Cisco IOS Voice Command Reference
Commands: call fallback jitter-probe precedence --> ces connect

Table Of Contents

call fallback jitter-probe precedence

call fallback jitter-probe priority-queue

call fallback key-chain

call fallback map address-list

call fallback map subnet

call fallback monitor

call fallback probe-timeout

call fallback reject-cause-code

call fallback threshold delay loss

call fallback threshold icpif

call fallback wait-timeout

call language voice

call language voice load

call rscmon update-timer

call rsvp-sync

call rsvp-sync resv-timer

call service stop

call spike

call start

call threshold global

call threshold interface

call threshold poll-interval

call treatment

call-agent

call-block (dial-peer)

call-denial

called-number (dial-peer)

caller-id

caller-id alerting dsp-pre-alloc

caller-id alerting line-reversal

caller-id alerting pre-ring

caller-id alerting ring

caller-id attenuation

caller-id block

caller-id block (ephone-dn)

caller-id enable

caller-number

call-forward all (ephone-dn)

call-forward busy (cm-fallback)

call-forward busy (ephone-dn)

call-forward noan (cm-fallback)

call-forward noan (ephone-dn)

call-forward pattern

calling-info pstn-to-sip

calling-info sip-to-pstn

calling-number outbound

call-manager-fallback

call-router

call-waiting

cap-list vfc

capacity update interval (dial peer)

capacity update interval (trunk group)

card type (t1/e1)

card type (t3/e3)

carrier-id (dial-peer)

carrier-id (global)

carrier-id (trunk group)

carrier-id (voice source group)

cause-code

ccm-manager application redundant-link port

ccm-manager config

ccm-manager fallback-mgcp

ccm-manager mgcp

ccm-manager music-on-hold

ccm-manager music-on-hold bind

ccm-manager redundant-host

ccm-manager switchback

ccm-manager switchover-to-backup

ccs connect (controller)

ccs connect (interface)

ccs encap frf11

ces cell-loss-integration-period

ces clockmode synchronous

ces connect


call fallback jitter-probe precedence

To specify the priority of the jitter-probe transmission, use the call fallback jitter-probe precedence command in global configuration mode. To restore the default priority, use the no form of this command.

call fallback jitter-probe precedence precedence-value

no call fallback jitter-probe precedence

Syntax Description

precedence-value

Jitter-probe precedence. Range is from 0 to 6. The default is 2.


Defaults

Enabled

Value set to 2

Command Modes

Global configuration

Command History

Release
Modification

12.1(3)T

This command was introduced.

12.2(2)XB1

This command was implemented on the Cisco AS5850.

12.2(4)T

The PSTN Fallback feature and enhancements were implemented on Cisco 7200 series and integrated into Cisco IOS Release 12.2(4)T.

12.2(4)T2

This command was implemented on the Cisco 7500 series.

12.2(8)T

Support for the Cisco AS5850 is not included in this release.

12.2(11)T

This command is supported on the Cisco AS5850 in this release.


Usage Guidelines

Every IP packet has a precedence header. Precedence is used by various queueing mechanisms in routers to determine the priority of traffic passing through the system.

Use the call fallback jitter-probe precedence command if there are different queueing mechanisms in your network. Enabling the call fallback jitter-probe precedence command sets the precedence for jitter probes to pass through your network.

If you require your probes to be sent and returned quickly, set the precedence to a low number (0 or 1): the lower the precedence, the higher the priority given.

The call fallback jitter-probe precedence command is mutually exclusive with the call fallback jitter-probe dscp command. Only one of these command can be enabled on the router. Usually the call fallback jitter-probe precedence command is enabled. When the call fallback jitter-probe dscp command is configured, the precedence value is replaced by the DSCP value. To disable DSCP and restore the default jitter probe precedence value, use the no call fallback jitter-probe dscp command.

Examples

The following example specifies a jitter-probe precedence of 5, or low priority. The following configuration changes the default jitter-probe precedence value. If the call fallback jitter-probe dscp command is configured on the same router, this configuration replaces the DSCP value with the precedence value:

call fallback jitter-probe precedence 5

The following configuration restores the default value for precedence:

no call fallback jitter-probe precedence

Related Commands

Command
Description

call fallback active

Enables a call request to fall back to alternate dial peers in case of network congestion.

call fallback jitter-probe dscp

Specifies the differentiated services code point (dscp) of the jitter-probe transmission.

call fallback jitter-probe num-packets

Specifies the number of packets in a jitter probe that are used to determine network conditions.

call fallback jitter-probe priority-queue

Assigns a priority queue for jitter-probe transmissions.

show call fallback config

Displays the call fallback configuration.


call fallback jitter-probe priority-queue

To assign a priority queue for jitter-probe transmissions, use the call fallback jitter-probe priority-queue command in global configuration mode. To return to the default state, use the no form of this command.

call fallback jitter-probe priority-queue

no call fallback jitter-probe priority-queue

Syntax Description

This command has no arguments or keywords.

Defaults

Disabled

Command Modes

Global configuration

Command History

Release
Modification

12.1(3)T

This command was introduced.

12.2(2)XB1

This command was implemented on the Cisco AS5850.

12.2(4)T

The PSTN Fallback feature and enhancements were implemented on Cisco 7200 series and integrated into Cisco IOS Release 12.2(4)T.

12.2(4)T2

This command was implemented on the Cisco 7500 series.

12.2(8)T

Support for the Cisco AS5850 is not included in this release.

12.2(11)T

This command is supported on the Cisco AS5850 in this release.


Usage Guidelines

This command is applicable only if the queueing method used is IP Real-Time Transport Protocol (RTP) Priority. This command is unnecessary when low latency queueing (LLQ) is used because these packets follow the priority queue path (or not) based on the LLQ classification criteria and not this command.

This command works by choosing between sending the probe on an odd or even Service Assurance Agent (SAA) port number. The SAA probe packets go out on randomly selected ports chosen from within the top end of the audio User Datagram Protocol (UDP) defined port range (16384 to 32767). The port pair (RTP Control Protocol [RTCP] port) is selected, and by default, SAA probes for call fallback use the RTCP port (odd) to avoid going into the priority queue, if enabled. If call fallback is configured to use the priority queue, the RTP port (even) is selected.

Examples

The following example specifies that a probe be sent to an SAA port:

Router(config)# call fallback jitter-probe priority-queue 

Note In order for this command to have any effect on the probes, the IP priority queueing must be set for UDP voice ports numbered from 16384 to 32767.


Related Commands

Command
Description

call fallback active

Enables a call request to fall back to alternate dial peers in case of network congestion.

call fallback jitter-probe num-packets

Specifies the number of packets in a jitter probe that are used to determine network conditions.

call fallback jitter-probe precedence

Specifies the jitter-probe precedence.

ip rtp priority

Provides a strict priority queueing scheme for delay-sensitive data.

show call fallback config

Displays the call fallback configuration.


call fallback key-chain

To specify the use of message digest 5 (MD5) algorithm authentication for sending and receiving Service Assurance Agents (SAA) probes, use the call fallback key-chain command in global configuration mode. To disable MD5, use the no form of this command.

call fallback key-chain name-of-chain

no call fallback key-chain name-of-chain

Syntax Description

name-of-chain

Name of the chain. This name is alphanumeric and case-sensitive text. There is no default value.


Defaults

MD5 authentication is not used.

Command Modes

Global configuration

Command History

Release
Modification

12.1(3)T

This command was introduced.

12.2(2)XB1

This command was implemented on the Cisco AS5850.

12.2(4)T

The PSTN Fallback feature and enhancements were implemented on Cisco 7200 series and integrated into Cisco IOS Release 12.2(4)T.

12.2(4)T2

This command was implemented on the Cisco 7500 series.

12.2(8)T

Support for the Cisco AS5850 is not included in this release.

12.2(11)T

This command is supported on the Cisco AS5850 in this release.


Usage Guidelines

This command is used to enable the SAA probe authentication using MD5. If MD5 authentication is used, the keys on the sender and receiver routers must match.

Examples

The following example specifies "secret" as the fallback key chain:

Router(config)# call fallback key-chain secret

Related Commands

Command
Description

call fallback active

Enables a call request to fall back to alternate dial peers in case of network congestion.

key chain

Enables authentication for routing protocols by identifying a group of authentication keys.

key-string

Specifies the authentication string for a key.

show call fallback config

Displays the call fallback configuration.


call fallback map address-list

To specify that the call fallback router keep a cache table by IP addresses of distances for several destination peers, use the call fallback map address-list command in global configuration mode. To restore the default values, use the no form of this command.

call fallback map map target ip-address address-list ip-address1 ... ip-address7

no call fallback map map target ip-address address-list ip-address1 ... ip-address7

Syntax Description

map

Fallback map. Range is from 1 to 16. There is no default.

target ip-address

Target IP address.

ip-address1 ... ip-address7

Lists the IP addresses that are kept in the cache table. The maximum number of IP addresses is seven.


Defaults

No call fallback maps are defined.

Command Modes

Global configuration

Command History

Release
Modification

12.1(3)T

This command was introduced.

12.2(2)XB1

This command was implemented on the Cisco AS5850.

12.2(4)T

The PSTN Fallback feature and enhancements were implemented on Cisco 7200 series and integrated into Cisco IOS Release 12.2(4)T.

12.2(4)T2

This command was implemented on the Cisco 7500 series.

12.2(8)T

Support for the Cisco AS5850 is not included in this release.

12.2(11)T

This command is supported on the Cisco AS5850 in this release.


Usage Guidelines

Use this command when several destination peers are in one common node.

Call fallback map setup allows the decongestion of traffic caused by a high volume of call probes sent across a network to query a large number of dial peers. One router/common node can keep the distances in a cache table of the numerous IP addresses/destination peers in a network. When the fallback is queried for network congestion to a particular IP address (that is, the common node), the map addresses are searched to find the target IP address. If a match is determined, the probes are sent to the target address rather than to the particular IP address.

In Figure 2, the three routers (1, 2, and 3) keep the cache tables of distances for the destination peers behind them. When a call probe comes from somewhere in the IP cloud, the cache routers check their distance tables for the IP address/destination peer where the call probe is destined. This distance checking limits congestion on the networks behind these routers by directing the probe to the particular IP address and not to the entire network.

Figure 2 Call Fallback Map with IP Addresses

Examples

The following example specifies call fallback map address-list configurations for 
172.32.10.1 and 172.46.10.1:

Router(config)# call fallback map 1 target 172.32.10.1
address-list 172.32.10.2 172.32.10.3 172.32.10.4 172.32.10.5

172.32.10.6 172.32.10.7 172.32.10.8

Router(config)# call fallback map 2 target 172.46.10.1
address-list 172.46.10.2 172.46.10.3 172.46.10.4 172.46.10.5

172.46.10.6 172.46.10.7 172.46.10.8

Related Commands

Command
Description

call fallback active

Enables a call request to fall back to alternate dial peers in case of network congestion.

call fallback map subnet

Specifies that the call fallback router keep a cache table by subnet addresses of distances for several destination peers that are sitting behind the router.

show call fallback config

Displays the call fallback configuration.


call fallback map subnet

To specify that the call fallback router keep a cache table by subnet addresses of distances for several destination peers, use the call fallback map subnet command in global configuration mode. To restore the default values, use the no form of this command.

call fallback map map target ip-address subnet ip-network netmask

no call fallback map map target ip-address subnet ip-network netmask

Syntax Description

map

Fallback map. Range is from 1 to 16. There is no default.

target ip-address

Target IP address.

subnet ip-network

Subnet IP address.

netmask

Network mask number.


Defaults

No call fallback maps are defined.

Command Modes

Global configuration

Command History

Release
Modification

12.1(3)T

This command was introduced.

12.2(2)XB1

This command was implemented on the Cisco AS5850.

12.2(4)T

The PSTN Fallback feature and enhancements were implemented on Cisco 7200 series and integrated into Cisco IOS Release 12.2(4)T.

12.2(4)T2

This command was implemented on the Cisco 7500 series.

12.2(8)T

Support for the Cisco AS5850 is not included in this release.

12.2(11)T

This command is supported on the Cisco AS5850 in this release.


Usage Guidelines

Use this command when several destination peers are in one common node.

Call fallback map setup allows the decongestion of traffic caused by a high volume of call probes sent across a network to query a large number of dial peers. One router/common node can keep the distances in a cache table of the numerous IP addresses within a subnet (destination peers) in a network. When the fallback is queried for network congestion to a particular IP address (that is, the common node), the map addresses are searched to find the target IP address. If a match is determined, the probes are sent to the target address rather than to the particular IP address.

In Figure 3, the three routers (1, 2, and 3) keep the cache tables of distances for the destination peers behind them. When a call probe comes from somewhere in the IP cloud, the cache routers check their distance tables for the subnet address/destination peer where the call probe is destined. This distance checking limits congestion on the networks behind these routers by directing the probe to the particular subnet address and not to the entire network.

Figure 3 Call Fallback Map with Subnet Addresses

Examples

The following examples specify the call fallback map subnet configuration for two different IP addresses:

Router(config)# call fallback map 1 target 209.165.201.225 subnet
209.165.201.224 255.255.255.224

Router(config)# call fallback map 2 target 209.165.202.225 subnet
209.165.202.224 255.255.255.224

Related Commands

Command
Description

call fallback active

Enables a call request to fall back to alternate dial peers in case of network congestion.

call fallback map address-list

Specifies that the call fallback router keep a cache table by IP addresses of distances for several destination peers that are sitting behind the router.

show call fallback config

Displays the call fallback configuration.


call fallback monitor

To enable the monitoring of destinations without call fallback to alternate dial peers, use the call fallback monitor command in global configuration mode. To disable monitoring without fallback, use the no form of this command.

call fallback monitor

no call fallback monitor

Syntax Description

This command has no arguments or keywords.

Defaults

Disabled

Command Modes

Global configuration

Command History

Release
Modification

12.1(3)T

This command was introduced.

12.2(2)XB1

This command was implemented on the Cisco AS5850.

12.2(4)T

The PSTN Fallback feature and enhancements were introduced on Cisco 7200 series and integrated into Cisco IOS Release 12.2(4)T.

12.2(4)T2

This command was implemented on the Cisco 7500 series.

12.2(8)T

Support for the Cisco AS5850 is not included in this release.

12.2(11)T

This command is supported on the Cisco AS5850 in this release.


Usage Guidelines

The call fallback monitor command is used as a statistics collector of network conditions based upon probes (detailing network traffic) and connected calls. There is no H.323 call checking/rejecting as with the call fallback active command. All call requests are granted regardless of network traffic conditions.

Configure the call fallback threshold delay loss or call fallback threshold icpif command to set threshold parameters. The thresholds are ignored, but for statistics collecting, configuring one of the thresholds allows you to monitor cache entries for either delay/loss or Calculated Planning Impairment Factor (ICPIF) values.

Examples

The following example enables the call fallback monitor command:

Router(config)# call fallback monitor

Related Commands

Command
Description

call fallback active

Enables a call request to fall back to alternate dial peers in case of network congestion.

call fallback threshold delay loss

Specifies that the call fallback threshold use only packet delay and loss values.

call fallback threshold icpif

Specifies that call fallback use the ICPIF threshold.

show call fallback config

Displays the call fallback configuration.


call fallback probe-timeout

To set the timeout for a Service Assurance Agent (SAA) probe for call fallback purposes, use the call fallback probe-timeout command in global configuration mode. To restore the default value, use the no form of this command.

call fallback probe-timeout seconds

no call fallback probe-timeout

Syntax Description

seconds

Interval, in seconds. Range is from 1 to 2147483. The default is 30.


Defaults

30 seconds

Command Modes

Global configuration

Command History

Release
Modification

12.1(3)T

This command was introduced.

12.2(2)XB1

This command was implemented on the Cisco AS5850.

12.2(4)T

The PSTN Fallback feature and enhancements were implemented on Cisco 7200 series and integrated into Cisco IOS Release 12.2(4)T.

12.2(4)T2

This command was implemented on the Cisco 7500 series.

12.2(8)T

Support for the Cisco AS5850 is not included in this release.

12.2(11)T

This command is supported on the Cisco AS5850 in this release.


Usage Guidelines

SAA probes collect network traffic information based upon configured delay and loss or Calculated Planning Impairment Factor (ICPIF) values and report this information to the cache for call request determination. Use the call fallback threshold delay loss or call fallback threshold icpif commands to set the threshold parameters.

When the probe timeout expires, a new probe is sent to collect network statistics. To reduce the bandwidth taken up by the probes, increase the probe-timeout interval (seconds). Probes do not have a great effect upon bandwidth unless several thousand destinations are involved. If this is the case in your network, use a longer timeout. If you need more network traffic information, and bandwidth is not an issue, use a lower timeout. The default interval, 30 seconds, is a low timeout.

When the call fallback cache-timeout command is configured or expires, new probes are initiated for data collection.

Examples

The following example configures a 120-second interval:

Router(config)# call fallback probe-timeout 120

Related Commands

Command
Description

call fallback active

Enables a call request to fall back to alternate dial peers in case of network congestion.

call fallback cache-timeout

Specifies the time after which the cache entries of network conditions are purged.

call fallback threshold delay loss

Specifies that the call fallback threshold use only packet delay and loss values.

call fallback threshold icpif

Specifies that call fallback use the ICPIF threshold.

show call fallback config

Displays the call fallback configuration.


call fallback reject-cause-code

To enable a specific call fallback reject cause code in case of network congestion, use the call fallback reject-cause-code command in global configuration mode. To reset the code to the default of 49, use the no form of this command.

call fallback reject-cause-code number

no call fallback reject-cause-code

Syntax Description

number

Specifies the cause code as defined in the International Telecommunication Union (ITU) standard Q.850 except the code for normal call clearing, which is code 16. The default is 49. See Table 12 for ITU cause-code numbers.


Defaults

49 (quality of service is unavailable)

Command Modes

Global configuration

Command History

Release
Modification

12.2(2)XA

This command was introduced.

12.2(4)T

The PSTN Fallback feature and enhancements were implemented on Cisco 7200 series and integrated into Cisco IOS Release 12.2(4)T.

12.2(4)T2

This command was implemented on the Cisco 7500 series.


Usage Guidelines

Enabling the call fallback reject-cause-code command determines the code to display when calls are rejected because of probing of network conditions.


Note Connected calls are not affected by this feature.


Table 12 lists the ITU cause codes and their associated displayed messages and meanings.

Cause Code
Displayed Message
Meaning

1

Unallocated (unassigned) number

Indicates that the called party cannot be reached because, although the called party number is in a valid format, it is not currently allocated (assigned).

2

No route to specified transit network (national use)

Indicates that the equipment that is sending this code has received a request to route the call through a particular transit network that it does not recognize. The equipment that is sending this code does not recognize the transit network either because the transit network does not exist or because that particular transit network, while it does exist, does not serve the equipment that is sending this cause. This code is supported on a network-dependent basis.

