Trunking Support

Revised: June 19, 2006

Trunking support features provide or enhance different types of trunk services. This chapter describes
the following topics:
• Direct FXO Trunk Lines, page 264
• QSIG Supplementary Services, page 272
• SIP Trunk Features, page 277

Note For more information about Cisco IOS voice features, see the entire Cisco IOS Voice Configuration
Library—including library preface and glossary, feature documents, and troubleshooting
information—at
http://www.cisco.com/en/US/products/ps6441/prod_configuration_guide09186a0080565f8a.html.

For publications related to Cisco Unified CallManager Express (Cisco Unified CME), see the
Cisco Unified CME documentation at
http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_documentation_roadmap09186a0
080189132.html.

Trunking Support Overview

Trunking support features provide or enhance trunk services from Cisco Unified CME to other
call-control devices in the PSTN or VoIP network. Table 27 summarizes trunking support features.

Table 27 Trunking Support Features

Feature Description Benefit Example

Direct FXO Trunk Lines System creates a private-line Phone user makes and A sales manager has a direct
automatic ringdown receives calls without going FXO trunk line with a number
off-premise extension for through Cisco Unified CME that is local to most company
direct connection to a PSTN and has a number provided by clients, and has another line
central office. the PSTN. on the phone that is a
Cisco Unified CME
extension, which is used for
in-house calls.

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Table 27 Trunking Support Features (continued)

Feature Description Benefit Example

QSIG Supplementary System enables H.450 You can interface Phone users have access to
Services supplementary services for Cisco Unified CME system to PBX applications.
QSIG interworking. PBX or PBX voice-mail
system.
SIP Trunk Features System enables functionality You can interface Phone users can make calls
that is necessary to interwork Cisco Unified CME system to and use some features across
with SIP networks. SIP networks. SIP networks.

Direct FXO Trunk Lines

Direct FXO trunk lines provide phone users direct access to a PSTN central office line. This section
describes the following topics:
• Direct FXO Trunk Lines Overview, page 264
• Configuring Direct FXO Trunk Lines, page 266
• Examples, page 270
• Feature History for Direct FXO Trunk Lines, page 271

Direct FXO Trunk Lines Overview

Note For a summary of the functionality introduced in different versions, see the “Feature History for Direct
FXO Trunk Lines” section on page 271.

For Cisco CME 3.2 and later versions, IP phones can be configured to have buttons for dedicated FXO
trunk lines, herein after referred to as FXO lines. Direct FXO trunk lines may used by companies whose
employees require private PSTN numbers. For example, a salesperson may need a special number that
customers can call without having to go through a main number. When a call comes in to the direct
number, the salesperson knows that the caller is a customer. In the salesperson’s absence the customer
can leave voice mail. Dedicated lines can use PSTN service provider voice mail: when the line button is
pressed, the PSTN-FXO line is seized, allowing the user to hear the stutter dial tone provided by the
PSTN to indicate that voice messages are available.
Because dedicated FXO lines behave as private lines, users do not have to dial a prefix, such as 9 or 8,
to reach an outside line. To reach phone users within the company, FXO-line users must dial numbers
that use the company's PSTN number. For calls to nonPSTN destinations, such as local IP phones, a
second ephone-dn must be provisioned.
Calls placed to or received on an FXO line have restricted Cisco Unified CME services (see the
“Restrictions” section on page 266) and cannot be transferred by Cisco Unified CME. However, phone
users are able to access hookflash-controlled PSTN services using the Flash soft key. See the fxo
hook-flash command in the Cisco Unified CallManager Express Command Reference.

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Direct FXO Trunk Lines

From a high level, configuration of a direct FXO trunk line consists of the following:
1. Configuring the FXO port for a private line automatic ringdown (PLAR) off-premises extension
(OPX) connection and declaring a private line’s number; for example:
voice-port 1/1/0
connection plar-opx 1020

2. Configuring dial peers for FXO port and declaring a trunk tag to bind the FXO port and its dial peer
to an ephone-dn; for example:
dial-peer voice 111 pots
destination-pattern 82
port 1/1/0

3. Configuring the ephone-dn and ephone; for example:

ephone-dn 12
number 1020

ephone-dn 1
mac-address 1111.1111.1111
button 1:12

4. Binding the ephone-dn to the FXO port with the trunk command; for example:
ephone-dn 12
number 1020
trunk 82 timeout 30

