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:
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:
The following example shows H.323 call service being enabled on a Cisco gateway:
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
The following example shows the gateway configured to use the H.323 Version 1 (Slow Connect) procedures.
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-avg—Average CPU utilization.
•io-mem—I/O memory utilization.
•proc-mem—Processor 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 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 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 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:
connection trunk 55Router0
destination pattern 5552150
session protocol frf11-trunk
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:
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:
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:
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:
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:
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:
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:
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:
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:
station number 4085551111
caller-id alerting ring 1
The following example configures a Cisco MC3810 voice port where caller ID information is received:
caller-id alerting ring 1
The following example configures a Cisco MC3810 voice port from which caller ID information is sent:
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:
The following example configures a Cisco MC3810 voice port where caller ID information is received:
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:
station number 4085551111
The following example configures a Cisco MC3810 voice port from which caller ID information is sent:
station number 4085551111
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:
The following example configures a Cisco 2600 series or Cisco 3600 series router voice port from which caller-ID information is sent:
station number 4085551111
The following example configures a Cisco MC3810 voice port where caller-ID information is received:
The following example configures a Cisco MC3810 voice port from which caller-ID information is sent:
station number 4085551111
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.
destination-pattern 5555555
caller-number 1111111 ring 2
caller-number 2222222 ring 1
caller-number 3333333 ring 1
destination-pattern 5552222
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# show running-config
Building configuration...
calling-info pstn-to-sip from name set CompanyA
calling-info pstn-to-sip from number set 5551000
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# show running-config
Building configuration...
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:
calling-number outbound range 4085556000 4085556001
calling Number Outbound is effective only for fgd_eana signaling
Voice-port configuration mode:
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:
calling-number outbound sequence 6000 6006 4000 5000 5025
Calling Number Outbound is effective only for fgd_eana signaling
Voice-port configuration mode:
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:
calling-number outbound null
Calling Number Outbound is effective only for fgd_eana signaling
Voice-port configuration mode:
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:
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:
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:
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:
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:
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.
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:
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
ccm-manager redundant-host 10.0.0.21
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:
•async—Asynchronous interface
•bvi—Bridge-Group Virtual interface
•ctunnel—CTunnel interface
•dialer—Dialer interface
•ethernet—IEEE 802.3
•lex—Lex interface
•loopback—Loopback interface
•mfr—Multilink Frame Relay bundle interface
•multilink—Multilink interface
•null—Null interface
•serial—Serial interface
•tunnel—Tunnel interface
•vif—PGM Multicast Host interface
•virtual-FrameRelay—Virtual frame relay interface
•virtual-Template—Virtual template interface
•virtual-TokenRing—Virtual 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:
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:
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
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Modification
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12.0(7)XK
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This command was introduced for the Cisco MC3810.
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12.1(2)T
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This command was integrated into Cisco IOS Release 12.1(2)T.
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12.1(2)XH
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This command was implemented on the Cisco 2600 series, Cisco 3600 series, Cisco 7200 series, and Cisco 7500 series.
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12.1(3)T
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This command was integrated into Cisco IOS Release 12.1(3)T.
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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:
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.
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Defaults
2500 milliseconds
Command Modes
Interface configuration
Command History
Release
|
Modification
|
11.3(1)MA
|
This command was introduced on the Cisco MC3810.
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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:
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.
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ces clockmode synchronous
|
Configures the ATM CES synchronous clock mode on the Cisco MC3810.
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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:
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.
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name
|
The name of the ATM PVC.
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vpi/
|
(Optional) The virtual path identifier value.
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vci
|
The virtual channel identifier value.
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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:
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.
|