Cisco Unified SCCP and SIP SRST System Administrator Guide (All Versions)
Cisco Unified SIP SRST 4.1
Downloads: This chapterpdf (PDF - 279.0KB) The complete bookPDF (PDF - 5.71MB) | Feedback

Table of Contents

Cisco Unified SIP SRST 4.1

Contents

Prerequisites for Cisco Unified SIP SRST 4.1

Restrictions for Cisco Unified SIP SRST 4.1

Information About Cisco Unified SIP SRST 4.1

Out-of-Dialog REFER

Digit Collection on SIP Phones

KPML Digit Collection

SIP Dial Plans

Caller ID Display

Disabling SIP Supplementary Services for Call Forward and Call Transfer

Idle Prompt Status

Enhanced 911 Services

How to Configure Cisco Unified SIP SRST 4.1 Features

Enabling KPML for SIP Phones

Restrictions

What to Do Next

Disabling SIP Supplementary Services for Call Forward and Call Transfer

Configuring Idle Prompt Status for SIP Phones

Prerequisites

Where to Go Next

Cisco Unified SIP SRST 4.1

Revised: February 3, 2011

This chapter describes the features and provides the configuration information for Cisco Unified SIP SRST 4.1:

  • Out-of-Dialog REFER(OOD-R)
  • Digit Collection on SIP Phones
  • Caller ID Display
  • Disabling SIP Supplementary Services for Call Forward and Call Transfer
  • Idle Prompt Status

Note With Cisco IOS Release 12.4(15)T, the number of SIP phones supported on each platform is now equivalent to the number of SCCP phones supported. For example, 3845 now supports 720 phones regardless of whether these are SIP or SCCP.


Prerequisites for Cisco Unified SIP SRST 4.1

Restrictions for Cisco Unified SIP SRST 4.1

  • Cisco Unified SRST does not support BLF speed-dial notification, call forward all synchronization, dial plans, directory services, or music-on-hold (MOH).
  • Prior to SIP phone load 8.0, SIP phones maintained dual registration with both Cisco Unified Communications Manager and Cisco Unified SRST simultaneously. In SIP phone load 8.0 and later versions, SIP phones use keepalive to maintain a connection with Cisco Unified SRST during active registration with Cisco Unified Communications Manager. Every 2 minutes, a SIP phone sends a keepalive message to Cisco Unified SRST. Cisco Unified SRST responds to this keepalive with a 404 message. This process repeats until fallback to Cisco Unified SRST occurs. After fallback, SIP phones send a keepalive message every two minutes to Cisco Unified Communications Manager while the phones are registered with Cisco Unified SRST. Cisco Unified SRST continues to support dual registration for SIP phone loads older than 8.0.

Information About Cisco Unified SIP SRST 4.1

Out-of-Dialog REFER

Out-of-dialog REFER (OOD-R) enables remote applications to establish calls by sending a REFER message to Cisco Unified SRST without an initial INVITE. After the REFER is sent, the remainder of the call setup is independent of the application and the media stream does not flow through the application. The application using OOD-R triggers a call setup request that specifies the Referee address in the Request-URI and the Refer-Target in the Refer-To header. The SIP messaging used to communicate with Cisco Unified SRST is independent of the end-user device protocol, which can be H.323, plain old telephone service (POTS), SCCP, or SIP. Click-to-dial is an example of an application that can be created using OOD-R.

A click-to-dial application enables users to combine multiple steps into one click for a call setup. For example, a user can click a web-based directory application from his or her PC to look up a telephone number, off-hook the desktop phone, and dial the called number. The application initiates the call setup without the user having to out-dial from his or her own phone. The directory application sends a REFER message to Cisco Unified SRST, which sets up the call between both parties based on this REFER.

For more information about OOD-R, see Out-of-Dialog REFER from the Cisco Unified Communications Manager Express System Administrator Guide.

Digit Collection on SIP Phones

Digit strings dialed by phone users must be collected and matched against predefined patterns to place calls to the destination corresponding to the user's input. Previously, SIP phones in a Cisco Unified SRST system required users to press the DIAL soft key or # key, or wait for the interdigit-timeout to trigger call processing. This could cause delays in processing the call.

Two new methods of collecting and matching digits are supported for SIP phones depending on the model of the phone:

KPML Digit Collection

The Key Press Markup Language (KPML) uses SIP SUBSCRIBE and NOTIFY methods to report user input digit by digit. Each digit dialed by the phone user generates its own signaling message to Cisco Unified SRST, which performs pattern recognition by matching a destination pattern to a dial peer as it collects the dialed digits. This process of relaying each digit immediately is similar to the process used by SCCP phones. It eliminates the need for the user to press the Dial soft key or wait for the interdigit timeout before the digits are sent to the Cisco Unified SRST for processing.

