Table Of Contents
Command Reference: N through Z
name (voice emergency response location)
name (voice hunt-group)
number (voice register pool)
num-buttons
outbound-proxy system
overlap-signal
pattern direct (vm-integration)
pattern ext-to-ext busy (vm-integration)
pattern ext-to-ext no-answer (vm-integration)
pattern trunk-to-ext busy (vm-integration)
pattern trunk-to-ext no-answer (vm-integration)
phoneload
pickup (call-manager-fallback)
preference (voice register pool)
proxy (voice register pool)
registrar server (SIP)
reset (call-manager-fallback)
secondary-dialtone (call-manager-fallback)
security
security-policy (voice register global)
show call-manager-fallback all
show call-manager-fallback dial-peer
show call-manager-fallback ephone-dn
show call-manager-fallback voice-port
show credentials
show ephone
show ephone cfa
show ephone dn
show ephone dnd
show ephone login
show ephone moh
show ephone offhook
show ephone overlay
show ephone phone-load
show ephone registered
show ephone remote
show ephone ringing
show ephone summary
show ephone tapiclients
show ephone telephone-number
show ephone unregistered
show ephone-dn
show ephone-dn callback
show ephone-dn loopback
show ephone-dn statistics
show ephone-dn summary
show sip-ua status registrar
show voice emergency
show voice emergency addresses
show voice emergency all
show voice emergency callers
show voice emergency zone
show voice moh-group
show voice moh-group statistics
show voice register all
show voice register dial-peers
show voice register dn
show voice register global
show voice register pool
show voice register pool after-hour-exempt
show voice register pool attempted-registrations
show voice register pool connected
show voice register pool ip
show voice register pool mac
show voice register pool network
show voice register pool on-hold
show voice register pool registered
show voice register pool remote
show voice register pool ringing
show voice register pool telephone-number
show voice register pool unregistered
show voice register statistics
subnet
system message (call-manager-fallback)
system message (voice register global)
time-format (call-manager-fallback)
timeouts busy (call-manager-fallback)
timeouts interdigit (call-manager-fallback)
timeouts ringing (call-manager-fallback)
transfer-digit-collect
transfer-pattern
transfer-pattern blocked (voice register pool)
transfer max-length (voice register pool)
transfer-system (call-manager-fallback)
translate (call-manager-fallback)
translate-outgoing (voice register pool)
translation-profile (call-manager-fallback)
translation-profile (voice register)
trustpoint (credentials)
user-locale (call-manager-fallback)
utf8
vad (voice register pool)
video (call-manager-fallback)
vm-integration
voice-class codec (voice register pool)
voice emergency response location
voice emergency response settings
voice emergency response zone
voice hunt-group
voice moh-group
voice register global
voice register pool
voicemail (call-manager-fallback)
voice moh-group
xmlschema (call-manager-fallback)
Command Reference: N through Z
Last Updated: March 15, 2015
This chapter contains commands to configure and maintain Cisco Unified Survivable Remote Site Telephony (SRST) and Cisco Unified SIP SRST. The commands are presented in alphabetical order. Some commands required for configuring Cisco Unified SRST and Cisco Unified SIP SRST may be found in other Cisco IOS command references. Use the command reference master index or search online to find these commands.
name (voice emergency response location)
To describe or identify an emergency response location, use the name command in voice emergency response location mode. To remove this definition, use the no form of this command.
name string
no name
Syntax Description
string
|
String (30 characters) used to describe or identify an ERL's location.
|
Command Default
The location is not described.
Command Modes
Voice emergency response location configuration (cfg-emrgncy-resp-location)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.4(15)XY
|
Cisco Unified CME 4.2(1) Cisco Unified SRST 4.2(1) Cisco Unified SIP SRST 4.2(1)
|
This command was introduced.
|
12.4(20)T
|
Cisco Unified CME 7.0 Cisco Unified SRST 7.0 Cisco Unified SIP SRST 7.0
|
This command was integrated into Cisco IOS Release 12.4(20)T.
|
Usage Guidelines
Use this command to enable a word or description of the ERL for administrative purposes. Most commonly, use this command to identify the location for the network administrator.
Examples
In this example, the location description is Widget Incorporated.
voice emergency response location 60
subnet 1 209.165.200.224 255.255.0.0
name Widget Incorporated,
Related Commands
Command
|
Description
|
address
|
Specifies a comma separated text entry (up to 247 characters) of an ERL's civic address.
|
elin
|
Specifies a PSTN number that will replace the caller's extension.
|
subnet
|
Defines which IP phones are part of this ERL.
|
voice emergency response location
|
Creates a tag for identifying an ERL for E911 services.
|
name (voice hunt-group)
To associate a name with a called voice hunt group, use the name command in voice hunt-group configuration mode. To dissociate the name of the called voice hunt group, use the no form of this command.
name "primary pilot name" [secondary "secondary pilot name"]
no name "primary pilot name" [secondary "secondary pilot name"]
Syntax Description
"primary pilot name"
|
Name of primary pilot number.
|
secondary "secondary pilot name"
|
(Optional) Name of secondary pilot number.
|
Command Default
No name is associated with the called voice hunt group.
Command Modes
Voice hunt-group configuration (config-voice-hunt-group)
Command History
Release
|
Modification
|
15.3(2)T
|
This command was introduced.
|
Usage Guidelines
In Cisco Unified SRST 9.5, when the secondary pilot name is not explicitly configured, the primary pilot name is applicable to both pilot numbers.
Note
Use quotes (") when input strings have spaces in between.
Examples
The following example configures the primary pilot name for both the primary and the secondary pilot numbers:
The following example configures different names for the primary and secondary pilot numbers:
name SALES secondary SALES-SECONDARY
The following example associates a two-word name for the primary pilot number and a one-word name for the secondary pilot number:
name "CUSTOMER SERVICE" secondary CS
The following example associates a one-word name for the primary pilot number and a two-word name for the secondary pilot number:
name FINANCE secondary "INTERNAL ACCOUNTING"
The following example associates two-word names for the primary and secondary pilot numbers:
name "INTERNAL CALLER" secondary "EXTERNAL CALLER"
When incoming call A reaches voice hunt group B and lands on final C, extension C does not show the name of the forwarder because the voice hunt group is not configured to display the name. To display the name of the forwarder and the final number, two separate names are required for the primary and secondary pilots.
The following example shows how the primary and secondary pilot names are configured in voice hunt-group configuration mode:
voice hunt-group 24 parallel
name SALES secondary SALES-SECONDARY
The following is a sample output of the show voice hunt-group command when the primary and secondary pilot names are configured in voice hunt-group configuration mode:
pilot number: 1000, peer-tag 2147483647
secondary number: 2000, peer-tag 2147483646
secondary name: SALES-SECONDARY
list of numbers: 2004,2005
Related Commands
Command
|
Description
|
voice hunt-group
|
Enters voice hunt-group configuration mode and creates a hunt group for phones in a Cisco Unified CME system.
|
show voice hunt-group
|
Displays configuration information associated with one or all voice hunt groups in a Cisco Unified CME system.
|
number (voice register pool)
To indicate the E.164 phone numbers that the registrar permits to handle the Register message from a Cisco Unified SIP IP phone, use the number command in voice register pool configuration mode. To disable number registration, use the no form of this command.
number tag {number-pattern [preference value] [huntstop] | dn dn-tag}
no number tag
Syntax Description
tag
|
Telephone number when there are multiple number commands. Range is 1 to 114.
|
number-pattern
|
Phone numbers (including wild cards and patterns) that are permitted by the registrar to handle the Register message from the Cisco Unified SIP IP phone.
|
preference value
|
(Optional) Defines the number list preference order. Range is 0 to 10. The highest preference is 0. There is no default.
|
huntstop
|
(Optional) Stops hunting when the dial peer is busy.
|
dn dn-tag
|
Identifies the directory number tag for this phone number as defined by the voice register dn command. Range is 1 to 288.
|
Command Default
Cisco Unified SIP IP phones cannot register in Cisco Unified CME.
Command Modes
Voice register pool configuration (config-register-pool)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco SIP SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco SIP SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
12.4(4)T
|
Cisco CME 3.4 Cisco SIP SRST 3.4
|
This command was added to Cisco CME and the dn keyword was added.
|
12.4(11)XJ
|
Cisco Unified CME 4.1 Cisco Unified SRST 4.1
|
This command was modified. The number-pattern argument and preference and huntstop keywords were removed from Cisco Unified CME.
|
12.4(15)T
|
Cisco Unified CME 4.1 Cisco Unified SRST 4.1
|
The modifications to this command were integrated into Cisco IOS Release 12.4(15)T.
|
15.2(4)M
|
Cisco Unified CME 9.1 Cisco Unified SIP SRST 9.1
|
This command was modified to increase the valid value of the tag argument to 114.
|
Usage Guidelines
The number command indicates the phone numbers that are permitted by the registrar to handle the Register message from the Cisco Unified SIP IP phone.
In Cisco Unified SRST, the keywords and arguments of this command allow for more explicit setting of user preferences regarding what number patterns should match the voice register pool.
Note
Configure the id (voice register pool) command before any other voice register pool commands, including the number command. The id command identifies a locally available, individual Cisco Unified SIP IP phone or a set of Cisco Unified SIP IP phones.
Examples
The following example shows three directory numbers assigned to Cisco Unified SIP IP phone 1 in Cisco Unified CME:
!
voice register pool 1
id mac 0017.E033.0284
type 7961
number 1 dn 10
number 2 dn 12
number 3 dn 13
codec g711ulaw
!
The following example shows directory numbers 10, 12, and 13 assigned to phone numbers 1, 2, and 55 of Cisco Unified SIP IP phone 2:
voice register pool 2
id mac 0017.E033.0284
type 7961
number 1 dn 10
number 2 dn 12
number 55 dn 13
codec g711ulaw
The following example shows a telephone number pattern set to 95... in Cisco Unified SRST. This means all five-digit numbers beginning with 95 are permitted by the registrar to handle the Register message.
voice register pool 3
id network 10.2.161.0 mask 255.255.255.0
number 1 95... preference 1
cor incoming call95 1 95011
Related Commands
Command
|
Description
|
id (voice register pool)
|
Explicitly identifies a locally available, individual Cisco Unified SIP IP phone or, when running Cisco Unified SIP SRST, a set of Cisco Unified SIP IP phones.
|
voice register dn
|
Enters voice register dn configuration mode to define an extension for a phone line, intercom line, voice-mail port, or a message-waiting indicator.
|
num-buttons
To set the number of line buttons supported by a phone type, use the num-buttons command in ephone-type configuration mode. To reset to the default, use the no form of this command.
num-buttons number
no num-buttons
Syntax Description
number
|
Number of line buttons. Range: 1 to 100. Default: 0. See Table 5 for the number of buttons supported by each phone type.
|
Command Default
No line buttons are supported by the phone type.
Command Modes
Ephone-type configuration (config-ephone-type)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.4(15)XZ
|
Cisco Unified CME 4.3 Cisco Unified SRST 4.3
|
This command was introduced.
|
12.4(20)T
|
Cisco Unified CME 7.0 Cisco Unified SRST 7.0
|
This command was integrated into Cisco IOS Release 12.4(20)T.
|
Usage Guidelines
This command defines the number of line buttons supported by the type of phone being added with an ephone-type template.
Table 5 Supported Values for Ephone-Type Commands
Supported Device
|
device-id
|
num-buttons
|
max-presentation
|
Cisco Unified IP Conference Station 7937G
|
431
|
1
|
6
|
Nokia E61
|
376
|
1
|
1
|
Examples
The following example shows that 1 line button is specified for the Nokia E61 when creating the ephone-type template.
Router(config)# ephone-type E61
Router(config-ephone-type)# num-buttons 1
Related Commands
Command
|
Description
|
device-id
|
Specifies the device ID for a phone type.
|
max-presentation
|
Sets the number of call presentation lines supported by a phone type.
|
type
|
Assigns the phone type to an SCCP phone.
|
outbound-proxy system
To specify that all Cisco Unified Communications Manager Express (Cisco Unified CME) line-side phones connected to a Cisco IOS voice gateway use the global settings for forwarding Session Initiation Protocol (SIP) messages to an outbound proxy, use the outbound-proxy system command in voice register global configuration mode. To disable the SIP outbound proxy function for Cisco Unified CME line-side SIP phones, use the no form of this command.
outbound-proxy system
no outbound-proxy system
Syntax Description
This command has no arguments or keywords.
Command Default
The SIP outbound proxy function for all SIP line-side phones in Cisco Unified CME is enabled and behavior is determined by the global setting on the Cisco IOS gateway.
Command Modes
Voice register global configuration (config-register-global)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.4(15)XZ
|
Cisco Unified CME 4.3 Cisco Unified SIP SRST 4.3
|
This command was introduced.
|
12.4(20)T
|
Cisco Unified CME 7.0 Cisco Unified SIP SRST 7.0
|
This command was integrated into Cisco IOS Release 12.4(20)T.
|
Usage Guidelines
If global configuration for outbound proxy is enabled on the Cisco IOS voice gateway and Cisco Unified CME receives a call, Cisco Unified CME forwards all SIP messages to the outbound proxy causing incoming calls to line-side SIP phones to fail. This is the default behavior.
To avoid these failed calls, use the no form of this command in voice register global configuration mode to override global outbound proxy settings for the gateway and disable the outbound proxy function for all line-side SIP phones connected to Cisco Unified CME.
To configure outbound proxy settings for an individual dial peer on the gateway, use the voice-class sip outbound-proxy command in dial peer voice configuration mode .
Examples
The following example shows how to disable the global outbound proxy feature for all line-side SIP phones on a Cisco Unified CME:
Router# configure terminal
Router(config)# voice register global
Router(config-register-global)# no outbound-proxy
Related Commands
Command
|
Description
|
voice-class sip outbound-proxy
|
Configures SIP outbound proxy settings for an individual dial peer that override global settings for the Cisco IOS voice gateway.
|
overlap-signal
To configure overlap dialing in SCCP or SIP IP phones, use the overlap-signal command in ephone, ephone-template, telephony-service, voice register pool, voice register global, or voice register template configuration mode.
overlap-signal
Syntax Description
This command has no arguments or keywords.
Command Default
Overlap-signal is disabled.
Command Modes
Call-manager-fallback
Ephone configuration (config-ephone)
Ephone-template configuration (config-ephone-template)
Telephony-service configuration (config-telephony)
Voice register pool (config-register-pool)
Voice register global configuration (config-register-global)
Voice register template (config-register-template)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
15.1(3)T
|
Cisco Unified CME 8.5 Cisco Unified SRST 8.5
|
This command was introduced.
|
Usage Guidelines
SCCP IP phones
In SCCP IP phones, overlap dialing is enabled when the overlap signal command is configured in ephone, ephone-template, and telephony-service configurations modes.
SIP IP phones
In SIP IP Phones, overlap dialing is enabled when the overlap signal command is configured in voice register pool, voice register global, and voice register template configuration modes.
Cisco Unified SRST
In Cisco Unified SRST, overlap dialing is enabled on SCCP IP phones when overlap signal command is configured in call-manager-fallback configuration mode.
Examples
The following example shows overlap-signal enabled on SCCP phones:
Router# show running config
!
!
telephony-service
max-ephones 25
max-dn 15
load 7906 SCCP11.8-5-3S.loads
load 7911 SCCP11.8-5-3S.loads
load 7921 CP7921G-1.3.3.LOADS
load 7941 SCCP41.8-5-3S.loads
load 7942 SCCP42.8-5-3S.loads
load 7961 SCCP41.8-5-3S.loads
load 7962 SCCP42.8-5-3S.loads
max-conferences 12 gain -6
web admin system name cisco password cisco
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
overlap-signal
!
ephone-template 1
button-layout 1 line
button-layout 3-6 blf-speed-dial
!
ephone-template 9
feature-button 1 Endcall
feature-button 3 Mobility
!
!
ephone-template 10
feature-button 1 Park
feature-button 2 MeetMe
feature-button 3 CallBack
button-layout 1 line
button-layout 2-4 speed-dial
button-layout 5-6 blf-speed-dial
overlap-signal
!
ephone 10
device-security-mode none
mac-address 02EA.EAEA.0010
overlap-signal
!
The following example shows overlap-signal configured in voice register global and voice register pool 10:
Router#show running config
!
!
!
voice service voip
ip address trusted list
ipv4 20.20.20.1
media flow-around
allow-connections sip to sip
!
voice class media 10
media flow-around
!
!
voice register global
max-pool 10
overlap-signal
!
voice register pool 5
overlap-signal
!
!
!
The following example shows overlap-signal configured in call-manager-fallback mode:
Router# show run | sec call-manager
call-manager-fallback
max-conferences 12 gain -6
transfer-system full-consult
overlap-signal
pattern direct (vm-integration)
To configure the dual-tone multifrequency (DTMF) digit pattern forwarding necessary to activate the voice-mail system when the user presses the messages button on the phone, use the pattern direct command in voice-mail integration configuration mode. To disable DTMF digit pattern forwarding when the user presses the messages button on the phone, use the no form of this command.
pattern direct tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
no pattern direct tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
Syntax Description
tag1
|
Alphanumeric string fewer than four DTMF digits in length. The alphanumeric string consists of a combination of four letters (A,B,C, and D), two symbols (* and #), and ten digits (0 to 9). The tag numbers match the numbers defined in the voice-mail system's integration file, immediately preceding either the number of the calling party, the number of the called party, or a forwarding number.
|
tag2 and tag3
|
(Optional) See tag1.
|
last-tag
|
(Optional) See tag1. This tag indicates the end of the pattern.
|
CGN
|
Calling number (CGN) information is sent to the voice-mail system.
|
CDN
|
Called number (CDN) information is sent to the voice-mail system.
|
FDN
|
Forwarding number (FDN) information is sent to the voice-mail system.
|
Defaults
This feature is disabled.
Command Modes
Voice-mail integration configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(2)XT
|
Cisco SRST 2.0
|
This command was introduced on the following platforms: Cisco 1750, Cisco 1751, Cisco 2600 series and Cisco 3600 series multiservice routers, and Cisco IAD2420 series IADs.
|
12.2(8)T
|
Cisco SRST 2.0
|
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725 and Cisco 3745 routers.
|
12.2(8)T1
|
Cisco SRST 2.0
|
This command was implemented on the Cisco 2600-XM and Cisco 2691 routers.
|
12.2(11)T
|
Cisco SRST 2.01
|
This command was integrated into Cisco IOS Release 12.2(11)T and implemented on the Cisco 1760 routers.
|
12.2(13)T
|
Cisco SRST 2.02
|
This command was introduced for Cisco SRST Version 2.02.
|
Usage Guidelines
The pattern direct command is used to configure the sequence of DTMF digits passed to a voice-mail system attached to the router through one or more voice ports. When a call is placed directly from a Cisco IP phone attached to the router, the voice-mail system expects to receive a sequence of DTMF digits at the beginning of the call that identify the mailbox of the user calling the voice-mail system accompanied by a string of digits indicating that the caller is attempting to access the designated mailbox in order to retrieve messages.
Note
Although it is unlikely that you will use multiple instances of the CGN, CDN, or FDN keyword in a single command line, it is permissible to do so.
Examples
The following example sets the DTMF pattern for a calling number (CGN) for a direct call to the voice-mail system:
Router(config)# vm-integration
Router(config-vm-integration)# pattern direct 2 CGN
Related Commands
Command
|
Description
|
pattern ext-to-ext busy (vm-integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system once an internal extension attempts to connect to a busy extension and the call is forwarded to voice mail
|
pattern ext-to-ext no-answer (vm- integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system once an internal extension fails to connect to an extension and the call is forwarded to voice mail.
|
pattern trunk-to-ext busy (vm-integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system once an external trunk call reaches a busy extension and the call is forwarded to voice mail.
|
pattern trunk-to-ext no-answer (vm- integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system when an external trunk call reaches an unanswered extension and the call is forwarded to voice mail.
|
vm-integration
|
Enters voice-mail integration configuration mode and enables voice-mail integration with DTMF and analog voice-mail systems.
|
pattern ext-to-ext busy (vm-integration)
To configure the dual-tone multifrequency (DTMF) digit pattern forwarding necessary to activate the voice-mail system once an internal extension attempts to connect to a busy extension and the call is forwarded to voice mail, use the pattern ext-to-ext busy command in voice-mail integration configuration mode. To disable DTMF digit pattern forwarding when an internal extension calls a busy extension and the call is forwarded to a voice-mail system, use the no form of this command.
pattern ext-to-ext busy tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
no pattern ext-to-ext busy tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
Syntax Description
tag1
|
Alphanumeric string fewer than four DTMF digits in length. The alphanumeric string consists of a combination of four letters (A,B,C, and D), two symbols (* and #), and ten digits (0 to 9). The tag numbers match the numbers defined in the voice-mail system's integration file, immediately preceding either the number of the calling party, the number of the called party, or a forwarding number.
|
tag2 and tag3
|
(Optional) See tag1.
|
last-tag
|
(Optional) See tag1. This tag indicates the end of the pattern.
|
CGN
|
Calling number (CGN) information is sent to the voice-mail system.
|
CDN
|
Called number (CDN) information is sent to the voice-mail system.
|
FDN
|
Forwarding number (FDN) information is sent to the voice-mail system.
|
Defaults
This feature is disabled.
Command Modes
Voice-mail integration configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(2)XT
|
Cisco SRST 2.0
|
This command was introduced on the following platforms: Cisco 1750, Cisco 1751, Cisco 2600 series and Cisco 3600 series multiservice routers, and Cisco IAD2420 series IADs.
|
12.2(8)T
|
Cisco SRST 2.0
|
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725 and Cisco 3745 routers.
|
12.2(8)T1
|
Cisco SRST 2.0
|
This command was implemented on the Cisco 2600-XM and Cisco 2691 routers.
|
12.2(11)T
|
Cisco SRST 2.01
|
This command was integrated into Cisco IOS Release 12.2(11)T and implemented on the Cisco 1760 routers.
|
12.2(13)T
|
Cisco SRST 2.02
|
This command was introduced for Cisco SRST Version 2.02.
|
Usage Guidelines
The pattern ext-to-ext busy command is used to configure the sequence of DTMF digits passed to a voice-mail system attached to the router through one or more voice ports. When a call is routed to the voice-mail system by call forward on busy from a Cisco IP phone attached to the router, the voice-mail system expects to receive a sequence of digits identifying the mailbox associated with the forwarding phone together with digits that identify the extension number of the calling IP phone.
Note
Although it is unlikely that you will use multiple instances of the CGN, CDN, or FDN keyword in a single command line, it is permissible to do so.
Examples
The following example sets the DTMF pattern for a local call forwarded on busy to the voice-mail system:
Router(config)# vm-integration
Router(config-vm-integration)# pattern ext-to-ext busy 7 FDN * CGN *
Related Commands
Command
|
Description
|
pattern direct (vm- integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system when the user presses the messages button on the phone.
|
pattern ext-to-ext no-answer (vm- integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system once an internal extension fails to connect to an extension and the call is forwarded to voice mail.
|
pattern trunk-to-ext busy (vm-integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system once an external trunk call reaches a busy extension and the call is forwarded to voice mail.
|
pattern trunk-to-ext no-answer (vm- integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system when an external trunk call reaches an unanswered extension and the call is forwarded to voice mail.
|
vm-integration
|
Enters voice-mail integration configuration mode and enables voice-mail integration with DTMF and analog voice-mail systems.
|
pattern ext-to-ext no-answer (vm-integration)
To configure the dual-tone multifrequency (DTMF) digit pattern forwarding necessary to activate the voice-mail system once an internal extension fails to connect to an extension and the call is forwarded to voice mail, use the pattern ext-to-ext no-answer command in voice-mail integration configuration mode. To disable DTMF digit pattern forwarding when an internal extension fails to connect to an extension and the call is forwarded to a voice-mail system, use the no form of this command.
pattern ext-to-ext no-answer tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
no pattern ext-to-ext no-answer tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
Syntax Description
tag1
|
Alphanumeric string fewer than four DTMF digits in length. The alphanumeric string consists of a combination of four letters (A,B,C, and D), two symbols (* and #), and ten digits (0 to 9). The tag numbers match the numbers defined in the voice-mail system's integration file, immediately preceding either the number of the calling party, the number of the called party, or a forwarding number.
|
tag2 and tag3
|
(Optional) See tag1.
|
last-tag
|
(Optional) See tag1. This tag indicates the end of the pattern.
|
CGN
|
Calling number (CGN) information is sent to the voice-mail system.
|
CDN
|
Called number (CDN) information is sent to the voice-mail system.
|
FDN
|
Forwarding number (FDN) information is sent to the voice-mail system.
|
Defaults
This feature is disabled.
Command Modes
Voice-mail integration configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(2)XT
|
Cisco SRST 2.0
|
This command was introduced on the following platforms: Cisco 1750, Cisco 1751, Cisco 2600 series and Cisco 3600 series multiservice routers, and Cisco IAD2420 series IADs.
|
12.2(8)T
|
Cisco SRST 2.0
|
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725 and Cisco 3745 routers.
|
12.2(8)T1
|
Cisco SRST 2.0
|
This command was implemented on the Cisco 2600-XM and Cisco 2691 routers.
|
12.2(11)T
|
2.01
|
This command was integrated into Cisco IOS Release 12.2(11)T and implemented on the Cisco 1760 routers.
|
12.2(13)T
|
2.02
|
This command was introduced for Cisco SRST Version 2.02.
|
Usage Guidelines
The pattern ext-to-ext no-answer command is used to configure the sequence of DTMF digits passed to a voice-mail system attached to the router through one or more voice ports. When a call is routed to the voice-mail system by call forward on no-answer from an IP phone attached to the router, the voice-mail system expects to receive a sequence of digits identifying the mailbox associated with the forwarding phone together with digits that identify the extension number of the calling IP phone.
Note
Although it is unlikely that you will use multiple instances of the CGN, CDN, or FDN keyword in a single command line, it is permissible to do so.
Examples
The following example sets the DTMF pattern for a local call forwarded on no-answer to the voice-mail system:
Router(config)# vm-integration
Router(config-vm-integration)# pattern ext-to-ext no-answer 5 FDN * CGN *
Related Commands
Command
|
Description
|
pattern direct (vm-integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system when the user presses the messages button on the phone.
|
pattern ext-to-ext busy (vm-integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system once an internal extension fails to connect to an extension and the call is forwarded to voice mail.
|
pattern trunk-to-ext busy (vm-integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system once an external trunk call reaches a busy extension and the call is forwarded to voice mail.
|
pattern trunk-to-ext no-answer (vm-integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system when an external trunk call reaches an unanswered extension and the call is forwarded to voice mail.
|
vm-integration
|
Enters voice-mail integration configuration mode and enables voice-mail integration with DTMF and analog voice-mail systems.
|
pattern trunk-to-ext busy (vm-integration)
To configure the dual-tone multifrequency (DTMF) digit pattern forwarding necessary to activate the voice-mail system once an external trunk call reaches a busy extension and the call is forwarded to voice mail, use the pattern trunk-to-ext busy command in voice-mail integration configuration mode. To disable DTMF digit pattern forwarding when an external trunk call reaches a busy extension and the call is forwarded to a voice-mail system, use the no form of this command.
pattern trunk-to-ext busy tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
no pattern trunk-to-ext busy tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
Syntax Description
tag1
|
Alphanumeric string fewer than four DTMF digits in length. The alphanumeric string consists of a combination of four letters (A,B,C, and D), two symbols (* and #), and ten digits (0 to 9). The tag numbers match the numbers defined in the voice-mail system's integration file, immediately preceding either the number of the calling party, the number of the called party, or a forwarding number.
|
tag2 and tag3
|
(Optional) See tag1.
|
last-tag
|
(Optional) See tag1. This tag indicates the end of the pattern.
|
CGN
|
Calling number (CGN) information is sent to the voice-mail system.
|
CDN
|
Called number (CDN) information is sent to the voice-mail system.
|
FDN
|
Forwarding number (FDN) information is sent to the voice-mail system.
|
Defaults
This feature is disabled.
Command Modes
Voice-mail integration configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(2)XT
|
Cisco SRST 2.0
|
This command was introduced on the following platforms: Cisco 1750, Cisco 1751, Cisco 2600 series and Cisco 3600 series multiservice routers, and Cisco IAD2420 series IADs.
|
12.2(8)T
|
Cisco SRST 2.0
|
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725 and Cisco 3745 routers.
|
12.2(8)T1
|
Cisco SRST 2.0
|
This command was implemented on the Cisco 2600-XM and Cisco 2691 routers.
|
12.2(11)T
|
Cisco SRST 2.01
|
This command was integrated into Cisco IOS Release 12.2(11)T and implemented on the Cisco 1760 routers.
|
12.2(13)T
|
Cisco SRST 2.02
|
This command was introduced for Cisco SRST Version 2.02.
|
Usage Guidelines
The pattern trunk-to-ext busy command is used to configure the sequence of DTMF digits passed to a voice-mail system attached to the router through one or more voice ports. When a call is routed to the voice-mail system by call forward on busy from an IP phone attached to the router, the voice-mail system expects to receive a sequence of digits identifying the mailbox associated with the forwarding phone together with digits indicating that the call originated from a PSTN or VoIP caller.
Note
Although it is unlikely that you will use multiple instances of the CGN, CDN, or FDN keyword in a single command line, it is permissible to do so.
Examples
The following example sets the DTMF pattern for call forwarding when an external trunk call reaches a busy extension and the call is forwarded to the voice-mail system:
Router(config)# vm-integration
Router(config-vm-integration)# pattern trunk-to-ext busy 6 FDN * CGN *
Related Commands
Command
|
Description
|
pattern direct (vm- integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system when the user presses the messages button on the phone.
|
pattern ext-to-ext busy (vm-integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system once an internal extension attempts to connect to a busy extension and the call is forwarded to voice mail.
|
pattern ext-to-ext no-answer (vm- integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system once an internal extension fails to connect to an extension and the call is forwarded to voice mail.
|
pattern trunk-to-ext no-answer (vm- integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system when an external trunk call reaches an unanswered extension and the call is forwarded to voice mail.
|
vm-integration
|
Enters voice-mail integration configuration mode and enables voice-mail integration with DTMF and analog voice-mail systems.
|
pattern trunk-to-ext no-answer (vm-integration)
To configure the dual-tone multifrequency (DTMF) digit pattern forwarding necessary to activate the voice-mail system when an external trunk call reaches an unanswered extension and the call is forwarded to voice mail, use the pattern trunk-to-ext no-answer command in voice-mail integration configuration mode. To disable DTMF digit pattern forwarding when an external trunk call reaches another extension where the called party does not answer and the call is forwarded to a voice-mail system, use the no form of this command.
pattern trunk-to-ext no-answer tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
no pattern trunk-to-ext no-answer tag1 {CGN | CDN | FDN} [tag2 {CGN | CDN | FDN}]
[tag3 {CGN | CDN | FDN}] [last-tag]
Syntax Description
tag1
|
Alphanumeric string fewer than four DTMF digits in length. The alphanumeric string consists of a combination of four letters (A,B,C, and D), two symbols (* and #), and ten digits (0 to 9). The tag numbers match the numbers defined in the voice-mail system's integration file, immediately preceding either the number of the calling party, the number of the called party, or a forwarding number.
|
tag2 and tag3
|
(Optional) See tag1.
|
last-tag
|
(Optional) See tag1. This tag indicates the end of the pattern.
|
CGN
|
Calling number (CGN) information is sent to the voice-mail system.
|
CDN
|
Called number (CDN) information is sent to the voice-mail system.
|
FDN
|
Forwarding number (FDN) information is sent to the voice-mail system.
|
Defaults
This feature is disabled.
Command Modes
Voice-mail integration configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(2)XT
|
Cisco SRST 2.0
|
This command was introduced on the following platforms: Cisco 1750, Cisco 1751, Cisco 2600 series and Cisco 3600 series multiservice routers, and Cisco IAD2420 series IADs.
|
12.2(8)T
|
Cisco SRST 2.0
|
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725 and Cisco 3745 routers.
|
12.2(8)T1
|
Cisco SRST 2.0
|
This command was implemented on the Cisco 2600-XM and Cisco 2691 routers.
|
12.2(11)T
|
Cisco SRST 2.01
|
This command was integrated into Cisco IOS Release 12.2(11)T and implemented on the Cisco 1760 routers.
|
12.2(13)T
|
Cisco SRST 2.02
|
This command was introduced for Cisco SRST Version 2.02.
|
Usage Guidelines
The pattern trunk-to-ext no-answer command is used to configure the sequence of DTMF digits passed to a voice-mail system attached to the router through one or more voice ports. When a call is routed to the voice-mail system by call forward on no-answer from an IP phone attached to the router, the voice-mail system expects to receive a sequence of digits identifying the mailbox associated with the forwarding phone together with digits indicating that the call originated from a PSTN or VoIP caller.
Note
Although it is unlikely that you will use multiple instances of the CGN, CDN, or FDN keyword in a single command line, it is permissible to do so.
Examples
The following example sets the DTMF pattern for call forwarding when an external trunk call reaches an unanswered extension and the call is forwarded (FDN) to a voice-mail system:
Router(config)# vm-integration
Router(config-vm-integration)# pattern trunk-to-ext no-answer 4 FDN * CGN *
Related Commands
Command
|
Description
|
pattern direct (vm-integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system when the user presses the messages button on the phone.
|
pattern ext-to-ext busy (vm-integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system once an internal extension attempts to connect to a busy extension and the call is forwarded to voice mail.
|
pattern ext-to-ext no-answer (vm-integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system once an internal extension fails to connect to an extension and the call is forwarded to voice mail.
|
pattern trunk-to-ext busy (vm-integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system once an external trunk call reaches a busy extension and the call is forwarded to voice mail.
|
vm-integration
|
Enters voice-mail integration configuration mode and enables voice-mail integration with DTMF and analog voice-mail systems.
|
phoneload
To define the phone firmware support for a phone type, use the phoneload command in ephone-type configuration mode. To remove firmware support, use the no form of this command.
phoneload
no phoneload
Syntax Description
This command has no arguments or keywords.
