Configuring Call Transfer and Forwarding

Last Updated: March 15, 2013

This chapter describes call transfer and forwarding features in Cisco Unified Communications Manager
Express (Cisco Unified CME) to enable interworking with various network requirements.

Finding Feature Information in This Module

Your Cisco Unified CME version may not support all of the features documented in this module. For a
list of the versions in which each feature is supported, see the “Feature Information for Call Transfer and
Forwarding” section on page 834.

Contents
• Information About Call Transfer and Forwarding, page 755
• How to Configure Call Transfer and Forwarding, page 778
• Configuration Examples for Call Transfer and Forwarding, page 822
• Where to Go Next, page 832
• Additional References, page 832
• Feature Information for Call Transfer and Forwarding, page 834

Information About Call Transfer and Forwarding

To configure transfer and forwarding features, you should understand the following concepts:
• Call Forwarding, page 756
• Call Forward Unregistered, page 757
• B2BUA Call Forwarding for SIP Devices, page 758
• Call Forward All Synchronization for SIP Phones, page 758
• Call Transfer, page 759
• Trunk-to-Trunk Transfer Blocking for Toll Fraud Prevention on Cisco Unified SIP IP Phones,
page 760
• H.450.2 and H.450.3 Support, page 764

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• Transfer Method Recommendations by Cisco Unified CME Version, page 767

• H.450.12 Support, page 768
• Hairpin Call Routing, page 768
• H.450 Tandem Gateways, page 771
• Dial Peers, page 773
• QSIG Supplementary Services, page 773
• Disabling SIP Supplementary Services for Call Forward and Call Transfer, page 774
• Typical Network Scenarios for Call Transfer and Call Forwarding, page 775

Call Forwarding
Call forwarding diverts calls to a specified number under one or more of the following conditions:
• All calls—When all-call call forwarding is activated by a phone user, all incoming calls are diverted.
The target destination for diverted calls can be specified in the router configuration or by the phone
user with a soft key or feature access code. The most recently entered destination is recognized by
Cisco Unified CME, regardless of how it was entered.
• No answer—Incoming calls are diverted when the extension does not answer before the timeout
expires. The target destination for diverted calls is specified in the router configuration.
• Busy—Incoming calls are diverted when the extension is busy and call waiting is not active. The
target destination for diverted calls is specified in the router configuration.
• Night service—All incoming calls are automatically diverted during night-service hours. The target
destination for diverted calls is specified in the router configuration.
A directory number can have all four types of call forwarding defined at the same time with a different
forwarding destination defined for each type of call forwarding. If more than one type of call forwarding
is active at one time, the order for evaluating the different types is as follows:
1. Call forward night-service
2. Call forward all
3. Call forward busy and call forward no-answer
H.450.3 capabilities are enabled globally on the router by default, and can be disabled either globally or
for individual dial peers. You can configure incoming patterns for using the H.450.3 standard.
Calling-party numbers that do not match the patterns defined with this command are forwarded using
Cisco-proprietary call forwarding for backward compatibility. For information about configuring
H.450.3 on a Cisco Unified CME system, see the “SCCP: Enabling Call Forwarding for a Directory
Number” section on page 784.

Selective Call Forwarding

You can apply call forwarding to a busy or no-answer directory number based on the number that is
dialed to reach the directory number: the primary number, the secondary number, or either of those
numbers expanded by a dial-plan pattern.
Cisco Unified CME automatically creates one POTS dial peer for each ephone-dn when it is assigned a
primary number. If the ephone-dn is assigned a secondary number, it creates a second POTS dial peer.
If the dialplan-pattern command is used to expand the primary and secondary numbers for ephone-dns,
it creates two more dial peers, resulting in the creation of the following four dial peers for the ephone-dn:

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• A POTS dial peer for the primary number

• A POTS dial peer for the secondary number
• A POTS dial peer for the primary number as expanded by the dialplan-pattern command
• A POTS dial peer for the secondary number as expanded by the dialplan-pattern command
Call forwarding is normally applied to all dial peers created for an ephone-dn. Selective call forwarding
allows you to apply call forwarding for busy or no-answer calls only for the dial peers you have specified,
based on the called number that was used to route the call to the ephone-dn.
For example, the following commands set up a single ephone-dn (ephone-dn 5) with four dial
peers:
telephony-service
dialplan-pattern 1 40855501.. extension-length 4 extension-pattern 50..

ephone-dn 5
number 5066 secondary 5067

In this example, selective call forwarding can be applied so that calls are forwarded when:
• callers dial the primary number 5066.
• when callers dial the secondary number 5067.
• when callers dial the expanded numbers 4085550166 or 4085550167.
For configuration information, see the “SCCP: Enabling Call Forwarding for a Directory Number”
section on page 784.

Call Forward Unregistered

The Call Forward Unregistered (CFU) feature allows you to forward a call to a different number if the
directory number (DN) is not associated with a phone or if the associated phone is not registered to
Cisco Unified CME. The CFU feature is very useful for wireless phone users when the wireless phone
is out of the access point or phone shuts down automatically because of an automatic shutdown feature.
The service is not available and the call can be forwarded to the CFU destination. Any unregistered or
floating DN can be forwarded using the CFU feature.
An unregistered DN indicates that none of its associated phones are registered to the
Cisco Unified CME. A registered phone will become unregistered when the Cisco Unified CME sends
an unregistration request or responses to a phone's unregistration request. Cisco Unified CME sends an
unregistration request under the following circumstances:
– When the keepalive timer expires.
– When a user issues a reset or restart command on the phone.
– When an extension mobility (EM) user logs into the phone. (All DNs configured under the
logout-profile are unregistered except for the shared ones that are associated with other
registered phones.)
– When an EM user logs out of the phone. (All DNs configured under the user-profile are
unregistered except for the shared ones that are associated with other registered phones.)
There is always a gap between the time the phone loses its connection with Cisco Unified CME and the
time when Cisco Unified CME claims the phone is unregistered. The length of the gap depends on the
keepalive timer. Cisco Unified CME considers the phone as registered and tries to associate DNs until
the keepalive timer expires. You can configure the expiration for the keepalive timer using the registrar

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server expires max <seconds> min <seconds> command under sip in voice service voip mode for SIP
IP phones. For more information, see the, “Configuring Keepalive Timer Expiration in SIP Phones:
Example” section on page 831.
Cisco Unified CME 8.6 supports the CFU feature on SIP IP phones using the call-forward b2bua
unregistered command under voice register dn tag. The CFU feature supports overlap dialing and
en-bloc dialing. A call to a floating DN is forwarded to its CFU destination, if configured. Calls to a DN
out of service point or phones losing connection are not forwarded to a CFU number until the phone
becomes unregistered. For more information on configuring call-forward unregistered, see the
“Configuring Call Forward Unregistered for SIP IP Phones: Example” section on page 831.

Note In earlier versions of Cisco Unified CME, a busy tone was played for callers when the callers are unable
to reach the SCCP phone number. In Cisco Unified CME 8.6 and later versions, a fast busy tone is played
instead of a busy tone for callers who are unable to reach the phone.

B2BUA Call Forwarding for SIP Devices

Cisco Unified CME 3.4 an d later versions acts as both UA server and UA client; that is, as a B2BUA.
Calls into a SIP phone can be forwarded to other SIP or SCCP devices (including Cisco Unity or
Cisco Unity Express, third-party voice mail systems, an auto attendant or an IVR system, such as
Cisco Unified IPCC and Cisco Unified IPCC Express). In addition, SCCP phones can 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 an MWI to a SIP phone when a call is forwarded. The SIP phone then displays the MWI when
indicated by the voice-messaging system.
The call-forward busy response is triggered when a call is sent to a SIP phone using a VoIP dial peer and
a busy response is received back from the phone. SIP-to-SIP call forwarding is invoked only if the phone
is dialed directly. Call forwarding is not invoked when the phone number is called through a sequential,
longest-idle, or peer hunt group.
You can configure call forwarding for an individual directory number, or for every number on a SIP
phone. If the information is configured in both, the information under voice register dn takes precedence
over the information configured under voice register pool.
For configuration information, see the “SIP: Configuring SIP-to-SIP Phone Call Forwarding” section on
page 812.

Call Forward All Synchronization for SIP Phones

The Call Forward All feature allows users to forward all incoming calls to a phone number that they
specify. This feature is supported on all SIP phones and can be provisioned from either
Cisco Unified CME or the individual SIP phone. Before Cisco Unified CME 4.1, there was no method
for exchanging the Call Forward All configuration between Cisco Unified CME and the SIP phone. If
Call Forward All was enabled on the phone, the configuration in Cisco Unified CME was not updated;
conversely, the configuration in Cisco Unified CME was not sent to the phone.
In Cisco Unified CME 4.1 and later, the following enhancements are supported for the
Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE to keep the
configuration consistent between Cisco Unified CME and the SIP phone:

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• When Call Forward All is configured on Cisco Unified CME with the call-forward b2bua all
command, the configuration is sent to the phone which updates the CfwdAll soft key to indicate that
Call forward All is enabled. Because Call Forward All is configured on a per line basis, the CfwdAll
soft key is updated only when Call Forward All is enabled for the primary line.
• When a user enables Call Forward All on a phone using the CfwdAll soft key, the uniform resource
identifier (URI) for the service (defined with the call-feature-uri command) and the call forward
number (unless Call Forward All is disabled) is sent to Cisco Unified CME. It updates its voice
register pool and voice register dn configuration with the call-forward b2bua all command to be
consistent with the phone configuration.
• Call Forward All supports KPML so that a user does not need to press the Dial or # key, or wait for
the interdigit timeout, to configure the Call Forward All number. Cisco Unified CME collects the
Call Forward All digits until it finds a match in the dial peers.
For configuration information, see the “SIP: Configuring Call-Forwarding-All Soft Key URI” section
on page 818.

Call Transfer
When you are connected to another party, call transfer allows you to shift the connection of the other
party to a different number. Call transfer methods must interoperate with systems in the other networks
with which you interface. Cisco CME 3.2 and later versions provide full call-transfer and
call-forwarding interoperability with call processing systems that support H.450.2, H.450.3, and
H.450.12 standards. For call processing systems that do not support H.450 standards, Cisco CME 3.2
and later versions provide VoIP-to-VoIP hairpin call routing.
Call transfers can be blind or consultative. A blind transfer is one in which the transferring extension
connects the caller to a destination extension 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.
You can configure blind or consultative transfer on a systemwide basis or for individual extensions. For
example, in a system that is set up for consultative transfer, a specific extension with an auto-attendant
that automatically transfers incoming calls to specific extension numbers can be set to use blind transfer,
because auto-attendants do not use consultative transfer.

Call Transfer Blocking

Transfers to all numbers except those on local phones are automatically blocked by default. During
configuration, you can allow transfers to nonlocal numbers. In Cisco Unified CME 4.0 and later
versions, you can prevent individual phones from transferring calls to numbers that are globally enabled
for transfer. This ensures that individual phones do not incur toll charges by transferring calls outside
the Cisco Unified CME system. Call transfer blocking can be configured for individual phones or
configured as part of a template that is applied to a set of phones.
Another way to eliminate toll charges on call transfers is to limit the number of digits that phone users
can dial when transferring calls. For example, if you specify a maximum of eight digits in the
configuration, users who are transferring calls can dial one digit for external access and seven digits
more, which is generally enough for a local number but not a long-distance number. In most locations,
this plan will limit transfers to nontoll destinations. Long-distance calls, which typically require ten
digits or more, will not be allowed. This configuration is only necessary when global transfer to numbers
outside the Cisco Unified CME system has been enabled using the transfer-pattern (telephony-service)
command. Transfers to numbers outside the Cisco Unified CME system are not permitted by default.

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For configuration information, see the “SCCP: Configuring Call Transfer Options for Phones” section
on page 788.

Trunk-to-Trunk Transfer Blocking for Toll Fraud Prevention on Cisco Unified

SIP IP Phones
In Cisco Unified CME 4.0 trunk-to-trunk transfer blocking for toll bypass fraud prevention is supported
on Cisco Unified Skinny Client Control Protocol (SCCP) IP phones.
In Cisco Unified CME 9.5 trunk-to-trunk transfer blocking for toll bypass fraud prevention is also
supported on Cisco Unified Session Initiation Protocol (SIP) IP phones.
Table 66 lists the transfer-blocking commands and the appropriate configuration modes for Cisco
Unified CME and Cisco Unified SRST.

Table 66 Configuration Modes for Transfer-Blocking Commands

Commands Cisco Unified CME

transfer-pattern telephony-service
transfer max-length voice register pool or
voice register template
transfer-pattern blocked voice register pool or
voice register template
conference transfer-pattern telephony-service

Note The call transfer and conference restrictions apply when transfers or conferences are initiated toward
external parties, like a PSTN trunk, a SIP trunk, or an H.323 trunk. The restrictions do not apply to
transfers to local extensions.

transfer-pattern
The transfer-pattern command for Cisco Unified SCCP IP phones is extended to Cisco Unified SIP IP
phones.
The transfer-pattern command specifies the directory numbers for call transfer. The command can be
configured up to 32 times using the following command syntax: transfer-pattern transfer-pattern
[blind].

Note The blind keyword in the transfer-pattern command applies to Cisco Unified SCCP IP phones only and
does not apply to Cisco Unified SIP IP phones.

With the transfer-pattern command configured, only call transfers to numbers that match the
configured transfer pattern are allowed to take place. With the transfer pattern configured, all or a subset
of transfer numbers can be dialed and the transfer to a remote party can be initiated.

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Note In Cisco Unified CME 9.5 and later versions, Cisco Unified SIP IP phones and Cisco Unified SCCP IP
phones registered to the same Cisco Unified CME are considered local and do not require
transfer-pattern configuration.

The following are examples of configurable transfer patterns:

• .T—This configuration allows call transfers to any destinations with one or more digits, like 123,
877656, or 76548765.
• 919........—This configuration only allows call transfers to remote numbers beginning with “919”
and followed by eight digits, like 91912345678. However, call transfers to 9191234 or
919123456789 are not allowed.

Backward Compatibility

To maintain backward compatibility, all call transfers from Cisco Unified SIP IP phones to any number
(local or over trunk) are allowed when no transfer patterns are configured through the transfer-pattern,
transfer-pattern blocked, or transfer max-length commands.
For Cisco Unified SCCP IP phones, call transfers over trunk continue to be blocked when no transfer
patterns are configured.

Dial Plans

Whatever dial plan is used for external calls, the same numbers should be configured as specific numbers
using the transfer-pattern command.
If a dial plan requires “9” to be dialed before an external call is made, then “9” should be a prefix of the
transfer-pattern number. For example, 12345678 is an external number that requires “9” to be dialed
before the external call can be made so the transfer-pattern number should be 912345678.

Note In Cisco Unified CME 9.5 and later versions, once transfer patterns are configured in telephony-service
configuration mode, the transfer patterns apply to both Cisco Unified SCCP IP phones and Cisco Unified
SIP IP phones.

transfer max-length
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.
For example, the maximum length is configured as 5, then only call transfers from Cisco Unified SIP IP
phones up to a five-digit directory number are allowed. All call transfers to directory numbers with more
than five digits are blocked.

transfer-pattern blocked
When the transfer-pattern blocked command is configured for a specific phone, no call transfers are
allowed from that phone over the trunk.

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This feature forces unconditional blocking of all call transfers from the 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.
Table 67 compares the behaviors of Cisco Unified SCCP and SIP IP phones for specific configurations.

Table 67 Behaviors of Cisco Unified IP Phones for Specific Configurations

Configuration Cisco Unified SCCP IP Phones Cisco Unified SIP IP Phones

No transfer patterns are All non-local call transfers are All non-local call transfers are
configured. blocked. allowed for backward
compatibility.
Specific transfer patterns are Call transfers to specific external Call transfers to specific external
configured. entities are allowed. entities are allowed.
The transfer-pattern blocked All non-local call transfers are All non-local call transfers are
command is configured. blocked. blocked.
Note The configuration Note The configuraton
reverts to the default, unconditionally blocks
where no transfer all non-local call
patterns are configured. transfers. It does not
return to the default,
where all non-local call
transfers are allowed.

conference transfer-pattern
When both the transfer-pattern and conference transfer-pattern commands are configured and dialed
digits match the configured transfer pattern, conference calls are allowed. However, when the dialed
digits do not match any of the configured transfer pattern, the conference call is blocked.
For configuration information, see the “SIP: Specifying Transfer Patterns for Trunk-to-Trunk Calls and
Conferences” section on page 792 and “SIP: Blocking Trunk-to-Trunk Call Transfers” section on
page 794.
For configuration examples, see the “Configuring Transfer Patterns: Example” section on page 823,
“Configuring Maximum Length of Transfer Number: Example” section on page 823, “Configuring
Conference Transfer Patterns: Example” section on page 824, and “Blocking All Call Transfers:
Example” section on page 824.

