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This chapter documents commands used to configure and monitor Voice over IP (VoIP) with gateway and gatekeeper functionality for the Cisco AS5800.
The following sections organize the VoIP commands by category.
The following commands are associated with system operations:
The following commands are associated with dial peer configuration:
The following commands are associated with system management:
The following commands are associated with gateway configuration:
The following commands are associated with gatekeeper configuration:
Table 5-1 describes the syntax used with the commands in this chapter.
| Convention | Description |
|---|---|
boldface font | Commands and keywords. |
italic font | Command input that is supplied by you. |
[ ] | Optional keywords or arguments. |
{ x | x | x } | Alternate but required arguments and keywords. Keywords (represented by x) appear in braces separated by vertical bars. You must select one. |
^ or Ctrl | Represent the key labeled Control. For example, when you read ^D or Ctrl-D, you should hold down the Control key while you press the D key. |
| Examples of information displayed on the screen. |
boldface screen font | Examples of information that you must enter. |
< > | Nonprinting characters, such as passwords. |
[ ] | Default responses to system prompts. |
Note | Means reader take note. Notes contain helpful suggestions or references to additional information and material. |
| Means the described action saves time. You can save time by performing the action described in the paragraph. |
| Means reader be careful. In this situation, you might do something that could result in equipment damage or loss of data. |
Use this command to define the accounting method list "H.323" with RADIUS as a method and with either stop-only or start-stop accounting options.
aaa accounting connection h323 {stop-only | start-stop} radius
stop-only | start-stop | Start-only or stop-only accounting options. |
radius | RADIUS is used as the method. |
[no] aaa accounting connection h323.
aaa accounting connection h323 {none | start-stop | stop-only | wait-start} radius
Global configuration
This command first appeared in Cisco IOS Release 11.3(6)NA2.
The method list has to be called "h323" and is activated for all voice interfaces.
This command line tells the system to create a method list called H.323 which has start-stop RADIUS as its method. The H.323 method list is static and is applied by default to all voice interfaces if the gw-accounting h323 command is also activated.
The aaa authentication login h323 radius command is used to define a method list called H.323 with RADIUS as a method. Enter the no form of this command to restore the default.
authentication login h323 radiusThis command has no keywords or arguments.
[no] authentication login h323 radius.
Global configuration
This command first appeared in Cisco IOS Release 11.3(6)NA2.
This command line registers the H.323 method list (also referred to as H.323 service) which has RADIUS as its only method in the router. The VoIP calls send all their authentication requests through the H.323 service. If this line does not exist in the router configuration VoIP authentication will not take place.
A significant difference in the usage of this command is that with Cisco IOS Release 11.3(T) the name of the method list is flexible and can be changed by the user. However, when using method list for configuring AAA with the VoIP service provider software, the method list is a specific name that you should not change.
The way method lists work in Cisco IOS software is that a list is defined using the aaa authentication login h323 radius command, and is then applied to an interface. For voice authentication, we never apply this list to any interface. When enabled, using this command applies to all voice interfaces. The function of this command is activated through the IVR application.
best-effort | Indicates that Resource Reservation Protocol (RSVP) makes no bandwidth reservation. |
controlled-load | Indicates that RSVP guarantees a single level of preferential service, presumed to correlate to a delay boundary. The controlled load service uses admission (or capacity) control to ensure that preferential service is received even when the bandwidth is overloaded. |
guaranteed-delay | Indicates that RSVP reserves bandwidth and guarantees a minimum bit rate and preferential queuing if the bandwidth reserved is not exceeded. |
Best-effort. Using the no form of this command is the same as the default.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the acc-qos dial peer command to generate an SNMP event when the quality of service for specified dial peer drops below the specified level. When a dial peer is used, the Cisco IOS software reserves a certain amount of bandwidth so that the selected quality of service can be provided. Cisco IOS software uses Resource Reservation Protocol (RSVP) to request quality of service guarantees from the network.
To select the most appropriate value for this command, you need to be familiar with the amount of traffic this connection supports and what kind of impact you are willing to have on it. The Cisco IOS software generates a trap message when the bandwidth required to provide the selected quality of service is not available.
This command is only applicable to VoIP peers.
The following example selects guaranteed-delay as the specified level below which an SNMP trap message will be generated:
dial-peer voice 10 voip acc-qos guaranteed-delay
You can use the master index or search online to find documentation of related commands.
req-qos
+ | (Optional) Plus sign (+), which can be used as the first digit to indicate an E.164 standard number. |
string | Series of digits that specify the E.164 or private dialing plan telephone number. Valid entries are: · Digits 0 through 9, letters A through D, pound sign (#), and asterisk (*), which represent specific digits that can be entered. · Comma (,), which inserts a pause between digits. · Period (.), which matches any entered digit. |
The default value is answer-address enabled using a null string for the E.164 telephone number.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the answer-address command to identify the origin (or dial peer) of incoming calls from the IP network. Cisco IOS software identifies the dial peers of a call in one of two ways: by identifying the interface through which the call is received or through the telephone number configured with the answer-address command. In the absence of a configured telephone number, the peer associated with the interface will be associated with the incoming call.
For calls coming in from a POTS interface, the answer-address command is not used to select an incoming dial peer. The incoming POTS dial peer is selected on the basis of the port configured for that dial peer.
This command is applicable to both VoIP and POTS dial peers.
The following example configures the E.164 telephone number, "379-9626," as the dial peer of an incoming call:
dial-peer voice 10 pots answer-address 3799626
destination-pattern
port
prefix
name | Indicates the name of the IVR script application the call should be handed to |
None. The call will be handed to the predefined session application.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(6)NA2.
application clid_authen_collect, CallID 90 got event IVR_EV-CALL_SETUP_IND : : ivr action: IVR_ACT_CALL_SETUP_ACK : : ivr action: IVR_ACT_CALL_PROCEEDING : : ivr action: IVR_ACT_CALL_CONNECT : ivr action: IVR_ACT_CALL_PROCEEDING : : ivr action: IVR_ACT_CALL_CONNECT
The arq reject-unknown-prefix command is used to control the behavior of the gatekeeper when it receives an Admission Request (ARQ) which does not match any configured zone prefixes. Use this command to force the gatekeeper to reject such requests. If however, the desired behavior is for the gatekeeper to try to service such requests, then use the no form of this command.
arq reject-unknown-prefix
no arq reject-unknown-prefix
This command has no arguments or keywords.
no arq reject-unknown-prefix.
Gatekeeper configuration
This command first appeared in the Cisco IOS Release 11.3(6)NA.
You can use the arq reject-unknown-prefix command to control the behavior of the gatekeeper when it receives an Admission Request (ARQ) for a destination E.164 address which does not match any configured zone prefixes.
When an endpoint or gateway initiates an H.323 call, it sends an ARQ to its gatekeeper. The gatekeeper uses the configured list of zone prefixes to determine to which zone the call should be directed. If the called address does not match any known zone prefixes, the gatekeeper will attempt to hairpin the call out through a local gateway with a matching technology prefix. If this is not the desired behavior, then use the arq reject-unknown-prefix command to mandate that such calls should be rejected.
This command is typically used either to restrict local gateway calls to a known set of prefixes, or to deliberately fail such calls so that an alternate choice on a gateway's rotary dial-peer can be selected.
The following example shows how this command affects the behavior of a gatekeeper. Consider a gatekeeper configured as follows:
zone local gk408 cisco.com
zone remote gk415 cisco.com 172.21.139.91
zone prefix gk408 1408.......
zone prefix gk415 1415.......
In the above example, the gatekeeper manages a zone containing gateways to the 408 area code, and it knows about a peer gatekeeper with gateways to the 415 area code. These zones are configured with the appropriate prefixes so that calls to those area codes hop off in the optimal zone.
If an endpoint wishes to make a call to the 408 area code, the call will be routed out through a local gateway. If the call is to the 415 area code, it will be directed to the gk415 zone and hop off on a gateway there. But if a call is made to, say, the 212 area code, it will also be directed to a local gateway in the gk408 zone.
Now if you add the command:
arq reject-unknown-prefix
when a call is made to the 212 area code, it is now rejected because the destination address does not match any configured prefixes.
Use the audio-prompt load command to refresh the .au (audio) file in the memory. This is the file which contains the announcement prompt for the caller. The router will only load the .au file when the script initially plays that prompt, or on the router restart. If the .au file is changed, the user must run this EXEC command to reread the file. This will generate an error message if the file is not accessible, or if there is a format error.
audio-prompt load name
audio-prompt load | Initiates to load selected audio file from Flash memory into RAM. |
name | Indicates the location of the .au file to load. It can be loaded from memory, Flash memory, or an FTP server. Presently, with Cisco IOS Release 11.3(6)NA2, the URL pointer refers to the directory where Flash memory is stored. |
None.
Privileged EXEC
This command first appears in Cisco IOS Release 11.3(6)NA2.
The cas-group command is used to configure a channel group for CAS signaling. The cas-group command is used to specify the channels/timeslots to be allocated for the CAS group and the CAS signaling type. Use the no form of this command to disable channel associated signaling for one or more timeslots.
Two new service type attributes (voice and data) have been added to the cas group command.
The command syntax is as follows:
cas-group channel timeslots range type signal [ [tone] [service] ]
no cas-group channel timeslots range type signal [ [tone] [service]
The service type are:
For e&m-fgb and e&m fgd, there are also tone type:
The default tone type is dtmf.
For these two CAS signaling types the tone type is always set to mf and dnis is always required. They can be used for both voice and data calls.
channel | Specifies a single channel group number, which can be between 0 and 23. |
timeslots range | Specifies a timeslot range of values from 1 to 24. |
type signal | Specifies the type of robbed bit signaling. Choose one of the following signal types to configure: · e&m-fgb dtmf [dnis] · e&m-fgd dtmf [dnis] · e&m-immediate-start---Specifies ear and mouth channel signaling with immediate start support. · fxs-loop-start--- Specifies Foreign Exchange Station loopstart signaling support. · fxs-ground-start---Specifies Foreign Exchange Station ground start signaling support. · sas-loop-start---Specifies Special Access Station loopstart signaling support. · sas-ground-start---Specifies Special Access Station ground start signaling support. |
tone | Specify dtmf or mf. |
service | Specify voice or data. |
Controller configuration
This command first appeared in Cisco IOS Release 11.2.
Use this command to enable an integrated modem to receive and transmit incoming and outgoing call signaling (such as on-hook and off-hook) through each T1 controller.
By configuring DNIS as part of the cas-group command, the system can collect DNIS digits for incoming calls which can be directed as VoIP calls, or alternately can be redirected to specific modem pools setup for different customers or uses. To support modems you must be running MICA modems in the system and have at least 10% of your total modems in the default modem pool.
The following example configures the required signaling to support modem pooling and the digital number identification service (DNIS) over channelized T1 lines on a Cisco AS5800.
