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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 soft switches. 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.
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.
Release by any party other than controller causes the conference to be released when Packet Voice Digital Signal Processor (DSP) Module (PVDM2) is used.
Release by any party on cascading 3-Way Conference, releases all the calls.
Invocation of features such as Call Hold, Blind Transfer, Semi-Attended Transfers after establishment of 3-Way Conference, releases all the calls.
Semi-attended transfer is not possible between users connected to gateways using G729 codec with CUCM. With G711 codec, semi-attended transfer is possible using CUCM.
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:
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.
The sequence of placing a call on hold is summarized in the following steps:
User A and user B are active with a call.
By pressing hookflash, user A initiates a call hold.
SIP sends a call hold indication to user B.
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. |
User A receives a second dial tone and presses hookflash.
The figure below shows the initiation of the calls hold sequence.
User A and User B reconnect.
The figure below shows the calls on hold resume sequence.
The table below summarizes the hookflash support for Call Hold services.
|
State |
Action |
Result |
Response to FXS Line |
|---|---|---|---|
|
Active call |
Hookflash |
Call placed on hold for remote party. |
Second dial tone for FXS phone. |
|
Call on hold |
Hookflash |
Active call. |
FXS line connects to call. |
|
Call on hold and active call |
Hookflash |
Active and call on hold are swapped. |
FXS line connects to previous held call. |
|
On hook |
Active call is dropped. |
Held call still active. Reminder ring on FXS line. |
|
|
Call on hold goes on hook |
Call on hold is dropped. |
None. |
|
|
Active call goes on hook |
Active call is dropped. |
Silence. Reconnects to held call after the value you specify for disc-toggle-time expires. See "How to Configure Disconnect Toggle Time". |
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:
User A is active with a call to user B
User C calls user B
User B presses hookflash.
The call between user A and user B is held.
User B connects to user C.
The figure below shows the call waiting sequence.
The table below summarizes hookflash support for Call Waiting services.
|
State |
Action |
Result |
Response to FXS Line |
|---|---|---|---|
|
Active call and waiting call |
Hookflash |
Swap active call and waiting call. |
FXS line connects to waiting call. |
|
Active call disconnects |
Active call is disconnected. |
Silence. |
|
|
Waiting call goes disconnects |
Stay connected to active call. |
None. |
|
|
Call disconnects |
Active call is dropped. |
Reminder ring on FXS line. |
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.
The following describes a typical Blind call-transfer scenario:
User A calls user B.
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". |
Before the Blind call transfer trigger timer expires, user B disconnects, and the call between user A and user B is terminated.
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.
The figure below shows the call sequence for a Blind call transfer.
The following is a typical semi-attended transfer scenario:
User A calls user B.
User B places user A on hold and dials user C.
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.
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.
The figure below shows the call details for a semi-attended transfer.
The following describes a typical attended transfer:
User A calls user B.
User B places user A on hold and dials user C.
After user C answers, user B goes on-hook to initiate a transfer, and the call between user A and user B is terminated.
User A is transferred to user C. If user C is busy when user B calls, user A hears a busy tone.
If user C is not busy and answers, user A and user C connect.
The figure below shows the call details for an attended transfer.
The table below summarizes the hookflash support for Call Transfer services.
|
State |
Action |
Result |
Response to FXS Line |
|---|---|---|---|
|
Active call |
Hookflash. |
Call placed on hold. |
Second dial tone. |
|
Call on hold and outgoing dialed or alerting, or active call |
On hook. |
Call on hold and active call transferred. |
-- |
|
Call on hold and outgoing alerting call |
Hookflash |
Active call dropped. |
FXS line connects to call o hold. |
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 supports only those SIP calls that use the g711 or g729 codecs. This feature also supports specification GR-577-CORE. |
The following describes a typical 3-way conference scenario:
User A is talking with user B (a second-party call).
User A presses hookflash, receives a dial-tone, and dials user C.
