Cisco SPA8800 Configuring

Application Note

Revised June 10th 2009

Configuring the Cisco SPA8800 IP Telephony Gateway in

®

an Asterisk

Environment

Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment

TableofContents

Introduction 5

Audience 5

Scope 5

Related Documents

• Asterisk: http://www.asterisk.org

• Asterisk Book from O'Reilly: http://www.asteriskdocs.org/

• Cisco SPA8800 Administration Guide

Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment

• Cisco Small Business IP Phones Admin Guide

• Cisco Small Business IP Phones User Guide

• Cisco Community Central: Small Business Community ATA Support

Overview

Configuring the SPA8800 is a relatively trivial task and is similar to configuring any of the Sipura / Linksys / Cisco ATA and SPA9000 Voice System devices. Troubleshooting configuration problems due to incorrectly typed information in configuration fields requires advanced network and Asterisk troubleshooting skills. This application note walks you through configuring a SPA8800 and also provides sample traces that may be of use to you when you are troubleshooting your SPA8800 in an Asterisk environment.

By the end of this document, an Asterisk phone user, analog or IP, will be able to pick up a phone and dial out via the PSTN or ITSP, depending on the steering digit they use.

Summary of Tasks in this Document

You must complete the following tasks in order to use the SPA8800 in an Asterisk environment:

1. Gather Basic Information

2. Configure the Asterisk Server a. Edit the sip.conf file b. Edit the extensions.conf file c. Connect to the Asterisk Server's console d. Reload Asterisk modules

3. Connect the SPA8800

4. Configure the SPA8800 a. Configure static IP address and related information b. Upgrade the SPA8800's firmware

5. Configure phone extensions on the SPA8800 FXS Phone N ports

6. Configure FXO line parameters on the SPA8800 LINE N ports

7. Configure FXO line dial plans for inbound PSTN call routing

8. Test the phone system for appropriate behavior

Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment

Requirements

You need the following equipment and services:

• A functional Asterisk server

• A functional LAN with network connectivity to the Internet and an optional Internet Telephony

Service Provider (ITSP)

• A SPA8800

• One to four analog phones

• Optional IP phones such as the SPA525G, SPA9x2, SPA9x1, or WIP310 (wireless) IP phones

Configuring Asterisk for a SPA8800

Before you configure your Asterisk server for the SPA8000, you need to gather some basi c information:

1. Static IP address for the SPA8800. By default, this device is a DHCP client, but will be of no use to you if it is assigned a new dynamic IP address periodically. In this document, I use

192.168.2.237/24

2. Gateway / Default router, DNS, and NTP server IP addresses

3. Extension numbers for up to four analog phones to be connected to the SPA8800's PHONE FXS ports 1-4. In this document, I use 101, 102, 103, and 104.

4. Decide how many PSTN lines will be connected to the SPA8800's LINE FXO ports 1-4. In this document, I use two PSTN lines connected to FXO LINE ports 2 and 3 [UDP 5161 and 5261 respectively]

5. Decide what steering digits to use. In this document, I use 8 for PSTN and 9 for ITSP

6. Decide what phone or phones to route inbound PSTN and ITSP calls to. In this document, I will route all inbound calls to the analog phone associated with extension 101.

7. Decide what to name the SPA8800 context group in the extensions.conf file. In this document, I use the [fxsgroup] context.

Summary:

• SPA8800 static IP address: 192.168.2.237/24

• Gateway / Default router, DNS, and NTP server IP addresses

• Analog phones: 101-104

• PSTN lines: 2, LINE 2 [UDP 5161] and LINE 3 [UDP 5261]

• Steering digits: 8 for PSTN, 9 for ITSP

• Inbound PSTN call target: 101

• Inbound ITSP call target: 101

Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment

Configuring the Asterisk Server

Once you have gathered all of the basic information, you can begin configuring the Asterisk server.