5800# configure terminal Enter configuration commands, one per line. End with CNTL/Z. 5800(config)# controller t1 0 5800(config-controller)# cas-group 0 timeslots 1-24 type e&m-fgb dtmf dnis 5800(config-controller)# exit 5800(config)# 5800(config)# modem-pool accounts1 5800(config-modem-pool)# pool-range 30-50 5800(config-modem-pool)# called-number 2000 max-conn 21 5800(config-modem-pool)# exit 5800(config)#
g711alaw | G.711 A-Law 64000 bits per second (bps). |
g711ulaw | G.711 u-Law 64000 bps. |
g729r8 | G.729 8000 bps. |
g723r53 | G.723.1 5300 bps. |
g723r63 | G.723.1 6300 bps. |
g726r16 | G.726 16000 bps. |
g726r24 | G.726 24000 bps. |
g726r32 | G.726 32000 bps. |
g728 | G.728 16000 bps. |
g729abr8 | G.729 ANNEX-A & B 8000 bps. |
g729ar8 | G.729 ANNEX-A 8000 bps. |
g729br8 | G.729 ANNEX-B 8000 bps. |
gsmfr | GSMFR 13200 bps. |
The default value for this command is g729r8.
Dial peer configuration
This command first appeared in Cisco IOS Release 12.0(4)XL.
Use the codec command to define a specific voice coder rate of speech for a dial peer.
For toll quality, use g711alaw or g711ulaw. These values provide high-quality voice transmission but use a significant amount of bandwidth. For almost toll quality (and a significant savings in bandwidth), use the g729r8 value.
If codec values for the VoIP peers of a connection do not match, the call will fail.
This command is only applicable to VoIP peers.
The following example configures a voice coder rate that provides toll quality and uses a relatively high amount of bandwidth:
dial-peer voice 10 voip codec g711alaw
+ | (Optional) Plus sign (+), which can be used as the first digit to indicate an E.164 standard number. |
string | Series of digits that specify the E.164 or private dialing plan telephone number. Valid entries are: · Digits 0 through 9, letters A through D, pound sign (#), and asterisk (*), which represent specific digits that can be entered. · Comma (,), which inserts a pause between digits. · Period (.), which matches any entered digit. |
The default value for this command is destination-pattern enabled using a null string.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the destination-pattern command to define the E.164 telephone number for this dial peer. This pattern is used to match dialed digits to a dial peer. The dial peer is then used to complete the call.
This command is applicable to both VoIP and POTS dial peers.
The following example configures the E.164 telephone number, "479-7922," for a dial peer:
dial-peer voice 10 pots destination-pattern 4797922
answer-address
prefix
max-size number | Specifies the maximum size of the call history table. Valid entries are from 0 to 500 table entries. A value of 0 will prevent any history from being retained. |
retain-timer number | Specifies the length of time, in minutes, for entries in the call history table. Valid entries are from 0 to 2147483647 minutes. A value of 0 will prevent any history from being retained. |
The default call history table length is 50 table entries. The default retain timer is 15 minutes.
Global configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
The following example configures the call history table to hold 400 entries, with each entry remaining in the table for 10 minutes:
configure terminal dial-control-mib max-size 400 dial-control-mib retain-timer 10
This command has no arguments or keywords.
The default value for this command is disabled.
Dial peer configuration.
This command first appeared in Cisco IOS Release 11.3 NA.
Use the direct-inward-dial command to enable the DID call treatment for the incoming called numbers. When this feature is enabled, the incoming call is treated as if the digits are received from the DID trunk. The called number is used to select the outgoing dial peer. No dial tone will be presented to the caller.
Use the no form of this command to disable this feature. When disabled, the called number is used to select the outgoing dial peer. The caller will be prompted for a called number via dial tone.
This command is only applicable to POTS peers.
The following example enables DID call treatment for incoming called numbers:
dial peer voice 10 pots direct-inward-dial
session-protocol
Use this command to specify how an H.323 gateway relays DTMF tones between telephony interfaces and an IP network, use the dtmf-relay dial-peer configuration command. Use the no form of this command to remove all signaling options and transmit the DTMF tones as part of the audio stream.
dtmf-relay [cisco-rtp] [h245-alphanumeric] [h245-signal]
cisco-rtp | (Optional) Forwards DTMF tones by using RTP protocol with a Cisco proprietary payload type. |
h245-alphanumeric | (Optional) Forwards DTMF tones by using the H.245 "alphanumeric" User Input Indication method. Supports tones 0-9, *, #, and A-D. |
h245-signal | (Optional) Forwards DTMF tones by using the H.245 "signal" User Input Indication method. Supports tones 0-9, *, #, and A-D. |
Disabled
Dial-peer configuration
This command first appeared in Cisco IOS Release 12.0(4)XL.
The principal advantage of this feature is that it transmits DTMF tones with greater fidelity than is possible in band for most low-bandwidth CODECs, such as G.729 and G.723. Without the use of DTMF-relay, calls established with low-bandwidth CODECs may have trouble accessing automated DTMF-based systems, such as voicemail, menu-based ACD systems, and automated-banking systems.
The following example configures DTMF-relay with cisco-rtp when sending DTMF tones to dial-peer 103:
5800# conf t 5800(config)# dial-peer voice 103 voip 5800(config-dial-peer)# dtmf-relay cisco-rtp 5800(config-dial-peer)# end 5800#
The next example configures the gateway to send DTMF inband (the default) when sending DTMF tones to dial-peer 103:
5800# conf t 5800(config)# dial-peer voice 103 voip 5800(config-dial-peer)# no dtmf-relay 5800(config-dial-peer)# end
The gateway only sends DTMF tones in the format you specify if the remote device supports it. If the remote device supports multiple formats, the gateway chooses the format based on the following priority:
codec
value | Integer(s) that represent the ITU specification for quality of voice as described in G.113. Valid entries are from 0 to 20, with 0 representing toll quality. |
The default value for this command is 10.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Voice over IP monitors the quality of voice received over the network. Use the expect-factor command to specify when the router will generate an SNMP trap to the network manager.
This command is only applicable to VoIP peers.
The following example configures toll quality of voice when connecting to a dial peer:
dial-peer voice 10 voip expect-factor 0
2400 | Specifies a fax transmission speed of 2400 bits per second (bps). |
4800 | Specifies a fax transmission speed of 4800 bps. |
7200 | Specifies a fax transmission speed of 7200 bps. |
9600 | Specifies a fax transmission speed of 9600 bps. |
14400 | Specifies a fax transmission speed of 14,400 bps. |
disable | Disables fax relay transmission capability. |
voice | Specifies the highest possible transmission speed allowed by voice rate. |
The default value for this command is voice.
Dial peer configuration
This command first appeared in Cisco IOS Release 12.0(4)XL.
Use the fax-rate command to specify the fax transmission rate to the specified dial peer.
The values for this command apply only to the fax transmission speed and do not affect the quality of the fax itself. The higher values provide a faster transmission speed but monopolize a significantly larger portion of the available bandwidth. Slower transmission speeds use less bandwidth.
If the fax-rate command is set above the CODEC rate in the same dial peer, the data sent over the network for fax transmission will be above the bandwidth reserved for RSVP. Because more network bandwidth will be monopolized by the fax transmission, we do not recommend setting the fax-rate value higher than the codec value. If the fax-rate value is set lower than the codec value, faxes will take longer to transmit but will use less bandwidth.
This command is only applicable to VoIP peers.
The following example configures a facsimile rate of 9600 bps for faxes sent to a dial peer:
dial-peer voice 10 voip fax-rate 9600
codec
To enable the H.323 VoIP gateway, use the gateway global configuration command. Use the no form of this command to unregister this gateway with the gatekeeper.
gatewayThis command has no keywords or arguments.
The gateway is unregistered.
Global configuration
This command first appeared in Cisco IOS Release 11.3(6)NA2.
Use the gateway command to enable H.323 VoIP gateway functionality. After you enable the gateway, it will attempt to discover a gatekeeper by using the H.323 RAS GRQ message.
The gw-accounting command is used to enable or disable gateway specific accounting.
gw-accounting [h323 | syslog]
| Field | Description |
|---|---|
h323 | H.323 method uses RADIUS to output accounting CDRs. |
syslog | Syslog uses the system logging facility to output CDRs. |
[no] gw-accounting [h323 | syslog].
Global configuration
This command first appeared in Cisco IOS Release 11.3(6)NA2.
To configure a technology prefix in the gatekeeper, use the gw-type-prefix command. To remove the technology prefix, use the no form of the command.
gw-type-prefix type-prefix[hopoff gkid][default-technology]
type-prefix | A technology prefix is recognized and is stripped before checking for the zone prefix. It is strongly recommended that you select technology prefixes that do not lead to ambiguity with zone prefixes. Do this by using the # character to |
hopoff gkid | (Optional) Use this option to specify the gatekeeper or zone where the call is to hop off, regardless of the zone prefix in the destination address. The gkid |
default-technology | (Optional) Gateways registering with this prefix option are used as the default for routing any addresses that are otherwise unresolved. |
gw ipaddr ipaddr [port] | (Optional) Use this option to indicate that the gateway is incapable of registering technology prefixes. When it registers, it adds the gateway to the group for this type-prefix, just as if it had sent the technology prefix in its registration. This parameter can be repeated to associate more than one gateway with a technology prefix. |
No technology prefix is defined.
Gatekeeper configuration
This command first appeared in Cisco Release 11.3(6)NA2.
zone prefix.
The following example specifies 4# as the default technology prefix:
gw-type-prefix 4# default-technology
To configure the H.323 name of the gateway identifying this gateway to its associated gatekeeper, use the h323-gateway voip h.323-id interface configuration command. Use the no form of this command to disable this defined gatekeeper name.
h323-gateway voip h323-id interface-id
interface-id | H.323 name (ID) used by this gateway when this gateway communicates with its associated gatekeeper. Usually, this ID is the name of the gateway with the gatekeeper's domain name appended to the end: name@domain-name. |
No gateway identification is defined.
Interface configuration
This command first appeared in Cisco IOS Release 11.3(6)NA2.
The following example configures Ethernet interface 0.0 as the gateway interface. In this example, the gateway ID is GW13@cisco.com.
interface FastEthernet0/0/0 ip address 172.9.53.13 255.255.255.0 h323-gateway voip interface h323-gateway voip id GK15.cisco.com ipaddr 172.9.53.15 1719 h323-gateway voip h323-id GW13@cisco.com h323-gateway voip tech-prefix 13#
h323-gateway voip id
h323-gateway voip interface
h323-gateway voip tech-prefix 13#
To define the name and location of the gatekeeper for this gateway, use the h323-gateway voip id interface configuration command. Use the no form of this command to disable this gatekeeper identification.
h323-gateway voip id gatekeeper-id {ipaddr ip-address [port-number] | multicast}
gatekeeper-id | Indicates the H.323 identification of the gatekeeper. This value must exactly match the gatekeeper ID in the gatekeeper configuration. The recommended format is name.doman-name. |
ipaddr | Indicates that the gateway will use an IP address to locate the gatekeeper |
ip-address | Defines the IP address used to identify the gatekeeper. |
multicast | Indicates that the gateway will use multicast to locate the gatekeeper. |
No gatekeeper identification is defined.
Interface configuration
This command first appeared in Cisco IOS Release 11.3(6)NA2.
This command tells the H.323 gateway associated with this interface which H.323 gatekeeper to talk to and where to locate it. The gatekeeper ID configured here must exactly match the gatekeeper ID in the gatekeeper configuration.
