ADMINISTRATION
GUIDE
Cisco Small Business Pro
SPA2102, SPA3102, SPA8000, PAP2T, WRP400
Analog Telephone Adapters
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OL-17901-01
About This Document ix
Chapter 1: Introducing Cisco Small Business Analog Telephone Adapters 16
Comparison of ATA Devices 17
ATA Connectivity Requirements 20
PAP2T Connectivity 21
SPA2102 Connectivity 22
SPA3102 Connectivity 23
SPA8000 Connectivity 24
ATA Software Features 25
Voice Supported Codecs 25
SIP Proxy Redundancy 27
Contents
Other ATA Software Features 27
Chapter 2: Basic Administration and Configuration 35
Basic Services and Equipment Required 35
Downloading Firmware 36
Basic Installation and Configuration 36
Upgrading the Firmware for the ATA Device 36
Setting up Your ATA Device 37
Using the Administration Web Server 38
Connecting to the Administration Web Server 39
Setting Up the WAN Configuration for Your ATA Device 39
Registering to the Service Provider 41
Advanced Configurations 42
Upgrading, Rebooting, and Resyncing Your ATA Device 42
Upgrade URL 42
Resync URL 43
ATA Administration Guide i
Reboot URL 44
Provisioning Your ATA Device 44
Provisioning Capabilities 44
Configuration Profile 45
Chapter 3: Configuring Your System for ITSP Interoperability 47
Network Address Translation (NAT) and Voice over IP (VoIP) 47
NAT Mapping with Session Border Controller 48
NAT Mapping with SIP-ALG Router 48
Configuring NAT Mapping with a Static IP Address 48
Configuring NAT Mapping with STUN 50
Determining Whether the Router Uses Symmetric or Asymmetric NAT
52
Contents
Firewalls and SIP 53
Configuring SIP Timer Values 53
Chapter 4: Configuring Voice Services 54
Supported Codecs 54
Using a FAX Machine (SPA2102, SPA3102 or SPA8000) 55
Fax Troubleshooting 57
Managing Caller ID Service 58
Silence Suppression and Comfort Noise Generation 60
Configuring Dial Plans 61
About Dial Plans 61
Editing Dial Plans 70
Secure Call Implementation 72
Enabling Secure Calls 72
Secure Call Details 73
ATA Administration Guide ii
Contents
Using a Mini-Certificate 74
Generating a Mini Certificate 75
SIP Trunking and Hunt Groups on the SPA8000 77
About SIP Trunking 78
Setting the Trunk Group Call Capacity 80
Inbound Call Routing for a Trunk Group 80
Contact List for a Trunk Group 81
Outgoing Call Routing for a Trunk Group 83
Configuring a Trunk Group 84
Trunk Group Management 85
Setting the Hunt Policy 86
Additional Notes About Trunk Groups 87
Chapter 5: Configuring Music on Hold 88
Using the Internal Music Source for Music On Hold 88
Using the Internal Music Source 88
Changing the Music File for the Internal Music Source 89
Configuring a Streaming Audio Server 90
About the Streaming Audio Server 90
Configuring the Streaming Audio Server 92
Using the IVR with an SAS Line 93
Chapter 6: Configuring the PSTN (FXO) Gateway on the SPA3102 94
Connecting to PSTN and VoIP Services 94
How VoIP-To-PSTN Calls Work 95
One-Stage Dialing 95
Two-Stage Dialing 97
How PSTN-To-VoIP Calls Work 98
Terminating Gateway Calls 99
VoIP Outbound Call Routing 101
ATA Administration Guide iii
Configuring VoIP Failover to PSTN 102
Sharing One VoIP Account Between the FXS and PSTN Lines 103
Other Options 104
PSTN Call to Ring Line 1 104
Symmetric RTP 104
Call Progress Tones 105
Call Scenarios 105
PSTN to VoIP Call with and Without Ring-Thru 106
VoIP to PSTN Call With and Without Authentication 106
Call Forwarding to PSTN Gateway 109
Appendix A: ATA Routing Field Reference 111
Contents
Router Status page 111
Product Information section 112
System Status section 112
WAN Setup page 113
Internet Connection Settings section 113
Static IP Settings section 114
PPPoE Settings section 114
Optional Settings section 115
MAC Clone Settings section 116
Remote Management section 116
QOS Settings section 116
VLAN Settings section 117
LAN Setup page 117
Networking Service section 117
LAN Networking Settings section 118
Static DHCP Lease Settings section 118
Application page 118
ATA Administration Guide iv
Port Forwarding Settings section 119
DMZ Settings section 119
Miscellaneous Settings section 120
System Reserved Ports Range section 120
Appendix B: ATA Voice Field Reference 121
Info page 122
Product Information section 122
System Status section 123
Line Status section 123
System Information section (PAP2T) 126
PSTN Line Status section (SPA3102) 126
Trunk Status section (SPA8000) 129
Contents
System page 130
System Configuration section 130
Internet Connection Type section (PAP2T) 131
Optional Network Configuration section (PAP2T) 131
Miscellaneous Settings section (not used with PAP2T) 132
SIP page 133
SIP Parameters section 133
SIP Timer Values (sec) section 135
Response Status Code Handling section 137
RTP Parameters section 138
SDP Payload Types section 140
NAT Support Parameters section 141
Trunking Parameters section (SPA8000) 144
Regional page 145
Call Progress Tones section 146
Distinctive Ring Patterns section 148
Distinctive Call Waiting Tone Patterns section 149
ATA Administration Guide v
Contents
Distinctive Ring/CWT Pattern Names section 150
Ring and Call Waiting Tone Spec section 151
Control Timer Values (sec) section 151
Vertical Service Activation Codes section 153
Vertical Service Announcement Codes section (SPA2102, SPA8000)
159
Outbound Call Codec Selection Codes section 159
Miscellaneous section 161
Line page 165
Line Enable section 166
Streaming Audio Server (SAS) section 166
NAT Settings section 167
Network Settings section 168
SIP Settings section 169
Call Feature Settings section 172
Proxy and Registration section 173
Subscriber Information section 174
Supplementary Service Subscription section 175
Audio Configuration section 178
Gateway Accounts section (SPA3102) 178
VoIP Fallback to PSTN section (SPA3102) 179
Dial Plan section 179
FXS Port Polarity Configuration section 181
Trunk Group page (SPA8000) 181
Line Enable section 182
Network Settings section 182
SIP Settings section 182
Subscriber Information section 186
Dial Plan section 188
NAT Settings section 188
Proxy and Registration section 189
ATA Administration Guide vi
Contents
PSTN Line page (SPA3102) 190
Line Enable section 191
NAT Settings section 191
Network Settings section 192
SIP Settings section 193
Proxy and Registration section 195
Subscriber Information section 197
Audio Configuration section 198
Dial Plans section 201
VoIP-To-PSTN Gateway Setup section 202
VoIP Users and Passwords (HTTP Authentication) section 204
Ring Settings section 205
FXO (PSTN) Timer Values (sec) section 205
PSTN Disconnect Detection section 207
International Control (Settings) section 211
User page 213
Call Forward Settings section 214
Selective Call Forward Settings section 215
Speed Dial Settings section 215
Supplementary Service Settings section 216
Distinctive Ring Settings section 217
Ring Settings section 218
PSTN User page (SPA3102 Only) 219
PSTN-To-VoIP Selective Call Forward Settings section 219
PSTN-To-VoIP Speed Dial Settings section 219
PSTN Ring Thru Line 1 Distinctive Ring Settings section 220
PSTN Ring Thru Line 1 Ring Settings section 220
ATA Administration Guide vii
Appendix C: Provisioning Reference (WRP400) 221
Appendix D: Troubleshooting 235
Appendix E: Environmental Specifications 239
PAP2T 239
SPA2102 240
SPA3102 240
SPA8000 241
WRP400 242
Contents
WRTP54G 242
Appendix F: Where to Go From Here 244
Product Resources 244
Related Documentation 245
Appendix G: Additional Information 247
Appendix H: Support Contacts 248
ATA Administration Guide viii
About This Document
This guide is intended to help VARs and Service Providers to manage and
configure the Cisco Analog Telephone Adapters (ATAs). This preface provides
helpful information about this guide and other resources that are available to you.