Editing the /etc/asterisk/sip.conf file

# vim /etc/asterisk/sip.conf

[general] … … ;register => <DID>@<ITSP>:<password>:<DID>@<ITSP>/101 register =>

3615551212@sip.broadvoice.com:mypassword:3615551212@sip.broadvoice.com/101 … … ; ;SPA8800 Changes ;define SPA8800 analog phone 1 extension 101 [101] type=friend username=101 secret=101 qualify=yes nat=no host=dynamic canreinvite=no context=fxsgroup regext=101 ; ;define SPA8800 analog phone 2 extension 102 [102] type=friend username=102 secret=102 qualify=yes nat=no host=dynamic canreinvite=no context=fxsgroup regext=102 ; ;define SPA8800 analog phone 3 extension 103 [103] type=friend username=103 secret=103 qualify=yes nat=no host=dynamic

Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment

canreinvite=no context=fxsgroup regext=103 ; ;define SPA8800 analog phone 4 extension 104 [104] type=friend username=104 secret=104 qualify=yes nat=no host=dynamic canreinvite=no context=fxsgroup regext=104 ; ;define SPA8800 pstn2 user [pstn2] type=friend host=192.168.2.237 ;IP address of the SPA8800 port=5161 ;5161 is the default SIP port for line 2 on the SPA8800 dtmfmode=rfc2833 context=pstn2 insecure=very ; ;define SPA8800 pstn3 user [pstn3] type=friend host=192.168.2.237 ;IP address of the SPA8800 port=5261 ;5261 is the default SIP port for line 2 on the SPA8800 dtmfmode=rfc2833 context=pstn3 insecure=very ; [itsp1] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=3612887272 secret=TitSSTQqpp username=3612887272 insecure=very authname=3612887272 dtmfmode=inband dtmf=inband canreinvite=no qualify=yes

Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment

nat=yes context=itsp1 ;eof

Editing the /etc/asterisk/extensions.conf file

# vim /etc/asterisk/extensions.conf

… … ; SPA8800 Changes ;outbound dialing [fxsgroup] ; ; ; dial 7 to explicitly use FXS3 exten => _7.,1,Dial(SIP/${EXTEN:1}@pstn3,60,r) ;

8 as a steering digit:

; dial ; if FXS2 is not available, FXS3 will be used. ; if FXS3 is not available, the user hears congestion exten => _8.,1,Dial(SIP/${EXTEN:1}@pstn2,60,r) exten => _8.,2,Dial(SIP/${EXTEN:1}@pstn3,60,r) ; ; dial 9 to explicitly use ITSP exten => _9.,1,Dial(SIP/${EXTEN:1}@itsp1,30,r) ; ; r causes ringing for calling party but audio is not ; passed until called party answers call ; T allows caller to transfer with # exten => 101,1,Dial(SIP/101,60,rT) exten => 102,1,Dial(SIP/102,60,rT) exten => 103,1,Dial(SIP/103,60,rT) exten => 104,1,Dial(SIP/104,60,rT) exten => 200,1,Dial(SIP/200,60,rT) exten => 201,1,Dial(SIP/201,60,rT) ; ;inbound from PSTN [pstn2] ; t allows called person to transfer with a # exten => 101,1,Dial(SIP/101,60,rt) [pstn3] exten => 201,1,Dial(SIP/201,60,rt)

;inbound calls from ITSP [itsp1] exten => 3615551212,1,Answer ;enable ring group of extensions 101, 102, and 200 exten => 3615551212,2,Dial(SIP/101&SIP/102&SIP/200,25,rt) exten => 3615551212,3,Hangup ;eof … …

Loading the Modified Configuration

1. Connect to the Asterisk console:

Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment

$ sudo asterisk –r *CLI>

2. Use the reload command to load the changed configuration:

*CLI> module reload

This completes the Asterisk server configuration. You must now configure the SPA8800 to register to the Asterisk server.

Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment

Configuring the SPA8800

SPA8800 Ports and Connections

Figure 1The photograph shows the rear of the SPA8800 and its connections

Connect the SPA8800

a. Connect the ETHERNET port of the SPA8800 to the LAN switch. b. Connect analog phones to the PHONE 1-4 FXS ports or use the multiport RJ-21

50-pin connector.

c. Connect PSTN lines to the LINE 1-2 F XO ports or use the multiport RJ-21 50-pin

connector. d. Connect the power adapter. e. Going off-hook with the analog phones will result in a fast-busy because the

SPA8800 has not yet been configured.

Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment

Figure 2: An example of deploying 2 SPA8800s in an Asterisk Environment

Factory Resetting the SPA8800

You should factory reset the SPA8800 so that you start from a known starting point.

f. Connect an analog phone to the SPA8800 PHONE 1 port g. Go off-hook, ignore the fast-busy or silence h. Dial **** [four asterisks or stars] i. Dial 73738# when prompted j. Press 1 to confirm reset k. Hang up when prompted

Configuring Static IP Address Addressing

You must configure the SPA8800 with a static IP address because this address is defined in the Asterisk Server's /etc/asterisk/sip.conf file.

l. Determine the SPA8800's Dynamically Assigned IP Address

i. Connect an analog phone to the SPA8800 PHONE 1 port

ii. Go off-hook, ignore the fast-busy or silence iii. Dial **** [four asterisks or stars] iv. Dial 110# when prompted

v. Document the IP address vi. Hang up

Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment

m. Direct your browser to the SPA8800's web user interface (web-ui)

http://<IP_address_of_SPA8800>/admin/advanced

n. Change the following parameters:

Network tab > Wan Status tab:

i. Internet Connection Settings > Connection Type: Static IP

ii. Static IP Settings:

1. Static IP:

2. Netmask:

3. Gateway:

iii. Optional Settings:

1. Primary DNS

2. Secondary DNS

3. Primary NTP Server

o. Click Submit All Changes

Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment

Upgrading the SPA8800's Firmware

1. Direct your browser to the SPA8800's web user interface (web-ui) http://<IP_address_of_SPA8800>/admin/advanced

2. Verify that Compare Network tab > Status tab > Product Information > Software Version: is up to date with SPA8800 firmware available at the Cisco.com site. If newer firmware is available, save it to disk and upgrade the SPA8800 as follows:

3. Copy the downloaded firmware image to your TFTP server's root directory

4. Cause the SPA8800 to retrieve the firmware by TFTP and install the new image:

http://IPADDRESSofSPA/upgrade?tftp://TFTPADDRESS/SPAFILE.bin

Where:

• IPADDRESSofSPA is the SPA8800 IP address

• TFTPADDRESS is the TFTP server's IP address

• SPAFILE.bin is the name of the downloaded firmware image

Example:

http://192.168.2.237/upgrade?tftp://192.168.2.20/spa8800-6-1-7-GW.bin

Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment

Configuring Phone Extensions on the SPA8800 FXS Phone Ports

In this section, you will point the SPA8800 to the Asterisk Server as the SIP proxy and the provide user credentials that you defined earlier in the sip.conf and extensions.conf Asterisk files. This configuration defines the characteristics of the analog phone connected to the SPA8800 FXS PHONE port.

1. Direct your browser to the SPA8800's web user interface (web-ui) http://<IP_address_of_SPA8800>/admin/advanced

2. Voice tab > Phone 1 tab > Line Enable: yes

3. Voice tab > Phone 1 tab > Proxy and Registration > Proxy: 192.168.2.20 Where this is the IP address of the Asterisk Server

4. Voice tab > Phone 1 tab > Subscriber Information >

5. Display Name: SPA8k8Phone1 Where this name is assigned by you for easy identification

6. User ID: 101 Where 101 [username] is defined in the /etc/asterisk/sip.conf and /etc/asterisk/extensions.conf files

7. Password: 101 Where the password [secret] is defined in the /etc/asterisk/sip.conf file

Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment

8. Configure the remaining phones using the parameters that you defined in the /etc/asterisk/sip.conf and /etc/asterisk/extensions.conf files.

9. Click Submit All Changes if you do not intend to complete the next step at this time.

Configuring FXO Line Ports on the SPA8800

In this section, you will point the SPA8800 to the Asterisk Server as the SIP proxy for the FXO ports and the provide user credentials that you defined earlier in the sip.conf and extensions.conf

Asterisk files. This configuration defines the characteristics of the FXO port connected to the PSTN line.

