caller-ID updates and then disables that capability on dial peer 24.
Router(config)# voice service voip Router(config-voi-serv)# supplementary-service h225-notify cid-update Router(config-voi-serv)# exit Router(config)# dial-peer voice 24 voip Router(config-dial-peer)# no supplementary-service h225-notify cid-update Router(config-dial-peer)# exitsupplementary-service h450.2 (dial peer)
To enable H.450.2 supplementary services capabilities exchange for call transfers across a VoIP network for an individual dial peer, use the supplementary-service h450.2 command in dial peer configuration mode. To disable H.450.2 capabilities for an individual dial peer, use the no form of this command.
Usage Guidelines
This command specifies the use of the H.450.2 standard protocol for call transfers across a VoIP network for the calls handled by an individual dial peer. Use the supplementary-service h450.2 command in voice-service configuration mode to specify H.450.2 capabilities at a global level.
If this command is enabled globally and enabled on a dial peer, the functionality is enabled for the dial peer. This is the default.
If this command is enabled globally and disabled on a dial peer, the functionality is disabled for the dial peer.
If this command is disabled globally and either enabled or disabled on a dial peer, the functionality is disabled for the dial peer.
Examples
The following example disables H.450.2 services for dial peer 37.
Router(config)# dial-peer voice 37 voip Router(config-dial-peer)# destination-pattern 555.... Router(config-dial-peer)# session target ipv4:10.5.6.7 Router(config-dial-peer)# no supplementary-service h450.2 Router(config-dial-peer)# exitsupplementary-service h450.2 (voice-service)
To globally enable H.450.2 supplementary services capabilities exchange for call transfers across a VoIP network, use the supplementary-service h450.2command in voice-service configuration mode. To disable H.450.2 capabilities globally, use the no form of this command.
Usage Guidelines
This command specifies global use of the H.450.2 standard protocol for call transfers for all calls across a VoIP network. Use the no supplementary-service h450.2 command in dial-peer configuration mode to disable H.450.2 capabilities for individual dial peers.
If this command is enabled globally and enabled on a dial peer, the functionality is enabled for the dial peer. This is the default.
If this command is enabled globally and disabled on a dial peer, the functionality is disabled for the dial peer.
If this command is disabled globally and either enabled or disabled on a dial peer, the functionality is disabled for the dial peer.
supplementary-service h450.3 (dial peer)
To enable H.450.3 supplementary services capabilities exchange for call forwarding across a VoIP network for an individual dial peer, use the supplementary-service h450.3command in dial peer configuration mode. To disable H.450.3 capabilities for an individual dial peer, use the no form of this command.
Usage Guidelines
This command specifies use of the H.450.3 standard protocol for call forwarding for calls handled by an individual dial peer. Use the supplementary-service h450.3 command in voice-service configuration mode to specify H.450.3 capabilities at a global level.
If this command is enabled globally and enabled on a dial peer, the functionality is enabled for the dial peer. This is the default.
If this command is enabled globally and disabled on a dial peer, the functionality is disabled for the dial peer.
If this command is disabled globally and either enabled or disabled on a dial peer, the functionality is disabled for the dial peer.
Examples
The following example disables H.450.3 capabilities for dial peer 37.
Router(config)# dial-peer voice 37 voip Router(config-dial-peer)# destination-pattern 555.... Router(config-dial-peer)# session target ipv4:10.5.6.7 Router(config-dial-peer)# no supplementary-service h450.3 Router(config-dial-peer)# exitsupplementary-service h450.3 (voice-service)
To globally enable H.450.3 supplementary services capabilities exchange for call forwarding across a VoIP network, use the supplementary-service h450.3 command in voice-service configuration mode. To disable H.450.3 capabilities globally, use the no form of this command.
Usage Guidelines
This command specifies global use of the H.450.3 standard protocol for call forwarding across a VoIP network. Use the no supplementary-service h450.3command in dial-peer configuration mode to disable H.450.3 capabilities for individual dial peers.