Before you begin to use this guide, refer to the following topics:
• “Purpose,” on page ix
• “Audience,” on page ix
• “Firmware,” on page x
• “Organization,” on page xi
Preface
Purpose
Audience
• “Document Conventions,” on page x
• “Finding Information in PDF Files,” on page xiii
This document provides information that administrators can use to configure and
manage Cisco ATAs that are used in conjunction with the SPA9000 Voice System.
This document is written for the following audience:
• Service providers offering services using LVS products
• VARs and resellers who need LVS configuration references
• System administrators or anyone who performs LVS installation and
administration
NOTE This guide does not provide the configuration information required by specific
service providers. Please consult with the service provider for specific service
parameters.
ATA Administration Guide ix
Firmware
Preface
This guide describes the features that are available in the following firmware
releases.
Product Firmware Version
PAP2T 5.1.6
SPA2102 5.2.5
SPA3102 5.1.7
SPA8000 6.1.3
WRP400 1.00.06
Document Conventions
The following are the typographic conventions used in this document.
Typographic
Element
Boldface
Italic
Meaning
May indicate either of the following:
• A user interface element that you need to click, select, or
otherwise act on
• A literal value to be entered in a field.
May indicate either of the following:
• A variable that should be replaced with a literal value.
• The name of a page, section, or field in the user interface
Monospaced
Font
ATA Administration Guide x
Indicates code samples or system output.
Organization
Preface
The information in this guide is organized into the following chapters and
appendices:
Chapter Contents
Chapter 1, “Introducing
Cisco Small Business
Analog Telephone
Adapters”
Chapter 2, “Basic
Administration and
Configuration”
Chapter 3, “Configuring
You r Sy stem for ITS P
Interoperability”
Chapter 4, “Configuring
Voice Services”
Chapter 5, “Configuring
Music on Hold”
Chapter 6, “Configuring
the PSTN (FXO) Gateway
on the SPA3102”
This chapter introduces the functionality of the ATA
devices and describes the features that are
available.
This chapter describes the equipment and
services that are required to install your ATA device
and explains how to complete the basic
administration and configuration tasks.
This chapter provides configuration details to help
you to ensure that your infrastructure properly
supports voice services.
This chapter describes how to configure your ATA
device to meet the customer’s requirements for
voice services.
This chapter explains how to configure Music on
Hold using either a music file or streaming audio.
This chapter describes how to configure the
Linksys SPA3102 and AG310 devices to provide
PSTN connectivity.
Appendix A, “ATA Routing
Field Reference”
Appendix B, “ATA Voice
Field Reference”
Appendix C, “Provisioning
Reference (WRP400)”
ATA Administration Guide xi
This chapter describes the settings that you can
configure under the Router and Network tabs in the
administration web server pages.
This chapter describes the settings that you can
configure under the Voice tab in the administration
web server pages.
This chapter provides information about the
parameters that can be provisioned from an XML
profile by using the profile compiler tool (SPC).
Chapter Contents
Preface
Appendix D,
“Troubleshooting”
Appendix F, “Where to Go
From Here”
Appendix G, “Additional
Information”
Appendix H, “Support
Contacts”
This appendix provides solutions to problems that
may occur during the installation and operation of
the ATA devices.
These appendices provide information about other
resources that may be useful to you.
ATA Administration Guide xii
Finding Information in PDF Files
The SPA9000 Voice System documents are published as PDF files. The PDF Find/
Search tool within Adobe® Reader® lets you find information quickly and easily
online. You can perform the following tasks:
• Search an individual PDF file.
• Search multiple PDF files at once (for example, all PDFs in a specific folder or
disk drive).
• Perform advanced searches.
Finding Text in a PDF
Follow this procedure to find text in a PDF file.
STEP 1 Enter your search terms in the Find text box on the toolbar.
Preface
NOTE By default, the Find tool is available at the right end of the Acrobat toolbar. If
the Find tool does not appear, choose Edit > Find.
STEP 2 Optionally, click the arrow next to the Find text box to refine your search by
choosing special options such as Whole Words Only.
STEP 3 Press Enter.
STEP 4 Acrobat displays the first instance of the search term.
STEP 5 Press Enter again to continue to more instances of the term.
ATA Administration Guide xiii
Finding Text in Multiple PDF Files
The
Search
on your PC or local network. The PDF files do not need to be open.
STEP 1 Start Acrobat Professional or Adobe Reader.
window lets you search for terms in multiple PDF files that are stored
Preface
STEP 2 Choose Edit > Search, or click the arrow next to the
Open Full Acrobat Search.
STEP 3 In the
a. Enter the text that you want to find.
b. Choose All PDF Documents in.
c. If you want to specify additional search criteria, click Use Advanced Search
d. Click Search.
Search
From the drop-down box, choose Browse for Location. Then choose the
location on your computer or local network, and click OK.
Options, and choose the options you want.
window, complete the following steps:
Find
box and then choose
ATA Administration Guide xiv
Preface
STEP 4 When the Results appear, click + to open a folder, and then click any link to open
the file where the search terms appear.
For more information about the Find and Search functions, see the Adobe Acrobat
online help.