FXO Trunk Enhancements in Cisco Unified CME 4.0

The following FXO trunk enhancements were introduced in Cisco Unified CME 4.0 to improve the
keyswitch emulation behavior of PSTN lines in a Cisco Unified CME system.
• FXO port monitoring—Allows the line button on IP phones to reliably show the status of an FXO
port when the port is in use. The status indicator, either a lamp or an icon, depending on the phone
model, accurately displays the status of the FXO port during the duration of the call, even after the
call is forwarded or transferred. The same FXO port can be monitored by multiple phones using
multiple trunk ephone-dns. Enable this feature for a trunk ephone-dn by using the trunk command
with the monitor-port keyword.
• Transfer recall—If a transfer-to phone does not answer after a specified timeout, the call is returned
to the phone that initiated the transfer and it resumes ringing on the FXO line button. The ephone-dn
must be dual-line and must have the huntstop channel command configured. Enable this feature for
a trunk ephone-dn by using the trunk command with the transfer-timeout keyword.
• Transfer-to button optimization—When an FXO call is transferred to a private extension button on
another phone, and that phone has a shared line button for the FXO port, after the transfer is
committed and the call is answered, the connected call displays on the FXO line button of the
transfer-to phone. This frees up the private extension line on the transfer-to phone. The ephone-dn
must be dual-line and it must have the huntstop channel command configured. Enable this feature
for a trunk ephone-dn by using the trunk command with the monitor-port keyword.
• Dual-line ephone-dns— Ephone-dns for FXO trunk lines can now be configured for dual-line to
support the FXO monitoring, transfer recall, and transfer-to button optimization features.

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Restrictions
• An ephone-dn with a trunk line cannot be configured for call forward, busy, or no answer.
• Numbers entered after a trunk line is seized will not be displayed. Only the trunk tag is displayed
on IP phones.
• Numbers entered after trunk line is seized will not appear in call history or call detail records
(CDRs) of a Cisco Unified CME router. Only the trunk tag is logged for calls made from trunk lines.
• FXO trunk lines do not support the CFwdALL, Transfer, Pickup, GPickUp, Park, CallBack, and
NewCall soft keys.
• FXO trunk lines do not support conference initiator dropoff.
• FXO trunk lines do not support on-hook redial. The phone user must explicitly select the FXO trunk
line before pressing the Redial button.
• FXO trunk lines do not support call transfer to IP phones. However, the call initiator can conference
an FXO line with an IP phone by pressing the Hold button, which leaves the FXO trunk line and IP
phone connected. The conference initiator is unable to participate in the conference, but can place
calls on other lines.
• FXO port monitoring has the following restrictions:
– Requires Cisco Unified CME 4.0 or later.
– Supported only for analog FXO loop-start and ground-start ports and T1/E1 FXO CAS ports.
FXS loop-start and ground-start ports and PRI/BRI PSTN trunks are not supported.
– Not supported for analog ports on the Cisco VG 224 or Cisco ATA 180 Series.
– T1 CAS DS0 group must be configured per time slot (cannot bundle more than one time slot
into a ds0-group).
• Transfer recall and transfer-to button optimization are supported on dual-line ephone-dns only in
Cisco Unified CME 4.0 and later.
• Transfer-to button optimization is not supported for call forwarding, call-park recall, call pickup on
hold, or call pickup at alert.
• Transfer recall is not supported for analog ports on the Cisco VG 224 or Cisco ATA 180 Series.

Configuring Direct FXO Trunk Lines

This procedure sets up a direct FXO trunk line on an IP phone.

SUMMARY STEPS

1. enable
2. configure terminal
3. voice-port port
4. connection plar opx digits
5. exit
6. dial-peer voice tag pots
7. destination-pattern string[T]
8. port port

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9. exit
10. ephone-dn dn-tag [dual-line]
11. number number
12. trunk digit-string [timeout seconds] [transfer-timeout seconds] [monitor-port port]
13. huntstop channel
14. exit
15. ephone phone-tag
16. mac-address tag
17. button button-number{separator}dn-tag [[button-number{separator}dn-tag] ...]
18. 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-port port Enters voice-port configuration mode.
• Port argument is platform-dependent; type ? to display
Example: syntax. For more information, see the Cisco IOS Voice
Router(config)# voice-port 0/0/0 Command Reference.
Step 4 connection plar opx digits Specifies a PLAR OPX connection.
• Using this option, the local voice port provides a local
Example: response before the remote voice port receives an
Router(config-voice-port)# connection plar opx answer. On FXO interfaces, the voice port does not
5550111 answer until the remote side has answered.
• digits—Specifies the destination telephone number.
Valid entries are any series of digits that specify the
E.164 telephone number.
Step 5 exit Exits voice-port configuration mode.

