Table Of Contents
Configuring Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode (for Version 3.4 and Version 4.0 Only)
Contents
Prerequisites for Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode
Restrictions for Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode
Information About Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode
Cisco Unified SIP SRST and Cisco SIP CallManager Express Feature Crossover
How to Configure Cisco Unified SIP SRST
Configuring SIP Phone Features
Configuring SIP-to-SIP Call Forwarding
Configuring Call Blocking Based on Time of Day, Day of Week, or Date
SIP Call Hold and Resume
Configuration Examples for Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode
Cisco Unified SIP SRST: Example
Configuring Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode (for Version 3.4 and Version 4.0 Only)
Note 
Prior to version 4.0, the name of this product was Cisco SIP SRST.
This chapter describes Cisco Unified Cisco Unified Survivable Remote Site Telephony (SRST) support for standardized RFC 3261 features for SIP phones. Features include call blocking and call forwarding.
Note The Cisco IOS Voice Configuration Library includes a standard library preface, glossary, and feature and troubleshooting documents and is located at http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/vcl.htm.
Contents
• Prerequisites for Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode
• Restrictions for Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode
• Information About Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode
• How to Configure Cisco Unified SIP SRST
• Configuration Examples for Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode
Prerequisites for Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode
•Complete the prerequisites documented in the "Prerequisites for Configuring Cisco Unified SIP SRST" section in the "Cisco Unified SIP SRST Feature Overview" chapter.
•Complete the necessary tasks found in the "Getting Started" chapter. Specific tasks include the required task that is documented in the "Enabling SIP-to-SIP Connection Capabilities" section on page 21.
•Configure the SIP registrar. The SIP registrar gives users control of accepting or rejecting registrations. To configure acceptance of incoming SIP Register messages, see the "Configuring the SIP Registrar" section on page 24.
Restrictions for Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode
See the restrictions documented in the "Restrictions for Configuring Cisco Unified SIP SRST" section in the "Cisco Unified SIP SRST Feature Overview" chapter.
Information About Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode
A Cisco Unified SRST system can now support SIP phones with standard-based RFC 3261 feature support locally and across SIP WAN networks. With Cisco Unified SIP SRST, SIP phones can place calls across SIP networks with similar features, as SCCP phones do. For example, most SCCP phone features such as caller ID, speed dial, and redial are supported now on SIP networks, which gives users the opportunity to choose SCCP or SIP.
Cisco Unified SIP SRST also uses a back-to-back user agent (B2BUA), which is a separate call agent that has more features than Cisco SIP SRST 3.0, which used a redirect server that only accepted and forwarded calls. The main advantage of a B2BUA call agent is in call forwarding, because it forwards calls on behalf of the phone. In addition, it maintains a presence as call middleman in the call path.
Cisco SIP SRST 3.4 supports the following call combinations:
•SIP phone to SIP phone
•SIP phone to PSTN / router voice port
•SIP phone to SCCP phone
See Figure 1 on page 6 and Figure 2 on page 7 for an illustration of Cisco Unified SIP SRST using a B2BUA.
Cisco Unified SIP SRST and Cisco SIP CallManager Express Feature Crossover
Cisco Unified SIP SRST uses is a voice register dn configuration mode. However, in a typical Cisco Unified SIP SRST setup, voice register dn commands are not used, so they are not discussed in this book. Although you are not restricted from using voice register dn commands, they are not likely to be needed in a Cisco Unified SIP SRST environment. The voice register dn commands are most likely to be used in a Cisco Unified SIP CallManager Express (CME) environment. If you work in a Cisco Unified SIP CME environment and would like to know which commands are also applicable to Cisco Unified SIP SRST, Table 5 lists Version 3.4 commands for CME and SRST. Commands marked under the column "Cisco (SIP) CME Mode Only" show up if mode cme is configured in voice register global configuration mode; these commands apply to Cisco CME only.
Procedures for configuring Cisco Unified SIP CME and complete descriptions of all CME and voice register dn commands are found in the Cisco CallManager Express Version 3.4 documentation.
Note Table 5 is not all-inclusive; additional commands may exist.