If this command is enabled globally and enabled on a dial peer, the functionality is enabled for the dial peer. This is the default.
If this command is enabled globally and disabled on a dial peer, the functionality is disabled for the dial peer.
If this command is disabled globally and either enabled or disabled on a dial peer, the functionality is disabled for the dial peer.
supplementary-service h450.7
To globally enable H.450.7 supplementary services capabilities exchange for message-waiting indication (MWI) across a VoIP network, use the supplementary-service h450.7 command in voice-service or dial-peer configuration mode. To return to the default, use the no form of this command.
Command Modes
Voice service configuration (config-voi-serv)
Dial-peer configuration (config-dial-peer)Usage Guidelines
Use this command when you are implementing QSIG supplementary service features that use the H.450.7 standard.
Use this command in voice-service configuration mode to affect all dial peers globally. Use this command in dial-peer configuration mode to affect an individual dial peer:
If the supplementary-service h450.7 command is not in use, the services are globally disabled by default.
If the supplementary-service h450.7 command is not in use in voice-service configuration mode, you can use this command in dial-peer configuration mode to enable the services on individual dial peers.
If the supplementary-service h450.7 command is in use in voice-service configuration mode, the services are globally enabled and you cannot disable the services on individual dial peers.
supplementary-service h450.12 (dial peer)
To enable H.450.12 supplementary services capabilities exchange for call transfers across a VoIP network for an individual dial peer, use the supplementary-service h450.12 command in dial peer configuration mode. To disable H.450.12 capabilities for an individual dial peer, use the no form of this command.
Usage Guidelines
This command specifies use of the H.450.12 standard protocol for call transfers across a VoIP network for calls handled by an individual dial peer. Use the supplementary-service h450.12 command in voice-service configuration mode to specify H.450.12 capabilities at a global level.
If this command is enabled globally and enabled on a dial peer, the functionality is enabled for the dial peer.
If this command is enabled globally and disabled on a dial peer, the functionality is enabled for the dial peer.
If this command is disabled globally and enabled on a dial peer, the functionality is enabled for the dial peer.
If this command is disabled globally and disabled on a dial peer, the functionality is disabled for the dial peer. This is the default.
Examples
The following example enables H.450.12 capabilities on dial peer 37.
Router(config)# dial-peer voice 37 voip Router(config-dial-peer)# destination-pattern 555.... Router(config-dial-peer)# session target ipv4:10.5.6.7 Router(config-dial-peer)# supplementary-service h450.12 Router(config-dial-peer)# exitsupplementary-service h450.12 (voice-service)
To globally enable H.450.12 supplementary services capabilities exchange for call transfers across a VoIP network, use the supplementary-service h450.12command in voice-service configuration mode. To disable H.450.12 capabilities globally, use the no form of this command.
Usage Guidelines
The H.450.12 standard provides a means to advertise and discover H.450.2 call transfer and H.450.3 call forwarding capabilities in voice gateway endpoints on a call-by-call basis. When H.450.12 is enabled, use of H.450.2 and H.450.3 standards is disabled for call transfers and call forwards unless a positive H.450.12 indication is received from all the other VoIP endpoints involved in the call. If positive H.450.12 indications are received, the router uses the H.450.2 standard for call transfers and the H.450.3 standard for call forwarding. If a positive H.450.12 indication is not received, the router uses the alternative method that you have configured for call transfers and forwards, which, for Cisco CallManager Express (Cisco CME) 3.1 systems, may be either hairpin call routing or an H.450 tandem gateway. This command is useful when you have a mixed network with some endpoints that support H.450.2 and H.450.3 standards and other endpoints that do not support those standards.
This command specifies the global use of the H.450.12 standard protocol for all calls across a VoIP network. Use the supplementary-service h450.12 command in dial-peer configuration mode to specify H.450.12 capabilities for individual dial peers.