Call-Transfer Recall
The Call-Transfer Recall feature in Cisco Unified CME 4.3 and later versions returns a transferred call
to the phone that initiated the transfer if the destination is busy or does not answer. After a phone user
completes a transfer to a directory number on a local phone, if the transfer-to party does not answer
before the configured recall timer expires, the call is directed back to the transferor phone. The message
“Transfer Recall From xxxx” displays on the transferor phone.
The transfer-to directory number cannot have Call Forward Busy enabled and cannot be a member of
any hunt group. If the transfer-to directory number has Call Forward No Answer (CFNA) enabled,
Cisco Unified CME recalls the call only if the transfer-recall timeout is set to less than the CFNA
timeout. If the transfer-recall timeout is set to more than the CFNA timeout, the call is forwarded to the
CFNA target number after the transfer-to party does not answer.

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If the transferor phone is busy, Cisco Unified CME attempts the recall again after a 15-second
retry-timer expires. Cisco Unified CME attempts a recall up to three times. If the transferor phone
remains busy, the call is disconnected after the third recall attempt.
The transferor phone and transfer-to phone must be registered to the same Cisco Unified CME, however
the transferee phone can be remote.
For configuration information, see the “Enabling Call Transfer and Forwarding at System-Level”
section on page 779.

Consultative-Transfer Enhancements in Cisco Unified CME 4.3 and Later Versions

Cisco Unified CME 4.3 modifies the digit-collection process for consultative call transfers. After a
phone user presses the Transfer soft key to make a consultative transfer, a new consultative call leg is
created and the Transfer soft key is not displayed again until the dialed digits of the transfer-to number
are matched to a transfer pattern and the consultative call leg is in the alerting state.
Transfer-to digits dialed by the phone user are no longer buffered. The dialed digits, except the call park
FAC code, are collected on the seized consultative call-leg until the digits match a pattern for
consultative transfer, blind transfer, park-slot transfer, park-slot transfer blocking, or PSTN transfer
blocking. The existing pattern matching process is unchanged, and you have the option of using this new
transfer digit-collection method or reverting to the former method.
Before Cisco Unified CME 4.3, the consultative transfer feature collects dialed digits on the original call
leg until the digits either match a transfer pattern or blocking pattern. When the transfer-to number is
matched, and PSTN blocking is not enabled, the original call is put on hold and an idle line or channel
is seized to send the dialed digits from the buffer.
The method of matching a pattern for consultative transfer, blind transfer, park-slot transfer, park-slot
transfer blocking, PSTN transfer blocking, and after-hours blocking remain the same. When the
transfer-to number matches the pattern for a blind transfer or park-slot transfer, Cisco Unified CME
terminates the consultative call leg and transfers the call.
After the transfer-to digits are collected, if the transfer is not committed before the transfer-timeout
expires in 30 seconds, the consultation call leg is disconnected.
These enhancements are 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 the transferor's directory number
(ephone-dn).
• An idle line or channel is available for seizing, digit collection, and dialing.
Cisco Unified CME 4.3 and later versions enable these transfer enhancements by default.
To revert to the digit-collection method used in previous versions of Cisco Unified CME, see the
“Enabling Call Transfer and Forwarding at System-Level” section on page 779.

Consultative Transfer With Direct Station Select

Direct Station Select (DSS) is a feature that allows a multibutton phone user to transfer calls to an idle
monitored line by pressing the Transfer key and the appropriate monitored line button. A monitored line
is one that appears on two phones; one phone can use the line to make and receive calls and the other
phone simply monitors whether the line is in use. For Cisco CME 3.2 and later versions, consultative
transfers can occur during Direct Station Select (transferring calls to idle monitored lines).

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If the person sharing the monitored line does not want to accept the call, the person announcing the call
can reconnect to the incoming call by pressing the EndCall soft key to terminate the announcement call
and pressing the Resume soft key to reconnect to the original caller.
Direct Station Select consultative transfer is enabled with the transfer-system full-consult dss
command, which defines the call transfer method for all lines served by the router. The transfer-system
full-consult dss command supports the keep-conference command. See “Configuring Conferencing”
on page 941.

H.450.2 and H.450.3 Support

H.450.2 is a standard protocol for exchanging call-transfer information across a network, and H.450.3
is a standard protocol for exchanging call-forwarding information across a network. Cisco CME 3.0 and
later versions support the H.450.2 call-transfer standards and the H.450.3 call-forwarding standards that
were introduced in Cisco ITS V2.1. Using the H.450.2 and H.450.3 standards to manage call transfer
and forwarding in a VoIP network provides the following benefits:
• The final call path from the transferred party to the transfer destination is optimal, with no
hairpinned routes or excessive use of resources.
• Call parameters (for example, codec) can be different for the different call legs.
• This solution is scalable.
• There is no limit to the number of times a call can be transferred.
Considerations for using the H.450.2 and H.450.3 standards include the following:
• Cisco IOS Release 12.2(15)T or a later release is required on all voice gateways in the network.
• Support of H.450.2 and H.450.3 is required on all voice gateways in the network. H.450.2 and
H.450.3 are used regardless of whether the transfer-to or forward-to target is on the same
Cisco Unified CME system as the transferring party or the forwarding party, so the transferred party
must also support H.450.2 and the forwarded party must also support H.450.3. The exception is calls
that can be reoriginated through hairpin call routing or through the use of an H.450 tandem gateway.
• Call forwarding over SIP networks uses the 302 Moved Temporarily SIP response, which works in
a manner similar to the way in which the H.450.3 standard is used for H.323 networks. To enable
call forwarding, you must specify a pattern that matches the calling-party numbers of the calls that
you want to be able to forward.
• Cisco Unified CME supports all SIP Refer method call transfer scenarios, but you must ensure that
call transfer is enabled using H.450.2 standards.
• H.450 standards are not supported by Cisco Unified Communications Manager, Cisco BTS, or
Cisco PGW, although hairpin call routing or an H.450 tandem gateway can be set up to handle calls
to and from those types of systems.
The following series of figures depicts a call being transferred using H.450.2 standards. Figure 31 on
page 765 shows A calling B. Figure 32 on page 765 shows B consulting with C and putting A on hold.
Figure 33 on page 765 shows that B has connected A and C, and Figure 34 on page 766 shows A and C
directly connected, with B no longer involved in the call.

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Figure 31 Call Transfer Using H.450.2: A Calls B

H.323
V
Cisco Unified
Cisco Unified CME 1 CME 3
IP
Phone A Phone C

Cisco Unified CME 2

146629
IP
Phone B

Figure 32 Call Transfer Using H.450.2: B Consults with C

H.323 V
Cisco Unified
Cisco Unified CME 1 CME 3

IP
Cisco Unified CME 2 Phone C
Phone A

H.450.2 connection H.450.2 connection

146634
IP
Phone B

Figure 33 Call Transfer Using H.450.2: B Transfers A to C

H.323 V
Cisco Unified
Unified
Cisco CME 1 CME 1 CME 3
IP
Phone A Cisco Unified CME 2 Phone C
146633

IP
Phone B

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Figure 34 Call Transfer Using H.450.2: A and C Are Connected

H.323 V
Cisco Unified
Cisco
Cisco Unified
Unified
CME 1 CME 1 CME 3
CME 1
IP
Phone A Cisco Unified
Cisco Unified CME 2 Phone C
CME 2

344518
IP
Phone B

Tips for Using H.450 Standards

Use H.450 standards when a network meets the following conditions:
• The router that you are configuring uses Cisco CME 3.0 or a later version, or Cisco ITS V2.1.
• For Cisco CME 3.0 or Cisco ITS V2.1 systems, all endpoints in the network must support H.450.2
and H.450.3 standards. For Cisco CME 3.1 or later systems, if some of the endpoints do not support
H.450 standards (for example, Cisco Unified Communications Manager, Cisco BTS, or
Cisco PGW), you can use hairpin call routing or an H.450 tandem gateway to handle transfers and
forwards with those endpoints. Also, either you must explicitly disable H.450.2 and H.450.3 on the
dial peers that handle those calls or you must enable H.450.12 capability to automatically detect the
calls that support H.450.2 and H.450.3 and those calls that do not.
Support for the H.450.2 standard and the H.450.3 standard is enabled by default and can be disabled
globally or for individual dial peers. For configuration information, see the “Enabling Call Transfer and
Forwarding at System-Level” section on page 779.

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Transfer Method Recommendations by Cisco Unified CME Version

You must specify the method to use for call transfers: H.450.2 standard signaling or Cisco proprietary
signaling, and whether transfers should be blind or allow consultation. Table 68 summarizes transfer
method recommendations for all Cisco Unified CME versions.

Table 68 Transfer Method Recommendations

Cisco Unified CME transfer-system transfer-system

Version Command Default Keyword to Use Transfer Method Recommendation
4.0 and later full-consult full-consult Use H.450.2 for call transfer, which is the default for this
or version. You do not need to use the transfer-system command
full-blind unless you want to use the full-blind or dss keyword.
Optionally, you can use the proprietary Cisco method by using
the transfer-system command with the blind or local-consult
keyword.
Use H.450.7 for call transfer using QSIG supplementary
services
3.0 to 3.3 blind full-consult Use H.450.2 for call transfer. You must explicitly configure the
or transfer-system command with the full-consult or full-blind
full-blind keyword because H.450.2 is not the default for this version.
Optionally, you can use the proprietary Cisco method by using
the transfer-system command with the blind or local-consult
keyword.
2.1 blind blind Use the Cisco proprietary method, which is the default for this
or version. You do not need to use the transfer-system command
local-consult unless you want to use the local-consult keyword.
Optionally, you can use the transfer-system command with the
full-consult or full-blind keyword. You must also configure the
router with a Tcl script that is contained in the
app-h450-transfer.x.x.x.x.zip file. This file is available from the
Cisco Unified CME software download website at
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-iostsp.
Earlier than 2.1 blind blind Use the Cisco proprietary method, which is the default for this
version. You do not need to use the transfer-system command
unless you want to use the local-consult keyword.

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H.450.12 Support
Cisco CME 3.1 and later versions support the H.450.12 call capabilities standard, which provides a
means to advertise and dynamically discover H.450.2 and H.450.3 capabilities in voice gateway
endpoints on a call-by-call basis. When discovered, the calls associated with non-H.450 endpoints can
be directed to use non-H.450 methods for transfer and forwarding, such as hairpin call routing or H.450
tandem gateway.
When H.450.12 is enabled, H.450.2 and H.450.3 services are disabled for call transfers and call forwards
unless a positive H.450.12 indication is received from all other VoIP endpoints involved in the call. If a
positive H.450.12 indication is received, the router uses the H.450.2 standard for call transfers and the
H.450.3 standard for call forwarding. If a positive H.450.12 indication is not received, the router uses
the alternative method that you have configured for call transfers and forwards, either hairpin call routing
or an H.450 tandem gateway.
You can have either of the following situations in your network:
• All gateway endpoints support H.450.2 and H.450.3 standards. In this situation, no special
configuration is required because support for H.450.2 and H.450.3 standards is enabled on the
Cisco CME 3.1 or later router by default. H.450.12 capability is disabled by default, but it is not
required because all calls can use H.450.2 and H.450.3 standards.
• Not all gateway endpoints support H.450.2 and H.450.3 standards. Therefore, specify how
non-H.450 calls are to be handled by choosing one of the following options:
– Enable the H.450.12 capability in Cisco CME 3.1 and later to dynamically determine, on a
call-by-call basis, whether each call has H.450.2 and H.450.3 support. If H.450.12 is enabled
and a call is determined to have H.450 support, the call is transferred using H.450.2 standards
or forwarded using H.450.3 standards. See the “Enabling H.450.12 Capabilities” section on
page 795.
Support for the H.450.12 standard is disabled by default and can be enabled globally or for
individual dial peers.
If the call does not have H.450 support, it can be handled by a VoIP-to-VoIP connection that you
configure using dial peers and the “Enabling H.323-to-H.323 Connection Capabilities” section
on page 797. The connection can be used for hairpin call routing or routing to an H.450 tandem
gateway.
– Explicitly disable H.450.2 and H.450.3 capability on a global basis or by individual dial peer,
which forces all calls to be handled by a VoIP-to-VoIP connection that you configure using dial
peers and the“Enabling H.323-to-H.323 Connection Capabilities” section on page 797. This
connection can be used for hairpin call routing or routing to an H.450 tandem gateway.

Hairpin Call Routing

Cisco CME 3.1 and later supports hairpin call routing using a VoIP-to-VoIP connection to transfer and
forward calls that cannot use H.450 standards. When a call that originally terminated on a voice gateway
is transferred or forwarded by a phone or other application attached to the gateway, the gateway
reoriginates the call and routes the call as appropriate, making a VoIP-to-VoIP, or hairpin, connection.
This approach avoids any protocol dependency on the far-end transferred-party endpoint or
transfer-destination endpoint. Hairpin routing of transferred and forwarded calls also causes the
generation of separate billing records for each call leg, so that the transferred or forwarded call leg is
typically billed to the user who initiates the transfer or forward.

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In Cisco CME 3.2 and later versions, transcoding between G.711 and G.729 is supported when one leg
of a VoIP-to-VoIP hairpin call uses G.711 and the other leg uses G.729. For information about
transcoding, see “Configuring Transcoding Resources” on page 443.
Hairpin call routing provides the following benefits:
• Call transfer and forwarding is provided to non-H.450 endpoints, such as
Cisco Unified Communications Manager, Cisco BTS, or Cisco PGW.
• The network can also contain Cisco CME 3.0 or Cisco ITS 2.1 systems.
Hairpin call routing has the following disadvantages:
• End-to-end signaling and media delay are increased significantly.
• A single hairpinned call uses as much WAN bandwidth as two directly connected calls.
VoIP-to-VoIP hairpin connections can be made using dial peers if the allow-connections h323 to h323
command is enabled and at least one of the following is true:
• H.450.12 is used to detect calls on which H.450.2 or H.450.3 is not supported by the remote system.
• H.450.2 or H.450.3 is explicitly disabled.
• Cisco Unified CME automatically detects that the remote system is a
Cisco Unified Communications Manager.
Figure 35 on page 769 shows a call that is made from A to B. Figure 36 on page 770 shows that B has
forwarded all calls to C. Figure 37 on page 770 shows that A and C are connected by an H.323 hairpin.

Figure 35 Hairpin with H.323: A Calls B

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Figure 36 Hairpin with H.323: Call is Forwarded to C

H.323 V
Non-H.450
Cisco Unified CME 1 gateway
IP
Phone A Phone C
Cisco Unified CME 2

146630
IP Calls are forwarded
Phone B to phone C

Figure 37 Hairpin with H.323: A is Connected to C via B

H.323 V
Non-H.450
Cisco Unified CME 1 gateway
IP
Phone A Phone C

Cisco Unified CME 2

IP 146631

Phone B

Tips for Using Hairpin Call Routing

Use hairpin call routing when a network meets the following three conditions:
• The router that you are configuring uses Cisco CME 3.1 or a later version.
• Some or all calls require VoIP-to-VoIP routing because they cannot use H.450 standards, which can
happen for any of the following reasons:
– H.450 capabilities have been explicitly disabled on the router.
– H.450 capabilities do not exist in the network.
– H.450 capabilities are supported on some endpoints and not supported on other endpoints,
including those handled by Cisco Unified Communications Manager, Cisco BTS, and
Cisco PGW. When some endpoints support H.450 and others do not, you must enable H.450.12
capabilities on the router to detect which endpoints are H.450-capable or designate some dial
peers as H.450-capable. For more information about enabling H.450.12 capabilities, see the
“Enabling H.450.12 Capabilities” section on page 795.
• No voice gateway is available to act as an H.450 tandem gateway.
For information about configuring Cisco Unified CME to forward calls using local hairpin routing, see
the “Forwarding Calls Using Local Hairpin Routing” section on page 799.