ATA Administration Guide xv
Introducing Cisco Small Business Analog Telephone Adapters
This guide describes the administration and use of Cisco Small Business analog
telephone adapters (ATAs). These ATA devices are a key element in the end-toend IP Telephony solution. An ATA device provides user access to Internet phone
services through one or more standard telephone RJ-11 phone ports using
standard analog telephone equipment. The ATA device connects to a wide area IP
network, such as the Internet, through a broadband (DSL or cable) modem or
router.
1
V
V
Voice
gateway
PSTN
Layer 3
Telephone/fax
This chapter introduces the functionality of the ATA devices and describes the
features that are available.
Refer to the following topics:
V
Linksys ATA
Ethernet
Broadband CPE
(DSL, cable,
fixed wireless)
Broadband
IP infrastructure
SIP proxy
• “Comparison of ATA Devices,” on page17
• “ATA Connectivity Requirements,” on page 20
• “ATA Software Features,” on page 25
187254
ATA Administration Guide 16
Introducing Cisco Small Business Analog Telephone Adapters
Comparison of ATA Devices
Comparison of ATA Devices
Each ATA device is an intelligent low-density Voice over IP (VoIP) gateway that
enables carrier-class residential and business IP Telephony services delivered
over broadband or high-speed Internet connections. An ATA device maintains the
state of each call it terminates and makes the proper reaction to user input events
(such as on/off hook or hook flash). The ATA devices use the Session Initiation
Protocol (SIP) open standard so there is little or no involvement by a “middle-man”
server or media gateway controller. SIP allows interoperation with all ITSPs that
support SIP.
The following table summarizes the ports and features provided by the ATA
devices described in this document.
1
Product
Name
PAP2T 2 — 1 — 2 Voice adapter with
SPA2102 2 — 1 1 2 Voice adapter with
SPA3102 1 1 1 1 1 Voice adapter with
SPA8000 8 — 1 Mainte-
WRP400 2 — 1 4 2 Wireless-G IP router
FXS
(Analog
Phone)
FXO
PSTN
RJ-45
Internet
(WAN)
RJ-45
Ethernet
(LAN)
nance
only
Voice
Lines
8 Voice adapter with
Description
two FXS ports.
router.
router and PSTN
connectivity.
support for up to
eight FXS devices.
Supports SIP
Trunking for inbound
call routing to trunk
groups.
with two FXS ports.
Provides ATA device
functionality. Can be
remotely
provisioned.
WRTP54G 2 — 1 4 2 Wireless-G IP router
with two FXS ports.
Provides ATA device
functionality.
ATA Administration Guide 17
Introducing Cisco Small Business Analog Telephone Adapters
Comparison of ATA Devices
NOTE The information contained in this guide is not a warranty from Cisco. Customers
planning to use ATA devices in a VoIP service deployment are advised to test all
functionality they plan to support before putting the ATA device in service. By
implementing ATA devices with the SIP protocol, intelligent endpoints at the edges
of a network perform the bulk of the call processing. This allows the deployment of
a large network with thousands of subscribers without complicated, expensive
servers.
The following figure illustrates how the different ATA devices provide voice
connectivity in a VoIP network.
1
ATA Administration Guide 18
Introducing Cisco Small Business Analog Telephone Adapters
SPA3102
Broadband
router
Broadband
router
SPA8000,
PAP2T
DSL/cable
modem
WRP400,
WRTP54G,
and SPA2102
Ethernet/Wireless
LAN
Fax (up to 4
SPA8000)
Analog phone
(up to 8 with
SPA8000)
PSTN
Ethernet/Wired
LAN
Internet
187255-revised
Comparison of ATA Devices
Figure1 How ATAs Provide Voice Connectivity
1
• The SPA3102 and SPA8000 act as SIP-PSTN gateways. They provide PSTN
connectivity in addition to a single FXS port.
• The WRP400 and WRTP54G routers provide ports for analog telephone
devices and provide QoS in the form of priority packet queueing.
ATA Administration Guide 19
Introducing Cisco Small Business Analog Telephone Adapters
ATA Connectivity Requirements
ATA Connectivity Requirements
An ATA device can be connected to a local router, or directly to the Internet. Each
phone connected to an RJ-11 (analog) port on the ATA device connects to other
devices through SIP, which is transmitted over the IP network.
In order to ensure connectivity between the devices connected to its FXS ports,
the ATA device requires the following functionality to be supplied on the network
connected to its Ethernet port:
• Connection to an IP router with hairpinning support
• Connection to an outbound Proxy server
When a phone connected to the ATA device communicates with another phone, it
sends a SIP packet onto the internal LAN. The packet is then forwarded to the
external LAN or directly to the Internet. The source address and source port on the
original packet are assigned by the ATA device DHCP server. The address and
port are translated by the ATA device using Network Address Translation (NAT)
and Port Address Translation (PAT). The packet is then routed back to the internal
network on the ATA device by the local router or the ISP router.
1
Problems can occur with calls between phones connected to the ATA device
when an outbound proxy or a router with hairpinning support is not available. The
ATA device cannot directly connect the two telephone devices, but requires a
local or remote router to route the packet back to its destination on the local
network from which it originated.
The necessary routing can be provided by a router with hairpinning support, or by
an outbound SIP proxy, which is typically provided by the Internet Telephony
Service Provider (ITSP). When relying on the ITSP for interconnecting phones on
the ATA device, local phones connected to the ATA device are unable to
communicate with each other if the Internet connection is not available for any
reason. It is recommended you connect the ATA device to a local router that
provides hairpinning support to prevent this problem.
ATA Administration Guide 20
Introducing Cisco Small Business Analog Telephone Adapters
Line 1
Line 2
Internet
IP Router (with
hairpinning) or
Broadband mode
m
ITSP
ISP
PAP2T
LAN WAN
Ethernet
port
Administrative
IVR (Line 1 or
Line 2)
IP
IP
ATA Connectivity Requirements
PAP2T Connectivity
As shown in the following figure, the PAP2T has two FXS ports (voice lines 1 and
2).
1
NOTE
• The IVR functions are accessed by connecting an analog telephone to Line 1.
• For proper operation, the service provider should use an Outbound Proxy to
forward all voice traffic when the PAP2T is located behind a router. If
necessary, explicit port ranges can be specified for SIP and RTP.
ATA Administration Guide 21
Introducing Cisco Small Business Analog Telephone Adapters
ATA Connectivity Requirements
SPA2102 Connectivity
As shown in the following illustration, the SPA2102 has two FXS ports (voice lines
1 and 2).
Administrative
IVR (Line 1 or
Line 2)
Ethernet
Line 1
Line 2
LAN
2102
SPA
port
port
IP Router (with
hairpinning) or
Broadband mode
LAN
m
ISP
WAN
Administration
PC
Internet
IP
IP
ITSP
1
187257
NOTE
By default, the device attached to the LAN port is assigned the network address
192.168.0.0 with a subnet mask of 255.255.255.0. If there is a network address
conflict with a device on the Ethernet port, the network address of the device on
the LAN port is automatically changed to 192.168.1.0.
