SIP Configuration Guide, Cisco IOS Release 12.4T

Configuring SIP Support for Hookflash

This chapter contains information about the SIP Support for Hookflash feature that allows you to configure IP Centrex supplementary services on SIP-enabled, Foreign Exchange Station (FXS) lines. Supplementary services for the SIP Support for Hookflash feature include the following:

•Call hold

•Call waiting

•Call transfer

•3-Way conferencing

Use the service dsapp command to configure supplementary Centrex-like features on FXS phones to interwork with SIP-based softswitches. The SIP Support for Hookflash feature supports the concept of a dual-line (ACTIVE and STANDBY for active and held calls) for FXS calls to support supplementary services. Hookflash triggers supplementary services based on the current state of the call.

You can configure the service dsapp command on individual dial peers, or configure globally for all calls entering the gateway.

Feature History

for SIP Support for Hookflash

Contents

•Prerequisites for SIP Support for Hookflash

•Information About SIP Support for Hookflash

•How to Configure and Associate SIP Support for Hookflash

•Configuration Examples for SIP Support for Hookflash

•Additional References

Prerequisites for SIP Support for Hookflash

All Hookflash Features for FXS Ports

•Ensure that the gateway has voice functionality that is configurable for SIP.

•Establish a working IP network. For information on configuring IP, see the Cisco IOS IP Configuration Guide, Release 12.3.

•Configure VoIP.

Information About SIP Support for Hookflash

Use the service dsapp command to configure supplementary Centrex-like services on FXS phones to interwork with SIP-based softswitches. Hookflash triggers supplementary features based on the current state of the call and provides a simulation of dual-line capability for analog phones to allow one line to be active while the other line is used to control supplementary IP Centrex services. Supplementary services for the SIP Support for Hookflash feature include the following:

•Call Hold

•Call Waiting

•Call Transfers

•3-Way Conference

Call Hold

With the Call Hold feature, you can place a call on hold. When you are active with a call and you press hookflash, and there is no call that is waiting, you hear a dial tone.

If there is a call on hold, the hookflash switches between two calls; the call on hold becomes active while the active call is put on hold.

If you have a call on hold and the call hangs up, the call on hold is disconnected.

Call Holding Flows

The sequence of placing a call on hold is summarized in the following steps:

1. User A and user B are active with a call.

2. By pressing hookflash, user A initiates a call hold.

3. SIP sends a call hold indication to user B.

4. User A can now initiate another active call (user C), transfer the active call (call transfer), or respond to a call-waiting indication.

Note Use the offer call-hold command in sip-ua configuration mode to configure the method of hold used on the gateway. For detailed information on the offer call-hold command, see the Cisco IOS Voice Command Reference Guide.

5. User A receives a second dial tone and presses hookflash.

Figure 79 shows the initiation of the calls hold sequence.

Figure 79

Initiation of Call Hold

6. User A and User B reconnect.

Figure 80 shows the calls on hold resume sequence.

Figure 80

Resume Calls on Hold Flow

Table 48 summarizes the hookflash support for Call Hold services.

Call Waiting

With the Call Waiting feature, you can receive a second call while you are on the phone with another call. When you receive a second call, you hear a call-waiting tone (a tone with a 300 ms duration). Caller ID appears on phones that support caller ID. You can use hookflash to answer a waiting call and place the previously active call on hold. By using hookflash, you can toggle between the active and a call that is on hold. If the Call Waiting feature is disabled, and you hang up the current call, the second call will hear a busy tone.

The call-waiting sequence is summarizes in the following steps:

1. User A is active with a call to user B

2. User C calls user B

3. User B presses hookflash.

The call between user A and user B is held.

4. User B connects to user C.

Figure 81 shows the call waiting sequence.

Figure 81

Call-Waiting Sequence

Table 49 summarizes hookflash support for Call Waiting services.

Call Transfers

Call transfers are when active calls are put on hold while a second call is established between two users. After you establish the second call and terminate the active call, the call on hold will hear a ringback. The Call Transfer feature supports all three types of call transfers—blind, semi-attended, and attended.

Blind Call Transfer

The following describes a typical Blind call-transfer scenario:

1. User A calls user B.

2. User B (transferrer) presses hookflash, places user A (transferee) on hold, and dials user C (transfer-to).

Note User B will not hear alerting for the time you configure. See How to Configure Blind Transfer Wait Time.