The following example configures Ethernet interface 0.0 as the gateway interface. In this example, the gatekeeper ID is GW15.cisco.com and its IP address is 172.9.53.15 (using port 1719).
interface FastEthernet0/0/0 ip address 172.9.53.13 255.255.255.0 h323-gateway voip interface h323-gateway voip id GK15.cisco.com ipaddr 172.9.53.15 1719 h323-gateway voip h323-id GW13@cisco.com h323-gateway voip tech-prefix 13#
h323-gateway voip h323-id
h323-gateway voip interface
h323-gateway voip tech-prefix 13#
To configure this interface as an H.323 interface, use the h323-gateway voip interface interface configuration command. Use the no form of this command to disable H.323 functionality for this interface.
h323-gateway voip interfaceThis command has no arguments or keywords.
Disabled
Interface configuration
This command first appeared in Cisco IOS Release 11.3(6)NA2.
The following example configures Ethernet interface 0.0 as the gateway interface. In this example, the h323-gateway voip interface command configures this interface as an H.323 interface.
interface FastEthernet0/0/0 ip address 172.9.53.13 255.255.255.0 h323-gateway voip interface h323-gateway voip id GK15.cisco.com ipaddr 172.9.53.15 1719 h323-gateway voip h323-id GW13@cisco.com h323-gateway voip tech-prefix 13#
h323-gateway voip h323-id
h323-gateway voip id
h323-gateway voip tech-prefix 13#
To define the technology prefix that the gateway will register with the gatekeeper, use the h323-gateway voip tech-prefix interface configuration command. Use the no form of this command to disable this defined technology prefix.
h323-gateway voip tech-prefix prefix
prefix | Defines the numbers used as the technology prefix. Each technology prefix can contain up to 11 characters. Although not strictly necessary, a pound (#) symbol is frequently used as the last digit in a technology prefix. Valid characters 0 though 9, the pound (#) symbol, and the asterisk (*). |
Disabled
Interface configuration
This command first appeared in Cisco IOS Release 11.3(6)NA2.
This command defines a technology prefix that the gateway will then register with the gatekeeper. Technology prefixes can be used as a discriminator so that the gateway can tell the gatekeeper that a certain technology is associated with a particular call (for example, 15# could mean a fax transmission), or it can be used like an area code for more generic routing. No standard currently defines what the numbers in a technology prefix mean. By convention, technology prefixes are designated by a pound (#) symbol as the last character.
The following example configures Ethernet interface 0.0 as the gateway interface. In this example, the technology prefix is defined as 13#.
interface FastEthernet0/0/0 ip address 172.9.53.13 255.255.255.0 h323-gateway voip interface h323-gateway voip id GK15.cisco.com ipaddr 172.9.53.15 1719 h323-gateway voip h323-id GW13@cisco.com h323-gateway voip tech-prefix 13#
h323-gateway voip id
h323-gateway voip interface
h323-gateway voip h323-id
number | Integer, expressed in equipment impairment factor units, specifying the ICPIF value. Valid entries are 0 to 55. |
The default value for this command is 30.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the icpif command to specify the maximum acceptable impairment factor for the voice calls sent by the selected dial peer.
This command is applicable only to VoIP peers.
The following example disables the icpif command:
dial-peer voice 10 voip icpif 0
string | Specifies the destination telephone number. Valid entries are any series of digits that specify the E.164 telephone number. |
The default value for this command is no associated called number.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3 NA.
When the Cisco AS5800 is handling both modem and voice calls, it needs to be able to identify the service type of the call---meaning whether or not the incoming call to the server is a modem or a voice call. When the access server handles only modem calls, the service type identification is handled through modem pools. Modem pools associate calls with modem resources based on the called-number (DNIS). In a mixed environment, where the server receives both modem and voice calls, you need to identify the service type of a call by using the incoming called-number command.
Without this, the server attempts to resolve whether an incoming call is a modem or voice call based on the interface over which the call comes. If the call comes in over an interface associated with a modem pool, the call is assumed to be a modem call; if a call comes in over a voice port associated with a dial peer, the call is assumed to be a voice call.
By default, there is no called number associated with the dial peer, which means that incoming calls will be associated with dial peers based on matching calling number with answer address, call number with destination pattern, or calling interface with configured interface.
This command is applicable to both VoIP and POTS dial peers.
The following example configures calls coming in to the server with a called number of "3799262" as being voice calls:
dial peer voice 10 pots incoming called-number 3799262
number | Integer specifying the IP precedence value. Valid entries are from 0 to 7. A value of 0 means that no precedence (priority) has been set. |
The default value for this command is 0.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3 NA.
Use the ip precedence command to configure the value set in the IP precedence field when voice data packets are sent over the IP network. This command should be used if the IP link utilization is high and the quality of service for voice packets need to have a higher priority than other IP packets. The ip precedence command should also be used if RSVP is not enabled, and the user would like to give voice packets a higher priority over other IP data traffic.
This command is applicable to VoIP peers.
The following example sets the IP precedence to 5:
dial-peer voice 10 voip ip precedence 5
This command has no arguments or keywords.
The default value for this command is disabled.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the ip udp checksum command to enable UDP checksum calculation for each of the outbound voice packets. This command is disabled by default to speed up the transmission of the voice packets. If you suspect that the connection has a high error rate, you should enable ip udp checksum to prevent bad voice packets forwarded to the DSP.
This command is applicable to VoIP peers.
The following example calculates the UDP checksum for voice packets transmitted by this dial peer:
dial-peer voice 10 voip ip udp checksum
The lrq reject-unknown-prefix command is used to control the behavior of the gatekeeper when it receives a Location Request (LRQ) which does not match any configured zone prefixes. Use this command to force the gatekeeper to reject such requests. If however, the desired behavior is for the gatekeeper to try to service such requests, then use the no form of this command.
lrq reject-unknown-prefix
no lrq reject-unknown-prefix
This command has no arguments or keywords.
No lrq reject-unknown-prefix.
Gatekeeper configuration
This command first appeared in Cisco IOS Release 11.3(6)NA.
You can use the lrq reject-unknown-prefix command to control the behavior of the gatekeeper when it receives a Location Request (LRQ) which does not match any configured zone prefixes.
When the gatekeeper receives a Location Request asking about an E.164 address, it matches the target address against the list of configured zone prefixes. If the address matches a zone prefix, the behavior is unambiguous and well-defined:
However. if the target address does not match any known local or remote zone prefixes, then the default behavior is to attempt to service the call using one of the local zones. This default behavior may not be suitable for all sites, so the lrq reject-unknown-prefix command allows you to force the gatekeeper to reject such requests.
The following example shows how this command affects the behavior of a gatekeeper. Consider a gatekeeper configured as follows:
zone local gk408 cisco.com
zone local gk415 cisco.com
zone prefix gk408 1408.......
zone prefix gk415 1415.......
lrq reject-unknown-prefix
In the example, the gatekeeper manages two zones, one with gateways with interfaces in the 408 area code, and one with gateways in the 415 area code. These zones are configured with the appropriate prefixes so that calls to those area codes hop off in the optimal zone. If zone had been erroneously configured to route calls to the 212 area code to this gatekeeper, when the Location Request arrives, this gatekeeper fails to match the area code, and so the LRQ is rejected.
However, if this was your only site that had any gateways in it, and you wanted your other sites to route all calls requiring gateways to this gatekeeper, then you would undo the reject command:
no lrq reject-unknown-prefix
Now, with this command entered, when the gatekeeper receives an LRQ for the address 12125551234, it will attempt to find an appropriate gateway in either one of the zones gk408 or gk415 to service the call.
number | Integer specifying the maximum connections value. Valid values range from 1 to 2147483647. |
The default value for this command is no connection limit configured.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3 NA.
Use the maximum connections command to define the maximum number of connections allowed to and from this dial peer.
This command is applicable to both VoIP and POTS dial peers.
The following example specifies the maximum connection to and from VoIP dial peer 10 to be 3:
dial-peer voice 10 voip maximum connections 3
extension-number | Digit(s) defining an extension number for a particular dial peer. |
expanded-number | Digit(s) defining the expanded telephone number or destination pattern for the extension number listed. |
Global configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the num-exp global configuration command to define how to expand a particular set of numbers (for example, an extension number) into a particular destination pattern. With this command, you can map specific extensions and expanded numbers together by explicitly defining each number, or you can define extensions and expanded numbers using variables. You can also use this command to convert seven-digit numbers to numbers containing less than seven digits.
Use a period (.) as a variable or wild card, representing a single number. Use a separate period for each number you want to represent with a wildcard---meaning that if you want to replace four numbers in an extension with wildcards, enter four periods.
The following example expands the extension number 65541 to be expanded to 14085665541:
num-exp 65541 14085665541
The following example shows how to expand all five-digit extensions beginning with 6 to append the following numbers at the beginning of the extension number 1408566:
num-exp 6.... 1408566....
shelf/slot/port:D | Specifies the T1 or E1 controller; :D indicates the D-channel associated with ISDN PRI. |
No port is configured.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3 T.
Use the port configuration command to associate a destination number with a PRI span.
Use this command for calls incoming from a telephony interface to select an incoming dial peer and for calls coming from the VoIP network to match a port with the selected outgoing dial peer.
This command is applicable only to POTS peers.
The following example associates POTS dial peer 10 with voice port 1/0/0:D:
dial-peer voice 10 pots port 1/0/0:D
The preference command is used to indicate the preference order for matching dial peers in a rotary group. It is useful in selecting the desired dial peer when multiple dial peers are matched for a dial string. The no form of this command does not assign a prefrence.
dial-peer config mode
preference <value>
Value---An integer value [0..10]
No preference order is given.
value 0 [0 - 10]
(highest preference = 0)
This command first appeared in Cisco IOS Release 11.3(7)NA2.
The following examples show different dial peer configurations using the preference command.
Example 1
Dialpeer destpat preference session-target 1 4085271048 0 (highest) jmmurphy-voip 2 408527 0 sj-voip 3 408527 1 (lower) backup-sj-voip 4 .......... 1 0:D (interface) 5 .......... 0 anywhere-voip
If the destination number is 4085271048, the order of attempts will be 1,2,3,5,4.
Example 2
Dialpeer destpat preference 1 408527 0 2 4085271048 1 3 4085271 0 4 ..............4085271.........0
The number dialed is 4085271048, the order will be 2, 3, 4, 1.
string | Integers representing the prefix of the telephone number associated with the specified dial peer. Valid numbers are 0 through 9, and a comma (,). Use a comma to include a pause in the prefix. |
The default for this command is a null string.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the prefix command to specify a prefix for a specific dial peer. When an outgoing call is initiated to this dial peer, the prefix string value is sent to the telephony interface first, before the telephone number associated with the dial peer.
If you want to configure different prefixes for dialed numbers on the same interface, you need to configure different dial peers.
This command is applicable only to POTS peers.