Table 5 Version 3.4 New or Enhanced Commands for Cisco Unified SRST and Cisco Unified CME (Sorted by Configuration Mode)
Function—>Command
|
Dial Peer
|
Voice Register Mode
|
Configurable for Cisco Unified (SIP ) CME and Cisco Unified SIP SRST
|
Applicable to Cisco Unified (SIP) CME Only
|
after-hour exempt
|
X
|
dn
|
X
|
—
|
auto-answer
|
—
|
dn
|
—
|
X
|
call forward
|
X
|
dn
|
X
|
—
|
huntstop
|
X
|
dn
|
X
|
—
|
label
|
—
|
dn
|
—
|
X
|
name
|
—
|
dn
|
—
|
X
|
number
|
X
|
dn
|
X
|
—
|
preference
|
X
|
dn
|
X
|
—
|
application
|
X
|
global
|
X
|
—
|
authenticate
|
—
|
global
|
—
|
X
|
create
|
—
|
global
|
—
|
X
|
date-format
|
—
|
global
|
—
|
X
|
dst
|
—
|
global
|
—
|
X
|
external ring
|
—
|
global
|
X
|
—
|
file
|
—
|
global
|
—
|
X
|
hold-alert
|
—
|
global
|
—
|
X
|
load
|
—
|
global
|
—
|
X
|
logo
|
—
|
global
|
—
|
X
|
max-dn
|
—
|
global
|
X
|
—
|
max-pool
|
—
|
global
|
X
|
—
|
max-redirect
|
—
|
global
|
—
|
X
|
mode
|
—
|
global
|
X
|
—
|
mwi
|
—
|
global
|
—
|
X
|
reset
|
—
|
global
|
—
|
X
|
tftp-path
|
—
|
global
|
—
|
X
|
timezone
|
—
|
global
|
—
|
X
|
upgrade
|
—
|
global
|
—
|
X
|
url
|
—
|
global
|
—
|
X
|
voicemail
|
—
|
global
|
—
|
X
|
after-hour exempt
|
X
|
pool
|
X
|
—
|
application
|
X
|
pool
|
X
|
—
|
call-forward
|
—
|
pool
|
X
|
—
|
call-waiting
|
—
|
pool
|
—
|
X
|
codec
|
X
|
pool
|
X
|
—
|
description
|
—
|
pool
|
—
|
X
|
dnd-control
|
—
|
pool
|
—
|
X
|
dtmf-relay
|
—
|
pool
|
X
|
—
|
id
|
—
|
pool
|
X
|
—
|
keep-conference
|
—
|
pool
|
—
|
X
|
max-pool
|
—
|
pool
|
X
|
—
|
number
|
X
|
pool
|
X
|
—
|
preference
|
X
|
pool
|
X
|
—
|
proxy
|
X
|
pool
|
X
|
—
|
reset
|
—
|
pool
|
—
|
X
|
speed-dial
|
—
|
pool
|
—
|
X
|
template
|
—
|
pool
|
—
|
X
|
translate-outgoing
|
X
|
pool
|
X
|
—
|
type
|
—
|
pool
|
—
|
X
|
username
|
—
|
pool
|
—
|
X
|
vad
|
X
|
pool
|
X
|
—
|
anonymous
|
—
|
template
|
—
|
X
|
caller-id
|
—
|
template
|
—
|
X
|
conference
|
—
|
template
|
—
|
X
|
dnd-control
|
—
|
template
|
—
|
X
|
forward
|
—
|
template
|
—
|
X
|
transfer
|
—
|
template
|
—
|
X
|
How to Configure Cisco Unified SIP SRST
This section contains the following procedures:
• Configuring SIP Phone Features (optional)
• Configuring SIP-to-SIP Call Forwarding (required)
• Configuring Call Blocking Based on Time of Day, Day of Week, or Date (required)
• SIP Call Hold and Resume (no confguration necessary)
Configuring SIP Phone Features
Once a voice register pool has been set, this procedure adds optional features to increase functionality. Some features can be made per pool or globally.