3. Before the Blind call transfer trigger timer expires, user B disconnects, and the call between user A and user B is terminated.

4. User A is transferred to user C and hears a ringback if user C is available. If user C is busy, user A hears a busy tone; if user C is not busy and answers, user A and user C connect.

Figure 82 shows the call sequence for a Blind call transfer.

Figure 82

Blind Call Transfer Sequence

Semi-Attended Transfers

The following is a typical semi-attended transfer scenario:

1. User A calls user B.

2. User B places user A on hold and dials user C.

3. After user B hears a ringback and user C rings, user B initiates a transfer, and the call between user A and user B is terminated.

4. User A is transferred to user C and hears a ringback if user C is available. If user C is busy, user A hears a busy tone.

5. If user C is not busy and answers, user A and user C connect.

Figure 83 shows the call details for a semi-attended transfer.

Figure 83

Initiation of Semi-Attended Transfer

Attended Transfers

The following describes a typical attended transfer:

1. User A calls user B.

2. User B places user A on hold and dials user C.

3. After user C answers, user B goes on-hook to initiate a transfer, and the call between user A and user B is terminated.

4. User A is transferred to user C. If user C is busy when user B calls, user A hears a busy tone.

5. If user C is not busy and answers, user A and user C connect.

Figure 84 shows the call details for an attended transfer.

Figure 84

Attended Call Transfer

Table 50 summarizes the hookflash support for Call Transfer services.

3-Way Conference

You can use the 3-Way Conference feature to establish two calls with a single connection so that all three parties can talk together. If the 3-Way Conference feature is disabled, a second hookflash will toggle between the two calls.

Note The 3-Way Conference feature only supports those SIP calls using the g711 codex. This feature also supports specification GR-577-CORE.

Setting Up a 3-Way Conference

The following describes a typical 3-way conference scenario:

1. User A is talking with user B (a second-party call).

2. User A presses hookflash, receives a dial-tone, and dials user C.

3. User C answers. User A and user C are active in a second-party call.

4. User A presses hookflash to activate a 3-way conference.

In other terminology, user A is the host or controller; user B is the original call, and user C is the add-on.

Note The 3-Way Conference feature is available when the second-party call is outgoing. If the second-party call is incoming and you press hookflash, the phone toggles between the two calls.

Figure 85 shows the call details for 3-way conferencing.

Figure 85

3-Way Conference

Table 51 summarizes the hookflash support for 3-way conferencing services.

Terminating a 3-Way Conference

Table 52 summarizes the termination of a 3-way conference:

How to Configure and Associate SIP Support for Hookflash

This section describes the procedures for configuring and associating the SIP Support for Hookflash feature. These procedures include the following:

1. Configuring supplementary service by using the service dsapp command.

2. Associating the supplementary services with configured dial peers.

or

Associating the supplementary services as the global default application on a gateway.

This section provides configurations for the following supplementary services and provides configuration for associating supplementary services with dial peers:

•How to Configure Call Hold

•How to Configure Call Waiting

•How to Configure Call Transfer

•How to Configure 3-Way Conferencing

•How to Configure Disconnect Toggle Time

•How to Configure Blind Transfer Wait Time

•How to Associate Services with a Fixed Dial Peer

•How to Associate Services Globally on a Gateway

How to Configure Call Hold

To configure the Call Hold feature, follow these steps:

SUMMARY STEPS

1. enable

2. configure terminal

3. application

4. service dsapp

5. param callHold TRUE

6. exit

DETAILED STEPS

How to Configure Call Waiting

To enable call waiting for a DSAPP, follow these steps:

SUMMARY STEPS

1. enable

2. configure terminal

3. application

4. service dsapp

5. param callWaiting TRUE

6. exit

DETAILED STEPS

How to Configure Call Transfer

To configure call transfer, follow these steps:

SUMMARY STEPS

1. enable

2. configure terminal

3. application

4. service dsapp

5. param callTransfer TRUE

6. exit

DETAILED STEPS

How to Configure 3-Way Conferencing

To configure 3-way conferencing, follow these steps:

SUMMARY STEPS

1. enable

2. configure terminal

3. application

4. service dsapp

5. param callConference TRUE

6. exit

DETAILED STEPS

How to Configure Disconnect Toggle Time

You can configure the time to wait before switching to a call on hold if an active call disconnects (commonly known as disconnect toggle time). You can configure a time-to-wait range between 10 (default) and 30 seconds.