Command Default
Phone type supports firmware configuration.
Command Modes
Ephone-type configuration (config-ephone-type)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.4(15)XZ
|
Cisco Unified CME 4.3 Cisco Unified SRST 4.3
|
This command was introduced.
|
12.4(20)T
|
Cisco Unified CME 7.0 Cisco Unified SRST 7.0
|
This command was integrated into Cisco IOS Release 12.4(20)T.
|
Usage Guidelines
This command specifies whether the phone type defined in the phone-type template supports firmware configuration using the load command.
Examples
The following example shows that support for phone firmware is disabled for the Nokia E61 phone type:
Router(config)# ephone-type E61
Router(config-ephone-type)# no phoneload
Related Commands
Command
|
Description
|
device-name
|
Assigns a name to a phone type in an ephone-type template.
|
load
|
Associates a type of Cisco Unified IP phone with a phone firmware file.
|
pickup (call-manager-fallback)
To enable the PickUp soft key on all Cisco IP phones, allowing an external Direct Inward Dialing (DID) call coming into one extension to be picked up from another extension during SRST, use the pickup command in call-manager-fallback configuration mode. To disable the PickUp soft key on all Cisco IP phones during SRST, use the no form of this command.
pickup telephone-number
no pickup telephone-number
Syntax Description
telephone-number
|
The telephone number to match an incoming called number.
|
Defaults
The PickUp soft key is disabled.
Command Modes
Call-manager-fallback configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.3(7)XL
|
Cisco SRST 3.1.1
|
This command was introduced.
|
12.3(11)T
|
Cisco SRST 3.2
|
This command was integrated into Cisco IOS Release 12.3(11)T.
|
Usage Guidelines
Configuring the pickup command enables the PickUp soft key on all SRST phones. You can then press the PickUp key and answer any currently ringing IP phone that has a DID called number that matches the configured telephone-number. This command does not enable the Group PickUp (GPickUp) soft key.
When a user presses the PickUp soft key, SRST searches through all the SRST phones to find a ringing call that has a called number that matches the configured telephone-number. When a match is found, the call is automatically forwarded to the extension number of the phone that requested the call pickup.
The SRST pickup command is designed to operate in a manner compatible with Cisco Unified Communications Manager.
Note
The default phone load on Cisco Unified Communications Manager, Release 4.0(1), for the Cisco 7905 and Cisco 7912 IP phones does not enable the PickUp soft key during fallback. To enable the PickUp soft key on Cisco 7905 and Cisco 7912 IP phones, upgrade your default phone load to Cisco Unified Communications Manager, Release 4.0(1) Sr2. Alternatively, you can upgrade the phone load to cmterm-7905g-sccp.3-3-8.exe or cmterm-7912g-sccp.3-3-8.exe, respectively.
Examples
In SRST, the pickup command is best used with the alias command. The following output from the show running-config command shows the pickup command and the alias command configured to provide call routing for a pilot number of a hunt group:
alias 1 8005550100 to 5001
alias 2 8005550100 to 5002
alias 3 8005550100 to 5003
alias 4 8005550100 to 5004
When a DID incoming call to 800 555-0100 is received, the alias command routes the call at random to one of the four extensions (5001 to 5004). Because the pickup command is configured, if the DID call rings on extension 5002, the call can be answered from any of the other extensions (5001, 5003, 5004) by pressing the PickUp soft key.
The pickup command works by finding a match based on the incoming DID called number. In this example, a call from extension 5004 to extension 5001 (internal call) does not activate the pickup command because the called number (5001) does not match the configured pickup number (800 555-0100). Thus, the pickup command distinguishes between internal and external calls if multiple calls are ringing simultaneously.
Related Commands
Command
|
Description
|
alias (call-manager- fallback)
|
Provides a mechanism for rerouting calls to telephone numbers that are unavailable during Cisco Unified Communications Manager fallback.
|
call-manager-fallback
|
Enables Cisco Unified SRST support and enters call-manager-fallback configuration mode.
|
preference (voice register pool)
To set the preference order for creating the VoIP dial peers created for a number associated with a voice pool, use the preference command in voice register pool configuration mode. To put the number in default preference order, use the no form of this command.
preference preference-order
no preference
Syntax Description
preference-order
|
Preference order for the extension or telephone number associated with a pool. Range is 0 to 10. Default is 0, which is the highest preference.
|
Command Default
0 (highest preference order)
Command Modes
Voice register pool configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco SIP SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco SIP SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
12.4(4)T
|
Cisco CME 3.4 Cisco SIP SRST 3.4
|
This command was added to Cisco Communications Manager Express (Cisco CME).
|
Usage Guidelines
When you create a voice register pool for a SIP phone or a group of SIP phones in a Cisco Unified CME or Cisco Unified Session Initiation Protocol (SIP) Survivable Remote Site Telephony (SRST) environemnt, you automatically create a virtual voice port and one to four virtual dial peers to be used by the number associated with that pool. The preference value is passed transparently to dial peers created for the number. The preference value allows you to control the selection of a desired dial peer when multiple dial peers are matched on the same destination pattern (extension or phone number) associated with the pool. In this way, the preference command can be used to establish a hunt strategy for incoming calls.
Note
Configure the id (voice register pool) command before any other voice register pool commands, including the preference command. The id command identifies a locally available individual SIP phone or set of Cisco SIP phones.
Examples
The following example shows how to set a preference of 2 for extension number 3000:
In the following example, extension number 1222 under voice register dn 4 has a higher preference than number 1222 under voice register pool 5.
Related Commands
Command
|
Description
|
id (voice register pool)
|
Explicitly identifies a locally available individual Cisco SIP IP phone, or when running Cisco Unified SIP SRST, set of Cisco SIP IP phones.
|
voice register pool
|
Enters voice register pool configuration mode for SIP phones.
|
proxy (voice register pool)
To autogenerate additional VoIP dial peers to reach the main proxy whenever a Cisco Session Initiation Protocol (SIP) IP phone registers with a SIP Survivable Remote Site Telephony (SRST) gateway, use the proxy command in voice register pool configuration mode. To disable a dial peer as a SIP proxy, use the no form of this command.
proxy ip-address [preference value] [monitor probe {icmp-ping | rtr} [alternate-ip-address]]
no proxy
Syntax Description
ip-address
|
IP address of the SIP proxy.
|
preference value
|
(Optional) Defines the preference of the proxy dial peers that are created. Range is from 0 to 10. The highest preference is 0. There is no default.
|
monitor probe
|
(Optional) Enables monitoring of proxy dial peers.
• icmp-ping—Enables monitoring of proxy dial peers using ICMP ping.
• rtr—Enables monitoring of proxy dial peers using RTR probes.
• alternate-ip-address—(Optional) Enables monitoring of alternate IP addresses other than the proxy address. For example, to monitor a gateway front end to a SIP proxy.
|
Command Default
Proxy dial peer is disabled.
Command Modes
Voice register pool configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco SIP SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco SIP SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
Usage Guidelines
The proxy command autogenerates additional VoIP dial peers to reach the main proxy whenever a Cisco SIP IP phone registers with a Cisco Unified SIP SRST gateway. This autogeneration process enables all PSTN calls to be routed first to the main proxy before the backup dial peers for local Cisco SIP IP phones are tried.
Proxy dial peers can be monitored using ICMP ping or RTR probes, in case of WAN failure. If the Cisco Unified SIP SRST gateway loses probes to the main proxy, the proxy dial peers are temporarily set to an operational down state. Then the backup dial peers can be selected faster to lower the call setup time. In addition, the proxy dial peers can be monitored using an alternate location other than the main proxy to monitor the status of the WAN link.
Only one proxy address can be set per voice register pool.
For proxy monitoring to work, the call fallback active command must be configured.
Note
The id (voice register pool) command must be configured before any other voice register pool commands, including the proxy command. The id command identifies a locally available individual Cisco SIP IP phone or sets of Cisco SIP IP phones.
Examples
The following partial sample output from the show running-config command shows that voice register pool 1 has defined 10.2.161.187 as the SIP proxy and that it is monitored by ICMP ping:
cor incoming call91 1 91011
translate-outgoing called 1
proxy 10.2.161.187 preference 1 monitor probe icmp-ping
alias 1 94... to 91011 preference 8
Related Commands
Command
|
Description
|
call fallback active
|
Enables a call request to fall back to alternate dial peers in case of network congestion.
|
id (voice register pool)
|
Explicitly identifies a locally available individual Cisco SIP IP phone or set of Cisco SIP IP phones.
|
voice register pool
|
Enables SIP SRST voice register pool configuration commands.
|
registrar server (SIP)
To enable SIP registrar functionality, use the registrar server command in SIP configuration mode. To disable SIP registrar functionality, use the no form of the command.
registrar server [expires [max sec] [min sec] ]
no registrar server
Syntax Description
expires
|
(Optional) Sets the active time for an incoming registration.
|
max sec
|
(Optional) Maximum expires time for a registration, in seconds. The range is from 600 to 86400. The default is 3600.
|
min sec
|
(Optional) Minimum expires time for a registration, in seconds. The range is from 60 to 3600. The default is 60.
|
Defaults
Disabled
Command Modes
SIP configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco SIP SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco SIP SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
12.4(4)T
|
Cisco CME 3.4 and Cisco SIP SRST 3.4
|
This command was added to Cisco CME.
|
Usage Guidelines
When this command is entered, the router accepts incoming SIP Register messages. If SIP Register message requests are for a shorter expiration time than what is set with this command, the SIP Register message expiration time is used.
This command is mandatory for Cisco Unified SIP SRST or Cisco Unified CME and must be entered before any voice register pool or voice register global commands are configured.
If the WAN is down and you reboot your Cisco Unified CME or Cisco Unified SIP SRST router, when the router reloads it will have no database of SIP phone registrations. The SIP phones will have to register again, which could take several minutes, because SIP phones do not use a keepalive functionality. To shorten the time before the phones re-register, the registration expiry can be adjusted with this command. The default expiry is 3600 seconds; an expiry of 600 seconds is recommended.
Examples
The following partial sample output from the show running-config command shows that SIP registrar functionality is set:
voice service voip
allow-connections sip-to-sip
sip
registrar server expires max 1200 min 300
Related Commands
Command
|
Description
|
sip
|
Enters SIP configuration mode from voice service VoIP configuration mode.
|
voice register global
|
Enters voice register global configuration mode in order to set global parameters for all supported Cisco SIP phones in a Cisco Unified CME or Cisco Unified SIP SRST environment.
|
voice register pool
|
Enters voice register pool configuration mode for SIP phones.
|
reset (call-manager-fallback)
To reset Cisco IP phones, use the reset command in call-manager-fallback configuration mode.
reset {all seconds | mac-address mac-address}
Syntax Description
all
|
All Cisco IP phones.
|
seconds
|
Time interval, in seconds, that passes between each Cisco IP phone resetting. The range is from 0 to 60.
|
mac-address mac-address
|
MAC address of a particular Cisco IP phone.
|
Defaults
No default behavior or values.
Command Modes
Call-manager-fallback configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.1(5)YD
|
Cisco SRST 1.0
|
This command was introduced on the following platforms: Cisco 2600 series and Cisco 3600 series multiservice routers; Cisco IAD2420 series IADs.
|
12.2(2)XT
|
Cisco SRST 2.0
|
This command was implemented on Cisco 1750 and Cisco 1751 multiservice routers.
|
12.2(8)T
|
Cisco SRST 2.0
|
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725, Cisco 3745, and Cisco MC3810-V3 routers.
|
12.2(8)T1
|
Cisco SRST 2.0
|
This command was implemented on the Cisco 2600-XM and Cisco 2691 routers.
|
12.2(11)T
|
Cisco SRST 2.01
|
This command was integrated into Cisco IOS Release 12.2(11)T and implemented on the Cisco 1760 routers.
|
Usage Guidelines
This command does not have a no form.
Examples
The following example resets all Cisco IP phones in 8-second intervals:
Router(config)# call-manager-fallback
Router(config-cm-fallback)# reset all 8
The following example resets the Cisco IP phone with MAC address CFBA.321B.96FA:
Router(config)# call-manager-fallback
Router(config-cm-fallback)# reset mac-address CFBA.321B.96FA
Related Commands
Command
|
Description
|
call-manager-fallback
|
Enables Cisco Unified SRST support and enters call-manager-fallback configuration mode.
|
secondary-dialtone (call-manager-fallback)
To enable a secondary dial tone when a Cisco Unified IP phone user dials a defined PSTN access prefix, use the secondary-dialtone command in call-manager-fallback configuration mode. To disable the secondary dial tone, use the no form of this command.
secondary-dialtone digit-string
no secondary-dialtone digit-string
Syntax Description
digit-string
|
The number of the access prefix.
|
Defaults
Secondary dial tone is disabled.
Command Modes
Call-manager-fallback configuration
Command History
Cisco IOS Release
|
Cisco Unified Product
|
Modification
|
12.2(15)ZJ
|
Cisco Unified SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco Unified SRST 3.0
|
This command was integrated into Cisco Unified IOS Release 12.3(4)T.
|
Usage Guidelines
The secondary dial tone is turned off when the next number after the digit-string is pressed. For example, if 8 were the digit-string and a person were dialing 8 555-0100, the secondary dial tone would be turned off when the number 5 is pressed.
The tone value for the secondary dial is the skinny DtOutsideDialtone.
Examples
The following enables a secondary dial tone when a Cisco Unified IP phone users enters the number nine to get an outside line:
Router(config)# call-manager-fallback
Router(config-cm-fallback)# secondary-dialtone 9
Related Commands
Command
|
Description
|
call-manager-fallback
|
Enables Cisco Unified Unified SRST support and enters call-manager-fallback configuration mode.
|
security
To define whether a phone type supports security features, use the security command in ephone-type configuration mode. To disable security support, use the no form of this command.
security
no security
Syntax Description
This command has no arguments or keywords.
Command Default
Enabled (phone type supports security features).
Command Modes
Ephone-type configuration (config-ephone-type)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.4(15)XZ
|
Cisco Unified CME 4.3 Cisco Unified SRST 4.3
|
This command was introduced.
|
12.4(20)T
|
Cisco Unified CME 7.0 Cisco Unified SRST 7.0
|
This command was integrated into Cisco IOS Release 12.4(20)T.
|
Usage Guidelines
This command specifies whether security features are supported by the type of phone being added with a phone-type template.
Examples
The following example shows that support for security is disabled for the Nokia E61 when creating the ephone-type template:
Router(config)# ephone-type E61
Router(config-ephone-type)# no security
Related Commands
Command
|
Description
|
device-id
|
Specifies the device ID for a phone type.
|
type
|
Assigns a phone type to an SCCP phone.
|
security-policy (voice register global)
To define the security level of SIP phones allowed to register, use the security-policy command in voice register global configuration mode. To return to the default, use the no form of this command.
Note
The security-policy command only works with SRST. While it is possible to configure this command when in CME mode, TLS-based connections from Cisco Unified IP Phones will fail. This failure will occur even if using the "CME-as-SRST" failover model.
Cisco IOS Release 15.0(1)XA and later releases
security-policy secure
no security-policy secure
Syntax Description
secure
|
Requires SIP phones to use TLS for signaling transport. Non-secure SIP phones are blocked from registering. Valid for Cisco Unified SRST.
|
Command Default
All levels of phone security are permitted to register, also known as device-default mode.
Command Modes
Voice register global configuration (config-register-global)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
15.0(1)XA
|
Cisco Unified SRST 8.0
|
This command was introduced.
|
Usage Guidelines
The secure keyword configures the SIP registration security policy so that only encrypted phones may register to the Cisco Unified SRST device in the event of a failover from the primary call control. When this keyword is configured, non-secure phones using TCP or UDP for signaling transport, as well as authenticated phones using TLS/TCP for signaling transport, will be blocked from registering.
Examples
The following example shows that only registration requests from encrypted SIP phones in a Cisco Unified SRST system are permitted:
Router(config)# voice register global
Router(config-register-global)# security-policy secure
Related Commands
Command
|
Description
|
crypto signaling
|
Identifies the trustpoint and encryption restrictions used during the TLS handshake.
|
show call-manager-fallback all
To display the detailed configuration of all Cisco IP phones, directory numbers, voice ports, and dial peers in your network during Cisco Unified CallManager fallback, use the show call-manager-fallback all command in privileged EXEC mode.
show call-manager-fallback all
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.1(5)YD
|
Cisco SRST 1.0
|
This command was introduced on the following platforms: Cisco 2600 series and Cisco 3600 series multiservice routers; Cisco IAD2420 series IADs.
|
12.2(2)XT
|
Cisco SRST 2.0
|
This command was implemented on Cisco 1750 and Cisco 1751 multiservice routers.
|
12.2(8)T
|
Cisco SRST 2.0
|
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725, Cisco 3745, and Cisco MC3810-V3 routers.
|
12.2(8)T1
|
Cisco SRST 2.0
|
This command was implemented on the Cisco 2600-XM and Cisco 2691 routers.
|
12.2(11)T
|
Cisco SRST 2.01
|
This command was integrated into Cisco IOS Release 12.2(11)T and implemented on the Cisco 1760 routers.
|
12.3(4)T
|
Cisco SRST 3.0
|
The Version was added to output.
|
Examples
The following is sample output from the show call-manager-fallback all command:
Router #show call-manager-fallback
CONFIG (Version=4.1(0))
=====================
Version 4.1(0)
For on-line documentation please see:
www.cisco.com/univercd/cc/td/doc/product/access/ip_ph/ip_ks/index.htm
ip source-address 0.0.0.0 port 2000
max-ephones 0
max-dn 0
max-conferences 8 gain -6
dspfarm units 0
dspfarm transcode sessions 0
huntstop
cnf-file location: system:
cnf-file option: PER-PHONE-TYPE
network-locale[0] US (This is the default network locale for this box)
network-locale[1] US
network-locale[2] US
network-locale[3] US
network-locale[4] US
user-locale[0] US (This is the default user locale for this box)
user-locale[1] US
user-locale[2] US
user-locale[3] US
user-locale[4] US
srst mode auto-provision is OFF
srst ephone template is 0
srst dn template is 0
srst dn line mode is single
time-format 12
date-format mm-dd-yy
timezone 0 Greenwich Standard Time
no transfer-pattern is configured, transfer is restricted to local SCCP phones only.
keepalive 30 auxiliary 30
timeout interdigit 10
timeout busy 10
timeout ringing 180
timeout ringin-callerid 8
caller-id name-only: enable
Limit number of DNs per phone:
7910: 36
7935: 36
7936: 36
7940: 36
7960: 36
7970: 36
Log (table parameters):
max-size: 150
retain-timer: 15
transfer-system full-consult
local directory service: enabled.
Extension-assigner tag-type ephone-tag.
========================================
Table 6 describes the significant fields shown in the display.
Table 6 show call-manager-fallback all Field Descriptions
Field
|
Description
|
destination-pattern
|
Destination pattern (telephone number) configured for this dial peer.
|
dial-peer voice
|
Voice dial peer.
|
ephone-dn
|
Cisco IP phone directory number.
|
(no) huntstop
|
Whether or not huntstop is set.
|
ip source-address
|
IP address used by the Cisco IP phones to register with the router for service.
|
keepalive
|
Cisco IP phone keepalive period in seconds.
|
max-dn
|
Maximum directory numbers or virtual voice ports.
|
max-ephones
|
Maximum number of Cisco IP phones.
|
port
|
TCP port number used by the Cisco IP phones to communicate with the router.
|
station-id number
|
Number used for caller-ID purposes when calls are made using the line.
|
voice-port
|
(Virtual) voice port designator.
|
Version
|
SRST version number designation.
|
Related Commands
Command
|
Description
|
show call-manager- fallback dial-peer
|
Displays detailed configuration output for the dial peers in your network during Cisco Unified CallManager fallback.
|
show call-manager- fallback ephone-dn
|
Displays output for the Cisco IP phone directory numbers or virtual voice ports during Cisco Unified CallManager fallback.
|
show call-manager- fallback voice-port
|
Displays output for the voice ports while Cisco Unified CallManager is active.
|
show call-manager-fallback dial-peer
To display detailed configuration output for the dial peers in your network during Cisco Unified Communications Manager fallback, use the show call-manager-fallback dial-peer command in privileged EXEC mode.
show call-manager-fallback dial-peer
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.1(5)YD
|
Cisco SRST 1.0
|
This command was introduced on the following platforms: Cisco 2600 series and Cisco 3600 series multiservice routers; Cisco IAD2420 series IADs.
|
12.2(2)XT
|
Cisco SRST 2.0
|
This command was implemented on Cisco 1750 and Cisco 1751 multiservice routers.
|
12.2(8)T
|
Cisco SRST 2.0
|
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725, Cisco 3745, and Cisco MC3810-V3 routers.
|
12.2(8)T1
|
Cisco SRST 2.0
|
This command was implemented on the Cisco 2600-XM and Cisco 2691 routers.
|
12.2(11)T
|
Cisco SRST 2.01
|
This command was integrated into Cisco IOS Release 12.2(11)T and implemented on the Cisco 1760 routers.
|
Examples
The following is sample output from the show call-manager-fallback dial-peer command:
Router# show call-manager-fallback dial-peer
dial-peer voice 20046 pots
dial-peer voice 20047 pots
dial-peer voice 20048 pots
dial-peer voice 20049 pots
Table 7 describes the significant fields shown in the display.
Table 7 show call-manager-fallback dial-peer Field Descriptions
Field
|
Description
|
call-forward busy
|
Indicates call forwarding when a Cisco IP phone is busy.
|
call-forward noan
|
Indicates call forwarding when no answer is received from a Cisco IP phone.
|
destination-pattern
|
Destination pattern (telephone number) configured for this dial peer.
|
dial-peer voice
|
Voice dial peer.
|
port
|
(Virtual) voice port designator.
|
Related Commands
Command
|
Description
|
show call-manager- fallback all
|
Displays the detailed configuration of all Cisco IP phones, directory numbers, voice ports, and dial peers in your network during Cisco Unified Communications Manager fallback.
|
show call-manager- fallback ephone-dn
|
Displays output for the Cisco IP phone directory numbers or virtual voice ports during Cisco Unified Communications Manager fallback.
|
show call-manager- fallback voice-port
|
Displays output for the voice ports while Cisco Unified Communications Manager is active.
|
show call-manager-fallback ephone-dn
To display detailed configuration output for the Cisco IP phone directory numbers or virtual voice ports during Cisco Unified CallManager fallback, use the show call-manager-fallback ephone-dn command in privileged EXEC mode.
show call-manager-fallback ephone-dn
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.1(5)YD
|
Cisco SRST 1.0
|
This command was introduced on the following platforms: Cisco 2600 series and Cisco 3600 series multiservice routers; Cisco IAD2420 series IADs.
|
12.2(2)XT
|
Cisco SRST 2.0
|
This command was implemented on Cisco 1750 and Cisco 1751 multiservice routers.
|
12.2(8)T
|
Cisco SRST 2.0
|
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725, Cisco 3745, and Cisco MC3810-V3 routers.
|
12.2(8)T1
|
Cisco SRST 2.0
|
This command was implemented on the Cisco 2600-XM and Cisco 2691 routers.
|
12.2(11)T
|
Cisco SRST 2.01
|
This command was integrated into Cisco IOS Release 12.2(11)T and implemented on the Cisco 1760 routers.
|
12.3(4)T
|
Cisco SRST 3.0
|
The Version, ephone-dn,vand voice-port was added to output.
|
Examples
The following is sample output from the show call-manager-fallback ephone-dn command:
Router# Router #show call-manager-fallback
CONFIG (Version=4.1(0))
=====================
Version 4.1(0)
For on-line documentation please see:
www.cisco.com/univercd/cc/td/doc/product/access/ip_ph/ip_ks/index.htm
ip source-address 0.0.0.0 port 2000
max-ephones 0
max-dn 0
max-conferences 8 gain -6
dspfarm units 0
dspfarm transcode sessions 0
huntstop
cnf-file location: system:
cnf-file option: PER-PHONE-TYPE
network-locale[0] US (This is the default network locale for this box)
network-locale[1] US
network-locale[2] US
network-locale[3] US
network-locale[4] US
user-locale[0] US (This is the default user locale for this box)
user-locale[1] US
user-locale[2] US
user-locale[3] US
user-locale[4] US
srst mode auto-provision is OFF
srst ephone template is 0
srst dn template is 0
srst dn line mode is single
time-format 12
date-format mm-dd-yy
timezone 0 Greenwich Standard Time
no transfer-pattern is configured, transfer is restricted to local SCCP phones only.
keepalive 30 auxiliary 30
timeout interdigit 10
timeout busy 10
timeout ringing 180
timeout ringin-callerid 8
caller-id name-only: enable
Limit number of DNs per phone:
7910: 36
7935: 36
7936: 36
7940: 36
7960: 36
7970: 36
Log (table parameters):
max-size: 150
retain-timer: 15
transfer-system full-consult
local directory service: enabled.
Extension-assigner tag-type ephone-tag.
========================================
Table 8 describes the significant fields shown in the display.
Table 8 show call-manager-fallback ephone-dn Field Descriptions
Field
|
Description
|
ephone-dn
|
Cisco IP phone directory number.
|
(no) huntstop
|
Whether or not huntstop is set.
|
number
|
Cisco IP phone number.
|
translate called
|
The configured translation rule. Can be called or calling, plus the tag number by which the rule set is referenced.
|
Related Commands
Command
|
Description
|
show call-manager- fallback all
|
Displays the detailed configuration of all Cisco IP phones, directory numbers, voice ports, and dial peers in your network during Cisco Unified CallManager fallback.
|
show call-manager- fallback dial-peer
|
Displays detailed configuration output for the dial peers in your network during Cisco Unified CallManager fallback.
|
show call-manager- fallback voice-port
|
Displays output for the voice ports while Cisco Unified CallManager is active.
|
show call-manager-fallback voice-port
To display detailed configuration output for the voice ports while Cisco Unified Communications Manager is active, use the show call-manager-fallback voice-port command in privileged EXEC mode.
show call-manager-fallback voice-port
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.1(5)YD
|
Cisco SRST 1.0
|
This command was introduced on the following platforms: Cisco 2600 series and Cisco 3600 series multiservice routers, and Cisco IAD2420 series IADs.
|
12.2(2)XT
|
Cisco SRST 2.0
|
This command was implemented on Cisco 1750 and Cisco 1751 multiservice routers.
|
12.2(8)T
|
Cisco SRST 2.0
|
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725, Cisco 3745, and Cisco MC3810-V3 routers.
|
12.2(8)T1
|
Cisco SRST 2.0
|
This command was implemented on the Cisco 2600-XM and Cisco 2691 routers.
|
12.2(11)T
|
Cisco SRST 2.01
|
This command was integrated into Cisco IOS Release 12.2(11)T and implemented on the Cisco 1760 routers.
|
Examples
The following is sample output from the show call-manager-fallback voice-port command:
Router# show call-manager-fallback voice-port
Table 9 describes the significant fields shown in the display.
Table 9 show call-manager-fallback voice-port Field Descriptions
Field
|
Description
|
voice-port
|
(Virtual) voice port.
|
station-id number
|
The phone number used for caller-ID purposes for calls made from this voice port.
|
Related Commands
Command
|
Description
|
show call-manager- fallback all
|
Displays the detailed configuration of all Cisco IP phones, directory numbers, voice ports, and dial peers in your network during Cisco Unified Communications Manager fallback.
|
show call-manager- fallback dial-peer
|
Displays detailed configuration output for the dial peers in your network during Cisco Unified Communications Manager fallback.
|
show call-manager- fallback ephone-dn
|
Displays output for the Cisco IP phone directory numbers or virtual voice ports during Cisco Unified Communications Manager fallback.
|
show credentials
To display the credentials settings that have been configured for use during Cisco Unified CME phone authentication communications or secure Cisco Unified SRST fallback, use the show credentials command in privileged EXEC mode.
show credentials
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.3(14)T
|
Cisco SRST 3.3
|
This command was introduced for Cisco Unified SRST.
|
12.4(4)XC
|
Cisco Unified CME 4.0
|
This command was introduced for Cisco Unified CME.
|
Usage Guidelines
Cisco Unified CME
This command displays the credentials settings on a Cisco Unified CME router that has been configured with a CTL provider to be used with Cisco Unified CME phone authentication.
Cisco Unified SRST
This command displays the credentials settings on the Cisco Unified SRST router that are supplied to Cisco Unified Communications Manager for use during secure SRST fallback.
Examples
The following is sample output from the show credentials command:
Credentials IP: 10.1.1.22
Table 10 describes the fields in the sample output.
Table 10 show credentials Field Descriptions
Field
|
Description
|
Credentials IP
|
Cisco Unified CME—IP address where the CTL provider is configured.
Cisco Unified SRST—The specified IP address where certificates from Cisco Unified Communications Manager to the SRST router are received.
|
Credentials PORT
|
Cisco Unified CME—TCP port for credentials service communication. Default is 2444.
Cisco Unified SRST—The port to which the SRST router connects to receive messages from the Cisco Unified IP phones. The port number is from 2000 to 9999. The default port number is 2445.
|
Trustpoint
|
Cisco Unified CME—CTL provider trustpoint label that will be used for TLS sessions with the CTL client.
Cisco Unified SRST—The name of the trustpoint that is associated with the credentials service between the Cisco Unified Communications Manager client and the SRST router.
|
Related Commands
Command
|
Description
|
credentials
|
Enters credentials configuration mode to configure a Cisco Unified CME CTL provider certificate or a Cisco Unified SRST router certificate.
|
ctl-service admin
|
Specifies a user name and password to authenticate the CTL client during the CTL protocol.
|
debug credentials
|
Sets debugging on the credentials service that runs between a Cisco Unified CME CTL provider and the CTL client or between a Cisco Unified SRST router and Cisco Unified Communications Manager.
|
ip source-address (credentials)
|
Enables the Cisco Unified CME or SRST router to receive messages through the specified IP address and port.
|
trustpoint (credentials)
|
Specifies the name of the trustpoint to be associated with a Cisco Unified CME CTL provider certificate or with a Cisco Unified SRST router certificate.
|
show ephone
To display information about registered Cisco Unified IP phones, use the show ephone command in privileged EXEC mode.
show ephone [mac-address | phone-type]
Syntax Description
mac-address
|
(Optional) Displays information for the phone with the specified MAC address.
|
phone-type
|
(Optional) Displays information for phones of the specified phone type. Valid types are as follows:
• 7902—Cisco Unified IP Phone 7902G.
• 7905—Cisco Unified IP Phone 7905G.
• 7906—Cisco Unified IP Phone 7905G.
• 7910—Cisco Unified IP Phone 7910 and 7910G.
• 7911—Cisco Unified IP Phone 7911G.
• 7912—Cisco Unified IP Phone 7912G.
• 7914—Cisco Unified IP Phone 7914 Expansion Module.
• 7920—Cisco Unified Wireless IP Phone 7920.
• 7921—Cisco Unified Wireless IP Phone 7921.
• 7931—Cisco Unified Wireless IP Phone 7931G.
• 7935—Cisco Unified IP Conference Station 7935.
• 7936—Cisco Unified IP Conference Station 7936.
• 7940—Cisco Unified IP Phones 7940 and 7940G.
• 7941—Cisco Unified IP Phone 7941G.
• 7941GE—Cisco Unified IP Phone 7941G-GE.
• 7942—Cisco Unified IP Phone 7942.
• 7945—Cisco Unified IP Phone 7945.
• 7960—Cisco Unified IP Phones 7960 and 7960G.
• 7961—Cisco Unified IP Phone 7961G.
• 7961GE—Cisco Unified IP Phone 7961G-GE.
• 7962—Cisco Unified IP Phone 7962.
• 7965—Cisco Unified IP Phone 7965.
• 7970—Cisco Unified IP Phone 7970G.
• 7971—Cisco Unified IP Phone 7971G-GE.
• 7975—Cisco Unified IP Phone 7975
• 7985—Cisco Unified IP Phone 7985.