In voice register pool configuration, you can now configure several new options per pool (a pool can be one phone or a group of phones). There is also a new voice register global configuration mode for Cisco Unified SIP SRST. In voice register global mode, you can globally assign characteristics to phones.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register global tag
4. max-pool max-voice-register-pools
5. application application-name
6. external ring {bellcore-dr1 | bellcore-dr2 | bellcore-dr3 | bellcore-dr4 | bellcore-dr5}
7. exit
8. voice register pool tag
9. no vad
10. codec codec-type [bytes]
11. end
DETAILED STEPS
| |
Command or Action
|
Purpose
|
Step 1
|
enable
Example:
Router> enable
|
Enables privileged EXEC mode.
•Enter your password if prompted.
|
Step 2
|
configure terminal
Example:
Router# configure terminal
|
Enters global configuration mode.
|
Step 3
|
voice register global tag
Example:
Router(config)# voice register global 12
|
Enters voice register global configuration mode to set global parameters for all supported Cisco SIP IP phones in a Cisco Unified SIP SRST environment.
|
Step 4
|
max-pool max-voice-register-pools
Example:
Router(config-register-global)# max-pool 10
|
Sets the maximum number of SIP voice register pools that are supported in a Cisco Unified SIP SRST environment. The max-voice-register-pools argument represents the maximum number of SIP voice register pools supported by the Cisco Unified SIP SRST router. The upper limit of voice register pools is version- and platform-dependent; see Cisco IOS command-line interface (CLI) help. Default is 0.
|
Step 5
|
application application-name
Example:
Router(config-register-global)# application
global_app
|
Selects the session-level application for all dial peers associated with SIP phones. Use the application-name argument to define a specific interactive voice response (IVR) application.
|
Step 6
|
external-ring {bellcore-dr1 | bellcore-dr2 |
bellcore-dr3 | bellcore-dr4 | bellcore-dr5}
Example:
Router(config-register-global)# external-ring
bellcore-dr1
|
Specifies the type of ring sound used on Cisco SIP or Cisco SCCP IP phones for external calls. Each bellcore-dr 1-5 keyword supports standard distinctive ringing patterns as defined in the standard GR-506-CORE, LSSGR: Signaling for Analog Interfaces.
|
Step 7
|
exit
Example:
Router(config-register-global)# exit
|
Exits voice register global configuration mode.
|
Step 8
|
voice register pool tag
Example:
Router(config)# voice register pool 20
|
Enters voice register pool configuration mode for SIP phones.
•Use this command to control which phone registrations are to be accepted or rejected by a Cisco Unified SIP SRST device.
|
Step 9
|
no vad
Example:
Router(config-register-pool)# no vad
|
Disables voice activity detection (VAD) on the VoIP dial peer.
•VAD is enabled by default. Because there is no comfort noise during periods of silence, the call may seem to be disconnected. You may prefer to set no vad on the SIP phone pool.
|
Step 10
|
codec codec-type [bytes]
Example:
Router(config-register-pool)# codec g729r8
|
Specifies the codec supported by a single SIP phone or a VoIP dial peer in a Cisco Unified SIP SRST environment. The codec-type argument specifies the preferred codec and can be one of the following:
•g711alaw—G.711 a-law 64,000 bps.
•g711ulaw—G.711 mu-law 64,000 bps.
•g729r8—G.729 8000 bps (default).
The bytes argument is optional and specifies the number of bytes in the voice payload of each frame
|
Step 11
|
end
Example:
Router(config-register-pool)# end
|
Returns to privileged EXEC mode.
|
Configuring SIP-to-SIP Call Forwarding
SIP-to-SIP call forwarding (call routing) is available. Call forwarding is provided either by the phone or by using a back-to-back user agent (B2BUA), which allows call forwarding on any dial peer. Calls into a SIP device may be forwarded to other SIP or SCCP devices (including Cisco Unity, third-party voice-mail systems, or an auto attendant or IVR system such as IPCC and IPCC Express). In addition, SCCP IP phones may be forwarded to SIP phones.
Cisco Unity or other voice messaging systems connected by a SIP trunk or SIP user agent are able to pass a message-waiting indicator (MWI) when a message is left. The SIP phone then displays the MWI when indicated by the voice messaging system.
Note SIP-to-H.323 call forwarding is not supported.