3

No route to destination

Indicates that the called party cannot be reached because the network through which the call has been routed does not serve the destination desired. This code is supported on a network-dependent basis.

4

Send special information tone

Indicates that the called party cannot be reached for reasons that are of a long-term nature and that the special information tone should be returned to the calling party.

5

Misdialed trunk prefix (national use)

Indicates the erroneous inclusion of a trunk prefix in the called party number.

6

Channel unacceptable

Indicates that the channel most recently identified is not acceptable to the sending entity for use in this call.

7

Call awarded and being delivered in an established channel

Indicates that the user has been awarded the incoming call and that the incoming call is being connected to a channel that is already established to that user for similar calls (for example, packet-mode X.25 virtual calls).

8

Preemption

Indicates that the call is being preempted.

9

Preemption - circuit reserved for reuse

Indicates that the call is being preempted and that the circuit is reserved for reuse by the preempting exchange.

16

Normal call clearing

Indicates that the call is being cleared because one of the users involved in the call has requested that the call be cleared. Under normal situations, the source of this code is not the network.

17

User busy

Indicates that the called party is unable to accept another call because the user busy condition has been encountered. This code may be generated by the called user or by the network. If the user determines the busy, it is noted that the user equipment is compatible with the call.

18

No user responding

Indicates when a called party does not respond to a call establishment message with either an alerting or a connect indication within the prescribed period of time allocated.

19

No answer from user (user alerted)

Indicates when the called party has been alerted but does not respond with a connect indication within a prescribed period of time.

Note This code is not necessarily generated by ITU standard Q.931 procedures but may be generated by internal network timers.

20

Subscriber absent

Indicates when a mobile station has logged off, when radio contact is not obtained with a mobile station, or when a personal telecommunication user is temporarily not addressable at any user-network interface.

21

Call rejected

Indicates that the equipment that is sending this code does not wish to accept this call although it could have accepted the call because the equipment that is sending this code is neither busy nor incompatible.

The network may also generate this code, indicating that the call was cleared because of a supplementary service constraint. The diagnostic field may contain additional information about the supplementary service and reason for rejection.

22

Number changed

Indicates when the called-party number indicated by the calling party is no longer assigned. The new called-party number may optionally be included in the diagnostic field. If a network does not support this code, codeNo. 1, an unallocated (unassigned) number, shall be used.

26

Non-selected user clearing

Indicates that the user has not been sent the incoming call.

27

Destination out of order

Indicates that the destination indicated by the user cannot be reached because the interface to the destination is not functioning correctly. The term "not functioning correctly" indicates that a signaling message was unable to be delivered to the remote party; for example, a physical layer or data link layer failure at the remote party, or the equipment of the user is offline.

28

Invalid number format (address incomplete)

Indicates that the called party cannot be reached because the called party number is not in a valid format or is not complete.

29

Facility rejected

Indicates when a supplementary service requested by the user cannot be provided by the network.

30

Response to STATUS ENQUIRY

Indicates when the reason for generating the STATUS message was the prior receipt of a STATUS ENQUIRY message.

31

Normal, unspecified

Reports a normal event only when no other code in the normal class applies.

34

No circuit/channel available

Indicates that there is no appropriate circuit or channel presently available to handle the call.

38

Network out of order

Indicates that the network is not functioning correctly and that the condition is likely to last a relatively long period of time; for example, immediately reattempting the call is not likely to be successful.

39

Permanent frame mode connection out-of-service

Indicates in a STATUS message that a permanently established frame mode connection is out-of-service (for example, due to equipment or section failure) (see the ITU standard, Annex A/Q.933).

40

Permanent frame mode connection operational

Indicates in a STATUS message to indicate that a permanently established frame mode connection is operational and capable of carrying user information (see the ITU standard, Annex A/Q.933).

41

Temporary failure

Indicates that the network is not functioning correctly and that the condition is not likely to last a long period of time; for example, the user may wish to try another call attempt almost immediately.

42

Switching equipment congestion

Indicates that the switching equipment that is generating this code is experiencing a period of high traffic.

43

Access information discarded

Indicates that the network could not deliver access information to the remote user as requested, that is, user-to-user information, low layer compatibility, high layer compatibility, or subaddress, as indicated in the diagnostic. It is noted that the particular type of access information discarded is optionally included in the diagnostic.

44

Requested circuit/channel not available

Indicates when the circuit or channel indicated by the requesting entity cannot be provided by the other side of the interface.

46

Precedence call blocked

Indicates that there are no preemptable circuits or that the called user is busy with a call of an equal or higher preemptable level.

47

Resource unavailable, unspecified

Reports a resource-unavailable event only when no other cause in the resource-unavailable class applies.

49

Quality of service not available

Reports that the requested quality of service, as defined in ITU recommendation X.213, cannot be provided (for example, throughput or transit delay cannot be supported).

50

Requested facility not subscribed

Indicates that the user has requested a supplementary service that is implemented by the equipment that generated this cause but that the user is not authorized to use this service.

53

Outgoing calls barred within CUG

Indicates that although the calling party is a member of the closed user group (CUG) for the outgoing CUG call, outgoing calls are not allowed for this member of the CUG.

55

Incoming calls barred within CUG

Indicates that although the called party is a member of the CUG for the incoming CUG call, incoming calls are not allowed for this member of the CUG.

57

Bearer capability not authorized

Indicates that the user has requested a bearer capability that is implemented by the equipment that generated this cause but that the user is not authorized to use this capability.

58

Bearer capability not presently available

Indicates that the user has requested a bearer capability that is implemented by the equipment that generated this cause but that is not available at this time.

62

Inconsistency in designated outgoing access information and subscriber class

Indicates that there is an inconsistency in the designated outgoing access information and subscriber class.

63

Service or option not available, unspecified

Reports a service or option not available event only when no other cause in the service or option not available class applies.

65

Bearer capability not implemented

Indicates that the equipment that is sending this code does not support the bearer capability requested.

66

Channel type not implemented

Indicates that the equipment that is sending this code does not support the channel type requested.

69

Requested facility not implemented

Indicates that the equipment that is sending this code does not support the requested supplementary service.

70

Only restricted digital information bearer capability is available (national use)

Indicates that the calling party has requested an unrestricted bearer service but that the equipment that is sending this cause supports only the restricted version of the requested bearer capability.

79

Service or option not implemented, unspecified

Reports a service or option not implemented event only when no other code in the service or option not implemented class applies.

81

Invalid call reference value

Indicates that the equipment that is sending this code has received a message with a call reference that is not currently in use on the user-network interface.

82

Identified channel does not exist

Indicates that the equipment that is sending this code has received a request to use a channel not activated on the interface for a call. For example, if a user has subscribed to those channels on a PRI numbered from 1 to 12 and the user equipment or the network attempts to use channels 13 through 23, this cause is generated.

83

A suspended call exists, but this call identity does not

Indicates that a call resume has been attempted with a call identity that differs from that in use for any presently suspended call(s).

84

Call identity in use

Indicates that the network has received a call suspended request that contains a call identity (including the null call identity) that is already in use for a suspended call within the domain of interfaces over which the call might be resumed.

85

No call suspended

Indicates that the network has received a call resume request that contains a call identity information element that presently does not indicate any suspended call within the domain of interfaces over which calls may be resumed.

86

Call having the requested call identity has been cleared

Indicates that the network has received a call resume request that contains a call identity information element that indicates a suspended call that has in the meantime been cleared while suspended (either by network timeout or by the remote user).

87

User not member of CUG

Indicates that the called user for the incoming CUG call is not a member of the specified CUG or that the calling user is an ordinary subscriber that is calling a CUG subscriber.

88

Incompatible destination

Indicates that the equipment that is sending this code has received a request to establish a call that has low layer compatibility, high layer compatibility, or other compatibility attributes (for example, data rate) that cannot be accommodated.

90

Non-existent CUG

Indicates that the specified CUG does not exist.

91

Invalid transit network selection (national use)

Indicates that a transit network identification was received that is of an incorrect format as defined in ITU standard Annex C/Q.931.

95

Invalid message, unspecified

Reports an invalid message event only when no other code in the invalid message class applies.

96

Mandatory information element is missing

Indicates that the equipment that is sending this code has received a message that is missing an information element that must be present in the message before that message can be processed.

97

Message type non-existent or not implemented

Indicates that the equipment that is sending this code has received a message with a message type that it does not recognize because this is a message not defined or defined but not implemented by the equipment that is sending this cause.

98

Message not compatible with call state or message type non-existent or not implemented

Indicates that the equipment that is sending this code has received a message that the procedures do not indicate as a permissible message to receive while in the call state, or that a STATUS message that indicates an incompatible call state was received.

99

Information element/parameter non-existent or not implemented

Indicates that the equipment that is sending this code has received a message that includes information elements or parameters not recognized because the information element identifiers or parameter names are not defined or are defined but not implemented by the equipment sending the code. This code indicates that the information elements or parameters were discarded. However, the information element is not required to be present in the message for the equipment that is sending the code to process the message.

100

Invalid information element contents

Indicates that the equipment that is sending this code has received an information element that it has implemented; however, one or more fields in the information element are coded in a way that has not been implemented by the equipment that is sending this code.

101

Message not compatible with call state

Indicates that a message has been received that is incompatible with the call state.

102

Recovery on timer expired

Indicates that a procedure has been initiated by the expiration of a timer in association with error-handling procedures.

103

Parameter non-existent or not implemented - passed on

Indicates that the equipment that is sending this code has received a message that includes parameters not recognized because the parameters are not defined or are defined but not implemented by the equipment that is sending the code. The code indicates that the parameters were ignored. In addition, if the equipment that is sending this code is an intermediate point, this code indicates that the parameters were passed on unchanged.

110

Message with unrecognized parameter discarded

Indicates that the equipment that is sending this code has discarded a received message that includes a parameter that is not recognized.

111

Protocol error, unspecified

Reports a protocol error event only when no other code in the protocol error class applies.

127

Interworking, unspecified

Indicates that there has been interworking with a network that does not provide codes for actions it takes. Thus, the precise code for a message that is being sent cannot be ascertained.


Examples

The following example enables the call fallback reject-cause-code command and specifies cause code 34:

call fallback reject-cause-code 34

Related Commands

Command
Description

call fallback cache-size

Specifies the call fallback cache size for network traffic probe entries.

call fallback cache-timeout

Specifies the time after which the cache entries of network conditions are purged.

call fallback instantaneous-value-weight

Specifies that the call fallback subsystem take an average from the last two cache entries for call requests.

call fallback jitter-probe num-packets

Specifies the number of packets in a jitter probe that are used to determine network conditions.

call fallback jitter-probe precedence

Specifies the priority of the jitter-probe transmission.

call fallback jitter-probe priority-queue

Assigns a priority queue for jitter-probe transmissions.

call fallback key-chain

Specifies MD5 authentication for sending and receiving SAA probes.

call fallback map address-list

Specifies that the call fallback router keep a cache table by IP addresses of distances for several destination peers that are sitting behind the router.

call fallback map subnet

Specifies that the call fallback router keep a cache table by subnet addresses of distances for several destination peers that are sitting behind the router.

call fallback probe-timeout

Sets the timeout for an SAA probe for call fallback purposes.

call fallback threshold delay loss

Specifies that the call fallback threshold use only packet delay and loss values.

call fallback threshold icpif

Specifies that call fallback use the ICPIF threshold.

show call fallback config

Displays the call fallback configuration.


call fallback threshold delay loss

To specify that the call fallback threshold use only packet delay and loss values, use the call fallback threshold delay loss command in global configuration mode. To restore the default value, use the no form of this command.

call fallback threshold delay delay-value loss loss-value

no call fallback threshold delay delay-value loss loss-value

Syntax Description

delay-value

Sets the delay value, in milliseconds. Range is from 1 to 2147483647. There is no default value.

loss-value

Sets the loss value, expressed as a percentage. The valid range is from 0 to 100. There is no default value.


Defaults

No default behavior or values.

Command Modes

Global configuration

Command History

Release
Modification

12.1(3)T

This command was introduced.


Usage Guidelines

During times of heavy voice traffic, two parties in a conversation may notice a significant delay in transmission or hear only part of a conversation because of voice-packet loss.

Use the call fallback threshold delay loss command to configure parameters for voice quality. Lower values of delay and loss allow higher quality of voice. Call requests match the network information in the cache with the configured thresholds of delay and loss.

The amount of delay set by the call fallback threshold delay loss command should not be more than half the amount of the time-to-wait value set by the call fallback wait-timeout command; otherwise the threshold delay will not work correctly. Because the default value of the call fallback wait-timeout command is set to 300 milliseconds, the user can configure a delay of up to 150 milliseconds for the call fallback threshold delay loss command. If the user wants to configure a higher threshold, the time-to-wait delay has to be increased from its default (300 milliseconds) using the call fallback wait-timeout command.


Note The delay configured by call fallback threshold delay loss command corresponds to a one-way delay, whereas the time-to-wait period configured by call fallback wait-timeout command corresponds to a round-trip delay.


If you enable the call fallback active command, the call fallback subsystem uses the last cache entry compared with the configured delay/loss threshold to determine whether the call is connected or denied. If you enable the call fallback monitor command, all calls are connected, regardless of the configured threshold or voice quality. In this case, configuring the call fallback threshold delay loss command allows you to collect network statistics for further tracking.


Note The call fallback threshold delay loss command differs from the call fallback threshold icpif command because the call fallback threshold delay loss command uses only packet delay and loss parameters while the call fallback threshold icpif command uses packet delay and loss parameters plus other International Telecommunication Union (ITU) G.113 factors to gather impairment information.


Setting this command does not affect bandwidth. Available bandwidth for call requests is determined by the call fallback subsystem using probes. The number of probes on the network affects bandwidth.

Examples

The following example configures a threshold delay of 20 milliseconds and a threshold loss of 50 percent:

Router(config)# call fallback threshold delay 20 loss 50

Related Commands

Command
Description

call fallback active

Enables a call request to fall back to alternate dial peers in case of network congestion.

call fallback monitor

Enable the monitoring of destinations without call fallback to alternate dial peers.

call fallback threshold icpif

Specifies the ICPIF threshold.

show call fallback config

Displays the call fallback configuration.


call fallback threshold icpif

To specify that call fallback use the Calculated Planning Impairment Factor (ICPIF) threshold, use the call fallback threshold icpif command in global configuration mode. To restore the default value, use the no form of this command.

call fallback threshold icpif threshold-value

no call fallback threshold icpif

Syntax Description

threshold-value

Threshold value. Range is from 0 to 34. The default is 5.


Defaults

5

Command Modes

Global configuration

Command History

Release
Modification

12.1(3)T

This command was introduced on Cisco 2600 series, Cisco 3600 series, and Cisco MC3810.

12.2(2)XB1

This command was implemented on the Cisco AS5850.

12.2(4)T

The PSTN Fallback feature and enhancements were introduced on Cisco 7200 series routers and integrated into Cisco IOS Release 12.2(4)T.

12.2(4)T2

This command was implemented on the Cisco 7500 series.

12.2(8)T

Support for the Cisco AS5850 is not included in this release.

12.2(11)T

This command is supported on the Cisco AS5850 in this release.


Usage Guidelines

During times of heavy voice traffic, the parties in a conversation may notice a significant delay in transmission or hear only part of a conversation because of voice-packet loss.

Use the call fallback threshold icpif command to configure parameters for voice quality. A low ICPIF value allows for higher quality of voice. Call requests match the network information in the cache with the configured ICPIF threshold. If you enable the call fallback active command, the call fallback subsystem uses the last cache entry compared with the configured ICPIF threshold to determine whether the call is connected or denied. If you enable the call fallback monitor command, all calls are connected regardless of the configured threshold or voice quality. In this case, configuring the call fallback threshold icpif command allows you to collect network statistics for further tracking.

A lower ICPIF value tolerates less delay and loss of voice packets (according to ICPIF calculations). Use lower values for higher quality of voice. Configuring a value of 34 equates to 100 percent packet loss.

The ICPIF is calculated and used according to the International Telecommunication Union (ITU) G.113 specification.


Note The call fallback threshold delay loss command differs from the call fallback threshold icpif command because the call fallback threshold delay loss command uses only packet delay and loss parameters while the call fallback threshold icpif command uses packet delay and loss parameters plus other ITU G.113 factors to gather impairment information.


Setting this command does not affect bandwidth. Available bandwidth for call requests is determined by the call fallback subsystem using probes. The number of probes on the network affects bandwidth.

Examples

The following example sets the ICPIF threshold to 20:

Router(config)# call fallback threshold icpif 20

Related Commands

Command
Description

call fallback active

Enables a call request to fall back to alternate dial peers in case of network congestion.

call fallback monitor

Enables the monitoring of destinations without call fallback to alternate dial peers.

call fallback threshold delay loss

Specifies the call fallback threshold delay and loss values.

show call fallback config

Displays the call fallback configuration.


call fallback wait-timeout

To modify the time to wait for a response to a probe, use the call fallback wait-timeout command in global configuration mode. To return to the default value, use the no form of this command.

call fallback wait-timeout milliseconds

no call fallback wait-timeout milliseconds

Syntax Description

milliseconds

Specifies the time-to-wait value in milliseconds. The range is 100 to 3000 milliseconds.


Defaults

300 milliseconds

Command Modes

Global configuration

Command History

Release
Modification

12.2(15)T9

This command was introduced.


Usage Guidelines

This command is enabled by default and the time to wait for a response to a probe is set to 300 milliseconds. This command allows the user to modify the amount of time to wait for a response to a probe. The milliseconds argument allows the user to configure a time-to-wait value between 100 milliseconds and 3000 milliseconds. A user who has a higher-latency network may want to increase the value of the default timer.

The time-to-wait period set by the call fallback wait-timeout command should always be greater than or equal to twice the amount of the threshold delay time set by the call fallback threshold delay loss command; otherwise the probe will fail.


Note The delay configured by the call fallback threshold delay loss command corresponds to a one-way delay, whereas the time-to-wait period configured by call fallback wait-timeout command corresponds to a round-trip delay. The threshold delay time should be set at half the value of the time-to-wait value.


Examples

The following example sets the value of the amount of time to wait for a response to a probe to 200 milliseconds:

call fallback wait-timeout 200

Related Commands

Command
Description

call fallback threshold delay loss

Specifies the call fallback threshold delay and loss values.


call language voice

To configure an external Tool Command Language (TCL) module for use with an interactive voice response (IVR) application, use the call language voice command in global configuration mode.

call language voice language url

Syntax Description

language

Two-character prefix for the language; for example, "en" for English or "ru" for Russian.

url

URL that points to the TCL module.