• The IVR functions are accessed by connecting an analog telephone to Line 1.
• For proper operation, the service provider should use an Outbound Proxy to
forward all voice traffic when the SPA2102 is located behind a router. If
necessary, explicit port ranges can be specified for SIP and RTP.
ATA Administration Guide 22
Introducing Cisco Small Business Analog Telephone Adapters
Line 1
PSTN
Line 1
Internet
IP Router (with
hairpinning) or
Broadband mode
m
ITSP
ISP
SPA
3102
Ethernet
port
LAN
port
LAN WAN
Administrative
IVR (Line 1 or
Line 2)
IP
IP
Administration
PC
187259
PSTN
ATA Connectivity Requirements
SPA3102 Connectivity
As shown in the following figure, the SPA3102 has one FXS port (voice line 1).
1
NOTE
By default, the device on the LAN port is assigned the network address
192.168.0.0 with a subnet mask of 255.255.255.0. If there is a network address
conflict with a device on the Ethernet port, the network address of the device on
the LAN port is automatically changed to 192.168.1.0.
• The IVR functions are accessed by connecting an analog telephone to Line 1.
• For proper operation, the service provider should use an Outbound Proxy to
forward all voice traffic when the SPA3102 is located behind a router. If
necessary, explicit port ranges can be specified for SIP and RTP.
ATA Administration Guide 23
Introducing Cisco Small Business Analog Telephone Adapters
Line 1
Line 2
Internet
IP Router (with
hairpinning) or
Broadband modem
ITSP
ISP
SPA800
0
Line 4
Line 3
Line 6
Line 5
Line 8
Line 7
NAT/PAT
Internal DHCP
server
LAN WAN
Ethernet
port
AUX
port
Administrative
IVR (Line 1 or
Line 2
)
IP
IP
8 FXS (RJ-11/RJ-21 ) ports
Administration
PC
ATA Connectivity Requirements
SPA8000 Connectivity
1
As shown in the following illustration, the SPA8000 consists of eight voice ports
(voice lines 1-8).
ATA Administration Guide 24
By default, the device on the AUX port is assigned the network address
192.168.0.0 with a subnet mask of 255.255.255.0. If there is a network address
conflict with a device on the Ethernet port, the network address of the device on
the AUX port is automatically changed to 192.168.1.0.
In the illustration, one fax machine is connected to each pair of ports to illustrate
that only one T.38 connection is supported by each of the four pairs of RJ-11 ports.
Up to four fax machines can be connected to the SPA8000 router, but they must be
distributed as shown.
Introducing Cisco Small Business Analog Telephone Adapters
ATA S of t w ar e F ea tur es
NOTE
• With the SPA8000, use line 1 or line 2 to access the IVR functions. See the
SPA8000 Quick Installation Guide for IVR instructions.
• For proper operation, the service provider should use an Outbound Proxy to
forward all voice traffic when the SPA8000 is located behind a router. If
necessary, explicit port ranges can be specified for SIP and RTP.
• The SPA8000 is not designed to forward IP packets to devices connected to its
AUX port and that configuration is not supported.
• The SPA8000 also can be configured with trunk groups and trunk lines. See
“SIP Trunking and Hunt Groups on the SPA8000,” on page 77.
1
ATA Software Features
The ATA device is a full featured, fully programmable phone adapter that can be
custom provisioned within a wide range of configuration parameters. This section
contains a high-level overview of features to provide a basic understanding of the
feature breadth and capabilities of the ATA device.
The following sections describe the factors that contribute to voice quality:
• “Voice Supported Codecs,” on page 25
• “SIP Proxy Redundancy,” on page 27
• “Other ATA Software Features,” on page 27
Voice Supported Codecs
Negotiation of the optimal voice codec sometimes depends on the ability of the
ATA device to match a codec name with the codec used by the far-end device.
The ATA device allows the network administrator to individually name the various
codecs that are supported so that the ATA device can successfully negotiate the
codec with the far-end equipment. The administrator can select which low-bit-rate
ATA Administration Guide 25
Introducing Cisco Small Business Analog Telephone Adapters
ATA S of t w ar e F ea tur es
codec is to be used for each line. G.711a and G.711u are always enabled.
Configure your preferred codec in the (FXS) tab in the Administration Web Server.
See “ATA Voice Field Reference,” on page121. See also “Supported Codecs,” on
page 54 for a list of which codecs are supported on each ATA device.
1
Codec (Voice Compression
Algorithm)
G.711 (A-law and mμ-law) This very low complexity codec supports
G.726 This low complexity codec supports compressed 16,
G.729a The ITU G.729 voice coding algorithm is used to
G.723.1 The ATA device supports the use of ITU G.723.1 audio
Description
uncompressed 64 kbps digitized voice transmission at
one through ten 5 ms voice frames per packet. This
codec provides the highest voice quality and uses the
most bandwidth of any of the available codecs.
24, 32, and 40 kbps digitized voice transmission at one
through ten 10 ms voice frames per packet. This codec
provides high voice quality.
compress digitized speech. Cisco supports G.729.
G.729a is a reduced complexity version of G.729. It
requires about half the processing power to code
G.729. The G.729 and G.729a bit streams are
compatible and interoperable, but not identical.
codec at 6.4 kbps. Up to two channels of G.723.1 can be
used simultaneously. For example, Line 1 and Line 2 can
be using G.723.1 simultaneously, or Line 1 or Line 2 can
initiate a three-way conference with both call legs using
G.723.1.
NOTE: The WRP400 device does not support the
G.723.1 audio codec.
NOTE When no static payload value is assigned per RFC 1890, the ATA device can
support dynamic payloads for G.726.
ATA Administration Guide 26
Introducing Cisco Small Business Analog Telephone Adapters
ATA S of t w ar e F ea tur es
SIP Proxy Redundancy
In typical commercial IP Telephony deployments, all calls are established through
a SIP proxy server. An average SIP proxy server may handle thousands of
subscribers. It is important that a backup server be available so that an active
server can be temporarily switched out for maintenance. The ATA device supports
the use of backup SIP proxy servers (via DNS SRV) so that service disruption
should be nearly eliminated.
A relatively simple way to support proxy redundancy is to configure your DNS
server with a list of SIP proxy addresses. The ATA device can be instructed to
contact a SIP proxy server in a domain named in the SIP message. The ATA device
consults the DNS server to get a list of hosts in the given domain that provides SIP
services. If an entry exists, the DNS server returns an SRV record that contains a
list of SIP proxy servers for the domain, with their host names, priority, listening
ports, and so on. The ATA device tries to contact the list of hosts in the order of
their stated priority.