Example:
Router(config-voice-port)# exit
Step 6 dial-peer voice tag pots Enters dial-peer configuration mode for POTS.
• tag—Digits that define a particular dial peer. Range is
Example: 1 to 2147483647.
Router(config)# dial-peer voice 53 pots

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Command or Action Purpose

Step 7 destination-pattern [+]string[T] Defines a prefix, access code, or full E.164 telephone number
(depending on your dial plan) to be used for a dial peer.
Example:
Router(config-dial-peer)# destination-pattern 20
Step 8 port port Associates a dial peer with a specific voice port.
• Port argument is platform-dependent; type ? to display
Example: syntax. For more information, see the Cisco IOS Voice
Router(config-dial-peer)# port 0/0/0 Command Reference.
• Use the PLAR connection voice port configured by the
connection command.
Step 9 exit Exits dial-peer configuration mode.

Example:
Router(config-ephone-template)# exit
Step 10 ephone-dn dn-tag [dual-line] Enters ephone-dn configuration mode to create an extension
(ephone-dn) for a Cisco Unified IP phone line, an intercom
line, a paging line, a voice-mail port, or an MWI.
Example:
Router(config)# ephone-dn 1 dual-line • dn-tag—Unique sequence number that identifies an
ephone-dn during configuration tasks. Range is 1 to the
maximum number of ephone-dns allowed on the router
platform. Type ? to display range.
• dual-line—Enables dual-line mode for the ephone-dn.
Step 11 number number Associates a telephone or extension number with an
extension (ephone-dn).
Example: • number—String of up to 16 characters that represents an
Router(config-ephone-dn)# number 5550111 E.164 telephone number. Enter the PLAR number
configured by the connection command.

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Command or Action Purpose

Step 12 trunk digit-string [timeout seconds] Associates an ephone-dn with an FXO port’s trunk number
[transfer-timeout seconds] [monitor-port port] so the ephone-dn can support a direct FXO line.
• digit-string—Declares the number of the trunk line. Use
Example: the string argument specified in the destination-pattern
Router(config-ephone-dn)# trunk 20 timeout 30 command.
transfer-timeout 60 monitor-port 0/0/0:1
• timeout seconds—Sets the timeout between dialed
digits, in seconds. The phone user must either enter the
pound (#) key or wait for this interdigit timeout to
complete digit collection. Range is 3 to 30. Default is 3.
• transfer-timeout seconds—Enables a transferred or
forwarded call to be automatically recalled if the transfer
target does not answer after the specified number of
seconds. The call is withdrawn from the transfer-to
phone and the call resumes ringing on the phone that
initiated the transfer. Supported for dual-line ephone-dns
only. Range is 5 to 60000. Default is disabled.
• monitor-port port— Enables direct status monitoring of
the FXO port on the line button of the IP phone. The line
button indicator, either a lamp or an icon depending on
the phone, shows the in-use status of the FXO port
during the duration of the call, regardless of whether
after the call is forwarded or transferred. The same FXO
port can be monitored by multiple phones using multiple
trunk ephone-dns.
Note The monitor-port and transfer-timeout keywords
are not supported on ephone-dns for analog ports on
the Cisco VG 224 or Cisco ATA 180 Series.
Step 13 huntstop channel For dual-line ephone-dns, keeps incoming calls from hunting
to the second channel if the first channel is busy or does not
answer.
Example:
Router(config-ephone-dn)# huntstop channel • Required when using the monitor-port or
transfer-timeout keywords with the trunk command in
Step 12.
Step 14 end Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# end
Step 15 ephone phone-tag Enters ephone configuration mode for an IP phone.

Example:
Router(config)# ephone 1
Step 16 mac-address mac-address Associates the MAC address of a Cisco Unified IP phone
with an ephone configuration in a Cisco Unified CME
system.
Example:
Router(config-ephone)# mac-address • mac-address—The MAC address of an IP phone, which
CFBA.321B.96FA is found on a label located on the bottom of the phone.