Defaults

No default behavior or values

Command Modes

Global configuration

Command History

Release
Modification

12.2(2)T

This command was introduced.


Usage Guidelines

The built-in languages are English (en), Chinese (ch), and Spanish (sp). If you specify "en", "ch", or "sp", the new TCL module replaces the built-in language functionality. When you add a new TCL module, you create your own prefix to identify the language. When you configure and load the new languages, any upper-layer application (TCL IVR) can use the language.

You can use the language prefix in the language argument of any call application voice command. The language and the text-to-speech (TTS) notations are available for the IVR application to use after they are defined by the TCL module.

Examples

The following example adds Russian (ru) as a TCL module:

call language voice ru tftp://box/unix/scripts/multi-lang/ru_translate.tcl

Related Commands

Command
Description

call application voice

Configures an application.

debug voip ivr

Specifies the type of VoIP IVR debug output that you want to view.

show language voice

Displays information about configured languages and applications.


call language voice load

To load or reload a Tool Command Language (TCL) module from the configured URL location, use the call language voice load command in EXEC mode.

call language voice load language

Syntax Description

language

The two-character prefix configured with the call language voice command in global configuration mode; for example, "en" for English or "ru" for Russian.


Defaults

No default behavior or values

Command Modes

EXEC

Command History

Release
Modification

12.2(2)T

This command was introduced.


Usage Guidelines

You cannot use this command if the interactive voice response (IVR) application using the language that you want to configure has an active call. A language that is configured under an IVR application is not necessarily in use. To determine if a call is active, use the show call application voice command.

Examples

The following example loads French (fr) into memory:

call language voice load fr

Related Commands

Command
Description

call application voice load

Loads an application.

debug voip ivr

Specifies the type of VoIP IVR debug output that you want to view.

show language voice

Displays information about configured languages and applications.


call rscmon update-timer

To change the value of the resource monitor throttle timer, use the call rscmon update-timer command in privileged EXEC mode. To revert to the default value, use the no form of this command.

call rscmon update-timer duration

no call rscmon update-timer

Syntax Description

duration

Duration of the resource monitor throttle timer, in milliseconds. Range is from 20 to 3500. The default is 2000.


Defaults

2000 milliseconds

Command Modes

Privileged EXEC

Command History

Release
Modification

12.2(2)XA

This command was introduced.

12.2(4)T

The command introduced in Cisco IOS Release 12.2(2)XA was integrated into Cisco IOS Release 12.2(4)T. This command does not support the Cisco AS5300, Cisco AS5350, and Cisco AS5400 series in this release.

12.2(2)XB1

This command was implemented on the Cisco AS5850.

12.2(11)T

This command was integrated into Cisco IOS Release 12.2(11)T.


Usage Guidelines

This command specifies the duration of the resource monitor throttle timer. When events are delivered to the resource monitor process, the throttle timer is started and the event is processed after the timer expires (unless the event is a high-priority event). The timer ultimately affects the time it takes the gateway to send Resource Availability Indicator (RAI) messages to the gatekeeper. This command allows you to vary the timer according to your needs.

Examples

The following example shows how the timer is to be configured:

Router(config)# call rscmon update-timer 1000

Related Commands

Command
Description

resource threshold

Configures a gateway to report H.323 resource availability to its gatekeeper.


call rsvp-sync

To enable synchronization between Resource Reservation Protocol (RSVP) signaling and the voice signaling protocol, use the call rsvp-sync command in global configuration mode. To disable synchronization, use the no form of this command.

call rsvp-sync

no call rsvp-sync

Syntax Description

This command has no keywords or arguments.

Defaults

Synchronization is enabled between RSVP and the voice signaling protocol (for example, H.323).

Command Modes

Global configuration

Command History

Release
Modification

12.1(3)XI

This command was introduced on the Cisco 2600 series, 3600 series, 7200 series, Cisco AS5300, Cisco AS5800, and Cisco MC3810.

12.1(5)T

This command was integrated into Cisco IOS Release 12.1(5)T.

12.2(2)XB1

This command was implemented on the Cisco AS5850.

12.2(11)T

This command was integrated into Cisco IOS Release 12.2(11)T.


Usage Guidelines

The call rsvp-sync command is enabled by default.

Examples

The following example enables synchronization between RSVP and the voice signaling protocol:

call rsvp-sync

Related Commands

Command
Description

call rsvp-sync resv-timer

Sets the timer for reservation requests.

call start

Forces the H.323 Version 2 gateway to use fast connect or slow connect procedures for a dial peer.

debug call rsvp-sync events

Displays the events that occur during RSVP synchronization.

h323 call start

Forces an H.323 Version 2 gateway to use fast connect or slow connect procedures for all VoIP services.

ip rsvp bandwidth

Enables the use of RSVP on an interface.

show call rsvp-sync conf

Displays the RSVP synchronization configuration.

show call rsvp-sync stats

Displays statistics for calls that have attempted RSVP reservation.


call rsvp-sync resv-timer

To set the timer on the terminating VoIP gateway for completing RSVP reservation setups, use the call rsvp-sync resv-timer command in global configuration mode. To restore the default value, use the no form of this command.

call rsvp-sync resv-timer seconds

no call rsvp-sync resv-timer

Syntax Description

seconds

Number of seconds in which the reservation setup must be completed, in both directions. Range is from 1 to 60. The default is 10.


Defaults

10 seconds

Command Modes

Global configuration

Command History

Release
Modification

12.1(3)XI

This command was introduced on the following platforms: Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, Cisco AS5300, Cisco AS5800, and Cisco MC3810.

12.1(5)T

This command was integrated into Cisco IOS Release 12.1(5)T.

12.2(2)XB1

This command was implemented on the Cisco AS5850.

12.2(11)T

This command was integrated into Cisco IOS Release 12.2(11)T.


Usage Guidelines

The reservation timer is started on the terminating gateway when the session protocol receives an indication of the incoming call. This timer is not set on the originating gateway because the resource reservation is confirmed at the terminating gateway. If the reservation timer expires before the RSVP setup is complete, the outcome of the call depends on the acceptable quality of service (QoS) level configured in the dial peer; either the call proceeds without any bandwidth reservation or it is released. The timer must be set long enough to allow calls to complete but short enough to free up resources. The optimum number of seconds depends on the number of hops between the participating gateways and the delay characteristics of the network.

Examples

The following example sets the reservation timer to 30 seconds:

call rsvp-sync resv-timer 30

Related Commands

Command
Description

call rsvp-sync

Enables synchronization of RSVP and the H.323 voice signaling protocol.

debug call rsvp-sync events

Displays the events that occur during RSVP synchronization.

show call rsvp-sync conf

Displays the RSVP synchronization configuration.

show call rsvp-sync stats

Displays statistics for calls that have attempted RSVP reservation.


call service stop

To shut down VoIP call service under the H.323 or SIP submode on a gateway, use the call service stop command in voice service configuration mode. To enable VoIP call service, use the no form of this command.

call service stop [forced] [maintain-registration]

no call service stop

Syntax Description

forced

(Optional) Forces the gateway to immediately terminate all in-progress calls.

maintain-registration

(Optional) Forces the gateway to remain registered with the gatekeeper.


Defaults

Call service is enabled

Command Modes

Voice service configuration

Command History

Release
Modification

12.3(1)

This command was introduced.


Usage Guidelines

The call service stop command affects call processing only for the given submode. This command overrides the functionality of the shutdown command for the affected submode.

Examples

The following example shows SIP call service being shutdown on a Cisco gateway:

enable
 configure terminal
 voice service voip
 sip
 call service stop


The following example shows H.323 call service being enabled on a Cisco gateway:

enable
 configure terminal
 voice service voip
 h323
 no call service stop

Related Commands

Command
Description

shutdown (gateway)

Shuts down call processing on the gateway.


call spike

To configure limit on the number of incoming calls received in a short period of time, use the call spike command in global configuration mode. To disable this command, use the no form of this command.

call spike call-number [steps number-of-steps size milliseconds]

no call spike

Syntax Description

call-number

Incoming call numbers for spiking threshold. Range is 1 to 2147483647.

steps number-of-steps

(Optional) Number of steps for the spiking sliding window. Range is from 3 to 10. The default is 5.

size milliseconds

(Optional) Step size in milliseconds. Range is from 100 to 250. The default is 200.


Defaults

steps—The default is 5
size—The default is 200

Command Modes

Global configuration

Command History

Release
Modification

12.2(2)XA

This command was introduced.

12.2(4)T

The command was integrated into Cisco IOS Release 12.2(4)T. This release does not support the Cisco AS5300, Cisco AS5350, and Cisco AS5400 series in this release.

12.2(2)XB1

This command was implemented on the Cisco AS5850.

12.2(4)XM

This command was implemented on Cisco 1750 and Cisco 1751 routers. Support for other Cisco platforms is not included in this release.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.

12.2(11)T

This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850 in this release.


Usage Guidelines

A call spike occurs when a large number of incoming calls arrive from the Public Switched Telephone Network (PSTN) in a short period of time (for example, 100 incoming calls in 10 milliseconds). Setting this command allows you to control the number of call requests that can be received in a configured time period.

Examples

The following configuration of the call spike command has a call-number of 30, a sliding window of 10 steps, and a step size of 2000 milliseconds.

call spike 30 steps 10 size 2000

Related Commands

Command
Description

dtmf-relay
(Voice over IP)

Specifies how an H.323 gateway relays DTMF tones between telephony interfaces and an IP network.

show call spike status

Displays the configuration of the threshold for incoming calls.


call start

To force the H.323 Version 2 gateway to use fast connect or slow connect procedures for a dial peer, use the call start command in H.323 voice-service configuration mode. To restore the system setting, use the no form of this command.

call start {fast | slow | system}

no call start

Syntax Description

fast

Gateway uses H.323 Version 2 (fast connect) procedures.

slow

Gateway uses H.323 Version 1 (slow connect) procedures.

system

Gateway defaults to the voice-service configuration.


Defaults

system

Command Modes

H.323 voice-service configuration

Command History

Release
Modification

12.1(3)XI

This command was introduced on the following platforms: Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, Cisco AS5300, Cisco AS5800, and Cisco MC3810.

12.1(5)T

This command was integrated into Cisco IOS Release 12.1(5)T.

12.2(2)XA

This command was changed to use the H.323 voice service configuration mode from the voice-class configuration mode.

12.2(4)T

This command was integrated into Cisco IOS Release 12.2(4)T.

12.2(2)XB1

This command was implemented on the Cisco AS5850.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.

12.2(11)T

This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, Cisco AS5800, and Cisco AS5850 in this release.


Usage Guidelines

In Cisco IOS Release 12.1(3)XI and later, H.323 Voice over IP (VoIP) gateways by default use H.323 Version 2 (fast connect) for all calls, including those initiating RSVP. Previously, gateways used only slow connect procedures for RSVP calls. To enable Cisco IOS Release 12.1(3)XI gateways to be backward compatible with earlier releases of Cisco IOS Release 12.1 T, the call start command allows the originating gateway to initiate calls using slow connect.

The call start command is configured as part of the voice class assigned to an individual VoIP dial peer. It takes precedence over the h323 call start command, which applies globally to all VoIP calls, unless the system keyword is selected. If the system keyword is used, the gateway defaults to the Version 2.

Examples

The following example selects slow connect for the voice class 1000:

voice service class h323 1000
 call start slow
!
dial-peer voice 210 voip
 voice-class h323 1000

The following example shows the gateway configured to use the H.323 Version 1 (Slow Connect) procedures.

h323
 call start slow

Related Commands

Command
Description

acc-qos

Selects the acceptable quality of service for a dial peer.

call rsvp-sync

Enables synchronization between RSVP and the H.323 voice signaling protocol.

call rsvp-sync resv-timer

Sets the timer for RSVP reservation setup.

debug call rsvp-sync events

Displays the events that occur during RSVP synchronization.

h323

Enables H.323 voice service configuration commands.

req-qos

Selects the desired quality of service to use in reaching a dial peer.

show call rsvp-sync conf

Displays the RSVP synchronization configuration.

show call rsvp-sync stats

Displays statistics for calls that attempted RSVP reservation.

voice class h323

Enters voice-class configuration mode and creates a voice class for H.323 attributes.


call threshold global

To enable the global resources of a gateway, use the call threshold global command in global configuration mode. To disable the global resources of the gateway, use the no form of this command.

call threshold global trigger-name low value high value [busyout] [treatment]

no call threshold global trigger-name

Syntax Description

trigger-name

Specifies the global resources on the gateway.

The trigger-name argument can be one of the following:

cpu-5sec—CPU utilization in the last 5 seconds.

cpu-avgAverage CPU utilization.

io-memI/O memory utilization.

proc-memProcessor memory utilization.

total-calls—Total number of calls.

total-mem—Total memory utilization.

low value

Value of low threshold: Range is from 1 to 100% for the utilization triggers; 1 to 10000 calls for total-calls.

high value

Value of high threshold: Range is from 1 to 100% for the utilization triggers; 1 to 10000 calls for total-calls.

busyout

(Optional) Busyout the T1/E1 channels if the resource is not available.

treatment

(Optional) Applies call treatment from the session application if the resource is not available.


Defaults

The default is busyout and treatment for global resource triggers

Command Modes

Global configuration

Command History

Release
Modification

12.2(2)XA

This command was introduced.

12.2(4)T

The command was integrated into Cisco IOS Release 12.2(4)T. Support for the Cisco AS5300, Cisco AS5350, and Cisco AS5400 is not included in this release.

12.2(2)XB1

This command was implemented on the Cisco AS5850.

12.2(4)XM

This command was implemented on Cisco 1750 and Cisco 1751 routers. Support for other Cisco platforms is not included in this release

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. Support for the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 is not included in this release.

12.2(11)T

This command is supported on the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5800 in this release.


Usage Guidelines

Use this command to enable a trigger and define associated parameters to allow or disallow new calls on the router. Action is enabled when the trigger value goes above the value specified by the high keyword and is disabled when the trigger drops below the value specified by the low keyword.

You can configure these triggers to calculate Resource Availability Indicator (RAI) information. An RAI is forwarded to a gatekeeper so that it can make call admission decisions. You can configure a trigger that is global to a router or is specific to an interface.

Examples

The following example shows how to busy out the total calls when a low of 5 or a high of 5,000 is reached:

call threshold global total-calls low 5 high 5000 busyout

The following example shows how to busy out the average CPU utilization if a low of 5 percent or a high of 65 percent is reached:

call threshold global cpu-avg low 5 high 65 busyout

Related Commands

Command
Description

call threshold (interface)

Enables interface resources of a gateway.

call threshold poll-interval

Enables a polling interval threshold for CPU or memory.

clear call threshold

Clears enabled triggers and their associated parameters.

show call threshold

Displays enabled triggers, current values for configured triggers, and number of API calls that were made to global and interface resources.


call threshold interface

To enable the interface resources of a gateway, use the call threshold interface command in global configuration mode. To disable the interface resources of the gateway, use the no form of this command.

call threshold interface interface-name interface-number int-calls low value high value

no call threshold interface interface-name interface-number int-calls

Syntax Description

interface-name

Specifies the interface name.

interface-number

Number of calls through the interface.

int-calls

Number of calls transmitted through the interface.

low value

Value of low threshold: Range is from 1 to 100% for the utilization triggers; 1 to 10000 calls for int-calls.

high value

Value of high threshold: Range is from 1 to 100% for the utilization triggers; 1 to 10000 calls for int-calls.


Defaults

No default behavior or values

Command Modes

Global configuration

Command History

Release
Modification

12.2(2)XA

This command was introduced.

12.2(4)T

The command was integrated into Cisco IOS Release 12.2(4)T. Support for the Cisco AS5300, Cisco AS5350, and Cisco AS5400 is not included in this release.

12.2(2)XB1

This command was implemented on the Cisco AS5850.

12.2(4)XM

This command was implemented on Cisco 1750 and Cisco 1751 routers. This command does not support any other Cisco platforms in this release.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. This command does not support the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.


Usage Guidelines

Use this command to enable a trigger and define associated parameters to allow or disallow new calls on the router. You can configure these triggers to calculate Resource Availability Indicator (RAI) information. An RAI is forwarded to a gatekeeper so that it can make call admission decisions. You can configure a trigger that is specific to an interface. Use the interface attribute to apply interface-related triggers.

Examples

The following example enables thresholds as low as 5 and as high as 2500 for interface calls on interface Ethernet 0/1:

call threshold interface Ethernet 0/1 int-calls low 5 high 2500

Related Commands

Command
Description

call threshold (global)

Enables global resources of a gateway.

call threshold poll-interval

Enables a polling interval threshold for CPU or memory.

clear call threshold

Clears enabled triggers and their associated parameters.

show call threshold

Displays enabled triggers, current values for configured triggers, and number of API calls that were made to global and interface resources.


call threshold poll-interval

To enable a polling interval threshold for CPU or memory, use the call threshold poll-interval command in global configuration mode. To disable this command, use the no form of this command.

call threshold poll-interval {cpu-average | memory} seconds

no call threshold poll-interval {cpu-average | memory}

Syntax Description

cpu-average

The CPU average interval, in seconds. The default is 60.

memory

The average polling interval for the memory, in seconds. The default is 5.

seconds

Window of polling interval, in seconds. Range is from 10 to 300 for the CPU average interval, and from 1 to 60 for the memory average polling interval.


Defaults

Cpu-average: 60 seconds
Memory: 5 seconds

Command Modes

Global configuration

Command History

Release
Modification

12.2(2)XA

This command was introduced.

12.2(4)T

The command was integrated into Cisco IOS Release 12.2(4)T. Support for the Cisco AS5300, Cisco AS5350, and Cisco AS5400 is not included in this release.

12.2(2)XB1

This command was implemented on the Cisco AS5850.

12.2(4)XM

This command was implemented on Cisco 1750 and Cisco 1751 routers. This release does not support any other Cisco platforms in this release.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. This release does not support the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.

12.2(11)T

This command was integrated into Cisco IOS Release 12.2(11)T and support was added for Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5800.


Examples

The following example shows how to enable a polling interval threshold for memory of 10 seconds:

call threshold poll-interval memory 10

Related Commands

Command
Description

call threshold

Enables the global resources of the gateway.

clear call threshold

Clears enabled triggers and their associated parameters.

show call threshold

Displays enabled triggers, current values for configured triggers, and number of API calls that were made to global and interface resources.


call treatment

To configure how calls should be processed when local resources are unavailable, use the call treatment command in global configuration mode. To disable call treatment, use the no form of this command.

call treatment {on | action action [value] | cause-code cause-code | isdn-reject value}

no call treatment {on | action action [value] | cause-code cause-code | isdn-reject value}

Syntax Description

on

Enables call treatment from the default session application.

action action

Action to take when call treatment is triggered, where the action argument can be the following:

hairpin—Hairpin.

playmsg—Specifies the URL of the audio file to play.

reject—Disconnects the call and pass-down cause code.