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Support for VoIP-to-VoIP connections is disabled by default and can be enabled globally. For
configuration information, see the “Enabling H.323-to-H.323 Connection Capabilities” section on
page 797.

H.450 Tandem Gateways

H.450 tandem gateways address the limitations of hairpin call routing using a manner similar to hairpin
call routing but without the double WAN link traversal created by hairpin connections. An H.450 tandem
gateway is an additional voice gateway that serves as a “front-end” for a call processor that does not
support the H.450 standards, such as Cisco Unified Communications Manager, Cisco BTS Softswitch
(Cisco BTS), or Cisco PSTN Gateway (Cisco PGW). Transferred and forwarded calls that are intended
for non-H.450 endpoints are terminated instead on the H.450 tandem gateway and reoriginated there for
delivery to the non-H.450 endpoints. The H.450 tandem gateway can also serve as a PSTN gateway.
An H.450 tandem gateway is configured with a dial peer that points to the
Cisco Unified Communications Manager or other system for which the H.450 tandem gateway is
serving as a front end. The H.450 tandem voice gateway is also configured with dial peers that point to
all the Cisco Unified CME systems in the private H.450 network. In this way, Cisco Unified CME and
the Cisco Unified Communications Manager are not directly linked to each other, but are instead both
linked to an H.450 tandem gateway that provides H.450 services to the non-H.450 platform.
An H.450 tandem gateway can also work as a PSTN gateway for remote Cisco Unified CME systems
and for Cisco Unified Communications Manager (or other non-H.450 system). Use different inbound
dial peers to separate Cisco Unified Communications Manager-to-PSTN G.711 calls from tandem
gateway-to-Cisco Unified CME G.729 calls.

Note An H.450 tandem gateway that is used in a network to support non-H.450-capable call processing
systems requires the Integrated Voice and Video Services feature license. This feature license, which was
introduced in March 2004, includes functionality for H.323 gatekeeper, IP-to-IP Gateway, and H.450
tandem gateway. With Cisco IOS Release 12.3(7)T, an H.323 gatekeeper feature license is required with a
JSX Cisco IOS image on the selected router. Consult your Cisco Unified CME SE regarding the required
feature license. With Cisco IOS Release 12.3(7)T, you cannot use Cisco Unified CME and H.450 tandem
gateway functionality on the same router.

VoIP-to-VoIP connections can be made for an H.450 tandem gateway if the allow-connections h323 to
h323 command is enabled and one or more of the following is true:
• H.450.12 is used to dynamically detect calls on which H.450.2 or H.450.3 is not supported by the
remote VoIP system.
• H.450.2 or H.450.3 is explicitly disabled.
• Cisco CME 3.1 or later automatically detects that the remote system is a
Cisco Unified Communications Manager.
For Cisco CME 3.1 and earlier, the only type of VoIP-to-VoIP connection supported by
Cisco Unified CME is H.323-to-H.323. For Cisco CME 3.2 and later versions, H.323-to-SIP
connections are allowed only for Cisco Unified CME systems running Cisco Unity Express.
Figure 38 on page 772 shows a tandem voice gateway that is located between the central hub of the
network of a CPE-based Cisco CME 3.1 or later network and a Cisco Unified Communications Manager
network. This topology would work equally well with a Cisco BTS or Cisco PGW in place of the
Cisco Unified Communications Manager.

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In the network topology in Figure 38 on page 772, the following events occur (refer to the event numbers
on the illustration):
1. A call is generated from extension 4002 on phone 2, which is connected to a
Cisco Unified Communications Manager. The H.450 tandem gateway receives the H.323 call and,
acting as the H.323 endpoint, the H.450 tandem gateway handles the call connection to a
Cisco Unified IP phone in a CPE-based Cisco CME 3.1 or later network.
2. The call is received by extension 1001 on phone 3, which is connected to Cisco Unified CME 1.
Extension 1001 performs a consultation transfer to extension 2001 on phone 5, which is connected
to Cisco Unified CME 2.
3. When extension 1001 transfers the call, the H.450 tandem gateway receives an H.450.2 message
from extension 1001.
4. The H.450 tandem gateway terminates the call leg from extension 1001 and reoriginates a call leg
to extension 2001, which is connected to Cisco Unified CME 2.
5. Extension 4002 is connected with extension 2001.

Figure 38 H.450 Tandem Gateway

IP-to-IP
Gateway

Public VoIP Cisco Unified CallManager

323 323
1
H.323 Connection
IP IP in ICT mode using slow start
Phone 1 Phone 2
4001 4002 Media Termination Point (MTP)
V

Private H.450 Network H.450

H.450Tandem
Tandem
Gateway
Gateway
3 PSTN
V

H.450.2 Message
Private VoIP Telephone
Cisco Unified CME 1
2 Cisco Unified CME 2

V V
2 5
4
IP IP IP IP
Phone 3 Phone 4 Phone 5 Phone 6
1001 1002 3001 3002
146622

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Tips for Using H.450 Tandem Gateways

Use this procedure when a network meets the following conditions:
• The router that you are configuring uses Cisco CME 3.1 or a later version.
• Some endpoints in the network are not H.450-capable, including those handled by
Cisco Unified Communications Manager, Cisco BTS, and Cisco PGW.
Support for VoIP-to-VoIP connections is disabled by default and can be enabled globally. For more
information, see the “Enabling H.323-to-H.323 Connection Capabilities” section on page 797.
Use dial peers to set up an H.450 tandem gateway. See the “Dial Peers” section on page 773.

Dial Peers
Dial peers describe the virtual interfaces to or from which a call is established. All voice technologies
use dial peers to define the characteristics associated with a call leg. Attributes applied to a call leg
include specific quality of service (QoS) features, compression/decompression (codec), voice activity
detection (VAD), and fax rate. Dial peers are also used to establish the routing paths in your network,
including special routing paths such as hairpins and H.450 tandem gateways. Dial peer settings override
the global settings for call forward and call transfer. For information about configuring dial peers, see
the Dial Peer Configuration on Voice Gateway Routers guide.

QSIG Supplementary Services

QSIG is an intelligent inter-PBX signaling system widely adopted by PBX vendors. It supports a range
of basic services, generic functional procedures, and supplementary services. Cisco Unified CME 4.0
introduces supplementary services features that allow Cisco Unified CME phones to seamlessly
interwork using QSIG with phones connected to a PBX. One benefit is that IP phones can use a PBX
message center with proper MWI notifications. Figure 39 illustrates a topology for a
Cisco Unified CME system with some phones under the control of a PBX.

Figure 39 Cisco Unified CME System with PBX

IP 1001 IP 2001

IP 1002 IP 2002

IP 1003 IP 2003

Remote Cisco CME IP network Local Cisco CME

QSIG 3001

3002
PBX
3003

Message
135562

center

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The following QSIG supplementary service features are supported in Cisco Unified CME systems. They
follow the standards from the European Computer Manufacturers Association (ECMA) and the
International Organization for Standardization (ISO) on PRI and BRI interfaces.
• Basic calls between IP phones and PBX phones.
• Calling Line/Name Identification (CLIP/CNIP) presented on an IP phone when called by a PBX
phone; in the reverse direction, such information is provided to the called endpoint.
• Connected Line/Name Identification (COLP/CONP) information provided when a PBX phone calls
an IP phone and is connected; in the reverse direction, such information presented on an IP phone.
• Call Forward using QSIG and H.450.3 to support any combination of IP phone and PBX phone,
including an IP phone in the Cisco Unified CME system that is connected to a PBX or an IP phone
in another Cisco Unified CME system across an H.323 network.
• Call forward to the PBX message center according to the configured policy. The other two endpoints
can be a mixture of IP phone and PBX phones.
• Hairpin call transfer, which interworks with a PBX in transfer-by-join mode. Note that
Cisco Unified CME does not support the actual signaling specified for this transfer mode (including
the involved FACILITY message service APDUs) which are intended for an informative purpose
only and not for the transfer functionality itself. As a transferrer (XOR) host, Cisco Unified CME
simply hairpins two call legs to create a connection; as a transferee (XEE) or transfer-to (XTO) host,
it will not be aware of a transfer that is taking place on an existing leg. As a result, the final endpoint
may not be updated with the accurate identity of its peer. Both blind transfer and consult transfer are
supported.
• Message-waiting indicator (MWI) activation or deactivation requests are processed from the PBX
message center.
• The PBX message center can be interrogated for the MWI status of a particular ephone-dn.
• A user can retrieve voice messages from a PBX message center by making a normal call to the
message center access number.
For information about enabling QSIG supplementary services, see the “Enabling H.450.7 and QSIG
Supplementary Services at a System-Level” section on page 801 and “Enabling H.450.7 and QSIG
Supplementary Services on a Dial Peer” section on page 802.
For more information about configuring Cisco Unified CME to integrate with voice-mail systems, see
“Integrating Voice Mail” on page 531.

Disabling SIP Supplementary Services for Call Forward and Call Transfer
If a destination gateway does not support supplementary services, you can disable REFER messages for
call transfers and the redirect responses for call forwarding from being sent by Cisco Unified CME.
For configuration information, see the “Disabling SIP Supplementary Services for Call Forward and Call
Transfer” section on page 804.

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Typical Network Scenarios for Call Transfer and Call Forwarding

In a mixed network that involves two or more types of call agents or call-control systems, there can be
communication protocol discrepancies and dependencies, and therefore the opportunity for
interoperability errors. These discrepancies show up most often when a call is being transferred or
forwarded. This section provides descriptions of the specific mixed-network scenarios you might
encounter when configuring a router running Cisco CME 3.1 or a later version. Each of the following
sections point to the configuration instructions necessary to ensure call transfer and forwarding
capabilities throughout the network.
• Cisco CME 3.1 or Later and Cisco IOS Gateways, page 775
• Cisco CME 3.0 or an Earlier Version and Cisco IOS Gateways, page 776
• Cisco CME 3.1 or Later, Non-H.450 Gateways, and Cisco IOS Gateways, page 776
• Cisco Unified CME, Non-H.450 Gateways, and Cisco IOS Gateways, page 777
• Cisco CME 3.1 or Later, Cisco Unified Communications Manager, and Cisco IOS Gateways,
page 777
• Cisco CME 3.0 or an Earlier Version, Cisco Unified Communications Manager, and Cisco IOS
Gateways, page 778

Note Cisco Communications Manager Express 3.2 (Cisco CME 3.2) and later versions provide full
call-transfer and call-forwarding with call processing systems on the network that support H.450.2,
H.450.3, and H.450.12 standards. For interoperability with call processing systems that do not support
H.450 standards, Cisco CME 3.2 and later versions provide VoIP-to-VoIP hairpin call routing without
requiring the special Tool Command Language (Tcl) script that was needed in earlier versions of
Cisco Unified CME.

Cisco CME 3.1 or Later and Cisco IOS Gateways

In a network with Cisco CME 3.1 or a later version and Cisco IOS gateways, all systems that might
participate in calls that involve call transfer and call forwarding are capable of supporting the H.450.2,
H.450.3, and H.450.12 standards. This is the simplest environment for operating the Cisco CME 3.1 or
later features.
Configuration for this type of network consists of:
1. Setting up call-transfer and call-forwarding parameters for transfers and forwards that are initiated
on this router (H.450.2 and H.450.3 capabilities for transferred parties, transfer destinations,
forwarded parties, and forwarding destinations are enabled by default). See the “Enabling Call
Transfer and Forwarding at System-Level” section on page 779.
2. Enabling H.450.12 globally to detect any calls on which H.450.2 and H.450.3 standards are not
supported. Although this step is optional, we recommend it. See the “Enabling H.450.12
Capabilities” section on page 795.
3. Optionally setting up VoIP-to-VoIP connections (hairpin call routing or H.450 tandem gateway) to
route calls that do not support H.450.2 or H.450.3 standards. See the “Enabling H.323-to-H.323
Connection Capabilities” section on page 797.
4. Setting up dial peers to manage call legs within the network. See Dial Peer Configuration on Voice
Gateway Routers.

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Cisco CME 3.0 or an Earlier Version and Cisco IOS Gateways

Before Cisco CME 3.1, H.450.2 and H.450.3 standards are used for all calls by default and routers do
not support the H.450.12 standard.
Configuration for this type of network consists of:
1. Setting up call-transfer and call-forwarding parameters for transfers and forwards that are initiated
on this router (H.450.2 and H.450.3 capabilities for transferred parties, transfer destinations,
forwarded parties, and forwarding destinations are enabled by default). See the “Enabling Call
Transfer and Forwarding at System-Level” section on page 779.
2. Enabling H.450.12 in advertise-only mode on Cisco CME 3.1 or later systems. As each
Cisco CME 3.0 system is upgraded to Cisco CME 3.1 or later, enable H.450.12 in advertise-only
mode. Note that no checking for H.450.2 or H.450.3 support is done in advertise-only mode. When
all Cisco CME 3.0 systems in the network have been upgraded to Cisco CME 3.1 or later, remove
the advertise-only restriction. See the “Enabling H.450.12 Capabilities” section on page 795.
3. Optionally setting up VoIP-to-VoIP connections (hairpin call routing or H.450 tandem gateway) to
route calls that cannot use H.450.2 or H.450.3 standards. See the “Enabling H.323-to-H.323
Connection Capabilities” section on page 797.
4. Setting up dial peers to manage call legs within the network. See Dial Peer Configuration on Voice
Gateway Routers.

Cisco CME 3.1 or Later, Non-H.450 Gateways, and Cisco IOS Gateways
In a network with Cisco CME 3.1 or later, non-H.450 gateways, and Cisco IOS gateways, the H.450.2
and H.450.3 services are provided only to calling endpoints that use H.450.12 to explicitly indicate that
they are capable of H.450.2 and H.450.3 operations. Because the Cisco BTS and Cisco PGW do not
support the H.450.12 standard, calls to and from these systems that involve call transfer or forwarding
are handled using H.323-to-H.323 hairpin call routing.
Configuration for this type of network consists of:
1. Setting up call-transfer and call-forwarding parameters for transfers and forwards that are initiated
on this router (H.450.2 and H.450.3 capabilities for transferred parties, transfer destinations,
forwarded parties, and forwarding destinations are enabled by default). Optionally disable H.450.2
and H.450.3 capabilities on dial peers that point to non-H.450-capable systems such as
Cisco Unified Communications Manager, Cisco BTS, or Cisco PGW. See the “Enabling Call
Transfer and Forwarding at System-Level” section on page 779.
2. Enabling H.450.12 to detect any calls on which H.450.2 and H.450.3 standards are not supported,
either globally or for specific dial peers. See the “Enabling H.450.12 Capabilities” section on
page 795.
3. Setting up VoIP-to-VoIP connections (hairpin call routing or H.450 tandem gateway) to route calls
that do not support H.450.2 or H.450.3 standards. See the “Enabling H.323-to-H.323 Connection
Capabilities” section on page 797.
4. Setting up dial peers to manage call legs within the network. See Dial Peer Configuration on Voice
Gateway Routers.

Note If your network contains a Cisco Unified Communications Manager, also see the instructions in the
“Enabling Interworking with Cisco Unified Communications Manager” section on page 806.

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Cisco Unified CME, Non-H.450 Gateways, and Cisco IOS Gateways

Note Cisco CME 3.0 and Cisco ITS V2.1 systems do not have H.450.12 capabilities.

In a network that contains a mix of Cisco Unified CME versions and at least one non-H.450 gateway, the
simplest configuration approach is to globally disable all H.450.2 and H.450.3 services and force
H.323-to-H.323 hairpin call routing for all transferred and forwarded calls. In this case, you would
enable H.450.12 detection capabilities globally. Alternatively, you could select to enable H.450.12
capability for specific dial peers. In this case, you would not configure H.450.12 capability globally; you
would leave it in its default disabled state.
Configuration for this type of network consists of:
1. Setting up call-transfer and call-forwarding parameters for transfers and forwards that are initiated
on this router (H.450.2 and H.450.3 capabilities for transferred parties, transfer destinations,
forwarded parties, and forwarding destinations are enabled by default). See the “Enabling Call
Transfer and Forwarding at System-Level” section on page 779.
2. Enabling H.450.12 to detect any calls on which H.450.2 and H.450.3 standards are not supported,
either globally or on specific dial peers. See the “Enabling H.450.12 Capabilities” section on
page 795.
3. Setting up VoIP-to-VoIP connections (hairpin call routing or H.450 tandem gateway) to route all
transferred and forwarded calls. See the “Enabling H.323-to-H.323 Connection Capabilities”
section on page 797.
4. Setting up dial peers to manage call legs within the network. See Dial Peer Configuration on Voice
Gateway Routers.