To configure disconnect toggle time, proceed with the following steps:

SUMMARY STEPS

1. enable

2. configure terminal

3. application

4. service dsapp

5. param disc-toggle-time seconds

6. exit

DETAILED STEPS

How to Configure Blind Transfer Wait Time

To configure the time the system waits before establishing a call, so that you can transfer a call by placing the phone on hook, proceed with the following steps.

Note The transferrer will not hear the alert for the time you configure because the system delays the call in case blind transfer is initiated.

SUMMARY STEPS

1. enable

2. configure terminal

3. application

4. service dsapp

5. param blind-xfer-wait-time time

6. exit

DETAILED STEPS

How to Associate Services with a Fixed Dial Peer

After you have enabled and customized your services on a gateway by using the service dsapp command, you must associate these services with configured dial peers. You can associate individual dial peers, or alternately, you can configure these services globally on the gateway (see How to Associate Services Globally on a Gateway). If you associate these services globally, all calls entering from the FXS line side and from the SIP trunk side invoke the service dsapp services.

To configure a fixed dial peer used by DSAPP to set up a call to the SIP server (trunk) side, proceed with the following steps:

SUMMARY STEPS

1. enable

2. configure terminal

3. application

4. service dsapp

5. param dialpeer dial-peer-tag

6. exit

DETAILED STEPS

How to Associate Services Globally on a Gateway

After you have enabled and customized your services on a gateway by using the service dsapp command, you must associate these services with configured dial peers. You can associate individual dial peers (see How to Associate Services with a Fixed Dial Peer), or alternately, you can configure these services globally on the gateway. If you associate these services globally, all calls entering from the FXS line side and from the SIP trunk side will invoke the service dsapp services.

To associate services globally on a gateway, proceed with the following steps:

SUMMARY STEPS

1. enable

2. configure terminal

3. application

4. global

5. service default dsapp

6. exit

DETAILED STEPS

Configuration Examples for SIP Support for Hookflash

This section contains the following configuration examples:

•Configuring Call Hold: Example

•Configuring Call Waiting: Example

•Configuring Call Transfer: Example

•Configuring 3-Way Conferencing: Example

•Configuring Disconnect Toggle Time: Example

•Configuring Blind Transfer Wait Time: Example

•Configuring a Fixed Dial Peer Used for Outgoing Calls on SIP Trunk Side: Example

•Associating Services with a Fixed Dial Peer: Example

•Associating Services Globally on a Gateway: Example

Configuring Call Hold: Example

The following example shows how to enable the Call Hold feature:

Gateway# configure terminal

Enter configuration commands, one per line. End with CNTL/Z.

Gateway(conf)# application

Gateway(config-app)# service dsapp

Gateway(config-app-param)# param callHold TRUE

Configuring Call Waiting: Example

The following example shows the configuration to enable the Call Waiting feature.

Gateway# configure terminal

Enter configuration commands, one per line. End with CNTL/Z.

Gateway(conf)# application

Gateway(config-app)# service dsapp

Gateway(config-app-param)# param callWaiting TRUE

Configuring Call Transfer: Example

The following example shows how to enable the Call Transfer feature:

Gateway# configure terminal

Enter configuration commands, one per line. End with CNTL/Z.

Gateway(conf)# application

Gateway(config-app)# service dsapp

Gateway(config-app-param)# param callTransfer TRUE

Configuring 3-Way Conferencing: Example

The following example shows how to enable the 3-Way Conferencing feature:

Gateway# configure terminal

Enter configuration commands, one per line. End with CNTL/Z.

Gateway(conf)# application

Gateway(config-app)# service dsapp

Gateway(config-app-param)# param callConference TRUE

Configuring Disconnect Toggle Time: Example

In this example, a disconnect toggle time is configured; the toggle time specifies the amount of time in seconds the system waits before committing the call transfer, after the originating call is placed on hook.

Gateway# configure terminal

Enter configuration commands, one per line. End with CNTL/Z.