Note The hairpin keyword is not available on Cisco 1750 and Cisco 1751.

value

(Optional) (playmsg only) Specifies the audio file to play. URL format.

cause-code cause-code

Specifies the reason for the disconnection to the caller, where cause-code argument can be one of the following:

busy—Indicates that the gateway is busy.

no-QoS—Indicates that the gateway cannot provide quality of service (QoS).

no-resource—Indicates that the gateway has no resources available.

isdn-reject value

(ISDN interfaces only) Selects the ISDN reject cause code. Range is 34 to 47 (ISDN cause code for rejection).


Defaults

Treatment is inactive

Command Modes

Global configuration

Command History

Release
Modification

12.2(2)XA

This command was introduced.

12.2(4)T

The command was integrated into Cisco IOS Release 12.2(4)T. This command does not support the Cisco AS5300, Cisco AS5350, and Cisco AS5400 series in this release.

12.2(2)XB1

This command was implemented on the Cisco AS5850.

12.2(4)XM

This command was implemented on Cisco 1750 and Cisco 1751 routers. This command does not support any other Cisco platforms in this release.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on Cisco 7200 series routers. This command does not support the Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5850 in this release.

12.2(11)T

This command was integrated into Cisco IOS Release 12.2(11)T and support was added for Cisco AS5300, Cisco AS5350, Cisco AS5400, and Cisco AS5800.


Usage Guidelines

Use this command to enable a trigger and define associated parameters to disconnect (with cause code), or hairpin, or whether a message, or busy tone is played to the user.

Examples

The following example shows how to enable the call treatment feature with a "hairpin" action:

call treatment on
call treatment action hairpin

The following example shows how to enable the call treatment feature with a "playmsg" action. The file "congestion.au"plays to the caller when local resources are not available to handle the call.

call treatment on
call treatment action playmsg tftp://keyer/prompts/conjestion.au

The following example shows how to configure a call treatment cause code to reply with "no-Qos" when local resources are unavailable to process a call:

call treatment on
call treatment cause-code no-Qos

Related Commands

Command
Description

call threshold

Clears enabled triggers and their associated parameters.

clear call treatment stats

Clears the call treatment statistics.

show call treatment

Displays the call treatment configuration and statistics for handling calls on the basis of resource availability.


call-agent

To define the call agent for a Media Gateway Control Protocol (MGCP) profile, use the call-agent command in MGCP profile configuration mode. To return to the default values, use the no form of this command.

call-agent {dns-name | ip-address} [port] [service-type type] [version protocol-version]

no call-agent

Syntax Description

dns-name

Fully qualified domain name (including host portion) for the call agent. For example, "ca123.example.net".

ip-address

IP address of the call agent.

port

(Optional) User Datagram Protocol (UDP) port number over which the gateway sends messages to the call agent. Range is from 1025 to 65535.

The default call-agent UDP port is 2727 for MGCP 1.0, Network-based Call Signaling (NCS) 1.0, and Trunking Gateway Control Protocol (TGCP) 1.0.

The default call-agent UDP port is 2427 for MGCP 0.1 and Simple Gateway Control Protocol (SGCP).

service-type type

(Optional) Protocol service type valid values for the type argument are mgcp, ncs, sgcp, and tgcp. The default service type is mgcp.

version protocol-version

(Optional) Version number of the protocol. Valid values follow:

service-type mgcp—0.1, 1.0

service-type ncs—1.0

service-type sgcp—1.1, 1.5

service-type tgcp—1.0

The default service type and version is mgcp 0.1.


Defaults

The default call-agent UDP port is 2727 for MGCP 1.0, Network-based Call Signaling (NCS) 1.0, and Trunking Gateway Control Protocol (TGCP) 1.0.
The default call-agent UDP port is 2427 for MGCP 0.1 and Simple Gateway Control Protocol (SGCP).
The default service type and version is MGCP 0.1.

Command Modes

MGCP profile configuration

Command History

Release
Modification

12.2(2)XA

This command was introduced.

12.2(4)T

This command was integrated into Cisco IOS Release 12.2(4)T.

12.2(11)T

This command was implemented on the Cisco AS5300 and Cisco AS5850.


Usage Guidelines

This command is used when values for a MGCP profile are configured.

Call-agent configuration for an MGCP profile (with this command) and global call-agent configuration (with the mgcp call-agent command) are mutually exclusive; the first to be configured on an endpoint blocks configuration of the other on the same endpoint.

Identifying call agents by Domain Name System (DNS) name rather than by IP address in the call-agent command provides call-agent redundancy, because a DNS name can have more than one IP address associated with it. If a call agent is identified by a DNS name and a message from the gateway fails to reach the call agent, the max1 lookup and max2 lookup commands enable a search from the DNS lookup table for a backup call agent at a different IP address.

The port argument configures the call agent port number (the UDP port over which the gateway sends messages to the call agent). The reverse, or the gateway port number (the UDP port over which the gateway receives messages from the call agent), is configured by specifying a port number in the mgcp command.

The service type mgcp supports the Restart In Progress (RSIP) error messages sent by the gateway if the mgcp sgcp restart notify command is enabled. The service type sgcp ignores the RSIP messages.

Examples

The following example defines a call agent for the MGCP profile named "tgcp_trunk":

Router(config)# mgcp profile tgcp_trunk
Router(config-mgcp-profile)# call-agent 10.13.93.3 2500 service-type tgcp version 1.0

Related Commands

Command
Description

max1 lookup

Enables DNS lookup of the MGCP call agent address when the suspicion threshold value is reached.

max2 lookup

Enables DNS lookup of the MGCP call agent address when the disconnect threshold value is reached.

mgcp

Starts and allocates resources for the MGCP daemon.

mgcp call-agent

Configures the address of the call agent (media gateway controller).

mgcp profile

Initiates MGCP profile mode to create and configure a named MGCP profile associated with one or more endpoints or to configure the default profile.


call-block (dial-peer)

To enable blocking of incoming calls, use the call-block command in dial-peer configuration mode. To return to the default value, use the no form of this command.

call-block {disconnect-cause incoming {call-reject | invalid-number | unassigned-number | user-busy} | translation-profile incoming name}

no call-block {disconnect-cause incoming {call-reject | invalid-number | unassigned-number | user-busy} | translation-profile incoming name}

Syntax Description

disconnect-cause incoming

Associates a disconnect cause of incoming calls.

call-reject

Specifies call rejection as the cause for blocking a call during incoming call-number translation.

invalid-number

Specifies invalid number as the cause for blocking a call during incoming call-number translation.

unassigned-number

Specifies unassigned number as the cause for blocking a call during incoming call-number translation.

user-busy

Specifies busy as the cause for blocking a call during incoming call-number translation.

translation-profile incoming

Associates the translation profile for incoming calls.

name

Name of the translation profile.


Defaults

Disconnect cause: No Service (once the call-blocking translation profile is defined)
Translation profile: No default behavior or values

Command Modes

Dial-peer configuration

Command History

Release
Modification

12.2(11)T

This command was introduced.


Usage Guidelines

The incoming call can be blocked from the gateway if one of the call numbers (calling, called, or redirect) is matched with the reject translation rule of the incoming call-blocking translation profile.

The cause value is returned to the source of the call when a call is blocked during the incoming call-number translation.

This command is supported in POTS, VoIP, VoFR, and VoATM dial-peer configuration. For VoATM, only AAL5 calls are supported.

Examples

The following example assigns the translation profile "westcoast" to be used for incoming calls and returns the message "invalid number" as a cause for blocked calls:

Router(config)# dial-peer voice 5 pots
Router(config-dial-peer)# call-block translation-profile incoming westcoast
Router(config-dial-peer)# call-block disconnect-cause incoming invalid-number

Related Commands

Command
Description

dial-peer voice

Initiates the dial-peer voice configuration mode.

voice translation-profile

Defines a translation profile for voice calls.

voice translation-rule

Defines a translation rule for voice calls.


call-denial

The call-denial command is replaced by the call threshold global command. See the call threshold global command for more information.

called-number (dial-peer)

To enable an incoming Voice over Frame Relay (VoFR) call leg to get bridged to the correct plain old telephone service (POTS) call leg when a static FRF.11 trunk connection is used, use the called-number command in dial peer configuration mode. To disable a static trunk connection, use the no form of this command.

called-number string

no called-number

Syntax Description

string

A string of digits, including wildcards, that specifies the telephone number of the voice port dial peer.


Defaults

This command is disabled

Command Modes

Dial peer configuration

Command History

Release
Modification

12.0(4)T

This command was introduced on the Cisco 2600 series and Cisco 3600 series.


Usage Guidelines

This command applies to the Cisco 2600 and Cisco 3600 series routers only. It is ignored on the Cisco MC3810 and on the Cisco 7200 series.

The called-number command is used only when the dial peer type is VoFR and you are using the frf11-trunk (FRF.11) session protocol. It is ignored at all times on the Cisco MC3810 multiservice concentrator and on all other platforms when using the Cisco-switched session protocol.

Because FRF.11 does not provide any end-to-end messaging to manage a trunk, the called-number command is necessary to allow the router to establish an incoming trunk connection. The E.164 number is used to find a matching dial peer during call setup.

Examples

The following example shows how to configure a Cisco 2600 series routers or 3600 series router for a static FRF.11 trunk connection to a specific telephone number (555-2150), beginning in global configuration mode:

voice-port 1/0/0
 connection trunk 55Router0
 exit

dial-peer voice 100 pots
 destination pattern 5552150
 exit

dial-peer voice 200 vofr
 session protocol frf11-trunk
 called-number 5552150
 destination pattern 55Router0

Related Commands

Command
Description

codec (dial peer)

Specifies the voice coder rate of speech for a VoFR dial peer.

connection

Specifies a connection mode for a voice port.

destination-pattern

Specifies either the prefix, the full E.164 telephone number, or an ISDN directory number (depending on the dial plan) to be used for a dial peer.

dtmf-relay (VoFR)

Enables the generation of FRF.11 Annex A frames for a dial peer.

fax-rate

Establishes the rate at which a fax is sent to the specified dial peer.

preference

Indicates the preferred order of a dial peer within a rotary hunt group.

session protocol

Establishes a session protocol for calls between the local and remote routers via the packet network.

session target

Specifies a network-specific address for a specified dial peer or destination gatekeeper.

signal-type

Sets the signaling type to be used when connecting to a dial peer.

vad (dial peer)

Enables voice-activated dialing (VAD) for the calls using a particular dial peer.


caller-id

To enable caller ID, use the caller-id command in dial peer configuration mode. To disable caller ID, use the no form of the command.

caller-id

no caller-id

Syntax Description

This command contains no arguments or keywords.

Defaults

Caller ID is disabled

Command Modes

Dial peer configuration

Command History

Release
Modification

12.1.(2)XF

This command was introduced on the Cisco 800 series routers.

12.1(5)T

This command was integrated into Cisco IOS Release 12.1(5)T.


Usage Guidelines

This command is available on Cisco 800 series routers that have plain old telephone service (POTS) ports. The command is effective only if you subscribe to caller ID service. If you enable caller ID on a router without subscribing to the caller ID service, caller ID information does not appear on the telephone display.

The configuration of caller ID must match the device connected to the POTS port. That is, if a telephone supports the caller ID feature, use the command caller-id to enable the feature. If the telephone does not support the caller ID feature, use the command default or disable the caller ID feature. Odd ringing behavior might occur if the caller ID feature is disabled when it is a supported telephone feature or enabled when it is not a supported telephone feature.


Note Specific hardware is required to provide full support for the Caller ID features. To determine support for these features in your configuration, review the appropriate hardware documentation and data sheets. This information is available on Cisco.com.


Examples

The following example enables a router to use the caller ID feature:

dial-peer voice 1 pots
 caller-id

Related Commands

Command
Description

block-caller

Configures call blocking on caller ID.

debug pots csm csm

Activates events from which an application can determine and display the status and progress of calls to and from POTS ports.

isdn i-number

Configures several terminal devices to use one subscriber line.

pots call-waiting

Enables local call waiting on a router.

registered-caller ring

Configures the Nariwake service-registered caller ring cadence.


caller-id alerting dsp-pre-alloc

To statically allocate a digital signal processor (DSP) resource for receiving caller ID information for on-hook (Type 1) Caller ID at a receiving Foreign Exchange Office (FXO) voice port, use the caller-id alerting dsp-pre-alloc command in voice-port configuration mode. To disable the command's effect, use the no form of this command.

caller-id alerting dsp-pre-alloc

no caller-id alerting dsp-pre-alloc

Syntax Description

This command contains no keywords or arguments.

Defaults

No pre-allocation of DSP resources

Command Modes

Voice-port configuration

Command History

Release
Modification

12.1(2)XH

This command was introduced on the Cisco MC3810, Cisco 2600 series, and Cisco 3600 series.

12.1(3)T

This command was integrated into Cisco IOS Release 12.1(3)T.


Usage Guidelines

The caller-id alerting dsp-pre-alloc command may be required on an FXO port if the central office uses line polarity reversal to signal the start of Caller-ID information transmission. Pre-allocating a DSP allows the DSP to listen for Caller-ID information continuously without requiring an alerting signal from the CO.

This command is the FXO counterpart to the caller-id alerting line-reversal command, which is applied to the Foreign Exchange Station (sending) end of the Caller-ID call.


Note Specific hardware is required to provide full support for the Caller ID features. To determine support for these features in your configuration, review the appropriate hardware documentation and data sheets. This information is available on Cisco.com.


Examples

The following example configures a voice port on a Cisco 2600 series or Cisco 3600 series router where Caller-ID information is received:

voice-port 1/0/1
  cptone US
  caller-id enable
  caller-id alerting line-reversal
  caller-id alerting dsp-pre-alloc

The following example configures a voice port on a Cisco MC3810 where Caller-ID information is received:

voice-port 1/0
  cptone northamerica
  caller-id enable
  caller-id alerting line-reversal
  caller-id alerting dsp-pre-alloc

Related Commands

Command
Description

caller-id alerting line-reversal

Sets the line-reversal method of Caller-ID call alerting.


caller-id alerting line-reversal

To set the line-reversal alerting method for Caller-ID information for on-hook (Type 1) Caller ID at a sending Foreign Exchange Station (FXS) voice port, use the caller-id alerting line-reversal command in voice-port configuration mode. To disable the command's effect, use the no form of this command.

caller-id alerting line-reversal

no caller-id alerting line-reversal

Syntax Description

This command has no keywords or arguments.

Defaults

No line-reversal alert

Command Modes

Voice-port configuration

Command History

Release
Modification

12.1(2)XH

This command was introduced on the Cisco 2600 series, and Cisco 3600 series and Cisco MC3810.

12.1(3)T

This command was integrated into Cisco IOS Release 12.1(3)T.


Usage Guidelines

This command is only required when the telephone device attached to an FXS port requires the line-reversal method to signal the start of a Caller-ID transmission. Use it on FXS voice ports that send Caller-ID information.

This command is the FXS counterpart to the caller-id alerting dsp-pre-alloc command, which is applied to the FXO (receiving) end of the Caller-ID call with the line-reversal alerting method.


Note Specific hardware is required to provide full support for the Caller ID features. To determine support for these features in your configuration, review the appropriate hardware documentation and data sheets. This information is available on Cisco.com.


Examples

The following example configures a voice port on a Cisco 2600 or 3600 series router from which Caller-ID information is sent:

voice-port 1/0/1
   cptone US
   station name  A. Person
   station number 4085551111
   caller-id alerting line-reversal
   caller-id alerting dsp-pre-alloc

The following example configures a voice port on a Cisco MC3810 from which Caller-ID information is sent:

voice-port 1/0
   cptone northamerica
   station name  A. Person
   station number 4085551111
   caller-id alerting line-reversal
   caller-id alerting dsp-pre-alloc

Related Commands

Command
Description

caller-id alerting dsp-pre-alloc

At the receiving end of a line-reversal alerting Caller-ID call, pre-allocates DSPs for caller ID calls.


caller-id alerting pre-ring

To set a 250-millisecond pre-ring alerting method for caller ID information for on-hook (Type 1) Caller ID at a sending Foreign Exchange Station (FXS) voice port, use the caller-id alerting pre-ring command in voice-port configuration mode. To disable the command, use the no form of this command.

caller-id alerting pre-ring

no caller-id alerting pre-ring

Syntax Description

This command has no keywords or arguments.

Defaults

No pre-ring alert

Command Modes

Voice-port configuration

Command History

Release
Modification

12.1(2)XH

This command was introduced on the Cisco MC3810, Cisco 2600 series, and Cisco 3600 series.

12.1(3)T

This command was integrated into Cisco IOS Release 12.1(3)T.


Usage Guidelines

This command is required only when the telephone device attached to an FXS port requires the pre-ring (immediate ring) method to signal the start of caller ID transmission. Use it on FXS voice ports that send caller ID information. This command allows the FXS port to send a short pre-ring preceding the normal ring cadence. On an FXO port, an incoming pre-ring (immediate ring) is simply counted as a normal ring using the caller-id alerting ring command.


Note Specific hardware is required to provide full support for the Caller ID features. To determine support for these features in your configuration, review the appropriate hardware documentation and data sheets. This information is available on Cisco.com.


Examples

The following example configures a voice port on a Cisco 2600 series or Cisco 3600 series router from which caller ID information is sent:

voice-port 1/0/1
   cptone US
   station name  A. Person
   station number 4085551111
   caller-id alerting pre-ring

The following example configures a voice port on a Cisco MC3810 from which caller ID information is sent:

voice-port 1/0
   cptone northamerica
   station name A. Person
   station number 4085551111
   caller-id alerting pre-ring 1

Related Commands

Command
Description

caller-id alerting line-reversal

Enables caller ID operation and sets the line-reversal alerting type at an FXS port.

caller-id alerting ring

Enables caller ID operation and sets an alerting ring type at an FXO or FXS port.


caller-id alerting ring

To set the ring-cycle method for receiving caller ID information for on-hook (Type 1) Caller ID at a receiving Foreign Exchange Office (FXO) or a sending Foreign Exchange Station (FXS) voice port, use the caller-id alerting ring command in voice-port configuration mode. To set the command to the default, use the no form of this command.

caller-id alerting ring {1 | 2}

no caller-id alerting ring

Syntax Description

1

Use this setting if your telephone service provider specifies it to provide caller ID alerting (display) after the first ring at the receiving station. This is the most common setting.

2

Use this setting if your telephone service provider specifies it to provide caller ID alerting (display) after the second ring. This setting is used in Australia, where the caller ID information is sent following two short rings (double-pulse ring).


Defaults

1

Command Modes

Voice-port configuration

Command History

Release
Modification

12.1(2)XH

This command was introduced on the Cisco 2600 series, Cisco 3600 series and Cisco MC3810.