The following example specifies a prefix of "9" and then a pause:
dial-peer voice 10 pots prefix 9,
answer-address
destination-pattern
best-effort | Indicates that Resource Reservation Protocol (RSVP) makes no bandwidth reservation. |
controlled-load | Indicates that RSVP guarantees a single level of preferential service, presumed to correlate to a delay boundary. The controlled load service uses admission (or capacity) control to ensure that preferential service is received even when the bandwidth is overloaded. |
guaranteed-delay | Indicates that RSVP reserves bandwidth and guarantees a minimum bit rate and preferential queuing if the bandwidth reserved is not exceeded. |
The default value for this command is best-effort. The no form of this command restores the default value.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the req-qos command to request a specific quality of service to be used in reaching a dial peer. Like acc-qos, when you issue this command, the Cisco IOS software reserves a certain amount of bandwidth so that the selected quality of service can be provided. Cisco IOS software uses Resource Reservation Protocol (RSVP) to request quality of service guarantees from the network.
This command is applicable only to VoIP peers.
The following example configures guaranteed-delay as the desired quality of service to a dial peer:
dial-peer voice 10 voip req-qos guaranteed-delay
acc-qos
cisco | Specifies Cisco Session Protocol. |
The default value for this command is cisco.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
For this release, Cisco Session Protocol (cisco) is the only applicable session protocol. This command is applicable only to VoIP peers.
The following example selects Cisco Session Protocol as the session protocol:
dial-peer voice 10 voip session protocol cisco
session target
The session target command is used to identify the IP address of the destination gatekeeper. The field indicating if the RAS protocol is being used has been added. Enter the no form of this command to restore the default condition.
session target ras
ipv4:destination-address | IP address of the dial peer. |
dns:host-name | Indicates that the domain name server will be used to resolve the name of the IP address. Valid entries for this parameter are characters representing the name of the host device. (Optional) You can use one of the following four wildcards with this keyword when defining the session target for VoIP peers: · $s$.---Indicates that the source destination pattern will be used as part of the domain name. · $d$.---Indicates that the destination number will be used as part of the domain name. · $e$.---Indicates that the digits in the called number will be reversed, periods will be added in-between each digit of the called number, and that this string will be used as part of the domain name. · $u$.---Indicates that the unmatched portion of the destination pattern (such as a defined extension number) will be used as part of the domain name. |
loopback:rtp | Indicates that all voice data will be looped back to the originating source. This is applicable for VoIP peers. |
loopback:compressed | Indicates that all voice data will be looped back in compressed mode to the originating source. This is applicable for POTS peers. |
loopback:uncompressed | Indicates that all voice data will be looped-back in uncompressed mode to the originating source. This is applicable for POTS peers. |
ras | Indicates that the Registration, Admission, and Status (RAS) signaling function protocol is being used: a gatekeeper will be consulted to translate the E.164 address to an IP address. |
The default for this command is enabled with no IP address or domain name defined.
Dial-peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the session target command to specify a network-specific address or domain name for a dial peer. Whether you select a network-specific address or a domain name depends on the session protocol you select.
The session target loopback command is used for testing the voice transmission path of a call. The loopback point will depend on the call origination and the loopback type selected.
The session target dns command can be used with or without the specified wildcards. Using the optional wildcards can reduce the number of VoIP dial peer session targets you need to configure if you have groups of numbers associated with a particular router.
Use the session target ras command to specify that the RAS protocol is being used to determine the IP address of the session target.
The following example configures a session target using DNS for a host, "voice_router," in the domain "cisco.com":
dial-peer voice 10 voip session target dns:voice_router.cisco.com
The following example configures a session target using DNS, with the optional $u$. wildcard. In this example, the destination pattern has been configured to allow for any four-digit extension, beginning with the numbers 1310222. The optional wildcard $u$. indicates that the router will use the unmatched portion of the dialed number---in this case, the four-digit extension, to identify the dial peer. As in the previous example, the domain is "cisco.com."
dial-peer voice 10 voip destination-pattern 1310222.... session target dns:$u$.cisco.com
The following example configures a session target using dns, with the optional $d$. wildcard. In this example, the destination pattern has been configured for 13102221111. The optional wildcard $d$. indicates that the router will use the destination pattern to identify the dial peer in the "cisco.com" domain.
dial-peer voice 10 voip destination-pattern 13102221111 session target dns:$d$.cisco.com
The following example configures a session target using DNS, with the optional $e$. wildcard. In this example, the destination pattern has been configured for 12345. The optional wildcard $e$. indicates that the router will reverse the digits in the destination pattern, add periods between the digits, and then use this reverse-exploded destination pattern to identify the dial peer in the "cisco.com" domain.
dial-peer voice 10 voip destination-pattern 12345 session target dns:$e$.cisco.com
The following example configures a session target using RAS:
dial-peer voice 11 vofr destination-pattern 13102221111 session target ras
destination-pattern
session protocol
To show the active call table, use the show call active voice privileged EXEC command.
show call active voiceThis command contains no arguments or keywords.
Privileged EXEC
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the show call active voice privileged EXEC command to display the contents of the active call table, which shows all of the calls currently connected through the router.
For each call, there are two call legs, usually a POTS call leg and a VoIP call leg. A call leg is a discrete segment of a call connection that lies between two points in the connection. Each dial peer creates a call leg, as shown in Figure 5-1.
These two call legs are associated by the connection ID. The connection ID is global across the voice network, so that you can associate two call legs on one router with two call legs on another router, thereby providing an end-to-end view of a call.
The following is sample output from the show call active voice command:
5800# show call active voice GENERIC: SetupTime=179388054 ms Index=1 PeerAddress=+5.... PeerSubAddress= PeerId=5 PeerIfIndex=32 LogicalIfIndex=29 ConnectTime=179389793 ms CallState=4 CallOrigin=2 ChargedUnits=0 InfoType=2 TransmitPackets=532 TransmitBytes=10640 ReceivePackets=147 ReceiveBytes=2940 TELE: ConnectionId=[0xE3EA3FF8 0xFF6D0105 0x0 0x6AEC71E4] TxDuration=23230 ms VoiceTxDuration=2940 ms FaxTxDuration=0 ms CoderTypeRate=g729r8 NoiseLevel=-84 ACOMLevel=20 OutSignalLevel=-66 InSignalLevel=-66 InfoActivity=2 ERLLevel=20 SessionTarget= GENERIC: SetupTime=179388237 ms Index=1 PeerAddress=+3622 PeerSubAddress= PeerId=3 PeerIfIndex=31 LogicalIfIndex=0 ConnectTime=179389793 ms CallState=4 CallOrigin=1 ChargedUnits=0 InfoType=2 TransmitPackets=143 TransmitBytes=2860 ReceivePackets=580 ReceiveBytes=11600 VOIP: ConnectionId[0xE3EA3FF8 0xFF6D0105 0x0 0x6AEC71E4] RemoteIPAddress=172.24.96.200 RemoteUDPPort=16422 RoundTripDelay=37 ms SelectedQoS=best-effort SessionProtocol=cisco SessionTarget=ipv4:172.24.96.200 OnTimeRvPlayout=9920 GapFillWithSilence=0 ms GapFillWithPrediction=0 ms GapFillWithInterpolation=0 ms GapFillWithRedundancy=0 ms HiWaterPlayoutDelay=70 ms LoWaterPlayoutDelay=30 ms ReceiveDelay=30 ms VAD = enabled CoderTypeRate=g729r8
Table 5-2 provides an alphabetical listing of the possible show call active voice fields and a description of each field.
| Field | Description |
|---|---|
ACOM Level | Current ACOM level for the call. This value is the sum of the Echo Return Loss, Echo Return Loss Enhancement, and nonlinear processing loss for the call. |
CallOrigin | Call origin: answer or originate. |
CallState | Current state of the call. |
CoderTypeRate | Negotiated coder transmit rate of voice/fax compression during the call. |
ConnectionId | Global call identifier of a gateway call. |
ConnectTime | Time at which the call was connected. |
Dial-Peer | Tag of the dial peer transmitting this call. |
ERLLevel | Current Echo Return Loss (ERL) level for this call. |
FaxTxDuration | Duration of fax transmission from this peer to voice gateway for this call. You can derive the Fax Utilization Rate by dividing the FaxTxDuration value by the TxDuration value. |
GapFillWith Silence | Duration of voice signal replaced with silence because voice data was lost or not received on time for this call. |
GapFillWithPrediction | Duration of voice signal played out with signal synthesized from parameters or samples of data preceding in time because voice data was lost or not received in time from the voice gateway for this call. An example of such pullout is frame-eraser or frame-concealment strategies in G.729 and G.723.1 compression algorithms. |
GapFillWithInterpolation | Duration of voice signal played out with signal synthesized from parameters or samples of data preceding and following in time because voice data was lost or not received on time from voice gateway for this call. |
GapFillWith Redundancy | Duration of voice signal played out with signal synthesized from redundancy parameters available because voice data was lost or not received on time from voice gateway for this call. |
HiWaterPlayoutDelay | High water mark Voice Playout FIFO Delay during this call. |
Index | Dial peer identification number. |
InfoActivity | Active information transfer activity state for this call. |
InfoType | Information type for this call. |
InSignalLevel | Active input signal level from the telephony interface used by this call. |
LogicalIfIndex | Index number of the logical interface for this call. |
LoWaterPlayoutDelay | Low water mark Voice Playout FIFO Delay during the call. |
NoiseLevel | Active noise level for the call. |
OnTimeRvPlayout | Duration of voice playout from data received on time for this call. You can derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values. |
OutSignalLevel | Active output signal level to telephony interface used by this call. |
PeerAddress | Destination pattern associated with this peer. |
PeerId | ID value of the peer table entry to which this call was made. |
PeerIfIndex | Voice-port index number for this peer. |
PeerSubaddress | Subaddress to which this call is connected. |
ReceiveBytes | Number of bytes received by the peer during this call. |
ReceiveDelay | Average Playout FIFO Delay plus the Decoder Delay during the voice call. |
ReceivePackets | Number of packets received by this peer during this call. |
RemoteIPAddress | Remote system IP address for the VoIP call. |
RemoteUDPPort | Remote system UDP listener port to which voice packets are transmitted. |
RoundTripDelay | Voice packet round trip delay between the local and remote system on the IP backbone during the call. |
SelectedQoS | Selected RSVP quality of service (QoS) for the call. |
SessionProtocol | Session protocol used for an Internet call between the local and remote router via the IP backbone. |
SessionTarget | Session target of the peer used for the call. |
SetupTime | Value of the system UpTime when the call associated with this entry was started. |
TransmitBytes | Number of bytes transmitted from this peer during the call. |
TransmitPackets | Number of packets transmitted from this peer during the call. |
TxDuration | Duration of transmit path open from this peer to the voice gateway for the call. |
VADEnable | Whether or not voice activation detection (VAD) was enabled for this call. |
VoiceTxDuration | Duration of voice transmission from this peer to voice gateway for this call. You can derive the Voice Utilization Rate by dividing the VoiceTxDuration value by the TxDuration value. |
show call history voice
show dial-peer voice
show num-exp
show voice port
The show call application voice command defines the names of the audio files the script will play the operation of the abort keys, the prompts used, and caller interaction.
show call application voice [name | summary]
| Field | Description |
|---|---|
name | The name of the desired IVR application. |
summary | Enter this field to display a one line summary. If the command is entered without summary, a complete detailed description is displayed of the application. |
Privileged EXEC (also called enable mode)
This command first appeared in Cisco IOS Release 11.3(6)NA2.
