Analog FXS Port SIP Registration with CUCM

1

Configuration Example

Analog FXS port SIP

Registration with

CUCM

2

Version: 23

rd

February 2023

Introduction

This document helps configure FXS ports as SIP endpoints registered to Cisco Unified Communications

Manager (CUCM) in order to support supplementary services on SIP Endpoints.

Prerequisites

Cisco recommends to have knowledge of these subjects

SIP protocol

Foreign Exchange Station (FXS) ports

Cisco Unified Communications Manager (CUCM) Cisco Analog Voice Gateway (VG Series)

Requirements

The FXS ports for Supplementary Services support CUCM version 12.5.1 SU1 or later with IOS XE 16.12.1

and above.

CUCM and IOS release for all platforms

ISR4461/VG450

IOS: 16.12.1

CUCM: 12.5.1 SU1

VG420

IOS: 17.6.1

CUCM: 12.5.1 SU4, 14.0 SU1

ISR4K, C8300, C8200 and VG400

IOS: 17.8.1

CUCM: 14.0 SU1

Supported Platforms

Cisco ISR4451-X/K9

Cisco ISR4461/K9

Cisco ISR4431/K9

Cisco ISR4351/K9

Cisco ISR4331/K9

Cisco ISR4321/K9

C8300-2N2S-4T2X

C8300-2N2S-6T

C8300-1N1S-4T2X

C8300-1N1S-6T

C8200-1N-4T

C8200L-1N-4T

VG400-2FXS/2FXO

VG400-4FXS/4FXO

VG400-6FXS/6FXO

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VG400-8FXS

VG420-144FXS

VG420-84FXS/6FXO

VG420-132FXS/6FXO

VG450/K9*

VG450-72FXS/K9

VG450-144FXS/K9

*Requires a NIM

Supported Features:

The following supplementary services are supported.

Call Hold

Call Waiting

Call Transfer (unattended/attended)

Call Forward no Answer

Audio Message Waiting Indicator

Call Forwarding Unrestricted

Call Forward Busy

Call Park Directed

Call Pickup Directed

Call Pickup Group

Three-way Conference

Configure

To implement supplementary services for Foreign Exchange Station (FXS) ports the call control server

(CUCM) should be able to subscribe to the hookflash or onhook events. This requires FXS ports to be

registered to CUCM as SIP endpoints. The use of SIP over SCCP facilitates features such as SIP Header

modification, endpoint based call routing and enables new features such as directed call retrieval.

Network Diagram

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Configuration steps

This section describes the configuration required for this feature to work:

Configuring the Device Control Session Application

DSAPP (Device control Session Application) is the application that drives these Hook Flash features. It

can be configured globally or on a dial-peer basis.

SUMMARY STEPS

1. enable

2. configure terminal

3. application

4. global

5. service default dsapp

6. param dialpeer number

7. param callWaiting string

8. param callConference string

9. param callTransfer string

DETAILED STEPS

Command or Action

Purpose

Step 1

enable Example:

Router> enable

Enables privileged EXEC mode. Enter

your password if prompted.

Step 2

configure terminal Example:

Router# configure terminal

Enters global configuration mode.

Step 3

application global service default dsapp

Example:

Router(config)#application router(config-

app)#global

Router(app-global)#service default dsapp

(Optional) Enables the new hookflash

functionality globally. Device Control

Session Application (DSAPP) drives these

hookflash features and it must be

configured for new bookflash

functionality for an application

framework module in IOS. DSAPP can be

configured globally or on a dial-peer

basis.

Note

Only required if all calls on the gateway

are to be controlled

by DSAPP else it may lead to call failure.

Step 4

param dialpeer number

Example: router(config)#application

Router(config-app)#service dsapp

Router(app-global)#param dialpeer 100

If multiple dial-peer matches are made

for the destination-pattern,

dialpeer 100 command is used.

Note When you configure DSAPP on a

dial-peer basis, specify a VOIP dial-peer

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for any outbound call. If all outbound

calls that use the hookflash functionality

are on the same server, it is

recommended to use the param dial-

peer command.

When multiple matches are possible on

hookflash, enable peer parameters

callXXXX TRUE for DSAPP to interpret

hookflash to SIP supplementary service

messages

Step 5

param callWaiting string

Example:

Router(config)#application router(config-

app)#service dsapp

Router(app-global)#param dialpeer 100

Router(app-global)#param callWaiting TRUE

Enables call waiting feature.