To configure SIP-to-SIP call forwarding, you must first allow connections between specific types of endpoints in a Cisco IP-to-IP gateway. The allow-connections command grants this capability. For more information on setting the allow-connections command, see the "Enabling SIP-to-SIP Connection Capabilities" section on page 21. Once the SIP-to-SIP connections are allowed, you can configure call forwarding under an individual SIP phone pool. Any of the following commands can be used to configure call forwarding, according to your needs:
•Under voice register pool
–call-forward b2bua all directory-number
–call-forward b2bua busy directory-number
–call-forward b2bua mailbox directory-number
–call-forward b2bua noan directory-number [timeout seconds]
In a typical Cisco Unified SIP SRST setup, the call-forward b2bua mailbox command is not used; however it is likely to be used in a Cisco Unified SIP CallManager Express (CME) environment. Detailed procedures for configuring the call-forward b2bua mailbox command are found in Cisco CallManager Express Version 3.4 documentation.
SUMMARY STEPS
1. enable
2. configure terminal
3. voice register pool tag
4. call-forward b2bua all directory-number
5. call-forward b2bua busy directory-number
6. call-forward b2bua mailbox directory-number
7. call-forward b2bua noan directory-number timeout seconds
8. end
DETAILED STEPS
| |
Command or Action
|
Purpose
|
Step 1
|
enable
Example:
Router> enable
|
Enables privileged EXEC mode.
•Enter your password if prompted.
|
Step 2
|
configure terminal
Example:
Router# configure terminal
|
Enters global configuration mode.
|
Step 3
|
voice register pool tag
Example:
Router(config)# voice register pool 15
|
Enters voice register pool configuration mode.
•Use this command to control which phone registrations are accepted or rejected by a Cisco Unified SIP SRST device.
|
Step 4
|
call-forward b2bua all directory-number
Example:
Router(config-register-pool)# call-forward
b2bua all 5005
|
Enables call forwarding for a SIP back-to-back user agent (B2BUA) so that all incoming calls are forwarded to another extension:
•directory-number—Telephone number to which calls are forwarded. Represents a fully qualified E.164 number. Maximum length of the telephone number is 32.
|
Step 5
|
call-forward b2bua busy directory-number
Example:
Router(config-register-pool)# call-forward
b2bua busy 5006
|
Enables call forwarding for a SIP B2BUA so that incoming calls to a busy extension are forwarded to another extension.
•directory-number—Telephone number to which calls are forwarded. Represents a fully qualified E.164 number. Maximum length of the telephone number is 32.
|
Step 6
|
call-forward b2bua mailbox directory-number
Example:
Router(config-register-pool)# call-forward
b2bua mailbox 5007
|
Controls the specific voice-mail box selected in a voice-mail system at the end of a call forwarding exchange.
•directory-number—Telephone number to which calls are forwarded when the forwarded destination is busy or does not answer. Represents a fully qualified E.164 number. Maximum length of the telephone number is 32.
|
Step 7
|
call-forward b2bua noan directory-number
timeout seconds
Example:
Router(config-register-pool)# call-forward
b2bua noan 5010 timeout 10
|
Enables call forwarding for a SIP B2BUA so that incoming calls to an extension that does not answer after a configured amount of time are forwarded to another extension.
This command is used if a phone is registered with a Cisco Unified SIP SRST router, but the phone is not reachable because there is no IP connectivity (there is no response to Invite requests).
•directory-number—Telephone number to which calls are forwarded. Represents a fully qualified E.164 number. Maximum length of the telephone number is 32.
•timeout seconds—Duration, in seconds, that a call can ring with no answer before the call is forwarded to another extension. Range is 3 to 60000. The default value is 20.
|
Step 8
|
end
Example:
Router(config-register-pool)# end
|
Returns to privileged EXEC mode.
|
Configuring Call Blocking Based on Time of Day, Day of Week, or Date
Call blocking prevents the unauthorized use of phones and is implemented by matching a pattern of up to 32 digits during a specified time of day, day of week, or date. Cisco Unified SIP SRST provides SIP endpoints the same time-based call blocking mechanism that is currently provided for SCCP phones. The call blocking feature supports all incoming calls, including incoming SIP and analog FXS calls.