Cautionsblab115>show call application voice clid_authen_collect
Application clid_authen_collect has 10 states with 0 calls active
State start has 1 actions and 5 events
Do Action IVR_ACT_AUTHENTICATE. accountName=ani, pinName=dnis
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_CALL_SETUP_IND do action IVR_ACT_CALL_SETUP_ACK
and goto state start
If Event IVR_EV_AAA_SUCCESS goto state collect_dest
If Event IVR_EV_AAA_FAIL goto state get_account
State end has 1 actions and 3 events
Do Action IVR_ACT_END.
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_CALL_DISCONNECT_DONE do action IVR_ACT_CALL_DESTROY
and do nothing
State get_account has 4 actions and 7 events
Do Action IVR_ACT_PLAY.
URL: flash:enter_account.au
allowInt=1, pContent=0x60E4C564
Do Action IVR_ACT_ABORT_KEY. abortKey=*
Do Action IVR_ACT_TERMINATION_KEY. terminationKey=#
Do Action IVR_ACT_COLLECT_PATTERN. Pattern account is .+
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_PAT_COL_SUCCESS goto state get_pin
patName=account
If Event IVR_EV_ABORT goto state get_account
If Event IVR_EV_PLAY_COMPLETE do nothing
If Event IVR_EV_TIMEOUT goto state get_account count=0
If Event IVR_EV_PAT_COL_FAIL goto state get_account
State get_pin has 4 actions and 7 events
Do Action IVR_ACT_PLAY.
URL: flash:enter_pin.au
allowInt=1, pContent=0x0
Do Action IVR_ACT_ABORT_KEY. abortKey=*
Do Action IVR_ACT_TERMINATION_KEY. terminationKey=#
Do Action IVR_ACT_COLLECT_PATTERN. Pattern pin is .+
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_PAT_COL_SUCCESS goto state authenticate
patName=pin
If Event IVR_EV_PLAY_COMPLETE do nothing
If Event IVR_EV_ABORT goto state get_account
If Event IVR_EV_TIMEOUT goto state get_pin count=0
If Event IVR_EV_PAT_COL_FAIL goto state get_pin
State authenticate has 1 actions and 5 events
Do Action IVR_ACT_AUTHENTICATE. accountName=account, pinName=pin
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_AAA_SUCCESS goto state collect_dest
If Event IVR_EV_TIMEOUT do nothing count=0
If Event IVR_EV_AAA_FAIL goto state authenticate_fail
State collect_dest has 4 actions and 8 events
Do Action IVR_ACT_PLAY.
URL: flash:enter_destination.au
allowInt=1, pContent=0x0
Do Action IVR_ACT_ABORT_KEY. abortKey=*
Do Action IVR_ACT_TERMINATION_KEY. terminationKey=#
Do Action IVR_ACT_COLLECT_DIALPLAN.
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_PLAY_COMPLETE do nothing
If Event IVR_EV_ABORT goto state collect_dest
If Event IVR_EV_TIMEOUT goto state collect_dest count=0
If Event IVR_EV_DIAL_COL_SUCCESS goto state place_call
If Event IVR_EV_DIAL_COL_FAIL goto state collect_dest
If Event IVR_EV_TIMEOUT goto state collect_dest count=0
State place_call has 1 actions and 4 events
Do Action IVR_ACT_PLACE_CALL.
destination= called=
calling= account=
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_CALL_UP goto state active
If Event IVR_EV_CALL_FAIL goto state place_fail
State active has 0 actions and 2 events
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
State authenticate_fail has 1 actions and 2 events
Do Action IVR_ACT_PLAY.
URL: flash:auth_failed.au
allowInt=0, pContent=0x0
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
State place_fail has 1 actions and 2 events
Do Action IVR_ACT_PLAY_FAILURE_TONE.
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
sblab115>show call application voice clid_authen_collect
Application clid_authen_collect has 10 states with 0 calls active
State start has 1 actions and 5 events
Do Action IVR_ACT_AUTHENTICATE. accountName=ani, pinName=dnis
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_CALL_SETUP_IND do action IVR_ACT_CALL_SETUP_ACK
and goto state start
If Event IVR_EV_AAA_SUCCESS goto state collect_dest
If Event IVR_EV_AAA_FAIL goto state get_account
State end has 1 actions and 3 events
Do Action IVR_ACT_END.
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_CALL_DISCONNECT_DONE do action IVR_ACT_CALL_DESTROY
and do nothing
State get_account has 4 actions and 7 events
Do Action IVR_ACT_PLAY.
URL: flash:enter_account.au
allowInt=1, pContent=0x60E4C564
Do Action IVR_ACT_ABORT_KEY. abortKey=*
Do Action IVR_ACT_TERMINATION_KEY. terminationKey=#
Do Action IVR_ACT_COLLECT_PATTERN. Pattern account is .+
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_PAT_COL_SUCCESS goto state get_pin
patName=account
If Event IVR_EV_ABORT goto state get_account
If Event IVR_EV_PLAY_COMPLETE do nothing
If Event IVR_EV_TIMEOUT goto state get_account count=0
If Event IVR_EV_PAT_COL_FAIL goto state get_account
State get_pin has 4 actions and 7 events
Do Action IVR_ACT_PLAY.
URL: flash:enter_pin.au
allowInt=1, pContent=0x0
Do Action IVR_ACT_ABORT_KEY. abortKey=*
Do Action IVR_ACT_TERMINATION_KEY. terminationKey=#
Do Action IVR_ACT_COLLECT_PATTERN. Pattern pin is .+
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_PAT_COL_SUCCESS goto state authenticate
patName=pin
If Event IVR_EV_PLAY_COMPLETE do nothing
If Event IVR_EV_ABORT goto state get_account
If Event IVR_EV_TIMEOUT goto state get_pin count=0
If Event IVR_EV_PAT_COL_FAIL goto state get_pin
State authenticate has 1 actions and 5 events
Do Action IVR_ACT_AUTHENTICATE. accountName=account, pinName=pin
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_AAA_SUCCESS goto state collect_dest
If Event IVR_EV_TIMEOUT do nothing count=0
If Event IVR_EV_AAA_FAIL goto state authenticate_fail
State collect_dest has 4 actions and 8 events
Do Action IVR_ACT_PLAY.
URL: flash:enter_destination.au
allowInt=1, pContent=0x0
Do Action IVR_ACT_ABORT_KEY. abortKey=*
Do Action IVR_ACT_TERMINATION_KEY. terminationKey=#
Do Action IVR_ACT_COLLECT_DIALPLAN.
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_PLAY_COMPLETE do nothing
If Event IVR_EV_ABORT goto state collect_dest
If Event IVR_EV_TIMEOUT goto state collect_dest count=0
If Event IVR_EV_DIAL_COL_SUCCESS goto state place_call
If Event IVR_EV_DIAL_COL_FAIL goto state collect_dest
If Event IVR_EV_TIMEOUT goto state collect_dest count=0
State place_call has 1 actions and 4 events
Do Action IVR_ACT_PLACE_CALL.
destination= called=
calling= account=
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
If Event IVR_EV_CALL_UP goto state active
If Event IVR_EV_CALL_FAIL goto state place_fail
State active has 0 actions and 2 events
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
State authenticate_fail has 1 actions and 2 events
Do Action IVR_ACT_PLAY.
URL: flash:auth_failed.au
allowInt=0, pContent=0x0
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
State place_fail has 1 actions and 2 events
Do Action IVR_ACT_PLAY_FAILURE_TONE.
If Event IVR_EV_DEFAULT goto state end
If Event IVR_EV_CALL_DIGIT do nothing
To display the call history table, use the show call history voice privileged EXEC command.
show call history voice [last number | brief]
last number | (Optional) Displays the last calls connected, where the number of calls displayed is defined by the argument number. Valid values are from 1 to 2147483647. |
brief | (Optional) Displays a truncated version of the call history table. |
Privileged EXEC
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the show call history voice privileged EXEC command to display the call history table. The call history table contains a listing of all calls connected through this router in descending time order since Voice over IP was enabled. You can display subsets of the call history table by using specific keywords. To display the last calls connected through this router, use the keyword last, and define the number of calls to be displayed with the argument number.
The following is sample output from the show call history voice command:
5800# show call history voice brief
<ID>: <start>hs.<index> +<connect> +<disc> pid:<peer_id> <direction> <addr>
tx:<packets>/<bytes> rx:<packets>/<bytes> <disc-cause>(<text>)
IP <ip>:<udp> rtt:<time>ms pl:<play>/<gap>ms delay:<last>/<min>/<max>ms <codec>
Telephony <int>: tx:<tot>/<voice>/<fax>ms <codec> noise:<lvl>dBm acom:<lvl>dBm
234 : 158305740hs.1280 +241 +9199 pid:0 Answer +3...
tx:3804/76080 rx:1358/27160 10 (normal call clearing.)
IP 172.24.96.200:16468 rtt:33ms pl:25990/0ms delay:30/30/70ms g729r8
234 : 158305745hs.1281 +236 +9195 pid:6 Originate +68888
tx:1358/27160 rx:3804/76080 10 (normal call clearing.)
Telephony 0:D:22: tx:91850/76080/0ms g729r8 noise:-84dBm acom:20dBm
235 : 158344850hs.1282 +230 +28773 pid:0 Answer +3...
tx:11063/221260 rx:4604/92080 10 (normal call clearing.)
IP 172.24.96.200:16474 rtt:41ms pl:88260/290ms delay:40/30/130ms g729r8
235 : 158344856hs.1283 +224 +28769 pid:6 Originate +68888
tx:4604/92080 rx:11063/221260 10 (normal call clearing.)