• ata—Cisco ATA-186 or Cisco ATA-188.
|
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.1(5)YD
|
Cisco ITS 1.0 Cisco SRST 1.0
|
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco IAD2420 series.
|
12.2(2)XT
|
Cisco ITS 2.0 Cisco SRST 2.0
|
This command was implemented on the Cisco 1750 and Cisco 1751.
|
12.2(8)T
|
Cisco ITS 2.0 Cisco SRST 2.0
|
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725 and Cisco 3745.
|
12.2(8)T1
|
Cisco ITS 2.0 Cisco SRST 2.0
|
This command was implemented on the Cisco 2600XM and Cisco 2691.
|
12.2(11)T
|
Cisco ITS 2.01 Cisco SRST 2.01
|
The ata keyword was added and this command was implemented on the Cisco 1760.
|
12.2(11)YT
|
Cisco ITS 2.1 Cisco SRST 2.1
|
The 7914 keyword was added.
|
12.2(15)ZJ
|
Cisco CME 3.0 Cisco SRST 3.0
|
The 7902, 7905, and 7912 keywords were added.
|
12.3(7)T
|
Cisco CME 3.1 Cisco SRST 3.1
|
The 7920 and 7936 keywords were added.
|
12.3(11)XL
|
Cisco CME 3.2.1 Cisco SRST 3.2.1
|
The 7970 keyword was added.
|
12.3(14)T
|
Cisco CME 3.3 Cisco SRST 3.3
|
The 7971 keyword was added, and this command was integrated into Cisco IOS Release 12.3(14)T.
|
12.4(4)XC
|
Cisco Unified CME 4.0 Cisco Unified SRST 4.0
|
The 7911, 7941, 7941GE, 7961, and 7961GE keywords were added.
|
12.4(9)T
|
Cisco Unified CME 4.0 Cisco Unified SRST 4.0
|
The 7911, 7941, 7941GE, 7961, and 7961GE keywords were integrated into Cisco IOS Release 12.4(9)T.
|
12.4(6)XE
|
Cisco Unified CME 4.0(2)
|
The 7931 keyword was added for Cisco Unified CME.
|
12.4(4)XC4
|
Cisco Unified CME 4.0(3)
|
The 7931 keyword was added for Cisco Unified CME.
|
12.4(11)T
|
Cisco Unified CME 4.0(3)
|
The 7931 keyword for Cisco Unified CME was integrated in Cisco IOS Release 12.4(11)T.
|
12.4(11)XJ2
|
Cisco Unified CME 4.1 Cisco Unified SRST 4.1
|
The 7921 and 7985 keywords were introduced.
|
12.4(15)T1
|
Cisco Unified CME 4.1(1) Cisco Unified SRST 4.1(1)
|
The 7942, 7945, 7962, 7965, and 7975 keywords were introduced.
|
12.4(11)XW3
|
Cisco Unified CME 4.2 Cisco Unified SRST 4.2
|
The 7942, 7945, 7962, 7965, and 7975 keywords were introduced.
|
12.4(15)XY
|
Cisco Unified CME 4.2(1) Cisco Unified SRST 4.2(1)
|
Emergency response location (ERL) information displays in the output.
|
12.4(20)T
|
Cisco Unified CME 7.0 Cisco Unified SRST 7.0
|
This command was integrated into Cisco IOS Release 12.4(20)T.
|
Examples
Significant fields in the output from this command are described in Table 11.
The following sample output shows general information for registered phones:
ephone-1 Mac:0003.E3E7.F627 TCP socket:[2] activeLine:0 REGISTERED
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:10.0.0.2 51671 Telecaster 7940 keepalive 28 max_line 2
button 1: dn 1 number 4444
ephone-2 Mac:0030.94C3.F43A TCP socket:[1] activeLine:0 REGISTERED
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:10.0.0.3 50094 Telecaster 7960 keepalive 28 max_line 6
button 1: dn 3 number 5555
ephone-3 Mac:0003.6B40.99DA TCP socket:[3] activeLine:0 REGISTERED
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:10.2.168.200 51879 Telecaster 7960 keepalive 28 max_line 6
button 1: dn 2 number 3333
The following sample output shows general information for the phone with the MAC address 0003.E3E7.F627:
Router# show ephone 0003.E3E7.F627
ephone-1 Mac:0003.E3E7.F627 TCP socket:[2] activeLine:0 REGISTERED
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:10.0.0.2 51671 Telecaster 7940 keepalive 28 max_line 2
button 1: dn 1 number 4444
Active Call on DN 1:3001 10.0.0.51 31808 to 10.2.159.100 22708
Tx Pkts 452 bytes 41584 Rx Pkts 452 bytes 41584 Lost 0
The following sample output shows information for a phone that has two Cisco Unified IP Phone 7914 Expansion Modules attached. The output shows this module as a subsidiary type in addition to the main 7960 type for the phone itself. Subtype 3 means that one Cisco Unified IP Phone 7914 Expansion Module is attached to the main Cisco Unified IP Phone 7960 or 7960G, and subtype 4 means that two are attached.
ephone-2 Mac:0007.0EA6.39F8 TCP socket:[2] activeLine:0 REGISTERED
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:10.2.205.206 49278 Telecaster 7960 sub=4 keepalive 2723 max_line 34
button 1: dn 21 number 60021 CH1 IDLE
button 2: dn 22 number 60022 CH1 IDLE
button 7: dn 11 number 60011 CH1 IDLE monitor-ring
button 8: dn 12 number 60012 CH1 IDLE monitor-ring
button 9: dn 13 number 60013 CH1 IDLE monitor-ring
button 10: dn 14 number 60014 CH1 IDLE monitor-ring
button 11: dn 15 number 60015 CH1 IDLE monitor-ring
button 12: dn 16 number 60016 CH1 IDLE monitor-ring
button 13: dn 17 number 60017 CH1 IDLE monitor-ring
button 14: dn 18 number 60018 CH1 IDLE monitor-ring
button 15: dn 19 number 60019 CH1 IDLE monitor-ring
button 16: dn 20 number 60020 CH1 IDLE monitor-ring
button 17: dn 39 number 60039 CH1 IDLE CH2 IDLE monitor-ring
button 18: dn 40 number 60040 CH1 IDLE CH2 IDLE monitor-ring
button 19: dn 23 number 60023 CH1 IDLE monitor-ring
button 20: dn 24 number 60024 CH1 IDLE monitor-ring
button 21: dn 25 number 60025 CH1 IDLE monitor-ring
button 22: dn 26 number 60026 CH1 IDLE monitor-ring
button 23: dn 27 number 60027 CH1 IDLE monitor-ring
button 24: dn 28 number 60028 CH1 IDLE monitor-ring
button 25: dn 29 number 60029 CH1 IDLE monitor-ring
button 26: dn 30 number 60030 CH1 IDLE monitor-ring
button 27: dn 31 number 60031 CH1 IDLE CH2 IDLE monitor-ring
button 28: dn 32 number 60032 CH1 IDLE CH2 IDLE monitor-ring
button 29: dn 33 number 60033 CH1 IDLE CH2 IDLE monitor-ring
button 30: dn 34 number 60034 CH1 IDLE CH2 IDLE monitor-ring
button 31: dn 35 number 60035 CH1 IDLE CH2 IDLE monitor-ring
button 32: dn 36 number 60036 CH1 IDLE CH2 IDLE monitor-ring
button 33: dn 37 number 60037 CH1 IDLE CH2 IDLE monitor-ring
button 34: dn 38 number 60038 CH1 IDLE CH2 IDLE monitor-ring
The following sample output shows a phone that has a paging-dn and has received a page:
ephone-2 Mac:0087.0E76.B93C TCP socket:[4] activeLine:0 REGISTERED
mediaActive:1 offhook:0 ringing:0 reset:0 reset_sent:0 paging 1 debug:0
IP:10.50.50.20 49231 Telecaster 7910 keepalive 112 max_line 2 dual-line
button 1:dn 3 number 95021 CH1 IDLE
Table 11 describes significant fields in the output.
Table 11 show ephone Field Descriptions
Field
|
Description
|
Active Call
|
An active call is in progress.
|
activeLine
|
Line (button) on the phone that is in use. Zero indicates that no line is in use.
|
auto-dial number
|
Intercom extension that automatically dials number.
|
button number: dn number
|
Phone button number and the extension (ephone-dn) dn-tag number associated with that button.
|
bytes
|
Total number of voice data bytes sent or received by the phone.
|
Called Dn, Calling Dn
|
Ephone-dn tag numbers of the called and calling ephone-dn. Set to -1 if the call is not to or from an ephone-dn, or if there is no active call.
|
cfa number
|
Call-forward-all to number is enabled for this extension.
|
CH1 CH2
|
Status of channel 1 and, if this is a dual-line ephone-dn, the status of channel 2.
|
debug
|
1 indicates that debug for the phone is enabled. 0 indicates that debug is disabled.
|
DnD
|
Do Not Disturb is set on this phone.
|
DP tag
|
Not used.
|
ephone-number
|
Unique sequence number used to identify this phone during configuration (phone-tag).
|
IP
|
Assigned IP address of the Cisco Unified IP phone.
|
Jitter
|
Amount of variation (in milliseconds) of the time interval between voice packets received by the Cisco Unified IP phone.
|
keepalive
|
Number of keepalive messages received from the Cisco Unified IP phone by the router.
|
Latency
|
Estimated playout delay for voice packets received by the Cisco Unified IP phone.
|
line number
|
Button number on an IP phone. Line 1 is the button nearest the top of the phone.
|
Lost
|
Number of voice packets lost, as calculated by the Cisco Unified IP phone, on the basis of examining voice packet time-stamp and sequence numbers during playout.
|
Mac
|
MAC address.
|
Max Conferences
|
Maximum number of allowable conference calls and number of active conference calls.
|
max_line number
|
Maximum number of line buttons that can be configured on this phone.
|
mediaActive
|
1 indicates that an active conversation is in progress. 0 indicates that no conversation is ongoing.
|
monitor-ring
|
This button is set up as a monitor button.
|
number
|
Telephone or extension number associated with the Cisco Unified IP phone button and its dn-tag.
|
offhook
|
1 indicates that the phone is off-hook. 0 indicates that the phone is on-hook.
|
overlay
|
This button contains an overlay set. Use show ephone overlay to display the contents of overlay sets.
|
paging
|
1 indicates that the phone has received an audio page. 0 indicates that the phone has not received an audio page.
|
paging-dn
|
Ephone-dn that is dedicated for receiving audio pages on this phone. The paging-dn number is the number of the paging set to which this phone belongs.
|
Password
|
Authentication string that the phone user types when logging in to the web-based Cisco Unified CME GUI.
|
Port
|
Port used for TAPI transmissions.
|
REGISTERED
|
The Cisco Unified IP phone is active and registered. Alternative states are UNREGISTERED (indicating that the connection to the Cisco Unified IP phone was closed in a normal manner) and DECEASED (indicating that the connection to the Cisco Unified IP phone was closed because of a keepalive timeout).
|
reset
|
Pending reset.
|
reset_sent
|
Request for reset has been sent to the Cisco Unified IP phone.
|
ringing
|
1 indicates that the phone is ringing. 0 indicates that the phone is not ringing.
|
Rx Pkts
|
Number of received voice packets.
|
silent-ring
|
Silent ring has been set on this button and extension.
|
socket
|
TCP socket number used to connect to IP phone.
|
speed dial speed-tag:digit-string label-text
|
This button is a speed-dial button, assigned to the speed-dial sequence number speed-tag. It dials digit-string and displays the text label-text next to the button.
|
sub=3, sub=4
|
Subtype 3 means that one Cisco Unified IP Phone 7914 Expansion Module is attached to the main Cisco Unified IP Phones 7960 and 7960G, and subtype 4 means that two are attached.
|
Tag number
|
Dn-tag number, the unique sequence number that identifies an ephone-dn during configuration, followed by the type of ephone-dn it is.
|
TAPI Client IP Address
|
IP address of the PC running the TAPI client.
|
TCP socket
|
TCP socket number used to communicate with the Cisco Unified IP phone. This can be correlated with the output of other debug and show commands.
|
Telecaster model-number
|
Type and model of the Cisco Unified IP phone. This information is received from the phone during its registration with the router.
|
Tx Pkts
|
Number of transmitted voice packets.
|
Username
|
Username that the phone user types when logging in to the web-based Cisco Unified CME GUI.
|
Related Commands
Command
|
Description
|
show ephone-dn
|
Displays information about Cisco Unified IP phone extensions (ephone-dns).
|
show ephone login
|
Displays the login states of all local ephones.
|
show telephony-service all
|
Displays systemwide status and information for a Cisco Unified CME system.
|
show ephone cfa
To display status and information on the registered phones that have call-forward-all set on one or more of their extensions (ephone-dns), use the show ephone cfa command in privileged EXEC mode.
show ephone cfa
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
Examples
The following is sample output from the show ephone cfa command:
ephone-1 Mac:0007.0EA6.353A TCP socket:[2] activeLine:0 REGISTERED
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:1.2.205.205 52491 Telecaster 7960 keepalive 14 max_line 6
button 1: dn 11 number 60011 cfa 60022 CH1 IDLE
button 2: dn 17 number 60017 cfa 60021 CH1 IDLE
The show ephone command describes significant fields in this output.
Related Commands
Command
|
Description
|
show ephone
|
Displays statistical information about registered Cisco IP phones.
|
show ephone dn
To display phone information for specified dn-tag or for all dn-tags, use the show ephone dn command in privileged EXEC mode.
show ephone dn [dn-tag]
Syntax Description
dn-tag
|
(Optional) Unique sequence number that is used during configuration to identify a particular extension (ephone-dn).
|
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
Usage Guidelines
Use this command to identify the phone on which a particular dn-tag has been assigned.
Examples
The following is sample output for the two appearances of DN 5:
Tag 5, Normal or Intercom dn
ephone 1, mac-address 0030.94C3.CAA2, line 2
ephone 2, mac-address 0030.94c2.9919, line 3
The show ephone command describes significant fields in this output.
Related Commands
Command
|
Description
|
show ephone
|
Displays statistical information about registered Cisco IP phones.
|
show ephone dnd
To display information on the registered phones that have "do not disturb" set on one or more of their extensions (ephone-dns), use the show ephone dnd command in privileged EXEC mode.
show ephone dnd
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
Usage Guidelines
This command does not apply to Cisco Unified SRST.
Examples
The following is sample output from the show ephone dnd command:
ephone-1 Mac:0007.0EA6.353A TCP socket:[1] activeLine:0 REGISTERED
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:1.2.205.205 52486 Telecaster 7960 keepalive 2729 max_line 6 DnD
button 1: dn 11 number 60011 CH1 IDLE
The show ephone command describes significant fields in this output.
Related Commands
Command
|
Description
|
show ephone
|
Displays statistical information about registered Cisco IP phones.
|
show ephone login
To display the login states of all local IP phones, use the show ephone login command in privileged EXEC mode.
show ephone login
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
Usage Guidelines
The show ephone login command displays whether an ephone has a personal identification number (PIN) and whether its owner has logged in.
Examples
The following is sample output from the show ephone login command. It shows that a PIN is enabled for ephone 1 and that its owner has not logged in. The other phones do not have PINs associated with them.
Router# show ephone login
ephone 1 Pin enabled:TRUE Logged-in:FALSE
ephone 2 Pin enabled:FALSE
ephone 3 Pin enabled:FALSE
ephone 4 Pin enabled:FALSE
ephone 5 Pin enabled:FALSE
ephone 6 Pin enabled:FALSE
ephone 7 Pin enabled:FALSE
ephone 8 Pin enabled:FALSE
ephone 9 Pin enabled:FALSE
Table 12 describes significant fields in this output.
Table 12 show ephone login Field Descriptions
Field
|
Description
|
ephone phone-tag
|
Phone identified with its unique phone-tag sequence number.
|
Pin enabled
|
TRUE indicates that a PIN has been defined for this phone. FALSE indicates that no PIN has been defined for this phone.
|
Logged-in
|
TRUE indicates that a phone user is currently logged in on this phone. FALSE indicates that no phone user is currently logged in on this phone.
|
Related Commands
Command
|
Description
|
login (telephony-service)
|
Defines when users of IP phones in a Cisco Unified CME system are logged out automatically.
|
pin
|
Sets set a personal identification number (PIN) for an IP phone in a Cisco Unified CME system.
|
show ephone
|
Displays statistical information about registered Cisco IP phones.
|
show ephone moh
To display information about moh files in use, use the show ephone moh command in global configuration mode.
show ephone moh
Syntax Description
This command has no arguments or keywords
Command Modes
Global Configuration mode.
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
15.0(1)XA
|
Cisco Unified CME 8.0 Cisco Unified SRST 8.0
|
This command was introduced.
|
15.1(1)T
|
Cisco Unified CME 8.0 Cisco Unified SRST 8.0
|
This command was integrated into Cisco IOS Release 15.1(1)T.
|
Usage Guidelines
Use the show ephone moh to display information about the different MOH group configured. The following examples displays different MOH group configured.
Examples
Router #show ephone moh
Skinny Music On Hold Status (moh-group 1)
Active MOH clients 0 (max 830), Media Clients 0
File flash:/minuet.au (not cached) type AU Media_Payload_G711Ulaw64k 160 bytes
Moh multicast 239.10.16.6 port 2000
Skinny Music On Hold Status (moh-group 2)
Active MOH clients 0 (max 830), Media Clients 0
File flash:/audio/hello.au type AU Media_Payload_G711Ulaw64k 160 bytes
Moh multicast on 239.10.16.6 port 2000 via 0.0.0.0
Skinny Music On Hold Status (moh-group 3)
Active MOH clients 0 (max 830), Media Clients 0
File flash:/bells.au type AU Media_Payload_G711Ulaw64k 160 bytes
Moh multicast on 239.10.16.5 port 2000 via 0.0.0.0
Skinny Music On Hold Status (moh-group 4)
Active MOH clients 0 (max 830), Media Clients 0
File flash:/3003.au type AU Media_Payload_G711Ulaw64k 160 bytes
Moh multicast on 239.10.16.7 port 2000 via 0.0.0.0
Skinny Music On Hold Status (moh-group 5)
Active MOH clients 0 (max 830), Media Clients 0
File flash:/4004.au type AU Media_Payload_G711Ulaw64k 160 bytes
Moh multicast on 239.10.16.8 port 2000 via 0.0.0.0
Related Commands
Command
|
Description
|
show ephone-dn
|
Displays MOH group information for a phone directory number.
|
show ephone summary
|
Displays the information about the MOH files in use
|
show voice moh-group statistics
|
Displays the MOH subsystem statistics information
|
show ephone offhook
To display information and packet counts for the phones that are currently off hook, use the show ephone offhook command in privileged EXEC mode.
show ephone offhook
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
Examples
The following sample output is displayed when no phone is off hook:
Router# show ephone offhook
No ephone in specified type/condition.
The following sample output displays information for a phone that is off hook:
Router# show ephone offhook
ephone-5 Mac:000A.8A2C.8C6E TCP socket:[20] activeLine:1 REGISTERED
mediaActive:0 offhook:1 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:10.22.84.71 51228 Telecaster 7960 keepalive 43218 max_line 6
button 1:dn 9 number 59943 CH1 SIEZE silent-ring
button 2:dn 10 number 59943 CH1 IDLE
button 3:dn 42 number A4400 auto dial A4500 CH1 IDLE
button 4:dn 96 number 69943 auto dial 95259943 CH1 IDLE
button 5:dn 75 number 49943 auto dial 49943 CH1 IDLE
speed dial 1:57514 marketing
Active Call on DN 9 chan 1 :59943 0.0.0.0 0 to 0.0.0.0 2000 via 172.30.151.1
G711Ulaw64k 160 bytes vad
Tx Pkts 0 bytes 0 Rx Pkts 0 bytes 0 Lost 0
Jitter 0 Latency 0 callingDn -1 calledDn -1
Username:user1 Password:newuser
The following sample output displays information for a phone that has just completed a call:
Router# show ephone offhook
ephone-5 Mac:000A.8A2C.8C6E TCP socket:[20] activeLine:1 REGISTERED
mediaActive:1 offhook:1 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:10.22.84.71 51228 Telecaster 7960 keepalive 43224 max_line 6
button 1:dn 9 number 59943 CH1 CONNECTED silent-ring
button 2:dn 10 number 59943 CH1 IDLE
button 3:dn 42 number A4400 auto dial A4500 CH1 IDLE
button 4:dn 96 number 69943 auto dial 95259943 CH1 IDLE
button 5:dn 75 number 49943 auto dial 49943 CH1 IDLE
speed dial 1:57514 marketing
Active Call on DN 9 chan 1 :59943 10.23.84.71 22926 to 172.30.131.129 2000 via
172.30.151.1
G711Ulaw64k 160 bytes no vad
Tx Pkts 0 bytes 0 Rx Pkts 0 bytes 0 Lost 0
Jitter 0 Latency 0 callingDn -1 calledDn -1 (media path callID 19288 srcCallID 1
Username:user1 Password:newuser
The show ephone command describes significant fields in this output.
Related Commands
Command
|
Description
|
show ephone
|
Displays statistical information about registered Cisco IP phones.
|
show ephone overlay
To display information for the registered phones that have overlay ephone-dns associated with them, use the show ephone overlay in privileged EXEC mode.
show ephone overlay
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
Usage Guidelines
This command does not apply to Cisco Unified SRST.
Examples
The following is sample output from the show ephone overlay command:
Router# show ephone overlay
ephone-1 Mac:0007.0EA6.353A TCP socket:[1] activeLine:0 REGISTERED
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:10.2.225.205 52486 Telecaster 7960 keepalive 2771 max_line 6
button 1: dn 11 number 60011 CH1 IDLE overlay
button 2: dn 17 number 60017 CH1 IDLE overlay
button 3: dn 24 number 60024 CH1 IDLE overlay
button 4: dn 30 number 60030 CH1 IDLE overlay
button 5: dn 36 number 60036 CH1 IDLE CH2 IDLE overlay
button 6: dn 39 number 60039 CH1 IDLE CH2 IDLE overlay
overlay 1: 11(60011) 12(60012) 13(60013) 14(60014) 15(60015) 16(60016)
overlay 2: 17(60017) 18(60018) 19(60019) 20(60020) 21(60021) 22(60022)
overlay 3: 23(60023) 24(60024) 25(60025) 26(60026) 27(60027) 28(60028)
overlay 4: 29(60029) 30(60030) 31(60031) 32(60032) 33(60033) 34(60034)
overlay 5: 35(60035) 36(60036) 37(60037)
overlay 6: 38(60038) 39(60039) 40(60040)
The show ephone command describes significant fields in this output. Table 13 describes a field that is not in that table.
Table 13 show ephone overlay Field Descriptions
Field
|
Description
|
overlay number
|
Displays the contents of an overlay set, including each dn-tag and its associated extension number.
|
Related Commands
Command
|
Description
|
show ephone
|
Displays statistical information about registered Cisco IP phones.
|
show ephone phone-load
To display information about the phone firmware that is loaded on registered phones, use the show ephone phone-load command in privileged EXEC mode.
show ephone phone-load
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
Examples
The following is sample output that displays the phone firmware versions for all phones in the system:
Router# show ephone phone-load
DeviceName CurrentPhoneload PreviousPhoneload LastReset
=====================================================================
SEP0002B9AFC49F 3.2(2.14) 3.2(2.14) TCP-timeout
SEP003094C2D0B0 3.2(2.14) 3.2(2.14) TCP-timeout
SEP000C30F03707 3.2(2.14) 3.2(2.14) TCP-timeout
SEP003094C2999F 3.2(2.14) 3.2(2.14) TCP-timeout
SEP000A8A2C8C6E 3.2(2.14) 3.2(2.14) Initialized
SEP0002B9AFBB4D 3.2(2.14) 3.2(2.14) TCP-timeout
SEP00075078627F 3.2(2.14) 3.2(2.14) TCP-timeout
SEP0002FD659E59 3.2(2.14) 3.2(2.14) TCP-timeout
SEP00024BCCD626 3.2(2.14) CM-closed-TCP
SEP0008215F88C1 3.2(2.14) 3.2(2.14) TCP-timeout
SEP000C30F0390C 3.2(2.14) 3.2(2.14) TCP-timeout
SEP003094C30143 3.2(2.14) 3.2(2.14) TCP-timeout
Table 14 describes significant fields in this output.
Table 14 show ephone phone-load Field Descriptions
Field
|
Description
|
DeviceName
|
Device name.
|
CurrentPhoneLoad
|
Current phone firmware version.
|
PreviousPhoneLoad
|
Phone firmware version before last phone load.
|
LastReset
|
Reason for last reset of phone.
|
Related Commands
Command
|
Description
|
show ephone
|
Displays statistical information about registered Cisco IP phones.
|
show ephone registered
To display the status of registered phones, use the show ephone registered command in privileged EXEC mode.
show ephone registered
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
Examples
The following is sample output from the show ephone registered command:
Router# show ephone registered
ephone-2 Mac:000A.8A5C.5961 TCP socket:[1] activeLine:0 REGISTERED
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:192.168.0.50 50349 Telecaster 7940 keepalive 23738 max_line 2
button 1: dn 2 number 91450 CH1 IDLE CH2 IDLE
The show ephone command describes significant fields in this output.
Related Commands
Command
|
Description
|
show ephone
|
Displays statistical information about registered Cisco IP phones.
|
show ephone remote
To display nonlocal phones (phones with no Address Resolution Protocol [ARP] entry), use the show ephone remote command in privileged EXEC mode.
show ephone remote
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
Usage Guidelines
Phones without ARP entries are suspected not to be on the LAN. Use the show ephone remote command to identify phones without ARP entries that might have operational issues.
Examples
The following is sample output that identifies ephone 2 as not having an ARP entry:
Router# show ephone remote
ephone-2 Mac:0185.047C.993E TCP socket:[4] activeLine:0 REGISTERED
mediaActive:1 offhook:0 ringing:0 reset:0 reset_sent:0 paging 1 debug:0
IP:10.50.50.20 49231 Telecaster 7910 keepalive 112 max_line 2 dual-line
button 1:dn 3 number 95021 CH1 IDLE
The show ephone command describes significant fields in this output.
Related Commands
Command
|
Description
|
show ephone
|
Displays statistical information about registered Cisco IP phones.
|
show ephone ringing
To display information on phones that are ringing, use the show ephone ringing command in privileged EXEC mode.
show ephone ringing
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
Examples
The following is sample output from the show ephone ringing command:
Router# show ephone ringing
ephone-1 Mac:0005.5E37.8090 TCP socket:[1] activeLine:0 REGISTERED
mediaActive:0 offhook:0 ringing:1 reset:0 reset_sent:0 paging 0 debug:0
IP:10.50.50.10 49329 Telecaster 7960 keepalive 17602 max_line 6
button 1:dn 1 number 95011 CH1 RINGING CH2 IDLE
button 2:dn 2 number 95012 CH1 IDLE
The show ephone command describes significant fields in this output.
Related Commands
Command
|
Description
|
show ephone
|
Displays statistical information about registered Cisco IP phones.
|
show ephone summary
To display brief information about Cisco IP phones, use the show ephone summary command in privileged EXEC mode.
show ephone summary
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC (#)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.1(5)YD
|
Cisco CME 1.0 Cisco SRST 1.0
|
This command was introduced.
|
12.2(8)T
|
Cisco CME 2.0 Cisco SRST 2.0
|
This command was integrated into Cisco IOS Release 12.2(8)T .
|
15.0(1)XA
|
Cisco Unified CME 8.0 Cisco Unified SRST 8.0
|
This command was modified. The output was enhanced to show IPv6 or IPv4 addresses configured on ephones.
|
15.1(1)T
|
Cisco Unified CME 8.0 Cisco Unified SRST 8.0
|
This command was integrated into Cisco IOS Release 15.1(1)T.
|
15.1(2)T
|
Cisco Unified CME 8.1 Cisco Unified SRST 8.1
|
This command was modified. The output was enhanced to show voice-class stun-usage information.
|
Examples
The following is sample output from the show ephone summary command:
Router# show ephone summary
ephone-1[0] Mac:FCAC.3BAE.0000 TCP socket:[17] activeLine:0 whisperLine:0 REGISTERED
mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0
debug:0 primary_dn: 1*
IP:10.2.1.0 * SCCP Gateway (AN) keepalive 2966 music 0 1:1
voice-class stun is enabled
ephone-2[1] Mac:FCAC.3BAE.0001 TCP socket:[18] activeLine:0 whisperLine:0 REGISTERED
mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0
debug:0 primary_dn: 2*
IP:10.2.1.5 * SCCP Gateway (AN) keepalive 2966 music 0 1:2
voice-class stun is enabled
ephone-4 Mac:0030.94C3.F43A TCP socket:[-1] activeLine:0 REGISTERED
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 debug:0
IP:10.2.1.1 Telecaster 7960 keepalive 59
Max 48, Registered 1, Unregistered 0, Deceased 0, Sockets 1
Max Conferences 4 with 0 active (4 allowed)
Skinny Music On Hold Status
Active MOH clients 0 (max 72), Media Clients 0
The show ephone command describes significant fields in this output.
Related Commands
Command
|
Description
|
show ephone
|
Displays statistical information about registered Cisco IP phones.
|
show ephone tapiclients
To display status of ephone Telephony Application Programming Interface (TAPI) clients, use the show ephone tapiclients command in privileged EXEC mode.
show ephone tapiclients
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
Examples
The following is sample output from the show ephone tapiclients command:
Router# show ephone tapiclients
ephone-4 Mac:0007.0EA6.39F8 TCP socket:[2] activeLine:0 REGISTERED
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:192.168.1.18 50291 Telecaster 7960 sub=3 keepalive 728 max_line 20
button 1:dn 6 number 1004 CH1 IDLE CH2 IDLE
button 2:dn 1 number 1000 CH1 IDLE shared
button 3:dn 2 number 1000 CH1 IDLE shared
button 7:dn 3 number 1001 CH1 IDLE CH2 IDLE monitor-ring shared
button 8:dn 4 number 1002 CH1 IDLE CH2 IDLE monitor-ring shared
button 9:dn 5 number 1003 CH1 IDLE CH2 IDLE monitor-ring
button 10:dn 91 number A00 auto dial A01 CH1 IDLE
speed dial 1:2000 PAGE-STAFF
speed dial 2:2001 HUNT-STAFF
Username:userB Password:ge30qe
Username:userB status:REGISTERED Socket :[5]
Tapi Client IP address: 192.168.1.5 Port:2295
The show ephone command describes significant fields in this output.
Related Commands
Command
|
Description
|
show ephone
|
Displays statistical information about registered Cisco IP phones.
|
show ephone telephone-number
To display information for the phone associated with a specified number, use the show ephone telephone-number command in privileged EXEC mode.
show ephone telephone-number number
Syntax Description
number
|
Telephone number that is associated with an ephone.
|
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
Usage Guidelines
Use this command to find the phone on which a particular telephone number appears.
Examples
The following is sample output from the show ephone telephone-number command:
Router# show ephone telephone-number 91400
Tag 1, Normal or Intercom dn
ephone 1, mac-address 000A.0E51.19F0, line 1
The show ephone command describes significant fields in this output.
Related Commands
Command
|
Description
|
show ephone
|
Displays statistical information about registered Cisco IP phones.
|
show ephone unregistered
To display information about unregistered phones, use the show ephone unregistered command in privileged EXEC mode.
show ephone unregistered
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
Usage Guidelines
There are two ways that an ephone can become unregistered. The first way is when an ephone is listed in the running configuration but no physical device has been registered for that ephone. The second way is when an unknown device was registered at some time after the last router reboot but has since unregistered.
Examples
The following is sample output from the show ephone unregistered command:
Router# show ephone unregistered
ephone-1 Mac:0007.0E81.10F0 TCP socket:[-1] activeLine:0 UNREGISTERED
mediaActive:0 offhook:0 ringing:0 reset:0 reset_sent:0 paging 0 debug:0
IP:0.0.0.0 0 Unknown 0 keepalive 0 max_line 0
The show ephone command describes significant fields in this output.
Related Commands
Command
|
Description
|
show ephone
|
Displays statistical information about registered Cisco IP phones.
|
show ephone-dn
To display status and information for a Cisco IP phone destination number or for extensions (ephone-dns) in a Cisco Unified Communications Manager Express (Cisco Unified CME) or Cisco Unified Survivable Remote Site Telephony (SRST) environment, use the show ephone-dn command in privileged EXEC mode.
show ephone-dn [dn-tag]
Syntax Description
dn-tag
|
(Optional) For Cisco Unified CME, a unique sequence number that is used during configuration to identify a particular extension (ephone-dn).
(Optional) For Cisco Unified SRST, a destination number tag. The destination number can be from 1 to 288.
|
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.1(5)YD
|
Cisco CME 1.0 Cisco SRST 1.0
|
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco IAD2420 series.
|
12.2(2)XT
|
Cisco CME 2.0 Cisco SRST 2.0
|
This command was implemented on the Cisco 1750 and Cisco 1751.
|
12.2(8)T
|
Cisco CME 2.0 Cisco SRST 2.0
|
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725 and Cisco 3745.
|
12.2(8)T1
|
Cisco CME 2.0 Cisco SRST 2.0
|
This command was implemented on the Cisco 2600XM and Cisco 2691.
|
12.2(11)T
|
Cisco CME 2.01 Cisco SRST 2.01
|
This command was implemented on the Cisco 1760.
|
12.3(4)T
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
Examples
Cisco Unified CME
The following Cisco Unified CME sample output displays status and information for all ephone-dns:
EFXS 50/0/1 Slot is 50, Sub-unit is 0, Port is 1
Type of VoicePort is EFXS
Administrative State is UP
No Interface Down Failure
Noise Regeneration is enabled
Non Linear Processing is enabled
Non Linear Mute is disabled
Non Linear Threshold is -21 dB
Music On Hold Threshold is Set to -38 dBm
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancellation NLP mute is disabled
Echo Cancellation NLP threshold is -21 dB
Echo Cancel Coverage is set to 8 ms
Playout-delay Mode is set to adaptive
Playout-delay Nominal is set to 60 ms
Playout-delay Maximum is set to 200 ms
Playout-delay Minimum mode is set to default, value 40 ms
Playout-delay Fax is set to 300 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Call Disconnect Time Out is set to 60 s
Ringing Time Out is set to 180 s
Wait Release Time Out is set to 30 s
Region Tone is set for US
Station name None, Station number 91400
Translation profile (Incoming):
Translation profile (Outgoing):
Digit Duration Timing is set to 100 ms
EFXS 50/0/2 Slot is 50, Sub-unit is 0, Port is 2
Type of VoicePort is EFXS
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Noise Regeneration is enabled
Non Linear Processing is enabled
Non Linear Mute is disabled
Non Linear Threshold is -21 dB
Music On Hold Threshold is Set to -38 dBm
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancellation NLP mute is disabled
Echo Cancellation NLP threshold is -21 dB
Echo Cancel Coverage is set to 8 ms
Playout-delay Mode is set to adaptive
Playout-delay Nominal is set to 60 ms
Playout-delay Maximum is set to 200 ms
Playout-delay Minimum mode is set to default, value 40 ms
Playout-delay Fax is set to 300 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Call Disconnect Time Out is set to 60 s
Ringing Time Out is set to 180 s
Wait Release Time Out is set to 30 s
Region Tone is set for US
Station name None, Station number 91450
Translation profile (Incoming):
Translation profile (Outgoing):
Digit Duration Timing is set to 100 ms
Cisco Unified SRST
The following SRST sample output displays status and information for all ephone-dns:
EFXS 50/0/7 Slot is 50, Sub-unit is 0, Port is 7
Type of VoicePort is EFXS
Administrative State is UP
No Interface Down Failure
Noise Regeneration is enabled
Non Linear Processing is enabled
Non Linear Mute is disabled
Non Linear Threshold is -21 dB
Music On Hold Threshold is Set to -38 dBm
Out Attenuation is Set to 0 dB
Echo Cancellation is enabled
Echo Cancellation NLP mute is disabled
Echo Cancellation NLP threshold is -21 dB
Echo Cancel Coverage is set to 8 ms
Playout-delay Mode is set to default
Playout-delay Nominal is set to 60 ms
Playout-delay Maximum is set to 200 ms
Playout-delay Minimum mode is set to default, value 4 ms
Playout-delay Fax is set to 300 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 10 s
Interdigit Time Out is set to 10 s
Call Disconnect Time Out is set to 60 s
Ringing Time Out is set to 8 s
Wait Release Time Out is set to 30 s
Region Tone is set for US
Station name None, Station number None
Voice card specific Info Follows:
Digit Duration Timing is set to 100 ms
Table 15 describes significant fields in the output from this command.