Step 6

param callConference string

Enables call conference feature.

Step 7

param callTransfer string

Example:

Router(config)#application router(config-

app)#service dsapp

Router(app-global)#param dialpeer 100

Router(app-global)#param callWaiting TRUE

Router(app-global)#param callConference TRUE

Router(app-global)#param callTransfer TRUE

Enables call transfer feature.

Configuring the Outbound VoIP Dial-peer

Outbound dial-peer is configured like regular voip dial-peer for SIP. In addition to the parameters

required, the following configurations are required:

• service dsapp– specifies this dial-peer will be controlled by dsapp application

• session transport tcp – specifies only TCP signaling is supported now

• voice-class sip extension gw-ana – used to interop with CUCM

• voice-class sip bind control source-interface GigabitEthernet0/0/1 – need this interface’s mac

as the base mac

dial-peer voice 714281111 voip

service dsapp

destination-pattern .+

session protocol sipv2

session transport tcp

session target ipv4:172.16.10.10

incoming called-number 7141116...

voice-class sip bind control source-interface GigabitEthernet0/0/0

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codec g711ulaw

no shut

Note- G711 is the only codec supported for conference calls. It is recommended to add this command.

Configuring Pots Dial-peer

You can configure the pots dial-peer like a regular pots dial-peer for FXS. In addition to the parameters

required, you have to configure the following command under pots dial-peer to interpret hook flash

correctly and interop with CUCM:

• service dsapp — specifies this dial-peer to be controlled by DSAPP application.

• voice-class sip extension gw-ana – this parameter is used to interop with CUCM.

dial-peer voice 19993000 pots

service dsapp

destination-pattern 2124506300

voice-class sip extension

gw-ana port 3/0/0

Configuring Voice-card and SIP

When you configure the voice-card, all the traffic should go through the CUCM, the hairpin calls are not

supported. You have to configure no local-bypass command for the voice-card that have FXS SIP

endpoints.

For FXS SIP endpoints to register, configure the registrar IP address under the sip-ua mode and use the

TCP as the transport type. UDP protocol is not supported.

voice service voip

sip

bind control source-interface GigabitEthernet0/0/0

session transport tcp

no shut

!

voice-card 3/0

no local-bypass

no watchdog

!

sip-ua

registrar ipv4:172.16.10.10 expires 3600 tcp

protocol mode dual-stack

!

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Note - SIP service should be shut down before configuring the protocol mode. After configuring the

protocol mode as dual-stack, SIP service should be reenabled.

Enabling Device Control Session Application Line features

To register to CUCM as a SIP endpoint, and to distinguish line feature from trunk, configure the dsapp

line command.

SUMMARY STEPS

1. enable

2. configure

3. dsapp line

DETAILED STEPS

Command or Action

Purpose

Step 1

enable

Example:

Router> enable

Enables privileged EXEC mode. Enter your

password if prompted.

Step 2

configure terminal

Example:

Router# configure terminal

Enters global configuration mode.

Step 3

dsapp line

Example:

Router(config)#

Router(config)#dsapp line

Specifies the format of each call feature.

Note: If you do not configure the dsapp line

command, the gateway will act like a SIP trunk

and the analog phone may not register as SIP

endpoints. Also, you cannot configure the FAC.

Ensure to configure the dsapp line command to

enable the FXS for SIP supplementary services

Configuring Feature Access Code

The dsapp line feature access-code command invokes the feature to translate the Feature

Access Code (FAC) to the format that the CUCM understands. If you do not configure this

command, the whole FAC digits are sent to the CUCM and may not invoke features. You can

also change the default FAC in the sub-mode.

Analog phones do not have soft keys. The required supplementary service features are invoked

through FAC. By default, prefix of the FAC is ‘**’ and it can also be changed using the CLI

command.

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Router(config)#dsapp line feature access-code

Router(config-dsappline-fac)#prefix *#

Router(config-dsappline-fac)#cancel-call-waiting **4

Router(config-dsappline-fac)#exit

Router# show dsapp line feature

codes dsapp line feature

access-code prefix *#

call forward all *#1

call forward cancel *#2

pickup local *#5

pickup group *#7

pickup direct *#6

cancel-call-waiting **4 last-redial *#3

If the dsapp line feature access-code is not configured, the voice gateway does not translate

the FAC to the format that the CUCM understands. The whole FAC digits is sent to the CUCM.