Gateway(conf)# application

Gateway(config-app)# service dsapp

Gateway(config-app-param)# param disc-toggle-time 10

Configuring Blind Transfer Wait Time: Example

In this example, a blind transfer wait time is configured that specifies the amount of time in seconds the system waits before committing the call transfer after the originating call is placed on hook.

Gateway# configure terminal

Enter configuration commands, one per line. End with CNTL/Z.

Gateway(conf)# application

Gateway(config-app)# service dsapp

Gateway(config-app-param)# param blind-xfer-wait-time 10

Configuring a Fixed Dial Peer Used for Outgoing Calls on SIP Trunk Side: Example

In this example, a fixed dial peer is configured to set up the call to the SIP server (trunk) side.

Gateway# configure terminal

Enter configuration commands, one per line. End with CNTL/Z.

Gateway(conf)# application

Gateway(config-app)# service dsapp

Gateway(config-app-param)# param dialpeer 5000

Associating Services with a Fixed Dial Peer: Example

In this example, a fixed dial peer is configured to set up the call to the SIP server (trunk) side. The line in bold shows the dial peer statement.

Gateway# show running log

.

!

application

 service dsapp

 param dialpeer 1234

 param disc-toggle-time 15

 param callWaiting TRUE

 param callConference TRUE

 param blind-xfer-wait-time 10

 param callTransfer TRUE

!

voice-port 1/0/0

 station-id-name Example1

 station-id number 1234567890

!

voice-port 1/0/1

 station-id-name Example2

 station-id number 1234567891

!

voice-port 1/0/2

 station-id-name Example31

 station-id number 1234567892

!

dial-peer voice 1234 voip

 service dsapp

 destination-pattern.T

 session protocol sipv2

 session target ipv4:10.1.1.1

 dtmf-relay rtp-nte

 codec g711ulaw

!

dial-peer voice 100 pots

 service dsapp

 destination-pattern.1234567890

 port 1/0/0

 prefix 1234567890

!

dial-peer voice 101 pots

 service dsapp

 destination-pattern.1234567891

 port 1/0/1

 prefix 1234567891

!

dial-peer voice 102 pots

 service dsapp

 destination-pattern.1234567892

 port 1/0/2

 prefix 1234567892

!

!

sip-ua

 registar ipv4:10.1.1.1 expires 3600

!

Associating Services Globally on a Gateway: Example

In this example, the gateway is associated globally with supplementary services. The lines in bold show the dial peer statement.

Gateway# show running log

.

!

application

 service dsapp

 param disc-toggle-time 15

 param callWaiting TRUE

 param callConference TRUE

 param blind-xfer-wait-time 10

 param callTransfer TRUE

!

voice-port 1/0/0

 station-id-name Example1

 station-id number 1234567890

!

voice-port 1/0/1

 station-id-name Example2

 station-id number 1234567891

!

voice-port 1/0/2

 station-id-name Example31

 station-id number 1234567892

!

dial-peer voice 1234 voip

 service dsapp

 destination-pattern 1800T

 session protocol sipv2

 session target ipv4:10.1.1.1

 dtmf-relay rtp-nte

 codec g729r8

!

dial-peer voice 9753 voip

 service dsapp

 destination-pattern.T

 session protocol sipv2

 session target ipv4:10.1.1.1

 dtmf-relay rtp-nte

 codec g711ulaw

!

dial-peer voice 100 pots

 preference 8

 service dsapp

 destination-pattern.6234567890

 port 1/0/0

 prefix 6234567890

!

dial-peer voice 101 pots

 preference 8

 service dsapp

 destination-pattern.6234567892

 port 1/0/1

 prefix 6234567892

!

dial-peer voice 102 pots

 preference 8

 service dsapp

 destination-pattern.6234567893

 port 1/0/2

 prefix 6234567893

!

!

dial-peer hunt 2

!

sip-ua

 registar ipv4:10.1.1.1 expires 3600

!

Additional References

The following sections provide references related to the SIP Support for Hookflash feature.

MIBs

Technical Assistance

Any Internet Protocol (IP) addresses used in this document are not intended to be actual addresses. Any examples, command display output, and figures included in the document are shown for illustrative purposes only. Any use of actual IP addresses in illustrative content is unintentional and coincidental.

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