Table 15 show ephone-dn Field Descriptions
Field
|
Description
|
Administrative State
|
Administrative (configured) state of the voice port.
|
alert
|
The number of calls that were disconnected by the far-end device when the local IP phone was in the call alerting state (for example, because the far-end phone rang but was not answered and the far-end system decided to drop the call rather than let the phone ring for too long).
|
answered (incoming)
|
The number of incoming calls that were actually answered (the phone goes off hook when ringing).
|
answered (outgoing)
|
The number of outgoing call attempts that were answered by the far end.
|
busy
|
The number of outgoing call attempts that got a busy response.
|
Call Disconnect Time Out
|
Not applicable to the Cisco IP phone.
|
called, calling
|
Extension numbers of called and calling parties.
|
Caller ID Info Follows
|
Information about the caller ID.
|
Call Ref
|
A unique per-call identifier used by the SCCP protocol. The Call Ref values are assigned sequentially within the Cisco CME-SCCP interface, so this value also indicates the total number of SCCP calls since the router was last rebooted.
|
chan
|
Channel number of an ephone-dn.
|
CODEC
|
Codec type.
|
Companding Type
|
Not applicable to the Cisco IP phone.
|
connect
|
The number of calls that were disconnected by the far-end device when the local IP phone was in the call connected state.
|
Connection Mode
|
Not applicable to the Cisco IP phone.
|
Connection Number
|
Not applicable to the Cisco IP phone.
|
Description
|
Not applicable to the Cisco IP phone.
|
Digit Duration Timing
|
Not applicable to the Cisco IP phone.
|
DN STATE
|
Ephone-dn tag number and state of the phone line associated with an extension.
|
Echo Cancellation...
|
Not applicable to the Cisco IP phone.
|
Echo Cancel Coverage
|
Not applicable to the Cisco IP phone.
|
EFXS
|
Voice port type.
|
Far-end disconnect at...
|
See connect, alert, hold, and ring.
|
Final Jitter
|
The final voice packet receive jitter reported by the IP phone at the end of the call.
|
hold
|
The number of calls that were disconnected by the far-end device when the local IP phone was in the call hold state (for example, if the caller was left on hold for too long and got tired of waiting).
|
incoming
|
The number of incoming calls presented (the phone rings).
|
In Gain
|
Not applicable to the Cisco IP phone.
|
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.
|
Last 64 far-end disconnect cause codes
|
See the Mappings of PSTN Cause Codes to SIP Event table for a list of public switch telephone network (PSTN) cause codes that can be sent as an ISDN cause information element (IE) and the corresponding Session Interface Protocol (SIP) event.
|
Latency
|
The final voice packet receive latency reported by the IP phone at the end of the call.
|
Lost
|
Number of lost packets.
|
Music On Hold Threshold
|
Not applicable to the Cisco IP phone.
|
No Interface Down Failure
|
State of the interface.
|
Noise Regeneration
|
Not applicable to the Cisco IP phone.
|
Non Linear...
|
Not applicable to the Cisco IP phone.
|
Operation State
|
Operational state of the voice port.
|
Out Attenuation
|
Not applicable to the Cisco IP phone.
|
outgoing
|
The number of outgoing call attempts.
|
Playout-delay Maximum
|
Not applicable to the Cisco IP phone.
|
Playout-delay...
|
Not applicable to the Cisco IP phone.
|
Port
|
Port number for the interface associated with the voice interface card.
|
Region Tone
|
Not applicable to the Cisco IP phone.
|
ring
|
The number of calls that were disconnected by the far-end device when the local IP phone was in the ringing state (for example, if the call was not answered and the caller hung up).
|
Ringing Time Out
|
Duration, in seconds, for which ringing is to continue if a call is not answered. Set with the timeouts ringing command.
|
Rx Pkts, bytes
|
Number of packets and bytes received during the current or last call.
|
Signal Level to phone, peak
|
For G.711 calls only, this parameter indicates the most recent voice signal level in the voice IP packets sent from the router to the IP phone. This parameter is valid only for VoIP or PSTN G.711 calls to the IP phones. This parameter is not valid for calls between local IP phones, or calls that use codecs other than G.711. The peak field indicates the peak signal level seen during the entire call.
|
Slot
|
Slot used in the voice interface card for this port.
|
Station name
|
Station name.
|
Station number
|
Station number.
|
Sub-unit
|
Subunit used in the voice interface card for this port.
|
Tx Pkts, bytes
|
Number of packets and bytes transmitted during the current call or last call.
|
Type of VoicePort
|
Voice port type.
|
VAD
|
Voice activity detection.
|
Voice card specific info
|
Information specific to the voice card.
|
VPM STATE
|
State indication for the VPM software component.
|
VTSP STATE
|
State indication for the VTSP software component.
|
Wait Release Time Out
|
Time that a voice port stays in the call-failure state while the router sends a busy tone, reorder tone, or out-of-service tone to the port.
|
The following table lists the PSTN cause codes that can be sent as an ISDN cause information element (IE) and the corresponding SIP event for each. These are the far-end disconnect cause codes listed in the output for the show ephone-dn statistics command.
Table 16 Mappings of PSTN Cause Codes to SIP Events
PSTN Cause Code
|
Description
|
SIP Event
|
1
|
Unallocated number
|
410 Gone
|
3
|
No route to destination
|
404 Not found
|
16
|
Normal call clearing
|
BYE
|
17
|
User busy
|
486 Busy here
|
18
|
No user responding
|
480 Temporarily unavailable
|
19
|
No answer from the user
|
21
|
Call rejected
|
603 Decline
|
22
|
Number changed
|
302 Moved temporarily
|
27
|
Destination out of order
|
404 Not found
|
28
|
Address incomplete
|
484 Address incomplete
|
29
|
Facility rejected
|
501 Not implemented
|
31
|
Normal unspecified
|
404 Not found
|
34
|
No circuit available
|
503 Service unavailable
|
38
|
Network out of order
|
41
|
Temporary failure
|
42
|
Switching equipment congestion
|
44
|
Requested channel not available
|
47
|
Resource unavailable
|
55
|
Incoming class barred within CUG
|
603 Decline
|
57
|
Bearer capability not authorized
|
501 Not implemented
|
58
|
Bearer capability not presently available
|
63
|
Service or option unavailable
|
503 Service unavailable
|
65
|
Bearer cap not implemented
|
501 Not implemented
|
79
|
Service or option not implemented
|
87
|
User not member of CUG
|
603 Decline
|
88
|
Incompatible destination
|
400 Bad Request
|
95
|
Invalid message
|
102
|
Recover on timer expiry
|
408 Request timeout
|
111
|
Protocol error
|
400 Bad request
|
127
|
Interworking unspecified
|
500 Internal server error
|
Any code other than those listed above
|
500 Internal server error
|
Related Commands
Command
|
Description
|
show ephone-dn callback
|
Displays information about pending callbacks in a Cisco Unified CME or a Cisco Unified SRST environment.
|
show ephone-dn loopback
|
Displays information about loopback ephone-dns that have been created in a Cisco Unified CME or a Cisco Unified SRST environment.
|
show ephone-dn statistics
|
Displays display call statistics for a Cisco IP destination or for extensions (ephone-dns) in a Cisco Unified CME or a Cisco Unified SRST environment.
|
show ephone-dn summary
|
Displays brief information about Cisco IP phone destination numbers or for extensions (ephone-dns) in a Cisco Unified CME or a Cisco Unified SRST environment.
|
show ephone-dn callback
To display information about pending callbacks in a Cisco Unified Communications Manager Express (Cisco Unified CME) or a Cisco Unified Survivable Remote Site Telephony (Cisco Unified SRST) environment, use the show ephone-dn callback command in privileged EXEC mode.
show ephone-dn callback
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
Examples
The following sample output shows a callback placed by ephone-dn 1 against ephone-dn 3. Ephone-dn 3 has its channel 1 on hold and has just seized dial tone on its channel 2.
Router# show ephone-dn callback
DN 3 (95021) CallBack pending to DN 1 (95021) for ephone-1 age 7 seconds
State for DN 3 is CH1 HOLD CH2 SIEZE
The following sample output shows a callback placed by ephone-dn 1 against ephone-dn 3. Ephone-dn 3 has a call in progress on channel 1.
Router# show ephone-dn callback
DN 3 (95021) CallBack pending to DN 1 (95021) for ephone-1 age 8 seconds
State for DN 3 is CH1 CONNECTED
Significant fields in the output from this command are described in Table 17.
Table 17 show ephone-dn callback Field Descriptions
Field
|
Description
|
DN 3 (95021) CallBack pending to DN 1 (95021)
|
Callback originator is the extension with the dn-tag 1 (in this example), and the callback has been placed on the extension with the dn-tag 3 and the number 95021.
|
age
|
Number of seconds since the callback was placed.
|
State for DN 3 is CH1... CH2...
|
Call states for channel 1 and channel 2, if any, of the extension that the callback is for.
|
Related Commands
Command
|
Description
|
show ephone-dn
|
Displays status and information for a Cisco IP phone destination number or for extensions (ephone-dns) in a Cisco Unified CME or a Cisco Unified SRST environment.
|
show ephone-dn loopback
To display information about loopback ephone-dns that have been created in a Cisco Unified Communications Manager Express (Cisco Unified CME) or a Cisco Unified Survivable Remote Site Telephony (Cisco Unified SRST) environment, use the show ephone-dn loopback command in privileged EXEC mode.
show ephone-dn loopback
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.1(5)YD
|
Cisco CME 1.0 Cisco SRST 1.0
|
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco IAD2420 series.
|
12.2(2)XT
|
Cisco CME 2.0 Cisco SRST 2.0
|
This command was implemented on the Cisco 1750 and Cisco 1751.
|
12.2(8)T
|
Cisco CME 2.0 Cisco SRST 2.0
|
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725 and Cisco 3745.
|
12.2(8)T1
|
Cisco CME 2.0 Cisco SRST 2.0
|
This command was implemented on the Cisco 2600XM and Cisco 2691.
|
12.2(11)T
|
Cisco CME 2.01 Cisco SRST 2.01
|
This command was implemented on the Cisco 1760.
|
Examples
The following example displays information for a loopback using ephone-dn 21 and ephone-dn 22:
Router# show ephone-dn loopback
LOOPBACK DN status (min 21, max 22):
DN 21 51... Loopback to DN 22 CH1 IDLE
CallingDn -1 CalledDn -1 Called Calling G711Ulaw64k
Strip NONE, Forward 2, prefix 10 retry 10 Media 0.0.0.0 0
callID 0 srcCallID 0 ssrc 0 vector 0
DN 22 11... Loopback to DN 21 CH1 IDLE
CallingDn -1 CalledDn -1 Called Calling G711Ulaw64k
Strip NONE, Forward 2, prefix 50 retry 10 Media 0.0.0.0 0
callID 0 srcCallID 0 ssrc 0 vector 0
Significant fields in the output from this command are described in Table 18, in alphabetical order.
Table 18 show ephone-dn loopback Field Descriptions
Field
|
Description
|
Called, Calling
|
Called number and calling number when there is a call present.
|
CalledDn, CallingDn
|
Ephone-dn tag numbers of the called and calling ephone-dn. Set to -1 if the call is not to or from an ephone-dn, or if there is no active call.
|
callID
|
Internal call reference. This usage is the same as in other Cisco IOS voice gateway commands.
|
DN
|
Ephone-dn tag (sequence number).
|
Forward
|
Number of digits in the original called number to forward to the other ephone-dn in the loopback-dn pair.
|
G711...
|
G711Ulaw64k indicates G.711 codec, mu-law, 64000-bit stream. G711alaw64k indicates G.711 codec, A-law, 64000-bit stream.
|
Loopback to ...
|
Indicates the opposite ephone-dn in the loopback pair and the status of that ephone-dn.
|
Media
|
IP destination address, if any, for any voice packets that are passing through the loopback DN.
|
min, max
|
Lowest and highest dn-tag numbers of ephone-dns that are configured as loopback-dns.
|
prefix
|
Digit string to add to the beginning of forwarded called numbers.
|
retry
|
Number of seconds to wait before retrying the loopback target when is it busy.
|
srcCallID
|
Internal call reference for the destination.
|
ssrc
|
Real-time transport protocol (RTP) synchronization source (SSRC) of the most recent RTP packet.
|
Strip
|
Number of leading digits to strip before forwarding to the other extension in the loopback-dn pair.
|
vector
|
The following values describe the media path for voice packets that pass through the loopback-dn:
• 0—No media path or not a loopback-dn path (inactive).
• 1—Normal path. Loopback-dn has identified the final media destination as a local IP phone. The media IP address field shows a valid, non-zero value.
• 2—Hairpin. Media packets are routed back through paired loopback-dns. The final destination is not known. For example, this can be a VoIP-to-VoIP call path by a loopback-dn.
• 3—Hairpin. The final destination is an ephone-dn in a special mode such as paging.
• 4—Loopback-dn chain has been detected, in which two loopback-dn pairs have been connected together.
• 5—Loopback-dn chain has been detected in which more than two loopback-dn pairs are connected in series.
|
Related Commands
Command
|
Description
|
loopback-dn
|
Creates a virtual loopback voice port (loopback-dn) to establish a demarcation point for VoIP voice calls and supplementary services.
|
show ephone-dn
|
Displays status and information for a Cisco IP phone destination number or for extensions (ephone-dns) in a Cisco Unified CME or a Cisco Unified SRST environment.
|
show ephone-dn statistics
To display call statistics for a Cisco IP destination or for extensions (ephone-dns) in a Cisco Unified Communications Manager Express (Cisco Unified CME) or a Cisco Unified Survivable Remote Site Telephony (Cisco Unified SRST) environment, use the show ephone-dn command in privileged EXEC mode.
show ephone-dn [dn-tag] statistics
Syntax Description
dn-tag
|
(Optional) Unique sequence number that is used during configuration to identify a particular extension (ephone-dn).
|
statistics
|
Displays voice quality statistics on calls for a specified extension or for all extensions.
|
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ1
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco CME 3.0 Cisco SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
Examples
The following sample output displays statistics for all extensions (ephone-dns) in a Cisco Unified CME system. There are two ephone-dns (DN1 and DN3) in this example.
Router# show ephone-dn statistics
Stats may appear to be inconsistent for conference or shared line cases
DN 1 chan 1 incoming 36 answered 21 outgoing 60 answered 30 busy 6
Far-end disconnect at:connect 29 alert 18 hold 7 ring 15
Last 64 far-end disconnect cause codes
17 17 17 17 17 17 16 16 16 16 16 16 16 16 16 16
16 16 16 16 65 16 65 65 65 65 16 65 65 65 16 16
16 16 16 16 16 16 16 16 16 16 16 16 16 65 47 65
47 47 16 16 16 16 16 16 16 16 16 16 16 16 16 16
DN 1 chan 1 (95011) voice quality statistics for last call
Call Ref 103 called 91500 calling 95011
Total Tx Pkts 0 bytes 0 Rx Pkts 0 bytes 0 Lost 0
Final Jitter 30 Latency 0 Lost 0
Signal Level to phone 0 (-78 dB) peak 0 (-78 dB)
Packets counted by router 0
DN 1 chan 2 incoming 0 answered 0 outgoing 1 answered 0 busy 0
Far-end disconnect at:connect 0 alert 0 hold 0 ring 0
Last 64 far-end disconnect cause codes
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
DN 1 chan 2 (95011) voice quality statistics for last call
Call Ref 86 called calling
Total Tx Pkts 0 bytes 0 Rx Pkts 0 bytes 0 Lost 0
Final Jitter 0 Latency 0 Lost 0
Signal Level to phone 0 (-78 dB) peak 0 (-78 dB)
Packets counted by router 0
DN 3 chan 1 incoming 0 answered 0 outgoing 1 answered 1 busy 0
Far-end disconnect at:connect 0 alert 0 hold 0 ring 0
Last 64 far-end disconnect cause codes
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
DN 3 chan 1 (95021) voice quality statistics for current call
Call Ref 102 called 94011 calling 95021
Current Tx Pkts 241 bytes 3133 Rx Pkts 3304 bytes 515023 Lost 0
Worst Jitter 30 Worst Latency 0
Signal Level to phone 201 (-39 dB) peak 5628 (-12 dB)
Packets counted by router 3305
The following sample output displays voice quality statistics for the ephone-dn with dn-tag 2:
Router# show ephone-dn 2 statistics
DN 2 chan 1 incoming 0 answered 0 outgoing 2 answered 0 busy 0
Far-end disconnect at: connect 0 alert 0 hold 0 ring 0
Last 64 far-end disconnect cause codes
28 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
0 0 0 0 0 0 0 0 0 0 0 0 0 0 0 0
DN 2 chan 1 (91450) voice quality statistics for last call
Call Ref 2 called calling
Total Tx Pkts 0 bytes 0 Rx Pkts 0 bytes 0 Lost 0
Final Jitter 0 Latency 0 Lost 0
Signal Level to phone 0 (-78 dB) peak 0 (-78 dB)
Packets counted by router 0
The show ephone-dn command describes significant fields in the output from this command.
Related Commands
Command
|
Description
|
show ephone-dn
|
Displays status and information for a Cisco IP phone destination number or for extensions (ephone-dns) in a Cisco Unified CME or a Cisco Unified SRST environment.
|
show ephone-dn summary
To display brief information about Cisco IP phone destination numbers or for extensions (ephone-dns) in a Cisco Unified Communications Manager Express (Cisco Unified CME) or a Cisco Unified Survivable Remote Site Telephony (Cisco Unified SRST) environment, use the show ephone-dn summary command in privileged EXEC mode.
show ephone-dn summary
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.1(5)YD
|
Cisco CME 1.0 Cisco SRST 1.0
|
This command was introduced on the Cisco 2600 series, Cisco 3600 series, and Cisco IAD2420 series.
|
12.2(2)XT
|
Cisco CME 2.0 Cisco SRST 2.0
|
This command was implemented on the Cisco 1750 and Cisco 1751.
|
12.2(8)T
|
Cisco CME 2.0 Cisco SRST 2.0
|
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725 and Cisco 3745.
|
12.2(8)T1
|
Cisco CME 2.0 Cisco SRST 2.0
|
This command was implemented on the Cisco 2600XM and Cisco 2691.
|
12.2(11)T
|
Cisco CME 2.01 Cisco SRST 2.01
|
This command was implemented on the Cisco 1760.
|
Examples
The following is example output from the show ephone-dn summary command:
Router# show ephone-dn summary
PORT DN STATE CODEC VAD VTSP STATE VPM STATE
======== ========== ======== === ===================== =========
50/0/1 DOWN - - - EFXS_ONHOOK
50/0/2 DOWN - - - EFXS_ONHOOK
50/0/3 DOWN - - - EFXS_ONHOOK
50/0/4 INVALID - - - EFXS_INIT
50/0/5 INVALID - - - EFXS_INIT
50/0/6 INVALID - - - EFXS_INIT
Table 19 describes significant fields in the output from this command.
Table 19 show ephone-dn summary Field Descriptions
Field
|
Description
|
CODEC
|
Type of codec.
|
DN STATE
|
Status of the ephone-dn.
|
EFXS
|
Voice port type.
|
PORT
|
Port number (virtual) for this interface. The number that follows the last slash in the port number is the ephone-dn tag. For example, if the port number is 50/0/1, the dn-tag is 1.
|
VAD
|
Voice activity detection status.
|
VPM STATE
|
State indication for the voice port module (VPM) software component.
|
VTSP STATE
|
State indication for the voice telephony service provider (VTSP) software component.
|
Related Commands
Command
|
Description
|
show ephone-dn
|
Displays status and information for a Cisco IP phone destination number or for extensions (ephone-dns) in a Cisco Unified CME or a Cisco Unified SRST environment.
|
show sip-ua status registrar
To display all the SIP endpoints that are currently registered with the contact address, use the show sip-ua status registrar command in privileged EXEC mode.
show sip-ua status registrar
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco SIP SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco SIP SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
15.0(1)XA
|
Cisco SIP SRST 8.0
|
This command was updated to display the signaling transport protocol.
|
Examples
The following is a sample output from this command:
Router# show sip-ua status registrar
Line destination expires(sec) contact
============ =============== ============ ===============
3029991 10.2.30.108 388 10.2.30.108
TLS 00120014-4ae40064-f1a3e9fe-8d301072@10.2.30.1
3029993 10.2.30.103 382 10.2.30.103
TCP 001bd433-1c840052-655cd596-4e992eed@10.2.30.1
3029982 10.2.30.106 406 10.2.30.106
UDP 001d452c-dbba0056-0481d321-1f3f848d@10.2.30.1
Table 20 describes the significant fields shown in this output.
Table 20 show sip-ua status registrar Field Descriptions
Field
|
Description
|
call-id
|
A unique ID assigned for each call.
|
contact
|
The contact IP address provided by the Cisco SIP IP phone.
|
destination
|
The destination IP address.
|
expires (sec)
|
The amount of time, in seconds, until registration expires.
|
Line
|
The phone number that maintains registration of SIP devices.
|
peer
|
When an SIP IP phone registers, an associated VoIP dial peer is automatically generated. This dial peer contains general information on how to contact the phone. The information includes the directory number or numbers associated with the phone and the IP address and protocol of the phone.
|
Related Commands
Command
|
Description
|
registrar server
|
Enables SIP registrar functionality.
|
show voice emergency
To display the IP address, subnet mask, and ELIN for each emergency response location, use the show voice emergency command in user EXEC or privileged EXEC mode.
show voice emergency
Syntax Description
This command has no arguments or keywords.
Command Default
No default behavior or values
Command Modes
User EXEC (>)
Privileged EXEC (#)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.4(15)XY
|
Cisco Unified CME 4.2(1) Cisco Unified SRST 4.2(1) Cisco Unified SIP SRST 4.2(1)
|
This command was introduced.
|
12.4(20)T
|
Cisco Unified CME 7.0 Cisco Unified SRST 7.0 Cisco Unified SIP SRST 7.0
|
This command was integrated into Cisco IOS Release 12.4(20)T.
|
Usage Guidelines
This command displays the IP address, subnet mask, and ELIN for each emergency response location.
Examples
The following example shows a sample output which includes IP mask and ELIN information for each ERL:
EEMERGENCY RESPONSE LOCATIONS
ERL | ELIN 1 | ELIN2 | SUBNET 1 | SUBNET 2
1 | 6045550101 | | 10.0.0.0 | 255.0.0.0
2 | 6045550102 | 6045550106 | 192.168.0.0 | 255.255.0.0
3 | | 6045550107 | 172.16.0.0 | 255.255.0.0
4 | 6045550103 | | 192.168.0.0 | 255.255.0.0
5 | 6045550105 | | 209.165.200.224 | 255.0.0.0
6 6045550198 | | 6045550109 | 209.165.201.0 | 255.255.255.224
Related Commands
Command
|
Description
|
voice emergency response location
|
Creates a tag for identifying an ERL for E911 services.
|
show voice emergency addresses
To display address information for each emergency response location, use the show emergency addresses command in user EXEC or privileged EXEC mode.
show voice emergency addresses
Syntax Description
This command has no arguments or keywords.
Command Default
No default behavior or values
Command Modes
User EXEC (>)
Privileged EXEC (#)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.4(15)XY
|
Cisco Unified CME 4.2(1) Cisco Unified SRST 4.2(1) Cisco Unified SIP SRST 4.2(1)
|
This command was introduced.
|
12.4(20)T
|
Cisco Unified CME 7.0 Cisco Unified SRST 7.0 Cisco Unified SIP SRST 7.0
|
This command was integrated into Cisco IOS Release 12.4(20)T.
|
Usage Guidelines
This command displays the physical address of each emergency response location.
Examples
The following example shows a sample output which includes physical address information for the ERL:
Router# show voice emergency addresses
3850 Zanker Rd, San Jose,604,5550101
225 W Tasman Dr, San Jose,604,5550102
275 W Tasman Dr, San Jose,604,5550103
518 Bellew Dr,Milpitas,604,5550104
400 Tasman Dr,San Jose,604,5550105
3675 Cisco Way,San Jose,604,5550106
Related Commands
Command
|
Description
|
address
|
Specifies a comma separated text entry (up to 250 characters) of an ERL's civic address.
|
show voice emergency all
|
Displays all emergency response location information.
|
voice emergency response location
|
Creates a tag for identifying an ERL for E911 services.
|
show voice emergency all
To display all emergency response location information, use the show voice emergency all command in user EXEC or privileged EXEC mode.
show voice emergency all
Syntax Description
This command has no arguments or keywords.
Command Default
No default behavior or values
Command Modes
User EXEC (>)
Privileged EXEC (#)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.4(15)XY
|
Cisco Unified CME 4.2(1) Cisco Unified SRST 4.2(1) Cisco Unified SIP SRST 4.2(1)
|
This command was introduced.
|
12.4(20)T
|
Cisco Unified CME 7.0 Cisco Unified SRST 7.0 Cisco Unified SIP SRST 7.0
|
This command was integrated into Cisco IOS Release 12.4(20)T.
|
Usage Guidelines
This command displays all information configured for each emergency response location.
Examples
The following example shows a sample output, displaying all ERL-related information for ERL 1 and 3.
VOICE EMERGENCY RESPONSE SETTINGS
Callback Number: 6045550103
Emergency Line ID Number: 6045550155
EMERGENCY RESPONSE LOCATION 1
Address: 3850 Zanker Rd, San Jose,elin.1.3,elin.4.10
IP Address 1: 209.165.200.226 IP mask 1: 255.255.255.254
IP Address 2: 209.165.202.129 IP mask 2: 255.255.0.0
Emergency Line ID 1: 6045550180
Last Caller: 6045550188 [Jan 30 2007 16:05.52 PM]
Next ELIN For Emergency Call: 6045550166
EMERGENCY RESPONSE LOCATION 3
Address: 225 W Tasman Dr, San Jose,elin.1.3,elin.4.10
IP Address 1: 209.165.202.133 IP mask 1: 255.255.0.0
IP Address 2: 209.165.202.130 IP mask 2: 255.0.0.0
Emergency Line ID 2: 6045550150
Next ELIN For Emergency Call: 6045550151
Related Commands
Command
|
Description
|
address
|
Specifies a comma separated text entry (up to 250 characters) of an ERL's civic address.
|
elin
|
Specifies a PSTN number that will replace the caller's extension.
|
name
|
Specifies a string (up to 32-characters) used internally to identify or describe the emergency response location.
|
subnet
|
Defines which IP phones are part of this ERL.
|
voice emergency response location
|
Creates a tag for identifying an ERL for the E911 services.
|
show voice emergency callers
To display a list of 911 calls made over the last three hours, use the show emergency callers command in privileged EXEC mode.
show voice emergency callers
Syntax Description
This command has no arguments or keywords.
Command Default
No list of 911 calls is displayed.
Command Modes
Privileged EXEC (#)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.4(15)T
|
Cisco Unified CME 4.1 Cisco Unified SRST 4.1 Cisco Unified SIP SRST 4.1
|
This command was introduced. For Cisco Unified CME, this command is supported in SRST fallback mode only.
|
12.4(15)XY
|
Cisco Unified CME 4.2(1) Cisco Unified SRST 4.2(1) Cisco Unified SIP SRST 4.2(1)
|
This command was added to Cisco Unified CME.
|
12.4(20)T
|
Cisco Unified CME 7.0 Cisco Unified SRST 7.0 Cisco Unified SIP SRST 7.0
|
This command was integrated into Cisco IOS Release 12.4(20)T.
|
Usage Guidelines
This command displays a list of all 911 calls made in the past three hours. The list shows the originating number, the ELIN used, and the time the call was placed.
Examples
The following example shows a sample output, which includes the originating number, the ELIN used, and the time the call was placed:
router# show voice emergency callers
EMERGENCY CALLS CALL BACK TABLE
6045550181 | 8155550151 | Oct 12 2006 04:05:21
6045550182 | 8155550152 | Oct 12 2006 04:05:21
Related Commands
Command
|
Description
|
voice emergency response location
|
Creates a tag for identifying an ERL for the enhanced 911 service.
|
show voice emergency zone
To display each emergency response zone's list of locations in the order of priority, use the show voice emergency zone command in user EXEC or privileged EXEC mode.
show voice emergency zone
Syntax Description
This command has no arguments or keywords.
Command Default
No default behavior or values
Command Modes
User EXEC (>)
Privileged EXEC (#)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.4(15)XY
|
Cisco Unified CME 4.2(1) Cisco Unified SRST 4.2(1) Cisco Unified SIP SRST 4.2(1)
|
This command was introduced.
|
12.4(20)T
|
Cisco Unified CME 7.0 Cisco Unified SRST 7.0 Cisco Unified SIP SRST 7.0
|
This command was integrated into Cisco IOS Release 12.4(20)T.
|
Usage Guidelines
This command displays a list of the locations, in priority order, of all configured emergency response zones.
Examples
The following example shows a sample output which displays the ERL locations for emergency response zones 90 and 100.
Related Commands
Command
|
Description
|
location
|
Identifies locations within an emergency response zone.
|
voice emergency response location
|
Creates a tag for identifying an ERL for the enhanced 911 service.
|
voice emergency response zone
|
Creates an emergency response zone within which ERLs can be grouped.
|
show voice moh-group
To display information about voice moh-groups, use the show voice moh-group command in in privileged EXEC mode.
show voice moh-group
Syntax Description
This command has no arguments or keywords
Command Modes
Privileged EXEC (#)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
15.0(1)XA
|
Cisco Unified CME 8.0 Cisco Unified SRST 8.0
|
This command was introduced.
|
15.1(1)T
|
Cisco Unified CME 8.0 Cisco Unified SRST 8.0
|
This command was integrated into Cisco IOS Release 15.1(1)T.
|
Examples
The following sample output shows general information about voice moh-groups in Cisco Unifiede CME or Cisco Unified SRST.
Router# show voice moh-group
description this moh group is for sales
multicast moh 239.1.1.1 port 16386 route 239.1.1.3 239.1.1.3
extension-range 1000 to 1999
extension-range 2000 to 2999
extension-range 3000 to 3999
extension-range 20000 to 22000
extension-range A1000 to A1999
description (not configured)
multicast moh 239.23.4.10 port 2000
extension-range 7000 to 7999
extension-range 8000 to 8999
description This is for marketing
multicast moh 239.15.10.1 port 3000
extension-range 9000 to 9999
description (not configured)
multicast moh 239.16.12.1 port 4000
extension-range 10000 to 19999
description (not configured)
multicast moh 239.12.1.2 port 5000
extension-range ABCD to DECF
extension-range 0012 to 0024
extension-range 0934 to 0964
=== Total of 5 voice moh-groups ===
e
Examples
Command
|
Description
|
showcall-manager-fallback all
|
Displays the detailed configuration of all Cisco IP phones, directory numbers, voice ports, and dial peers in your network during Cisco Unified Communications Manager fallback.
|
show ephone summary
|
Displays the information about the MOH files in use
|
show voice moh-group statistics
|
Displays the MOH subsystem statistics information
|
show voice moh-group statistics
To display the MOH subsystem statistics information, use the show voice moh-group command in privileged EXEC mode.
show voice moh-group statistics
Syntax Description
This command has no arguments or keywords
Command Modes
Privileged EXEC (#)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
15.0(1)XA
|
Cisco Unified CME 8.0 Cisco Unified SRST 8.0
|
This command was introduced.
|
15.1(1)T
|
Cisco Unified CME 8.0 Cisco Unified SRST 8.0
|
This command was integrated into Cisco IOS Release 15.1(1)T.
|
Examples
In the following example, the MOH Group Streaming Interval Timing Statistics shows the media packet counts during streaming intervals.
Each packet counter is of 32 bit size and holds a count limit of 4294967296 intervals. This means that with 20 milliseconds packet interval (for G.711), the counters restart from 0 any time after 2.72 years (2 years 8 months). You must use the e clear voice moh-group statistics once in every two years to reset the packet counters.
MOH Group Packet Transmission Timing Statistics shows the maximum and minimum amount of time (in microseconds) taken by the MOH groups to send out media packets.