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Command or Action Purpose

Step 17 button button-number{separator}dn-tag Associates ephone-dns with individual buttons on
[[button-number{separator}dn-tag] ...] Cisco Unified IP phones and specifies ring behavior per
button.
Example: • button-number—Number of a line button on a
Router(config-ephone)# button 1:1 Cisco Unified IP phone to be associated with an
extension (ephone-dn). The maximum number of
button-ephone-dn pairs is determined by phone type.
• separator—Single character that denotes the
characteristics to be associated with this button and
extension or extensions. For a list of options, see the
button command description in the
Cisco Unified CallManager Express Command
Reference.
Note When o is used for the separator, the dn-tag
argument can contain up to ten individual DN tags,
separated by commas.

• dn-tag—Ephone-dn tag previously defined using the

ephone-dn command.
Step 18 end Returns to privileged EXEC mode.

Example:
Router(config-ephone)# end

Examples
The following example shows a configuration for the trunk monitoring feature.
!
controller T1 0/0/0
ds0-group 1 timeslots 1 type fxo-ground-start
ds0-group 2 timeslots 2 type fxo-ground-start
!
voice-port 0/0/0:1
connection plar opx 1001
!
voice-port 0/0/0:2
connection plar opx 1002
!
dial-peer voice 801 pots
destination-pattern 801
port 0/0/0:1
!
dial-peer voice 802 pots
destination-pattern 802
port 0/0/0:2
!
ephone-dn 1 dual-line
number 1001
huntstop channel
trunk 801 transfer-timeout 10 monitor-port 0/0/0:1
!
ephone-dn 2 dual-line

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number 1002
huntstop channel
trunk 802 transfer-timeout 10 monitor-port 0/0/0:2
!
ephone 1
mac-address 0001.0002.0003
type 7960
button 1:10 2:1 3:2
!
ephone 2
mac-address 0001.0002.0004
type 7960 addon 1 7914
button 1:11 7:1 8:2

The following example shows the configuration for one phone that has 2 buttons: the first button is for
making calls to local extensions and for receiving calls, and the second button is for a private line that
goes out an FXO port as a direct trunk.
voice-port 1/0/0
connection plar opx 1001

dial-peer voice 100 pots

destination-pattern 81
voice-port 1/0/0

ephone-dn 1
name MainExtension
number 1001

ephone-dn 2
name PrivateTrunkLine
trunk 81 timeout 5

ephone 1
mac-address 1111.1111.1110
button 1:1 2:2

Feature History for Direct FXO Trunk Lines

Cisco Unified CME

Version Modification
3.2 Direct FXO trunk line capability was introduced.
4.0 Support for dual lines was added and the transfer-timeout and monitor-port
keywords were added to the trunk command.

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QSIG Supplementary Services

QSIG Supplementary Services

QSIG supplementary services allow Cisco Unified CME phones to use QSIG to interwork with PBX
phones in a seamless fashion. This section describes the following topics:
• QSIG Supplementary Services Overview, page 272
• Configuring QSIG Supplementary Services, page 273
• Examples, page 283
• Feature History for QSIG Supplementary Services, page 277

QSIG Supplementary Services Overview

Note For a summary of the functionality introduced in different versions, see the “QSIG Supplementary
Services Overview” section on page 272.

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 29 illustrates a topology for a Cisco Unified CME
system with some phones under the control of a PBX.

Figure 29 Cisco Unified CME System with PBX

IP 1001 IP 2001

IP 1002 IP 2002

IP 1003 IP 2003

mote Cisco CME IP network Local Cisco CME

QSIG 3001

3002
PBX
3003

Message
135562

center

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.

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• 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.

Configuring QSIG Supplementary Services

This procedure enables QSIG supplementary services.

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. voicemail phone-number
5. transfer-system {blind | full-blind | full-consult | local-consult}
6. exit
7. ephone-dn dn-tag
8. mwi qsig
9. Configure call forwarding to the voice-mail number for this ephone-dn.
10. exit
11. voice service voip
or
dial-peer voice tag voip
12. supplementary-service h450.7
13. qsig decode

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14. exit
15. voice service pots
or
dial-peer voice tag pots
16. supplementary-service qsig call-forward
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 telephony-service Enters telephony-service configuration mode.