After the FAC is disabled and re-enabled, all the FAC and prefix are rolled back to the default

values.

Router(config)#no dsapp line feature

access-code Feature access-code

disabled

Router(config)#do show dsapp line feature

codes dsappline feature access-code

disabled

Router(config)#dsapp line feature access-code

Router(config-dsappline-fac)#do show dsapp line feature codes

dsapp line feature

access-code prefix **

call forward all **1

call forward cancel

**2 pickup local **5

pickup group **7

pickup direct **6

cancel-call-waiting

**9 last-redial **3

Router(config-dsappline-fac)#do show run | b dsapp line

dsapp line

!

dsapp line feature access-code

Auto Configuration

To enable the auto-configuration, use the ccm-manager sipana auto-config local command. To get the

XML configuration file, use the ccm-manager config server command to download the configuration file

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from the CUCM TFTP server. Configurations are needed on both CUCM and voice gateway. CUCM needs

to be configured first, then those configurations can be pushed to the voice gateway.

!

ccm-manager sipana auto-config local GigabitEthernetx/y/z

!

ccm-manager config server 172.xx.0.0

Note- Auto-Config only adds the dialpeers for each endpoint configured on CUCM. All other required SIP

CLI commands need to be added.

CUCM Configuration

VG450 is used in this example.

1. Navigate to Device> Gateway>Add New>Gateway Type.

2. Select SIP as the protocol and click Next

3. Enter the mac address of the interface used in sip bind control.

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4. Configure the voice module and individual voice ports.

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5. Add a directory Number > Save>Apply config.

7. The Analog port would now show as registered

Limitations

1. Shared line on SIP registered FXS port is not supported.

2. Line side SIP endpoints can be controlled by one CUCM.

3. CUCM failover for analog SIP phones is not supported.

4. Music on hold is not supported.

5. Third-party call agents are not supported.

6. Only non-secure calls are supported.

Verify

Use the following commands to verify the DSAPP configuration:

• show dsapp line device summary

• show dsapp line feature codes

• show ccm-manager config-download

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The show dsapp line device summary command shows whether the FXS ports are successfully

registered to the CUCM as SIP endpoints.

Router#show dsapp line device summary

Total Devices: 3

Port Device Registration Dev Directory Last Number Identifier

Name State Type Number Dialed

3/0/0 ANDD309DD761600 REGISTERED ALG 2124506300 Not Avail

3/0/1 ANDD309DD761601 REGISTERED ALG 2124506301 Not

Avail

3/0/2 ANDD309DD761602 UNREGISTERED ALG 2124506302 Not Avail

The show dsapp line feature codes command shows whether FAC is enabled and feature codes

Router#show dsapp line feature codes

dsapp line feature

access-code prefix **

call forward all **1

call forward cancel

**2 pickup local **5

pickup group **7

pickup direct **6

cancel-call-waiting

**9 last-redial **3

The show ccm-manager config-download command provides download status and history of the auto-

configuration.

Art_Utah_73#show ccm-manager config-download

SIP Line Side Analog auto-configuration status

===============================================================

Registered with Call Manager: Yes

Local interface: GigabitEthernet0/0/0 (2c5a.0fc8.8b70)

Current version-id: 1541004382-f60b9ac2-ce5b-439e-92e5-02b62e26d15c Current

config applied at: 16:47:40 UTC Oct 31 2018

Gateway downloads succeeded: 2 Gateway download attempts: 2

Last gateway download attempt: 16:47:40 UTC Oct 31 2018

Last successful gateway download: 16:47:40 UTC Oct 31 2018 Current TFTP

server: 172.19.156.84

Gateway resets: 1

Managed endpoints: 3

Endpoint downloads succeeded: 6 Endpoint download attempts: 6

Last endpoint download attempt: 16:47:40 UTC Oct 31 2018

Last successful endpoint download: 16:47:40 UTC Oct 31 2018

Endpoint resets: 0

Endpoint restarts: 0

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Configuration Error History:

Troubleshoot

For registration issues capture CUCM SDL/SDI traces and run the following debugs on the gateway

debug voip application session debug ccsip messages

debug ccsip states debug ccsip error

For auto-configuration issues run the debug debug ccm-manager config-download all

Related Information

Configuring the Cisco Fourth-Generation T1/E1 Voice and WAN Network Interface Module