The MOH Group Loopback Interval Timing Statistics is available when loopback interface is configured as part of the multicast MOH routes in Cisco Unified SRST . These counts are loopback packet counts within certain streaming timing intervals.
router# show voice moh-group statistics
MOH Group Streaming Interval Timing Statistics:
Grp# ~19 msec 20~39 40~59 60~99 100~199 200+ msec
==== ========== ========== ========== ========== ========== ==========
0: 25835 17559966 45148 0 0 1
1: 19766 17572103 39079 0 0 1
2: 32374 17546886 51687 0 0 1
3: 27976 17555681 47289 0 0 1
4: 34346 17542940 53659 0 0 1
5: 14971 17581689 34284 0 0 1
MOH Group Packet Transmission Timing Statistics:
==== ========== ==========
MOH Group Loopback Interval Timing Statistics:
loopback event array: svc_index=1542, free_index=1549, max_q_depth=31
Grp# ~19 msec 20~39 40~59 60~99 100~199 200+ msec
==== ========== ========== ========== ========== ========== ==========
0: 8918821 8721527 10023 0 1 1
1: 9007373 8635813 7184 0 1 1
2: 8864760 8772851 12758 0 1 1
3: 8924447 8715457 10464 0 1 1
4: 8858393 8778957 13017 0 1 1
5: 9005511 8639936 4919 0 1 1
Statistics collect time: 4 days 2 hours 5 minutes 39 seconds.
Related Commands
Command
|
Description
|
show ephone-dn
|
Displays MOH group information for a phone directory number.
|
show ephone summary
|
Displays the information about the MOH files in use
|
show voice moh-group
|
Displays the MOH groups configured
|
show voice register all
To display all Cisco Unified Session Initiation Protocol (SIP) Survivable Remote Site Telephony (SRST) or Cisco Unified Communications Manager Express (Cisco Unified CME) configurations and register information, use the show voice register all command in privileged EXEC mode.
show voice register all
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco SIP SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco SIP SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
12.4(4)T
|
Cisco CME 3.4 and Cisco SIP SRST 3.4
|
This command was added to Cisco CME.
|
15.0(1)XA
|
Cisco SIP SRST 8.0
|
This command was updated to display the signaling transport protocol.
|
15.1(2)T
|
Cisco Unified CME 8.1 Cisco Unified SIP SRST 8.1
|
This command was modified. The output display was modified.
|
Examples
Cisco Unified SIP SRST
The following is an example of show voice register all command:
Router# show voice register all
Outbound-proxy is enabled and will use global configured value
Security Policy: DEVICE-DEFAULT
network-locale[0] US (This is the default network locale for this box)
user-locale[0] US (This is the default user locale for this box)
user-locale[4] US Active registrations : 0
Total SIP phones registered: 0
Total Registration Statistics
Registration requests : 0
after last unregister : 0
Last register request time :
Last unregister request time :
Unregister success time :
Pool 1 has this DN configured for line 1
Pool 2 has this DN configured for line 1
Pool 3 has this DN configured for line 1, 2
Pool 1 has this DN configured for line 4
Pool 1 has this DN configured for line 3
Mac address is 001B.535C.D410
Proxy Ip address is 0.0.0.0
Reason for unregistered state:
No registration request since last reboot/unregister
Total SIP phones registered: 0
Total Registration Statistics
Registration requests : 0
after last unregister : 0
Last register request time :
Last unregister request time :
Unregister success time :
Mac address is 0015.C68E.6D13
Proxy Ip address is 0.0.0.0
Reason for unregistered state:
No registration request since last reboot/unregister
Total SIP phones registered: 0
Total Registration Statistics
Registration requests : 0
after last unregister : 0
Last register request time :
Last unregister request time :
Unregister success time :
Mac address is 0021.5553.8998
Proxy Ip address is 0.0.0.0
Reason for unregistered state:
No registration request since last reboot/unregister
Total SIP phones registered: 0
Total Registration Statistics
Registration requests : 0
after last unregister : 0
Last register request time :
Last unregister request time :
Unregister success time :
Cisco Unified CME
The following is an example of show voice register all command :
Router# show voice register all
1) show voice register all
Outbound-proxy is enabled and will use global configured value
Security Policy: DEVICE-DEFAULT
Source-address is 8.3.3.5 port 5060
Mwi registration for full E.164 is disabled
Forwarding local is enabled
Privacy-on-hold is disabled
Dst auto adjust is enabled
start at Apr week 1 day Sun time 02:00
stop at Oct week 8 day Sun time 02:00
ef (the MS 6 bits, 46, in ToS, 0xB8) for media
cs3 (the MS 6 bits, 24, in ToS, 0x60) for signal
af41 (the MS 6 bits, 34, in ToS, 0x88) for video
default (the MS 6 bits, 0, in ToS, 0x0) for service
Generate text file is disabled
Tftp files are created, current syncinfo 0001140473454008
OS79XX.TXT is not created
network-locale[0] US (This is the default network locale for this box)
user-locale[0] US (This is the default user locale for this box)
user-locale[4] US Active registrations : 0
Total SIP phones registered: 0
Total Registration Statistics
Registration requests : 0
after last unregister : 0
Last register request time :
Last unregister request time :
Unregister success time :
Pool 1 has this DN configured for line 1
call-forward b2bua noan 999 timeout 8
Pool 2 has this DN configured for line 1
Pool 7 has this DN configured for line 1
call-forward b2bua all 87687
Pool 3 has this DN configured for line 1, 2
Pool 1 has this DN configured for line 4
call-forward b2bua all 678
Pool 1 has this DN configured for line 3
Attended Transfer is enabled
Blind Transfer is enabled
Semi-attended Transfer is enabled
Caller-ID block is disabled
Anonymous call block is disabled
Pool 4 has this template configured
Template 1 has this dialplan configured
Pool 4 has this dialplan configured
Mac address is 001B.535C.D410
Proxy Ip address is 0.0.0.0
Busy trigger per button value is 0
call-forward phone all is 4566
call-forward b2bua all 4555
keep-conference is enabled
service-control mechanism is not supported
Privacy feature is not configured.
Privacy button is disabled
Reason for unregistered state:
No registration request since last reboot/unregister
Total SIP phones registered: 0
Total Registration Statistics
Registration requests : 0
after last unregister : 0
Last register request time :
Last unregister request time :
Unregister success time :
Mac address is 0015.C68E.6D13
Proxy Ip address is 0.0.0.0
Busy trigger per button value is 0
call-forward phone noan is 9886, timeout 98
keep-conference is enabled
username pool2 password lab
service-control mechanism is not supported
Privacy feature is not configured.
Privacy button is disabled
Reason for unregistered state:
No registration request since last reboot/unregister
Total SIP phones registered: 0
Total Registration Statistics
Registration requests : 0
after last unregister : 0
Last register request time :
Last unregister request time :
Unregister success time :
Mac address is 0021.5553.8998
Proxy Ip address is 0.0.0.0
Busy trigger per button value is 0
call-forward phone all is 45112
call-forward b2bua all 45111
keep-conference is enabled
service-control mechanism is not supported
Privacy feature is not configured.
Privacy button is disabled
Reason for unregistered state:
No registration request since last reboot/unregister
Total SIP phones registered: 0
Total Registration Statistics
Registration requests : 0
after last unregister : 0
Last register request time :
Last unregister request time :
Unregister success time :
Mac address is 8989.9867.8769
Proxy Ip address is 0.0.0.0
Busy trigger per button value is 0
keep-conference is enabled
service-control mechanism is not supported
Privacy feature is not configured.
Privacy button is disabled
Reason for unregistered state:
No registration request since last reboot/unregister
Total SIP phones registered: 0
Total Registration Statistics
Registration requests : 0
after last unregister : 0
Last register request time :
Last unregister request time :
Unregister success time :
Mac address is 0018.BAC8.D2B1
Proxy Ip address is 0.0.0.0
Busy trigger per button value is 0
keep-conference is enabled
service-control mechanism is not supported
Privacy feature is not configured.
Privacy button is disabled
Reason for unregistered state:
No registration request since last reboot/unregister
Total SIP phones registered: 0
Total Registration Statistics
Registration requests : 0
after last unregister : 0
Last register request time :
Last unregister request time :
Unregister success time :
Table 21 describes the significant fields shown in this output.
Table 21 show voice register all Field Descriptions
Field
|
Description
|
Pool Tag
|
Used with the all and pool keywords. Shows the assigned tag number of the current pool.
|
Config
|
Used with the all and pool keywords. Shows the voice register pool.
|
Network address and Mask
|
Used with the all and pool keywords. Shows network address and mask information if the id command is configured.
|
Number list, Pattern, and Preference
|
Used with the all and pool keywords. Shows the number command configuration.
|
Proxy IP address
|
Used with the all and pool keywords. Shows the proxy command configuration.
|
Default preference
|
Used with the all and pool keywords. Shows the default preference value of this pool.
|
Incoming called number
|
Used with the all and pool keywords. Shows the incoming called-number command configuration.
|
Translate outgoing called tag
|
Used with the all and pool keywords. Shows the translate-outgoing command configuration.
|
Class of Restriction List Tag
|
Used with the all and pool keywords. Shows the COR tag.
|
Incoming corlist name
|
Used with the all and pool keywords. Shows the cor command configuration.
|
Application
|
Used with the all and pool keywords. Shows the application command configuration for this pool.
|
Dialpeers created:
|
Used with the all and pool keywords. What follows is a list of all dial peers created and their contents. Dial-peer contents differ per application and are not described here.
|
Statistics
|
Used with the all, pool, and statistics keywords. Shows the registration statistics for this pool.
|
Active registrations
|
Used with the all, pool, and statistics keywords. Shows the current active registrations.
|
Total Registration Statistics
|
Used with the all, pool, and statistics keywords. Shows the total registration statistics for this pool.
|
Registration requests
|
Used with the all, pool, and statistics keywords. Shows the incoming registration requests.
|
Registration success
|
Used with the all, pool, and statistics keywords. Shows the successful registrations.
|
Registration failed
|
Used with the all, pool, and statistics keywords. Shows the failed registrations.
|
unRegister requests
|
Used with the all, pool, and statistics keywords. Shows the incoming unregister/registration expire requests.
|
unRegister success
|
Used with the all, pool, and statistics keywords. Reports the number of successful unregisters.
|
unRegister failed
|
Used with the all, pool, and statistics keywords. Reports the number of failed unregisters.
|
Related Commands
Command
|
Description
|
show sip-ua status registrar
|
Displays all the SIP endpoints currently registered with the contact address.
|
show voice register dial-peers
|
Displays details of all dynamically created VoIP dial peers associated with the Cisco Unified SIP SRST or Cisco Unified CME register event
|
show voice register pool
|
Displays all configuration information associated with a particular voice register pool.
|
show voice register dial-peers
To display details of all dynamically created VoIP dial peers associated with the Cisco Unified Session Initiation Protocol (SIP) Survivable Remote Site Telephony (SRST) or Cisco Unified CallManager Express (Cisco Unified CME) register event, use the show voice register dial-peers command in privileged EXEC mode.
show voice register dial-peers [pool tag]
Syntax Description
pool tag
|
Number of entries in attempted registrations table. Size range from 0 to 50.
|
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco SIP SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco SIP SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
12.4(4)T
|
Cisco CME 3.4 Cisco SIP SRST 3.4
|
This command was added to Cisco CME.
|
15.1(2)T
|
Cisco Unified CME 8.1 Cisco Unified SIP SRST 8.1
|
This command was modified. Pool tag keyword- argument was added. Command output display was also modified to display dial-peers specific to a pool.
|
Usage Guidelines
Use this command to display the dial-peers associated with a pool. To display the dynamic dial-peers associated with a specific pool, use the pool keyword followed by the pool tag. When using the pool keyword, you must specify the pool tag.
Examples
Cisco Unified CME adn Cisco Unified SIP SRST
The following is a sample output from this command displaying all dial-peers:
Router#show voice register dial-peers
Dial-peers for Pool 1
dial-peer voice 40001 voip
destination-pattern 45111
session target ipv4:8.3.3.111:5060
session protocol sipv2
call-fwd-all 4555
after-hours-exempt FALSE
dial-peer voice 40002 voip
destination-pattern 45113
session target ipv4:8.33.33.111:5060
session protocol sipv2
after-hours-exempt FALSE
Dial-peers for Pool 2
dial-peer voice 40003 voip
destination-pattern 45112
session target ipv4:8.33.33.112:5060
session protocol sipv2
call-fwd-noan-timeou 8
call-fwd-noan 999
after-hours-exempt TRUE
Cisco Unified CME and Cisco Unified SRST
The following is a sample output from this command displaying all statistical information related to pool 1:
Router# show voice register dial-peers pool 1
Dial-peers for Pool 1:
dial-peer voice 40004 voip
destination-pattern 1000
redirect ip2ip
session target ipv4:9.13.18.40:19633
session protocol sipv2
dtmf-relay rtp-nte sip-notify
digit collect kpml
codec g711ulaw bytes 160
after-hours-exempt FALSE
dial-peer voice 40001 voip
destination-pattern 2000
redirect ip2ip
session target ipv4:9.13.18.40:19634
session protocol sipv2
dtmf-relay rtp-nte sip-notify
digit collect kpml
codec g711ulaw bytes 160
after-hours-exempt FALSE
Related Commands
Command
|
Description
|
show sip-ua status registrar
|
Displays all the SIP endpoints currently registered with the contact address.
|
show voice register all
|
Displays all Cisco Unified SIP SRST and Cisco Unified CME configurations and register information.
|
show voice register pool
|
Displays all configuration information associated with a particular voice register pool.
|
show voice register dn
To display all configuration information associated with a specific voice register dn, use the show voice register dn command in privileged EXEC mode.
show voice register dn {tag | all}
Syntax Description
tag
|
Tag number of the voice register dn for which to display information. Range is 1 to 750.
|
all
|
(Optional) Displays configuration information associated with all voice register dns defined in a system.
|
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Version
|
Modification
|
12.4(4)T
|
Cisco CME 3.4 Cisco SIP SRST 3.4
|
This command was introduced.
|
15.1(2)T
|
Cisco CME 8.1 Cisco SIP SRST 8.1
|
This command was modified.The display output now shows pools that have DNs configured under them. All keyword was added to show configuration information for all voice register dns defined in system.
|
Usage Guidelines
In Cisco Unified CME 8.1 and Cisco Unified SIP SRST 8.1, the show voice register dn command displays the pools that have the DNs configured under them. When used with all keyword, the show voice register dn command displays configuration information for all the DNs defined in a system.
Examples
Cisco Unified SIP CME
The following is a sample output from this command:
Router# show voice register dn 1
Dn Tag 1
Config:
Number is 11
Preference is 10
Huntstop is enabled
Translation-profile incoming saaa
Allow watch is enabled
Pool 1 has this DN configured for line 1
Cisco Unified SIP SRST
The following is a sample output from this command:
Router# show voice register dn 2
Dn Tag 1
Config:
Number is 11
Preference is 10
Huntstop is enabled
Translation-profile incoming saaa
Allow watch is enabled
Pool 1 has this DN configured for line 1
Cisco Unified SIP SRST
The following is a sample output from this command displaying information for all the dns:
Translation-profile incoming saaa
Pool 1 has this DN configured for line 1
Pool 2 has this DN configured for line 1, 2
Cisco Unified SIP CME
The following is a sample output from this command displaying information for all the dns:
Router# show voice register dn all
call-forward b2bua noan 999 timeout 8
Pool 2 has this DN configured for line 1
Pool 7 has this DN configured for line 1
call-forward b2bua all 87687
call-forward b2bua all 87687
Pool 1 has this DN configured for line 1
Pool 3 has this DN configured for line 1, 2
Pool 1 has this DN configured for line 4
call-forward b2bua all 678
Pool 1 has this DN configured for line 3
Table 27 contains descriptions of significant fields shown in this output, listed in alphabetical order.
Table 22 show voice register dn Field Descriptions
Field
|
Description
|
Auto answer
|
Status of auto-answer feature defined with the auto-answer command.
|
Config
|
List of configuration options defined for this voice register dn.
|
Dn Tag
|
Tag number of the requested voice register dn.
|
Huntstop
|
Status of huntstop behavior defined with the huntstop command.
|
Number
|
Telephone or extension number set with the number command in voice register dn configuration mode.
|
Preference
|
Preference order set with the preference command in voice register dn configuration mode.
|
Related Commands
Command
|
Description
|
show voice register pool
|
Displays all configuration information associated with a particular voice register pool.
|
show voice register dn all
|
Displays information associated with all the dns configured in a system.
|
voice register dn
|
Enters voice register dn configuration mode to define an extension for a SIP phone line.
|
show voice register global
To display all global configuration parameters associated with SIP phones, use the show voice register global command in privileged EXEC mode.
show voice register global
Syntax Description
This command has no arguments or keywords.
Command Default
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.4(4)T
|
Cisco CME 3.4 Cisco SIP SRST 3.4
|
This command was introduced.
|
15.0(1)XA
|
Cisco SIP SRST 8.0
|
This command was updated to display the signaling transport protocol.
|
15.1(2)T
|
Cisco Unified CME 8.1 Cisco Unified SIP SRST 8.1
|
This command was modified.The output display now includes global statistics.
|
Examples
Cisco Unified CME
The following is sample output from this command:
Router# show voice register global
Outbound-proxy is enabled and will use global configured value
Security Policy: DEVICE-DEFAULT
Source-address is 8.3.3.5 port 5060
Mwi registration for full E.164 is disabled
Forwarding local is enabled
Privacy-on-hold is disabled
Dst auto adjust is enabled
start at Apr week 1 day Sun time 02:00
stop at Oct week 8 day Sun time 02:00
ef (the MS 6 bits, 46, in ToS, 0xB8) for media
cs3 (the MS 6 bits, 24, in ToS, 0x60) for signal
af41 (the MS 6 bits, 34, in ToS, 0x88) for video
default (the MS 6 bits, 0, in ToS, 0x0) for service
Generate text file is disabled
Tftp files are created, current syncinfo 0001140473454008
OS79XX.TXT is not created
network-locale[0] US (This is the default network locale for this box)
user-locale[0] US (This is the default user locale for this box)
user-locale[4] US Active registrations : 0
Total SIP phones registered: 0
Total Registration Statistics
Registration requests : 0
after last unregister : 0
Last register request time :
Last unregister request time :
Unregister success time :
Cisco Unified SIP SRST
Router# show voice register global
Outbound-proxy is enabled and will use global configured value
Security Policy: DEVICE-DEFAULT
network-locale[0] US (This is the default network locale for this box)
user-locale[0] US (This is the default user locale for this box)
user-locale[4] US Active registrations : 0
Total SIP phones registered: 0
Total Registration Statistics
Registration requests : 0
after last unregister : 0
Last register request time :
Last unregister request time :
Unregister success time :
Table 23 contains descriptions of significant fields shown in this output, listed in alphabetical order.
Table 23 show voice register global Field Descriptions
Field
|
Description
|
Date-format
|
Value of date-format command.
|
DST auto adjust
|
Setting of dst auto-adjust command.
|
Forwarding local
|
Setting of forwarding local command.
|
Generate text file
|
Setting of text file command.
|
Hold-alert
|
Setting of hold-alert command.
|
Load
|
Value of load command.
|
Max-dn
|
Reports the maximum number of SIP voice register directory numbers (dns) supported by the Cisco Unified SIP CME or Cisco Unified SIP SRST router as configured with the max-dn command. The maximum possible number is platform-dependent.
|
Max-pool
|
Reports the maximum number of SIP voice register pools supported by the Cisco Unified SIP SRST or Cisco Unified CME router as configured with the max-pool command. The maximum possible number is platform-dependent.
|
Max redirect number
|
Maximum number of redirects set with the max-redirect command.
|
Mode
|
Reports the mode as configured with the mode command. Value can be either Cisco Unified CME or Cisco Unified SIP SRST.
|
MWI registration
|
Setting of mwi command.
|
MWI stutter
|
Setting of mwi stutter command.
|
Time-format
|
Value of time-format command.
|
Time-zone
|
Number of the timezone selected with the timezone command.
|
TFTP path
|
Directory location of provisioning files for SIP phones that is specified with the tftp-path command.
|
Version
|
Reports the Cisco Unified SIP SRST or Cisco Unified CME version number.
|
Related Commands
Command
|
Description
|
show sip-ua status registrar
|
Displays all the SIP endpoints currently registered with the contact address.
|
show voice register all
|
Displays all Cisco Unified SIP SRST and Cisco Unified CME configurations and register information.
|
show voice register dial-peers
|
Displays details of all dynamically created VoIP dial peers associated with the Cisco Unified SIP SRST or Cisco Unified CME register event.
|
voice register global
|
Enters voice register global configuration mode in order to set global parameters for all supported Cisco SIP phones in a Cisco Unified CME or Cisco Unified SIP SRST environment.
|
show voice register pool
To display all configuration information associated with a specific voice register pool, use the show voice register pool command in privileged EXEC mode.
show voice register pool {pool-tag | all }[brief]
Syntax Description
pool-tag
|
Tag number of the voice register pool for which information is displayed. Range is 1 to 262.
Note The maximum number of pools is version and platform dependent.
|
all
|
Displays the information of all the voice register pools.
|
brief
|
(Optional) Displays brief information of all voice register pools.
|
Command Modes
Privileged EXEC (#)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco SIP SRST
|
This command was introduced.
|
12.3(4)T
|
Cisco SIP SRST
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
12.4(4)T
|
Cisco CME 3.4 Cisco SIP SRST 3.4
|
This command was added to Cisco CME.
|
12.4(15)XY
|
Cisco Unified CME 4.2(1) Cisco Unified SIP SRST 4.2(1)
|
This command was modified to include emergency response location information in the output display.
|
12.4(20)T
|
Cisco Unified CME 7.0 Cisco Unified SIP SRST 7.0
|
This command was integrated into Cisco IOS Release 12.4(20)T.
|
15.0(1)XA
|
Cisco Unified CME 8.0
|
This command was modified to include logical partitioning class of restriction (LPCOR) information in the output display.
|
15.1(1)T
|
Cisco Unified CME 8.0
|
This command was integrated into Cisco IOS Release 15.1(1)T.
|
15.1(2)T
|
Cisco Unified CME 8.1
|
This command was modified. The all and brief keywords were added. Voice-class stun-usage information is displayed in the output.
|
15.2(2)T
|
Cisco Unified CME 9.0
|
This command was modified to include conference admin, conference add mode, and conference drop mode in the output display.
|
15.2(4)M
|
Cisco Unified CME 9.1 Cisco Unified SIP SRST 9.1
|
This command was modified to include Key Expansion Module (KEM) data in the output display.
|
Examples
Cisco Unified CME
The following is a sample output of the show voice register pool command, displaying information for voice register pool 33 in Cisco Unified CME:
Router# show voice register pool 33
Mac address is 0009.B7F7.532E
Proxy Ip address is 0.0.0.0
Busy trigger per button value is 0
keep-conference is enabled
Emergency response location 3
Lpcor Incoming is sip_group
Lpcor Outgoing is sip_group
service-control mechanism is not supported
Privacy feature is not configured.
Privacy button is disabled
Total SIP phones registered: 0
Total Registration Statistics
Registration requests : 0
The following is a sample output of the show voice register pool command. The output shows that a meet-me hardware conference administrator has been assigned, the conference creator or any of the participants can add a new participant, and the conference creator can terminate the active video hardware conference by hanging up.
Router# show voice register pool 15
Mac address is 1C17.D340.81F0
Proxy Ip address is 0.0.0.0
Current Phone load version is Cisco-CP9951/9.0.1
DTMF Relay is enabled, sip-notify
Busy trigger per button value is 0
keep-conference is enabled
registration expires timer max is 86400 and min is 60
service-control mechanism is supported
registration Call ID is 1c17d340-81f00002-6c48fe8e-03013c10@1.5.40.105
Registration method: per line
Privacy feature is not configured.
Privacy button is disabled
active primary line is: 3915
contact IP address: 1.5.40.105 port 5060
conference drop mode: creator
paging-dn: config 0 [multicast] effective 0 [multicast]
The following is an example of a partial output of the show voice register pool all command, showing KEM data with the phone type information:
Router# show voice register pool all
Mac address is B4A4.E328.4698
Type is 9971 addon 1 CKEM
Proxy Ip address is 0.0.0.0
Busy trigger per button value is 0
keep-conference is enabled
registration expires timer max is 200 and min is 60
The following is a sample output of the show voice register pool all command, showing the three KEMs configured with phone type 9971:
Router# show voice register pool all
Mac address is B4A4.E328.4698
Type is 9971 addon 1 CKEM 2 CKEM 3 CKEM
Cisco Unified SIP SRST
The following is a sample output of the show voice register pool command, displaying all information for voice register pool 1 in Cisco Unified SIP SRST:
Router# show voice register pool 1
Network address is 192.168.0.0, Mask is 255.255.0.0
Number list 1 : Pattern is 50.., Preference is 2
Proxy Ip address is 0.0.0.0
Incoming called number is
Translate outgoing called tag is 1
Class of Restriction List Tag: default
Incoming corlist name is allowall
Application is default.new
dial-peer voice 40007 voip
corlist incoming allowall
incoming called-number 5001
session target ipv4:192.168.0.3
translate-outgoing called 1
Total Registration Statistics
Registration requests : 48
Registration success : 48
Emergency response location 6
Voice class stun usage
The following is a sample output of the show voice register pool command, displaying voice-class stun-usage information for voice register pool 51:
Router# show voice register pool 51
Mac address is 0011.209F.5D60
Proxy Ip address is 0.0.0.0
Current Phone load version is Cisco-SIPGateway/IOS-12.x
Busy trigger per button value is 0
keep-conference is enabled
service-control mechanism is not supported
registration Call ID is 2BA38EE3-17D311DB-800BCD81-A9AD11F0
Privacy feature is not configured.
Privacy button is disabled
active primary line is: 16263646
contact IP address: 192.168.0.87 port 5060
Reason for unregistered state:
No registration request since last reboot/unregister
voice-class stun-usage is enabled. tag is 1
Total SIP phones registered: 0
Total Registration Statistics
Registration requests : 2
after last unregister : 0
Last register request time : 13:43:27.839 IST Tue Apr 20 2010
Table 24 contains descriptions of significant fields shown in the Cisco Unified CME and Cisco Unified SIP SRST output, listed in alphabetical order.
Table 24 show voice register pool Field Descriptions
Field
|
Description
|
Active registrations
|
Shows the current active registrations.
|
Application
|
Shows the application command configuration for this pool.
|
Call Waiting
|
Shows the call-waiting command configuration.
|
Class of Restriction List Tag
|
Shows the COR tag.
|
Conference add mode
|
Shows the current setting of the hardware conference privilege for adding participants.
|
Conference admin
|
Shows whether the Cisco Unified SIP IP phone is assigned as the hardware conference administrator or not.
|
Conference drop mode
|
Shows who can terminate an active ad-hoc hardware conference by hanging up.
|
Config
|
Shows the voice register pool.
|
Default preference
|
Shows the default preference value of this pool.
|
Dialpeers created
|
Lists all the dial peers created and their contents. Dial-peer contents differ for each application and are not described here.
|
DnD
|
Shows the setting of the dnd-control command.
|
DTMF Relay
|
Shows the setting of the dtmf-relay command.
|
Emergency response location
|
Shows the ephone's emergency response location to which an emergency response team is dispatched when an emergency call is made.
|
Incoming called number
|
Shows the incoming called-number command configuration.
|
Incoming corlist name
|
Shows the cor command configuration.
|
keep-conference
|
Shows the status of the keep-conference command.
|
Lpcor Incoming
|
Shows the setting of the lpcor incoming command.
|
Lpcor Outgoing
|
Shows the setting of the lpcor outgoing command.
|
Lpcor Type
|
Shows the setting of the lpcor type command.
|
Mac address
|
Shows the MAC address of the Cisco Unified SIP IP phone as defined by the id command.
|
Network address and Mask
|
Shows network address and mask information when the id command is configured.
|
Number list, Pattern, and Preference
|
Shows the number command configuration.
|
Pool Tag
|
Shows the assigned tag number of the current pool.
|
Proxy IP address
|
Shows the proxy command configuration; that is, the IP address of the external SIP server.
|
Registration failed
|
Shows the failed registrations.
|
Registration requests
|
Shows the incoming registration requests.
|
Registration success
|
Shows the successful registrations.
|
Statistics
|
Shows the registration statistics for this pool.
|
Template
|
Shows the template-tag number for the template applied to the Cisco Unified SIP IP phone.
|
Total Registration Statistics
|
Shows the total registration statistics for this pool.
|
Translate outgoing called tag
|
Shows the translate-outgoing command configuration.
|
Type
|
Shows the phone type identified for the Cisco Unified SIP IP phone using the type command.
|
unRegister failed
|
Reports the number of failed unregisters.
|
unRegister requests
|
Shows the incoming unregister/registration expiry requests.
|
unRegister success
|
Reports the number of successful unregisters.
|
Username Password
|
Shows the values within the authentication credential.
|
Related Commands
Command
|
Description
|
application (voice register pool)
|
Selects the session-level application for the dial peer associated with an individual Cisco Unified SIP IP phone in a Cisco Unified CME environment or for a group of phones in a Cisco Unified SIP SRST environment.
|
call-waiting (voice register pool)
|
Enables the call-waiting option on a SIP phone.
|
cor (voice register pool)
|
Configures a class of restriction on the VoIP dial peers associated with directory numbers.
|
dnd-control (voice register template)
|
Enables the Do-Not-Disturb (DND) soft key on SIP phones.
|
dtmf-relay (voice register pool)
|
Specifies the list of dual-tone multifrequency (DTMF) relay methods that can be used to relay DTMF audio tones between SIP endpoints.
|
id (voice register pool)
|
Explicitly identifies a locally available, individual Cisco Unified SIP IP phone or, when running Cisco Unified SIP SRST, a set of Cisco Unified SIP IP phones.
|
incoming called-number (dial peer)
|
Specifies a digit string that can be matched by an incoming call to associate the call with a dial peer.
|
keep-conference (voice register pool)
|
Allows IP phone conference initiators to exit from conference calls and keep the remaining parties connected.
|
lpcor incoming
|
Associates an incoming call with a logical partitioning class of restriction (LPCOR) resource-group policy.
|
lpcor outgoing
|
Associates an outgoing call with an LPCOR resource-group policy.
|
lpcor type
|
Specifies the LPCOR type for an IP phone.
|
number (voice register pool)
|
Indicates the E.164 phone numbers that the registrar permits to handle the Register message from a Cisco Unified SIP IP phone.
|
proxy (voice register pool)
|
Autogenerates additional VoIP dial peers to reach the main proxy whenever a Cisco Unified SIP IP phone registers with a Cisco Unified SIP SRST gateway.
|
show sip-ua status registrar
|
Displays all the Cisco Unified SIP IP phones registered with the contact address.
|
show voice register all
|
Displays all Cisco Unified SIP SRST and Cisco Unified CME configurations and register information.
|
show voice register dial-peer
|
Displays details of all dynamically created VoIP dial peers associated with the Cisco Unified CME or Cisco Unified SIP SRST register event.
|
translate-outgoing (voice register pool)
|
Allows an explicit setting of translation rules on the VoIP dial peer to modify a phone number dialed by any Cisco Unified IP phone user.
|
type (voice register pool)
|
Defines a phone type for a SIP phone.
|
voice register pool
|
Enters voice register pool configuration mode for Cisco Unified SIP IP phones.
|
show voice register pool after-hour-exempt
To display the details of a phone that has after-hour-exempt enabled on it, use the show voice register after-hour-exempt command in privileged EXEC mode.
show voice register after-hour-exempt
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Version
|
Modification
|
15.1(2)T
|
Cisco Unified CME 8.1 Cisco Unified SRST 8.1
|
This command was introduced.
|
Usage Guidelines
Use this command to display the details of a phone that has after-hour-exempt enabled. Individual phones can be exempted from call blocking using the after-hour exempt.
Examples
Cisco Unified CME
The following is a sample output from this command displaying information for phones with after after-hour-exempt:
Router# show voice register pool after-hour-exempt
Pool ID IP Address Ln DN Number State
==== =============== =============== == === ==================== ============
1 001B.535C.D410 8.3.3.111 3 8 UNREGISTERED
2 0015.C68E.6D13 1 2 45112 UNREGISTERED
3 0021.5553.8998 1 3 45113 UNREGISTERED
7 0018.BAC8.D2B1 1 2 45112 UNREGISTERED
Cisco Unified SRST
The following is a sample output from this command displaying information for phones with after after-hour-exempt:
Router# show voice register pool after-hour-exempt
Pool ID IP Address Ln DN Number State
==== =============== =============== == === ==================== ============
1 9.13.18.40 9.13.18.40 1 1 1000 REGISTERED
Table 27 contains descriptions of significant fields shown in this output, listed in alphabetical order.
Table 25 show voice register pool after-hour exempt field descriptions
Field
|
Description
|
DN
|
Directory number of the phone.
|
IP Address/port
|
IP address and port number of the phones.
|
LN
|
Line number of the phone.
|
Number
|
Number of the phones that have after-hour exempt enabled.
|
Pool
|
Shows the current pool.
|
State
|
Registration state.
|
Related Commands
Command
|
Description
|
after-hour exempt(voice register pool)
|
Specifies that an IP phone does not have any of its outgoing calls blocked although call blocking is defined.
|
show voice register all
|
Displays all Cisco SIP SRST and Cisco CME configurations and register information.
|
show voice register pool
|
Displays all configuration information associated with a particular voice register pool.
|
voice register pool
|
Enters voice register pool configuration mode for SIP phones.
|
show voice register pool attempted-registrations
To display the details of phones that attempt to register with Cisco Unified CME or Cisco Unified SRST and fail, use the show voice register pool attempted-registrations command in privileged EXEC mode.
show voice register pool attempted-registrations
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Version
|
Modification
|
15.1(2)T
|
Cisco Unified CME 8.1 Cisco Unified SRST 8.1
|
This command was introduced.
|
Usage Guidelines
Use this command to display the details of the phones that attempt to register with Cisco Unified CME or Cisco Unified SRST and fail. If the phone registers successfully after some time, the attempted registration entry will still show up in the attempted-registration table. Use the clear voice register attempted-registrations command to remove the entry from the attempted registration table.