Note If your network contains a Cisco Unified Communications Manager, also see the instructions in the
“Enabling Interworking with Cisco Unified Communications Manager” section on page 806.

Cisco CME 3.1 or Later, Cisco Unified Communications Manager, and Cisco IOS Gateways
In a network with Cisco CME 3.1 or later, Cisco Unified Communications Manager, and Cisco IOS
gateways, Cisco CME 3.1 and later versions support automatic detection of calls to and from
Cisco Unified Communications Manager using proprietary signaling elements that are included with the
standard H.323 message exchanges. The Cisco CME 3.1 or later system uses these detection results to
determine the H.450.2 and H.450.3 capabilities of calls rather than using H.450.12 supplementary
services capabilities exchange, which Cisco Unified Communications Manager does not support. If a
call is detected to be coming from or going to a Cisco Unified Communications Manager endpoint, the
call is treated as a non-H.450 call. All other calls in this type of network are treated as though they
support H.450 standards. Therefore, this type of network should contain only Cisco CME 3.1 or later
and Cisco Unified Communications Manager call-processing systems.
Configuration for this type of network consists of:
1. Setting up call-transfer and call-forwarding parameters for transfers and forwards that are initiated
on this router (H.450.2 and H.450.3 capabilities for transferred parties, transfer destinations,
forwarded parties, and forwarding destinations are enabled by default). See the “Enabling Call
Transfer and Forwarding at System-Level” section on page 779.

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2. Enabling H.450.12 to detect any calls on which H.450.2 and H.450.3 standards are not supported,
either globally or on specific dial peers. See the “Enabling H.450.12 Capabilities” section on
page 795.
3. Setting up VoIP-to-VoIP connections (hairpin call routing or H.450 tandem gateway) to route all
transferred and forwarded calls that are detected as being to or from
Cisco Unified Communications Manager. See the “Enabling H.323-to-H.323 Connection
Capabilities” section on page 797.
4. Setting up specific parameters for Cisco Unified Communications Manager. See the instructions in
the “Enabling Interworking with Cisco Unified Communications Manager” section on page 806.
5. Setting up dial peers to manage call legs within the network. See Dial Peer Configuration on Voice
Gateway Routers.

Cisco CME 3.0 or an Earlier Version, Cisco Unified Communications Manager, and Cisco IOS
Gateways
Calls between the Cisco Unified Communications Manager and the older Cisco CME 3.0 or
Cisco ITS V2.1 networks need special consideration. Because Cisco CME 3.0 and Cisco ITS V2.1
systems do not support automatic Cisco Unified Communications Manager detection and also do not
natively support H.323-to-H.323 call routing, alternative arrangements are required for these systems.
To configure call transfer and forwarding on the Cisco CME 3.0 router, you can select from the
following three options:
• Use a Tcl script to handle call transfer and forwarding by invoking Tcl-script-based H.323-to-H.323
hairpin call routing (app-h450-transfer.2.0.0.9.tcl or a later version). Enable this script on all VoIP
dial peers and also under telephony-service mode, and set the local-hairpin script parameter to 1.
• Use a loopback-dn mechanism. See “Configuring Loopback Call Routing” on page 1185.
• Configure a loopback call path using router physical voice ports.
All three options force use of H.323-to-H.323 hairpin call routing for all calls regardless of whether the
call is from a Cisco Unified Communications Manager or other H.323 endpoint (including
Cisco CME 3.1 or later).

How to Configure Call Transfer and Forwarding

This section contains the following procedures:

SCCP
• Enabling Call Transfer and Forwarding at System-Level, page 779 (required)
• SCCP: Enabling Call Forwarding for a Directory Number, page 784 (required)
• SCCP: Enabling Call Transfer for a Directory Number, page 787 (required)
• SCCP: Configuring Call Transfer Options for Phones, page 788 (optional))
• SCCP: Verifying Call Transfer, page 791 (optional)
• SIP: Specifying Transfer Patterns for Trunk-to-Trunk Calls and Conferences, page 792
• SIP: Blocking Trunk-to-Trunk Call Transfers, page 794
• Enabling H.450.12 Capabilities, page 795 (optional)

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• Enabling H.323-to-H.323 Connection Capabilities, page 797 (optional)

• Forwarding Calls Using Local Hairpin Routing, page 799 (optional)
• Enabling H.450.7 and QSIG Supplementary Services at a System-Level, page 801 (optional)
• Enabling H.450.7 and QSIG Supplementary Services on a Dial Peer, page 802 (optional)
• Disabling SIP Supplementary Services for Call Forward and Call Transfer, page 804 (optional)
• Enabling Interworking with Cisco Unified Communications Manager, page 806 (optional)

SIP B2BUA
• SIP: Configuring SIP-to-SIP Phone Call Forwarding, page 812 (required)
• SIP: Configuring Call-Forwarding-All Soft Key URI, page 818 (optional)
• SIP: Configuring Call Forward Unregistered for SIP IP Phones, page 815 (optional)
• Configuring Keepalive Timer Expiration in SIP Phones, page 817 (optional)
• SIP: Specifying Number of 3XX Responses To be Handled, page 819 (optional)
• SIP: Configuring Call Transfer, page 820 (required)
• Disabling SIP Supplementary Services for Call Forward and Call Transfer, page 804 (optional)

Enabling Call Transfer and Forwarding at System-Level

To enable H.450 call transfers and forwards for transferring or forwarding parties; that is, to allow
transfers and forwards to be initiated from a Cisco Unified CME system, perform the following steps.

Note H.450.2 and H.450.3 capabilities are enabled by default for transferred or forwarded parties and
transfer-destination or forward-destination parties. Dial peer settings override the global setting.

Prerequisites
Cisco CME 3.0 or a later version, or Cisco ITS V2.1.

Restrictions
• Call transfers are handled differently depending on the Cisco Unified CME version. See Table 68
on page 767 for recommendations on selecting a transfer method for your Cisco Unified CME
version.
• The transfer-system local-consult command is not supported if the transfer-to destination is on the
Cisco ATA, Cisco VG224, or a SCCP-controlled FXS port.
• The H.450.2 and H.450.3 standards are not supported by Cisco Unified Communications Manager,
Cisco BTS, or Cisco PGW.
• In versions earlier than Cisco Unified CME 4.2, the caller ID displays correctly only after connect;
caller ID does not display correctly at Call Transfer or Call Forward.

Call-Transfer Recall
• Requires Cisco Unified CME 4.3 or a later version.

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• Transferor and transfer-to party must be on the same Cisco Unified CME router; transferee party can
be remote to the Cisco Unified CME router.
• Transfer recall is not supported if the transfer-to party has Call Forward Busy enabled or is a member
of any hunt group.
• If the transfer-to party has Call Forward No Answer enabled, Cisco Unified CME recalls a
transferred call only if the transfer-recall timeout is set to less than the timeout value set with the
call-forward noan command.
• Recall timer for trunk-line directory number has precedence (set on transferor using trunk
command with transfer-timeout keyword) over the transfer-recall timer. Transfer recall is not
initiated for hairpin transfers.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. transfer-system {blind | full-blind | full-consult [dss] | local-consult}
5. transfer-pattern transfer-pattern [blind]
6. call-forward pattern pattern
7. timeouts transfer-recall seconds
8. transfer-digit-collect {new-call | orig-call}
9. exit
10. voice service voip
11. supplementary-service h450.2
12. supplementary-service h450.3
13. exit
14. dial-peer voice tag voip
15. supplementary-service h450.2
16. supplementary-service h450.3
17. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 transfer-system {blind | full-blind | Specifies the call transfer method.
full-consult [dss] | local-consult}
• blind—Calls are transferred without consultation using
the Cisco proprietary method and a single phone line.
Example: This is the default in versions earlier than
Router(config-telephony)# transfer-system Cisco Unified CME 4.0.
full-consult
• full-blind—Calls are transferred without consultation
using H.450.2 standard methods.
• full-consult—Calls are transferred with consultation
using H.450.2 standard methods and a second phone
line if available. Calls fall back to full-blind if the
second line is unavailable. This is the default in
Cisco Unified CME 4.0 and later versions.

Transfer-system needs to be set at full-consult for the

“transfer by directory” to work. Transfer by directory is
supported by full-consult or blind transfer.

If you want to transfer using

directory/placed/missed/received calls, the
transfer-system needs to be set at full-consult for this to
work appropriately.

When changed to full-consult, you can do "blind

transfer" by selecting the number from the directory
and when the other phone rings, you can press the
soft-key "Transfer" and the call will be transferred to
the number selected and then you can hang up.
• dss—(Optional) Calls are transferred with consultation
to idle monitored lines. All other call-transfer behavior
is identical to full-consult.
• local-consult—Calls are transferred with local
consultation using a second phone line if available. The
calls fall back to blind for nonlocal consultation or
nonlocal transfer target. Not supported if transfer-to
destination is on the Cisco ATA, Cisco VG224, or a
SCCP-controlled FXS port.
• Cisco CME 3.0 and later versions—Use only the
full-blind or full-consult keyword.
• Before Cisco CME 3.0—Use the local-consult or
blind keyword. (Cisco ITS 2.1 can use the full-blind or
full-consult keyword by also using the Tcl script in the
file called app-h450-transfer.x.x.x.x.zip.)

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Command or Action Purpose

Step 5 transfer-pattern transfer-pattern [blind] Allows transfer of telephone calls by Cisco Unified IP
phones to specified phone number patterns. If no transfer
pattern is set, the default is that transfers are permitted only
Example:
Router(config-telephony)# transfer-pattern .T
to other local IP phones.
• transfer-pattern—String of digits for permitted call
transfers. Wildcards are allowed. A pattern of .T
transfers all calling parties using the H.450.2 standard.
• blind—(Optional) When H.450.2 consultative call
transfer is configured, forces transfers that match the
pattern specified in this command to be executed as
blind transfers. Overrides settings made using the
transfer-system and transfer-mode commands.
Note For transfers to nonlocal numbers, transfer-pattern
digit matching is performed before translation-rule
operations. Therefore, you should specify in this
command the digits actually entered by phone users
before they are translated. For more information,
see “Configuring Dialing Plans” on page 375.
Step 6 call-forward pattern pattern Specifies the H.450.3 standard for call forwarding.
• pattern—Digits to match for call forwarding using the
Example: H.450.3 standard. If an incoming calling-party number
Router(config-telephony)# call-forward pattern matches the pattern, it can be forwarded using the
.T H.450.3 standard. A pattern of .T forwards all calling
parties using the H.450.3 standard.
• Calling-party numbers that do not match the patterns
defined with this command are forwarded using Cisco
proprietary call forwarding for backward compatibility.
Note For forwarding to nonlocal numbers, pattern
matching is performed before translation-rule
operations. Therefore, you should specify in this
command the digits actually entered by phone users
before they are translated. For more information,
see “Configuring Dialing Plans” on page 375.
Step 7 timeouts transfer-recall seconds (Optional) Enables Cisco Unified CME to recall a
transferred call if the transfer-to party is busy or does not
answer.
Example:
Router(config-telephony)# timeouts • seconds—Duration, in seconds, to wait before recalling
transfer-recall 30 a transferred call. Range: 1 to 1800. Default: 0
(disabled).
• This command is supported in Cisco Unified CME 4.3
and later versions.
• This command can also be configured in ephone-dn and
ephone-dn-template configuration mode.

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Command or Action Purpose

Step 8 transfer-digit-collect {new-call | orig-call} (Optional) Selects the digit-collection method used for
consultative call transfers.
Example: • new-call—Digits are collected from the new call leg.
Router(config-telephony)# transfer-digit-collec Default value in Cisco Unified CME 4.3 and later
t orig-call versions.
• orig-call—Digits are collected from original call-leg.
Default behavior in versions earlier than
Cisco Unified CME 4.3.
• This command is supported in Cisco Unified CME 4.3
and later versions.
Step 9 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit
Step 10 voice service voip (Optional) Enters voice-service configuration mode to
establish global call transfer and forwarding parameters.
Example:
Router(config)# voice service voip
Step 11 supplementary-service h450.2 (Optional) Enables H.450.2 supplementary services
capabilities globally.
Example: • Default is enabled. Use the no form of this command to
Router(conf-voi-serv)# supplementary-service disable H.450.2 capabilities globally.
h450.2
• You can also use this command in dial-peer
configuration mode to enable H.450.2 services for a
single dial peer.
Step 12 supplementary-service h450.3 (Optional) Enables H.450.3 supplementary services
capabilities globally.
Example: • Default is enabled. Use the no form of this command to
Router(conf-voi-serv)# supplementary-service disable H.450.3 capabilities globally.
h450.3
• You can also use this command in dial-peer
configuration mode to enable H.450.3 services for a
single dial peer.
Step 13 exit (Optional) Exits voice-service configuration mode.

Example:
Router(conf-voi-serv)# exit
Step 14 dial-peer voice tag voip (Optional) Enters dial-peer configuration mode.

Example:
Router(config)# dial-peer voice 1 voip

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Command or Action Purpose

Step 15 supplementary-service h450.2 (Optional) Enables H.450.2 supplementary services
capabilities for an individual dial peer.
Example: • Default is enabled. You can also use this command in
Router(config-dial-peer)# no voice-service configuration mode to enable H.450.2
supplementary-service h450.2 services globally.
• If this command is enabled globally and enabled on a
dial peer, the functionality is enabled for the dial peer.
This is the default.
• If this command is enabled globally and disabled on a
dial peer, the functionality is disabled for the dial peer.
• If this command is disabled globally and either enabled
or disabled on a dial peer, the functionality is disabled
for the dial peer.
Step 16 supplementary-service h450.3 (Optional) Enables H.450.3 supplementary services
capabilities exchange for an individual dial peer.
Example: • Default is enabled. You can also use this command in
Router(config-dial-peer)# no voice-service configuration mode to enable H.450.3
supplementary-service h450.3 services globally.
• If this command is enabled globally and enabled on a
dial peer, the functionality is enabled for the dial peer.
This is the default.
• If this command is enabled globally and disabled on a
dial peer, the functionality is disabled for the dial peer.
• If this command is disabled globally and either enabled
or disabled on a dial peer, the functionality is disabled
for the dial peer.
Step 17 end Returns to privileged EXEC mode.

Example:
Router(config-dial-peer)# end

SCCP: Enabling Call Forwarding for a Directory Number

To define the conditions and target numbers for call forwarding for individual ephone-dns, and set other
restrictions for call forwarding, perform the following steps.

Note When defining call forwarding to nonlocal numbers, it is important to note that pattern digit matching is
performed before translation-rule operations. Therefore, you should specify in this command the digits
actually entered by phone users before they are translated. For more information, see the “Voice
Translation Rules and Profiles” section in “Configuring Dialing Plans” on page 375.

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Restrictions
• Call forwarding is invoked only if that phone is dialed directly. Call forwarding is not invoked when
the phone number is called through a sequential, longest-idle, or peer hunt group.
• If call forwarding is configured for hunt group member, call forward is ignored by the hunt group.
• Calls from an internal extension to an extension which is busy, is forwarded to the SNR destination
even if no forward local-calls is configured under the Directory Number.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. call-forward pattern pattern
5. exit
6. ephone-dn dn-tag [dual-line]
7. number number [secondary number] [no-reg [both | primary]]
8. call-forward all target-number
9. call-forward busy target-number [primary | secondary] [dialplan-pattern]
10. call-forward noan target-number timeout seconds [primary | secondary] [dialplan-pattern]
11. call-forward night-service target-number
12. call-forward max-length length
13. no forward local-calls
14. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)#

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Command or Action Purpose

Step 4 call-forward pattern pattern Specifies the H.450.3 standard for call forwarding.
Calling-party numbers that do not match the patterns
defined with this command are forwarded using
Example:
Router(config-telephony)# call-forward pattern
Cisco-proprietary call forwarding for backward
.T compatibility.
• pattern—Digits to match for call forwarding using the
H.450.3 standard. If an incoming calling-party number
matches the pattern, it is forwarded using the H.450.3
standard. A pattern of .T forwards all calling parties
using the H.450.3 standard.
Step 5 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit
Step 6 ephone-dn dn-tag [dual-line] Enters ephone-dn configuration mode, creates an
ephone-dn, and optionally assigns it dual-line status.
Example: • dual-line—(Optional) Enables an ephone-dn with one
Router(config)# ephone-dn 20 voice port and two voice channels, which supports
features such as call waiting, call transfer, and
conferencing with a single ephone-dn.
Step 7 number number [secondary number] [no-reg [both Configures a valid extension number for this ephone-dn
| primary]] instance.