User C answers. User A and user C are active in a second-party call.
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. |
The figure below shows the call details for 3-way conferencing.
The table below summarizes the hookflash support for 3-way conferencing services.
|
State |
Action |
Result |
Response to FXS Line |
|---|---|---|---|
|
Active call |
Hookflash |
Call place on hold. |
Second dial tone |
|
Call on hold and active call |
Join call on hold and active call. |
Media mixing of both calls |
The table below summarizes the termination of a 3-way conference:
|
State |
Action |
Result |
Response to FXS Line |
|---|---|---|---|
|
Active 3-way conference |
User A disconnects first |
3-Way conference terminates; all users are disconnected. |
Dial tone |
|
User B disconnects first |
User A and user C establish a second-party call. |
FXS line connects user A and user C. |
|
|
User C disconnects first |
User A and user B establish a second-party call. |
FXS line connects user A and user B. |
|
|
User A presses hookflash |
User C disconnects and user A and user B establish a second-party call. |
FXS line connects user A and user B. |
This section describes the procedures for configuring and associating the SIP Support for Hookflash feature. These procedures include the following:
Configuring supplementary service by using the service dsapp command.
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:
| Command or Action | Purpose | |
|---|---|---|
| Step 1 |
enable Example: |
Enables privileged EXEC mode.
|
| Step 2 |
configure terminal Example: |
Enters global configuration mode. |
| Step 3 |
application Example: |
Enters SIP gateway-application configuration mode. |
| Step 4 |
service dsapp Example: |
Enters DSAPP parameters mode. |
| Step 5 |
param callHold TRUE Example: |
Enables call hold. |
| Step 6 |
exit Example: |
Exits the current mode. |
To enable call waiting for a DSAPP, follow these steps:
| Command or Action | Purpose | |
|---|---|---|
| Step 1 |
enable Example: |
Enables privileged EXEC mode.
|
| Step 2 |
configure terminal Example: |
Enters global configuration mode. |
| Step 3 |
application Example: |
Enters SIP gateway-application configuration mode. |
| Step 4 |
service dsapp Example: |
Enters DSAPP parameters mode. |
| Step 5 |
param callWaiting TRUE Example: |
Enables call waiting. |
| Step 6 |
exit Example: |
Exits the current mode. |
| Command or Action | Purpose | |
|---|---|---|
| Step 1 |
enable Example: |
Enables privileged EXEC mode.
|
| Step 2 |
configure terminal Example: |
Enters global configuration mode. |
| Step 3 |
application Example: |
Enters SIP gateway-application configuration mode. |
| Step 4 |
service dsapp Example: |
Enters DSAPP parameters mode. |
| Step 5 |
param callTransfer TRUE Example: |
Enables call transfer. |
| Step 6 |
exit Example: |
Exits the current mode. |
| Command or Action | Purpose | |
|---|---|---|
| Step 1 |
enable Example: |
Enables privileged EXEC mode.
|
| Step 2 |
configure terminal Example: |
Enters global configuration mode. |
| Step 3 |
application Example: |
Enters SIP gateway-application configuration mode. |
| Step 4 |
service dsapp Example: |
Enters DSAPP parameters mode. |
| Step 5 |
param callConference TRUE Example: |
Enables 3-way conferencing. |
| Step 6 |
exit Example: |
Exits the current mode. |
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.
| Command or Action | Purpose | |
|---|---|---|
| Step 1 |
enable Example: |
Enables privileged EXEC mode.
|
| Step 2 |
configure terminal Example: |
Enters global configuration mode. |
| Step 3 |
application Example: |
Enters SIP gateway-application configuration mode. |
| Step 4 |
service dsapp Example: |
Enters DSAPP parameters mode. |
| Step 5 |
param disc-toggle-time seconds Example: |
Sets the time to wait before switching to a call on hold, if the active call disconnects (disconnect toggle time). You can specify a disconnect toggle time between 10 (default) and 30 seconds. |
| Step 6 |
exit Example: |
Exits the current mode. |
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. |
| Command or Action | Purpose | |
|---|---|---|
| Step 1 |
enable Example: |
Enables privileged EXEC mode.