Example:
Router(config)# telephony-service
Step 4 voicemail phone-number Specifies a directory number for voice mail. The PBX
message center number can be entered here.
Example: Note Ensure to configure a POTS dial peer and an ISDN
Router(config-telephony)# voicemail 74398 interface for the message center line.
Step 5 transfer-system {blind | full-blind | Defines the call transfer method to allow call transfer with
full-consult | local-consult} consultation for all lines served by the router.
Note For SIP networks, use only the full-blind keyword
Example: or the full-consult keyword.
Router(config-telephony)# transfer-system
full-consult • blind—Calls are transferred without consultation with
a single phone line using the Cisco-proprietary method.
• 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. The calls fall back to full-blind if the
second line is unavailable.
• 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.

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Command or Action Purpose

Step 6 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit
Step 7 ephone-dn dn-tag Enters ephone-dn configuration mode.
• dn-tag—Unique sequence number that identifies this
Example: ephone-dn during configuration tasks.
Router(config)# ephone-dn 25
Step 8 mwi qsig Specifies that the QSIG (PBX) message center should be
interrogated for MWI status for this ephone-dn.
Example:
Router(config-ephone-dn)# mwi qsig
Step 9 Configure call forwarding to the voice-mail number Configure one or more of the following commands with the
for this ephone-dn. the QSIG (PBX) message center as the call-forwarding
target number. If more than one type of call forwarding is
enabled, the preference order for evaluating the different
types is as follows:
1. call-forward night-service target-number
2. call-forward all target-number
3. call-forward busy target-number [primary |
secondary] [dialplan-pattern]
and
call-forward noan target-number timeout seconds
[primary | secondary] [dialplan-pattern]
Step 10 exit Exits ephone-dn configuration mode.

Example:
Router(config-ephone-dn)# exit
Step 11 voice service voip Enters VoIP voice-service configuration mode to define
or global call transfer and forwarding parameters.
dial-peer voice tag voip Enters dial-peer configuration mode to define parameters
for an individual dial peer.
Example:
Router(config)# voice service voip

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

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Command or Action Purpose

Step 12 supplementary-service h450.7 Enables H.450.7 supplementary services capabilities
exchange. This command is disabled by default.
Example: Note Use this command in voice-service configuration
Router(config-voi-serv)# supplementary-service mode to enable H.450.7 supplementary services
h450.7 globally, or use it in dial-peer configuration mode to
enable the services on a single dial peer.
Example:
Router(config-dial-peer)# supplementary-service
h450.7
Step 13 qsig decode Enables decoding for QSIG supplementary services.

Example:
Router(config-voi-serv)# qsig decode
Step 14 exit Exits VoIP voice-service configuration mode.

Example:
Router(config-voi-serv)# exit
Step 15 voice service pots Enters POTS voice-service configuration mode to define
or global call transfer and forwarding parameters.
dial-peer voice tag pots Enters dial-peer configuration mode to define parameters
for an individual dial peer.
Example:
Router(config)# voice service pots

Example:
Router(config)# dial-peer voice 2 pots
Step 16 supplementary-service qsig call-forward Enables QSIG call-forwarding supplementary services
(ISO 13873) to forward calls to another number. This
command is disabled by default.
Example:
Router(config-voi-serv)# supplementary-service Note Use this command in voice-service configuration
qsig call-forward mode to enable QSIG call-forwarding services
globally, or use it in dial-peer configuration mode to
Example: enable the services on a single dial peer.
Router(config-dial-peer)# supplementary-service
qsig call-forward
Step 17 end Returns to privileged EXEC mode.

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

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Examples
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

Feature History for QSIG Supplementary Services

Cisco Unified CME

Version Modification
4.0 QSIG supplementary services capability was introduced.

SIP Trunk Features

SIP trunk features enable interoperability between Cisco Unified CME phones and SIP networks. This
section describes the following topics:
• SIP Trunk Features Overview, page 278
• Configuring SIP Trunk Support, page 279
• Examples, page 283
• Verifying SIP Trunk Support Features, page 285
• Feature History for SIP Trunk Features, page 285
• Related Features, page 286

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SIP Trunk Features Overview

Note For a summary of the functionality introduced in different versions, see the “SIP Trunk Features
Overview” section on page 278.