Telephony 0:D:22: tx:287590/221280/0ms g729r8 noise:-75dBm acom:20dBm
Table 5-3 provides an alphabetical listing of the possible fields for the show call history voice command and a description of each field.
| Field | Description |
|---|---|
ACOMLevel | Average ACOM level for this call. This value is the sum of the Echo Return Loss, Echo Return Loss Enhancement, and nonlinear processing loss for the call. |
CallOrigin | Call origin: answer or originate. |
CoderTypeRate | Negotiated coder rate. This value specifies the transmit rate of voice/fax compression to its associated call leg for the call. |
ConnectionID | Global call identifier for the gateway call. |
ConnectTime | Time the call was connected. |
DisconnectCause | Description explaining why the call was disconnected. |
DisconnectText | Descriptive text explaining the disconnect reason. |
DisconnectTime | Time the call was disconnected. |
FaxDuration | Duration of fax transmitted from this peer to the voice gateway for this call. You can derive the Fax Utilization Rate by dividing this value by the TxDuration value. |
GapFillWithSilence | Duration of voice signal replaced with silence because the voice data was lost or not received on time for this call. |
GapFillWithPrediction | Duration of voice signal played out with signal synthesized from parameters or samples of data preceding in time because the voice data was lost or not received on time from the voice gateway for this call. |
GapFillWithInterpolation | Duration of voice signal played out with signal synthesized from parameters or samples of data preceding and following in time because the voice data was lost or not received on time from the voice gateway for this call. |
GapFillWithRedundancy | Duration of voice signal played out with signal synthesized from redundancy parameters available because the voice data was lost or not received on time from the voice gateway for this call. |
HiWaterPlayoutDelay | High water mark Voice Playout FIFO Delay during the voice call. |
Index | Index number identifying the dial peer for this call. |
InfoType | Information type for this call. |
LogicalIfIndex | Index of the logical voice port for this call. |
LoWaterPlayoutDelay | Low water mark Voice Playout FIFO Delay during the voice call. |
NoiseLevel | Average noise level for this call. |
OnTimeRvPlayout | Duration of voice playout from data received on time for this call. You can derive the Total Voice Playout Duration for Active Voice by adding the OnTimeRvPlayout value to the GapFill values. |
PeerAddress | Destination pattern or number to which this call is connected. |
PeerId | ID value of the peer entry table to which this call was made. |
PeerIfIndex | Index number of the logical interface through which this call was made. For ISDN media, this would be the index number of the B channel used for the call. |
PeerSubAddress | Subaddress to which this call is connected. |
ReceiveBytes | Number of bytes received by the peer during this call. |
ReceiveDelay | Average Playout FIFO Delay plus the decoder delay during the voice call. |
ReceivePackets | Number of packets received by this peer during the call. |
RemoteIPAddress | Remote system IP address for the call. |
RemoteUDPPort | Remote system UDP listener port to which voice packets for this call are transmitted. |
RoundTripDelay | Voice packet round trip delay between the local and remote system on the IP backbone for this call. |
SelectedQoS | Selected RSVP quality of service for the call. |
Session Protocol | Session protocol to be used for an Internet call between the local and remote router via the IP backbone. |
Session Target | Session target of the peer used for the call. |
SetUpTime | Value of the system UpTime when the call associated with this entry was started. |
TransmitBytes | Number of bytes transmitted by this peer during the call. |
TransmitPackets | Number of packets transmitted by this peer during the call. |
TxDuration | Duration of the transmit path open from this peer to the voice gateway for the call. |
VADEnable | Whether or not voice activation detection (VAD) was enabled for this call. |
VoiceTxDuration | Duration of voice transmitted from this peer to voice gateway for this call. You can derive the Voice Utilization Rate by dividing the VoiceTxDuration by the TxDuration value. |
show call active voice
show dial-peer voice
show num-exp
show voice port
This command shows the information related to Call Switching Module (CSM), which is that the CSM state machine is in for the call associated to that DSP channel, the start time of the call, the end time of the call, and the channel on the controller used by the call.
show csm voice [shelf/slot/port]
Shelf | This is currently 1. |
Slot | This is the slot the card is in. The values are 0 - 11. |
Channel number | This is the channel number within the card. The values are 0 - 191. |
Privileged EXEC
This command first appeared in Cisco IOS Release 12.0(4)XL.
Use the show csm voice privileged EXEC command to display the call connection information.
The following is sample output from the show csm voice command:
5800# show csm voice 1/8/19 shelf 1, slot 8, port 19 VDEV_INFO:slot 8, port 19 vdev_status(0x00000401):VDEV_STATUS_ACTIVE_CALL.VDEV_STATUS_HASLOCK. csm_state(0x00000406)=CSM_OC6_CONNECTED, csm_event_proc=0x60868B8C, current call thru PRI line invalid_event_count=0, wdt_timeout_count=0 watchdog timer is not activated wait_for_bchan:False pri_chnl=(T1 1/0/0:22), vdev_chnl=(s8, c19) start_chan_p=0, chan_p=62436D58, call_id=0x800D, bchan_num=22 The calling party phone number = The called party phone number = 7511 ring_no_answer=0, ic_failure=0, ic_complete=0 dial_failure=0, oc_failure=0, oc_complete=1 oc_busy=0, oc_no_dial_tone=0, oc_dial_timeout=0 remote_link_disc=0, busyout=0, modem_reset=0 call_duration_started=3d16h, call_duration_ended=00:00:00, total_call_duration=00:00:00
number | (Optional) Displays configuration for the dial peer identified by the argument number. Valid entries are any integers that identify a specific dial peer, from 1 to 32767. |
Privileged EXEC
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the show dial-peer voice privileged EXEC command to display the configuration for all VoIP and POTS dial peers configured for the router. To show configuration information for only one specific dial peer, use the argument number to identify that dial peer.
The following is sample output from the show dial-peer voice command for a POTS dial peer:
5800# show dial-peer voice 1
VoiceEncapPeer1
tag = 1, dest-pat = \Q+14085291000',
answer-address = \Q',
group = 0, Admin state is up, Operation state is down
Permission is Both,
type = pots, prefix = \Q',
session-target = \Q', voice-port =
Connect Time = 0, Charged Units = 0
Successful Calls = 0, Failed Calls = 0
Accepted Calls = 0, Refused Calls = 0
Last Disconnect Cause is ""
Last Disconnect Text is ""
Last Setup Time = 0
The following is sample output from the show dial-peer voice command for a VoIP dial peer:
5800# show dial-peer voice 10
VoiceOverIpPeer10
tag = 10, dest-pat = \Q',
incall-number = \Q+14087',
group = 0, Admin state is up, Operation state is down
Permission is Answer,
type = voip, session-target = \Q',
sess-proto = cisco, req-qos = bestEffort,
acc-qos = bestEffort,
fax-rate = voice, codec = g729r8,
Expect factor = 10,Icpif = 30, VAD = disabled, Poor QOV Trap = disabled,
Connect Time = 0, Charged Units = 0
Successful Calls = 0, Failed Calls = 0
Accepted Calls = 0, Refused Calls = 0
Last Disconnect Cause is ""
Last Disconnect Text is ""
Last Setup Time = 0
Table 5-4 explains the fields contained in both examples in alphabetical order.
| Field | Description |
|---|---|
AcceptedCalls | Number of calls from this peer accepted since system startup. |
acc-qos | Lowest acceptable quality of service configured for calls for this peer. |
Admin state | Administrative state of this peer. |
Charged Units | Total number of charging units applying to this peer since system startup. |
codec | Default voice coder rate of speech for this peer. |
Connect Time | Accumulated connect time to the peer since system startup for both incoming and outgoing calls. The unit of measure is in hundredths of seconds. |
dest-pat | Destination pattern (telephone number) for this peer. |
Expect factor | User-requested Expectation Factor of voice quality for calls via this peer. |
fax-rate | Fax transmission rate configured for this peer. |
Failed Calls | Number of failed call attempts to this peer since system startup. |
group | Group number associated with this peer. |
ICPIF | Configured Calculated Planning Impairment Factor (ICPIF) value for calls sent by a dial peer. |
incall-number | Full E.164 telephone number to be used to identify the dial peer. |
Last Disconnect Cause | Encoded network cause associated with the last call. This value will be updated whenever a call is started or cleared and depends on the interface type and session protocol being used on this interface. |
Last Disconnect Text | ASCII text describing the reason for the last call termination. |
Last Setup Time | Value of the System Up Time when the last call to this peer was started. |
Operation state | Operational state of this peer. |
Permission | Configured permission level for this peer. |
Poor QOV Trap | Whether Poor Quality of Voice trap messages have been enabled or disabled. |
Refused Calls | Number of calls from this peer refused since system startup. |
req-qos | Configured requested quality of service for calls for this dial peer. |
session-target | Session target of this peer. |
sess-proto | Session protocol to be used for Internet calls between local and remote router via the IP backbone. |
Successful Calls | Number of completed calls to this peer. |
tag | Unique dial peer ID number. |
VAD | Whether or not voice activation detection (VAD) is enabled for this dial peer. |
show call active voice
show call-history voice
show num-exp
show voice port
shelf/slot/port | Specifies the T1 or E1 controller. |
cas-group number | Specifies the CAS group number. |
D | Indicates the D channel associated with ISDN PRI. |
dial string | Specifies a particular destination pattern (telephone number). |
Privileged EXEC
This command first appeared in Cisco IOS Release 11.3(1)T.
Occasionally, an incoming call cannot be matched to a dial peer in the dial peer database. One reason this might occur is that the specified destination cannot be reached via the voice interface through which the incoming call came. Use the show dialplan incall number command as a troubleshooting method to resolve the call destination by pairing voice ports and telephone numbers together until there is a match.
The following example tests whether the telephone extension 57681 can be reached through voice port 1/0/0:D:
show dialplan incall 0:D number 57681
show dialplan number
dial-string | Specifies a particular destination pattern (telephone number). |
Privileged EXEC
This command first appeared in Cisco IOS Release 12.0(4)XL.
The following example displays the dial peer associated with the destination pattern of 54567:
show dialplan number 54567
show dialplan incall number
The show gateway command is used to display the current gateway status.
show gatewayThis command has no keywords or arguments.
no show gateway
Privileged EXEC (also called enable mode)
This command first appeared in Cisco IOS Release 11.3(6)NA2.
gte-AS5800-2#show gateway Gateway voip2@vm1lab is registered to Gatekeeper gk1.vm1lab
To display the gateway-type prefix table, use the show gatekeeper gw-type-prefix EXEC command.
show gatekeeper gw-type-prefixPrivileged EXEC (also called enable mode)
This command first appeared in the Cisco IOS Release 11.3 NA.
The following is sample output from the show gatekeeper gw-type-prefix command:
5800# show gatekeeper gw-type-prefix
(GATEWAYS-TYPE PREFIX TABLE ================================ Prefix: 3#* (Hopoff- gk408) Prefix: 4#* (Default gateway-technology) Static Configured Gateways: Prefix: 7#* (Hopoff gk408) Static Configured Gateways: 1.1.1.1:1720 2.2.2.2:1720
| Field | Description |
|---|---|
Prefix: | The tech-prefix defined with the gw-type-prefix command. |
(Hopoff gk408) | Calls specifying tech-prefix 3# or 7# will always be routed to zone gk408, regardless of the actual zone-prefix in the destination address. |
(Default gateway-technology) | The address associated with the technology prefix is a gateway used as the default for routing any addresses that are otherwise unresolveable. |
Static Configured Gateways: | Lists all IP addresses and port numbers of statically configured gateways. |
To show overall gatekeeper status that includes authorization and authentication status, zone status, and so on, use the show gatekeeper status EXEC command.
show gatekeeper statusThis command has no arguments or keywords.
Privileged EXEC (also called enable mode)
This command first appeared in Cisco IOS Release 11.3 NA.
The following is sample output from the show gatekeeper status command:
5800# show gatekeeper status Gatekeeper State: UP Zone Name: gk-px4.cisco.com Accounting: DISABLED Security: DISABLED
| Field | Description |
|---|---|
Gatekeeper State |
|
Zone Name | Zone name. |
Accounting | Authorization and accounting status. |
Security | Security status. |
To display the zone prefix table, use the show gatekeeper zone prefix EXEC command.
show gatekeeper zone prefixPrivileged EXEC (also called enable mode)
This command first appeared in Cisco IOS Release 11.3 NA.
The following is sample output from the show gatekeeper zone prefix command:
5800# show gatekeeper zone prefix
ZONE PREFIX TABLE ================= GK-NAME E164-PREFIX ------- ----------- gk.zone13 212....... gk.zone14 415.......
gk.zone14 408.......
| Field | Description |
|---|---|
GK-NAME | The gatekeeper name. |
E164-PREFIX | The E.164 prefix and a dot that acts as a wildcard for matching each remaining number in the telephone number. |
To show the number expansions configured, use the show num-exp privileged EXEC command.
show num-exp [dialed- number]
dialed-number | (Optional) Displays number expansion for the specified dialed number. |
Privileged EXEC
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the show num-exp privileged EXEC command to display all of the number expansions configured for this router. To display number expansion for only one number, specify that number by using the dialed-number argument.