1. Direct your browser to the SPA8800's web user interface (web-ui) http://<IP_address_of_SPA8800>/admin/advanced

2. Voice tab > Line 2 tab > Line Enable: yes

3. Voice tab > Line N tab > Proxy and Registration >

a. Proxy: 192.168.2.20 [This field does not need to be completed, but is a good

reminder of which device is being used] b. Make Call Without Reg: yes c. Ans Call Without Reg: yes

4. Voice tab > Line N tab > Subscriber Information > d. Display Name: SPA8k8Line2 e. User ID: pstn2

Where pstn2 is defined in the /etc/asterisk/sip.conf and /etc/asterisk/extensions.conf files

f. Password: pstn2

Where the pstn2 password [secret] is defined in the /etc/asterisk/sip.conf file [This field does not need to be completed, the device does not need to register]

g. Configure the remaining lines using the parameters that you defined in the

/etc/asterisk/sip.conf and /etc/asterisk/extensions.conf files.

5. Click Submit All Changes if you do not intend to complete the next step at this time.

Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment

Configuring FXO Line Dial Plans for inbound PSTN call Routing

In this section, you will configure FXO port dial plans. These dial plans affect inbound PSTN call routing and work in conjunction with definitions made in the /etc/asterisk/sip.conf and /etc/asterisk/extensions.conf files.

The [general] section of sip.conf contains the register directive which instructs the SIP proxy on where (101) to send inbound calls:

[general] register => 3615551212@sip.broadvoice.com:mypassword:3615551212@sip.broadvoice.com/

The [pstn2] and [pstn3] contexts of the extensions.conf file describes how inbound calls must be routed:

[pstn2] exten => 101,1,Dial(SIP/101,60,rt) [pstn3] exten => 101,1,Dial(SIP/201,60,rt

1. Direct your browser to the SPA8800's web user interface (web-ui)

101

http://<IP_address_of_SPA8800>/admin/advanced

2. Voice tab > Line 2 tab > Dial Plans >

Edit any dial plan to route inbound calls from the PSTN line connected to this FXO line. In this example, Dial Plan 8 is edited with: (<:101@192.168.2.20>S0) where: All inbound calls will be routed to extension 101 of the Asterisk server [192.168.2.20].

3. Voice tab > Line 2 tab > PSTN-To-VoIP Gateway Setup > PSTN Caller Default DP: 8

Where this number must match the dial plan used in the previous step. Refer to the Failing Inbound from PSTN

in the Troubleshooting section to see a sample

Asterisk console error message that results from an incorrect dial plan entry.

Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment

4. Repeat for Line 3.

5. Click Submit All Changes.

6. The entire configuration process is complete once the SPA8800 has reb ooted.

Testing the Phone System

Test the phone system as follows for appropriate behavior:

1. Test internal calls: a. Verify that analog phones can call each other internally. For example

call from 101 to 102.

b. Optional: Verify that analog phones can call IP phones internally. For

example, call from 101 to 200 [if configured]

2. Test inbound calls: a. From the PSTN, call a phone line attached to the SPA8800's FXO line.

Verify that the appropriate phone rings, analog phone 101 in this document's example.

b. From the PSTN, call a DID associated with the Asterisk server.

Verify that the appropriate phone rings, analog phone 101 in this document's example.

3. Test outbound calls: a. From an analog phone, make an outbound call using the PSTN by using

the appropriate steering digit, 8 in this document's example.

b. From an analog phone, make another simultaneous outbound call usin g

the PSTN by using the appropriate steering digit, 8 in this document's example. Verify that both calls succeed, using both configured outbound PSTN lines.

c. From an analog pho ne, make an outbound call using the ITSP by using

the appropriate steering digit, 9 in this document's example.

Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment

T.38 Faxing

The SPA8800 supports fax with G.711 pass-through or real-time fax over IP via T.38 fax relay. The only change from default setting for fax pass-through is to change from the default named

signaling event (NSE) to ReINVITE: Web-ui > Voice tab > Line N tab > Audio Configuration > FAX Passthru Method: ReINVITE

Optionally, you can change the FAX Line Toggle code from the default of #99. Predialing #99 as a prefix forces the call to be a fax call. This will case the INVITE to indicate T.38 in the SDP without relying on a reINVITE to switch to T.38. The default can be changed from #99 with the web-ui > Voice tab > Regional tab > Vertical Service Activation Codes > FAX Line Toggle Code:

Refer to the

Cisco SPA8800 Configuring

Specifications and Main Features

Frequently Asked Questions

User Manual