12.1(3)T

This command was integrated into Cisco IOS Release 12.1(3)T.


Usage Guidelines

This setting is determined by the Bellcore/Telcordia or ETSI standard that your telephone service provider uses for caller ID. Use it on FXO loop-start and ground-start voice ports where caller ID information arrives and on FXS voice ports from which caller ID information is sent.

This setting must match on the sending and receiving ends on both ends of the telephone line connection.


Note Specific hardware is required to provide full support for the Caller ID features. To determine support for these features in your configuration, review the appropriate hardware documentation and data sheets. This information is available on Cisco.com.


Examples

The following example configures a Cisco 2600 series or Cisco 3600 series router voice port where caller ID information is received:

voice-port 1/0/1
   cptone US
   caller-id alerting ring 1

The following example configures a Cisco 2600 series or Cisco 3600 series router voice port from which caller ID information is sent:

voice-port 1/0/1
   cptone northamerica
   station name A. Person
   station number 4085551111
   caller-id alerting ring 1

The following example configures a Cisco MC3810 voice port where caller ID information is received:

voice-port 1/0
   cptone northamerica
   caller-id alerting ring 1

The following example configures a Cisco MC3810 voice port from which caller ID information is sent:

voice-port 1/0
   cptone northamerica
   station name A. Person
   station number 4085551111
   caller-id alerting ring 1

Related Commands

Command
Description

caller-id alerting line-reversal

Enables caller ID operation and sets the line-reversal alerting type at an FXS port.

caller-id alerting pre-ring

Enables caller ID operation and sets the pre-ring alerting method at an FXS port.


caller-id attenuation

To set the attenuation for caller ID at a receiving Foreign Exchange Office (FXO) voice port, use the caller-id attenuation command in voice-port configuration mode. To set the command to the default, use the no form of this command.

caller-id attenuation [attenuation]

no caller-id attenuation

Syntax Description

attenuation

Specifies the attenuation, in decibels (dB). Range is from 0 to 64. The default is 14.


Defaults

The default value is 14 decibels (dB), signal level of -14 dBm

Command Modes

Voice-port configuration

Command History

Release
Modification

12.1(2)XH

This command was introduced on and Cisco 2600 series, Cisco 3600 series and the Cisco MC3810.

12.1(3)T

This command was integrated into Cisco IOS Release 12.1(3)T.


Usage Guidelines

Use this setting to specify the attenuation for a caller ID FXO port. If the setting is not used, the attenuation is set to 14 decibels (dB), signal level of -14 dBm.


Note Specific hardware is required to provide full support for the Caller ID features. To determine support for these features in your configuration, review the appropriate hardware documentation and data sheets. This information is available on Cisco.com.


Examples

The following example configures a Cisco 2600 series or Cisco 3600 series router voice port where caller ID information is received:

voice-port 1/0/1
   cptone US
   caller-id attenuation 0

The following example configures a Cisco MC3810 voice port where caller ID information is received:

voice-port 1/0
   cptone northamerica
   caller-id attenuation 0

caller-id block

To request the blocking of the display of caller ID information at the far end of a call from calls originated at a Foreign Exchange Station (FXS) port, use the caller-id block command in voice-port configuration mode at the originating FXS voice port. To allow the display of caller ID information, use the no form of this command.

caller-id block

no caller-id block

Syntax Description

This command has no keywords or arguments.

Defaults

No blocking of caller ID information

Command Modes

Voice-port configuration

Command History

Release
Modification

12.1(2)XH

This command was introduced on and Cisco 2600 series, Cisco 3600 series and the Cisco MC3810.

12.1(3)T

This command was integrated into Cisco IOS Release 12.1(3)T.


Usage Guidelines

This command is used on FXS voice ports that are used to originate on-net telephone calls. This command affects all calls sent to a far-end FXS station from the configured originating FXS station. Calling number and called number are provided in the H.225 setup message for VoIP, through the H.225 Octet 3A field. Calling name information is included in a display information element.


Note Cisco-switched calls using Voice over Frame Relay (VoFR) and Voice over ATM (VoATM) carry calling party information in the Cisco proprietary setup message. For standards-based, point-to-point VoFR (FRF.11) trunks where transparent signaling is applied for FXS-to-FXO calls, only pass-through of in-band Automatic Number Identification (ANI) is supported. ANI information is always unblocked for these communications. Interface technology using transparent channel associated signaling (CAS) can support only ANI through Feature Group D (in-band MF signaling). The Caller ID feature cannot be used with fixed point-to-point trunk connects created using the connection trunk command.


This command applies to the Cisco MC3810 and to Cisco 2600 series and 3600 series routers.


Note Specific hardware is required to provide full support for the Caller ID features. To determine support for these features in your configuration, review the appropriate hardware documentation and data sheets. This information is available on Cisco.com.


Examples

The following example configures a Cisco 2600 series or Cisco 3600 series router voice port from which caller ID information is sent:

voice-port 1/0/1
   cptone US
   station name A. Person
   station number 4085551111
   caller-id block

The following example configures a Cisco MC3810 voice port from which caller ID information is sent:

voice-port 1/0
   cptone northamerica
   station name A. Person
   station number 4085551111
   caller-id block

Related Commands

Command
Description

caller-id enable

Enables caller ID operation.


caller-id block (ephone-dn)

To configure caller-ID blocking for outbound calls, use the caller-id block command in ephone-dn configuration mode. To disable caller-ID blocking, use the no form of this command.

caller-id block

no caller-id block

Syntax Description

This command has no arguments or keywords.

Defaults

Caller ID is not blocked on calls originating from a Cisco IP phone

Command Modes

Ephone-dn configuration

Command History

Release
Modification

12.1(5)YD

This command was introduced on the following platforms: Cisco 2600 series and Cisco 3600 series, and Cisco IAD2420 series.

12.2(2)XT

This command was implemented on the Cisco 1750 and Cisco 1751.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725 and Cisco 3745 routers.

12.2(8)T1

This command was implemented on the Cisco 2600-XM and Cisco 2691.

12.2(11)T

This command was implemented on the Cisco 1760.


Usage Guidelines

The caller-id block command sets caller-ID blocking for outbound calls originating from the specific directory number (ephone-dn). This command requests that the far-end gateway device block display of the calling party information, for calls received by the far-end gateway from the ephone-dn. This command does not effect the ephone-dn calling party information display for inbound calls received by the ephone-dn.

Examples

The following example shows how to set caller ID blocking for the directory number 5001:

Router(config) ephone-dn 1
Router(config-ephone-dn)# number 5001
Router(config-ephone-dn)# caller-id block

Related Commands

Command
Description

ephone

Enters ephone configuration mode.

ephone-dn

Enters ephone-dn configuration mode.


caller-id enable

To allow the sending or receiving of caller-ID information, use the caller-id enable command in voice-port configuration mode at the sending foreign exchange station (FXS) voice port or the receiving foreign exchange office (FXO) voice port. To disable the sending or receiving of caller-ID information, use the no form of this command.

caller-id enable

no caller-id enable

Syntax Description

This command has no keywords or arguments.

Defaults

No sending or receiving of caller-ID information

Command Modes

Voice-port configuration

Command History

Release
Modification

12.1(2)XH

This command was introduced on and Cisco 2600 series, Cisco 3600 series and the Cisco MC3810.

12.1(3)T

This command was integrated into Cisco IOS Release 12.1(3)T.


Usage Guidelines

This command applies to FXS voice ports that send caller-ID information and to FXO ports that receive caller-ID information. Calling number and called number are provided in the H.225.0 setup message for VoIP, through the H.225.0 Octet 3A field. Calling name information is included in a display information element.


Note Cisco-switched calls using Voice over Frame Relay (VoFR) and Voice over ATM (VoATM) carry calling party information in the Cisco proprietary setup message. For standards-based, point-to-point VoFR (FRF.11) trunks where transparent signaling is applied for FXS-to-FXO calls, only pass-through of in-band automatic number identification (ANI) is supported. ANI information is always unblocked for these communications. Interface technology using transparent channel-associated signaling (CAS) can support only ANI through Feature Group D (in-band multifrequency signaling). The Caller ID feature cannot be used with fixed point-to-point trunk connections created using the connection trunk command.


If the station name, station number, or a caller-id alerting command is configured on the voice port, these automatically enable caller ID, and the caller-id enable command is not necessary.


Note The no form of this command also clears all other caller-ID configuration settings for the voice port.


This command applies to the Cisco MC3810 and to Cisco 2600 and Cisco 3600 series routers.


Note Specific hardware is required to provide full support for the caller-ID features. To determine support for these features in your configuration, review the appropriate hardware documentation and data sheets. This information is available on Cisco.com.


Examples

The following example configures a Cisco 2600 series or Cisco 3600 series router voice port at which caller-ID information is received:

voice-port 1/0/1
   cptone US
   caller-id enable

The following example configures a Cisco 2600 series or Cisco 3600 series router voice port from which caller-ID information is sent:

voice-port 1/0/1
   cptone northamerica
   station name A. Person
   station number 4085551111
   caller-id enable

The following example configures a Cisco MC3810 voice port where caller-ID information is received:

voice-port 1/0
   cptone northamerica
   caller-id enable

The following example configures a Cisco MC3810 voice port from which caller-ID information is sent:

voice-port 1/0
   cptone northamerica
   station name A. Person
   station number 4085551111
   caller-id enable

Related Commands

Command
Description

caller-id alerting line-reversal

Enables caller ID operation and sets the line-reversal alerting type at an FXS port.

caller-id alerting pre-ring

Enables caller ID operation and sets the pre-ring alerting method at an FXS port.

caller-id alerting ring

Enables caller ID operation and sets an alerting ring type at an FXO or FXS port.

caller-id block

Disables the sending of caller ID information from an FXS port.

station name

Enables caller ID operation and sets the name sent from an FXS port.

station number

Enables caller ID operation and sets the number sent from an FXS port.


caller-number

To associate a type of ring cadence with a specific caller ID, use the caller-number command in dial-peer voice configuration mode. To disable the type of ring cadence for a specific caller ID, use the no form of this command.

caller-number number ring cadence

no caller-number number ring cadence

Syntax Description

number

Caller ID for which the user wishes to set the cadence. Twenty numbers along with their respective cadences may be set for each of the plain old telephone service (POTS) ports.

ring cadence

Ring cadence level. There are three cadence levels (0, 1, and 2), which differ in duration and cadence. The levels are as follows:

0—The ring cadence is 1 second on and 2 seconds off (NTT-defined regular ring).

1—The ring cadence is 0.25 seconds on, 0.2 seconds off, 0.25 seconds on, and 2.3 seconds off (NTT-defined nonregular ring).

2—The ring cadence is 0.5 seconds on, 0.25 seconds off, 0.25 seconds on, and 2 seconds off (Cisco-defined nonregular ring).


Defaults

The router does not associate any caller ID with a cadence level. Therefore, there is no distinctive ring.

Command Modes

Dial-peer voice configuration

Command History

Release
Modification

12.2(8)T

This command was introduced on Cisco 803, Cisco 804, and Cisco 813 routers.


Usage Guidelines

You can enter the caller-number command for each POTS port. There is a maximum of 20 caller IDs that can be associated with distinct ring cadences. After 20 numbers per port have been set, you cannot set more numbers (and their ring cadences) for that port until you have removed any of the numbers that have already been set. To remove already-set numbers and their ring cadences, use the no form of the caller-number command.

The command must be set within each dial peer. Because there are 6 dial peers available, you can specify 20 caller IDs per port, for a maximum of 120 caller ID numbers.


Note If you have already subscribed to Nariwake service, the priority goes to the Nariwake caller ID cadence.


To disable distinctive ringing based on a caller ID number, configure the no caller-number command. Disabling the ringing removes the specific cadence that has been set for that particular number. If you have set 20 numbers and their ring cadences, you need to set the no caller-number command for each of the 20 numbers.

Use the show running-config command to check distinctive ringing status.

Examples

The following output examples show that three caller ID numbers and their ring cadences have been set for POTS port 1 and that five caller ID numbers and their ring cadences have 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 2
 caller-number 2222222 ring 1
 caller-number 3333333 ring 1

dial-peer voice 2 pots
 destination-pattern 5552222
 port 2
 no call-waiting
 ring 0
 volume 2
 caller-number 4444444 ring 1
 caller-number 6666666 ring 2
 caller-number 7777777 ring 0
 caller-number 8888888 ring 1
 caller-number 9999999 ring 2

Related Commands 

Command
Description

call-waiting

Enables call waiting.

volume

Configures the receiver volume level in the router.


call-forward all (ephone-dn)

To configure call forwarding so that all the incoming calls on one of the lines of a Cisco IP phone are forwarded from that telephone to another telephone, use the call-forward all command in ephone-dn configuration mode. To disable call forwarding, use the no form of this command.

call-forward all directory-number

no call-forward all [directory-number]

Syntax Description

directory-number

Selected directory number. Represents a fully qualified E.164 number.


Defaults

No default behavior or values

Command Modes

Ephone-dn configuration

Command History

Release
Modification

12.1(5)YD

This command was introduced on the following platforms: Cisco 2600 series and Cisco 3600 series, and Cisco IAD2420 series.

12.2(2)XT

This command was implemented on the Cisco 1750 and Cisco 1751.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725 and Cisco 3745.

12.2(8)T1

This command was implemented on the Cisco 2600-XM and Cisco 2691.

12.2(11)T

This command was implemented on the Cisco 1760.


Usage Guidelines

The call forwarding mechanism is applied to the individual telephone line (directory number) and cannot be configured for individual Cisco IP phones.


Note The call-forward all command takes precedence over the call-forward busy and call-forward noan commands.


Examples

The following example shows how to set call forwarding of all calls on line 1, directory number 5001, to directory number 5005. All incoming calls destined for extension 5001 are forwarded to another Cisco IP phone with the extension number 5005:

Router(config)# ephone-dn 1
Router(config-ephone-dn)# number 5001
Router(config-ephone-dn)# call-forward all 5005

Related Commands

Command
Description

call-forward busy

Configures call forwarding to another number when a Cisco IP phone is busy.

call-forward noan

Configures call forwarding to another number when no answer is received from a Cisco IP phone.

ephone

Enters ephone configuration mode.

ephone-dn

Enters ephone-dn configuration mode.


call-forward busy (cm-fallback)

To configure call forwarding to another number when a Cisco IP phone is busy, use the call-forward busy command in call-manager-fallback configuration mode. To disable call forwarding, use the no form of this command.

call-forward busy directory-number

no call-forward busy [directory-number]

Syntax Description

directory-number

Selected directory number. Represents a fully qualified E.164 number.


Defaults

No default behavior or values

Command Modes

Call-manager-fallback configuration

Command History

Release
Modification

12.2(2)XT

This command was introduced on the following platforms: Cisco 1750, Cisco 1751, Cisco 2600 series, Cisco 3600 series, and Cisco IAD2420 series.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725, Cisco 3745, and Cisco MC3810-V3.

12.2(8)T1

This command was implemented on the Cisco 2600-XM and Cisco 2691.

12.2(11)T

This command was implemented on the Cisco 1760.


Usage Guidelines

The call-forward busy command configures call forwarding to another number when a Cisco IP phone is busy. The call forwarding mechanism is applied globally to all phones that register during fallback.

Examples

The following example shows how to set call forwarding to extension number 5005 on busy for an incoming call to any IP phone extension number:

Router(config)# call-manger-fallback 
Router(config-cm-fallback)# call-forward busy 5005

Note You can forward an incoming Voice over IP (VoIP) call only to destination numbers local to the router. VoIP calls can not be forwarded to an alternate (on-net) VoIP destination.


Related Commands

Command
Description

call-forward noan

Configures call forwarding to another number when no answer is received from a Cisco IP phone.

call-manager-fallback

Enables SRS Telephony feature support and enters call-manager-fallback configuration mode.


call-forward busy (ephone-dn)

To configure call forwarding to another number when the Cisco IP phone is busy, use the call-forward busy command in ephone-dn configuration mode. To disable call forwarding, use the no form of this command.

call-forward busy directory-number

no call-forward busy [directory-number]

Syntax Description

directory-number

Selected directory number. Represents a fully qualified E.164 number.


Defaults

No default behavior or values

Command Modes

Ephone-dn configuration

Command History

Release
Modification

12.1(5)YD

This command was introduced on the following platforms: Cisco 2600 series and Cisco 3600 series, and Cisco IAD2420 series IADs.

12.2(2)XT

This command was implemented on the Cisco 1750 and Cisco 1751.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725 and Cisco 3745.

12.2(8)T1

This command was implemented on the Cisco 2600-XM and Cisco 2691.

12.2(11)T

This command was implemented on the Cisco 1760.


Usage Guidelines

The call forwarding mechanism is applied to the individual telephone line (directory number) and cannot be configured individual Cisco IP phones.


Note The call-forward all command takes precedence over the call-forward busy and call-forward noan commands.


Examples

The following example shows how to set call forwarding of incoming calls to directory number 5005 when line 1, directory number 5001, is busy:

Router(config)# ephone-dn 1
Router(config-ephone-dn)# number 5001
Router(config-ephone-dn)# call-forward busy 5005

Related Commands

Command
Description

call-forward all

Configures call forwarding for all the incoming calls on one of the lines of a Cisco IP phone.

call-forward noan

Configures call forwarding to another number when no answer is received from a Cisco IP phone.

ephone

Enters ephone configuration mode.

ephone-dn

Enters ephone-dn configuration mode.


call-forward noan (cm-fallback)

To configure call forwarding to another number when no answer is received from a Cisco IP phone, use the call-forward noan command in call-manager-fallback configuration mode. To disable call forwarding, use the no form of this command.

call-forward noan directory-number timeout seconds

no call-forward noan [directory-number]

Syntax Description

directory-number

Selected directory number. Represents a fully qualified E.164 number.

timeout

Ringing no answer timeout duration. It is the waiting time before the call is forwarded to another phone.

seconds

Time set, in seconds before call forwarding starts. Range is from 3 to 60000. There is no default value.


Defaults

No default behavior or values

Command Modes

Call-manager-fallback configuration

Command History

Release
Modification

12.2(2)XT

This command was implemented on the Cisco 1750, Cisco 1751, Cisco 2600 series, Cisco 3600 series, and Cisco IAD2420 series IADs.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725 and Cisco 3745.

12.2(8)T1

This command was implemented on the Cisco 2600-XM and Cisco 2691.

12.2(11)T

This command was implemented on the Cisco 1760.


Usage Guidelines

The call-forward noan command configures call forwarding to another number when no answer is received from a Cisco IP phone. The call forwarding mechanism is applied globally to all phones that register during fallback. The timeout keyword sets the waiting time before the call is forwarded to another phone.