If this command is enabled globally and enabled on a dial peer, the functionality is enabled for the dial peer.
If this command is enabled globally and disabled on a dial peer, the functionality is enabled for the dial peer.
If this command is disabled globally and enabled on a dial peer, the functionality is enabled for the dial peer.
If this command is disabled globally and disabled on a dial peer, the functionality is disabled for the dial peer. This is the default.
Use the advertise-only keyword on a Cisco CME 3.1 system when you have only Cisco CME 3.0 systems in your network in addition to Cisco CME 3.1 systems. Cisco CME 3.0 systems can use H.450.2 and H.450.3 standards, but are unable to respond to H.450.12 queries. The advertise-only keyword enables a Cisco CME 3.1 system to bypass the requirement that a system respond to an H.450.12 query in order to use H.450.2 and H.450.3 standards for transferring and forwarding calls.
Examples
The following example enables H.450.12 capabilities at a global level.
Router(config)# voice service voip Router(config-voi-serv)# supplementary-service h450.12 Router(config-voi-serv)# exitThe following example enables H.450.12 capabilities at a global level in advertise-only mode on a Cisco CME 3.1 system to enable call transfers using the H.450.2 standard and call forwards using the H.450.3 standard with Cisco CME 3.0 systems in the network.
Router(config)# voice service voip Router(config-voi-serv)# supplementary-service h450.12 advertise-only Router(config-voi-serv)# exitsupplementary-service media-renegotiate
To globally enable midcall media renegotiation for supplementary services, use the supplementary-service media-renegotiate command in voice-service configuration mode. To disable midcall media renegotiation for supplementary services, use the no form of this command.
Usage Guidelines
This command enables midcall media renegotiation, or key renegotiation, for all calls across a VoIP network. To implement media encryption, the two endpoints controlled by Cisco Unified Communications Manager Express (Cisco Unified CME) need to exchange keys that they will use to encrypt and decrypt packets. Midcall key renegotiation is required to support interoperation and supplementary services among multiple VoIP suites in a secure media environment using Secure Real-Time Transport Protocol (SRTP).
NoteThe video part of a video stream will not play if the supplementary-service media-renegotiate command is configured in voice-service configuration mode.
supplementary-service qsig call-forward
To specify that calls are using QSIG and require supplementary services for call forwarding, use the supplementary-service qsig call-forwardcommand in voice-service or dial-peer configuration mode. To return to the default, use the no form of this command.
Command Modes
Voice service configuration (config-voi-serv)
Dial-peer configuration (dial-peer-config)Usage Guidelines
This command provides QSIG call-forwarding supplementary services (ISO 13873) when necessary to forward calls to another number.
Use this command in voice-service configuration mode, which is enabled by the voice service potscommand, to affect all POTS dial peers globally. Use this command in dial-peer configuration mode, which is enabled by the dial-peer voice command, to affect a single POTS dial peer.
If you are not using the supplementary-service qsig call-forwardcommand, the services are globally disabled by default.
If you are not using the supplementary-service qsig call-forward command in voice-service configuration mode, you can use this command in dial-peer configuration mode to enable the services on individual POTS dial peers.
If you are using the supplementary-service qsig call-forward command in voice-service configuration mode, this feature is globally enabled and you cannot disable the services on individual POTS dial peers.
Examples
The following example shows how to enable QSIG call-forwarding treatment for all POTS calls:
Router(config)# voice service pots Router(conf-voi-serv)# supplementary-service qsig call-forwardThe following example shows how to enable QSIG call-forwarding treatment for calls on POTS dial-peer 23:
Router(config)# dial-peer voice 23 pots Router(config-dial-peer)# supplementary-service qsig call-forwardsupplementary-service sip
To enable SIP supplementary service capabilities for call forwarding and call transfers across a SIP network, use the supplementary-service sip command in dial peer voice or voice service VOIP configuration mode. To disable supplementary service capabilities, use the no form of this command.
supplementary-service sip { handle-replaces | moved-temporarily | refer }
no supplementary-service sip { handle-replaces | moved-temporarily | refer }
Command Modes
Dial peer voice configuration (config-dial-peer)
Voice service configuration (conf-voi-serv)
Command History
Release
Modification
12.4(11)XJ
This command was introduced.