Example:
Router(config-ephone-dn)# number 2777 secondary
2778
Step 8 call-forward all target-number Forwards all calls for this extension to the specified number.
• target-number—Phone number to which calls are
Example: forwarded.
Router(config-ephone-dn)# call-forward all 2411
Note After you use this command to specify a target
number, the phone user can activate and cancel the
call-forward-all state from the phone using the
CFwdAll soft key or a feature access code (FAC).
Step 9 call-forward busy target-number [primary | Forwards calls for a busy extension to the specified number.
secondary] [dialplan-pattern]

Example:
Router(config-ephone-dn)# call-forward busy
2513
Step 10 call-forward noan target-number timeout seconds Forwards calls for an extension that does not answer.
[primary | secondary] [dialplan-pattern]

Example:
Router(config-ephone-dn)# call-forward noan
2513 timeout 45

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Command or Action Purpose

Step 11 call-forward night-service target-number Automatically forwards incoming calls to the specified
number when night service is active.
Example: • target-number—Phone number to which calls are
Router(config-ephone-dn)# call-forward forwarded.
night-service 2879
Note Night service must also be configured. See
“Configuring Call Coverage Features” on page 837.
Step 12 call-forward max-length length (Optional) Limits the number of digits that can be entered
for a target number when using the CfwdAll soft key on an
IP phone.
Example:
Router(config-ephone-dn)# call-forward • length—Number of digits that can be entered using the
max-length 5 CfwdAll soft key on an IP phone.
Step 13 no forward local-calls (Optional) Specifies that local calls (calls from ephone-dns
on the same Cisco Unified CME system) will not be
forwarded from this extension.
Example:
Router(config-ephone-dn)# no forward • If this extension is busy, an internal caller hears a busy
local-calls signal.
• If this extension does not answer, the internal caller
hears ringback.
Step 14 end Returns to privileged EXEC mode.

Example:
Router(config-ephone-dn)# end

SCCP: Enabling Call Transfer for a Directory Number

To enable call transfer for a specific directory number, perform the following steps. This procedure
overrides the global setting for blind or consultative transfer for individual directory numbers.

Prerequisites
Call transfer must be enabled globally. See the “Enabling Call Transfer and Forwarding at
System-Level” section on page 779.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-dn dn-tag [dual-line]
4. transfer-mode {blind | consult}
5. timeouts transfer-recall seconds
6. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-dn dn-tag [dual-line] Enters ephone-dn configuration mode, creates an ephone-dn,
and optionally assigns it dual-line status.
Example: • dual-line—(Optional) Enables an ephone-dn with one
Router(config)# ephone-dn 20 voice port and two voice channels, which supports
features such as call waiting, call transfer, and
conferencing with a single ephone-dn.
Step 4 transfer-mode {blind | consult} Specifies the type of call transfer for an individual directory
number using the H.450.2 standard, allowing you to override
the global setting.
Example:
Router(config-ephone-dn)# transfer-mode blind • Default: system-level value set with the transfer-system
command.
Step 5 timeouts transfer-recall seconds (Optional) Enables call-transfer recall and sets the number of
seconds that Cisco Unified CME waits before recalling a
transferred call if the transfer-to party does not answer or is
Example:
Router(config-ephone-dn)# timeouts
busy.
transfer-recall 30 • seconds—Duration, in seconds, to wait before recalling a
transferred call. Range: 1 to 1800. Default: 0 (disabled).
• This command is supported in Cisco Unified CME 4.3
and later versions.
• This command can also be configured in
ephone-dn-template and telephony-service configuration
mode.
Step 6 end Returns to privileged EXEC mode.

Example:
Router(config-ephone-dn)# end

SCCP: Configuring Call Transfer Options for Phones

To specify a maximum number of digits for transfer destinations or block transfers to external
destinations by individual phones, perform the following steps.

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Restrictions
• Transfers made to speed-dial numbers are not blocked when the transfer-pattern blocked
command is used.
• Transfers made using speed-dial are not blocked by the after-hours block pattern command.

SUMMARY STEPS

1. enable
2. configure terminal
3. ephone-template template-tag
4. transfer-pattern blocked
5. transfer max-length digit-length
6. exit
7. ephone phone-tag
8. ephone-template template-tag
9. restart
10. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 ephone-template template-tag Enters ephone-template configuration mode.
• template-tag—Unique number that identifies this
Example: template during configuration tasks. Range: 1 to 20.
Router(config)# ephone-template 1
Step 4 transfer-pattern blocked (Optional) Prevents directory numbers on the phone to
which this template is applied from transferring calls to
patterns specified in the transfer-pattern
Example:
Router(config-ephone-template)#
(telephony-service) command.
transfer-pattern blocked Note This command is also available in ephone
configuration mode to block external transfers from
individual phones without using a template.

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Command or Action Purpose

Step 5 transfer max-length digit-length (Optional) Specifies the maximum number of digits the user
can dial when transferring a call.
Example: • digit-length—Number of digits allowed in a number to
Router(config-ephone-template)# transfer which a call is being transferred. Range: 3 to 16.
max-length 8 Default: 16.
Step 6 exit Exits ephone-template configuration mode.

Example:
Router(config-ephone-template)# exit
Step 7 ephone phone-tag Enters ephone configuration mode.

Example:
Router(config)# ephone 25
Step 8 ephone-template template-tag Applies a template to a phone.
• template-tag—Template number that you want to apply
Example: to this phone.
Router(config-ephone)# ephone-template 1
Step 9 restart Performs a fast reboot of this phone without contacting the
DHCP server for updated information.
Example: • Repeat Step 6 to Step 9 for each phone on which you
Router(config-ephone)# restart want to limit transfer capabilities.
Step 10 end Exits to privileged EXEC mode.

Example:
Router(config-ephone)# end

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SCCP: Verifying Call Transfer

Step 1 Use the show running-config command to verify your configuration. Transfer method and patterns are
listed in the telephony-service portion of the output. You can also use the show telephony-service
command to display this information.
Router# show running-config
!
telephony-service
fxo hook-flash
load 7910 P00403020214
load 7960-7940 P00305000600
load 7914 S00103020002
load 7905 CP7905040000SCCP040701A
max-ephones 100
max-dn 500
ip source-address 10.115.33.177 port 2000
max-redirect 20
no service directed-pickup
timeouts ringing 10
voicemail 7189
max-conferences 8 gain -6
moh music-on-hold.au
web admin system name cisco password cisco
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern 92......
transfer-pattern 91..........
transfer-pattern 93......
transfer-pattern 94......
transfer-pattern 95......
transfer-pattern 96......
transfer-pattern 97......
transfer-pattern 98......
transfer-pattern 99......
transfer-pattern .T
secondary-dialtone 9
!
create cnf-files version-stamp 7960 Jul 13 2004 03:39:28

Step 2 If you have used the transfer-mode command to override the global transfer mode for an individual
ephone-dn, use the show running-config or show telephony-service ephone-dn command to verify that
setting.
Router# show running-config
!
ephone-dn 40 dual-line
number 451
description Main Number
huntstop channel
no huntstop
transfer-mode blind

Step 3 Use the show telephony-service ephone-template command to view ephone-template configurations.

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SIP: Specifying Transfer Patterns for Trunk-to-Trunk Calls and Conferences

To specify transfer patterns that will enable trunk-to-trunk calls and conferences, perform the following
steps.

Prerequisites
Cisco Unified CME 9.5 or a later version.

Restrictions
Call transfer and conference restrictions apply when transfers or conferences are initiated toward
external parties, like a PSTN trunk, a SIP trunk, or an H.323 trunk. The restrictions do not apply to
transfers to local extensions.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. transfer-pattern transfer-pattern
5. exit
6. voice register pool pool-tag
or
voice register template template-tag
7. transfer max-length max-length
8. exit
9. telephony-service
10. conference transfer-pattern
11. restart
12. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode for
configuring Cisco Unified CME.
Example:
Router(config)# telephony-service
Step 4 transfer-pattern transfer-pattern Allows the transfer of calls from Cisco IP phones to
specified directory numbers of phones other than Cisco IP
phones.
Example:
Router(config-telephony)# transfer-pattern • transfer-pattern—String of digits for permitted call
1234... transfers. Wildcards are allowed. A maximum of 32
Router(config-telephony)# transfer-pattern
transfer patterns can be entered, using a separate
2468..
command for each one.
Step 5 exit Exits telephony-service configuration mode and enters
global configuration mode.
Example:
Router(config-telephony)# exit
Step 6 voice register pool pool-tag Enters voice register pool configuration mode and creates a
or pool configuration for a Cisco Unified SIP IP phone in
voice register template template-tag
Cisco Unified CME or for a set of Cisco Unified SIP IP
phones in Cisco Unified SIP SRST.
Example: • pool-tag—Unique number assigned to the pool. Range
Router(config)# voice register pool 25
is 1 to 100.
or
Enters voice register template configuration mode and
defines a template of common parameters for Cisco Unified
SIP IP phones.
• template-tag—Declares a template tag. Range is 1 to
10.
Step 7 transfer max-length max-length (Optional) Specifies the maximum length of the transfer
number.
Example: • max-length—Maximum length of the transfer number.
Router(config-register-pool)# transfer Range is 3 to 16.
max-length 7

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Command or Action Purpose

Step 8 exit Enters global configuration mode.

Example:
Router(config-register-pool)# exit
Step 9 telephony-service Enters telephony-service configuration mode for
configuring Cisco Unified CME.
Example:
Router(config)# telephony-service
Step 10 conference transfer-pattern Enables a Cisco Unified CME system to apply transfer
patterns to a conference call using conference softkeys or
feature buttons.
Example:
Router(config-telephony)# conference
transfer-pattern
Step 11 end Exits telephony-service configuration mode and enters
privileged EXEC mode.
Example:
Router(config-telephony)# end

SIP: Blocking Trunk-to-Trunk Call Transfers

To block call transfers to external destinations, perform the following steps.

Prerequisites
Cisco Unified CME 9.5 or a later version.

Restrictions
Call transfer restrictions apply when transfers are initiated toward external parties, like a PSTN trunk, a
SIP trunk, or an H.323 trunk. The restrictions do not apply to transfers to local extensions.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register pool pool-tag
or
voice register template template-tag
4. transfer-pattern blocked
5. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register pool pool-tag Enters voice register pool configuration mode and creates a
or pool configuration for a Cisco Unified SIP IP phone in
voice register template template-tag
Cisco Unified CME or for a set of Cisco Unified SIP IP
phones in Cisco Unified SIP SRST.
Example: • pool-tag—Unique number assigned to the pool. Range
Router(config)# voice register template 5
is 1 to 100.
Enters voice register template configuration mode and
defines a template of common parameters for Cisco Unified
SIP IP phones.
• template-tag—Declares a template tag. Range is 1 to
10.
Step 4 transfer-pattern blocked Blocks all call transfers for a specific Cisco Unified SIP IP
phone or a set of Cisco Unified SIP IP phone.
Example:
Router(config-register-temp)# transfer-pattern
blocked
Step 5 end Exits voice register template configuration mode and enters
privileged EXEC mode.
Example:
Router(config-register-temp)# end

Enabling H.450.12 Capabilities

To enable H.450.12 capabilities globally or by individual dial peer when not all gateway endpoints in
your network support H.450.2 and H.450.3 standards, perform the following steps. H.450.12 capabilities
are disabled by default to minimize the risk of compatibility issues with other types of H.323 systems.
Settings for individual dial peers override the global setting.

Restrictions
Cisco CME 3.0 and earlier versions do not support H.450.12.

SUMMARY STEPS

1. enable

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2. configure terminal
3. voice service voip
4. supplementary-service h450.12 [advertise-only]
5. exit
6. dial-peer voice tag voip
7. supplementary-service h450.12
8. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice service voip (Optional) Enters voice service configuration mode to
establish global call transfer and forwarding parameters.
Example:
Router(config)# voice service voip
Step 4 supplementary-service h450.12 [advertise-only] (Optional) Enables H.450.12 supplementary services
capabilities globally for VoIP endpoints.
Example: • This command enables call-by-call detection of H.450
Router(conf-voi-serv)# supplementary-service capabilities when some endpoints in your mixed
h450.12 network are H.450-capable and other endpoints are not.
This command is disabled by default.
• advertise-only—(Optional) Advertises H.450
capabilities to the remote end but does not require
H.450.12 responses. Use this keyword on
Cisco CME 3.1 or later systems if you have a mixed
network containing Cisco CME 3.0 systems.
This command is also used in dial-peer configuration mode
to affect an individual dial peer.
Step 5 exit (Optional) Exits voice-service configuration mode.

Example:
Router(conf-voi-serv)# exit
Step 6 dial-peer voice tag voip (Optional) Enters dial-peer configuration mode.

Example:
Router(config)# dial-peer voice 1 voip

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Command or Action Purpose

Step 7 supplementary-service h450.12 (Optional) Enables H.450.12 supplementary services
capabilities for an individual dial peer. This command is
disabled by default.
Example:
Router(config-dial-peer)# supplementary-service This command is also used in voice-service configuration
h450.12 mode to enable H.450.12 services globally.
• If this command is enabled globally and enabled on a
dial peer, the functionality is enabled for the dial peer.
• If this command is enabled globally and disabled on a
dial peer, the functionality is enabled for the dial peer.
• If this command is disabled globally and enabled on a
dial peer, the functionality is enabled for the dial peer.
• If this command is disabled globally and disabled on a
dial peer, the functionality is disabled for the dial peer.
This is the default.
Step 8 end Returns to privileged EXEC mode.

Example:
Router(config-dial-peer)# end

Enabling H.323-to-H.323 Connection Capabilities

VoIP-to-VoIP connections permit the termination and reorigination of transferred and forwarded calls
over the VoIP network. VoIP-to-VoIP connections are used for hairpin call routing and for H.450 tandem
gateways. The only type of VoIP-to-VoIP connection that is supported by Cisco CME 3.1 or a later
version is H.323-to-H.323 connection.
VoIP-to-VoIP connections are disabled on the router by default, and they must be explicitly enabled to
make use of hairpin call routing or an H.450 tandem gateway. In addition, you must configure a
mechanism to direct transferred or forwarded calls to the hairpin or the H.450 tandem gateway, using
one of the following methods:
• Enable H.450.12 capabilities globally or on the routes that your transfers and forwards take. See the
“Enabling H.450.12 Capabilities” section on page 795.
• Explicitly disable H.450.2 and H.450.3 capabilities globally or on the routes that your transfers and
forwards take. See the “Enabling Call Transfer and Forwarding at System-Level” section on
page 779.

Restrictions
• Codecs on all the VoIP dial peers of the H.450 tandem gateway must be the same.
• Only one codec type is supported in the VoIP network at a time, and there are only two codec
choices: G.711 (A-law or mu-law) or G.729.
• Transcoding is not supported.
• Codec renegotiation is not supported. For example, if an H.323 call that uses a G.729 codec is
received by a Cisco Unified CME system and is forwarded to a voice-mail system that requires a
G.711 codec, the codec cannot be renegotiated from G.729 to G.711.