1
If the ATA device is currently using a lower priority proxy server, it periodically
probes the higher priority proxy to see whether it is back on line, and switches
back to the higher priority proxy when possible. SIP Proxy Redundancy is
configured in the Line and PSTN Line tabs in the Administration Web Server. See
“ATA Routing Field Reference,” on page111.
Other ATA Software Features
The following table summarizes other features provided by ATA devices.
Feature Description
Streaming Audio
Server
T.38 Fax Relay See “Using a FAX Machine (SPA2102, SPA3102 or
Silence
Suppression
See “Configuring a Streaming Audio Server,” on page 90.
SPA8000),” on page 55.
See “Silence Suppression and Comfort Noise
Generation,” on page 60.
ATA Administration Guide 27
Introducing Cisco Small Business Analog Telephone Adapters
ATA S of t w ar e F ea tur es
Feature Description
1
Modem and Fax
Pass-Through
Adaptive Jitter
Buffer
• Modem pass-through mode can be triggered only by
predialing the number set in the
(Set in the Regional tab.)
Modem Line Toggle Code.
• FAX pass-through mode is triggered by a CED/CNG tone or
an NSE event.
• Echo canceller is automatically disabled for Modem pass-
through mode.
• Echo canceller is disabled for FAX pass-through if the
parameter
for that line (in that case FAX pass-through is the same as
Modem pass-through).
FAX Di sa ble ECAN
(Line 1 or 2 tab) is set to “yes”
• Call waiting and silence suppression is automatically
disabled for both FAX and Modem pass-through. In addition,
out-of-band DTMF Tx is disabled during modem or fax passthrough.
The ATA device can buffer incoming voice packets to
minimize out-of-order packet arrival. This process is
known as jitter buffering. The jitter buffer size proactively
adjusts or adapts in size, depending on changing network
conditions.
The ATA device has a Network Jitter Level control setting
for each line of service. The jitter level determines how
aggressively the ATA device tries to shrink the jitter buffer
over time to achieve a lower overall delay. If the jitter level
is higher, it shrinks more gradually. If jitter level is lower, it
shrinks more quickly.
Adaptive Jitter Buffer is configured in the Line and PSTN
Line tabs. See “ATA Voice Field Reference,” on page121.
International Caller
ID Delivery
Secure Calls A user (if enabled by service provider or administrator)
In addition to support of the Bellcore (FSK) and Swedish/
Danish (DTMF) methods of Caller ID (CID) delivery, ATAs
provide a large subset of ETSI-compliant methods to
support international CID equipment. International CID is
configured in the Line and PSTN Line tabs. See “ATA Voice
Field Reference,” on page121.
has the option to make an outbound call secure in the
sense that the audio packets in both directions are
encrypted. See ”Secure Call Implementation” section on
page 72.
ATA Administration Guide 28
Introducing Cisco Small Business Analog Telephone Adapters
ATA S of t w ar e F ea tur es
Feature Description
1
Adjustable Audio
Frames Per Packet
DTMF The ATA device may relay DTMF digits as out-of-band
Call Progress Tone
Generation
This feature allows the user to set the number of audio
frames contained in one RTP packet. Packets can be
adjusted to contain from 1–10 audio frames. Increasing the
number of packets decreases the bandwidth utilized, but
it also increases delay and may affect voice quality. See
the RTP Packet Size parameter found in the SIP tab in the
“ATA Voice Field Reference,” on page121.
events to preserve the fidelity of the digits. This can
enhance the reliability of DTMF transmission required by
many IVR applications such as dial-up banking and airline
information. DTMF is configured in the
parameter found in the Line tabs. See the “ATA Voice Field
Reference,” on page121.
The ATA device has configurable call progress tones. Call
progress tones are generated locally on the ATA device so
an end user is advised of status (such as ringback).
Parameters for each type of tone (for instance a dial tone
played back to an end user) may include frequency and
amplitude of each component, and cadence information.
See the Regional tab in the “ATA Voice Field Reference,”
on page121.
DTMF Tx Mode
Call Progress Tone
Pass Through
Echo Cancellation Impedance mismatch between the telephone and the IP
This feature allows the user to hear the call progress tones
(such as ringing) that are generated from the far-end
network. See the Regional tab in the “ATA Voice Field
Reference,” on page121.
Telephony gateway phone port can lead to near-end echo.
The ATA device has a near-end echo canceller that
compensates for impedance match. The ATA device also
implements an echo suppressor with comfort noise
generator (CNG) so that any residual echo is not
noticeable. Echo Cancellation is configured in the
Regional, Line, and PSTN Line tabs. See “ATA Voice Field
Reference,” on page121.
ATA Administration Guide 29
Introducing Cisco Small Business Analog Telephone Adapters
ATA S of t w ar e F ea tur es
Feature Description
1
Signaling Hook
Flash Event
Configurable Dial
Plan with Interdigit
Time rs
The ATA device can signal hook flash events to the remote
party on a connected call. This feature can be used to
provide advanced mid-call services with third-party-callcontrol. Depending on the features that the service
provider offers using third-party-call-control, the following
ATA features may be disabled to correctly signal a hookflash event to the softswitch:
• Call Waiting Service (parameter
Line tab)
• Three Way Conference Service (parameter
set in the Line tab)
serv
• Three Way Call Service (parameter
in the Line tab)
You can configure the length of time allowed for detection
of a hook flash using the Hook Flash Timer parameter on
the Regional tab of the administration web server. See
“ATA Voice Field Reference,” on page121.
The ATA device has three configurable interdigit timers:
Initial timeout (T)—Signals that the handset is off the hook
and that no digit has been pressed yet.
call waiting serv
three-way conf
three-way call serv
set in the
set
Long timeout (L)—Signals the end of a dial string; that is,
no more digits are expected.
Short timeout (S)—Used between digits; that is after a
digit is pressed a short timeout prevents the digit from
being recognized a second time.
See “Configuring Dial Plans,” on page 61 for more
information.
Polarity Control The ATA device allows the polarity to be set when a call is
connected and when a call is disconnected. This feature is
required to support some pay phone system and
answering machines. Polarity Control is configured in the
Line and PSTN Line tabs. See “ATA Voice Field Reference,”
on page121.
ATA Administration Guide 30
Introducing Cisco Small Business Analog Telephone Adapters
ATA S of t w ar e F ea tur es
Feature Description
1
Calling Party
Control
Report Generation
and Event Logging
Syslog and Debug
Server Records
Calling Party Control (CPC) signals to the called party
equipment that the calling party has hung up during a
connected call by removing the voltage between the tip
and ring momentarily. This feature is useful for autoanswer equipment, which then knows when to disengage.
CPC is configured in the Regional, Line, and PSTN Line
tabs. See “ATA Voice Field Reference,” on page121.