12.4(15)T
This command was integrated into Cisco IOS Release 12.4(15)T.
15.2(2)T1
This command was modified. The handle-replaces keyword was introduced.
15.3(1)T
This command was modified. With CSCub47586, if an INVITE (incoming call or incoming forward) with a diversion header is received while the no supplementary-service sip moved-temporarily form of this command is enabled, on either an inbound call leg or an outbound call leg, the call is disconnected.
Usage Guidelines
The supplementary-service sip refer command enables REFER message pass-through on a router.
The no form of the supplementary-service sip command allows you to disable a supplementary service feature (call forwarding or call transfer) if the destination gateway does not support the supplementary service. You can disable the feature either globally or for a specific SIP trunk (dial peer).
- The no supplementary-service sip handle-replaces command replaces the Dialog-ID in the Replaces Header with the peer Dialog-ID.
- The no supplementary-service sip moved-temporarily command prevents the router from sending a redirect response to the destination for call forwarding. SDP Passthrough is not supported in 302-consumption mode or Refer-consumption mode. With CSCub47586, if an INVITE (incoming call or incoming forward) with a diversion header is received while SDP Pass through is enabled on either an inbound call leg or an outbound call leg, the call is disconnected.
- The no supplementary-service sip refer command prevents the router from forwarding a REFER message to the destination for call transfers. The router instead attempts to initiate a hairpin call to the new target.
If this command is enabled globally and disabled on a dial peer, the functionality is disabled for the dial peer.
If this command is disabled globally and either enabled or disabled on a dial peer, the functionality is disabled for the dial peer.
On Cisco Unified Communications Manager Express (CME), this command is supported for calls between SIP phones and for calls between SCCP phones. It is not supported for a mixture of SCCP and SIP phones; for example, it has no effect for calls from an SCCP phone to a SIP phone. On the Cisco UBE, this command is supported for SIP trunk-to-SIP trunk calls.
Examples
The following example shows how to disable SIP call transfer capabilities for dial peer 37:
Device(config)# dial-peer voice 37 voip Device(config-dial-peer)# destination-pattern 555.... Device(config-dial-peer)# session target ipv4:10.5.6.7 Device(config-dial-peer)# no supplementary-service sip referThe following example shows how to disable SIP call forwarding capabilities globally:
Device(config)# voice service voip Device(conf-voi-serv)# no supplementary-service sip moved-temporarilyThe following example shows how to enable a REFER message pass-through on the Cisco UBE globally and how to disable the Refer-To header modification:
Device(config)# voice service voip Device(conf-voi-serv)# supplementary-service sip refer Device(conf-voi-serv)# sip Device(conf-serv-sip)# referto-passingThe following example shows how to enable a REFER message consumption on the Cisco UBE globally:
Device(config)# voice service voip Device(conf-voi-serv)# no supplementary-service sip referThe following example shows how to enable REFER message consumption on the Cisco UBE for dial peer 22:
Device(config)# dial-peer voice 22 voip Device(config-dial-peer)# no supplementary-service sip referThe following example shows how to enable a REFER message to replace the Dialog-ID in the Replaces Header with the peer Dialog-ID on the Cisco UBE for dial peer:
Device(config)# dial-peer voice 34 voip Device(config-dial-peer)# no supplementary-service sip handle-replaces [system]The following example shows how to enable a REFER message to replace the Dialog-ID in the Replaces Header with the peer Dialog-ID on the Cisco UBE globally:
Device(config)# voice service voip Device(conf-voi-serv)# no supplementary-service sip handle-replacesRelated Commands
Command
Description
supplementary-service h450.2 (voice-service)
Globally enables H.450.2 capabilities for call transfer.
supplementary-service h450.3 (voice-service)
Globally enables H.450.3 capabilities for call forwarding.
referto-passing
Disables dial peer lookup and modification of the Refer-To header while passing across REFER message on the Cisco UBE during a call transfer.
supported language
To configure Session Initiation Protocol (SIP) Accept-Language header support, use the supported-languagecommand in voice service or dial-peer voice configuration mode. To disable Accept-Language header support, use the no form of this command.