When Cisco Unified CME and SCCP phones are used with SIP networks, the following features may
need special settings:
• Call Forwarding over SIP Networks, page 278
• Call Transfer over SIP Networks, page 278
• DTMF Relay, page 278
• SIP Register Support, page 279

Call Forwarding over SIP Networks

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, use the call-forward pattern command and specify a pattern that matches the calling-party
numbers of the calls that you want to be able to forward. Use the call-forward pattern command with
the .T pattern to allow all calls for all possible SIP calling parties to be forwarded using the SIP 302
response.

Call Transfer over SIP Networks

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. Note that the transfer-system command must be configured
with the full-blind or full-consult keyword for SIP Refer to be invoked.

DTMF Relay
SCCP phones used with Cisco Unified CME systems relay dual tone multifrequency (DTMF) digits out
of band. To interwork with SIP applications that expect in-band DTMF digits, you must enable a
conversion. Two types of conversions are possible:
• RFC 2833 (Standard) DTMF Relay—For remote SIP-based IVR or voice-mail application
• SIP Notify (Nonstandard) DTMF Relay—For Cisco Unity Express on a SIP network

RFC 2833 (Standard) DTMF Relay

To use remote voice-mail or interactive voice response (IVR) applications on SIP networks from
Cisco Unified CME phones, you must enable conversion of the out-of-band dual tone multifrequency
(DTMF) digits used by the Cisco Unified CME phones to the RFC 2833 in-band DTMF relay
mechanism used by SIP phones. You select this method in the SIP VoIP dial peer using the dtmf-relay
rtp-nte command.
The SIP DTMF relay method is needed in the following situations:
• When SIP is used to connect a Cisco Unified CME system to a remote SIP-based IVR or voice-mail
application.

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• When SIP is used to connect a Cisco Unified CME system to a remote SIP-PSTN voice gateway that
goes through the PSTN to a voice-mail or IVR application.
Note that the need to use out-of-band DTMF relay conversion is limited to SCCP phones. SIP phones
natively support in-band DTMF relay as specified in RFC 2833.
To enable SIP DTMF relay using RFC 2833, the commands in this section must be used on both
originating and terminating gateways.

SIP Notify (Nonstandard) DTMF Relay

To use voice mail on a SIP network that connects to a Cisco Unity Express system, use a nonstandard
SIP Notify format. To configure the Notify format, use the sip-notify keyword with the dtmf-relay
command. Using the keyword sip-notify may be required for backward compatibility with
Cisco CME 3.0 and 3.1.

SIP Register Support

SIP register support enables a SIP gateway to register E.164 numbers with a SIP proxy or SIP registrar,
similar to the way that H.323 gateways can register E.164 numbers with a gatekeeper. SIP gateways
allow registration of E.164 numbers to a SIP proxy or registrar on behalf of analog telephone voice ports
(FXS) and IP phone virtual voice ports (EFXS) for local SCCP phones. This support is enabled using
the register command in SIP UA configuration mode.
When registering E.164 numbers in dial peers with an external registrar, you can also register them with
a secondary SIP proxy or registrar to provide redundancy. The secondary registration can be used if the
primary registrar fails.
For more detailed information, see SIP Gateway Enhancements, Cisco IOS Release 12.2(15)ZJ.

Note No commands allow registration between the H.323 and SIP protocols.

By default, SIP gateways do not generate SIP Register messages, so the following procedure is needed
to set up the gateway to register the gateway’s E.164 telephone numbers with an external SIP registrar.

Configuring SIP Trunk Support

This procedure enables four SIP trunk support parameters:
• Call forwarding over SIP networks—call-forward pattern and calling-number local commands
• Call transfer over SIP networks—transfer-system and transfer-pattern commands
• DTMF relay—dtmf-relay rtp-nte or dtmf-relay sip-notify command and notify telephone-event
max-duration command
• SIP registrar—registrar, retry, and timers commands

SUMMARY STEPS

1. enable
2. configure terminal
3. telephony-service
4. call-forward pattern pattern