The ATA device reports a variety of status and error
reports to assist service providers to diagnose problems
and evaluate the performance of their services. The
information can be queried by an authorized agent, using
HTTP with digested authentication, for instance. The
information may be organized as an XML page or HTML
page. Report Generation and Event Logging are
configured in the System, Line, and PSTN Line tabs. See
“ATA Voice Field Reference,” on page121.
Syslog and Debug Sever Records log more details than
Report Generation and Event Logging. Using the
configuration parameters, the ATA device allows you to
select which type of activity/events should be logged.
Syslog and Debug Server allow the information captured
to be sent to a Syslog Server. Syslog and Debug Server
Records are configured in the System, Line, and PSTN
Line tabs. See “ATA Voice Field Reference,” on page121.
SIP Over TCP To guarantee state-oriented communications, SPA2102
and SPA3102 devices allow you to choose TCP as the
transport protocol for SIP. This protocol is “guaranteed
delivery”, which assures that lost packets are
retransmitted. TCP also guarantees that the SIP packages
are received in the same order that they were sent. As a
result, TCP overcomes the main disadvantages of UDP. In
addition, for security reasons, most corporate firewalls
block UDP ports. With TCP, new ports do not need to be
opened or packets dropped, because TCP is already in
use for basic activities such as Internet browsing or ecommerce. SIP over TCP is configured in the Line tabs.
See “ATA Voice Field Reference,” on page121.
ATA Administration Guide 31
Introducing Cisco Small Business Analog Telephone Adapters
ATA S of t w ar e F ea tur es
Feature Description
SIP Over TLS SPA2102, SPA3102, and WRP400 devices allow the use
of SIP over Transport Layer Security (TLS). SIP over TLS is
designed to eliminate the possibility of malicious activity
by encrypting the SIP messages of the service provider
and the end user. SIP over TLS relies on the widelydeployed and standardized TLS protocol. SIP Over TLS
encrypts only the signaling messages and not the media.
A separate secure protocol such as Secure Real-Time
Transport Protocol (SRTP) can be used to encrypt voice
packets. SIP over TLS is configured in the SIP Transport
parameter configured in the Line tab(s). See “ATA Voice
Field Reference,” on page121.
Media Loopback SPA2102, SPA3102, and PAP2T devices allow service
providers to use media loopback to quantitatively and
qualitatively measure the voice quality experienced by the
end user. One device acts as the audio transmitter and
receiver while the other device acts as the audio mirror.
The audio mirror transmits the audio packets that it
receives back to the transmitter/receiver instead of
transmitting the data sampled on its local microphone (IP
phone) or attached analog telephone (ATA-type device).
Media loopback is configured in the User tab. See “ATA
Voice Field Reference,” on page121.
1
ATA Administration Guide 32
Introducing Cisco Small Business Analog Telephone Adapters
ATA S of t w ar e F ea tur es
Feature Description
1
Register Retry
Enhancements
The Register Retry Enhancements feature for SPA2102,
SPA3102, and PAP2T devices adds flexibility to the delay
timers that are activated when the SIP REGISTER of a
device fails. Once a SIP REGISTER failure response code
is sent, a delay timer is selected depending on the type of
registration failure response code. The delay timers can
be one of the following:
• Reg Retry Random Delay—Random delay range (in
seconds) to add to the
retrying a SIP REGISTER after a failure. The default is 0,
which disables this feature.
Register Retry Intvl
parameter when
• Reg Retry Long Random Delay—Random delay range (in
seconds) to add to the
when retrying a SIP REGISTER after a failure. The default is
0, which disables this feature.
Register Retry Long Intvl
parameter
• Reg Retry Intvl Cap—The maximum value to cap the
exponential back-off retry delay. The exponential back-off
retry delay starts with the setting found in the
parameter and doubles it on every REGISTER retry after
Intvl
a failure. In other words, the retry interval after a failure is
always set to the seconds configured in the
parameter. If this feature is enabled, the
Intvl
Random Delay
back-off adjusted delay value. The default value is 0, which
disables the exponential back-off feature.
setting is added on top of the exponential
Register Retry
Register Retry
Reg Retry
Register Retry is configured in the SIP tab. See “ATA Voic e
Field Reference,” on page121.
ATA Administration Guide 33
Introducing Cisco Small Business Analog Telephone Adapters
ATA S of t w ar e F ea tur es
Feature Description
1
DHCP Renewal on
Time ou t
SPA2102, SPA3102, and PAP2T voice devices typically
operate in a network where a DHCP server assigns IP
addresses to the devices. Because IP addresses are a
limited resource, the DHCP server periodically renews the
device lease on the IP address. Therefore, if an ATA device
loses its IP address for any reason, or if some other device
on the network is assigned its IP address, the
communication between the SIP proxy and the device is
either severed or degraded.
Whenever an expected SIP response is not received
within a programmable amount of time after the
corresponding SIP command is sent, the DHCP Renewal
on Timeout feature automatically causes the device to
request a renewal of its IP address. If the DHCP server
returns the IP address that it originally assigned to the
device, the ATA device is presumed to be operating
correctly. If it returns a different address, the ATA device
changes its IP address to the new address provided by
the DHCP server. The ATA device then resets, and once
again sends a SIP register request for the DHCP server to
accept.
ATA Administration Guide 34
Basic Administration and Configuration
This chapter describes the equipment and services that are required to install
your ATA device and explains how to complete the basic administration and
configuration tasks.
Refer to the following topics:
• ”Basic Services and Equipment Required” section on page 35
2
• ”Downloading Firmware” section on page 36
• ”Basic Installation and Configuration” section on page 36
• ”Upgrading the Firmware for the ATA Device” section on page 36
• ”Setting up Your ATA Device” section on page 37
• ”Using the Administration Web Server” section on page 38
• ”Upgrading, Rebooting, and Resyncing Your ATA Device” section on page 42
• ”Provisioning Your ATA Device” section on page 44
Basic Services and Equipment Required
To configure your ATA devices, you need the following services and equipment:
• An integrated access device or modem for broadband access to the Internet
• Internet Telephony Service Provider (ITSP) for Voice Over IP Telephone service
• You must have to following information about your account:
• SIP Proxy (IP address or name)
• Account information and Password
• Computer with Microsoft Windows XP or Windows Vista (for system
configuration)
ATA Administration Guide 35
Basic Administration and Configuration
Downloading Firmware
• Analog phones
• UPS (uninterruptible Power Source) recommended for devices such as the
Integrated Access Device, network switch, router, and PoE switch to ensure
that your phone system continues to work during a power failure, just like your
home phone.
Downloading Firmware
Always download and install the latest firmware for your ATA device before doing
any configurations. You can find the latest firmware at www.cisco.com/go/
smallbiz.