KPML is supported on Cisco Unified IP Phones 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE. For configuration information, see the “Enabling KPML for SIP Phones” section.

SIP Dial Plans

A dial plan is a set of dial patterns that SIP phones use to determine when digit collection is complete after a user goes off-hook and dials a destination number. Dial plans enable SIP phones to perform local digit collection and recognize dial patterns as user input is collected. After a pattern is recognized, the SIP phone sends an INVITE message to Cisco Unified SRST to initiate the call to the number matching the user's input. All of the digits entered by the user are presented as a block to Cisco Unified SRST for processing. Because digit collection is done by the phone, dial plans reduce signaling messages overhead compared to KPML digit collection.

SIP dial plans eliminate the need for a user to press the Dial soft key or # key or to wait for the interdigit timeout to trigger an outgoing INVITE. You configure a SIP dial plan and associate the dial plan with a SIP phone. The dial plan is downloaded to the phone in the configuration file.

You can configure SIP dial plans and associate them with the following SIP phones:

  • Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE: These phones use dial plans and support KPML. If both a dial plan and KPML are enabled, the dial plan has priority.

If a matching dial plan is not found and KPML is disabled, the user must wait for the interdigit timeout before the SIP NOTIFY message is sent to Cisco Unified SRST. Unlike other SIP phones, these phones do not have a Dial soft key to indicate the end of dialing, except when on-hook dialing is used.

  • Cisco Unified IP Phone 7905, 7912, 7940, and 7960: These phones use dial plans and do not support KPML. If you do not configure a SIP dial plan for these phones, or if the dialed digits do not match a dial plan, the user must press the Dial soft key or wait for the interdigit timeout before digits are sent to Cisco Unified SRST for processing.

When you reset a phone, the phone requests its configuration files from the TFTP server, which builds the appropriate configuration files depending on the type of phone.

  • Cisco Unified IP Phone 7905 and 7912: The dial plan is a field in their configuration files.
  • Cisco Unified IP Phone 7911G, 7940, 7941G, 7941GE, 7960, 7961G, 7961GE, 7970G, and 7971GE: The dial plan is a separate XML file that is pointed to from the normal configuration file.

The Cisco Unified SRST supports SIP dial plans if they are provisioned in Cisco Unified Communications Manager. You cannot configure dial plans in Cisco Unified SRST.

Caller ID Display

The name and number of the caller is included in the Caller ID display on the
Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE. Other SIP phones display only the number of the caller. Also, the caller ID information is updated on the destination phone when there is a change in the caller ID of the originating party such as with call forwarding or call transfer. No new configuration is required to support these enhancements.

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 redirect responses for call forwarding from being sent by Cisco Unified SRST.

Disabling supplementary services is supported if all endpoints use SCCP or all endpoints use SIP. It is not supported for a mix of SCCP and SIP endpoints.

Idle Prompt Status

A message displays on the status line of a SIP phone after the phone registers to Cisco Unified SRST to indicate that Cisco Unified SRST is providing fallback support for the Cisco Unified Communications Manager. This message informs the user that the phone is operating in fallback mode and that not all features are available. The default message that displays “CM Fallback Service Operating” is taken from the phone dictionary file. You can customize the message by using the system message command on the Cisco Unified SRST router. Cisco Unified SRST updates the idle prompt message when a SIP phone registers or when you modify the message through the configuration. The message displays until a phone switches back to the Cisco Unified Communications Manager.

The idle prompt status message is supported for the Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE with Cisco Unified SRST 4.1 and later versions. For versions earlier than Cisco Unified SRST 4.1, the phones display the default message from the dictionary file.

Enhanced 911 Services

Enhanced 911 Services for Cisco Unified SRST enables 911 operators to:

  • Immediately pinpoint the location of the 911 caller based on the calling number
  • Callback the 911 caller if a disconnect occurs

Before this feature was introduced, Cisco Unified SRST supported only outbound calls to 911. With basic 911 functionality, calls were simply routed to a Public Safety Answering Point (PSAP). The 911 operator at the PSAP would then have to verbally gather the emergency information and location from the caller, before dispatching a response team from the ambulance service, fire department, or police department. Calls could not be routed to different PSAPs, based on the specific geographic areas that they cover.

With Enhanced 911 Services, 911 calls are selectively routed to the closest PSAP based on the caller’s location. In addition, the caller’s phone number and address automatically display on a terminal at the PSAP. Therefore, the PSAP can quickly dispatch emergency help, even if the caller is unable to communicate the location. Also, if the caller disconnects prematurely, the PSAP has the information it needs to contact the 911 caller.

See Configuring Enhanced 911 Services from Cisco Unified Communications Manager Express System Administrator Guide for more information.