Examples
Cisco Unified CME and Cisco Unified SRST
The following is a sample output from this command displaying information for show voice register pool attempted-registrations:
Router# show voice register pool attempted-registrations
Phones that have attempted registrations and have failed:
MAC address: 001b.535c.d410
IP address : 8.3.3.111
Attempts : 5
Time of first attempt : *10:49:51.542 UTC Wed Oct 14 2009
Time of latest attempt: *10:50:00.886 UTC Wed Oct 14 2009
Reason for failure :
No pool match for the registration request
MAC address: 0015.c68e.6d13
IP address : 8.33.33.112
Attempts : 4
Time of first attempt : *10:49:53.418 UTC Wed Oct 14 2009
Time of latest attempt: *10:50:00.434 UTC Wed Oct 14 2009
Reason for failure :
No pool match for the registration request
MAC address: 0009.43E9.0B35
IP address : 9.13.40.83
Attempts : 1
Time of first attempt : *10:49:57.866 UTC Wed Oct 14 2009
Time of latest attempt: *10:49:57.866 UTC Wed Oct 14 2009
Reason for failure :
No pool match for the registration request
The following is a sample output from this command displaying information for show voice register pool attempted-registrations when none of the phones fail:
Router# show voice register pool attempted-registrations
Phones that have attempted registrations and have failed: NONE
Related Commands
Command
|
Description
|
attempted-registrations size
|
Allows to set the size of the table that stores information related to SIP phones that attempt to register and fail.
|
clear voice register attempted-registrations
|
Clears entries from the attempted-registration table.
|
show voice register pool connected
To display the details of SIP phones that are in connected state, use the show voice register pool connected command in privileged EXEC mode.
show voice register pool connected [brief]
Syntax Description
brief
|
(Optional) Displays brief details of SIP phones that are in connected state.
|
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
15.1(2)T
|
Cisco Unified CME 8.1 Cisco Unified SRST 8.1
|
This command was introduced.
|
Usage Guidelines
Use this command to display the details of the phone that are currently in connected state (in conversation). The output for show voice register pool connected command shows details of both calls originating from the SIP phones and calls made towards SIP phones. When used with brief keyword, the show voice register pool connected command displays a brief detail of phones in connected state.
Cisco Unified CME and Cisco Unified SRST
The following is sample output from this command displaying all statistical information:
Router# show voice register pool connected
Outbound calls from SIP line phones:
Pool tag: 1
==============
MAC Address : 001B.535C.D410
Contact IP : 8.3.3.111
Phone Number : 45111
Remote Number : 45112
Call 2
SIP Call ID : 001b535c-d4100010-79612b5a-336b0db5@8.3.3.111
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 45111
Called Number : 45112
Bit Flags : 0xC0401C 0x100 0x4
CC Call ID : 7
Source IP Address (Sig ): 8.3.3.5
Destn SIP Req Addr:Port : [8.3.3.111]:5060
Destn SIP Resp Addr:Port: [8.3.3.111]:50076
Destination Name : 8.3.3.111
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 7
Stream Type : voice-only (0)
Stream Media Addr Type : 1
Negotiated Codec : g729r8 (20 bytes)
Codec Payload Type : 18
Negotiated Dtmf-relay : inband-voice
Dtmf-relay Payload Type : 0
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [8.3.3.5]:17580
Media Dest IP Addr:Port : [8.3.3.111]:26298
Options-Ping ENABLED:NO ACTIVE:NO
Inbound calls to SIP line phones:
Pool tag: 2
==============
MAC Address : 0015.C68E.6D13
Contact IP : 8.33.33.112
Phone Number : 45112
Remote Number : 45111
Call 3
SIP Call ID : 4DA52F97-ADA311DE-8019803A-FF3E4CBC@8.3.3.5
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 45111
Called Number : 45112
Bit Flags : 0xC04018 0x100 0x80
CC Call ID : 8
Source IP Address (Sig ): 8.3.3.5
Destn SIP Req Addr:Port : [8.33.33.112]:5060
Destn SIP Resp Addr:Port: [8.33.33.112]:5060
Destination Name : 8.33.33.112
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 8
Stream Type : voice-only (0)
Stream Media Addr Type : 1
Negotiated Codec : g729r8 (20 bytes)
Codec Payload Type : 18
Negotiated Dtmf-relay : inband-voice
Dtmf-relay Payload Type : 0
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [8.3.3.5]:16384
Media Dest IP Addr:Port : [8.33.33.112]:30040
The following is sample output from this command displaying brief statistical information:
Router# show voice register pool connected brief
Pool IP Address Number Remote Number
==== =============== ==================== ====================
1 8.3.3.111 45111 45112
Inbound calls to SIP line phones:
Pool IP Address Number Remote Number
==== =============== ==================== ====================
2 8.33.33.112 45112 45111
Related Commands
Command
|
Description
|
show sip-ua calls
|
Displays active user agent client (UAC) and user agent server (UAS) information on SIP calls
|
show voice register all
|
Displays all Cisco Unified SIP SRST and Cisco Unified CME configurations and register information.
|
show voice register pool
|
Displays all configuration information associated with a particular voice register pool.
|
show voice register pool ip
To display the details of a SIP phone with a specific IP address, use the show voice register pool ip command in privileged EXEC mode.
show voice register pool ip ip-address
Syntax Description
ip-address
|
IPv4 address of the SIP phone .
|
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
15.1(2)T
|
Cisco Unified CME 8.1 Cisco Unified SRST 8.1
|
This command was introduced.
|
Usage Guidelines
Use this command to display the details of a phone with a specific IP-address. When the pool ID is configured as a mac address or an IP address the registered pools contain the IP address information. The pool information is displayed if the IP addresses match.
When the pool ID is IP and the pool is unregistered, IP address configured under pool is compared with the input IP. When the pool ID is network contact, the IP address of each phone that is registered is compared with the input IP address.
Examples
Cisco Unified CME and Cisco Unified SRST
The following is sample output from this command displaying all statistical information:
Router# show voice register pool ip 8.3.3.111
Pool ID IP Address Ln DN Number State
==== =============== =============== == === ==================== ============
1 001B.535C.D410 8.3.3.111 1 1 45111 REGISTERED
4 7 451110 UNREGISTERED
Table 27 contains descriptions of significant fields shown in this output, listed in alphabetical order.
Table 26 show voice register pool ip field descriptions
Field
|
Description
|
DN
|
Voice register DN tag of the line.
|
ID
|
Phone identification (ID) address.
|
IP Address
|
IP address of the SIP phone.
|
LN
|
Line number of the telephone number.
|
Number
|
Number of the phones that have a mac address.
|
Pool
|
Tag ID of the pool.
|
State
|
Registration state of the line.
|
Related Commands
Command
|
Description
|
show voice register all
|
Displays all Cisco Unified SIP SRST and Cisco Unified CME configurations and register information.
|
show voice register pool
|
Displays all configuration information associated with a particular voice register pool.
|
show voice register pool mac
To display the details of voice register pool associated with a specific phone type, use the show voice register pool mac command in privileged EXEC mode.
show voice register pool mac H.H.H
Syntax Description
H.H.H
|
MAC address of the SIP phone attempting to register.
|
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
15.1(2)T
|
Cisco Unified CME 8.1 Cisco Unified SRST 8.1
|
This command was introduced.
|
Usage Guidelines
Use this command to display the details of the phone with the mac address H.H.H. The command displays only the pools that are configured with an ID as mac.
Examples
Cisco Unified CME and Cisco Unified SRST
The following is sample output from this command displaying all statistical information:
Router# show voice register pool mac 001B.535C.D410
Pool ID IP Address Ln DN Number State
==== =============== =============== == === ==================== ============
1 001B.535C.D410 8.3.3.111 1 1 45111 REGISTERED
4 7 451110 UNREGISTERED
Table 27 contains descriptions of significant fields shown in this output, listed in alphabetical order.
Table 27 show voice register pool mac field descriptions
Field
|
Description
|
DN
|
Voice register DN tag of the line.
|
ID
|
Phone identification (ID) address.
|
IP Address
|
IP address of the SIP phone.
|
LN
|
Line number of the telephone number.
|
Number
|
Number of the phones that have a mac address.
|
Pool
|
Tag ID of the pool.
|
State
|
Registration state of the line.
|
Related Commands
Command
|
Description
|
show voice register all
|
Displays all Cisco Unified SIP SRST and Cisco Unified CME configurations and register information.
|
show voice register pool
|
Displays all configuration information associated with a particular voice register pool.
|
show voice register pool network
To display the details of a phone with a specific network address, use the show voice register pool network command in privileged EXEC mode.
show voice register pool network network-address
Syntax Description
network-address
|
Network address of the SIP phone.
|
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
15.1(2)T
|
Cisco Unified SRST 8.1
|
This command was introduced.
|
Usage Guidelines
Use this command to display the details of pools that have network ID configured and whose network address matches the specific network address provided by the user.
Examples
The following is sample output from this command displaying all statistical information:
Router# show voice register pool network 78.89.0.0
Pool ID IP Address Ln DN Number State
==== =============== =============== == === ==================== ============
7 78.89.0.0 1 1 6576 UNREGISTERED
Table 27 contains descriptions of significant fields shown in this output, listed in alphabetical order.
Table 28 show voice register pool network field descriptions
Field
|
Description
|
DN
|
Directory number of the phone.
|
ID
|
Phone identification (ID) address.
|
IP Address
|
IP address and port number of the phones
|
LN
|
Line number of the phone.
|
Number
|
Number of the phone that have network address.
|
Pool
|
Shows the current pool.
|
State
|
Registration state.
|
Related Commands
Command
|
Description
|
show voice register all
|
Displays all Cisco Unified SIP SRST and Cisco Unified CME configurations and register information.
|
show voice register pool
|
Displays all configuration information associated with a particular voice register pool.
|
show voice register pool ip
|
Displays the voice register pool details of a phone with a specific IP address.
|
show voice register pool on-hold
To display the details of phones that are currently on-hold, use the show voice register pool oh-hold command in privileged EXEC mode.
show voice register pool on-hold [brief]
Syntax Description
brief
|
(Optional) Displays brief details of SIP phones that are currently on-hold.
|
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Version
|
Modification
|
15.1(2)T
|
Cisco Unified CME 8.1 Cisco Unified SRST 8.1
|
This command was introduced.
|
Usage Guidelines
Use this command to display the details of the phone that are currently on-hold. The show voice register pool on-hold command output also displays a field to show if the hold was a locally initiated hold (initiated on the phone) or if the hold was initiated on the remote end. When used with brief keyword, the show voice register pool on-hold command displays a brief information of the phones that are currently put on hold by the remote caller or have put the remote caller on hold. The "Hold-Origin" field specifies the type of the hold, which can be either remote or local. Local indicates that the call is placed on hold by the local phone and remote indicates that call is placed on hold by the remote phone. In case of double-hold, the hold origin will display the value "Local and Remote".
Examples
Cisco Unified CME and Cisco Unified SRST
The following is a sample output from this command displaying information for phones ringing in a voice register pool:
Router# show voice register pool on-hold brief
Outbound calls from SIP line phones:
Pool IP Address Number Remote Number Hold Origin
==== =============== ==================== ==================== ==============
1 8.3.3.111 45111 45112 Remote & Local
Inbound calls to SIP line phones:
Pool IP Address Number Remote Number Hold Origin
==== =============== ==================== ==================== ==============
2 8.33.33.112 45112 45111 Remote & Local
Cisco Unified CME and Cisco Unified SRST
The following is a sample output from this command displaying information for phones on-hold:
Router# show voice register pool on-hold
Outbound calls from SIP line phones:
Pool tag: 1
==============
MAC Address : 001B.535C.D410
Contact IP : 8.3.3.111
Phone Number : 45111
Remote Number : 45112
Local Hold : CALL HOLD Pressed on SIP Phone
Call 4
SIP Call ID : 001b535c-d4100010-79612b5a-336b0db5@8.3.3.111
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 45111
Called Number : 45112
Bit Flags : 0xC0401C 0x10100 0x4
CC Call ID : 7
Source IP Address (Sig ): 8.3.3.5
Destn SIP Req Addr:Port : [8.3.3.111]:5060
Destn SIP Resp Addr:Port: [8.3.3.111]:50076
Destination Name : 8.3.3.111
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 7
Stream Type : voice-only (0)
Stream Media Addr Type : 1
Negotiated Codec : g729r8 (20 bytes)
Codec Payload Type : 18
Negotiated Dtmf-relay : inband-voice
Dtmf-relay Payload Type : 0
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [8.3.3.5]:17580
Media Dest IP Addr:Port : [8.3.3.111]:26298
Options-Ping ENABLED:NO ACTIVE:NO
Inbound calls to SIP line phones:
Pool tag: 2
==============
MAC Address : 0015.C68E.6D13
Contact IP : 8.33.33.112
Phone Number : 45112
Remote Number : 45111
Remote Hold : SIP Phone has received CALL HOLD
Call 5
SIP Call ID : 4DA52F97-ADA311DE-8019803A-FF3E4CBC@8.3.3.5
State of the call : STATE_ACTIVE (7)
Substate of the call : SUBSTATE_NONE (0)
Calling Number : 45111
Called Number : 45112
Bit Flags : 0xC04018 0x4100 0x80
CC Call ID : 8
Source IP Address (Sig ): 8.3.3.5
Destn SIP Req Addr:Port : [8.33.33.112]:5060
Destn SIP Resp Addr:Port: [8.33.33.112]:5060
Destination Name : 8.33.33.112
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 8
Stream Type : voice-only (0)
Stream Media Addr Type : 1
Negotiated Codec : g729r8 (20 bytes)
Codec Payload Type : 18
Negotiated Dtmf-relay : inband-voice
Dtmf-relay Payload Type : 0
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [8.3.3.5]:16384
Media Dest IP Addr:Port : [8.33.33.112]:30040
Options-Ping ENABLED:NO ACTIVE:NO
Related Commands
Command
|
Description
|
show voice register all
|
Displays all Cisco SIP SRST and Cisco CME configurations and register information.
|
show sip-ua calls
|
Displays active user agent client (UAC) and user agent server (UAS) information on SIP calls
|
show voice register pool
|
Displays all configuration information associated with a particular voice register pool.
|
show voice register pool registered
To display the details of phones that successfully register to Cisco Unified Communications Manager Express (Cisco Unified CME), use the show voice register pool registered command in privileged EXEC mode.
show voice register pool registered
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC (#)
Command History
Cisco IOS Release
|
Version
|
Modification
|
15.1(2)T
|
Cisco Unified CME 8.1 Cisco Unified SRST 8.1
|
This command was introduced.
|
15.2(4)M
|
Cisco Unified CME 9.1 Cisco Unified SIP SRST 9.1
|
This command was modified to display Key Expansion Module (KEM) details with the phone type information.
|
Usage Guidelines
Use the show voice register pool registered command to display the details of phones that are successfully registered to Cisco Unified CME and Cisco Unified Survivable Remote Site Telephony (Cisco Unified SRST).
Examples
Cisco Unified CME
The following is a sample output displaying information for a registered voice register pool in Cisco Unified CME:
Router# show voice register pool registered
Mac address is 001B.535C.D410
Proxy Ip address is 0.0.0.0
Current Phone load version is Cisco-CP7960G/8.0
Busy trigger per button value is 0
call-forward phone all is 4566
call-forward b2bua all 4555
keep-conference is enabled
service-control mechanism is supported
registration Call ID is 001b535c-d410790d-17a6877e-5d04bbc5@8.3.3.111
Privacy feature is not configured.
Privacy button is disabled
active primary line is: 45111
contact IP address: 8.3.3.111 port 5060
dial-peer voice 40001 voip
destination-pattern 45111
session target ipv4:8.3.3.111:5060
Total SIP phones registered: 1
Total Registration Statistics
Registration requests : 1
after last unregister : 0
Last register request time : *11:40:32.263 UTC Wed Oct 14 2009
Last unregister request time :
Register success time : *11:40:32.267 UTC Wed Oct 14 2009
Unregister success time :
The following is a sample output displaying information for a registered voice register pool with a Cisco Unified 9971 Session Initiation Protocol (SIP) IP phone attached to a Cisco SIP IP Phone CKEM 36-Button Line Expansion Module:
Router# show voice register pool registered
Mac address is B4A4.E328.4698
Type is 9971 addon 1 CKEM
Proxy Ip address is 0.0.0.0
Busy trigger per button value is 0
keep-conference is enabled
registration expires timer max is 200 and min is 60
Cisco Unified SRST
The following is a sample output displaying information for a registered voice register pool in Cisco
Unified SRST:
Router# show voice register pool registered
Ip address is 9.13.18.40, Mask is 255.255.0.0
Proxy Ip address is 0.0.0.0
DTMF Relay is enabled, rtp-nte, sip-notify
dial-peer voice 40004 voip
session target ipv4:9.13.18.40:19633
dtmf-relay rtp-nte sip-notify
dial-peer voice 40001 voip
session target ipv4:9.13.18.40:19634
dtmf-relay rtp-nte sip-notify
dial-peer voice 40002 voip
session target ipv4:9.13.18.40:19635
dtmf-relay rtp-nte sip-notify
dial-peer voice 40003 voip
session target ipv4:9.13.18.40:19636
dtmf-relay rtp-nte sip-notify
Total SIP phones registered: 1
Total Registration Statistics
Registration requests : 4
after last unregister : 0
Last register request time : .05:22:55.604 UTC Tue Oct 6 2009
Last unregister request time :
Register success time : .05:22:55.604 UTC Tue Oct 6 2009
Unregister success time :
Table 29 contains descriptions of significant fields shown in the show voice register pool registered command output, listed in alphabetical order.
Table 29 show voice register pool registered Field Descriptions
Field
|
Description
|
Active registrations
|
Shows the current active registrations.
|
Application
|
Shows the application command configuration for this pool.
|
Call Waiting
|
Shows the setting of the call-waiting command.
|
Class of Restriction List Tag
|
Shows the COR tag.
|
Config
|
Shows the voice register pool.
|
Current phone-load
|
Shows the current version of the phone load.
|
Default preference
|
Shows the default preference value of this pool.
|
Dialpeers created
|
Results in a list of all dial peers created and their contents. Dial-peer contents differ for each application and are not described here.
|
DnD
|
Shows the setting of the dnd-control command.
|
DTMF Relay
|
Shows the setting of the dtmf-relay command.
|
Emergency response location
|
Shows the ephone's emergency response location to which an emergency response team is dispatched when an emergency call is made.
|
Incoming called number
|
Shows the incoming called-number command configuration.
|
Incoming corlist name
|
Shows the cor command configuration.
|
keep-conference
|
Shows the status of the keep-conference command.
|
Lpcor Incoming
|
Shows the setting of the lpcor incoming command.
|
Lpcor Outgoing
|
Shows the setting of the lpcor outgoing command.
|
Lpcor Type
|
Shows the setting of the lpcor type command.
|
Mac address
|
Shows the MAC address of this SIP phone as defined by the id command.
|
Network address and Mask
|
Shows network address and mask information when the id command is configured.
|
Number list, Pattern, and Preference
|
Shows the number command configuration.
|
Pool Tag
|
Shows the assigned tag number of the current pool.
|
Previous phone-load
|
Shows the version of the previous phone load.
|
Proxy IP address
|
Shows the proxy command configuration; that is, the IP address of the external SIP server.
|
Registration failed
|
Shows the failed registrations.
|
Registration requests
|
Shows the incoming registration requests.
|
Registration success
|
Shows the successful registrations.
|
Statistics
|
Shows the registration statistics for this pool.
|
statistics time-stamps
|
Shows the registration statistics for this pool with specific time stamps.
|
Template
|
Shows the template-tag number for the template applied to this SIP phone.
|
Total Registration Statistics
|
Shows the total registration statistics for this pool.
|
Translate outgoing called tag
|
Shows the translate-outgoing command configuration.
|
Type
|
Shows the phone type identified for this SIP phone using the type command.
|
unRegister failed
|
Reports the number of failed unregisters.
|
unRegister requests
|
Shows the incoming unregister/registration expiry requests.
|
unRegister success
|
Reports the number of successful unregisters.
|
Username Password
|
Shows the values within the authentication credential.
|
Related Commands
Command
|
Description
|
application (voice register pool)
|
Selects the session-level application for the dial peer associated with an individual Cisco Unified SIP IP phone in a Cisco Unified CME environment or for a group of phones in a Cisco Unified SIP SRST environment.
|
call-waiting (voice register pool)
|
Enables the call-waiting option on a SIP phone.
|
cor (voice register pool)
|
Configures a class of restriction on the VoIP dial peers associated with directory numbers.
|
dnd-control (voice register template)
|
Enables the Do-Not-Disturb (DND) soft key on SIP phones.
|
dtmf-relay (voice register pool)
|
Specifies the list of dual-tone multifrequency (DTMF) relay methods that can be used to relay DTMF audio tones between SIP endpoints.
|
id (voice register pool)
|
Explicitly identifies a locally available, individual Cisco Unified SIP IP phone or, when running Cisco Unified SIP SRST, a set of Cisco Unified SIP IP phones.
|
incoming called-number (dial peer)
|
Specifies a digit string that can be matched by an incoming call to associate the call with a dial peer.
|
keep-conference (voice register pool)
|
Allows IP phone conference initiators to exit from conference calls and keep the remaining parties connected.
|
lpcor incoming
|
Associates an incoming call with a logical partitioning class of restriction (LPCOR) resource-group policy.
|
lpcor outgoing
|
Associates an outgoing call with an LPCOR resource-group policy.
|
lpcor type
|
Specifies the LPCOR type for an IP phone.
|
number (voice register pool)
|
Indicates the E.164 phone numbers that the registrar permits to handle the Register message from a Cisco Unified SIP IP phone.
|
proxy (voice register pool)
|
Autogenerates additional VoIP dial peers to reach the main proxy whenever a Cisco Unified SIP IP phone registers with a Cisco Unified SIP SRST gateway.
|
show voice register all
|
Displays all Cisco Unified SIP SRST and Cisco Unified CME configurations and register information.
|
show voice register dial-peers
|
Displays details of all dynamically created VoIP dial peers associated with the Cisco Unified SIP SRST or Cisco Unified CME register event.
|
show voice register pool
|
Displays all configuration information associated with a particular voice register pool.
|
show voice register pool unregistered
|
Displays the details of voice register pools that do not have any phones registered.
|
translate-outgoing (voice register pool)
|
Allows an explicit setting of translation rules on the VoIP dial peer to modify a phone number dialed by any Cisco Unified IP phone user.
|
type (voice register pool)
|
Defines a phone type for a SIP phone.
|
voice register pool
|
Enters voice register pool configuration mode for SIP phones.
|
show voice register pool remote
To display the details of phones that are at a remote location, use the show voice register pool remote command in privileged EXEC mode.
show voice register pool remote
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Version
|
Modification
|
15.1(2)T
|
Cisco Unified CME 8.1 Cisco Unified SRST 8.1
|
This command was introduced.
|
Usage Guidelines
Use this command to display the details of the phones that are at remote location and do not have an address resolution protocol (ARP) entry.If the pool id is MAC or IP, the entire pool detail is displayed in a brief format. If the pool id is network, only the line details with remote contact IP address are displayed. In Cisco Unified SRST, if the pool id is IP and if the pool is not registered, the configured IP is checked to see if it is a remote IP.
Examples
Cisco Unified CME
The following is a sample output from this command displaying information for remote phones:
Router# show voice register pool remote
Pool ID IP Address Ln DN Number State
==== =============== =============== == === ==================== ============
1 001B.535C.D410 8.3.3.111 1 1 45111 REGISTERED
3 8 UNREGISTERED
4 7 451110 UNREGISTERED
2 8.3.3.112 1 2 45112 REGISTERED
3 8.3.0.0 1 3 45113 REGISTERED
Cisco Unified SRST
The following is a sample output from this command displaying information for remote phones:
Router# show voice register pool remote
Pool ID IP Address Ln DN Number State
==== =============== =============== == === ==================== ============
1 001B.535C.D410 8.33.33.111 1 1 45111 REGISTERED
3 8 UNREGISTERED
4 7 451110 UNREGISTERED
2 8.33.33.112 8.33.33.112 1 2 45112 REGISTERED
3 8.3.0.0 8.3.44.116 1 3 45113 REGISTERED
Table 27 contains descriptions of significant fields shown in this output, listed in alphabetical order.
Table 30 show voice register pool telephone number field descriptions
Field
|
Description
|
DN
|
Directory number of the phone.
|
ID
|
Phone identification (ID) address.
|
IP Address
|
IP address and port number of the phones
|
LN
|
Line number of the phone.
|
Number
|
Number of the phones.
|
Pool
|
Shows the current pool.
|
State
|
Registration state.
|
Related Commands
Command
|
Description
|
show voice register all
|
Displays all Cisco SIP SRST and Cisco CME configurations and register information.
|
show voice register dial-peer
|
Displays details of all dynamically created VoIP dial peers associated with the Cisco SIP SRST or Cisco CME register event.
|
show voice register pool
|
Displays all configuration information associated with a particular voice register pool.
|
voice register pool
|
Enters voice register pool configuration mode for SIP phones.
|
show voice register pool ringing
To display the details of phones that are currently in ringing state, use the show voice register pool ringing command in privileged EXEC mode.
show voice register pool ringing [brief]
Syntax Description
brief
|
(Optional) Displays brief details of SIP phones that are currently in ringing state.
|
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Version
|
Modification
|
15.1(2)T
|
Cisco Unified CME 8.1 Cisco Unified SRST 8.1
|
This command was introduced.
|
Usage Guidelines
Use this command to display the details of the phone that are currently in ringing state. When used with the brief keyword, the show voice register pool ringing brief command only displays information related to calls that are bound towards the SIP phones.
Examples
Cisco Unified CME and Cisco Unified SRST
The following is a sample output from this command displaying information for phones ringing in a voice register pool:
Router# show voice register pool ringing brief
Pool IP Address Number Remote Number
==== =============== ==================== ====================
2 8.33.33.112 45112 45111
Cisco Unified CME and Cisco Unified SRST
The following is a sample output from this command displaying information for phones ringing in a voice register pool:
Router# show voice register pool ringing
Pool tag: 2
==============
MAC Address : 0015.C68E.6D13
Contact IP : 8.33.33.112
Phone Number : 45112
Remote Number : 45111
Call 1
SIP Call ID : C0B5DA7-ADA311DE-8011803A-FF3E4CBC@8.3.3.5
State of the call : STATE_RECD_PROCEEDING (4)
Substate of the call : SUBSTATE_PROCEEDING_PROCEEDING (2)
Calling Number : 45111
Called Number : 45112
Bit Flags : 0xC00018 0x100 0x280
CC Call ID : 5
Source IP Address (Sig ): 8.3.3.5
Destn SIP Req Addr:Port : [8.33.33.112]:5060
Destn SIP Resp Addr:Port: [8.33.33.112]:5060
Destination Name : 8.33.33.112
Number of Media Streams : 1
Number of Active Streams: 1
RTP Fork Object : 0x0
Media Mode : flow-through
Media Stream 1
State of the stream : STREAM_ACTIVE
Stream Call ID : 5
Stream Type : voice+dtmf (1)
Stream Media Addr Type : 1
Negotiated Codec : No Codec (0 bytes)
Codec Payload Type : 255 (None)
Negotiated Dtmf-relay : inband-voice
Dtmf-relay Payload Type : 0
QoS ID : -1
Local QoS Strength : BestEffort
Negotiated QoS Strength : BestEffort
Negotiated QoS Direction : None
Local QoS Status : None
Media Source IP Addr:Port: [8.3.3.5]:16882
Related Commands
Command
|
Description
|
show sip-ua calls
|
Displays active user agent client (UAC) and user agent server (UAS) information on SIP calls
|
show voice register all
|
Displays all Cisco SIP SRST and Cisco CME configurations and register information.
|
show voice register pool
|
Displays all configuration information associated with a particular voice register pool.
|
show voice register pool telephone-number
To display the details of a phone line with a specific telephone-number, use the show voice register pool telephone-number command in privileged EXEC mode.
show voice register pool telephone-number number
Syntax Description
number
|
Number identifying a specific phone.
|
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
15.1(2)T
|
Cisco Unified CME 8.1 Cisco Unified SRST 8.1
|
This command was introduced.
|
Usage Guidelines
Use this command to display the details of the phone line with the specified telephone-number. If the line is registered, the contact ip address will be displayed. When the phone line is not registered and the pool ID type is network IP, the IP address is not displayed. When the phone line is not registered but some other line is registered for the same pool with MAC or IP address, then the IP address is displayed.
Examples
Cisco Unified CME
The following is a sample output from this command displaying all statistical information:
Router# show voice register pool telephone number 45112
Pool ID IP Address Ln DN Number State
==== =============== =============== == === ==================== ============
2 0015.C68E.6D13 1 2 45112 UNREGISTERED
7 0018.BAC8.D2B1 1 2 45112 UNREGISTERED
Cisco Unified SRST
Router# show voice register pool telephone-number 1000
Pool ID IP Address Ln DN Number State
==== =============== =============== == === ==================== ============
1 9.13.18.40 9.13.18.40 1 1 1000 REGISTERED
Table 27 contains descriptions of significant fields shown in this output, listed in alphabetical order.
Table 31 show voice register pool telephone number field descriptions
Field
|
Description
|
DN
|
Directory number of the phone.
|
ID
|
Phone identification (ID) address.
|
IP Address
|
IP address and port number of the phones
|
LN
|
Line number of the phone.
|
Number
|
Number of the phones.
|
Pool
|
Shows the current pool.
|
State
|
Registration state.
|
Related Commands
Command
|
Description
|
show voice register all
|
Displays all Cisco Unified SIP SRST and Cisco Unified CME configurations and register information.
|
show voice register pool
|
Displays all configuration information associated with a particular voice register pool.
|
show voice register pool detail all
|
Displays the details of all the pools defined in the system.
|
show voice register pool unregistered
To display the details of the voice registration pools that do not have any phones registered, use the show voice register pool unregistered command in privileged EXEC mode.
show voice register pool unregistered
Syntax Description
This command has no arguments or keywords.
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Version
|
Modification
|
15.1(2)T
|
Cisco Unified CME 8.1 Cisco Unified SRST 8.1
|
This command was introduced.
|
Usage Guidelines
Use this command to display the details of the pools that do not have any active registrations. In Cisco Unified SRST, if multiple phones are trying to register through the same pool and if one phone successfully registers and the others do not, the pool is not considered as an unregistered pool, as it does have an active registration of the registered phone.
Examples
Cisco Unified CME and Cisco Unified SRST
The following is a sample output from this command displaying information for pools with no active registeration:
Router# show voice register pool unregistered
Pool Tag: 2
MAC Address : 0015.C68E.6D13
No. of attempts to register: 0
Unregister time :
Last register request time :
Reason for state unregister:
No registration request since last reboot/unregister
Pool Tag: 3
MAC Address : 0021.5553.8998
No. of attempts to register: 0
Unregister time :
Last register request time :
Reason for state unregister:
No registration request since last reboot/unregister
Pool Tag: 4
MAC Address : 8989.9867.8769
No. of attempts to register: 0
Unregister time :
Last register request time :
Reason for state unregister:
No registration request since last reboot/unregister
Related Commands
Command
|
Description
|
show voice register all
|
Displays all Cisco SIP SRST and Cisco CME configurations and register information.
|
show voice register pool
|
Displays all configuration information associated with a particular voice register pool.
|
show voice register pool registered
|
Displays details of phones that sucessfully register to Cisco Unified CME or Cisco Unified SRST.
|
voice register pool
|
Enters voice register pool configuration mode for SIP phones.
|
show voice register statistics
To display statistics associated with the registration event, use the show voice register statistics command in privileged EXEC mode.
show voice register statistics [global | pool tag]
Syntax Description
global
|
(Optional) Displays aggregate statistics associated with the SIP phone registration event.
|
pool tag
|
(Optional) Displays registration pool statistics associated with a specific pool tag. The maximum number of pools is version and platform dependent. Type ? to display a list of values.
|
Command Modes
Privileged EXEC
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco SIP SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco SIP SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
12.4(4)T
|
Cisco CME 3.4 Cisco SIP SRST 3.4
|
This command was added to Cisco CME.
|
15.1(2)T
|
Cisco CME 8.1 Cisco SIP SRST 8.1
|
This command was modified. The global and pool keywords and tag argument were added. The output display was also modified to show more information about pools in unregistered state and time-stamps of registration event.
|
Usage Guidelines
When using the show voice register statistics command, you can verify that the number of Registration and unRegister successes for global statistics are the sum of the values in the individual pools. Because some Registrations fail even before matching a voice register pool, for Registration and unRegister failed statistics the value is not the sum of the values in the individual pools. Immediate failures are accounted in the global statistics.
In Cisco Unified CME 8.1 and Cisco Unified SIP SRST 8.1, the time-stamps for the events is displayed along with other registration related statistics. The command output also displays the reason for pools in unregistered state. Use the show voice register statistics command with pool tag keyword to display registration pool statistics associated with a specific pool.
When using the global keyword, the show voice register command output displays the aggregate statistics associated with SIP phone registration. The output of this command also displays the attempted-registrations table.