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5. calling-number local
6. transfer-system {full-blind | full-consult}
7. transfer-pattern transfer-pattern
8. exit
9. dial-peer voice tag voip
10. dtmf-relay rtp-nte
11. dtmf-relay sip-notify
12. exit
13. sip-ua
14. notify telephone-event max-duration time
15. registrar {dns:host-name | ipv4:ip-address} expires seconds [tcp] [secondary]
16. retry register number
17. timers register time
18. 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 call-forward pattern pattern Specifies the H.450.3 standard or SIP 302 redirection
method for call forwarding. Calling-party numbers that do
not match the patterns defined with this command are
Example:
Router(config-telephony)# call-forward pattern
forwarded using Cisco-proprietary call forwarding for
4... backward compatibility.
• pattern—Digits to match for call forwarding using the
H.450.3 standard or SIP 302 redirection method. A
pattern of .T matches all calling-party numbers.
Note When defining forwards 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 on
page 120.
Step 5 calling-number local (Optional) Replaces a calling-party number and name with
the forwarding-party (local) number and name.
Example:
Router(config-telephony)# calling-number local
Step 6 transfer-system {full-blind | full-consult} Defines the call transfer method for all lines served by the
router.
Example: Note For SIP networks, use only the full-blind keyword
Router(config-telephony)# transfer-system or the full-consult keyword. For more information,
full-consult see the Cisco IOS SIP Configuration Guide.

• 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. The calls fall back to full-blind if the
second line is unavailable.

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Command or Action Purpose

Step 7 transfer-pattern transfer-pattern 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
to other local IP phones.
52540.. • transfer-pattern—String of digits for permitted call
transfers. Wildcards are allowed.
Note When defining transfers to nonlocal numbers, it is
important to note that transfer-pattern digit
matching is performed before translation-rule
operations. Therefore, you should specify in this
command the digits that are actually entered by
phone users before they are translated. For more
information, see the “Voice Translation Rules and
Profiles” section on page 120.
Step 8 exit Exits telephony-service configuration mode.

Example:
Router(config-telephony)# exit
Step 9 dial-peer voice tag voip Enters dial-peer configuration mode.

Example:
Router(config)# dial-peer voice 2 voip
Step 10 dtmf-relay rtp-nte Forwards DTMF tones by using Real-Time Transport
Protocol (RTP) with the Named Telephone Event (NTE)
payload type. This enables DTMF relay using the
Example:
Router(config-dial-peer)# dtmf-relay rtp-nte
RFC 2833 standard method.
Step 11 dtmf-relay sip-notify Forwards DTMF tones using SIP NOTIFY messages.

Example:
Router(config-dial-peer)# dtmf-relay sip-notify
Step 12 exit Exits dial-peer configuration mode.

Example:
Router(config-dial-peer)# exit
Step 13 sip-ua Enters SIP user-agent configuration mode.

Example:
Router(config)# sip-ua
Step 14 notify telephone-event max-duration time Configures the maximum time interval allowed between
two consecutive NOTIFY messages for a single DTMF
event.
Example:
Router(config-sip-ua)# notify telephone-event • max-duration time—Time interval between
max-duration 2000 consecutive NOTIFY messages for a single DTMF
event, in milliseconds. Range is 500 to 3000. Default
is 2000.

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Command or Action Purpose

Step 15 registrar {dns:host-name | ipv4:ip-address} Registers E.164 numbers on behalf of analog telephone
expires seconds [tcp] [secondary] voice ports (FXS) and IP phone virtual voice ports (EFXS)
with an external SIP proxy or SIP registrar server.
Example: • dns:host-name—Domain name server that resolves
Router(config-sip-ua)# registrar the name of the dial peer to receive calls.
ipv4:10.8.17.40 expires 3600 secondary
• ipv4:ip-address—IP address of the dial peer to receive
calls.
• expires seconds—Default registration time, in
seconds.
• tcp—(Optional) Sets the transport layer protocol to
TCP. UDP is the default.
• secondary—(Optional) Specifies registration with a
secondary SIP proxy or registrar for redundancy
purposes.
Step 16 retry register number Sets the total number of SIP Register messages that the
gateway should send.
Example: • number—Number of Register message retries. Range
Router(config-sip-ua)# retry register 10 is 1 to 10. Default is 10.
Step 17 timers register time Sets how long the SIP user agent (UA) waits before sending
Register requests.
Example: • time—Waiting time, in milliseconds. Range is
Router(config-sip-ua)# timers register 500 100 to 1000. Default is 500.
Step 18 end Returns to privileged EXEC mode.