Note Pin-based exemptions and the "Login" toll-bar override are not supported in Cisco Unified SIP SRST.
The commands used for SIP phone call blocking are the same commands that are used for SCCP phones on your Cisco Unified SRST system. The Cisco SRST session application accesses the current after-hours configuration under call-manager-fallback mode and applies it to calls originated by Cisco SIP phones that are registered to the Cisco SRST router. The commands used in call-manager-fallback mode that set block criteria (time/date/block pattern) are the following:
•after-hours block pattern pattern-tag pattern [7-24]
•after-hours day day start-time stop-time
•after-hours date month date start-time stop-time
When a user attempts to place a call to digits that match a pattern that has been specified for call blocking during a time period that has been defined for call blocking, the call is immediately terminated and the caller hears a fast busy.
In SRST (call-manager-fallback configuration mode), there is no phone- or pin-based exemption to after-hours call blocking. However, in Cisco Unified SIP SRST (voice register pool mode), individual IP phones can be exempted from all call blocking using the after-hours exempt command.
SUMMARY STEPS
1. enable
2. configure terminal
3. call-manager-fallback
4. after-hours block pattern tag pattern [7-24]
5. after-hours day day start-time stop-time
6. after-hours date month date start-time stop-time
7. exit
8. voice register pool tag
9. after-hour exempt
10. end
DETAILED STEPS
| |
Command or Action
|
Purpose
|
Step 1
|
enable
Example:
Router> enable
|
Enables privileged EXEC mode.
•Enter your password if prompted.
|
Step 2
|
configure terminal
Example:
Router# configure terminal
|
Enters global configuration mode.
|
Step 3
|
call-manager-fallback
Example:
Router(config)# call-manager-fallback
|
Enters call-manager-fallback configuration mode.
|
Step 4
|
after-hours block pattern tag pattern [7-24]
Example:
Router(config-cm-fallback)# after-hours block
pattern 1 91900
|
Defines a pattern of outgoing digits to be blocked. Up to 32 patterns can be defined, using individual commands.
•If the 7-24 keyword is specified, the pattern is always blocked, 7 days a week, 24 hours a day.
•If the 7-24 keyword is not specified, the pattern is blocked during the days and dates that are defined using the after-hours day and after-hours date commands.
|
Step 5
|
after-hours day day start-time stop-time
Example:
Router(config-cm-fallback)# after-hours day mon
19:00 07:00
|
Defines a recurring time period based on the day of the week during which calls are blocked to outgoing dial patterns that are defined using the after-hours block pattern command.
•day—Day of the week abbreviation. The following are valid day abbreviations: sun, mon, tue, wed, thu, fri, sat.
•start-time stop-time—Beginning and ending times for call blocking, in an HH:MM format using a 24-hour clock. If the stop time is a smaller value than the start time, the stop time occurs on the day following the start time. For example, "mon 19:00 07:00" means "from Monday at 7 p.m. until Tuesday at 7 a.m."
The value 24:00 is not valid. If 00:00 is entered as a stop time, it is changed to 23:59. If 00:00 is entered for both start time and stop time, calls are blocked for the entire 24-hour period on the specified date.
|
Step 6
|
after-hours date month date start-time
stop-time
Example:
Router(config-cm-fallback)# after-hours date
jan 1 00:00 00:00
|
Defines a recurring time period based on month and date during which calls are blocked to outgoing dial patterns that are defined using the after-hours block pattern command.
•month—Month abbreviation. The following are valid month abbreviations: jan, feb, mar, apr, may, jun, jul, aug, sep, oct, nov, dec.
•date—Date of the month. Range is from 1 to 31.
•start-time stop-time—Beginning and ending times for call blocking, in an HH:MM format using a 24-hour clock. The stop time must be larger than the start time.
The value 24:00 is not valid. If 00:00 is entered as a stop time, it is changed to 23:59. If 00:00 is entered for both start time and stop time, calls are blocked for the entire 24-hour period on the specified date.
|
Step 7
|
exit
Example:
Router(config-cm-fallback)# exit
|
Exits call-manager-fallback configuration mode.
|
Step 8
|
voice register pool tag
Example:
Router(config)# voice register pool 12
|
Enters voice register pool configuration mode.