2
Basic Installation and Configuration
See your the Quick Installation Guide and the User Guide the ATA model that you
are installing. If you are configuring the complete SPA9000 Voice System, also
refer to the documentation for the SPA9000 Voice System.
Upgrading the Firmware for the ATA Device
In this procedure, you install the firmware files that you downloaded previously.
STEP 1 Determine the address of the ATA device:
a. Connect an analog telephone to the Phone 1 or Phone 2 port on the ATA
device.
b. Press **** on the keypad to access the IVR menu.
c. Press 110# to determine the Internet (WAN) IP address.
STEP 2 Make a note of the IP address that is announced.
NOTE If the administration computer is connected to the Ethernet port of the ATA
device, the default IP address is 192.168.0.1.
ATA Administration Guide 36
Basic Administration and Configuration
Setting up Your ATA Device
STEP 3 Use the administration computer to install the latest firmware:
a. Extract the Zip file, and then run the executable file to upgrade the firmware.
2
b. When the
c. In the next window that appears, enter the IP address of the ATA device, and
then click OK.
d. In the
product number appear. Then click Upgrade.
e. A progress message appears while the upgrade is in progress. The success
window appears when the upgrade is completed. The device reboots.
f. Click OK to close the confirmation message.
g. To verify the upgrade, point the web browser to the IP address of the ATA
device. Check the
show the firmware version that you installed.
NOTE You may need to refresh your browser to display the updated page
reflecting the new version number.
Firmware Upgrade Warning
Confirm Upgrade
window, verify that the correct device information and
Router > Status
window appears, click Continue.
page. The
Software Version
field should
Setting up Your ATA Device
After installation and basic configuration of your ATA device, you will use the
administration web server to finish your configuration.
ATA devices support two levels of administration privileges: Administrator and
User. Both privileges can be password protected.
NOTE By default, there are no passwords assigned for either the Administrator account or
the User account.
ATA Administration Guide 37
Basic Administration and Configuration
Using the Administration Web Server
The Administrator account can modify all the web profile parameters and the
passwords of both Administrator and User account. The User account can access
only part of the web profile parameters. The parameters that the User account can
access are specified using the Administrator account on the Provisioning page of
the administration web server.
To directly access the Administrator account level privilege, use the following URL:
http://<ipaddress>/admin/voice
If the password has been set for the Administrator account, the browser prompts
for authentication. The User account name and the Administrator account name
cannot be changed.
When browsing pages with the Administrator account privilege, you can switch to
User account privilege by clicking the User Login link.
If the User account password is set, the browser prompts for authentication when
you click the User Login link. From the User account, you can switch to the
Administrator account by clicking the Admin Login link. Authentication is required
if the Administrator account password has been set.
2
NOTE Switching between User and Administrator accounts or between basic and
advanced views discards any uncommitted changes on the web pages.
Using the Administration Web Server
This section describes how to use the administration web server to configure the
advanced settings of the ATA device. It includes the following topics:
• ”Connecting to the Administration Web Server” section on page 39
• ”Setting Up the WAN Configuration for Your ATA Device” section on page 39
• ”Registering to the Service Provider” section on page 41
• ”Advanced Configurations” section on page 42
ATA Administration Guide 38
Basic Administration and Configuration
Using the Administration Web Server
Connecting to the Administration Web Server
To access the ATA administration web server, perform the following steps.
STEP 1 Start Internet Explorer on a computer that is connected to the same network as the
ATA d ev ic e.
STEP 2 Determine the address of the ATA device.
a. Connect an analog telephone to the Phone 1 port of the ATA device.
b. Press **** on the keypad to access the IVR menu.
c. Press 110# to determine the Internet (WAN) IP address.
2
NOTE For more information on the IVR menu, see your Quick Installation Guide or
User Guide for your device, or the LVS Administration Guide.
STEP 3 Direct the browser to the IP address of the ATA device.
STEP 4 The
Router > Status
on to the administrator view by clicking Admin Login, near the top right corner of
the page. Then click Advanced.
NOTE By default, no password is required. You can assign an administrative
password later, but it is convenient not to use a password during the initial
configuration.
page appears. By default, the page is in Basic User mode. Log
Setting Up the WAN Configuration for Your ATA Device
STEP 1 Start Internet Explorer, connect to the administration web server, and choose
Admin access with Advanced settings.
STEP 2 Click Network tab > WAN Setup.
ATA Administration Guide 39
Basic Administration and Configuration
Using the Administration Web Server
STEP 3 Complete the WAN configuration for DHCP, static IP addressing, or PPPoE.
For DHCP:
2
a. Select DHCP from the
b. If you use a cable modem, you may need to configure the MAC Clone Settings.
(Contact your ISP for more information.)
c. If your service uses a specific PC MAC address, then select yes from the
Enable MAC Clone Service
d. Then enter the PC’s MAC address in the
For Static IP Addressing:
a. Select Static IP from the
b. In the Static IP Settings section, enter the IP address in the
subnet mask in the
Gateway
c. In the Optional Settings section, enter the DNS server address(es) in the
Primary DNS
For PPPoE:
a. Select PPPoE from the
setting for most DSL users.
field.
and optional
Connection Type
setting.
Connection Type
NetMask
field, and the default gateway IP address in the
Secondary DNS
Connection Type
drop-down menu.
Cloned MAC Address
drop-down menu.
fields.
drop-down menu. This is the correct
field.
Static IP
field, the
b. Enter the values provided by the ITSP in the following fields:
• PPPoE Login Name
• PPPoE Login Password
• PPPoE Service Name
STEP 4 Click Submit All Changes. The ATA device reboots.
STEP 5 To verify your progress, click the Router tab and then click Status. Under
Status
Gateway
ATA Administration Guide 40
, confirm the
, and
Primary DNS
WAN Connection Type, Current IP, Current Netmask, Current
.
System
Basic Administration and Configuration
Using the Administration Web Server
Registering to the Service Provider
To use VoIP phone service, you must configure your ATA device to the Service
Provider.
STEP 1 Start Internet Explorer, connect to the administration web server, and choose
Admin access with Advanced settings.
2
STEP 2 Click Voice tab > Line
STEP 3 Enter the account information for your ITSP. The following is the minimum required
configuration to connect the ATA device to an ITSP:
N
, where N is the line number that you want to configure.
• User ID: The account number or logon name for your ITSP account (Subscriber
Information section)
• Password: The password for your ITSP account (Subscriber Information
section)
• Proxy: The proxy server for your ITSP account (Proxy and Registration section)
STEP 4 After making any necessary changes, click the Submit All Changes button.
STEP 5 To verify your progress, perform the following tasks:
• After the devices reboot, click Voice tab > Info. Scroll down to the
Status
following example.
section of the page. Verify that the line is registered. Refer to the
Line 1
• Use an external phone to place an inbound call to the telephone number that
was assigned by your ITSP. Assuming that you have left the default settings in
place, the phone should ring and you can pick up the phone to get two-way
audio.