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• H.323-to-SIP hairpin call routing is supported only with Cisco Unity Express. For more
information, see Integrating Cisco CallManager Express with Cisco Unity Express.
• Cisco Unified Communications Manager must use a media termination point (MTP), intercluster
trunk (ICT) mode, and slow start.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice service voip
4. allow-connections h323 to h323
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice service voip Enters voice service configuration mode to establish global
call transfer and forwarding parameters.
Example:
Router(config)# voice service voip
Step 4 allow-connections h323 to h323 Enables VoIP-to-VoIP call connections. Use the no form of
the command to disable VoIP-to-VoIP connections; this is
the default.
Example:
Router(conf-voi-serv)# allow-connections h323
to h323
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-voi-serv)# end

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Forwarding Calls Using Local Hairpin Routing

When Cisco Unified CME is used to forward calls that originate on phones that do not support the
H.450.3 standard such as Cisco Unified Communications Manager phones, local hairpin routing must be
used to forward the calls. For calling parties whose numbers match the pattern specified, the system
automatically detects whether H.450.3 is supported and uses the appropriate method to forward calls.
To enable hairpin routing, you must denote the originating and terminating legs of the hairpin. To
forward calls to Cisco Unity Express, connections must be allowed to a SIP trunk.
Optionally, you can disable the use of H.450.3 but this is not required because the system automatically
detects calls on which H.450.3 is not supported and local hairpin routing is required when the
calling-party numbers match the pattern specified.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. call-forward pattern pattern
5. calling-number local
6. exit
7. voice service voip
8. allow connections from-type to to-type
9. supplementary-service h450.3
10. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service

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Command or Action Purpose

Step 4 call-forward pattern pattern Specifies the calling-party numbers for which to allow call
forwarding with automatic detection of whether H.450.3 is
supported. If H.450.3 is supported, H.450.3 is used for the
Example:
Router(config-telephony)# call-forward pattern
forward and, if not, local hairpin is used.
6000 • pattern—Digits to match for call forwarding. A pattern
of .T forwards all calling parties.
Step 5 calling-number local (Optional) Replaces a calling-party number and name with
the forwarding-party (local) number and name for
hairpin-forwarded calls only.
Example:
Router(config-telephony)# calling-number local • Before Cisco CME 3.3, this command must be used
with Tool Command Language (Tcl) script
app-h450-transfer.2.0.0.7 or a later version. The
local-hairpin attribute-value (AV) pair must be set to 1.
Step 6 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit
Step 7 voice service voip Enters voice-service configuration mode.

Example:
Router(config)# voice service voip
Step 8 allow connections from-type to to-type Allows connections between specific types of endpoints in
a network.
Example: • from-type—Originating endpoint type. Valid choices
Router(conf-voi-serv)# allow connections h323 are h323 and sip.
to sip
• to-type—Terminating endpoint type. Valid choices are
h323 and sip.
Step 9 supplementary-service h450.3 (Optional) Enables H.450.3 supplementary services
capabilities exchange globally. This is the default. Use the
no form of this command to disable H.450.3 capabilities
Example:
Router(conf-voi-serv)# no supplementary-service
globally. This command can also be used in dial-peer
h450.3 configuration mode to disable H.450.3 functionality for a
single dial peer.
Note If this command is disabled globally and either
enabled or disabled on a dial peer, the functionality
is disabled for the dial peer.
Step 10 end Exits to privileged EXEC mode.

Example:
Router(config-voi-serv)# end

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Enabling H.450.7 and QSIG Supplementary Services at a System-Level

To enable H.4350.7 capabilities and QSIG supplementary services on all dial peers, perform the
following steps.

Prerequisites
Cisco Unified CME 4.0 or a later version.

Restrictions
• QSIG integration supports SCCP phones only.
• QSIG integration is exclusive; once QSIG integration is configured, QSIG transit node capability is
disabled. There is no dial-peer control to enable either transit or originate/terminate capability on a
call by call basis.
• If you enable QSIG supplementary services at a system-level, you cannot disable the capability on
individual dial peers.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice service voip
4. supplementary-service h450.7
5. qsig decode
6. exit
7. voice service pots
8. supplementary-service qsig call-forward
9. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 voice service voip Enters VoIP voice-service configuration mode to define
global call transfer and forwarding parameters.
Example:
Router(config)# voice service voip

Step 4 supplementary-service h450.7 Enables H.450.7 supplementary services capabilities

exchange at a system-level.
Example:
Router(config-voi-serv)# supplementary-service
h450.7
Step 5 qsig decode Enables decoding for QSIG supplementary services.

Example:
Router(config-voi-serv)# qsig decode
Step 6 exit Exits VoIP voice-service configuration mode.

Example:
Router(config-voi-serv)# exit
Step 7 voice service pots Enters POTS voice-service configuration mode to define
global call transfer and forwarding parameters.
Example:
Router(config)# voice service pots
Step 8 supplementary-service qsig call-forward Enables QSIG call-forwarding supplementary services
(ISO 13873) to forward calls to another number.
Example:
Router(config-voi-serv)# supplementary-service
qsig call-forward
Step 9 end Exits to privileged EXEC mode.

Example:
Router(config-voi-serv)# end

Enabling H.450.7 and QSIG Supplementary Services on a Dial Peer

To enable H.4350.7 capabilities and QSIG supplementary services on an individual dial peer, perform
the following steps.

Prerequisites
Cisco Unified CME 4.0 or a later version.

Restrictions
• QSIG integration supports SCCP phones only.

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• QSIG integration is exclusive; once QSIG integration is configured, QSIG transit node capability is
disabled. There is no dial-peer control to enable either transit or originate/terminate capability on a
call by call basis.
• If you enable QSIG supplementary services at a system-level, you cannot enable or disable the
capability on individual dial peers.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice service voip
4. qsig decode
5. exit
6. dial-peer voice tag voip
7. supplementary-service h450.7
8. exit
9. dial-peer voice tag pots
10. supplementary-service qsig call-forward
11. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice service voip Enters VoIP voice-service configuration mode to define
global call transfer and forwarding parameters.
Example:
Router(config)# voice service voip
Step 4 qsig decode Enables decoding for QSIG supplementary services.

Example:
Router(config-voi-serv)# qsig decode
Step 5 exit Exits VoIP voice-service configuration mode.

Example:
Router(config-voi-serv)# exit

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Command or Action Purpose

Step 6 dial-peer voice tag voip Enters dial-peer configuration mode to define parameters
for an individual dial peer.
Example:
Router(config)# dial-peer voice 1 voip
Step 7 supplementary-service h450.7 Enables H.450.7 supplementary services capabilities
exchange on a single dial peer.
Example:
Router(config-dial-peer)# supplementary-service
h450.7
Step 8 exit Exits dial-peer configuration mode.

Example:
Router(config-dial-peer)# exit
Step 9 dial-peer voice tag pots Enters dial-peer configuration mode to define parameters
for an individual dial peer.
Example:
Router(config)# dial-peer voice 2 pots
Step 10 supplementary-service qsig call-forward Enables QSIG call-forwarding supplementary services
(ISO 13873) to forward calls to another number.
Example:
Router(config-dial-peer)# supplementary-service
qsig call-forward
Step 11 end Exits to privileged EXEC mode.

Example:
Router(config-dial-peer)# end

Disabling SIP Supplementary Services for Call Forward and Call Transfer
To disable REFER messages for call transfers or redirect responses for call forwarding from being sent
to the destination by Cisco Unified CME, perform the following steps. You can disable these
supplementary features if the destination gateway does not support them.

Prerequisites
Cisco Unified CME 4.1 or a later version.

Restrictions
• In Cisco Unified CME 4.2 and 4.3, when the supplementary-service sip refer command is enabled
(default) and both the caller being transferred (transferee) and the phone making the transfer
(transferor) are SIP, but the transfer-to phone is SCCP, Cisco Unified CME hairpins the call to the
transfer-to phone after receiving the REFER request from transferor instead of sending the REFER
request to the transferee.

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SUMMARY STEPS

1. enable
2. configure terminal
3. voice service voip
or
dial-peer voice tag voip
4. no supplementary-service sip moved-temporarily
5. no supplementary-service sip refer
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice service voip Enters voice-service configuration mode to set global
or parameters for VoIP features.
dial-peer voice tag voip or
Enters dial peer configuration mode to set parameters for a
Example: specific dial peer.
Router(config)# voice service voip
or
Router(config)# dial-peer voice 99 voip
Step 4 no supplementary-service sip moved-temporarily Disables SIP redirect response for call forwarding either
globally or for a dial peer.
Example: • Sending redirect message to the destination is the
Router(conf-voi-serv)# no supplementary-service default behavior.
sip moved-temporarily
or
Router(config-dial-peer)# no
supplementary-service sip moved-temporarily

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Command or Action Purpose

Step 5 no supplementary-service sip refer Disables SIP REFER message for call transfers either
globally or for a dial peer.
Example: • Sending REFER message to the destination is the
Router(conf-voi-serv)# no supplementary-service default behavior.
sip refer
or
Router(config-dial-peer)# no
supplementary-service sip refer
Step 6 end Exits to privileged EXEC mode.

Example:
Router(config-voi-serv)# end
or
Router(config-dial-peer)# end

Enabling Interworking with Cisco Unified Communications Manager

If Cisco CME 3.1 or later and Cisco Unified Communications Manager are used in the same network,
some additional configuration is necessary, as described in the following sections:
• Configuring Cisco CME 3.1 or Later to Interwork with Cisco Unified Communications Manager,
page 807
• Enabling Cisco Unified Communications Manager to Interwork with Cisco Unified CME, page 810
• Troubleshooting Transfer and Forwarding Configuration, page 811
Figure 40 shows a network containing Cisco Unified CME and Cisco Unified Communications
Manager systems.

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Figure 40 Network with Cisco Unified CME and Cisco Unified Communications Manager

Cisco Unified CallManager 1 Cisco Unified CallManager 2

IP IP
Cisco Unified CallManager 3
Phone 1 Phone 2
4001 4002 H.323 Connection
in ICT mode using slow start

Media Termination Point (MTP)

V

Cisco Unified CallManager Network VoIP

Cisco Unified CME Network

PSTN

Cisco Unified CME 1 Cisco Unified CME 2 Cisco Unified CME 3

V V V
Telephone

IP IP IP IP IP IP

146621
Phone 3 Phone 4 Phone 5 Phone 6 Phone 7 Phone 8
1001 1002 2001 2002 3001 3002

Prerequisites
• Cisco Unified CME must be configured to forward calls using local hairpin routing. For
configuration information, see the “Forwarding Calls Using Local Hairpin Routing” section on
page 799.

Configuring Cisco CME 3.1 or Later to Interwork with Cisco Unified Communications Manager
All of the commands in this section are optional because they are set by default to work with
Cisco Unified Communications Manager. They are included here only to explain how to implement
optional capabilities or return nondefault settings to their defaults.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice service voip
4. h323
5. telephony-service ccm-compatible
6. h225 h245-address on-connect

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7. exit
8. supplementary-service h225-notify cid-update
9. exit
10. voice class h323 tag
11. telephony-service ccm-compatible
12. h225 h245-address on-connect
13. exit
14. dial-peer voice tag voip
15. supplementary-service h225-notify cid-update
16. voice-class h323 tag
17. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice service voip Enters voice-service configuration mode to establish global
parameters.
Example:
Router(config)# voice service voip
Step 4 h323 Enters H.323 voice-service configuration mode.

Example:
Router(conf-voi-serv)# h323
Step 5 telephony-service ccm-compatible (Optional) Globally enables a Cisco CME 3.1 or later
system to detect Cisco Unified Communications Manager
and exchange calls with it. This is the default.
Example:
Router(conf-serv-h323)# telephony-service • Use the no form of this command to disable
ccm-compatible Cisco Unified Communications Manager detection and
exchange. We do not recommend using the no form of
the command.
• Using this command in an H.323 voice class definition
allows you to specify this behavior for an individual
dial peer.

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Command or Action Purpose

Step 6 h225 h245-address on-connect (Optional) Globally enables a delay for the H.225 message
exchange of an H.245 transport address until a call is
connected. The delay allows
Example:
Router(conf-serv-h323)# h225 h245-address
Cisco Unified Communications Manager to generate local
on-connect ringback for calls to Cisco Unified CME phones. This is the
default.
• The no form of this command disables the delay. We do
not recommend using the no form of the command.
• Using this command in an H.323 voice class definition
allows you to specify this behavior for an individual
dial peer.
Step 7 exit Exits H.323 voice-service configuration mode.

Example:
Router(conf-serv-h323)# exit
Step 8 supplementary-service h225-notify cid-update (Optional) Globally enables H.225 messages with caller-ID
updates to be sent to Cisco Unified Communications
Manager. This is the default.
Example:
Router(conf-voi-serv)# supplementary-service • The no form of the command disables caller-ID update.
h225-notify cid-update We do not recommend using the no form of the
command.
This command is also used in dial-peer configuration mode
to affect a single dial peer.
• If this command is enabled globally and enabled on a
dial peer, the functionality is enabled for that dial peer.
This is the default.
• If this command is enabled globally and disabled on a
dial peer, the functionality is disabled for that dial peer.
• If this command is disabled globally and either enabled
or disabled on a dial peer, the functionality is disabled
for that dial peer.
Step 9 exit Exits voice-service configuration mode.

Example:
Router(config-voice-service)# exit
Step 10 voice class h323 tag (Optional) Creates a voice class that contains commands to
be applied to one or more dial peers.
Example:
Router(config)# voice class h323 48
Step 11 telephony-service ccm-compatible (Optional) Enables the dial peer to exchange calls with a
Cisco Unified Communications Manager system when this
voice class is applied to a dial peer. This is the default.
Example:
Router(config-voice-class)# telephony-service • The no form of the command disables call exchange
ccm-compatible with Cisco Unified Communications Manager. We do
not recommend using the no form of the command.

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Command or Action Purpose

Step 12 h225 h245-address on-connect (Optional) Enables the calls that use this dial peer to delay
the exchange of H.225 messages that contain the H.245
transport address until calls are connected, when this voice
Example:
Router(config-voice-class)# h225 h245-address
class is applied to a dial peer. The delay allows the playing
on-connect of local ringback for calls from
Cisco Unified Communications Manager. This is the
default.
• The no form of this command disables the delay. We do
not recommend using the no form of the command.
Step 13 exit Exits voice-class configuration mode.

Example:
Router(config-voice-class)# exit
Step 14 dial-peer voice tag voip (Optional) Enters dial-peer configuration mode to set
parameters for an individual dial peer.
Example:
Router(config)# dial-peer voice 28 voip
Step 15 supplementary-service h225-notify cid-update (Optional) Enables H.225 messages with caller-ID updates
to Cisco Unified Communications Manager for a specific
dial peer. This is the default.
Example:
Router(config-dial-peer)# no • The no form of the command disables caller-ID
supplementary-service h225-notify cid-update updates. We do not recommend using the no form of the
command.
Step 16 voice-class h323 tag (Optional) Applies the previously defined voice class with
the specified tag number to this dial peer.
Example:
Router(config-dial-peer)# voice-class h323 48
Step 17 end Exits to privileged EXEC mode.

Example:
Router(config-dial-peer)# end

What to Do Next

Set up Cisco Unified Communications Manager using the configuration procedure in the “Enabling
Cisco Unified Communications Manager to Interwork with Cisco Unified CME” section on page 810.

Enabling Cisco Unified Communications Manager to Interwork with Cisco Unified CME
To enable Cisco Unified Communications Manager to interwork with Cisco CME 3.1 or a later version,
perform the following steps in addition to the normal Cisco Unified Communications Manager
configuration.

SUMMARY STEPS

1. Set Cisco Unified Communications Manager service parameters.

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2. Configure Cisco Unified CME as an ICT in the Cisco Unified Communications Manager network.
3. Ensure that the Cisco Unified Communications Manager network uses an MTP.
4. Set up dial peers to establish routing.

DETAILED STEPS

Step 1 Set Cisco Unified Communications Manager service parameters. From Cisco Unified Communications
Manager Administration, choose Service Parameters. Choose the Cisco Unified Communications
Manager service, and make the following settings:
• Set the H323 FastStart Inbound service parameter to False.
• Set the Send H225 User Info Message service parameter to H225 Info for Ring Back.
Step 2 Configure Cisco Unified CME as an ICT in the Cisco Unified Communications Manager network. For
information about different intercluster trunk types and configuration instructions, see the
Cisco Unified Communications Manager documentation.
Step 3 Ensure that the Cisco Unified Communications Manager network uses an MTP. The MTP is required to
provide DSP resources for transcoding and for sending and receiving G.729 calls to Cisco Unified CME.
All media streams between Cisco Unified Communications Manager and Cisco Unified CME must pass
through the MTP because Cisco CME 3.1 does not support transcoding. For more information, see the
Cisco Unified Communications Manager documentation.
Step 4 Set up dial peers to establish routing using the instructions in the Dial Peer Configuration on Voice
Gateway Routers guide.

