Cisco Small Business SPA8800 Manuel utilisateur

Application Note

Revised June 10

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 5

Overview 6

Summary of Tasks in this Document 6

Requirements 7

Configuring Asterisk for a SPA8800 7

Summary: 7

Configuring the Asterisk Server 8

Editing the /etc/asterisk/sip.conf file 8

Editing the /etc/asterisk/extensions.conf file 10

Loading the Modified Configuration 10

Configuring the SPA8800 12

SPA8800 Ports and Connections 12

Connect the SPA8800 12

Factory Resetting the SPA8800 13

Configuring Static IP Address Addressing 13

Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment

Upgrading the SPA8800's Firmware 15

Configuring Phone Extensions on the SPA8800 FXS Phone Ports 16

Configuring FXO Line Ports on the SPA8800 17

Configuring FXO Line Dial Plans for inbound PSTN call Routing 19

Testing the Phone System 20

T.38 Faxing 21

Codecs 23

Troubleshooting 24

Troubleshooting Rejected Because Extension not Found 25

Failing Inbound from ITSP 25

Failing Inbound from PSTN 25

SPA8800 Debug and syslog 26

SPA8800 SIP Debugging 27

Troubleshooting with Asterisk CLI Commands 29

sip show peers 29

sip show peer <PeerName> 29

sip show users 31

sip show user <UserName> 31

Sample Traces 31

Trace of Asterisk Server Registering to ITSP 32

Trace of SPA922 Registering to an Asterisk Server 33

Trace of SPA8800 Phone Ports Registering 36

Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment

Trace of Call between SPA8800 FXS1 and FXS2 42

Trace of SPA8800 FXS Port Calling SPA922 IP Phone 49

Trace of SPA8800 FXS 1 Making Outbound Call 52

Trace of SPA8800 FXS Receiving Inbound Call 60

Trace of FAX Line Toggle Code #99 68

Gathering Information for Support 69

Configuring the Cisco SPA8800 IP Telephony Gateway in an Asterisk Environment

Introduction

The Cisco®SPA8800 IP Telephony Gateway provides four RJ-11 FXS and four FXO ports, a

10/100BASE-T RJ-45 Ethernet interface to connect to either a router or multilayer switch, and an

auxiliary port for local administration. It also provides a single multiport RJ-21 50-pin connector.

The SPA8800 can in only a matter of minutes, be easily be configured as an Asterisk®FXO

gateway.

Calls originating from the public switched telephone network (PSTN) can be terminated by the

SPA8800's FXO ports and routed to analog or IP phones based on an Asterisk server's

configuration. Analog phones connected to the SPA8800 can make low-cost VoIP calls via an

Internet Telephony Service Provider (ITSP) or can make calls via the PSTN.

Many interesting call routing options are possible using Asterisk to control the SPA8800 gateway.

Asterisk can be configured to trunk the SPA8800's four FXO ports together into a trunk group. A

trunk group allows the PSTN lines connected to the FXO ports to be over-subscribed and shared

among all configured analog and IP phones, effectively lowering telephony costs by not requiring a

dedicated line per phone. For example, you can have 4 connected PSTN lines and share them with

any number of phones. When all 4 lines are busy, the 5th user will hear a congestion tone.

Additionally, the SPA8800's FXS ports can be used in other ways, including connecting analog

phones, door phones, and fax machines.

The SPA8800 supports fax with G.711 pass-through or real-time fax over IP via T.38 fax relay and

also supports the G.711 A-law, G.711 µ-law, G.726, G.729A, G.723.1 voice codecs.

In the event that you only need FXS ports and do not need any FXO ports, consider using the

Cisco SPA8000 8-Port IP Telephony Gateway. The SPA8000 configuration is very similar to the

SPA8800.

Audience

This application note is targeted at anyone who needs an FXO gateway for their Asterisk server. It

is expected that readers of this document are familiar with the administration tasks involved with

configuring VoIP in an Asterisk environment.

Scope

This scope of this document is limited to configuring the SPA8800 in an Asterisk environment and

does not address the following topics:

•Installing an Asterisk server

•Advanced Asterisk configuration

•SPA8800 localization

•Security

Refer to the Related Documents for additional configuration and background information.

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 Nports

6. Configure FXO line parameters on the SPA8800 LINE Nports

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 basic

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

[email protected]:mypassword:[email protected]/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)

;

; dial8 as a steering digit:

; 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:

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