•Use this command to control which registrations are accepted or rejected by a Cisco Unified SIP SRST device.
|
Step 9
|
after-hour exempt
Example:
Router(config-register-pool)# after-hour exempt
|
Specifies that for a particular voice register pool, none its outgoing calls are blocked even though call blocking is enabled.
|
Step 10
|
end
Example:
Router(config-register-pool)# end
|
Returns to privileged EXEC mode.
|
Examples
The following example defines several patterns of digits for which outgoing calls are blocked. Patterns 1 and 2, which block calls to external numbers that begin with 1 and 011, are blocked on Monday through Friday before 7 a.m. and after 7 p.m. Pattern 3 blocks calls to 900 numbers 7 days a week, 24 hours a day.
after-hours block pattern 1 91
after-hours block pattern 2 9011
after-hours block pattern 3 91900 7-24
after-hours day mon 19:00 07:00
after-hours day tue 19:00 07:00
after-hours day wed 19:00 07:00
after-hours day thu 19:00 07:00
after-hours day fri 19:00 07:00
The following example exempts a Cisco SIP phone pool from the configured blocking criteria:
Verification
To verify the feature's configuration, enter one of the following commands:
•show voice register dial-peer—Displays all the dial peers created dynamically by phones that have registered. This command also displays configurations for after hours blocking and call forwarding.
•show voice register pool <tag>—Displays information regarding a specific pool.
•debug ccsip message—Debugs basic B2BUA calls.
SIP Call Hold and Resume
Cisco Unified SRST supports the ability for SIP phones to place calls on hold and to resume from calls placed on hold. This also includes support for a consultative hold where A calls B, B places A on hold, B calls C, and B disconnects from C and then resumes with A. Support for call hold is signaled by SIP phones using "re-INVITE c=0.0.0.0" and also by the receive-only mechanism.
No configuration is necessary.
Note Music on hold (MOH) is not supported for call hold invoked from a SIP phone. A caller hears only silence when placed on hold by a SIP phone.
Configuration Examples for Cisco Unified SIP SRST Features Using Back-to-Back User Agent Mode
This section provides the following configuration example.
• Cisco Unified SIP SRST: Example
Note IP addresses and hostnames in examples are fictitious.
Cisco Unified SIP SRST: Example
This section provides a configuration example to match the configuration tasks in the previous sections.
Router# show running-config
Building configuration...
Current configuration : 1462 bytes
configuration mode exclusive manual
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
service internal
!
boot-start-marker
boot-end-marker
!
logging buffered 8000000 debugging
!
no aaa new-model
!
resource policy
!
clock timezone edt -5
clock summer-time edt recurring
ip subnet-zero
!
!
!
ip cef
!
!
!
voice-card 0
no dspfarm
!
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
registrar server expires max 600 min 60
!
!
!
voice register global
max-dn 10
max-pool 10
!
! Define call forwarding under a voice register pool
voice register pool 1
id mac 0012.7F57.60AA
number 1 1000
call-forward b2bua all 2412
call-forward b2bua busy 2413
call-forward b2bua noan 2414 timeout 30
codec g711ulaw
!
voice register pool 2
id mac 0012.7F3B.9025
number 1 2800
codec g711ulaw
!
voice register pool 3
id mac 0012.7F57.628F
number 1 2801
codec g711ulaw
!
!
!
interface GigabitEthernet0/0
ip address 10.0.2.99 255.255.255.0
duplex auto
speed auto
!
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
ip classless
ip route 0.0.0.0 0.0.0.0 GigabitEthernet0/0
!
ip http server
!
!
!
control-plane
!
!
!
dial-peer voice 1000 voip
destination-pattern 24..
session protocol sipv2
session target ipv4:10.0.2.5
codec g711ulaw
!
! Define call blocking under call-manager-fallback mode
call-manager-fallback
max-conferences 4 gain -6
after-hours block pattern 1 2417
after-hours date Dec 25 12:01 20:00
!
!
line con 0
exec-timeout 0 0
line aux 0
line vty 0 4
login
!
scheduler allocate 20000 1000
ntp server 10.0.2.10
!
end