Troubleshooting Transfer and Forwarding Configuration

Step 1 If you encounter lack of ringback on direct calls from a Cisco Unified Communications Manager phone
to an IP phone on a Cisco Unified CME system, check the show running-config command output to
ensure that the following two commands do not appear: no h225 h245-address on-connect and no
telephony-service ccm-compatible. These commands should be enabled, which is their default state.
Step 2 Use the debug h225 asn1 command to display the H.323 messages that are sent from the
Cisco Unified CME system to the Cisco Unified Communications Manager system to see if the H.245
address is being sent too early.
Step 3 For calls that are routed using VoIP-to-VoIP connections, use the show voip rtp connections detail
command to display the call identification number, IP addresses, and port numbers involved for all VoIP
call legs. This command includes VoIP-to-POTS and VoIP-to-VoIP call legs. The following is sample
output for this command:
Router# show voip rtp connections detail

VoIP RTP active connections :

No. CallId dstCallId LocalRTP RmtRTP LocalIP RemoteIP
1 7 8 16586 22346 172.27.82.2 172.29.82.2
2 8 7 17010 16590 172.27.82.2 209.165.202.129
Found 2 active RTP connections

Step 4 Use the show call prompt-mem-usage detail command to see information on ringback tone generation
that uses the interactive voice response (IVR) prompt playback mechanism. This ringback is needed for
hairpin transfers that are committed during the alerting-of-the-transfer-destination phase of the call and

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for calls to destinations that do not provide in-band ringback tone, such as IP phones (FXS analog ports
do provide in-band ringback tone). Ringback tone is played to the transferred party by the
Cisco Unified CME system that performs the transfer (the system attached to the transferring party). The
system automatically generates tone prompts as needed based on the network-locale setting for the
Cisco Unified CME system.
If you are not getting ringback tone when you should, use the show call prompt-mem-usage command
to ensure that the correct prompt is loaded and playing. The following sample output indicates that a
prompt is playing (“Number of prompts playing”) and indicates the country code used for the prompt
(GB for Great Britain) and the codec.
Router# show call prompt-mem-usage detail

Prompt memory usage:

config'd wait active free mc total ms total
file(s) 0200 0001 -001 00200 00001 00002
memory 02097152 00003000 00000000 02094152 00003000
Prompt load counts: (counters reset 0)
success 0(1st try) 0(2nd try), failure 0
Other mem block usage:
mcDynamic mcReader
gauge 00001 00001
Number of prompts playing: 1
Number of start delays : 0
MCs in the ivr MC sharing table
===============================
Media Content: NoPrompt (0x83C64554)
URL:
cid=0, status=MC_READY size=24184 coding=g711ulaw refCount=0
Media Content: tone://GB_g729_tone_ringback (0x83266EC8)
URL: tone://GB_g729_tone_ringback

SIP: Configuring SIP-to-SIP Phone Call Forwarding

To configure SIP-to-SIP call forwarding using a back-to-back user agent (B2BUA) which allows call
forwarding on any dial peer, perform the following steps.

Prerequisites
• Cisco CME 3.4 or a later version.
• Connections between specific types of endpoints in a Cisco IP-to-IP gateway must be configured by
using the allow-connections command. For configuration information, see the “Enabling Calls in
Your VoIP Network” on page 88.

Restrictions
• SIP-to-SIP call forwarding is invoked only if that phone is dialed directly. Call forwarding is not
invoked when the phone number is called through a sequential, longest-idle, or peer hunt group.
• If call forwarding is configured for a hunt group member, call forward is ignored by the hunt group.
• In Cisco Unified CME 4.1 and later versions, Call Forward All requires SIP phones to be configured
with a directory number (using dn keyword in number command); direct line numbers are not
supported.

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SUMMARY STEPS

1. enable
2. configure terminal
3. voice register dn dn-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. call-forward b2bua unreachable directory-number
9. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register dn dn-tag Enters voice register dn mode to define a directory number
for a SIP phone, intercom line, voice port, or an MWI.
Example:
Router(config)# voice register dn 1
Step 4 call-forward b2bua all directory- number Enables call forwarding for a SIP back-to-back user agent so
that all incoming calls will be forwarded to the designated
directory-number.
Example:
Router(config-register-dn)# call-forward b2bua • In Cisco CME 3.4 and Cisco Unified CME 4.0, this
all 5005 command is also available in voice register pool
configuration mode. The configuration under voice
register dn takes precedence over the configuration
under voice register pool.
• If the call-forward b2bua all command is configured in
voice register pool configuration mode, it applies to all
directory numbers on the phone.

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Command or Action Purpose

Step 5 call-forward b2bua busy directory- number Enables call forwarding for a SIP back-to-back user agent so
that incoming calls to an extension that is busy will be
forwarded to the designated directory number.
Example:
Router(config-register-dn)# call-forward b2bua • In Cisco CME 3.4 and Cisco Unified CME 4.0, this
busy 5006 command is also available in voice register pool
configuration mode. The configuration under voice
register dn takes precedence over the configuration
under voice register pool.
Step 6 call-forward b2bua mailbox directory- number Enables call forwarding for a SIP back-to-back user agent so
that incoming calls that have been forwarded to a busy or
no-answer extension will be forwarded to the recipient’s
Example:
Router(config-register-dn)# call-forward b2bua
voice mail.
mailbox 5007 • In Cisco CME 3.4 and Cisco Unified CME 4.0, this
command is also available in voice register pool
configuration mode. The configuration under voice
register dn takes precedence over the configuration
under voice register pool.
Step 7 call-forward b2bua noan directory- number Enables call forwarding for a SIP back-to-back user agent so
timeout seconds that incoming calls to an extension that does not answer will
be forwarded to the designated directory number.
Example: • In Cisco CME 3.4 and Cisco Unified CME 4.0, this
Router(config-register-dn)# call-forward b2bua command is also available in voice register pool
noan 5010 timeout 10
configuration mode. The configuration under voice
or
register dn takes precedence over the configuration
Router(config-register-pool)# call-forward under voice register pool.
b2bua noan 5010 timeout 10
• timeout seconds—Duration that a call can ring before it
is forwarded to the destination directory number. Range:
3 to 60000. Default: 20.
Step 8 call-forward b2bua unreachable directory- (Optional) Enables call forwarding for a SIP back-to-back
number user agent so that calls can be forwarded to a phone that has
not registered in Cisco Unified CME.
Example: • Target directory-number must be configured in
Router(config-register-dn)# call-forward b2bua Cisco Unified CME.
unreachable 5009
or • In Cisco CME 3.4 and Cisco Unified CME 4.0, this
command is also available in voice register pool
Router(config-register-pool)# call-forward
b2bua unreachable 5009 configuration mode. The configuration under voice
register dn takes precedence over the configuration
under voice register pool.
• This command was removed in Cisco Unified CME 4.1.
Step 9 end Exits to privileged EXEC mode.

Example:
Router(config-register-dn)# end

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SIP: Configuring Call Forward Unregistered for SIP IP Phones

To configure Call Forward Unregistered (CFU) on SIP IP phones, follow these steps:

Prerequisites
• Cisco Unified CME 8.6 or a later version.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register dn tag
4. call-forward b2bua unregistered directory-number
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register dn tag Enters voice register dn mode to define a directory number for a
SIP phone, intercom line, voice port, or an MWI.
Example:
Router(config)#voice register dn 20
Step 4 call-forward b2bua unregistered Enables call forwarding for a SIP back-to-back user agent so that
directory-number all incoming calls are forwarded to the unregistered
directory-number.
Example:
Router(config-register-dn)#call-forward
b2bua unregistered 2345
Step 5 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

Troubleshooting Tips
– Use the show dial-peer voice summary command to check whether a CFU dial peer is created
or removed.

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– Enable deb voice reg event, deb voice reg state, and deb voice reg error commands to trace
the creation and deletion of the CFU dial peer.
– Enable deb voice reg event, deb voip ccapi inout, deb voip app callsetup, deb voip app core,
deb voip app state, and deb voip app error commands to trace the call flow for CFU.

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Configuring Keepalive Timer Expiration in SIP Phones

To configure keepalive timer expiration in SIP phones, follow these steps:

SUMMARY STEPS

1. enable
2. configure terminal
3. voice service voip
4. sip
5. registrar server [expires [max seconds] [min seconds] ]
6. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice service voip Enters voice-service configuration mode and specifies
voice-over-IP encapsulation.
Example:
Router#(conf)voice service voip
Step 4 sip Enters SIP configuration mode.

Example:
Router#(conf-serv)sip
Step 5 registrar server [expires [max seconds] Enables SIP registrar functionality in Cisco Unified CME.
[min seconds]] • expires—(Optional) Sets the active time for an incoming
registration.
Example: • max sec—(Optional) Maximum time for a registration to
Router(conf-serv-sip)#registrar server expire, in seconds. Range: 120 to 86400.
expires max 250 min 75
• min sec—(Optional) Minimum time for a registration to
expire, in seconds.

Step 6 end Returns to privileged EXEC mode.

Example:
Routerconf-serv-sip)# end

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SIP: Configuring Call-Forwarding-All Soft Key URI

To specify the uniform resource identifier (URI) for the call forward all (CfwdAll) soft key on supported
SIP phones, perform the following steps. This URI and the call forward number is sent to
Cisco Unified CME when a user enables Call Forward All on a SIP phone.

Prerequisites
• Cisco Unified CME 4.1 or a later version.
• The mode cme command must be enabled in Cisco Unified CME.
• Call Forward All must be enabled on the directory number. For information, see “SIP: Configuring
SIP-to-SIP Phone Call Forwarding” on page 812.

Restrictions
• This feature is supported only on Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE,
7970G, and 7971GE.
• If a user enables Call Forward All using the CfwdAll soft key, it is enabled on the primary line.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register global
4. call-feature-uri cfwdall service-uri
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register global Enters voice register global configuration mode to set
global parameters for all supported SIP phones in a
Cisco Unified CME environment.
Example:
Router(config)# voice register global

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Command or Action Purpose

Step 4 call-feature-uri cfwdall service-uri Specifies the URI for soft keys on SIP phones connected to
a Cisco Unified CME router.
Example:
Router(config-register-global)#
call-feature-uri cfwdall
http://1.4.212.11/cfwdall
Step 5 end Exits to privileged EXEC mode.

Example:
Router(config-register-global)# end

SIP: Specifying Number of 3XX Responses To be Handled

To specify how many subsequent 3XX responses an originating SIP phone can handle for a single call
when the terminating side is a forwarding party which does not use B2BUA, perform the following steps.

Prerequisites
• Cisco CME 3.4 or a later version.
• The mode cme command must be enabled

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register global
4. phone-redirect-limit number
5. end

DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal

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Command or Action Purpose

Step 3 voice register global Enters voice register global configuration mode to set
parameters for all supported SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register global
Step 4 phone-redirect-limit number Changes the default number of 3XX responses a SIP phone
that originates a call can handle for a single call.
Example: • Default: 5.
Router(config-register-global)#
phone-redirect-limit 8
Step 5 end Exits to privileged EXEC mode.

Example:
Router(config-register-global)# end

SIP: Configuring Call Transfer

To create and apply a template to enable call transfer softkeys on an individual SIP phone in
Cisco Unified CME, perform the following steps.

Prerequisites
Cisco CME 3.4 or a later version.

Restrictions
• Blind transfer is not supported on certain phones such as Cisco Unified IP Phone 7911G, 7941G,
7941GE, 7961G, 7961GE, 7970G, or 7971GE.
• In Cisco Unified CME 4.1, the soft key display can be customized only for certain IP phones, such
as Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE. For
configuration information, see “SIP: Modifying Soft-Key Display” on page 1343.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice register template template-tag
4. transfer-attended
5. transfer-blind
6. exit
7. voice register pool pool-tag
8. template template-tag
9. end

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DETAILED STEPS

Command or Action Purpose

Step 1 enable Enables privileged EXEC mode.
• Enter your password if prompted.
Example:
Router> enable
Step 2 configure terminal Enters global configuration mode.

Example:
Router# configure terminal
Step 3 voice register template template-tag Enters voice register template configuration mode to define
a template of common parameters for SIP phones in
Cisco Unified CME.
Example:
Router(config)# voice register template 1 • Range: 1 to 5.
Step 4 transfer-attended Enable a soft key for attended transfer on any supported SIP
phone that uses a template in which this command is
configure.
Example:
Router(config-register-template)#
transfer-attended
Step 5 transfer-blind Enable a soft key for blind transfer on any supported SIP
phone that uses a template in which this command is
configure.
Example:
Router(config-register-template)#
transfer-blind
Step 6 exit Exits configuration mode to the next highest mode in the
configuration mode hierarchy.
Example:
Router(config-register-template)# exit
Step 7 voice register pool pool-tag Enters voice register pool configuration mode to set
phone-specific parameters for SIP phones.
Example:
Router(config)# voice register pool 3
Step 8 template template-tag Applies a template created with the voice register template
command.
Example: • template-tag—Range: 1 to 5.
Router(config-register-pool)# voice register
pool 1
Step 9 end Exits to privileged EXEC mode.

Example:
Router(config-register-pool)# end

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Configuration Examples for Call Transfer and Forwarding

The following configuration examples are included in this section:
• H.450.2 and H.450.3: Example, page 822
• Basic Call Forwarding: Example, page 823
• Call Forwarding Blocked for Local Calls: Example, page 823
• Configuring Transfer Patterns: Example, page 823
• Configuring Maximum Length of Transfer Number: Example, page 823
• Configuring Conference Transfer Patterns: Example, page 824
• Blocking All Call Transfers: Example, page 824
• Selective Call Forwarding: Example, page 824
• Call Transfer: Example, page 824
• Call-Transfer Recall: Example, page 825
• H.450.12: Example, page 826
• H.450.7 and QSIG Supplementary Services: Example, page 826
• Cisco Unified CME and Cisco Unified Communications Manager in Same Network: Example,
page 826
• H.450 Tandem Gateway Working with Cisco Unified CME and
Cisco Unified Communications Manager: Example, page 828
• Forwarding Calls to Cisco Unity Express: Example, page 830
• Configuring Call Forward Unregistered for SIP IP Phones: Example, page 831
• Configuring Keepalive Timer Expiration in SIP Phones: Example, page 831

H.450.2 and H.450.3: Example

The following example sets all transfers and forwards that are initiated by a Cisco CME 3.0 or later
system to use the H.450 standards, globally enables H.450.2 and H.450.3 capabilities, and disables those
capabilities for dial peer 37. The supplementary-service commands under voice-service configuration
mode are not necessary because these values are the default, but they are shown here for illustration.
telephony-service
transfer-system full-consult
transfer-pattern .T
call-forward pattern .T
!
voice service voip
supplementary-service h450.2
supplementary-service h450.3
!
dial-peer voice 37 voip
destination-pattern 555....
session target ipv4:10.5.6.7
no supplementary-service h450.2
no supplementary-service h450.3

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Basic Call Forwarding: Example

The following example sets up forwarding for extension 2777 to extension 2513 on all calls, busy, and
no answer. During night service hours, calls are forwarded to a different number, extension 2879.
ephone-dn 20
number 2777
call-forward all 2513
call-forward busy 2513
call-forward noan 2513 timeout 45
call-forward night-service 2879

Call Forwarding Blocked for Local Calls: Example

In the following example, extension 2555 is configured to not forward local calls that are internal to the
Cisco Unified CME system. Extension 2222 dials extension 2555. If 2555 is busy, the caller hears a busy
tone. If 2555 does not answer, the caller hears ringback. The internal call is not forwarded.
ephone-dn 25
number 2555
no forward local-calls
call-forward busy 2244
call-forward noan 2244 timeout 45

Configuring Transfer Patterns: Example

The following example shows how to configure transfer patterns beginning with 1234:
Router# configure terminal
Router(config)# telephony-service
Router(config-telephony)# transfer-pattern 1234

Configuring Maximum Length of Transfer Number: Example

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

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Configuring Conference Transfer Patterns: Example