Examples

The following example shows how to set call forwarding of incoming calls to directory number 5005 when line 1, directory number 5001, does not answer. The timeout before the call is forwarded to the directory number 5005 is set for 10 seconds:

Router(config)# call-manager-fallback
Router(config-cm-fallback)# call-forward noan 5005 timeout 10

Note An incoming Voice over IP (VoIP) call can be forwarded only to destination numbers local to the router. VoIP calls cannot be forwarded to an alternate (on-net) VoIP destination.


Related Commands

Command
Description

call-forward busy

Configures call forwarding to another number when a Cisco IP phone is busy.

call-manager-fallback

Enables SRS Telephony feature support and enters call-manager-fallback configuration mode.


call-forward noan (ephone-dn)

To configure call forwarding to another number when no answer is received from a Cisco IP phone, use the call-forward noan command in ephone-dn configuration mode. To disable call forwarding, use the no form of this command.

call-forward noan directory-number timeout seconds

no call-forward noan [directory-number]

Syntax Description

directory-number

Selected directory number. Represents a fully qualified E.164 number.

timeout

Ringing no answer timeout duration. It is the waiting time before the call is forwarded to another phone.

seconds

Time, set in seconds for the call forwarding to start. Range is from 3 to 60000. There is no default value.


Defaults

No default behavior or values

Command Modes

Ephone-dn configuration

Command History

Release
Modification

12.1(5)YD

This command was introduced on the following platforms: Cisco 2600 series and Cisco 3600 series, and Cisco IAD2420 series.

12.2(2)XT

This command was implemented on the Cisco 1750 and Cisco 1751.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725 and Cisco 3745.

12.2(8)T1

This command was implemented on the Cisco 2600-XM and Cisco 2691.

12.2(11)T

This command was implemented on the Cisco 1760.


Usage Guidelines

The call forwarding mechanism is applied to the individual telephone line (directory number) and cannot be configured for individual Cisco IP phones.

Examples

The following example shows how to set call forwarding of incoming calls to directory number 5005 when line 1, directory number 5001, does not answer. The timeout before the call is forwarded to the directory number 5005 is set for 10 seconds:

Router(config)# ephone-dn 1
Router(config-ephone-dn)# number 5001
Router(config-ephone-dn)# call-forward noan 5005 timeout 10

Related Commands

Command
Description

call-forward all

Configures call forwarding for all the incoming calls on one of the lines of a Cisco IP phone.

call-forward busy

Configures call forwarding to another number when a Cisco IP phone is busy.

ephone

Enters ephone configuration mode.

ephone-dn

Enters ephone-dn configuration mode.


call-forward pattern

To specify a pattern for calling-party numbers that are able to support the ITU-T H.450.3 standard for call forwarding, use the call-forward pattern command in telephony-service configuration mode. To remove the pattern, use the no form of this command.

call-forward pattern pattern

no call-forward pattern pattern

Syntax Description

pattern

String that consists of one or more digits and wildcard markers or dots (.) to define a specific pattern. Calling parties that match a defined pattern use the H.450.3 standard if they are forwarded. A pattern of .T specifies the H.450.3 forwarding standard for all incoming calls.


Defaults

No default behavior or values

Command Modes

Telephony-service configuration

Command History

Release
Modification

12.2(11)YT

This command was introduced.

12.2(15)T

This command was integrated into Cisco IOS Release 12.2(15)T.


Usage Guidelines

Use this command with Cisco IOS Telephony Service (ITS) V2.1 or a later version.

When H.450.3 call forwarding is selected, the router must be configured with a Tool Command Language (TCL) script that supports the H.450.3 protocol. The TCL script is loaded on the ITS router with the call application voice command.

The pattern match in this command is against the phone number of the calling party. When an ITS directory number has forwarded its calls and an incoming call is received for that number, the ITS router sends an H.450.3 response back to the original calling party to request that the call be placed again using the forward-to destination.

Calling numbers that do not match the patterns defined with this command are forwarded using Cisco-proprietary call forwarding for backward compatibility.

Examples

The following example specifies that all 4-digit directory numbers beginning with 4 should use the H.450.3 standard whenever they are forwarded:

Router(config)# telephony-service
Router(config-telephony-service)# call-forward pattern 4...

The following example forwards all calls using the H.450.3 standard:

Router(config)# telephony-service
Router(config-telephony-service)# call-forward pattern .T

Related Commands

Command
Description

call application voice

Defines an application, indicates the location of the corresponding TCL files that implement the application, and loads the selected TCL script.

telephony-service

Enables Cisco ITS and enters telephony-service configuration mode.


calling-info pstn-to-sip

To specify calling information treatment for PSTN-to-SIP calls, use the calling-info pstn-to-sip command in SIP user agent configuration mode. To disable calling information treatment for PSTN-to-SIP calls, use the no form of this command.

calling-info pstn-to-sip {unscreened discard | {from | remote-party-id {name set name | number set number}}}

no calling-info pstn-to-sip

Syntax Description

unscreened discard

(Optional) Specifies that the calling name and number be discarded.

from name set name

(Optional) Specifies that the display-name of the From header is unconditionally set to the configured ASCII string in the forwarded INVITE message.

from number set number

(Optional) Specifies that the user part of the From header is unconditionally set to the configured ASCII string in the forwarded INVITE message.

remote-party-id name set name

(Optional) Specifies that the display-name of the Remote-Party-ID header is unconditionally set to the configured ASCII string in the forwarded INVITE message.

remote-party-id number set number

(Optional) Specifies that the user part of the Remote-Party-ID header is unconditionally set to the configured ASCII string in the forwarded INVITE message.


Defaults

This command is disabled.

Command Modes

SIP user agent configuration

Command History

Release
Modification

12.2(13)T

This command was introduced.


Usage Guidelines

When a call exits the gateway, the calling-info pstn-to-sip treatments are applied.

Examples

The following example enables calling information treatment for PSTN-to-SIP calls and sets the company name and number:

Router(config-sip-ua)# calling-info pstn-to-sip from name set CompanyA
Router(config-sip-ua)# calling-info pstn-to-sip from number set 5551000
Router(config-sip-ua)# exit
Router(config)# exit
Router# show running-config
Building configuration...

.
.
.
!
sip-ua 
 calling-info pstn-to-sip from name set CompanyA
 calling-info pstn-to-sip from number set 5551000
 no remote-party-id
!
.
.
.

Related Commands

Command
Description

calling-info sip-to-pstn

Specifies calling information treatment for SIP-to-PSTN calls.

debug ccsip events

Enables tracing of SIP SPI events.

debug ccsip messages

Enables tracing SIP messages exchanged between the SIP UA client and the access server.

debug isdn q931

Displays call setup and teardown of ISDN connections.

debug voice ccapi error

Enables tracing error logs in the call control API.

debug voip ccapi in out

Enables tracing the execution path through the call control API.


calling-info sip-to-pstn

To specify calling information treatment for SIP-to-PSTN calls, use the calling-info sip-to-pstn command in SIP user agent configuration mode. To disable calling information treatment for SIP-to-PSTN calls, use the no form of this command.

calling-info sip-to-pstn {unscreened discard | name set name | number set number}

no calling-info sip-to-pstn

Syntax Description

unscreened discard

(Optional) Specifies that the calling name and number be discarded.

name set name

(Optional) Specifies that the calling name be unconditionally set to the configured ASCII string in the forwarded Setup mesage.

number set number

(Optional) Specifies that he calling number be unconditionally set to the configured ASCII string in the forwarded Setup message.


Defaults

This command is disabled.

Command Modes

SIP user agent configuration

Command History

Release
Modification

12.2(13)T

This command was introduced.


Usage Guidelines

When a call enters the gateway, the calling-info sip-to-pstn treatments are applied.

Examples

The following example enables calling information treatment for SIP-to-PSTN calls and sets the company name to CompanyA and the number to 5551000:

Router(config-sip-ua)# calling-info sip-to-pstn name set CompanyA
Router(config-sip-ua)# calling-info sip-to-pstn number set 5551000
Router(config-sip-ua)# exit
Router(config)# exit
Router# show running-config
Building configuration...

.
.
.
!
sip-ua 
 calling-info sip-to-pstn name set CompanyA
	 calling-info sip-to-pstn number set 5551000
!
.
.
.

Related Commands

Command
Description

debug ccsip events

Enables tracing of SIP SPI events.

debug ccsip messages

Enables SIP SPI message tracing.

debug isdn q931

Displays call setup and teardown of ISDN connections.

debug voip ccapi in out

Enables tracing the execution path through the call control API.

calling-info pstn-to-sip

Specifies calling information treatment for PSTN-to-SIP calls.


calling-number outbound

To specify automatic number identification (ANI) to be sent out when T1-channel associated signaling (T1-CAS) Feature Group D-Exchange Access North American (FGD-EANA) is configured as the signaling type, use the calling-number outbound command in dial peer or voice-port configuration mode. To disable the calling-number outbound command, use no form of this command.

calling-number outbound {range string1 string2 | sequence string1... string5| null}

no calling-number outbound {range string1 string2 | sequence string1... string5| null}

Syntax Description

range

Generates the sequence of ANI by rotating through the specified range (string1 to string2).

sequence

Configures a sequence of discrete strings (string1... string5) to be passed out as ANI for successive calls using the peer.

null

Suppresses ANI. If used, no ANI is passed when this dial peer is selected.

string#...

Valid E.164 telephone number strings. Strings must be of equal length and cannot be more than 32-digits long.


Defaults

No outbound calling number is specified

Command Modes

Dial peer configuration

Voice-port configuration

Command History

Release
Modification

12.1(3)T

This command was introduced on the Cisco AS5300.


Usage Guidelines

This command is effective only for Feature Group D-Exchange Access North American (FGD-EANA) signaling.

Examples

Use the calling-number outbound command to enable or disable the passing of ANI on T1-CAS FGD-EANA configured T1 interface for outgoing calls. Syntax for this command is the same for both voice-port mode and dial peer mode. Examples are given for both modes.

calling-number outbound Range

calling-number outbound range string1 string2 

The values string1 and string2 are valid E.164 telephone number strings. Both strings must be of the same length and cannot be more than 32 digits long. Only the last four digits are used for specifying the range (string1 to string2) and for generating the sequence of ANI by rotating through the range until string2 is reached and then starting from string1 again. If strings are less than four digits in length, then entire strings is used.

ANI is generated by using the 408555 prefix and by rotating through 6000 to 6001 for each call using this peer.

Dial peer configuration mode:

dial-peer voice 1 pots
 calling-number outbound range 4085556000 4085556001
 calling Number Outbound is effective only for fgd_eana signaling

Voice-port configuration mode:

voice-port 1:D
 calling-number outbound range 4085556000 4085556005
 Calling Number Outbound is effective only for fgd_eana signaling

calling-number outbound Sequence

calling-number outbound sequence string1 string2 string3 
string4 string5

This option configures a sequence of discrete strings (string1...string5) to be passed out as ANI for successive calls using the peer. The limit is five strings. All strings must be valid E.164 numbers, up to 32 digits in length.

Dial peer configuration mode:

dial-peer voice 1 pots
 calling-number outbound sequence 6000 6006 4000 5000 5025
 Calling Number Outbound is effective only for fgd_eana signaling

Voice-port configuration mode:

voice-port 1:D
 calling-number outbound sequence 6000 6006 4000 5000 5025
 Calling Number Outbound is effective only for fgd_eana signaling

calling-number outbound Null

calling-number outbound null

This option suppresses ANI. If used, no ANI is passed when this dial peer is selected.

Dial peer configuration mode:

dial-peer voice 1 pots
 calling-number outbound null
 Calling Number Outbound is effective only for fgd_eana signaling

Voice-port configuration mode:

voice-port 1:D
 calling-number outbound null
 Calling Number Outbound is effective only for fgd_eana signaling

Related Commands

Command
Description

info-digits string1

Configures two information digits to be prepended to the ANI string.


call-manager-fallback

To enable Survivable Remote Site (SRS) Telephony support and enter call-manager-fallback mode, use the call-manager-fallback command in global configuration mode. To disable SRS Telephony support, use the no form of this command.

call-manager-fallback

no call-manager-fallback

Syntax Description

This command has no arguments or keywords.

Defaults

No default behavior or values

Command Modes

Global configuration

Command History

Release
Modification

12.1(5)YD

This command was introduced on the following platforms: Cisco 2600 series and Cisco 3600 series, and Cisco IAD2420 series.

12.2(2)XT

This command was implemented on Cisco 1750 and Cisco 1751.

12.2(8)T

This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725 and Cisco 3745.

12.2(8)T1

This command was implemented on the Cisco 2600-XM and Cisco 2691.

12.2(11)T

This command was implemented on the Cisco 1760.


Usage Guidelines

The call-manager-fallback command is a top-level command in the hierarchy of commands in call-manager-fallback configuration mode.

Examples

The following example shows how to enter the call-manager-fallback configuration mode:

Router(config)# call-manager-fallback 
Router(config-cm-fallback)#

Related Commands

Command
Description

access-code

Configures trunk access codes for each type of line so that the Cisco IP phones can access trunk lines.

alias

Provides a mechanism for servicing calls to telephone numbers that are unavailable during CallManager fallback.

call-forward busy

Configures call forwarding to another number when a Cisco IP phone is busy.

call-forward noan

Configures call forwarding to another number when no answer is received from a Cisco IP phone.

cor

Configures a COR on the dial peers associated with directory numbers.

default-destination

Assigns a default destination number for incoming telephone calls.

dialplan-pattern

Creates a global prefix that can be used to expand the abbreviated extension numbers into fully qualified E.164 numbers.

huntstop

Sets huntstop for the dial peers associated with a Cisco IP phone lines.

ip source-address

Enables the router to receive messages from Cisco IP phones through the specified IP addresses and ports.

keepalive

Configures the time interval between sending keepalive messages to the router used by Cisco IP phones.

max-dn

Sets the maximum number of directory numbers or virtual voice ports that can be supported by the router.

max-ephone

Configures the maximum number of Cisco IP phones that can be supported by the router.

reset

Resets Cisco IP phones.

timeouts interdigit

Configures the interdigit timeout value for all Cisco IP phones attached to the router.

transfer-pattern

Allows transfer of telephone calls by Cisco IP phones to other phone numbers.

translate

Applies a translation rule to modify the phone number dialed by any Cisco IP phone user during the CallManager fallback mode.

voicemail

Configures the telephone number that is speed-dialed when the message button on a Cisco IP phone is pressed.


call-router

To enable the Annex G border element (BE) configuration commands, use the call-router command in global configuration mode. To remove the definition of a BE, use the no form of this command.

call-router h323-annexg border-element-id

no call-router

Syntax Description

h323-annexg

Keyword to invoke H.323 Annex G configuration mode.

border-element-id

Identifier of the BE that you are provisioning. Possible values are any International Alphabet 5 (IA5) string, without spaces and up to 20 characters in length. This value must match the value that you specified for the BE ID in the border-element command.


Defaults

No default behaviors or values

Command Modes

Global configuration

Command History

Release
Modification

12.2(2)XA

This command was introduced.

12.2(4)T

This command was integrated into Cisco IOS Release 12.2(4)T. This command does not support the Cisco AS5300, Cisco AS5350, and Cisco AS5400 series in this release.

12.2(2)XB1

This command was implemented on the Cisco AS5850.

12.2(11)T

This command was integrated into Cisco IOS Release 12.2(11)T.


Usage Guidelines

Use this command to enter Annex G configuration mode and to identify BEs.

Examples

The following example shows that Annex G configuration mode is being entered for a BE named "be20".

Router(config)# call-router h323-annexg be20

Related Commands

Command
Description

show call history

Displays the fax call history table for a fax transmission.

show call-router status

Displays the Annex G BE status.


call-waiting

To enable call waiting, use the call-waiting command in interface configuration mode. To disable call waiting, use the no form of this command.

call-waiting

no call-waiting

Syntax Description

This command has no arguments or keywords.

Defaults

Call waiting is enabled

Command Modes

Interface configuration

Command History

Release
Modification

12.0(3)T

This command was introduced on the Cisco 800 series.


Usage Guidelines

This command is applicable to Cisco 800 series routers.

You must specify this command when creating a dial peer. This command does not work if it is not specified within the context of a dial peer. For information on creating a dial peer, refer to the Cisco 800 Series Routers Software Configuration Guide.

Examples

The following example disables call waiting:

no call-waiting

Related Commands

Command
Description

destination-pattern

Specifies either the prefix, the full E.164 telephone number, or an ISDN directory number (depending on the dial plan) to be used for a dial peer.

dial peer voice

Enters dial peer configuration mode, defines the type of dial peer, and defines the tag number associated with a dial peer.

port (dial peer)

Enables an interface on a PA-4R-DTR port adapter to operate as a concentrator port.

ring

Sets up a distinctive ring for telephones, fax machines, or modems connected to a Cisco 800 series router.

show dial peer voice

Displays configuration information and call statistics for dial peers.


cap-list vfc

To add a voice codec overlay file to the capability file list, use the cap-list vfc command in global configuration mode. To disable a particular codec overlay file that has been added to the capability list, use the no form of this command.

cap-list filename vfc slot-number

no cap-list filename vfc slot-number

Syntax Description

filename

Identifies the codec file stored in voice feature card (VFC) Flash memory.

slot-number

Identifies the slot where the VFC is installed. Range is 0 to 2. There is no default value.


Defaults

No default behavior or values

Command Modes

Global configuration

Command History

Release
Modification

11.3 NA

This command was introduced on the Cisco AS5300.


Usage Guidelines

When VCWare is unbundled, it automatically adds DSPWare to Flash memory, creates both the capability and default file lists, and populates these lists with the default files for the particular version of VCWare. The capability list defines the available voice codecs for H.323 capability negotiation. Use the cap-list vfc command to add the indicated voice codec overlay file (defined by filename) to the capability file list in Flash memory.

Examples

The following example adds the following codec to the list included in Flash memory:

config terminal
 cap-list cdc-g711-1.0.14.0.bin vfc 0

Related Commands

Command
Description

default-file vfc

Specifies an additional (or different) file from the ones in the default file list and stored in VFC Flash memory.


capacity update interval (dial peer)

To change the capacity update for prefixes associated with this dial peer, use the capacity update interval command in dial peer configuration mode. To return to the default, use the no form of this command.

capacity update interval seconds

no capacity update interval seconds

Syntax Description

seconds

Interval, in seconds, between the sending of periodic capacity updates. This can be a number in the range 10 to 1000. The default value is 25 seconds.


Defaults

25 seconds

Command Modes

Dial peer configuration

Command History

Release
Modification

12.3(1)

This command was introduced.


Usage Guidelines

The update interval should be set depending on the number of updates that are sent. Updates are sent more often when more calls are coming in, which can lead to data getting out of sync. If the interval is too short for the amount of updates, the location server can be overwhelmed. If this dial peer gets too much traffic, set the seconds argument to a higher value.