Syntax Description
language -code
Any of 139 languages designated by a two-letter ISO-639 country code.
qvalue
The priority of the language, with languages sorted in descending order according the assigned parameter value. Valid values include zero, one, or a decimal fraction in the range .001 through .999. Default is 1, the highest priority.
language -param
Specifies language preferences by associating a parameter with the language being configured.
Command Modes
Dial-peer voice configuration (config-dial-peer)
Voice service configuration (config-voi-serv)Usage Guidelines
To include the Accept-Language header in outgoing SIP INVITE messages, and enable Accept-Language header support on specific trunk groups with different language requirements, use dial-peer voice configuration mode, which is enabled by the dial-peer voice command . To enable Accept-Language headers to be included in both SIP INVITE messages and OPTIONS responses, use voice service configuration mode, enabled by the voice service pots command. If both voice service and dial-peer voice mode accept-language support are configured, and there are no dial-peer matches, the outgoing INVITE message contains the voice service specified languages. Otherwise, the INVITE contains the dial-peer configured languages.
The SIP Accept-Language Header Support feature supports 139 languages which are designated by a two-letter ISO-639 country code. The following is a partial list of supported language codes and languages. To display a complete listing use the help command supported-language ?.
suppress
To suppress accounting for a specific call leg, use the suppress command in gateway accounting AAA configuration mode. To reenable accounting for that leg, use the no form of this command.
Usage Guidelines
Use this command to turn off accounting for a specific call leg.
If both incoming and outgoing call legs are of the same type, no accounting packets are generated.
Use the rotary keyword to suppress excess start and stop accounting records. This causes only one pair of records to be generated for every connection attempt through a dial peer.
suspend-resume (SIP)
To enable SIP Suspend and Resume functionality, use the suspend-resume command in SIP user agent configuration mode. To disable SIP Suspend and Resume functionality, use the no form of this command.
Usage Guidelines
Session Initiation Protocol (SIP) gateways are now enabled to use Suspend and Resume. Suspend and Resume are basic functions of ISDN and ISDN User Part (ISUP) signaling procedures. A Suspend message temporarily halts communication (call hold), and a Resume message is received after a Suspend message and continues the communication.
switchback interval
To set the amount of time that the digital signal processor (DSP) farm waits before polling the primary Cisco Unified CallManager when the current Cisco Unified CallManager switchback connection fails, use the switchback intervalcommand in SCCP Cisco Unified CallManager configuration mode. To reset the amount of time to the default value, use the no form of this command.
Usage Guidelines
The optimum setting for this command depends on the platform and your individual network characteristics. Adjust the switchback interval value to meet your needs.
Examples
The following example sets the length of time the DSP farm waits to before polling the primary Cisco Unified CallManager to 120 seconds (2 minutes):
Router(conf-sccp-ccm)# switchback interval 120Related Commands
Command
Description
connect interval
Specifies how many times a given profile attempts to connect to the specific CiscoUnified CallManager.
sccp ccm group
Creates a Cisco CallManger group and enters SCCP Cisco CallManager configuration mode.
switchback method
Sets the method that Cisco Unified CallManager uses to initiate the switchback process.
switchover method
Sets the switchover method that the SCCP client uses when the communication between the active Cisco Unified CallManager and the SCCP client goes down.
switchback method
To set the Cisco Unified CallManager switchback method, use the switchback methodcommand in Skinny SCCP Cisco Unified CallManager configuration mode. To reset to the default value, use the no form of this command.
switchback method { graceful | guard [timeout-guard-value] | immediate | uptime uptime-timeout-value }
no switchback method
Syntax Description
graceful
Selects the graceful switchback method.
guard
Selects the graceful with guard switchback method.
guard timeout value
(Optional) Timeout value, in seconds. Range is from 60 to 172800. Default is 7200.
immediate
Selects the immediate switchback method.
uptime
Selects the uptime-delay switchback method.
uptime timeout value
(Optional) Timeout value, in seconds. Range is from 60 to 172800. Default is 7200.