Examples
Cisco Unified CME and Cisco Unified SRST
The following is a sample output from this command displaying all statistical information:
Router# show voice register statistics
Total SIP phones registered: 2
Total Registration Statistics
Registration requests : 3
after last unregister : 1
Last Register Request Time : *11:42:31.783 UTC Wed Sep 16 2009
Last Unregister Request Time :
Register Success Time : *11:11:56.707 UTC Wed Sep 16 2009
Unregister Success Time :
Register pool 1 statistics
Total SIP phones registered: 1
Total Registration Statistics
Registration requests : 1
after last unregister : 0
Last Register Request Time : *11:11:54.615 UTC Wed Sep 16 2009
Last Unregister Request Time :
Register Success Time : *11:11:54.623 UTC Wed Sep 16 2009
Unregister Success Time :
Register pool 2 statistics
Total SIP phones registered: 1
Total Registration Statistics
Registration requests : 1
after last unregister : 0
Last Register Request Time : *11:11:56.707 UTC Wed Sep 16 2009
Last Unregister Request Time :
Register Success Time : *11:11:56.707 UTC Wed Sep 16 2009
Unregister Success Time :
Cisco Unified CME and Cisco Unified SRST
The following is a sample output from this command displaying all statistical information:
Router# show voice register statistics global
Global Statistics:
Active registrations : 1
Total SIP phones registered: 2
Total Registration Statistics
R egistration requests : 97715
Registration success : 3
Registration failed : 97712
unRegister requests : 1
unRegister success : 1
unRegister failed : 0
Attempts to register
after last unregister : 97712
Last register request time : *06:45:11.127 UTC Wed Oct 14 2009
Last unregister request time : *11:56:22.179 UTC Tue Oct 13 2009
Register success time : *12:10:37.263 UTC Tue Oct 13 2009
Unregister success time : *11:56:22.182 UTC Tue Oct 13 2009
Phones that have attempted registrations and have failed:
MAC address: 001b.535c.d410
IP address : 8.3.3.111
Attempts : 97712
Time of first attempt : *12:20:32.775 UTC Tue Oct 13 2009
Time of latest attempt: *06:46:14.815 UTC Wed Oct 14 2009
Reason for failure :
Unauthorized registration request
Cisco Unified CME and Cisco Unified SRST
The following is a sample output from this command displaying all statistical information associated with pool 1:
Router# show voice register statistics pool 1
Total SIP phones registered: 1
Total Registration Statistics
Registration requests : 2
after last unregister : 0
Last register request time : *12:10:37.259 UTC Tue Oct 13 2009
Last unregister request time : *11:56:22.179 UTC Tue Oct 13 2009
Register success time : *12:10:37.263 UTC Tue Oct 13 2009
Unregister success time : *11:56:22.182 UTC Tue Oct 13 2009
Reason for unregistered state:
No registration request since last reboot/unregister
Table 32 describes the significant fields shown in this output.
Table 32 show voice register statistics Field Descriptions
Field
|
Description
|
Statistics:
|
Used with the all, pool, and statistics keywords. Shows the registration statistics for this pool.
|
Active registrations
|
Used with the all, pool, and statistics keywords. Shows the current active registrations.
|
Last Register Request Time
|
Used with all, pool, and statistics keywords. Shows details such as day, date, and time when the phones requested to register the last time.
|
Last unRegister Request Time
|
Used with all, pool, and statistics keywords. Shows details such as day, date, and time when the phones requested to unregister the last time.
|
Total Registration Statistics
|
Used with the all, pool, and statistics keywords. Shows the total registration statistics for this pool.
|
Registration requests
|
Used with the all, pool, and statistics keywords. Shows the incoming registration requests.
|
Registration success
|
Used with the all, pool, and statistics keywords. Shows the successful registrations.
|
Registration failed
|
Used with the all, pool, and statistics keywords. Shows the failed registrations.
|
unRegister requests
|
Used with the all, pool, and statistics keywords. Shows the incoming unregister/registration expire requests.
|
unRegister success
|
Used with the all, pool, and statistics keywords. Reports the number of successful unregisters.
|
unRegister failed
|
Used with the all, pool, and statistics keywords. Reports the number of failed unregisters.
|
Global statistics
|
Used with the statistics keyword. Details all active registrations.
|
Register pool number statistics
|
Used with the statistics keyword. Details specific pool statistics.
|
Related Commands
Command
|
Description
|
show voice register all
|
Displays all Cisco Unified SIP SRST and Cisco Unified CME configurations and register information.
|
show voice register pool
|
Displays all configuration information associated with a particular voice register pool.
|
show voice register pool attempted-registrations
|
Displays the details of phones that attempt to register with Cisco Unified CME or Cisco Unified SRST and fail.
|
subnet
To define which IP phones are part of an emergency response location (ERL) for the enhanced 911 service, use the subnet command in voice emergency response location configuration mode. To remove the subnet definition, use the no form of this command.
subnet [1 | 2] IPaddress mask
no subnet [1 | 2]
Syntax Description
IPaddress
|
IP address that identifies which IP phones are part of the ERL.
|
mask
|
IP subnet mask for the network segment that is part of the ERL.
|
Command Default
No subnets are defined.
Command Modes
Voice emergency response location configuration (cfg-emrgncy-resp-location)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.4(15)T
|
Cisco Unified CME 4.1 Cisco Unified SRST 4.1 Cisco Unified SIP SRST 4.1
|
This command was introduced. For Cisco Unified CME, this command is supported in SRST fallback mode only.
|
12.4(15)XY
|
Cisco Unified CME 4.2(1) Cisco Unified SRST 4.2(1) Cisco Unified SIP SRST 4.2(1)
|
This command was added to Cisco Unified CME.
|
12.4(20)T
|
Cisco Unified CME 7.0 Cisco Unified SRST 7.0 Cisco Unified SIP SRST 7.0
|
This command was integrated into Cisco IOS Release 12.4(20)T.
|
Usage Guidelines
This command defines the groups of IP phones that are part of an ERL. You can create up to 2 different subnets. To include all phones on a single ERL, you can set the subnet mask to 0.0.0.0 to indicate a "catch-all" subnet.
Examples
In the following example, all IP phones with the IP address of 10.X.X.X or 192.168.X.X are automatically associated with this ERL. If one of the phones dials 911, its extension is replaced with 408 555-0100 before it goes to the PSAP. The PSAP will see that the caller's number is 408 555-0100.
voice emergency response location 1
subnet 10.0.0.0 255.0.0.0
subnet 2 192.168.0.0 255.255.0.0
Related Commands
Command
|
Description
|
elin
|
Specifies a PSTN number that will replace the caller's extension.
|
system message (call-manager-fallback)
To customize the system message text displayed on all Cisco IP phones units in fallback mode that are connected to a Cisco Unified Survivable Remote Site Telephony ( SRST) router, use the system message command in call-manager-fallback configuration mode. To disable the customized message and return to the default system message, use the no form of this command.
system message {primary primary-string | secondary secondary-string}
no system message {primary primary-string | secondary secondary-string}
Syntax Description
primary
|
Sets the system message for Cisco IP phones that can support static text messages during fallback, such as the Cisco IP Phone 7940 and the Cisco IP Phone 7960.
|
primary-string
|
Text string of less than 32 characters.
|
secondary
|
Sets the system message for Cisco IP phones that do not support static text messages and that have a limited display space, such as the Cisco IP Phone 7910.
|
secondary-string
|
Text string of less than 20 characters.
|
Defaults
The default fallback display message for Cisco IP phones that support static text messages is "CM Fallback Service Operating." For Cisco IP phones that do not support static text messages, the default message is "CM Fallback Service."
Command Modes
Call-manager-fallback configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
Usage Guidelines
Changes to the display message configuration occur only after a phone reset, at the end of each call, or on receipt of the next keepalive message from an idle phone.
The normal in-service static text message is controlled by Cisco Unified Communications Manager.
Secondary IP phones flash system messages during fallback.
Examples
The following example sets the system message to "Customized Message" for all Cisco IP Phone 7940 and Cisco IP Phone 7960 units connected to a Cisco Unified SRST router:
Router(config)# call-manager-fallback
Router(config-cm-fallback)# system message primary Customized Message
Related Commands
Command
|
Description
|
call-manager-fallback
|
Enables Cisco Unified SRST feature support and enters call-manager-fallback configuration mode.
|
system message (voice register global)
To define a message that displays on SIP phones in a Cisco Unified Survivable Remote Site Telephony (Cisco Unified SRST) system, use the system message command in voice register global configuration mode. To return to the default, use the no form of this command.
system message string
no system message
Syntax Description
string
|
Message that displays on SIP phones after the phones failover to Cisco Unified SRST. String can contain a maximum of 32 alphanumeric characters.
|
Command Default
"CM Fallback Service Operating" message from dictionary file displays.
Command Modes
Voice register global configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.4(11)XJ
|
Cisco Unified SRST 4.1
|
This command was introduced.
|
12.4(15)T
|
Cisco Unified CME 4.1
|
This command was integrated into Cisco IOS Release 12.4(15)T.
|
Usage Guidelines
The command allows you to customize the idle prompt message that displays on the status line of SIP phones after the phones lose connection with Cisco Unified Communications Manager and failover to Cisco Unified SRST. The default message that displays is from the dictionary file for the phone. The configured message displays until the phones fallback to Cisco Unified Communications Manager. For versions earlier than Cisco Unified SRST 4.1, the phones display the default message from the dictionary file.
This command is not supported on the Cisco Unified IP Phone 7905, 7912, 7940, or 7960.
Examples
The following example shows that SIP phones will display the message, "SRST service active" after the phones register to Cisco Unified SRST.
Router(config)# voice register global
Router(config-register-global)# system message SRST service active
Related Commands
Command
|
Description
|
show voice register global
|
Displays all global configuration parameters associated with SIP phones.
|
time-format (call-manager-fallback)
To set the time display format on all Cisco IP phones attached to a router, use the time-format command in call-manager-fallback configuration mode. To disable the time display format, use the no form of this command.
time-format {12 | 24}
no time-format {12 | 24}
Syntax Description
12
|
Sets format to 12-hour increments.
|
24
|
Sets format to 24-hour increments.
|
Defaults
12-hour format
Command Modes
Call-manager-fallback configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(2)XT
|
Cisco SRST 2.0
|
This command was introduced on the following platforms: Cisco 1750, Cisco 1751, Cisco 2600 and Cisco 3600 series multiservice routers, Cisco IAD2420 series IADs.
|
12.2(8)T
|
Cisco SRST 2.0
|
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725, Cisco 3745, and Cisco MC3810-V3 routers.
|
12.2(8)T1
|
Cisco SRST 2.0
|
This command was implemented on the Cisco 2600-XM and Cisco 2691 routers.
|
12.2(11)T
|
Cisco SRST 2.01
|
This command was integrated into Cisco IOS Release 12.2(11)T and implemented on the Cisco 1760 routers.
|
Examples
The following example shows the time format on the Cisco IP phones being set to the 24-hour format:
Router(config)# call-manager-fallback
Router(config-cm-fallback)# time-format 24
Related Commands
Command
|
Description
|
call-manager-fallback
|
Enables Cisco Unified Survivable Remote Site Telephony (SRST) support and enters call-manager-fallback configuration mode.
|
timeouts busy (call-manager-fallback)
To set the timeout value for call transfers to busy destinations, use the timeouts busy command in call-manager-fallback configuration mode. To return to the default value, use the no form of this command.
timeouts busy seconds
no timeouts busy
Syntax Description
seconds
|
Number of seconds after connection to a busy destination before a transferred call is disconnected. Range is from 0 to 30 seconds. Default is 10.
|
Command Default
10 seconds
Command Modes
Call-manager-fallback configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(8)T
|
Cisco SRST 2.0
|
This command was introduced.
|
Usage Guidelines
For calls that are transferred to busy destinations, this command sets the amount of time after connection to the busy destination before the call is disconnected.
Note that the timeout set by this command applies only to calls that are transferred to busy destinations and not to calls that directly dial busy destinations.
Examples
The following example sets the ringing timeout to 10 seconds:
Router(config)# call-manager-fallback
Router(config-cm-fallback)# timeouts busy 20
Related Commands
Command
|
Description
|
call-manager-fallback
|
Enables Cisco Unified Survivable Remote Site Telephony (SRST) and enters call-manager-fallback configuration mode.
|
timeouts interdigit (call-manager-fallback)
To configure the interdigit timeout value for all Cisco IP phones attached to a router, use the timeouts interdigit command in call-manager-fallback configuration mode. To return the interdigit timeout value to its default, use the no form of this command.
timeouts interdigit seconds
no timeouts interdigit
Syntax Description
seconds
|
Interdigit timeout duration, in seconds, for all Cisco IP phones. Valid entries are integers from 2 to 120.
|
Defaults
No default behavior or values.
Command Modes
Call-manager-fallback configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(2)XB
|
Cisco SRST 1.0
|
This command was introduced on the following platforms: Cisco 2600 series and Cisco 3600 series multiservice routers; Cisco IAD2420 series IADs; Cisco 7200 series routers.
|
12.2(2)XT
|
Cisco SRST 2.0
|
This command was implemented on Cisco 1750 and Cisco 1751 multiservice routers.
|
12.2(8)T
|
Cisco SRST 2.0
|
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725, Cisco 3745, and Cisco MC3810-V3 routers.
|
12.2(8)T1
|
Cisco SRST 2.0
|
This command was implemented on the Cisco 2600-XM and Cisco 2691 routers.
|
12.2(11)T
|
Cisco SRST 2.01
|
This command was integrated into Cisco IOS Release 12.2(11)T and implemented on the Cisco 1760 routers.
|
Usage Guidelines
The timeouts interdigit command specifies how long, in seconds, the system waits after a caller enters the initial digit or a subsequent digit of the dialed string. The interdigit timer is activated when the caller enters a digit and is restarted each time the caller enters subsequent digits until the destination address is identified. If the configured timeout value is exceeded before the destination address is identified, a tone sounds and the call is terminated.
Examples
The following example sets the interdigit timeout value to 5 seconds for all Cisco IP phones:
Router(config)# call-manager-fallback
Router(config-cm-fallback)# timeouts interdigit 5
In this example, the 5 seconds refers to the elapsed time after which an incompletely dialed number times out. For example, if you dial nine digits (408555010) instead of the required ten digits (4085550100), you hear a busy tone after the 5 "timeout" seconds have elapsed.
Related Commands
Command
|
Description
|
call-manager-fallback
|
Enables Cisco Unified Survivable Remote Site Telephony (SRST) support and enters call-manager-fallback configuration mode.
|
timeouts interdigit (voice port)
|
Configures the interdigit timeout value for a specified voice port.
|
timeouts ringing (call-manager-fallback)
To set the time before a disconnect code is returned on phones without a call-forward no-answer configuration, use the timeouts ringing command in call-manager-fallback configuration mode. To disable the time setting, use the no form of this command.
timeouts ringing seconds
no timeouts ringing
Syntax Description
seconds
|
The duration, in seconds, for which a voice port allows ringing to continue if a call is not answered. The range is from 5 to 60000.
|
Defaults
No default behavior or values.
Command Modes
Call-manager-fallback configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
Usage Guidelines
This mechanism protects against hung inbound calls through interfaces that do not have forward disconnect supervision, such as Foreign Exchange Office (FXO).
Expiration of the timeout causes incoming calls to return a disconnect code to the caller.
Examples
The following example sets the ringing timeout to 10 seconds:
Router(config)# call-manager-fallback
Router(config-cm-fallback)# timeouts ringing 10
Related Commands
Command
|
Description
|
call-manager-fallback
|
Enables Cisco Unified Survivable Remote Site Telephony (SRST) and enters call-manager-fallback configuration mode.
|
transfer-digit-collect
To select the digit-collection method for consultative call-transfers, use the transfer-digit-collect command in telephony-service configuration mode for Cisco Unified CME or in call-manager-fallback configuration mode for Cisco Unified SRST. To reset to the default value, use the no form of this command.
transfer-digit-collect {new-call | orig-call}
no transfer-digit-collect
Syntax Description
new-call
|
Dialed digits are collected from new call leg. Default value.
|
orig-call
|
Dialed digits are collected from original call leg.
|
Command Default
Digits are collected from the new consultative call-leg (new-call keyword).
Command Modes
Telephony-service configuration (config-telephony)
Call-manager-fallback configuration (config-cm-fallback)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.4(15)XZ
|
Cisco Unified CME 4.3 Cisco Unified SRST 4.3
|
This command was introduced.
|
12.4(20)T
|
Cisco Unified CME 7.0 Cisco Unified SRST 7.0
|
This command was integrated into Cisco IOS Release 12.4(20)T.
|
Usage Guidelines
This command specifies whether the dialed digits of the target number are collected on the original call leg or on the new call leg that is created when a phone user initiates a consultative call-transfer.
For consultative transfers, a local number is matched on the number command in ephone-dn configuration mode; a PSTN number is matched on the transfer-pattern command in telephony service mode.
The orig-call keyword selects the method used in versions before Cisco Unified CME 4.3 and Cisco Unified SRST 4.3. After a phone user presses the Transfer soft key, the dialed digits of the target number are collected on the original call leg and buffered until either a local ephone-dn or transfer-pattern is matched. When the transfer-to number is matched, the original call is put on hold and an idle line or channel is seized to send the dialed digits from the buffer.
The new-call keyword selects the default method that is used in Cisco Unified CME 4.3 and later versions and Cisco Unified SRST 4.3 and later versions. The transfer-to digits are collected on a new consultative call-leg that is created when the user presses the Transfer soft key. The consultative call-leg is seized and the dialed digits are passed on without being buffered until the digits are matched and the consultative call-leg moves to the alerting state.
The new-call keyword is supported only if:
•
The transfer-system full-consult command (default) is set in telephony-service configuration mode.
•
The transfer-mode consult command (default) is set for transferor's directory number (ephone-dn).
•
An idle line or channel is available for seizing, digit collection, and dialing.
A consultative transfer is one in which the transferring party either connects the caller to a ringing phone (ringback heard) or speaks with the third party before connecting the caller to the third party.
Examples
The following example shows the digit-collection set to the method used in versions before Cisco Unified CME 4.3 and Cisco Unified SRST 4.3:
Router(config)# telephony-service
Router(config-telephony)# transfer-digit-collect orig-call
Related Commands
Command
|
Description
|
transfer-mode
|
Specifies the type of call transfer for an individual directory number that uses the ITU-T H.450.2 standard.
|
transfer-pattern (telephony-service)
|
Allows the transfer of calls to phones outside the local Cisco Unified CME network.
|
transfer-system
|
Specifies the call transfer method for all IP phones on a Cisco Unified CME router using the ITU-T H.450.2 standard.
|
transfer-pattern
To allow Cisco IP phones to transfer telephone calls from callers outside the local IP network to another Cisco IP phone, use the transfer-pattern command in call-manager-fallback configuration mode. To disable transfer of calls to other numbers, use the no form of this command.
transfer-pattern transfer-pattern
no transfer-pattern
Syntax Description
transfer-pattern
|
String of digits for permitted call transfers. Wildcards are allowed.
|
Defaults
This feature is enabled.
Command Modes
Call-manager-fallback configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.1(5)YD
|
Cisco SRST 1.0
|
This command was introduced on the following platforms: Cisco 2600 series and Cisco 3600 series multiservice routers, and Cisco IAD2420 series IADs.
|
12.2(2)XT
|
Cisco SRST 2.0
|
This command was implemented on Cisco 1750 and Cisco 1751 multiservice routers.
|
12.2(8)T
|
Cisco SRST 2.0
|
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725, Cisco 3745, and Cisco MC3810-V3 routers.
|
12.2(8)T1
|
Cisco SRST 2.0
|
This command was implemented on the Cisco 2600-XM and Cisco 2691 routers.
|
12.2(11)T
|
Cisco SRST 2.01
|
This command was integrated into Cisco IOS Release 12.2(11)T and implemented on the Cisco 1760 routers.
|
Usage Guidelines
The transfer-pattern command allows you to transfer a call from a non-IP phone number to another Cisco IP phone on the same IP network using the specified transfer pattern. By default, all Cisco IP phone directory numbers or virtual voice ports are allowed as transfer targets.
When you define transfers to nonlocal numbers, it is important to note that transfer-pattern digit matching is performed before translation-rule operations. Therefore, you should specify in this command the digits that are actually entered by phone users before they are translated. For more information, see the "Enabling Digit Translation Rules" section in the Cisco IOS Survivable Remote Site Telephony Version 3.3 System Administrator Guide.
Examples
The following example sets a transfer pattern:
Router(config)# call-manager-fallback
Router(config-cm-fallback)# transfer-pattern 55501..
A maximum of 32 transfer patterns can be entered. In this example, 55501.. (the two decimal points are used here as wildcards) permits transfers to any numbers in the range from 555-0100 to 555-0199.
Related Commands
Command
|
Description
|
call-manager-fallback
|
Enables Cisco Unified Survivable Remote Site Telephony (SRST) support and enters call-manager-fallback configuration mode.
|
transfer-pattern blocked (voice register pool)
To block all call transfers for a specific Cisco Unified SIP IP phone or a set of Cisco Unified SIP IP phone, use the transfer-pattern blocked command in voice register pool and voice register template configuration mode. To allow call transfers, use the no form of this command.
transfer-pattern blocked
no transfer-pattern blocked
Syntax Description
This command has no arguments or keywords.
Command Default
Call transfers for a specific Cisco Unified SIP IP phone or a set of Cisco Unified SIP IP phone are allowed.
Command Modes
Voice register pool configuration (config-register-pool)
Voice register template configuration ((config-register-temp)
Command History
Release
|
Modification
|
15.3(2)T
|
This command was introduced.
|
Usage Guidelines
When the transfer-pattern blocked command is configured for a specific phone, no call transfers are allowed from that phone over the trunk.
This feature forces unconditional blocking of all call transfers from a specific phone to any other non-local numbers (external calls from one trunk to another trunk). No call transfers from this specific phone are possible even when a transfer pattern matches the dialed digits for transfer.
Examples
The following example shows how to block all call transfers for voice register pool 5:
Router(config)# voice register pool 5
Router(config-register-pool)# transfer-pattern ?
blocked global transfer pattern not allowed
Router(config-register-pool)# transfer-pattern blocked
The following example shows how to block all call transfers for a set of Cisco Unified SIP IP phones defined by voice register template 9:
Router(config)# voice register template 9
Router(config-register-temp)# transfer-pattern ?
blocked global transfer pattern not allowed
Router(config-register-temp)# transfer-pattern blocked
Related Commands
Command
|
Description
|
voice register pool
|
Enters voice register pool configuration mode and creates a pool configuration for a Cisco Unified SIP IP phone in Cisco Unified CME or for a set of Cisco Unified SIP IP phones in Cisco Unified SIP SRST.
|
voice register template
|
Enters voice register template configuration mode and defines a template of common parameters for Cisco Unified SIP IP phones.
|
transfer max-length (voice register pool)
To specify the maximum length of the transfer number, use the transfer max-length command in voice register pool or voice register template configuration mode. To disable the maximum length, use the no form of this command.
transfer max-length max-length
no transfer max-length max-length
Syntax Description
max-length
|
Maximum length of the transfer number. Range is 3 to 16.
|
Command Default
No maximum length is specified for the transfer number.
Command Modes
Voice register pool configuration (config-register-pool)
Voice register template configuration ((config-register-temp)
Command History
Release
|
Modification
|
15.3(2)T
|
This command was introduced.
|
Usage Guidelines
The transfer max-length command is used to indicate the maximum length of the number being dialed for a call transfer. When only a specific number of digits are to be allowed during a call transfer, a value between 3 and 16 is configured.When the number dialed exceeds the maximum length configured, then the call transfer is blocked.
Examples
The following example shows how to configure the maximum length of the transfer number under voice register pool 1. Because the maximum length is configured as 5, only call transfers to Cisco Unified SIP IP phones with a five-digit directory number are allowed. All call transfers to directory numbers with more than five digits are blocked.
Router# configure terminal
Router(config)# voice register pool 1
Router(config-register-pool)# transfer max-length 5
The following example shows how to configure the maximum length of the transfer number for a set of phones under voice register template 2:
Router# configure terminal
Router(config)# voice register template 2
Router(config-register-temp)# transfer max-length 10
Command
|
Description
|
voice register pool
|
Enters voice register pool configuration mode and creates a pool configuration for a SIP IP phone in Cisco Unified CME or for a set of SIP phones in Cisco Unified SIP SRST
|
voice register template
|
Enters voice register template configuration mode and defines a template of common parameters for SIP phones.
|
transfer-system (call-manager-fallback)
To specify the call-transfer method for all IP phones on a Cisco Unified Survivable Remote Site Telephony (SRST) router using the ITU-T H.450.2 standard, use the transfer-system command in call-manager-fallback configuration mode. To disable the call-transfer method, use the no form of this command.
transfer-system {blind | full-blind | full-consult | local-consult}
no transfer-system
Syntax Description
blind
|
Transfers calls without consultation using a single phone line and the Cisco proprietary method. The keyword blind is not recommended. Use either the full-blind or full-consult keyword instead.
|
full-blind
|
Transfers calls without consultation using H.450.2 standard methods.
|
full-consult
|
Transfers calls using H.450.2 with consultation using the second phone line if available, or the calls fall back to full-blind if the second line is unavailable.
|
local-consult
|
Transfers calls with local consultation using the second phone line if available, or the calls fall back to blind for nonlocal consultation or transfer target. This method is intended for use primarily in VoFR networks because the Cisco VoFR call-transfer protocol does not support an end-to-end transfer with consultation mechanism.
|
Defaults
No default behavior or values.
Command Modes
Call-manager-fallback configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
Usage Guidelines
Call transfers using the H.450.2 standard can be blind or consultative. A blind transfer is one in which the transferring phone connects the caller to a destination line before ringback begins. A consultative transfer is one in which the transferring party either connects the caller to a ringing phone (ringback heard) or speaks with the third party before connecting the caller to the third party. When H.450.2 call transfer is selected using the full-blind or full-consult keyword, the router must be configured with a Tool Command Language (Tcl) script that supports the H.450.3 protocol. The Tcl script is loaded on the Cisco Unified SRST router with the call application voice command.
Note
Note: The keyword blind is not recommended. Use either the full-blind or full-consult keyword instead.
Examples
The following example sets full consultation as the call-transfer method for this Cisco Unified SRST phone network:
Router(config)# call-manager-fallback
Router(config-cm-fallback)# transfer-system full-consult
Related Commands
Command
|
Description
|
call application voice
|
Defines an application, indicates the location of the corresponding Tcl files that implement the application, and loads the selected Tcl script.
|
call-manager-fallback
|
Enables Cisco Unified SRST support and enters call-manager-fallback configuration mode.
|
translate (call-manager-fallback)
To apply a translation rule to modify the phone number dialed or received by any Cisco IP phone user during Cisco Unified Communications Manager fallback, use the translate command in call-manager-fallback configuration mode. To disable this feature, use the no form of this command.
translate {called | calling} translation-rule-tag
no translate {called | calling} translation-rule-tag
Syntax Description
called
|
Translation rule to apply to the number called by a Cisco IP phone.
|
calling
|
Translation rule to apply to the calling party number sent in the call setup message for calls originated from a Cisco IP phone.
|
translation-rule-tag
|
Tag number by which the rule set is referenced. This is an arbitrarily chosen number. The range is from 1 to 2147483647.
|
Defaults
No default behavior or values.
Command Modes
Call-manager-fallback configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(2)XT
|
Cisco SRST 2.0
|
This command was introduced on the following platforms: Cisco 1750, Cisco 1751, Cisco 2600 series and Cisco 3600 series multiservice routers, Cisco IAD2420 series IADs.
|
12.2(8)T
|
Cisco SRST 2.0
|
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725, Cisco 3745, and Cisco MC3810-V3 routers.
|
12.2(8)T1
|
Cisco SRST 2.0
|
This command was implemented on the Cisco 2600-XM and Cisco 2691 routers.
|
12.2(11)T
|
Cisco SRST 2.01
|
This command was integrated into Cisco IOS Release 12.2(11)T and implemented on the Cisco 1760 routers.
|
Usage Guidelines
The translate command allows you to apply a previously configured number-translation rule to modify the number dialed or received by a specific extension. Translation rules are a powerful general-purpose number-manipulation mechanism that performs operations such as automatically adding telephone area and prefix codes to dialed numbers.
Examples
The following example applies translation rule 20 to the inbound called number:
Router(config)# translation-rule 20
Router(config-translate)# rule 0 1234 2345 abbreviated
Router(config-translate)# exit
Router(config)# call-manager-fallback
Router(config-cm-fallback)# translate called 20
Related Commands
Command
|
Description
|
call-manager-fallback
|
Enables Cisco Unified Survivable Remote Site Telephony (SRST) support and enters call-manager-fallback configuration mode.
|
translation-profile (call-manager-fallback)
|
Assigns a translation profile for incoming or outgoing call legs on a Cisco IP phone.
|
translation-rule
|
Creates a translation name and enters translation-rule configuration mode.
|
translate-outgoing (voice register pool)
To allow an explicit setting of translation rules on the VoIP dial peer in order to modify a phone number dialed by any Cisco IP phone user, use the translate-outgoing command in voice register pool configuration mode. To disable translation rules, use the no form of this command.
translate-outgoing {called | calling} rule-tag
no translate-outgoing {called | calling}
Syntax Description
called
|
Called party requires translation.
|
calling
|
Calling party requires translation.
|
rule-tag
|
The rule-tag is an arbitrarily chosen number by which the rule set is
referenced. The range is from 1 to 2147483.
|
Defaults
None
Command Modes
Voice register pool configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco SIP SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco SIP SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
12.4(4)T
|
Cisco CME 3.4 Cisco SIP SRST 3.4
|
This command was added to Cisco CME.
|
Usage Guidelines
Translation rules are a powerful general-purpose number-manipulation mechanism that perform operations such as automatically adding telephone area and prefix codes to dialed numbers. The translation rules are applied to VoIP dial peers created by Cisco Unified Session Initiation Protocol (SIP) Survivable Remote Site Telephony (SRST) or Cisco Unified Communications Manager Express (Cisco Unified CME)
.
During registration, a dial peer is created, and that dial peer includes a default translation rule. The translate-outgoing command allows you to change the translation rule, if desired. The translate-outgoing command allows you to select a preconfigured number translation rule to modify the number dialed by a specific extension.
Note
Translation rules must be set by using the translate-outgoing command before the alias command is configured in Cisco Unified SIP SRST.
Configure the id (voice register pool) command before any other voice register pool commands, including the translate-outgoing command. The id command identifies a locally available individual SIP phone or set of SIP phones.
Examples
Cisco Unified CME
The following is partial sample output from the show running-config command showing that called-party 1 requires translation.
cor incoming call91 1 91011
translate-outgoing called 1
Cisco Unified SIP SRST
The following is partial sample output from the show running-config command showing that called-party 1 requires translation.
cor incoming call91 1 91011
translate-outgoing called 1
proxy 10.2.161.187 preference 1 monitor probe icmp-ping
alias 1 94... to 91011 preference 8
Related Commands
Command
|
Description
|
alias (voice register pool)
|
Allows Cisco SIP IP phones to handle inbound PSTN calls to telephone numbers that are unavailable when the main proxy is not available.
|
id (voice register pool)
|
Explicitly identifies a locally available individual Cisco SIP IP phone, or when running Cisco Unified SIP SRST, set of Cisco SIP IP phones.
|
translate-outgoing (dial-peer)
|
Applies a translation rule to manipulate dialed digits on an outbound POTS or VoIP call leg.
|
voice register pool
|
Enters voice register pool configuration mode for SIP phones.
|
translation-profile (call-manager-fallback)
To assign a translation profile for incoming or outgoing call legs on a Cisco IP phone, use the translation-profile command in call-manager-fallback configuration mode. To delete the translation profile from the voice port, use the no form of this command.
translation-profile {incoming | outgoing} name
no translation-profile {incoming | outgoing} name
Syntax Description
incoming
|
Specifies that this translation profile handles incoming calls.
|
outgoing
|
Specifies that this translation profile handles outgoing calls.
|
name
|
Name of the translation profile.
|
Defaults
No default behavior or values.
Command Modes
Call-manager-fallback configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.3(11)T
|
Cisco SRST 3.2
|
This command was introduced.
|
Usage Guidelines
Cisco Unified Survivable Remote Site Telephony (SRST) 3.2 and later versions support translation profiles. Translation profiles allow you to group translation rules together and to associate translation rules with the following:
•
Called numbers
•
Calling numbers
•
Redirected called numbers
Use the translation-profile command to assign a global predefined translation profile to an incoming or outgoing call leg. For example, a company number can be assigned to overwrite an individual caller's phone number. That is, the translation-profile command modifies the phone number dialed or received by a Cisco IP phone user while in Communications Manager fallback mode.
Cisco IP phones support one incoming and one outgoing translation profile when in SRST mode.
Examples
The following example shows a configuration in which a translation profile called name1 is created with two voice translation rules. Rule1 consists of associated calling numbers, and rule2 consists of redirected called numbers. The Cisco IP phones in SRST mode are configured with name1.
voice translation-profile name1
translation calling rule1
translation called-direct rule2
translation-profile incoming name1
Related Commands
Command
|
Description
|
call-manager-fallback
|
Enables Cisco SRST support and enters call-manager-fallback configuration mode.
|
show voice translation-profile
|
Displays the configuration of a translation profile.
|
translate (call-manager- fallback)
|
Applies a translation rule to modify the phone number dialed or received by any Cisco IP phone user during Communications Manager fallback.
|
translation-rule
|
Creates a translation name and enters translation-rule configuration mode to apply rules to the translation name.
|
voice translation-profile
|
Defines a translation profile for voice calls.
|
translation-profile (voice register)
To apply a translation profile to incoming or outgoing call legs on a SIP phone in a Cisco Unified SRST system, use the translation-profile command in voice register dn or voice register pool configuration mode. To remove the translation profile, use the no form of this command.
translation-profile {incoming | outgoing} name
no translation-profile {incoming | outgoing}
Syntax Description
incoming
|
Specifies that this translation profile handles incoming calls.
|
outgoing
|
Specifies that this translation profile handles outgoing calls.
|
name
|
Name of the translation profile.
|
Defaults
Translation profile is not assigned to call legs on the phone.