How to Configure Cisco Unified SIP SRST 4.1 Features

This section contains the following tasks:

Enabling KPML for SIP Phones

Perform the following steps to enable KPML digit collection on a SIP phone.

Restrictions

  • This feature is supported only on Cisco Unified IP Phone 7911G, 7941G, 7941GE, 7961G, 7961GE, 7970G, and 7971GE.
  • A dial plan assigned to a phone has priority over KPML.

SUMMARY STEPS

1. enable

2. configure terminal

3. voice register pool pool-tag

4. digit collect kpml

5. end

6. show voice register dial-peer

DETAILED STEPS

 

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

voice register pool pool-tag

 

Router(config)# voice register pool 4

Enters voice register pool configuration mode to set phone-specific parameters for a SIP phone.

  • pool-tag : Unique sequence number of the SIP phone to be configured. Range is version and platform-dependent; type ? to display range. You can modify the upper limit for this argument with the max-pool command.

Step 4

digit collect kpml

 

Router(config-register-pool)# digit collect kpml

Enables KPML digit collection for the SIP phone.

Note This command is enabled by default for supported phones in Cisco Unified CME and Cisco Unified SRST.

Step 5

end

 

Router(config-register-pool)# end

Exits to privileged EXEC mode.

Step 6

show voice register dial-peers

 

Router# show voice register dial-peers

Displays details of all dynamically created VoIP dial peers associated with the Cisco Unified CME SIP register including the defined digit collection method.

What to Do Next

After changing the KPML configuration in Cisco Unified SRST, you do not need to create new configuration profiles and restart the phones. Enabling or disabling KPML is effective immediately in Cisco Unified SRST.

Disabling SIP Supplementary Services for Call Forward and Call Transfer

Perform the following steps to disable REFER messages for call transfers and redirect responses for call forwarding from being sent to the destination by Cisco Unified SRST. You can disable these supplementary features if the destination gateway does not support them.

SUMMARY STEPS

1. enable

2. configure terminal

3. voice service voip
or
dial-peer voice tag voip

4. no supplementary-service sip { moved-temporarily | refer }

5. end

DETAILED STEPS

 

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

voice service voip

or

dial-peer voice tag voip

 

Router(config)# voice service voip

or

Router(config)# dial-peer voice 99 voip

Enters voice-service configuration mode to set global parameters for VoIP features.

or

Enters dial peer configuration mode to set parameters for a specific dial peer.

Step 4

no supplementary-service sip { moved-temporarily | refer }

 

Router(conf-voi-serv)# no supplementary-service sip refer

or

Router(config-dial-peer)# no supplementary-service sip refer

Disables SIP call forwarding or call transfer supplementary services globally or for a dial peer.

  • moved-temporarily : SIP redirect response for call forwarding.
  • refer : SIP REFER message for call transfers.
  • Sending REFER and redirect messages to the destination is the default behavior.
Note This command is supported for calls between SIP phones and calls between SCCP phones. It is not supported for a mixture of SCCP and SIP endpoints.

Step 5

end

 

Router(config-voi-serv)# end

or

Router(config-dial-peer)# end

Exits to privileged EXEC mode.

Configuring Idle Prompt Status for SIP Phones

Perform the following steps to customize the message that displays on SIP phones after the phones failover to Cisco Unified SRST.


Note You do not need to create new configuration files with the create profile command and restart the phones after changing the idle status message in Cisco Unified SRST. Modifying the status message takes effect immediately in Cisco Unified SRST.


Prerequisites

Cisco Unified SRST 4.1 or a later version.

SUMMARY STEPS

1. enable

2. configure terminal

3. voice register global

4. system message string

5. end

6. show voice register global

DETAILED STEPS

 

Command or Action
Purpose

Step 1

enable

 

Router> enable

Enables privileged EXEC mode.

  • Enter your password if prompted.

Step 2

configure terminal

 

Router# configure terminal

Enters global configuration mode.

Step 3

voice register global

 

Router(config)# voice register global

Enters voice register global configuration mode to set global parameters for all supported SIP phones in a Cisco Unified CME environment.

Step 4

system message string

 

Router(config-register-global)# system message fallback active

Defines a status message that displays on SIP phones registered to Cisco Unified SRST.

  • string : Up to 32 alphanumeric characters. Default is “CM Fallback Service Operating.”

Step 5

end

 

Router(config-register-global)# end

Exits to privileged EXEC mode.

Step 6

show voice register global

 

Router# show voice register global

Displays all global configuration parameters associated with SIP phones.

Where to Go Next

The next step is configuring Cisco Unified IP phones using SCCP. For instructions, see the “Setting Up Cisco Unified IP Phones using SCCP” section.

For additional information, see the “Additional References” section in the “Cisco Unified SCCP and SIP SRST Feature Overview” chapter.