Example:
Router(config-sip-ua)# end

Examples
This section contains the following examples:
• Call Forwarding over SIP Networks: Example, page 283
• Call Transfer over SIP Networks: Example, page 284
• DTMF Relay using RFC 2833: Example, page 284
• DTMF Relay using SIP Notify: Example, page 284
• SIP Register Support: Example, page 284

Call Forwarding over SIP Networks: Example

The following example enables call forwarding using the H.450.3 standard or SIP 302 response:
dial-peer voice 100 pots
destination-pattern 9.T
port 1/0/0
!
dial-peer voice 4000 voip
destination-pattern 4...

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session protocol sipv2

session-target ipv4:10.1.1.1
!
telephony-service
call-forward pattern 4...

Call Transfer over SIP Networks: Example

The following example specifies transfer with consultation using the H.450.2 standard for all IP phones
serviced by the router:
!
dial-peer voice 100 pots
destination-pattern 9.T
port 1/0/0
!
dial-peer voice 4000 voip
destination-pattern 4...
session protocol sipv2
session-target ipv4:10.1.1.1
!
telephony-service
transfer-pattern 4...
transfer-system full-consult

DTMF Relay using RFC 2833: Example

The following example specifies use of the RFC 2833 method for in-band DTMF relay for calls using
dial peer 2.
dial-peer voice 2 voip
dtmf-relay rtp-nte

sip-ua
notify telephone-event max-duration 2000

DTMF Relay using SIP Notify: Example

The following example specifies use of the SIP notify method for in-band DTMF relay for calls using
dial peer 4.
dial-peer voice 4 voip
dtmf-relay sip-notify

sip-ua
notify telephone-event max-duration 2000

SIP Register Support: Example

The following example sets up the gateway to register the gateway’s E.164 telephone numbers with an
external SIP registrar.
sip-ua
registrar ipv4:10.8.17.40 expires 3600 secondary
retry register 10
timers register 500

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Verifying SIP Trunk Support Features

For information about verifying commands, see the Cisco Unified CallManager Express Command
Reference.

Step 1 The show sip-ua status command output displays the time interval between consecutive NOTIFY
messages for a telephone event. In the following example, the time interval is 2000 ms.
Router# show sip-ua status

SIP User Agent Status

SIP User Agent for UDP :ENABLED
SIP User Agent for TCP :ENABLED
SIP User Agent bind status(signaling):DISABLED
SIP User Agent bind status(media):DISABLED
SIP early-media for 180 responses with SDP:ENABLED
SIP max-forwards :6
SIP DNS SRV version:2 (rfc 2782)
NAT Settings for the SIP-UA
Role in SDP:NONE
Check media source packets:DISABLED
Maximum duration for a telephone-event in NOTIFYs:2000 ms
SIP support for ISDN SUSPEND/RESUME:ENABLED
Redirection (3xx) message handling:ENABLED

SDP application configuration:

Version line (v=) required
Owner line (o=) required
Timespec line (t=) required
Media supported:audio image
Network types supported:IN
Address types supported:IP4
Transport types supported:RTP/AVP udptl

Step 2 Use the show sip-ua timers command to show the waiting time before Register requests are sent; that
is, the value that has been set with the timers register command.
Step 3 Use the show sip-ua register status command to show the status of local E.164 registrations.
Step 4 Use the show sip-ua statistics command to show the Register messages that have been sent.

Feature History for SIP Trunk Features

Cisco Unified CME

Version Modification
3.1 Support for SIP networks was introduced.
3.2 DTMF relay for SIP trunks was introduced.

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Related Features
Call Forwarding and Call Transfer on a SIP Network
After using call forwarding commands for Cisco Unified CME, you need to configure SIP call
forwarding and transfer, which are described in the “Configuring SIP Call Transfer” chapter of the
Cisco IOS SIP Configuration Guide.

Call Forwarding and Call Transfer for SCCP Phones

For a more complete discussion of Cisco Unified CME call forwarding and transfer methods, see the
“Transfer and Forwarding Support” section on page 221.
For information about configuring call forwarding and transfer for SCCP phones, see the “Call
Forwarding” section on page 372 and the “Call Transfer” section on page 466.

Call Forwarding and Call Transfer for SIP Phones

For information about configuring call forwarding and transfer for SIP phones, see the
Cisco CallManager Express 3.4 Configuration Guide.

Prefixes for SIP Unsolicited MWI Notify Messages

When SIP trunks are used to connect sites to a voice-mail server, a prefix can be added to extension
numbers in order to distinguish the extensions at different sites. For more information, see the “MWI
Prefix Specification for SIP Voice-Mail Applications” section on page 310.

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