Examples

The following example shows that POTS dial peer 10 is having the capacity update occur every 35 seconds:

Router(config)# dial-peer voice 10 pots
Router(config-dial-peer)# capacity update interval 35

Related Commands

Command
Description

dial-peer voice

Enters dial-peer configuration mode and specifies the method of voice-related encapsulation.


capacity update interval (trunk group)

To change the capacity update for carriers or trunk groups, use the capacity update interval command in trunk group configuration mode. To return to the default, use the no form of this command.

capacity {carrier | trunk-group} update interval seconds

no capacity {carrier | trunk-group} update interval seconds

Syntax Description

carrier

Carrier capacity.

trunk-group

Trunk group capacity.

seconds

Interval, in seconds, between the sending of periodic capacity updates. This can be a number in the range 10 to 1000. The default value is 25 seconds.


Defaults

25 seconds

Command Modes

Trunk group configuration

Command History

Release
Modification

12.3(1)

This command was introduced.


Usage Guidelines

The update interval should be set depending on the number of updates that are sent. Updates are sent more often when more calls are coming in, which can lead to data getting out of sync. If the interval is too short for the amount of updates, the location server can be overwhelmed. If this trunk group or carrier group gets too much traffic, set the seconds argument to a higher value.

Examples

The following example sets the capacity update for trunk group 101 to occur every 45 seconds:

Router(config)# trunk group 101
Router(config-trunkgroup)# capacity trunk-group update interval 45

Related Commands

Command
Description

trunk group

Defines the trunk group and enters trunk group configuration mode.


card type (t1/e1)

To configure the card type, use the card type command in global configuration mode. To restore the default value, use the no form of this command.

card type {t1 | e1} slot [bay]

no card type {t1 | e1} slot [bay]

Syntax Description

t1

Specifies T1 connectivity of 1.544 Mbps through the telephone switching network, using AMI or B8ZS coding.

e1

Specifies a wide-area digital transmission scheme used predominantly in Europe that carries data at a rate of 2.048 Mbps.

slot

Slot (port) number of the interface.

bay

(Optional) Card interface bay number in a slot (route/switch processor [RSP] platform only). This option is not available on other platforms.


Defaults

No default behavior or values

Command Modes

Global configuration

Command History

Release
Modification

12.0(5)XE

This command was introduced.

12.0(7)T

This command was integrated into Cisco IOS Release 12.0(7)T.

12.3(1)

This command was integrated into Cisco IOS Release 12.3(1) and support was added for Cisco 2610XM, Cisco 2611XM, Cisco 2620XM, Cisco 2621XM, Cisco 2650XM, Cisco 2651XM, Cisco 2691, Cisco 3631, Cisco 3660, Cisco 3725, and Cisco 3745 platforms.


Usage Guidelines

Changes made using this command do not take effect unless the reload command is used or the router is rebooted.

Examples

The following example configures T1 data transmission on slot 1 (port 1) of the router:

card type t1 1

Related Commands

Command
Description

controller

Configures a T1 or E1 controller and enters controller configuration mode.

reload

Reloads the operating system.


card type (t3/e3)

To configure the card type on the T3 or E3 controller, use the card type command in global configuration mode. To restore the default value, use the no form of this command.

card type {t3 | e3} slot

no card type {t3 | e3} slot

Syntax Description

t3

Specifies T3 connectivity of 44210 kbps through the network, using B8ZS coding.

e3

Specifies a wide-area digital transmission scheme used predominantly in Europe that carries data at a rate of 34010 kbps.

slot

Slot number of the interface.


Defaults

No default behavior or values.

Command Modes

Global configuration

Command History

Release
Modification

12.1(1)T

This command was introduced.

12.2(11)YT

This command was integrated into Cisco IOS Release 12.2(11)YT and implemented on the following platforms: Cisco 2650XM, Cisco 2651XM, Cisco 2691, Cisco 3660 series, Cisco 3725, and Cisco 3745 routers.

12.2(15)T

This command was integrated into Cisco IOS Release 12.2(15)T.


Usage Guidelines

Once a card type is issued, the user can enter the no card type command and then another card type command to configure a new card type. The user must save the configuration to the NVRAM and reboot the router in order for the new configuration to take effect.

When the router comes up, the software comes up with the new card type. Note that the software will reject the configuration associated with the old controller and old interface. The user will now have to configure the new controller and serial interface and save it.

Examples

The following example shows T3 data transmission configured in slot 1:

card type t3 1

Related Commands

Command
Description

controller

Configures a T3 or E3 controller and enters controller configuration mode.

reload

Reloads the operating system.


carrier-id (dial-peer)

To specify the carrier associated with a VoIP call in a dial peer, use the carrier-id command in dial-peer configuration mode. To delete the source carrier ID, use the no form of this command.

carrier-id {source | target} name

no carrier-id {source | target} name

Syntax Description

source

Indicates the carrier that the dial peer uses as a matching key for inbound dial-peer matching.

target

Indicates the carrier that the dial peer uses as a matching key for outbound dial-peer matching.

name

Specifies the ID of the carrier to use for the call. Valid carrier IDs contain a maximum of 127 alphanumeric characters.


Defaults

No default behavior or values

Command Modes

Dial-peer configuration

Command History

Release
Modification

12.2(11)T

This command was introduced.


Usage Guidelines

A Gatekeeper Transaction Message Protocol (GKTMP) route server-based application at the terminating gateway uses the source carrier ID to select a target carrier that routes the call over a plain old telephone service (POTS) line.

The terminating gateway uses the target carrier ID to select a dial peer for routing the call over a POTS line.

Examples

The following example indicates that dial peer 112 should use carrier ID "east17" for outbound dial-peer matching in the terminating gateway:

Router(config)# dial-peer voice 112 pots
Router(config-dial-peer)# carrier-id target east17

The following example indicates that dial peer 111 should use carrier ID "beta23" for inbound dial-peer matching in the terminating gateway:

Router(config)# dial-peer voice 111 voip
Router(config-dial-peer)# carrier-id source beta23

Related Commands

Command
Description

translation-profile (dial-peer)

Associates a translation profile with a dial peer.

trunkgroup (dial-peer)

Assigns a trunk group to a source IP group or dial peer for trunk group label routing.


carrier-id (global)

To set the carrier ID for trunk groups when a local carrier ID is not configured, use the carrier-id command in global configuration mode. To disable the carrier ID, use the no form of this command.

carrier-id name [cic]

no carrier-id name[cic]

Syntax Description

name

Identifier for the carrier ID. Must be 4-digit numeric carrier identification code to be advertised as a TRIP carrier family but can be alphanumeric if used otherwise.

cic

Specifies that the carrier ID is a circuit identification code(CIC).


Defaults

No default behavior or values

Command Modes

Global configuration

Command History

Release
Modification

12.3(1)

This command was introduced.


Usage Guidelines

To advertise the carrier as a TRIP carrier family, the cic keyword must be used. When cic is used, only numeric values can be accepted for the name value. If cic is not used, the name value can be alphanumeric but is not advertised to TRIP location servers.

Examples

The following example shows a carrier ID using the circuit identification code:

Router(config)# carrier-id 1234 cic

Related Commands

Command
Description

carrier-id (trunk group)

Configures the carrier ID locally on the trunk group.


carrier-id (trunk group)

To specify the carrier associated with a trunk group, use the carrier-id command in trunk group configuration mode. To delete the source carrier ID, use the no form of this command.

carrier-id name [cic]

no carrier-id name [cic]

Syntax Description

name

Specifies the ID of the carrier to use for the call. Valid carrier IDs contain a maximum of 127 alphanumeric characters.

To be advertised as a TRIP carrier family, this must be set to a 4-digit numeric carrier identification code.

cic

Specifies that the carrier ID is a circuit identification code.


Defaults

No default behavior or values

Command Modes

Trunk group configuration

Command History

Release
Modification

12.2(11)T

This command was introduced.

12.3(1)

The cic keyword was added.


Usage Guidelines

In a network, calls are routed over incoming trunk groups and outgoing trunk groups. The name arguments identifies the carrier that handles the calls for a specific trunk group. In some cases, the same trunk group may be used to carry both incoming calls and outgoing calls.

The carrier ID configured locally on the trunk group supersedes the globally configured carrier ID.

To advertise the carrier as a TRIP carrier family, the cic keyword must be used. When cic is used, only numeric values can be accepted for the name value. If cic is not used, the name value can be alphanumeric but is not advertised to TRIP location servers.

Examples

The following example indicates that carrier "alpha1" carries calls for trunk group 5:

Router(config)# trunk group 5
Router(config-trunk-group)# carrier-id alpha1

The following example shows that the carrier with circuit identification code 1234 carries calls for trunk group 101. This trunk group can carry TRIP advertisements:

Router(config)# trunk group 101
Router(config-trunk-group)# carrier-id 1234 cic

Related Commands

Command
Description

carrier-id (global)

Configures the carrier ID globally for all trunk groups.

translation-profile (trunk group)

Associates a translation profile with a trunk group.

trunk group

Initiates the definition of a trunk group.


carrier-id (voice source group)

To specify the carrier associated with a VoIP call, use the carrier-id command in voice source group configuration mode. To delete the source carrier ID, use the no form of this command.

carrier-id {source | target} name

no carrier-id {source | target} name

Syntax Description

source

Indicates the carrier ID associated with an incoming VoIP call at the terminating gateway.

target

Indicates the carrier ID used by the terminating gateway to match an outbound dial peer.

name

Specifies the ID of the carrier to use for the call. Valid carrier IDs contain a maximum of 127 alphanumeric characters.


Defaults

No default behavior or values

Command Modes

Voice source group configuration

Command History

Release
Modification

12.2(11)T

This command was introduced.


Usage Guidelines

A Gatekeeper Transaction Message Protocol (GKTMP) server application at the terminating gateway uses the source carrier ID to select a target carrier that routes the call over a plain old telephone service (POTS) line. The terminating gateway uses the target carrier ID to select a dial peer for routing the call over a POTS line.


Note If an incoming H.323 VoIP call matches a source IP group that has a target carrier ID, the source IP group's target carrier ID overrides the VoIP call's H.323 setup message.


Examples

The following example indicates that voice source IP group "florida" should use carrier ID named "north3" for incoming VoIP calls and carrier ID named "east17" for outbound dial-peer matching in the terminating gateway:

Router(config)# voice source-group florida
Router(cfg-source-grp)# carrier-id source north3
Router(cfg-source-grp)# carrier-id target east17

Related Commands

Command
Description

voice source-group

Initiates the definition of a source IP group.


cause-code

To represent internal failures with former and nonstandard H.323 or SIP cause codes, use the cause-code command in voice service VoIP configuration mode. To use standard cause-code categories, use the no form of this command.

cause-code legacy

no cause-code legacy

Syntax Description

legacy

Sets the internal cause code to the former and nonstandard set of values. Used for backward compatibility purposes.


Defaults

The default for SIP and H.323 is to use standard cause-code categories.

Command Modes

Voice service VoIP configuration

Command History

Release
Modification

12.2(11)T

This command was introduced.


Examples

The following example sets the internal cause codes to the former and nonstandard set of SIP and H.323 values for backward compatibility.

Router(config)# voice service voip
Router(config-voi-srv)# cause-code legacy

Related Commands

Command
Description

show call history voice

Displays the call history table for voice calls.


ccm-manager application redundant-link port

To configure the port number for the redundant link application, use the ccm-manager application redundant-link port command in global configuration mode. To disable the configuration, use the no form of this command.

ccm-manager application redundant-link port number

no ccm-manager application

Syntax Description

number

Port number for the transport protocol. The protocol may be the User Data Protocol (UDP), Reliable User Datagram Protocol (RDUP), or Transmission Control Protocol (TCP). Range is from 0 to 65535, and it must not be a well-known reserved port number. The default is 2428.


Defaults

Port 2428

Command Modes

Global configuration

Command History

Release
Modification

12.1(3)T

This command was introduced with Cisco CallManager Version 3.0 and the Cisco Voice Gateway 200 (VG200).

12.2(2)XA

The command was implemented on Cisco 2600 series and Cisco 3600 series.

12.2(4)T

The command was integrated into Cisco IOS Release 12.2(4)T.


Usage Guidelines

This command is optional. Use this command only when defining an application-specific port other than the default.

Examples

In the following example, the port number of the redundant link application is 2429:

ccm-manager application redundant-link port 2429

Related Commands

Command
Description

ccm-manager redundant-host

Configures the IP address or the DNS name of up to two backup Cisco CallManagers.

ccm-manager switchback

Configures the switchback mode that determines when the primary Cisco CallManager is used if it becomes available again while a backup Cisco CallManager is being used.


ccm-manager config

To enable the local Media Gateway Control Protocol (MGCP) voice gateway with the TFTP server IP address or logical name from which to download XML configuration files and to enable the download of the configuration, use the ccm-manager config command in global configuration mode. To disable the dial-peer and server configurations, use the no form of this command.

ccm-manager config {dialpeer-prefix | server {ip-address | name}}

no ccm-manager config {dialpeer-prefix | server {ip-address | name}}

Syntax Description

dialpeer-prefix

Dial peer created for a voice dial-peer tag. Range is from 1 to 2147483647. The default is 999.

server

IP address or logical name of the TFTP server from which the XML configuration files are downloaded.

The arguments are as follows:

ip-address—IP address of the TFTP server from which to download the XML configuration files to the local MGCP voice gateway.

name—Logical (symbolic) name of the TFTP server from which to download XML configuration files to the local MGCP voice gateway.


Defaults

The configuration download feature is disabled.
dialpeer-prefix: 999

Command Modes

Global configuration

Command History

Release
Modification

12.2(2)XN

This command was introduced on the Cisco 2600 series, Cisco 3600 series and the Cisco VG200.

12.2(11)T

This command was integrated into Cisco IOS Release 12.2(11)T and Cisco CallManager Version 3.2 and implemented on Cisco IAD2420 series.


Usage Guidelines

The ccm-manager config command is optional. If you separate the MGCP and H.323 dial peers under different dial-peer tags, ensure that the MGCP dial peers are configured before the H.323 dial peers. Direct-inward-dial (DID) is required for E1 PRI dial peers.

Examples

The following example shows the configuration on the command:

ccm-manager config

In the following example, the IP address of the TFTP server from which a configuration file is downloaded is identified:

ccm-manager config server 10.0.0.21
! Enter configuration commands, one per line.
ctrl z

Related Commands

Command
Description

debug ccm-manager config

Displays dialog during configuration download from the Cisco CallManager to the gateway.

show ccm-config

Displays whether or not the ccm-manager config is enabled.


ccm-manager fallback-mgcp

To enable the gateway fallback feature on a Media Gateway Control Protocol (MGCP) voice gateway, use the ccm-manager fallback-mgcp command in global configuration mode. To disable fallback on the MGCP voice gateway, use the no form of this command.

ccm-manager fallback-mgcp

no ccm-manager fallback-mgcp

Syntax Description

This command has no arguments or keywords.

Defaults

Enabled

Command Modes

Global configuration

Command History

Release
Modification

12.2(2)XN

This command was introduced on the Cisco 2600 series, Cisco 3600 series, and the Cisco VG200.

12.2(11)T

This command was integrated into Cisco IOS Release 12.2(11)T and Cisco CallManager Version 3.2 and implemented on Cisco IAD2420 series.


Usage Guidelines

The ccm-manager fallback-mgcp command must be enabled to cause the gateway to fall back. The mode and timing are set by default.

Examples

The following example enables the gateway fallback feature on an MGCP voice gateway.

ccm-manager fallback-mgcp

Related Commands

Related Command
Purpose

show ccm-manager fallback-mgcp

Displays the status of the MGCP gateway fallback feature.


ccm-manager mgcp

To enable the gateway to communicate with the Cisco CallManager through the Media Gateway Control Protocol (MGCP) and to supply redundant control agent services, use the ccm-manager mgcp command in global configuration mode. To disable communication with the Cisco CallManager and redundant control agent services, use the no form of this command.

ccm-manager mgcp

no ccm-manager mgcp

Syntax Description

This command has no arguments or keywords.

Defaults

Cisco CallManager does not communicate with the gateway through MGCP

Command Modes

Global configuration

Command History

Release
Modification

12.1(3)T

This command was introduced with Cisco CallManager Version 3.0 and the Cisco VG200.

12.2(2)XA

The command was implemented on Cisco 2600 series and Cisco 3600 series.

12.2(2)XN

Support for enhanced MGCP voice gateway interoperability was added to Cisco CallManager Version 3.1 for the Cisco 2600 series, 3600 series, and Cisco VG200.

12.2(4)T

The command was integrated into Cisco IOS Release 12.2(4)T.

12.2(11)T

This command was integrated into the Cisco IOS Release 12.2(11)T and Cisco CallManager Version 3.2 and was implemented on the Cisco IAD2420 series routers.

12.2(11)YU

This command was implemented on the Cisco 1760 gateway.


Usage Guidelines

This command enables the gateway to communicate with Cisco CallManager through MGCP. This command also enables control agent redundancy when a backup Cisco CallManager server is available.

Examples

In the following example, support for Cisco CallManager and redundancy is enabled within MGCP:

ccm-manager mgcp 

Related Commands

Command
Description

ccm-manager redundant-host

Configures the IP address or the DNS name of up to two backup Cisco CallManagers.

ccm-manager switchback

Configures the switchback mode that determines when the primary Cisco CallManager is used if it becomes available again while a backup Cisco CallManager is being used.

mgcp

Enables Media Gateway Control Protocol mode.


ccm-manager music-on-hold

To enable the multicast music-on-hold (MOH) feature on a Media Gateway Control Protocol (MGCP) voice gateway, use the ccm-manager music-on-hold command in global configuration mode. To disable the MOH feature on the voice gateway, use the no form of this command.

ccm-manager music-on-hold

no ccm-manager music-on-hold

Syntax Description

This command has no arguments or keywords.

Defaults

Disabled

Command Modes

Global configuration

Command History

Release
Modification

12.2(2)XN

This command was introduced on the Cisco 2600 series, Cisco 3600 series, and the Cisco VG200.

12.2(11)T

This command was integrated into Cisco IOS Release 12.2(11)T and Cisco CallManager Version 3.2 and implemented on the Cisco IAD2420 series routers.


Examples

The following example shows multicast MOH configured for an MGCP voice gateway:

mgcp call-agent 10.0.0.21 2427 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
mgcp rtp unreachable timeout 1000
mgcp modem passthrough voip mode cisco
mgcp package-capability rtp-package
mgcp package-capability sst-package
no mgcp timer receive-rtcp
call rsvp-sync
!
ccm-manager redundant-host 10.0.0.21 
ccm-manager mgcp
ccm-manager music-on-hold
ccm-manager config server 10.0.0.21 
!