The following is sample output from the show num-exp command:
5800# show num-exp Dest Digit Pattern = '0...' Translation = '+14085270...' Dest Digit Pattern = '1...' Translation = '+14085271...' Dest Digit Pattern = '3..' Translation = '+140852703..' Dest Digit Pattern = '4..' Translation = '+140852804..' Dest Digit Pattern = '5..' Translation = '+140852805..' Dest Digit Pattern = '6....' Translation = '+1408526....' Dest Digit Pattern = '7....' Translation = '+1408527....' Dest Digit Pattern = '8...' Translation = '+14085288...'
Table 5-5 explains the fields in the sample output.
| Field | Description |
|---|---|
Dest Digit Pattern | Index number identifying the destination telephone number digit pattern. |
Translation | Expanded destination telephone number digit pattern. |
show call active voice
show call history voice
show dial-peer voice
show voice port
shelf/slot/port | Specifies the T1 or E1 controller. |
D | Indicates the D channel associated with ISDN PRI. |
Privileged EXEC
This command first appeared in Cisco IOS Release 11.3(2)NA.
Use the show voice port privileged EXEC command to display configuration and voice interface card-specific information about a specific port.
The following is sample output from the show voice port command:
5800#show voice port 1/0/0:DISDN 1/0/0:DType of VoicePort is ISDNOperation State is DORMANTAdministrative State is UPNo Interface Down FailureDescription is ""Noise Regeneration is enabledNon Linear Processing is enabledMusic On Hold Threshold is Set to -38 dBmIn Gain is Set to 0 dBOut Attenuation is Set to 0 dBEcho Cancellation is enabledEcho Cancel Coverage is set to 16 msConnection Mode is normalConnection Number is not setInitial Time Out is set to 10 sInterdigit Time Out is set to 10 sRegion Tone is set for US
Table 5-6 explains the fields in the sample output.
| Field | Description |
|---|---|
Type of VoicePort | Indicates the voice port type. |
Operational State | Operational state of the voice port. |
Administrative State | Administrative state of the voice port. |
Clear Wait Duration Timing | Time of inactive seizure signal to declare call cleared. |
Currently Processing | Type of call currently being processed: none, voice, or fax. |
Operations State | Operation state of the port. |
Operation Type | Operation of the E&M signal: two-wire or four-wire. |
Noise Regeneration | Whether or not background noise should be played to fill silent gaps if VAD is activated. |
Non-Linear Processing | Whether or not Non-Linear Processing is enabled for this port. |
Music-On-Hold Threshold | Configured Music-On-Hold Threshold value for this interface. |
In Gain | Amount of gain inserted at the receiver side of the interface. |
Out Attenuation | Amount of attenuation inserted at the transmit side of the interface. |
Pulse Rate Timing | Pulse dialing rate in pulses per second (pps). |
Echo Cancellation | Whether or not echo cancellation is enabled for this port. |
Echo Cancel Coverage | Echo Cancel Coverage for this port. |
Connection Mode | Connection mode of the interface. |
Connection Number | Full E.164 telephone number used to establish a connection with the trunk or PLAR mode. |
Initial Time Out | Amount of time the system waits for an initial input digit from the caller. |
Interdigit Time Out | Amount of time the system waits for a subsequent input digit from the caller. |
Regional Tone | Configured regional tone for this interface. |
show call active voice
show call history voice
show dial-peer voice
show num-exp
To monitor DSP usage and states on VFCs, use the show vrm privileged EXEC command.
show vrm [vdevices [dial shelf slot number [0-13] [voice device number [1-96]]]| summary]
vdevices | Indicates DSPs. |
active_calls | Indicates active voice calls on the channel. |
dial shelf slot number | Slot number of the dial shelf. Valid number is 0 to 13. |
voice device number | DSP number. Valid number is 1 to 96. |
summary | List synopsis of voice feature card DSP mappings, capabilities, and resource states. |
all | Lists all active calls for VFC slots. |
Privileged EXEC
This command first appeared in Cisco IOS Release 11.0(4)XL.
Use the show vrm vdevice to display detailed information for a specific DSP or a brief summary display for all VFCs. The display provides information on the number of channels, channels per DSP, bitmap of DSPMs, version numbers, and so on. This information is useful in monitoring the current state of your VFCs.
The display for a specific DSP provides information on the codec that each channel is using, if active, or last used, if the channel is not currently transmitting cells. It also displays the state of the resource. In most cases, if there is an active call on that channel, the resource should be marked active. If the resource is marked as reset and/or bad, this may be an indication of a response loss for the VFC on a reset request. If this condition persists, you might experience a problem with the communication link between the router shelf and the VFC.
Use the show vrm active_calls to display active-only voice calls either for a specific VFC or all VFCs. Each active call occupies a block of information describing the call. This information provides basically the same information as the show vrm vdevice command.
The following is sample output from the show vrm vdevice command specifying dial shelf slot number and DSP number:
5800# show vrm vdevices 6 1 slot = 6 virtual voice dev (tag) = 1 channel id = 1 capabilities list map = 9FFF last/current codec loaded/used = None TDM timeslot = 0 Resource (vdev_common) status = 401 means :active others tot ingress data = 101 tot ingress control = 1194 tot ingress data drops = 0 tot ingress control drops = 0 tot egress data = 39722 tot egress control = 1209 tot egress data drops = 0 tot egress control drops = 0 slot = 6 virtual voice dev (tag) = 1 channel id = 2 capabilities list map = 9FFF last/current codec loaded/used = None TDM timeslot = 1 Resource (vdev_common) status = 401 means :active others tot ingress data = 21 tot ingress control = 1167 tot ingress data drops = 0 tot ingress control drops = 0 tot egress data = 19476 tot egress control = 1163 tot egress data drops = 0 tot egress control drops = 0
The following is sample output from the show vrm devices command specifying a summary list. In the Voice Device Mapping area, the C_Ac column indicates number of active calls for a specific DSP. If there are any non zero numbers under the C_Rst and/or C_Bad column, this indicates a reset request was sent but it was lost; possible faulty DSP.
5800# show vrm vdevices summary *********************************************************** ******summary of voice devices for all voice cards********* *********************************************************** slot = 6 major ver = 0 minor ver = 1 core type used = 2 number of modules = 16 number of voice devices (DSPs) = 96 chans per vdevice = 2 tot chans = 192 tot active calls = 178 module presense bit map = FFFF tdm mode = 1 num_of_tdm_timeslots = 384 auto recovery is on number of default voice file (core type images) = 2 file 0 maj ver = 0 min ver = 0 core_type = 1 trough size = 2880 slop value = 0 built-in codec bitmap = 0 loadable codec bitmap = 0 fax codec bitmap = 0 file 1 maj ver = 3 min ver = 1 core_type = 2 trough size = 2880 slop value = 1440 built-in codec bitmap = 40B loadable codec bitmap = BFC fax codec bitmap = 7E -------------------Voice Device Mapping------------------------ Logical Device (Tag) Module# DSP# C_Ac C_Busy C_Rst C_Bad --------------------------------------------------------------- 1 1 1 2 0 0 0 2 1 2 2 0 0 0 3 1 3 2 0 0 0 4 1 4 2 0 0 0 5 1 5 2 0 0 0 6 1 6 2 0 0 0 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ 7 2 1 2 0 0 0 8 2 2 2 0 0 0 9 2 3 2 0 0 0 10 2 4 1 0 0 0 11 2 5 2 0 0 0 12 2 6 1 0 0 0 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ <information deleted> +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ 91 16 1 2 0 0 0 92 16 2 2 0 0 0 93 16 3 1 0 0 0 94 16 4 2 0 0 0 95 16 5 2 0 0 0 96 16 6 2 0 0 0 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ Total active call channels = 178 Total busied out channels = 0 Total channels in reset = 0 Total bad channels = 0 Note :Channels could be in multiple states
The following is sample output from the show vrm active_calls command specifying dial shelf slot number:
5800# show vrm active_calls 6 slot = 6 virtual voice dev (tag) = 61 channel id = 2 capabilities list map = 9FFF last/current codec loaded/used = None TDM timeslot = 241 Resource (vdev_common) status = 401 means :active others tot ingress data = 24 tot ingress control = 1308 tot ingress data drops = 0 tot ingress control drops = 0 tot egress data = 22051 tot egress control = 1304 tot egress data drops = 0 tot egress control drops = 0 slot = 6 virtual voice dev (tag) = 40 channel id = 2 capabilities list map = 9FFF last/current codec loaded/used = None TDM timeslot = 157 Resource (vdev_common) status = 401 means :active others
This command has no arguments or keywords.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
When a dial peer is shut down, you cannot initiate calls to that peer. This command is applicable to both VoIP and POTS peers.
The following example changes the administrative state of voice telephony dial peer 10 to down:
configure terminal dial-peer voice 10 pots shutdown
This command has no arguments or keywords.
Disabled.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the snmp enable peer-trap poor qov command to generate poor quality of voice notifications for applicable calls associated with this dial peer. If you have an SNMP manager that will use SNMP messages when voice quality drops, you might want to enable this command. Otherwise, you should disable this command to reduce unnecessary network traffic.
This command is applicable only to VoIP peers.
The following example enables poor quality of voice notifications for calls associated with VoIP dial peer 10:
dial-peer voice 10 voip snmp enable peer-trap poor-qov
snmp-server enable trap voice poor-qov
snmp trap link-status
To enable the router to send SNMP traps, use the snmp-server enable traps global configuration command. Use the no form of this command to disable SNMP traps.
snmp-server enable traps [trap-type] [trap-option]
trap-type | (Optional) Type of trap to enable. If no type is specified, all traps are sent (including the envmon and repeater traps). The trap type can be one of the following keywords: · bgp---Sends Border Gateway Protocol (BGP) state change traps. · config---Sends configuration traps. · entity---Sends Entity MIB modification traps. · envmon---Sends Cisco enterprise-specific environmental monitor traps when an environmental threshold is exceeded. When the envmon keyword is used, you can specify a trap-option value. · frame-relay---Sends Frame Relay traps. · isdn---Sends Integrated Services Digital Network (ISDN) traps. When the isdn keyword is used on Cisco 1600 series routers, you can specify a trap-option value. · repeater---Sends Ethernet hub repeater traps. When the repeater keyword is selected, you can specify a trap-option value. · rtr---Sends response time reporter (RTR) traps. · snmp---Sends Simple Network Management Protocol (SNMP) traps. When the snmp keyword is used, you can specify a trap-option value. · syslog---Sends error message traps (Cisco Syslog MIB). Specify the level of messages to be sent with the logging history level command. · voice---Sends SNMP poor quality of voice traps, when used with the qov trap-option. |
trap-option | (Optional) When the envmon keyword is used, you can enable a specific environmental trap type, or accept all trap types from the environmental monitor system. If no option is specified, all environmental types are enabled. The option can be one or more of the following keywords: voltage, shutdown, supply, fan, and temperature. When the isdn keyword is used on Cisco 1600 series routers, you can specify the call-information keyword to enable an SNMP ISDN call information trap for the ISDN MIB subsystem, or you can specify the isdnu-interface keyword to enable an SNMP ISDN U interface trap for the ISDN U interface MIB subsystem. When the repeater keyword is used, you can specify the repeater option. If no option is specified, all repeater types are enabled. The option can be one or more of the following keywords: · health---Enables IETF Repeater Hub MIB (RFC 1516) health trap. · reset---Enables IETF Repeater Hub MIB (RFC 1516) reset trap. When the snmp keyword is used, you can specify the authentication option to enable SNMP Authentication Failure traps. (The snmp-server enable traps snmp authentication command replaces the snmp-server trap-authentication command.) If no option is specified, all SNMP traps are enabled. When the voice keyword is used, you can enable SNMP poor quality of voice traps by using the qov option. |
This command is disabled by default. No traps are enabled.