Usage Guidelines
Use this command to set the Cisco Unified CallManager switchback method. When a switch-over happens with the secondary Cisco Unified CallManager initiates the switchback process with that higher-order Cisco Unified CallManager. The available switchback methods follow:
- graceful--The Cisco Unified CallManager switchback happens only after all the active sessions are terminated gracefully.
- guard--The Cisco Unified CallManager switchback happens either when the active sessions are terminated gracefully or when the guard timer expires, whichever happens first.
- immediate--Performs the Cisco Unified CallManager switchback to the higher order CiscoUnified CallManager immediately as soon as the timer expires, whether there is an active connection or not.
- uptime--Once the higher-order Cisco Unified CallManager comes alive, the uptime timer in initiated.
NoteThe optimum setting for this command depends on the platform and your individual network characteristics. Adjust the switchback method to meet your needs.
Examples
The following example sets the Cisco Unified CallManager switchback method to happen only after all the active sessions are terminated gracefully.
Router(config-sccp-ccm)# switchback method gracefulRelated Commands
Command
Description
connect interval
Specifies the amount of time that a DSP farm profile waits before attempting to connect to a Cisco Unified CallManager when the current Cisco Unified CallManager fails to connect.
sccp ccm group
Creates a Cisco CallManger group and enters SCCP Cisco CallManager configuration mode.
switchback interval
Sets the amount of time that the DSP farm waits before polling the primary Cisco Unified CallManager when the current Cisco Unified CallManager fails to connect.
switchover method
Sets the switchover method that the SCCP client uses when the communication between the active Cisco Unified CallManager and the SCCP client goes down.
switchover method
To set the switchover method that the Skinny Client Control Protocol (SCCP) client uses when the communication link between the active Cisco Unified CallManager and the SCCP client goes down, use the switchover methodcommand in SCCP Cisco Unified CallManager configuration mode. To reset the switchover method to the default, use the no form of this command.
Usage Guidelines
When the communication link between the active Cisco Unified CallManager and the SCCP client goes down the SCCP client tries to connect to one of the secondary Cisco Unified CallManagers using one of the following switchover methods:
- graceful--The Cisco Unified CallManager switchover happens only after all the active sessions are terminated gracefully.
- immediate--Regardless of whether there is an active connection or not the SCCP client switches over to one of the secondary Cisco Unified CallManagers immediately. If the SCCP Client is not able to connect to a secondary Cisco CUnified allManager, it continues polling for a CiscoUnified CallManager connection.
NoteThe optimum setting for this command depends on the platform and your individual network characteristics. Adjust the switchover method to meet your needs.
Examples
The following example sets the switchover method that the SCCP client uses to connect to a secondary Cisco Unified CallManager to happen only after all the active sessions are terminated gracefully:
Router (config-sccp-ccm)# switchover method gracefulRelated Commands
Command
Description
connect interval
Specifies the amount of time that a DSP farm profile waits before attempting to connect to a Cisco Unified CallManager when the current Cisco Unified CallManager fails to connect.
sccp ccm group
Creates a Cisco CallManger group and enters the SCCP Cisco CallManager configuration mode.
switchback interval
Sets the amount of time that the DSP farm waits before polling the primary Cisco Unified CallManager when the current Cisco Unified CallManager fails to connect.
switchback method
Sets the method that Cisco Unified CallManager uses to initiate the switchback process.