The following example configures transfer patterns that allow conference calls:
Router# configure terminal
Router(config)# telephony-service
Router(config-telephony)# transfer-pattern 1357
Router(config-telephony)# transfer-pattern 222....
Router(config-telephony)# conference transfer-pattern

Blocking All Call Transfers: Example

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

Selective Call Forwarding: Example

The following example sets call forwarding on busy and no answer for ephone-dn 38 only for its primary
number, 2777. Callers who dial 2778 will hear a busy signal if the ephone-dn is busy or ringback if there
is no answer.
ephone-dn 38
number 2777 secondary 2778
call-forward busy 3000 primary
call-forward noan 3000 primary timeout 45

Call Transfer: Example

The following example limits transfers from ephone 6, extension 2977, to numbers containing a
maximum of 8 digits.
telephony-service
load 7910 P00403020214
load 7960-7940 P00305000600
load 7914 S00103020002
load 7905 CP7905040000SCCP040701A
load 7912 CP7912040000SCCP040701A
max-ephones 100
max-dn 500

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ip source-address 10.104.8.205 port 2000

max-redirect 20
system message XYZ Inc.
create cnf-files version-stamp 7960 Jul 13 2004 03:39:28
voicemail 7189
max-conferences 8 gain -6
moh music-on-hold.au
web admin system name admin1 password admin1
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern 91..........
transfer-pattern 92......
transfer-pattern 93......
transfer-pattern 94......
transfer-pattern 95......
transfer-pattern 96......
transfer-pattern 97......
transfer-pattern 98......
transfer-pattern 99......
secondary-dialtone 9
fac standard

ephone-template 2
transfer max-length 8

ephone-dn 4
number 2977

ephone 6
button 1:4
ephone-template 2

Call-Transfer Recall: Example

The following example shows that transfer recall is enabled globally. After 60 seconds an unanswered
call is forwarded back to the phone that initiated the transfer (transferor).
telephony-service
max-ephones 100
max-dn 240
timeouts transfer-recall 60
max-conferences 8 gain -6
transfer-system full-consult

The following example shows that transfer recall is enabled for extension 1030 (ephone-dn 103), which
is assigned to ephone 3. If extension 1030 forwards a call and the transfer-to party does not answer, after
60 seconds the unanswered call is sent back to extension 1030 (transferor). The timeouts transfer-recall
command can also be set in an ephone-dn template and applied to one or more directory numbers.
ephone-dn 103
number 1030
name Smith, John
timeouts transfer-recall 60
!
ephone 3
mac-address 002D.264E.54FA
type 7962
button 1:103

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H.450.12: Example
The following example globally disables H.450.12 capabilities and then enables them only on
dial peer 24.
voice service voip
no supplementary-service h450.12
!
dial-peer voice 24 voip
destination-pattern 555....
session target ipv4:10.5.6.7
supplementary-service h450.12

H.450.7 and QSIG Supplementary Services: Example

The following example implements QSIG supplementary services on extension 74367 and globally
enables H.450.7 supplementary services and QSIG call-forwarding supplementary services.
telephony-service
voicemail 74398
transfer-system full-consult

ephone-dn 25
number 74367
mwi qsig
call-forward all 74000

voice service voip

supplementary-service h450.7

voice service pots

supplementary-service qsig call-forward

Cisco Unified CME and Cisco Unified Communications Manager in Same

Network: Example
The following example shows a running configuration for a Cisco CME 3.1 or later router that has a
Cisco Unified Communications Manager in its network.
Router# show running-config

version 12.3
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
enable password pswd
!
aaa new-model
!
!
aaa session-id common
no ip subnet-zero
!
ip dhcp pool phone1
host 172.24.82.3 255.255.255.0

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client-identifier 0100.07eb.4629.9e
default-router 172.24.82.2
option 150 ip 172.24.82.2
!
ip dhcp pool phone2
host 172.24.82.4 255.255.255.0
client-identifier 0100.0b5f.f932.58
default-router 172.24.82.2
option 150 ip 172.24.82.2
!
ip cef
no ip domain lookup
no mpls ldp logging neighbor-changes
no ftp-server write-enable
!
voice service voip
allow-connections h323 to h323
!
voice class codec 1
codec preference 1 g711ulaw
!
no voice hpi capture buffer
no voice hpi capture destination
!
interface FastEthernet0/0
ip address 172.24.82.2 255.255.255.0
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip bind srcaddr 172.24.82.2
!
ip classless
ip route 0.0.0.0 0.0.0.0 172.24.82.1
ip route 192.168.254.254 255.255.255.255 172.24.82.1
!
ip http server
!
tftp-server flash:P00303020700.bin
!
voice-port 1/0/0
!
voice-port 1/0/1
!
dial-peer cor custom
!
dial-peer voice 1001 voip
description points-to-CCM
destination-pattern 1.T
voice-class codec 1
session target ipv4:172.26.82.10
!
dial-peer voice 1002 voip
description points to router
destination-pattern 4...
voice-class codec 1
session target ipv4:172.25.82.2
!
dial-peer voice 1 pots
destination-pattern 3000
port 1/0/0
!
dial-peer voice 1003 voip
destination-pattern 26..
session target ipv4:10.22.22.38

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Configuration Examples for Call Transfer and Forwarding

!
!
telephony-service
load 7960-7940 P00303020700
max-ephones 48
max-dn 15
ip source-address 172.24.82.2 port 2000
create cnf-files version-stamp Jan 01 2002 00:00:00
keepalive 10
max-conferences 4
moh minuet.au
transfer-system full-consult
transfer-pattern ....
!
ephone-dn 1
number 3001
name abcde-1
call-forward busy 4001
!
ephone-dn 2
number 3002
name abcde-2
!
ephone-dn 3
number 3003
name abcde-3
!
ephone-dn 4
number 3004
name abcde-4
!
ephone 1
mac-address 0003.EB27.289E
button 1:1 2:2
!
ephone 2
mac-address 000D.39F9.3A58
button 1:3 2:4
!
line con 0
exec-timeout 0 0
logging synchronous
line aux 0
line vty 0 4
password pswd
!
end

H.450 Tandem Gateway Working with Cisco Unified CME and

Cisco Unified Communications Manager: Example
The following example shows a sample configuration for a Cisco CME 3.1 or later system that is linked
to an H.450 tandem gateway that serves as a proxy for Cisco Unified Communications Manager.
Router# show running-config

Building configuration...

Current configuration : 1938 bytes

!
version 12.3

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service timestamps debug datetime msec

service timestamps log datetime msec
no service password-encryption
!
hostname Router
!
boot-start-marker
boot-end-marker
!
enable password pswd
!
aaa new-model
!
aaa session-id common
no ip subnet-zero
!
ip cef
no ip domain lookup
no ftp-server write-enable
no scripting tcl init
no scripting tcl encdir
!
voice call send-alert
!
voice service voip
allow-connections h323 to h323
supplementary-service h450.12
h323
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
codec preference 3 g729br8
!
interface FastEthernet0/0
ip address 172.27.82.2 255.255.255.0
duplex auto
speed auto
h323-gateway voip interface
h323-gateway voip h323-id host24
!
ip classless
ip route 0.0.0.0 0.0.0.0 172.26.82.1
ip route 0.0.0.0 0.0.0.0 172.27.82.1
ip http server
!
dial-peer cor custom
!
dial-peer voice 1001 voip
description points-to-CCM
destination-pattern 4...
session target ipv4:172.24.89.150
!
dial-peer voice 1002 voip
description points to CCME1
destination-pattern 28..
session target ipv4:172.24.22.38
!
dial-peer voice 1003 voip
description points to CCME3
destination-pattern 9...
session target ipv4:192.168.1.29
!

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Configuration Examples for Call Transfer and Forwarding

dial-peer voice 1004 voip

description points to CCME2
destination-pattern 29..
session target ipv4:172.24.22.42
!
line con 0
exec-timeout 0 0
logging synchronous
line aux 0
line vty 0 4
password pswd
!
end

Forwarding Calls to Cisco Unity Express: Example

The following example enables the ability to forward calls that originate from
Cisco Unified Communications Manager phones and are routed through a Cisco Unified CME system to
a Cisco Unity Express extension. Call forwarding is enabled for all calling parties, H.450.3 is disabled,
and connections are allowed to SIP endpoints.
telephony-service
call-forward pattern .T

voice service voip

no supplementary-service h450.3
allow connections from h323 to sip

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Configuring Call Forward Unregistered for SIP IP Phones: Example

The following example shows CFU configured for voice register dn 20:
!
!
!
voice service voip
allow-connections sip to sip
sip
registrar server expires max 250 min 75
!
!
voice register global
mode cme
source-address 10.100.109.10 port 5060
bandwidth video tias-modifier 256 negotiate end-to-end
max-dn 200
max-pool 42
url directory http://1.4.212.11/localdirectory
create profile sync 0004625832149157
!
voice register dn 20
number 10
call-forward b2bua unregistered 2345
!
voice register pool 1
number 1 dn 20
id mac 1111.1111.1111
camera
video
!
voice register pool 2
id mac 0009.A3D4.1234

Configuring Keepalive Timer Expiration in SIP Phones: Example

The following example shows the minimum and maximum registrar server expiration time for SIP
phones:

Router#show run
!
!
!
!
!
!
voice service voip
allow-connections sip to sip
sip
registrar server expires max 250 min 75
!
!
voice register global
mode cme
source-address 10.100.109.10 port 5060
bandwidth video tias-modifier 256 negotiate end-to-end
max-dn 200

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Where to Go Next

Where to Go Next
If you are finished modifying the configuration, generate a new configuration file and restart the phones.
See “Generating Configuration Files for Phones” on page 353.

Soft Keys
To block the function of the call-forward-all or transfer soft key without removing the key display or to
remove the soft key from one or more phones, see the “How to Customize Soft Keys” section on
page 1339.

Feature Access Codes (FACs)

Phone users can activate and deactivate a phone’s call-forward-all setting by using a feature access code
(FAC) instead of a soft key on the phone if standard or custom FACs have been enabled for your system.
The following are the standard FACs for call forward all:
• callfwd all—Call forward all calls. Standard FAC is **1 plus an optional target extension.
• callfwd cancel—Cancel call forward all calls. Standard FAC is **2.
For more information about FACs, see “Configuring Feature Access Codes” on page 1131.

Night Service
Calls can be automatically forwarded during night service hours, but you must define the night-service
periods, which are the dates or days and hours during which night service will be active. For instance,
you may want to designate night service periods that include every weeknight between 5 p.m. and 8 a.m.
and all day every Saturday and Sunday. For more information, see “Configuring Call Coverage
Features” on page 837.

Additional References
The following sections provide references related to Cisco Unified CME features.

Related Documents
Related Topic Document Title
Cisco Unified CME configuration • Cisco Unified CME Command Reference
• Cisco Unified CME Documentation Roadmap
Cisco IOS commands • Cisco IOS Voice Command Reference
• Cisco IOS Software Releases 12.4T Command References
Cisco IOS configuration • Cisco IOS Voice Configuration Library
• Cisco IOS Software Releases 12.4T Configuration Guides
Phone documentation for Cisco Unified CME • User Documentation for Cisco Unified IP Phones

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Additional References

Technical Assistance
Description Link
The Cisco Support website provides extensive online http://www.cisco.com/techsupport
resources, including documentation and tools for
troubleshooting and resolving technical issues with
Cisco products and technologies.
To receive security and technical information about
your products, you can subscribe to various services,
such as the Product Alert Tool (accessed from Field
Notices), the Cisco Technical Services Newsletter, and
Really Simple Syndication (RSS) Feeds.
Access to most tools on the Cisco Support website
requires a Cisco.com user ID and password.

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Feature Information for Call Transfer and Forwarding

Feature Information for Call Transfer and Forwarding

Table 69 lists the features in this module and enhancements to the features by version.
To determine the correct Cisco IOS release to support a specific Cisco Unified CME version, see the
Cisco Unified CME and Cisco IOS Software Version Compatibility Matrix at
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/requirements/guide/33matrix.htm.
Use Cisco Feature Navigator to find information about platform support and software image support.
Cisco Feature Navigator enables you to determine which Cisco IOS software images support a specific
software release, feature set, or platform. To access Cisco Feature Navigator, go to
http://www.cisco.com/go/cfn. An account on Cisco.com is not required.

Note Table 69 lists the Cisco Unified CME version that introduced support for a given feature. Unless noted
otherwise, subsequent versions of Cisco Unified CME software also support that feature.

Table 69 Feature Information for Call Transfer and Forwarding

Cisco Unified CME

Feature Name Version Feature Information
Trunk-to-Trunk Transfer Blocking for 9.5 Introduced support Trunk-to-Trunk Transfer Blocking for
Toll Fraud Prevention on Cisco Unified Toll Fraud Prevention on Cisco Unified SIP IP Phones.
SIP IP Phones
Call Forwarding 4.1 • Call Forward All synchronization between
Cisco Unified CME and SIP phones was added.
• Disabling SIP supplementary services for call forward
and call transfer was added.
4.0 • Automatic call forwarding during night service was
introduced.
• Selective call forwarding was introduced.
• Forwarding of local (internal) calls can be blocked.
• H.450.7 standards support and QSIG supplementary
services capability was introduced.
3.4 Calls into a SIP device can be forwarded to other SIP or
SCCP devices including Cisco Unity, third- party voice
mail systems, or an auto-attendant (AA) or other interactive
voice response (IVR) devices. SCCP devices may also be
forwarded to SIP devices.
3.1 • Number of digits that can be entered using the
CfwdALL (call-forward all) soft key can be limited.
• H.450.12 standards support, which provide dynamic
detection of H.450.2 and H.450.3 capabilities on a
call-by-call basis, was introduced.

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Table 69 Feature Information for Call Transfer and Forwarding

Cisco Unified CME

Feature Name Version Feature Information
3.0 • CFwdALL soft key was introduced.
• Local hairpin call routing was supported as an option
for networks that cannot support H.450 call transfer and
forwarding. This feature requires installation of the Tcl
script app_h450_transfer.2.0.0.8.tcl or a later version.
2.1 Call forwarding using the H.450.3 standard was introduced.
Call Forwarding 1.0 Call forwarding for all calls, busy conditions, and
no-answer conditions was introduced, using a
Cisco-proprietary method.
Call Forward Unregistered 8.6 The Call Forward Unregistered (CFU) feature was
introduced for SIP phones.
Call Transfer 4.3 • Call-Transfer Recall was added.
• Consultative Call Transfer digit-collection process was
modified.
4.1 • Disabling SIP supplementary services for call transfer
and call forward was added.
4.0 • Default for the transfer-system command was changed
from the blind keyword to the full-consult keyword.
• Transfers to phones outside the Cisco Unified CME
system can be blocked for individual ephones.
• Number of digits in transfer destination numbers can be
limited.
3.4 Support for attended and blind transfer s using SIP IP phone
directly connected to Cisco CME.
3.2 • Consultative transfer to monitored lines using direct
station select was introduced.
• Transcoding between G.711 and G.729 is supported
when one leg of a Voice over IP (VoIP)-to-VoIP hairpin
call uses G.711 and the other leg uses G.729.

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Table 69 Feature Information for Call Transfer and Forwarding

Cisco Unified CME

Feature Name Version Feature Information
3.1 Support was introduced for the following:
• Enhancements for VoIP networks which contain a mix
of platforms that support H.450.2 and H.450.3
standards, such as Cisco CME 3.1, Cisco CME 3.0,
Cisco ITS V2.1, and platforms that do not support
H.450.2 and H.450.3 standards, such as
Cisco Unified Communications Manager, Cisco BTS
Softswitch (BTS), and Cisco PSTN Gateway (PGW).
• H.450.12 standards, which provide dynamic detection
of H.450.2 and H.450.3 capabilities on a call-by-call
basis.
• Automatic detection of Cisco Unified Communications
Manager endpoints.
• Hairpin VoIP-to-VoIP call routing and routing to an
H.450 tandem gateway.
• Hairpin call routing does not require a Tcl script.
3.0 Local hairpin call routing was supported as an option for
networks that cannot support H.450 call transfer and
forwarding. This feature requires installation of the Tcl
script app_h450_transfer.2.0.0.8.tcl or a later version.
2.1 Consultative transfer using the ITU-T H.450.2 standard was
introduced.
1.0 Call transfer was introduced, using a Cisco proprietary
method.

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