Command Modes
Voice register dn configuration (config-register-dn)
Voice register pool configuration (config-register-pool)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.4(22)YB
|
Cisco Unified SIP SRST 7.1
|
This command was introduced.
|
12.4(24)T
|
Cisco Unified SIP SRST 7.1
|
This command was integrated into Cisco IOS Release 12.4(24)T.
|
Usage Guidelines
This command assigns a predefined translation profile to incoming or outgoing call legs to and from the Cisco Unified SRST router. Use this command to apply the translation profile to a specific directory number or to all directory numbers on a SIP phone. The translation profile that you assign is created by using the voice translation-profile command.
Examples
The following example assigns the translation profile named "profile1" to handle translation of outgoing calls from SIP phone 21:
Router(config)# voice register pool 21
Router(config-register-pool)# translation-profile outgoing profile1
The following example assigns the translation profile named "profile2" to handle translation of incoming calls to extension 1200:
Router(config)# voice register dn 12
Router(config-register-pool)# number 1200
Router(config-register-pool)# translation-profile incoming profile2
Related Commands
Command
|
Description
|
show voice translation-profile
|
Displays the configuration of a translation profile.
|
translate (translation profiles)
|
Assigns a translation rule to a translation profile.
|
voice translation-profile
|
Defines a translation profile for voice calls.
|
trustpoint (credentials)
To specify the name of the trustpoint to be associated with a Cisco Unified Communications Manager Express (Cisco Unified CME) CTL provider certificate or with the Cisco Unified Survivable Remote Site Telephony (SRST) router certificate, use the trustpoint command in credentials configuration mode. To change the specified trustpoint, use the no form of this command.
trustpoint trustpoint-name
no trustpoint
Syntax Description
trustpoint-name
|
Name of the trustpoint to be associated with the Cisco Unified CME CTL provider certificate or the Cisco Unified SRST device certificate.
|
Command Default
No default behavior or values.
Command Modes
Credentials configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.3(14)T
|
Cisco SRST 3.3
|
This command was introduced for Cisco Unified SRST.
|
12.3(14)T
|
Cisco Unified CME 4.0
|
This command was introduced for Cisco Unified CME.
|
Usage Guidelines
Cisco Unified CME
This command is used with Cisco Unified CME phone authentication to define the trustpoint for the CTL provider. This trustpoint will be used for TLS sessions with the CTL client.
Cisco Unified SRST
The name of the trustpoint must be consistent with the trustpoint name of the Cisco Unified SRST router.
Examples
Cisco Unified CME
The following example sets up a CTL provider on the Cisco Unified CME router with the IP address 172.19.245.1.
Router(config)# credentials
Router(config-credentials)# ip source-address 172.19.245.1 port 2444
Router(config-credentials)# trustpoint ctlpv
Router(config-credentials)# ctl-service admin user4 secret 0 c89L8o
Cisco Unified SRST
The following example enters credentials configuration mode, sets the IP source address and port, and specifies the trustpoint:
Router(config)# credentials
Router(config-credentials)# ip source-address 10.6.21.4 port 2445
Router(config-credentials)# trustpoint srstca
Related Commands
Command
|
Description
|
credentials
|
Provides the Cisco Unified CME CTL provider certificate or the Cisco Unified SRST router certificate and enters credentials configuration mode.
|
ctl-service admin
|
Specifies a user name and password to authenticate the CTL client during the CTL protocol.
|
debug credentials
|
Sets debugging on the credentials service.
|
ip source-address (credentials)
|
Enables the router to receive messages through the specified IP address and port.
|
show credentials
|
Displays the credentials settings.
|
user-locale (call-manager-fallback)
To set the language by country for displays on the Cisco Unified IP Phone 7905G, Cisco Unified IP Phone 7912G, Cisco UnifiedIP Phone 7940G and Cisco UnifiedIP Phone 7960G, use the user-locale command in call-manager-fallback configuration mode. To disable the country selection and use the default country (United States), use the no form of this command.
user-locale country-code
no user-locale country-code
Syntax Description
country-code
|
The following ISO-3166 codes are available to Cisco Unified Survivable Remote Site Telephony (SRST) systems running under Cisco Unified Communications Manager V3.2 or later:
• DE—German.
• DK—Danish.
• ES—Spanish.
• FR—French.
• IT—Italian.
• JP—Japanese Katakana (available under Cisco Unified Communications Manager V4.0 or later).
• NL—Dutch.
• NO—Norwegian.
• PT—Portuguese.
• RU—Russian.
• SE—Swedish.
• US—United States English (default).
|
Defaults
The default country code is US (United States).
Command Modes
Call-manager-fallback configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(11)YT
|
Cisco SRST 2.1
|
This command was introduced for use on the Cisco IP Phone 7940 and Cisco IP Phone 7960 for Cisco SRST systems running Cisco Communications Manager V3.2. Support includes country codes for France, Germany, Italy, Portugal, Spain, and the United States (default).
|
12.3(4)T
|
Cisco SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
12.3(7)XL
|
Cisco SRST 3.1.1
|
The JP keyword was added, providing support for Japanese Katakana.
|
12.3(11)T
|
Cisco SRST 3.2
|
This command was integrated into Cisco IOS Release 12.3(11)T and support was increased to include the Cisco Unified IP Phone 7905G and Cisco Unified IP Phone 7912G.
|
Usage Guidelines
Japanese Katakana is now supported with the JP keyword and is available to Cisco Unified SRST systems running under Cisco Unified Communications Manager V4.0. All other country-code options are available to Cisco Unified SRST systems running under Cisco Unified Communications Manager V3.2. Systems running Cisco Unified Communications Manager prior to V3.2 can use only the default country, United States (US).
Examples
The following example shows how to set the user locale to the ISO-3166 code for Spain:
Router(config)# call-manager-fallback
Router(config-cm-fallback)# user-locale ES
Related Commands
Command
|
Description
|
call-manager-fallback
|
Enables Cisco Unified SRST support and enters call-manager-fallback configuration mode.
|
utf8
To define Unicode UTF-8 support for a phone type, use the utf8 command in ephone-type configuration mode. To reset to the default value, use the no form of this command.
utf8
no utf8
Syntax Description
This command has no arguments or keywords.
Command Default
Phone type supports Unicode UTF-8.
Command Modes
Ephone-type configuration (config-ephone-type)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.4(15)XZ
|
Cisco Unified CME 4.3 Cisco Unified SRST 4.3
|
This command was introduced.
|
Usage Guidelines
This command specifies whether Unicode UTF-8 is supported by the type of phone that is being added with the phone-type template.
Examples
The following example shows that UTF-8 support is set to disabled for the Nokia E61 when creating the ephone-type template:
Router(config)# ephone-type E61
Router(config-ephone-type)# no utf8
Related Commands
Command
|
Description
|
device-id
|
Specifies the device ID for a phone type.
|
type
|
Assigns the phone type to an SCCP phone.
|
vad (voice register pool)
To enable voice activity detection (VAD) on a VoIP dial peer, use the vad command in voice register pool configuration mode. To disable VAD, use the no form of this command.
vad
no vad
Syntax Description
This command has no arguments or keywords.
Defaults
Enabled
Command Modes
Voice register pool configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.4(4)T
|
Cisco CME 3.4 Cisco SIP SRST 3.4
|
This command was introduced.
|
Usage Guidelines
VAD detects periods of silence in the voice signal and temporarily discontinues transmission of the signal during these periods to save bandwidth. Because VAD is enabled by default, there is no comfort noise during periods of silence. As a result, the call may seem to be disconnected and you may prefer to set no vad on the SIP phone pool.
Examples
The following example shows how to disable VAD for pool 1:
Router(config)# voice register pool 1
Router(config-register-pool)# no vad
Related Commands
Command
|
Description
|
voice register pool
|
Enters voice register pool configuration mode for SIP phones.
|
video (call-manager-fallback)
To enter video configuration mode for a Cisco Unified SRST router, use the video command in call-manager-fallback configuration mode. To reset global video parameters, use the no form of this command.
video
no video
Syntax Description
This command has no arguments or keywords.
Command Default
Global video parameters are configured.
Command Modes
Call-manager-fallback configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.4(4)XC
|
Cisco Unified SRST 4.0
|
This command was introduced.
|
Usage Guidelines
Use the video command to enter video configuration mode and set video parameters for all applicable Cisco IP phones in a Cisco Unified SRST system.
Examples
The following example shows how to enter video configuration mode for a Cisco Unified SRST system. You must enter video configuration mode to set video parameters, such as maximum bit rate.
Router(config)# call-manager-fallback
Router(config-call-manager-fallback)# video
Router(conf-cm-fallback-video)# maximum bit-rate 256
Related Commands
Command
|
Description
|
show call active video
|
Displays call information for SCCP video calls in progress.
|
show call history video
|
Displays call history information for SCCP video calls.
|
video (telephony-service)
|
Enters video configuration mode to set video parameters for all applicable Cisco IP phones associated with a Cisco Unified CME router.
|
vm-integration
To enter voice-mail integration configuration mode and enable voice-mail integration with dual tone multifrequency (DTMF) and analog voice-mail systems, use the vm-integration command in global configuration mode. To disable voice-mail integration, use the no form of this command.
vm-integration
no vm-integration
Syntax Description
This command has no arguments or keywords.
Defaults
No voice-mail integration is defined.
Command Modes
Global configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(11)YT
|
Cisco SRST 2.1
|
This command was introduced for Cisco Survivable Remote Site Telephony (SRST).
|
12.2(2)XT
|
Cisco CME 2.0
|
This command was introduced on the following platforms: Cisco 1750, Cisco 1751, Cisco 2600 series, Cisco 3600 series, and Cisco IAD2420 series.
|
12.2(8)T
|
Cisco CME 2.0
|
This command was integrated into Cisco IOS Release 12.2(8)T and implemented on the Cisco 3725 and Cisco 3745.
|
12.2(8)T1
|
Cisco CME 2.0
|
This command was implemented on the Cisco 2600XM and Cisco 2691.
|
12.2(11)T
|
Cisco CME 2.01
|
This command was integrated into Cisco IOS Release 12.2(11)T and implemented on the Cisco 1760.
|
Usage Guidelines
The vm-integration command is used to enter voice-mail integration configuration mode. Use voice-mail integration configuration mode to integrate a Cisco Unified Communications Manager Express (Cisco Unified CME) system with an analog voice-mail system.
Examples
The following example shows how to enter the voice-mail integration configuration mode:
Router(config) vm-integration
Router(config-vm-integration) pattern direct 2 CGN *
Related Commands
Command
|
Description
|
pattern direct (vm-integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system when a user presses the Messages button on a phone.
|
pattern ext-to-ext busy (vm-integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system once an internal extension reaches a busy extension and the call is forwarded to voice mail.
|
pattern ext-to-ext no-answer (vm-integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system once an internal extension fails to connect to an extension and the call is forwarded to voice mail.
|
pattern trunk-to-ext busy (vm-integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system once an external trunk call reaches a busy extension and the call is forwarded to voice mail.
|
pattern trunk-to-ext no-answer (vm-integration)
|
Configures the DTMF digit pattern forwarding necessary to activate the voice-mail system when an external trunk call reaches an unanswered extension and the call is forwarded to voice mail.
|
voice-class codec (voice register pool)
To assign a previously configured codec selection preference list, use the voice-class codec command in voice register pool configuration mode. To remove the codec preference assignment from the voice register pool, use the no form of this command.
voice-class codec tag
no voice-class codec
Syntax Description
tag
|
Unique number assigned to the voice class. Range is from 1 to 10000. The tag number maps to the tag number created by using the voice class codec command in dial-peer configuration mode.
|
Command Default
None
Command Modes
Voice register pool configuration
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.2(15)ZJ
|
Cisco SIP SRST 3.0
|
This command was introduced.
|
12.3(4)T
|
Cisco SIP SRST 3.0
|
This command was integrated into Cisco IOS Release 12.3(4)T.
|
12.4(4)T
|
Cisco CME 3.4 Cisco SIP SRST 3.4
|
This command was added to Cisco CME.
|
Usage Guidelines
During Cisco Unified Session Initiation Protocol (SIP) Survivable Remote Site Telephony (SRST) or Cisco Unified Communications Manager Express (Cisco Unified CME) registration, a dial peer is created, and that dial peer includes codec g729r8 by default. The voice-class codec command allows you to change the automatically selected default codec, if desired.
You can assign one voice class to each voice register pool. If you assign another voice class to a pool, the last voice class assigned replaces the previous voice class.
Note
The id (voice register pool) command is required and must be configured before any other voice register pool commands. The id command identifies a locally available individual Cisco SIP IP phone or set of Cisco SIP IP phones.
Examples
The following partial sample output from the show running-config command shows that voice register pool 1 has been set up to use the previously configured codec voice class 1:
cor incoming call91 1 91011
translate-outgoing called 1
proxy 10.2.161.187 preference 1 monitor probe icmp-ping
alias 1 94... to 91011 preference 8
Related Commands
Command
|
Description
|
codec (voice register pool)
|
Specifies the codec supported by a single Cisco SIP phone or a VoIP dial peer in a Cisco Unified SIP SRST or a Cisco Unified CME environment.
|
id (voice register pool)
|
Explicitly identifies a locally available individual Cisco SIP IP phone, or when running Cisco Unified SIP SRST, set of Cisco SIP IP phones.
|
voice register pool
|
Enters voice register pool configuration mode for SIP phones.
|
voice class codec (dial-peer)
|
Assigns a previously configured codec selection preference list (codec voice class) to a VoIP dial peer.
|
voice emergency response location
To create a tag for identifying an emergency response location (ERL) for E911 services, use the voice emergency response location command in global configuration mode. To remove the ERL tag, use the no form of this command.
voice emergency response location tag
no voice emergency response location tag
Syntax Description
tag
|
Unique number that identifies this ERL tag.
|
Command Default
No ERL tag is created.
Command Modes
Global configuration (config)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.4(15)T
|
Cisco Unified CME 4.1 Cisco Unified SRST 4.1 Cisco Unified SIP SRST 4.1
|
This command was introduced. For Cisco Unified CME, this command is supported in SRST fallback mode only.
|
12.4(15)XY
|
Cisco Unified CME 4.2(1) Cisco Unified SRST 4.2(1) Cisco Unified SIP SRST 4.2(1)
|
Address and name commands introduced under voice emergency response location command. This command was added for Cisco Unified CME.
|
12.4(20)T
|
Cisco Unified CME 7.0 Cisco Unified SRST 7.0 Cisco Unified SIP SRST 7.0
|
This command was integrated into Cisco IOS Release 12.4(20)T.
|
Usage Guidelines
This command creates an ERL that identifies an area where emergency teams can quickly locate a 911 caller. The ERL definition optionally includes which ELINs are associated with the ERL and which IP phones are located in the ERL. You can define two or fewer unique IP subnets and two or fewer ELINs. If you define one ELIN, this ELIN is always used for phones calling from this ERL. If you define two ELINs, the system alternates between using each ELIN. If you define zero ELINs and phones use this ERL, the outbound calls do not have their calling numbers translated. The PSAP sees the original calling numbers for these 911 calls. You can optionally add the civic address using the address command and an address description using the name command.
Examples
In the following example, all IP phones with the IP address of 10.X.X.X or 192.168.X.X are automatically associated with this ERL. If one of the phones dials 911, its extension is replaced with 408 555-0100 before it goes to the PSAP. The PSAP will see that the caller's number is 408 555-0100. The civic address, 410 Main St, Tooly, CA, and a descriptive identifier, Bldg 3 are included.
voice emergency response location 1
subnet 1 10.0.0.0 255.0.0.0
subnet 2 192.168.0.0 255.255.0.0
address 1,408,5550100,410,Main St.,Tooly,CA
Related Commands
Command
|
Description
|
address
|
Specifies a comma separated text entry (up to 250 characters) of an ERL's civic address.
|
elin
|
Specifies a PSTN number that will replace the caller's extension.
|
name
|
Specifies a string (up to 32-characters) used internally to identify or describe the emergency response location.
|
subnet
|
Defines which IP phones are part of this ERL.
|
voice emergency response settings
To define 911 call behavior settings, use the voice emergency response settings command in global configuration mode. To remove the settings, use the no form of this command.
voice emergency response settings
no voice emergency response settings
Syntax Description
This command has no arguments or keywords.
Command Default
No default behavior or values
Command Modes
Global configuration (config)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.4(15)XY
|
Cisco Unified CME 4.2(1) Cisco Unified SRST 4.2(1) Cisco Unified SIP SRST 4.2(1)
|
This command was introduced.
|
12.4(20)T
|
Cisco Unified CME 7.0 Cisco Unified SRST 7.0 Cisco Unified SIP SRST 7.0
|
This command was integrated into Cisco IOS Release 12.4(20)T.
|
Usage Guidelines
This command enables definition of several 911 of the following call behavior settings:
•
elin: Default ELIN to use if a 911 caller's IP phone's address does not match the subnet of any location in any zone.
•
expiry: Number of minutes a 911 call is associated to an ELIN in case of a call back from the 911 operator.
•
callback: Default number to contact if a 911 call back cannot find the last 911 caller.
•
logging: Syslog informational message that is printed to the console each time an emergency call is made. This feature is enabled by default, however you can disable this feature by entering the no form of this command.
Examples
In the following example, if the 911 caller's IP phone address does not match any of the voice emergency response locations, the ELIN defined in the voice emergency response settings configuration (4085550101) is used. After the 911 call is placed to the PSAP, the PSAP has 120 minutes (2 hours) to call back 408 555-0101 to reach the 911 caller. If during a call back, the last caller's extension number cannot be found, the call is routed to extension 7500. The outbound 911 calls do not cause a syslog message to the logging facility (for example, to the local buffer, console, or remote host).
voice emergency response settings
Related Commands
Command
|
Description
|
callback
|
Default phone number to contact if a 911 callback cannot find the last 911 caller from the ERL.
|
elin
|
E.164 number used as the default ELIN if no matching ERL to the 911 caller's IP phone address is found.
|
expiry
|
Number of minutes a 911 call is associated to an ELIN in case of a callback from the 911 operator.
|
logging
|
Syslog informational message printed to the console every time an emergency call is made.
|
voice emergency response zone
To create an emergency response zone, use the voice emergency response zone command in global configuration mode. To remove the defined voice emergency response zone, use the no form of this command.
voice emergency response zone tag
no voice emergency response zone tag
Syntax Description
tag
|
Identifier (1-100) for the voice emergency response zone.
|
Command Default
No default behavior or values
Command Modes
Global configuration (config)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.4(15)XY
|
Cisco Unified CME 4.2(1) Cisco Unified SRST 4.2(1) Cisco Unified SIP SRST 4.2(1)
|
This command was introduced.
|
12.4(20)T
|
Cisco Unified CME 7.0 Cisco Unified SRST 7.0 Cisco Unified SIP SRST 7.0
|
This command was integrated into Cisco IOS Release 12.4(20)T.
|
Usage Guidelines
This command creates voice emergency response zones that allow routing of 911 calls to different PSAPs.
Examples
The following example shows an assignment of ERLs to a voice emergency response zone. The calls have an ELIN from ERLs 8, 9, and 10. The locations for ERLs in zone 10 are searched in the order each CLI was entered for a phone address match because no priority order is assigned.
voice emergency response zone 10
Related Commands
Command
|
Description
|
location
|
Identifies locations within an emergency response zone and optionally assigns a priority order to the location.
|
voice hunt-group
To create a hunt group for phones in a Cisco Unified Communications Manager Express (Cisco Unified CME) or Cisco Unified Session Initiation Protocol (SIP) Survivable Remote Site Telephony (SRST) system, use the voice hunt-group command in global configuration mode. To delete a hunt group, use the no form of this command.
voice hunt-group hunt-tag {longest-idle | parallel | peer | sequential}
no voice hunt-group hunt-tag
Syntax Description
hunt-tag
|
Unique sequence number that identifies the hunt group. Range is 1 to 100.
|
longest-idle
|
Allows an incoming call to go first to the number that has been idle the longest for the number of hops specified when the hunt group was defined. The longest-idle time is determined from the last time that a phone registered, reregistered, or went on-hook.
|
parallel
|
Allows an incoming call to simultaneously ring all the numbers in the hunt group member list.
|
peer
|
Allows a round-robin selection of the first extension to ring. Ringing proceeds in a circular manner from left to right. The round-robin selection starts with the number left of the number that answered when the hunt-group was last called.
|
sequential
|
Allows an incoming call to ring all the numbers in the left-to-right order in which they were listed when the hunt group was defined.
|
Command Default
No voice hunt group is created.
Command Modes
Global configuration (config)
Command History
Cisco IOS Release
|
Cisco Product
|
Modification
|
12.4(4)T
|
Cisco CME 3.4
|
This command was introduced.
|
12.4(15)XZ
|
Cisco Unified CME 4.3
|
This command was modified to add support for Cisco Unified SCCP IP phones.
|
12.4(20)T
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Cisco Unified CME 7.0
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This command was integrated into Cisco IOS Release 12.4(20)T.
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15.2(4)M
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Cisco Unified SIP SRST 9.1
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This command was introduced in Cisco Unified SIP SRST 9.1.
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Usage Guidelines
The voice hunt-group command enters voice hunt-group configuration mode to define a hunt group. A hunt group is a list of phone numbers that take turns receiving incoming calls to a specific number (pilot number), which is defined with the pilot command. The specific extensions included in the hunt group and the order and maximum number of extensions allowed in the list are defined with the list command.
If a number in the list is busy or does not answer, the call is redirected to the next number in the list. The last number tried is the final number, which is defined with the final command. If the number of times that a call is redirected to a new number exceeds 5, you must use the max-redirect command to increase the allowable number of redirects in the Cisco Unified CME or Cisco Unified SIP SRST system.
To configure a new hunt group, you must specify the longest-idle, parallel, peer, or sequential keyword. To change an existing hunt group configuration, the keyword is not required. To change the type of hunt group, for instance from peer to sequential or sequential to peer, you must remove the existing hunt group first by using the no form of this command and then re-create it.
The parallel keyword creates a dial peer to allow an incoming call to ring multiple phones simultaneously. The use of parallel hunt groups is also referred to as application-level forking because it enables the forking of a call to multiple destinations. A pilot dial peer cannot be used as a voice hunt group and a hunt group at the same time.
While ephone hunt groups only support Cisco Unified SCCP IP phones, a voice hunt group supports either a Cisco Unified SCCP IP phone or a Cisco Unified SIP IP phone.
With the voice hunt group feature preconfigured in the Cisco Unified SIP SRST router, voice hunt groups continue to be supported after phones fallback from a Cisco Unified Communications Manager (Cisco Unified CM) to a Cisco Unified SIP SRST router.
Examples
The following example shows how to define longest-idle hunt group 1 with pilot number 7501, final number 8000, and nine numbers in the list. After a call is redirected six times (makes 6 hops), it is redirected to the final number 8000.
Router(config)# voice hunt-group 1 longest-idle
Router(config-voice-hunt-group)# pilot 7501
Router(config-voice-hunt-group)# list 7001, 7002, 7023, 7028, 7045, 7062, 7067, 7072, 7079
Router(config-voice-hunt-group)# final 8000
Router(config-voice-hunt-group)# hops 6
Router(config-voice-hunt-group)# timeout 20
Router(config-voice-hunt-group)# exit
The following example shows how to define peer hunt group number 2. Callers dial the pilot number 5610 to reach the hunt group. The first extension to ring the first time that this hunt group is called is 5601. If 5601 does not answer, the hunt proceeds from left to right, beginning with the extension directly to the right. If none of those extensions answer, the call is forwarded to extension 6000, which is the number for the voice-mail service.
The second time someone calls the hunt group, the first extension to ring is 5602 if 5601 was answered during the previous call.
Router(config)# voice hunt-group 2 peer
Router(config-voice-hunt-group)# pilot 5610
Router(config-voice-hunt-group)# list 5601, 5602, 5617, 5633
Router(config-voice-hunt-group)# final 6000
Router(config-voice-hunt-group)# timeout 30
Router(config-voice-hunt-group)# exit
The following example shows how to define sequential hunt group number 3. When callers dial extension 5601, the first phone to ring is 5001, then 5002, 5017, and 5028. If none of those extensions answer, the call is forwarded to extension 6000, which is the number for the voice-mail service.
Router(config)# voice hunt-group 3 sequential
Router(config-voice-hunt-group)# pilot 5601
Router(config-voice-hunt-group)# list 5001, 5002, 5017, 5028
Router(config-voice-hunt-group)# final 6000
Router(config-voice-hunt-group)# timeout 30
Router(config-voice-hunt-group)# exit
The following example shows how to define a parallel hunt group. When callers dial extension 1000, extensions 1001, 1002, and so forth ring simultaneously. The first extension to answer is connected. All other call legs are disconnected. If none of the extensions answer, the call is forwarded to extension 2000, which is the number for the voice-mail service.
Router(config)# voice hunt-group 4 parallel
Router(config-voice-hunt-group)# pilot 1000
Router(config-voice-hunt-group)# list 1001, 1002, 1003, 1004
Router(config-voice-hunt-group)# final 2000
Router(config-voice-hunt-group)# timeout 20
Router(config-voice-hunt-group)# exit
Related Commands
Command
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Description
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final (voice hunt-group)
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Defines the last extension in a voice hunt group.
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hops (voice hunt-group)
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Defines the number of times that a call is redirected to the next phone number in a peer voice hunt-group list before proceeding to the final phone number.
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list (voice hunt-group)
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Defines the phone numbers that participate in a voice hunt group.
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max-redirect
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Changes the number of times that a call can be redirected by call forwarding or transfer within a Cisco Unified CME system.
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pilot (voice hunt-group)
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Defines the phone number that callers dial to reach a voice hunt group.
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timeout (voice hunt-group)
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Sets the number of seconds after which a call that is not answered is redirected to the next number in the hunt-group list and defines the last phone number in the hunt group.
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voice moh-group
To enter voice-moh-group configuration mode and set up music on hold (MOH) group parameters, use the voice moh-group command in global configuration mode. To remove the music on hold (MOH) group parameters from the configuration for SCCP IP phones, use the no form of this command.
voice moh-group moh-group tag
no voice moh-group tag
Syntax Description
tag
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Specifies a moh-group number tag (1-5) to be used for music on hold group parameters.
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Command Default
No voice-moh-group is enabled.
Command Modes
Global configuration (config)
Command History
Cisco IOS Release
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Cisco Product
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Modification
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15.0(1)XA
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Cisco Unified CME 8.0 Cisco Unified SRST 8.0
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This command was introduced.
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15.1(1)T
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Cisco Unified CME 8.0 Cisco Unified SRST 8.0
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This command was integrated into Cisco IOS Release 15.1(1)T.
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Usage Guidelines
This command enters the voice-moh-group configuration mode for configuring music on hold (MOH) group parameters for SCCP IP phones in Cisco Unified CME or in Cisco Unified SRST.
Examples
The following example shows how to enter voice-moh-group configuration mode for configuring a moh group in Cisco Unified CME. This example also includes the command to configure a music on hold (MOH) flash file for this voice-moh- group.
Router(config)# voice-moh-group 1
Router(config-voice-moh-group)#moh minuet.wav
Related Commands
moh
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Enables music on hold from a flash audio feed.
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multicast moh
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Enables multicast of the music-on-hold audio stream.
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extension-range
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Defines extension range for a clients calling a voice-moh-group.
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voice register global
To enter voice register global configuration mode in order to set global parameters for all supported Cisco SIP IP phones in a Cisco Unified CME or Cisco Unified Session Initiation Protocol (SIP) Survivable Remote Site Telephony (SRST) environment, use the voice register global command in global configuration mode. To automatically remove the existing DNs, pools, and global dialplan patterns, use the no form of this command.
voice register global
no voice register global
Syntax Description
This command has no arguments or keywords.
Command Default
There are no system-level parameters configured for SIP IP phones.
Command Modes
Global configuration (config)
Command History
Cisco IOS Release
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Cisco Product
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Modification
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12.4(4)T
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Cisco CME 3.4 Cisco SIP SRST 3.4
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This command was introduced.
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15.0(1)XA
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Cisco SIP SRST 8.0
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This command was updated to display the signaling transport protocol.
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15.1(2)T
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Cisco Unified CME 8.1 Cisco Unified SRST 8.1
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The no form of the command was modified.
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Usage Guidelines
Cisco Unified CME
Use this command to set provisioning parameters for all supported SIP phones in a Cisco Unified CME system.
Cisco Unified SIP SRST
Use this command to set provisioning parameters for multiple pools; that is, all supported Cisco SIP IP phones in a SIP SRST environment.
Cisco Unified CME 8.1 enhances the no form of voice register global command. The no voice register global command clears global configuration along with pools and DN configuration and also removes the configurations for voice register template, voice register dialplan, and voice register session-server. A confirmation is sought before the cleanup is made.
In Cisco Unified SRST 8.1 and later versions, the no voice register global command removes pools and DNs along with the global configuration.
Examples1
Cisco Unified CME
The following is a partial sample output from the show voice register global command. All of the parameters listed were set under voice register global configuration mode:
Router# show voice register global
CONFIG [Version=4.0(0)]
========================
Version 4.0(0)
Mode is cme
Max-pool is 48
Max-dn is 48
Source-address is 10.0.2.4 port 5060
Load 7960-40 is P0S3-07-4-07
Time-format is 12
Date-format is M/D/Y
Time-zone is 5
Hold-alert is disabled
Mwi stutter is disabled
Mwi registration for full E.164 is disabled
Dst auto adjust is enabled
start at Apr week 1 day Sun time 02:00
stop at Oct week 8 day Sun time 02:00
Examples2
Cisco Unified CME and Cisco Unified SRST
The following is a sample output from no voice register global command:
Router(config)# no voice register global
This will remove all the existing DNs, Pools, Templates,
Dialplan-Patterns, Dialplans and Feature Servers on the system.
Are you sure you want to proceed? Yes/No? [no]:
Related Commands
Command
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Description
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allow connections sip to sip
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Allows connections between SIP endpoints in a Cisco multiservice IP-to-IP gateway.
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application (voice register global)
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Selects the session-level application for all dial peers associated with SIP phones.
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mode (voice register global)
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Enables the mode for provisioning SIP phones in a Cisco Unified system.
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voice register pool
To enter voice register pool configuration mode for SIP phones, use the voice register pool command in global configuration mode. To remove the pool configuration, use the no form of this command.
voice register pool pool-tag
no voice register pool pool-tag
Syntax Description
pool-tag
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Unique number assigned to the pool. Range is 1 to 100.
Note For Cisco Unified Communications Manager Express (Cisco Unified CME) systems, the upper limit for this argument is defined by the max-pool command.
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Defaults
No default behavior or values
Command Modes
Global configuration
Command History
Cisco IOS Release
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Cisco Product
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Modification
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12.2(15)ZJ
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Cisco SIP SRST 3.0
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This command was introduced.
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12.3(4)T
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Cisco SIP SRST 3.0
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This command was integrated into Cisco IOS Release 12.3(4)T.
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12.4(4)T
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Cisco CME 3.4 Cisco SIP SRST 3.4
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This command was added to Cisco CME.
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Usage Guidelines
Cisco Unified CME
Use this command to set phone-specific parameters for SIP phones in a Cisco Unified CME system. Before using this command, enable the mode cme command and set the maximum number of SIP phones supported in your system by using the max-pool command.
Cisco Unified SIP SRST
Use this command to enable user control on which registrations are to be accepted or rejected by a SIP SRST device. The voice register pool command mode can be used for specialized functions and to restrict registrations on the basis of MAC, IP subnet, and number range parameters.
Examples
Cisco Unified CME
The following example shows how to enter voice register pool configuration mode and forward calls to extension 9999 when extension 2001 is busy:
Router(config)# voice register pool 10
Router(config-register-pool)# type 7960
Router(config-register-pool)# number 1 2001
Router(config-register-pool)# call-forward busy 9999 mailbox 1234
Cisco Unified SIP SRST
The following partial sample output from the show running-config command shows that several voice register pool commands are configured within voice register pool 3:
id network 10.2.161.0 mask 255.255.255.0
number 1 95... preference 1
cor outgoing call95 1 95011
Related Commands
Command
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Description
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max-pool (voice register global)
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Sets the maximum number of SIP phones that are supported by a Cisco Unified CME system.
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mode (voice register global)
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Enables the mode for provisioning SIP phones in a Cisco Unified CME system.
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number (voice register pool)
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Configures a valid number for a SIP phone.
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type (voice register pool)
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Defines a Cisco IP phone type.
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voice register dn
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Enters voice register dn configuration mode to define an extension for a SIP phone line.
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voice register global
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Enters voice register global configuration mode in order to set global parameters for all supported Cisco SIP phones in a Cisco Unified CME or Cisco Unified SIP SRST environment.
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voicemail (call-manager-fallback)
To configure the telephone number that is speed-dialed when the messages button on a Cisco IP phone is pressed, use the voicemail command in call-manager-fallback configuration mode. To disable the messages button, use the no form of this command.
voicemail phone-number
no voicemail
Syntax Description
phone-number
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Phone number configured as a speed-dial number for retrieving messages.
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Defaults
No phone number is configured, and the messages button is ineffective.
Command Modes
Call-manager-fallback configuration
Command History
Cisco IOS Release
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Cisco Product
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Modification
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