|
| Step 2 |
configure terminal Example: |
Enters global configuration mode. |
| Step 3 |
application Example: |
Enters SIP gateway-application configuration mode. |
| Step 4 |
service dsapp Example: |
Enters DSAPP parameters mode. |
| Step 5 |
param blind-xfer-wait-time time Example: |
Enables call waiting. |
| Step 6 |
exit Example: |
Exits the current mode. |
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:
| Command or Action | Purpose | |
|---|---|---|
| Step 1 |
enable Example: |
Enables privileged EXEC mode.
|
| Step 2 |
configure terminal Example: |
Enters global configuration mode. |
| Step 3 |
application Example: |
Enters SIP gateway-application configuration mode. |
| Step 4 |
service dsapp Example: |
Enters DSAPP parameters mode. |
| Step 5 |
param dialpeer dial-peer-tag Example: |
Configures a fixed dial peer used by DSAPP to set up a call to the SIP server (trunk) side, where dial-peer-tag is the tag of the dial peer used to place an outgoing call on the IP trunk side. The dial-peer-tag must be the same tag as the dial peer configured to the SIP server. |
| Step 6 |
exit Example: |
Exits the current mode. |
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 ("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.
| Command or Action | Purpose | |
|---|---|---|
| Step 1 |
enable Example: |
Enables privileged EXEC mode.
|
| Step 2 |
configure terminal Example: |
Enters global configuration mode. |
| Step 3 |
application Example: |
Enters SIP gateway-application configuration mode. |
| Step 4 |
global Example: |
Enters SIP gateway-application-global configuration mode. |
| Step 5 |
service default dsapp Example: |
Globally sets dsapp as the default application. All calls entering the gateway (from the FXS line side and the SIP trunk side) invoke the dsapp application. |
| Step 6 |
exit Example: |
Exits the current mode. |
After the 3-way conference is established, perform this task to verify the codec used for the conference.
| Step 1 |
enable Example:Enables privileged EXEC mode. |
| Step 2 |
show call active voice compact Example: |
You can use the following commands to troubleshoot the SIP Support for Hookflash feature:
debug voice application session
debug ccsip all (SIP message level debug)
debug voice ccapi inout
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
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
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
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
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
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
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
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 9753 voip
service dsapp
destination-pattern.T
session protocol sipv2
session target ipv4:15.0.0.15
dtmf-relay rtp-nte
codec g729r8
!
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
!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
!The following sections provide references related to the SIP Support for Hookflash feature.
|
MIB |
MIBs Link |
|---|---|
|
None |
To locate and download MIBs for selected platforms, Cisco IOS releases, and feature sets, use Cisco MIB Locator found at the following URL: |
|
Description |
Link |
|---|---|
|
The Cisco Technical Support website contains thousands of pages of searchable technical content, including links to products, technologies, solutions, technical tips, and tools. Registered Cisco.com users can log in from this page to access even more content. |
The following table provides release information about the feature or features described in this module. This table lists only the software release that introduced support for a given feature in a given software release train. Unless noted otherwise, subsequent releases of that software release train also support that feature.
Use Cisco Feature Navigator to find information about platform support and Cisco software image support. To access Cisco Feature Navigator, go to www.cisco.com/go/cfn. An account on Cisco.com is not required.|
Feature Name |
Releases |
Feature Information |
|---|---|---|
|
SIP Support for Hookflash |
12.4(11)T |
This feature was introduced. SIP Support for Hookflash feature allows you to configure IP Centrex supplementary services on SIP-enabled, Foreign Exchange Station (FXS) lines. |
|
SIP Support for Hookflash |
15.4(2)T |
The feature was enhanced to support hookflash using G729 codec for 3-way conference. |
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