If you enter this command with no keywords, the default is to enable all trap types.
Some trap types cannot be controlled with this command. These traps are either always enabled or enabled by some other means. For example, the linkUpDown messages are disabled by the no snmp trap link-status command.
Global configuration
This command first appeared in Cisco IOS Release 11.1.
This command is useful for disabling traps that are generating a large amount of uninteresting or useless noise.
If you do not enter an snmp-server enable traps command, no traps controlled by this command are sent. In order to configure the router to send these SNMP traps, you must enter at least one snmp-server enable traps command. If you enter the command with no keywords, all trap types are enabled. If you enter the command with a keyword, only the trap type related to that keyword is enabled. To enable multiple types of traps, you must issue a separate snmp-server enable traps command for each trap type and option.
The snmp-server enable traps command is used in conjunction with the snmp-server host command. Use the snmp-server host command to specify which host or hosts receive SNMP traps. To send traps, you must configure at least one snmp-server host command.
For a host to receive a trap controlled by this command, both the snmp-server enable traps command and the snmp-server host command for that host must be enabled. If the trap type is not controlled by this command, only the appropriate snmp-server host command must be enabled.
The trap types used in this command all have an associated MIB object that allows them to be globally enabled or disabled. Not all of the trap types available in the snmp-server host command have notificationEnable MIB objects, so some of these cannot be controlled using the snmp-server enable traps command.
The following example enables the router to send SNMP poor quality of voice traps:
configure terminal
snmp-server enable trap voice poor-qov
The following example enables the router to send all traps to the host, "myhost.cisco.com," using the community string, "public":
snmp-server enable traps snmp-server host myhost.cisco.com public
The following example enables the router to send Frame Relay and environmental monitor traps to the host, "myhost.cisco.com," using the community string, "public":
snmp-server enable traps frame-relay snmp-server enable traps envmon temperature snmp-server host myhost.cisco.com public
The following example will not send traps to any host. The BGP traps are enabled for all hosts, but the only traps enabled to be sent to a host are ISDN traps.
snmp-server enable traps bgp snmp-server host bob public isdn
snmp enable peer-trap peer-qov
snmp trap link-status
This command contains no arguments or keywords.
Enabled.
Voice-port configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the snmp trap link-status command to enable SNMP trap messages (linkup and linkdown) to be generated whenever this voice port is brought online or offline.
If you are managing the equipment with an SNMP manager, this command should be enabled. Enabling link-status messages will allow the SNMP manager to learn of a status change without polling the equipment. If you are not using an SNMP manager, this command should be disabled to avoid unnecessary network traffic.
The following example enables SNMP trap messages for voice-port 2/1/0:
voice-port 2/1/0 snmp trap link-stat
smnp enable peer-trap poor-qov
snmp-server enable traps
number | Defines the numbers used as the technology prefix. Each technology prefix can contain up to 11 characters. Although not strictly necessary, a pound (#) symbol is frequently used as the last digit in a technology prefix. Valid characters 0 though 9, the pound (#) symbol, and the asterisk (*). |
No technology prefix is defined.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(6)NA2.
Technology prefixes are used to distinguish between gateways having specific capabilities within a given zone. In the exchange between the gateway and the gatekeeper, the technology prefix is used to select a gateway after the zone has been selected. Use the tech-prefix command to define technology prefixes.
Technology prefixes can be used as a discriminator so that the gateway can tell the gatekeeper that a certain technology is associated with a particular call (for example, 15# could mean a fax transmission), or it can be used like an area code for more generic routing. No standard defines what the numbers in a technology prefix mean; by convention, technology prefixes are designated by a pound (#) symbol as the last character.
In most cases, there is a dynamic protocol exchange between the gateway and the gatekeeper that enables the gateway to inform the gatekeeper about technology prefixes and where to forward calls. If, for some reason, that dynamic registry feature is not in effect, you can statically configure the gatekeeper to query the gateway for this information by configuring the gw-type-prefix command on the gatekeeper. Use the show gatekeeper gw-type-prefix to display how the gatekeeper has mapped the technology prefixes to local gateways.
The following example defines a technology prefix of 14# for the specified dial peer. In this example, the technology prefix means that the H.323 gateway will ask the RAS gatekeeper to direct calls using the technology prefix of 14#.
dial-peer voice 10 voip destination-pattern 14... tech-prefix 14#
gw-type-prefix
show gatekeeper gw-type-prefix
This command has no arguments or keywords.
Enabled.
Dial peer configuration
This command first appeared in Cisco IOS Release 11.3(1)T.
Use the vad command to enable voice activity detection. With VAD, silence is not transmitted over the network, only audible speech. If you enable VAD, the sound quality will be slightly degraded, but the connection will monopolize much less bandwidth. If you use the no form of this command, VAD is disabled and voice data is continuously transmitted to the IP backbone.
This command is applicable only to VoIP peers.
The following example enables VAD:
dial-peer voice 10 voip vad
none
To apply a CODEC preference list to a specific dial peer, use the voice-class codec dial-peer configuration command.
voice-class codecThis command has no arguments or keywords.
No default behavior or values.
Dial-peer configuration
This command first appeared in Cisco IOS Release 12.0(4)XL.
To specify a zone controlled by a gatekeeper, use the zone local gatekeeper configuration command. To remove a zone controlled by a gatekeeper, use the no form of this command. This command can also be used to change the IP address used by the gatekeeper.
zone local gatekeeper-name domain-name [rasIPaddress]
gatekeeper-name
| The gatekeeper's name or zone name. This is usually the fully domain-qualified host name of the gatekeeper. For example, if the domain-name is cisco.com, the gatekeeper-name might be gk1.cisco.com. However, if the gatekeeper is controlling multiple zones, the gatekeeper-name for each zone should be some unique string that has a mnemonic value. |
domain-name | The domain name served by this gatekeeper. |
rasIPaddress | The IP address of one of the interfaces on the gatekeeper. When the gatekeeper responds to gatekeeper discovery messages, it signals the endpoint or gateway to use this address in future communications. Setting this address for one local zone makes it the address used for all local zones. |
No local zone is defined.
Gatekeeper configuration
This command first appeared in Cisco IOS Release 11.3 NA.
Multiple local zones can be defined. The gatekeeper manages all configured local zones. Intrazone and interzone behavior remains the same (zones are controlled by the same or different gatekeepers.)
Only one rasIPaddress argument can be defined for all local zones. You cannot configure each zone to use a different RAS IP address. If you define this in the first zone definition, you can omit it for all subsequent zones, which automatically pick up this address. If you set it in a subsequent zone local command, it also changes the RAS address of all previously configured local zones. After it is defined, you can change it by re-issuing any zone local command with a different rasIPaddress argument.
If the rasIPaddress argument is an HSRP virtual address, it automatically puts the gatekeeper into HSRP mode. In this mode, the gatekeeper assumes STANDBY or ACTIVE status according to whether the HSRP interface is on STANDBY or ACTIVE status.
You cannot remove a local zone if there are endpoints or gateways registered in it. To remove the local zone, shut down the gatekeeper first, which forces unregistration.
Multiple zones are controlled by multiple logical gatekeepers on the same Cisco IOS release.
The following example creates a zone controlled by a gatekeeper in the domain called cisco.com:
zone local gk1.cisco.com cisco.com show gatekeeper zone statue
zone remote
To configure the gatekeeper with knowledge of its own and any remote zone's prefixes, use the zone prefix gatekeeper configuration command. To remove knowledge of zone prefixes, use the no form of this command.
zone prefix gatekeeper-name e164-prefix
gatekeeper-name
| The name of a local or remote gatekeeper, which must have been defined using the zone local or zone remote command. |
e164-prefix | An E.164 prefix in standard form followed by dots (.) that each represent a For example, 212....... is matched by 212 and any seven numbers. |
No knowledge of its own or any other zone's prefix is defined.
Gatekeeper configuration
Although a dot representing each digit in an E.164 address is the preferred configuration method, you may also enter an asterisk (*) to match any number of digits.
A gatekeeper may handle more than one zone prefix, but a zone prefix cannot be shared by more than one gatekeeper. If you have defined a zone prefix as being handled by a gatekeeper, and now define it as being handled by a second gatekeeper, the second assignment will cancel the first.
When a zone handles several prefixes, all gateways in that zone constitute a common pool which can be used to hop off to any of those prefixes. You may however wish to partition your gateways by prefix, for instance you have a gateway which interfaces to the 408 area code, and another which interfaces to the 415 area code, and for cost reasons you want each gateway only to be used for calls to its area code. In that case, you can define several local zones on the gatekeeper, each responsible for a prefix, and have each gateway register to the zone handling its prefix. For example, you can define local zone gk-408 handling prefix 408....... and local zone gk-415 handling 415....... and have the gateway interfacing to the 408 area code register with gk-408, and the gateway with the 415 interface register to gk-415.
zone local
zone remote
The following example matches the 212 area code and any seven digits as the zone prefix for gk-ny:
zone prefix gk-ny 212.......
To statically specify a remote zone if DNS is unavailable or undesirable, use the zone remote gatekeeper configuration command. To remove the remote zone, use the no form of this command.
zone remote other-gatekeeper-name other-domain-name other-gatekeeper-ip-address
other-gatekeeper-name | Name of the remote gatekeeper. |
other-domain-name | Domain name of the remote gatekeeper. |
other-gatekeeper-ip-address | IP address of the remote gatekeeper. |
port-number | (Optional) RAS signaling port number for the remote zone. Value ranges from 1 to 65535. If this is not set, the default is the well-known RAS port number 1719. |
No remote zone is defined. DNS will locate the remote zone.
Gatekeeper configuration
This command first appeared in Cisco IOS Release 11.3 NA.
All gatekeepers do not have to be in DNS. For those that are not, use the zone remote command so that the local gatekeeper knows how to access them. In addition, you may wish to improve call response time slightly for frequently accessed zones. If the zone remote command is configured for a particular zone, you do not need to make a DNS lookup transaction.
The following example configures the local gatekeeper to reach targets of the form xxx.cisco.com by sending queries to the gatekeeper named sj3.cisco.com at IP address 1.2.3.4:
zone remote sj3.cisco.com cisco.com 1.2.3.4 show gatekeeper zone statue
zone local
Posted: Wed Nov 3 16:59:32 PST 1999
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