Related Commands

Command
Description

ccm-manager music-on-hold bind

Enables the multicast MOH feature on MGCP voice gateways.

debug ccm-manager music-on-hold

Displays debugging information for MOH.

show ccm-manager music-on-hold

Displays the MOH information.


ccm-manager music-on-hold bind

To bind the multicast music-on-hold (MOH) feature to a designated interface, use the ccm-manager music-on-hold bind command in global configuration mode. To unbind the MOH feature on the voice gateway, use the no form of this command.

ccm-manager music-on-hold bind type slot/port

no ccm-manager music-on-hold bind type slot/port

Syntax Description

type

Interface type to which the MOH feature is bound. The options follow:

asyncAsynchronous interface

bviBridge-Group Virtual interface

ctunnelCTunnel interface

dialerDialer interface

ethernetIEEE 802.3

lexLex interface

loopbackLoopback interface

mfrMultilink Frame Relay bundle interface

multilinkMultilink interface

nullNull interface

serialSerial interface

tunnelTunnel interface

vifPGM Multicast Host interface

virtual-FrameRelay—Virtual frame relay interface

virtual-TemplateVirtual template interface

virtual-TokenRingVirtual Token Ring

slot/port

Number of the slot being configured. Refer to the appropriate hardware manual for slot and port information.


Defaults

This feature is not enabled.

Command Modes

Global configuration

Command History

Release
Modification

12.2(11)T

This command was introduced.


Examples

The following example shows multicast MOH bound to serial interface 0/0:

ccm-manager music-on-hold bind serial 0/0

Related Commands

Command
Description

ccm-manager music-on-hold

Enables the MOH feature.

debug ccm-manager music-on-hold

Displays debugging information for MOH.

show ccm-manager music-on-hold

Displays the MOH information.


ccm-manager redundant-host

To configure the IP address or the Domain Name System (DNS) name of one or two backup Cisco CallManager servers, use the ccm-manager redundant-host command in global configuration mode. To disable the use of backup Cisco CallManager servers as call agents, use the no form of this command.

ccm-manager redundant-host {ip-address | dns-name} [ip-address | dns-name]

no ccm-manager redundant-host {ip-address | dns-name} [ip-address | dns-name]

Syntax Description

ip-address

IP address of the backup Cisco CallManager server.

dns-name

DNS name of the backup Cisco CallManager server.


Defaults

If you do not configure a backup Cisco CallManager, the redundancy is disabled.

Command Modes

Global configuration

Command History

Release
Modification

12.1(3)T

This command was introduced with Cisco CallManager Version 3.0 and the Cisco Voice Gateway 200 (VG200).

12.2(2)XA

The command was implemented on Cisco 2600 series and Cisco 3600 series. The DNS-name argument was added.

12.2(4)T

The command was integrated into Cisco IOS Release 12.2(4)T.

12.2(2)XN

Support for enhanced MGCP voice gateway interoperability was added to Cisco CallManager Version 3.1 for the Cisco 2600 series, 3600 series, and the Cisco VG200.

12.2(11)T

This command was integrated into the Cisco IOS Release 12.2(11)T and Cisco CallManager Version 3.2 and implemented on the Cisco IAD2420 series routers.


Usage Guidelines

You can configure one or two backup Cisco CallManager servers. The list of IP addresses or DNS names is an ordered and prioritized list. The Cisco CallManager server that was defined with the mgcp call-agent command has the highest priority (that is, it is the primary Cisco CallManager server). The gateway selects a Cisco CallManager server on the basis of the order of its appearance in this list.

Examples

In the following example, the IP address of the backup Cisco CallManager is 10.0.0.50:

ccm-manager redundant-host 10.0.0.50

Related Commands

Command
Description

ccm-manager application

Configures the port number for the redundant link application.

ccm-manager switchback

Configures the switchback mode that determines when the primary Cisco CallManager is used if it becomes available again while a backup Cisco CallManager is being used.

ccm-manager switchover-to-backup

Redirects (manually and immediately) a Cisco 2600 series router or Cisco 3600 series router to the backup Cisco CallManager server.

mgcp call-agent

Defines the Cisco CallManager server as the highest priority.


ccm-manager switchback

To specify the time when control is to be returned to the primary Cisco CallManager server once it becomes available, use the ccm-manager switchback command in global configuration mode. To disable the setting for when the primary server takes control, use the no form of this command.

ccm-manager switchback {graceful | immediate | schedule-time hh:mm | uptime-delay minutes}

no ccm-manager switchback

Syntax Description

graceful

Specifies that control is returned to the primary Cisco CallManager server after the last active call ends (when there is no voice call in active setup mode on the gateway).

immediate

Specifies an immediate switchback to the primary Cisco CallManager server when the TCP link to the primary Cisco CallManager server is established, regardless of current call conditions.

schedule-time hh:mm

Specifies an hour and minute, based on a 24-hour clock, when control is returned to the primary Cisco CallManager server. If the specified time is earlier than the current time, the switchback occurs at the specified time on the following day.

uptime-delay minutes

Specifies the number, of minutes the primary Cisco CallManager server must run after the TCP link to is reestablished and control is returned to that primary call agent. Valid values are from 1 to 1440 (1 minute to 24 hours).


Defaults

Graceful

Command Modes

Global configuration

Command History

Release
Modification

12.1(3)T

This command was introduced with Cisco CallManager Version 3.0 and the Cisco VG200.

12.2(2)XA

The command was implemented on Cisco 2600 series and Cisco 3600 series.

12.2(2)XN

Support for enhanced MGCP voice gateway interoperability was added to Cisco CallManager Version 3.1 for the Cisco 2600 series, 3600 series, and the Cisco VG200.

12.2(4)T

The command was integrated into Cisco IOS Release 12.2(4)T.

12.2(11)T

This command was integrated into the Cisco IOS Release 12.2(11)T and Cisco CallManager Version 3.2 and implemented on the Cisco IAD2420 series routers.


Usage Guidelines

This command allows you to configure switchback to the higher priority Cisco CallManager when it becomes available. Switchback allows call control to revert back to the original (primary) Cisco CallManager once service has been restored.

Examples

In the following example, the primary Cisco CallManager is used as soon as it becomes available:

ccm-manager switchback immediate

Related Commands

Command
Description

ccm-manager application

Configures the port number for the redundant link application.

ccm-manager redundant-host

Configures the IP address or the DNS name of up to two backup Cisco CallManagers.

ccm-manager switchover-to-backup

Redirects a Cisco 2600 series or Cisco 3600 series router to the backup Cisco CallManager.


ccm-manager switchover-to-backup

To manually redirect a gateway to the backup Cisco CallManager server, use the ccm-manager switchover-to-backup command in privileged EXEC mode.

ccm-manager switchover-to-backup

Syntax Description

This command has no arguments or keywords.

Defaults

No default behavior or value

Command Modes

Privileged EXEC

Command History

Release
Modification

12.2(2)XN

This command was introduced on the Cisco 2600 series, Cisco 3600 series, and the Cisco VG200.

12.2(11)T

This command was integrated into Cisco IOS Release 12.2(11)T and Cisco CallManager Version 3.2 and implemented on Cisco IAD2420 series.


Usage Guidelines

Switchover to the backup Cisco CallManager server occurs immediately. This command does not switch the gateway to the backup Cisco CallManager server if you have the ccm-manager switchback command option set to "immediate" and the primary Cisco CallManager server is still running.

Examples

In the following example, the backup Cisco CallManager server is used as soon as it becomes available:

ccm-manager switchover-to-backup

Related Commands

Command
Description

ccm-manager application redundant-link

Configures the port number for the redundant link application (that is, for the secondary Cisco CallManager server).

ccm-manager redundant-host

Configures the IP address or the DNS name of up to two backup Cisco CallManager servers.

ccm-manager switchback

Specifies the time at which control is returned to the primary Cisco CallManager server once the server is available.


ccs connect (controller)

To configure a common channel signaling (CCS) connection on an interface configured to support CCS frame forwarding, use the ccs connect command in controller configuration mode. To disable the CCS connection on the interface, use the no form of this command.

ccs connect {serial | atm} number [dlci | pvc vpi/vci | pvc name] [cidnumber]

no ccs connect {serial | atm} number [dlci | pvc vpi/vci | pvc name] [cidnumber]

Syntax Description

serial

Makes a serial CCS connection for Frame Relay.

atm

Makes an ATM CCS connection.

dlci

(Optional) Specifies the data link connection identifier (DLCI) number.

pvc vpi/vci

(Optional) Specifies the permanent virtual circuit (PVC) virtual path identifier or virtual channel identifier. Range is from 0 to 255; the slash is required.

pvc name

(Optional) Specifies the PVC string that names the PVC for recognition.

cidnumber

(Optional) If you have executed the ccs encap frf11 command, the cidnumber option allows you to specify any channel identification (CID) number from 5 to 255.


Defaults

No CCS connection is made

Command Modes

Controller configuration

Command History

Release
Modification

12.0(2)T

This command was introduced on the Cisco MC3810.

12.0(7)XK

The CID syntax was added; the dlci keyword and vcd options were removed.

12.1(2)T

The CID syntax addition and removal of the dlci keyword and vcd options were integrated into Cisco IOS Release 12.1(2)T.

12.1(2)XH

This command was implemented on the Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, and Cisco 7500 series.

12.1(3)T

This command was integrated into Cisco IOS Release 12.1(3)T.


Usage Guidelines

Use this command to configure a CCS connection. If the CCS connection is over Frame Relay, specify a serial interface and the DLCI. If the CCS connection is over ATM, specify atm, the slot number (0 only on the Cisco MC3810), and the PVC.

If you have executed the ccs encap frf11 command, the cidnumber option allows you to specify any CID from 5 to 255. If you do not issue the ccs encap frf11 command, Cisco encapsulation is used, and any CID value other than 254 is ignored.


Note CDP and keepalives are disabled by default on a D-channel interface.


Examples

To configure a Frame Relay CCS frame-forwarding connection on DLCI 100 by using the default CID of 254, enter the following command:

ccs connect serial 1 100

or:

ccs connect serial 1 100 10

To configure a CCS frame-forwarding connection over an ATM PVC, enter the following command:

ccs connect atm0 pvc 100/10

or:

ccs connect atm0 pvc 10/100 21

or:

ccs connect atm0 pvc mypvc_10 21

To configure a Frame Relay CCS frame-forwarding connection on DLCI 100 using a CID of 110, enter the following command:

ccs connect serial 1 100 110

Related Commands

Command
Description

ccs encap frf11

Allows the specification of the standard Annex-C FRF.11 format.


ccs connect (interface)

To configure a common channel signaling (CCS) connection on an interface configured to support CCS frame forwarding, use the ccs connect command in interface configuration mode. To disable the CCS connection on the interface, use the no form of this command.

ccs connect {serial | atm} number [dlci | pvc vpi/vci | pvc name] [cid-number]

no ccs connect {serial | atm} number [dlci | pvc vpi/vci | pvc name] [cid-number]

Syntax Description

serial

Serial CCS connection for Frame Relay.

atm

ATM CCS connection for ATM.

dlci

(Optional) Data-link connection identifier (DLCI) number.

pvc vpi/vci

(Optional) Permanent virtual circuit (PVC) virtual path identifier or virtual channel identifier. Range is from 0 to 255; the slash is required.

pvc name

(Optional) PVC string that names the PVC for recognition.

cid-number

(Optional) If you have executed the ccs encap frf11 command, the cid-number option allows you to specify any channel identification (CID) number from 5 to 255.


Defaults

No CCS connection is made

Command Modes

Interface configuration

Command History

Release
Modification

12.0(2)T

This command was introduced on the Cisco MC3810.

12.0(7)XK

The CID syntax was added; the dlci keyword and vcd options were removed.

12.1(2)T

This command was integrated into Cisco IOS Release 12.1(2)T.

12.2(2)T

This command was implemented on the Cisco 7200 series router and integrated into Cisco IOS Release 12.2(2)T.


Usage Guidelines

Use this command to configure a CCS connection. If the CCS connection is over Frame Relay, specify a serial interface and the DLCI. If the CCS connection is over ATM, specify atm, the interface number (0), and the PVC. If you have executed the ccs encap frf11 command, the cid-number option allows you to specify any CID from 5 to 255. If you do not issue the ccs encap frf11 command, Cisco encapsulation is used, and any CID value other than 254 is ignored.


Note Cisco Discovery Protocol (CDP) and keepalives are disabled by default on a D-channel interface.


This configuration is applicable only to the MC3810 multiservice access concentrator.

Examples

To configure a Frame Relay CCS frame-forwarding connection on DLCI 100 by using the default CID of 254, enter the following command:

ccs connect serial 1 100

or

ccs connect serial 1 100 10

To configure a CCS frame-forwarding connection over an ATM PVC, enter the following command:

ccs connect atm0 pvc 100/10

or

ccs connect atm0 pvc 10/100 21

or

ccs connect atm0 pvc mypvc_10 21

To configure a Frame Relay CCS frame-forwarding connection on DLCI 100 using a CID of 110, enter the following command:

ccs connect serial 1 100 110

Related Commands

Command
Description

ccs encap frf11

Allows the specification of the standard Annex-C FRF.11 format.


ccs encap frf11

To configure the common channel signaling (CCS) packet encapsulation format for FRF.11, use the ccs encap frf11 command in interface configuration mode. To disable ccs encapsulation for FRF11, use the no form of this command.

ccs encap frf11

no ccs encap frf11

Syntax Description

This command has no keywords or arguments.

Defaults

By default, the format is a Cisco packet format, using a channel ID (CID) of 254

Command Modes

Interface configuration

Command History

Release
Modification

12.0(7)XK

This command was introduced for the Cisco MC3810.

12.1(2)T

This command was integrated into Cisco IOS Release 12.1(2)T.

12.1(2)XH

This command was implemented on the Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, and Cisco 7500 series.

12.1(3)T

This command was integrated into Cisco IOS Release 12.1(3)T.


Usage Guidelines

This command allows the specification of the standard Annex-C format. Use this command to define the packet format for the CCS packet; it places the FRF.11 Annex-C (Data Transfer Syntax) standard header on the CCS packets only.

Once the ccs encap frf11 command is executed, you can use the ccs connect command to specify a CID other than 254.

Examples

The following example shows how to configure a serial interface for Frame Relay:

interface Serial1:15
 ccs encap frf11
 ccs connect Serial0 990 100

Related Commands

Command
Description

mode ccs frame-forwarding

Set to forward frames on the controller.


ces cell-loss-integration-period

To set the circuit emulation service (CES) cell-loss integration period, use the ces cell-loss-integration-period command in interface configuration mode. To delete the cell-loss integration period, use the no form of this command.

ces cell-loss-integration-period period

no ces cell-loss-integration-period period

Syntax Description

period

Time, in milliseconds, for the cell-loss integration period. Range is from 1 to 2147483647.


Defaults

2500 milliseconds

Command Modes

Interface configuration

Command History

Release
Modification

11.3(1)MA

This command was introduced on the Cisco MC3810.


Usage Guidelines

This command applies to ATM configuration on the Cisco MC3810 multiservice concentrator.

This command is supported on serial ports 0 and 1 with encapsulation atm-ces.

Examples

The following example configures the CES cell-loss integration period on serial port 0 to 1056:

interface serial 0
 ces cell-loss-integration-period 1056

Related Commands

Command
Description

cbr

Configures the CBR for the ATM CES for an ATM PVC on the Cisco MC3810.

ces clockmode synchronous

Configures the ATM CES synchronous clock mode on the Cisco MC3810.

ces connect

Maps the CES service to an ATM PVC on the Cisco MC3810.

ces initial-delay

Configures the size of the receive buffer of a CES circuit on the Cisco MC3810 multiservice concentrator.

ces max-buf-size

Configures the send buffer of a CES circuit on the Cisco MC3810.

ces partial-fill

Configures the number of user octets per cell for the ATM CES on the Cisco MC3810.

ces service

Configures the ATM CES type on the Cisco MC3810.

encapsulation atm-ces

Enables CES ATM encapsulation on the Cisco MC3810.


ces clockmode synchronous

To configure the ATM circuit emulation service (CES) synchronous clock mode, use the ces clockmode synchronous command in interface configuration mode. To restore the default value, use the no form of this command.

ces clockmode synchronous

no ces clockmode synchronous

Syntax Description

This command has no arguments or keywords.

Defaults

Enabled

Command Modes

Interface configuration

Command History

Release
Modification

11.3(1)MA

This command was introduced on the Cisco MC3810.


Usage Guidelines

This command maps into the transmit clock source of the constant bit rate (CBR) interface. This command is supported on serial ports 0 and 1 when set for CES ATM encapsulation.

Examples

The following example sets the ATM CES clock to synchronous mode on serial port 0:

interface serial 0
 ces clockmode synchronous

Related Commands

Command
Description

encapsulation atm-ces

Enables CES ATM encapsulation on the Cisco MC3810.


ces connect

To map the circuit emulation service (CES) service to an ATM permanent virtual circuit (PVC) on the Cisco MC3810 multiservice concentrator, use the ces connect command in interface configuration mode. To delete the CES map to the ATM PVC, use the no form of this command.

ces connect atm-interface pvc {name | [vpi/] vci}

no ces connect atm-interface pvc {name | [vpi/] vci}

Syntax Description

atm-interface

Number of the ATM interface. The only valid option on the Cisco MC3810 is ATM0.

pvc

Specifies that the connection is to an ATM PVC.

name

The name of the ATM PVC.

vpi/

(Optional) The virtual path identifier value.

vci

The virtual channel identifier value.


Defaults

No ATM interface is defined

Command Modes

Interface configuration

Command History

Release
Modification

11.3(1)MA

This command was introduced on the Cisco MC3810.


Usage Guidelines

This command is supported on serial ports 0 and 1. The ATM interface must be configured to encapsulation atm-ces, and the vpi/vci must be defined on the interface.

Examples

The following example maps the CES service to PVC 20 on ATM port 0:

ces connect atm0 pvc 20

Related Commands

Command
Description

cbr

Configures the CBR for the ATM CES for an ATM PVC on the Cisco MC3810.

ces cell-loss-integration-period

Sets the CES cell-loss integration period on the Cisco MC3810.

ces clockmode synchronous

Configures the ATM CES synchronous clock mode on the Cisco MC3810.

ces initial-delay

Configures the size of the receive buffer of a CES circuit on the Cisco MC3810.

ces max-buf-size

Configures the send buffer of a CES circuit on the Cisco MC3810.

ces partial-fill

Configures the number of user octets per cell for the ATM CES on the Cisco MC3810.

ces service

Configures the ATM CES type on the Cisco MC3810.

encapsulation atm-ces

Enables CES ATM encapsulation on the Cisco MC3810.