• If the line is not registered, you may need to refresh the browser several times
because it can take a few seconds for the registration to succeed. Also verify
that your DNS is configured properly.
ATA Administration Guide 41
Basic Administration and Configuration
Upgrading, Rebooting, and Resyncing Your ATA Device
NOTE If the device has more than one Line tab, each line tab must be configured
separately. Each line tab can be configured for a different ITSP.
Advanced Configurations
Other parameters may need to be changed from the defaults, depending on the
requirements of a specific ITSP. Some of the commonly configured parameters
include the following:
• Streaming Audio Server—You can enable an external music source for music
on hold. See the “Configuring a Streaming Audio Server,” on page 90 for further
information.
2
• NAT Settings—You can adjust these settings to resolve issues that arise when
using a ATA on a network behind a Network Address Translation (NAT) device.
See the “Network Address Translation (NAT) and Voice over IP (VoIP),” on
page 47 for further information.
• Subscriber Information—You can configure security parameters. See the
“Secure Call Implementation,” on page 72 for further information.
• Dial Plan—You can configure a dial plan for a specific line. See the “Configuring
Dial Plans,” on page 61 for further information.
Upgrading, Rebooting, and Resyncing Your ATA Device
The administration web server supports upgrading, rebooting, and resyncing
functions through special URLs. Administrator account privilege is needed for
these functions.
Upgrade URL
The Upgrade URL lets you upgrade the ATA device to the firmware specified by
the URL, which can identify either a TFTP or HTTP server.
ATA Administration Guide 42
Basic Administration and Configuration
Upgrading, Rebooting, and Resyncing Your ATA Device
2
NOTE If the value of the
cannot upgrade the ATA device even if the web page indicates otherwise.
The syntax of the Upgrade URL is as follows:
http://spa-ip-addr/admin/upgrade?[protocol://][server-name[:port]][/
firmware-pathname]
Both HTTP and TFTP are supported for the upgrade operation.
If no
protocol
host that requests the URL is used as
If no port specified, the default port of the protocol is used. (69 for TFTP or 80 for
HTTP)
firmware-pathname
The
directory on the TFTP or HTTP server. If no
spa.bin
http://192.168.2.217/admin/upgrade?tftp://192.168.2.251/spa.bin
is assumed, as in the following example:
is specified, TFTP is assumed. If no
Resync URL
Upgrade Enable
is typically the file name of the binary located in a
parameter in the Provisioning page is No, you
server-name
server-name
firmware-pathname
.
is specified, the
is specified,
/
The Resync URL lets you force the ATA device to do a resync to a profile specified
in the URL, which can identify either a TFTP, HTTP, or HTTPS server. The syntax of
the Resync URL is as follows:
http://spa-ip-addr/admin/resync?[[protocol://][server-name[:port]]/profilepathname]
NOTE The SPA resyncs only when it is idle.
If no parameter follows
page is used.
If no
protocol
host that requests the URL is used as
If no port is specified, the default port is used (69 for TFTP, 80 for HTTP, and 443
for HTTPS).
is specified, TFTP is assumed. If no
/resync?,
the Profile Rule setting from the Provisioning
server-name
server-name
.
is specified, the
ATA Administration Guide 43
Basic Administration and Configuration
Provisioning Your ATA Device
The profile-path is the path to the new profile with which to resync, for example:
http://192.168.2.217admin/resync?tftp://192.168.2.251/spaconf.cfg
Reboot URL
The Reboot URL lets you reboot the ATA device. The Reboot URL is as follows:
http://spa-ip-addr/admin/reboot
NOTE The ATA device reboots only when it is idle.
Provisioning Your ATA Device
2
This section describes the provisioning functionality of the ATA device. This
section includes the following topics:
• ”Provisioning Capabilities” section on page 44
• ”Configuration Profile” section on page 45
For detailed information about provisioning your ATA device, refer to the SPA
Provisioning Guide.
Provisioning Capabilities
The ATA device provides for secure provisioning and remote upgrade.
Provisioning is achieved through configuration profiles transferred to the device
via TFTP, HTTP, or HTTPS. To configure Provisioning, go to Provisioning tab in the
administration web server.
The ATA device can be configured to automatically resync its internal
configuration state to a remote profile periodically and on power up. The
automatic resyncs are controlled by configuring the desired profile URL into the
device.
The ATA device accepts profiles in XML format, or alternatively in a proprietary
binary format, which is generated by a profile compiler tool available from Cisco.
Find the Profiler Compiler for your ATA at http://www.cisco.com/web/partners/
sell/smb/products/voice_and_conferencing.html#~vc_technical_resources.
ATA Administration Guide 44
Basic Administration and Configuration
Provisioning Your ATA Device
The ATA device supports up to 256-bit symmetric key encryption of profiles. For
the initial transfer of the profile encryption key (initial provisioning stage), the ATA
device can receive a profile from an encrypted channel (HTTPS), or it can resync
to a binary profile generated by the Cisco-supplied profile compiler. In the latter
case, the profile compiler can encrypt the profile specifically for the target ATA
device, without requiring an explicit key exchange.
Remote firmware upgrade is achieved via TFTP or HTTP (firmware upgrades
using HTTPS are not supported). Remote upgrades are controlled by configuring
the desired firmware image URL into the ATA device via a remote profile resync.
For further information about remote provisioning refer to the SPA Provisioning
Guide.
Configuration Profile
2
The ATA configuration profile can be either an XML file or a binary file with a
proprietary format.
The XML file consists of a series of elements (one per configuration parameter),
encapsulated within the element tags <flat-profile> … </flat-profile>. The
encapsulated elements specify values for individual parameters. Here is an
example of a valid XML profile:
<flat-profile>
<Admin_Passwd>some secret</Admin_Passwd>
<Upgrade_Enable>Yes</Upgrade_Enable>
</flat-profile>
Binary format profiles contain ATA parameter values and user access permissions
for the parameters. By convention, the profile uses the extension .cfg (for example,
spa2102.cfg). The Profile Compiler (SPC) tool compiles a plain-text file containing
parameter-value pairs into a properly formatted and encrypted .cfg file. The SPC
tool is available for the Win32 environment and Linux-i386-elf environment.
Requests for SPC tools compiled on other platforms are evaluated on a case-bycase basis. Please contact your sales representative for further information about
obtaining the SPC tool.
The syntax of the plain-text file accepted by the profile compiler is a series of
parameter-value pairs, with the value in double quotes. Each parameter-value pair
is followed by a semicolon. Here is an example of a valid text source profile for
input to the SPC tool:
Admin_Passwd “some secret”;
Upgrade_Enable “Yes”;
Refer to the SPA Provisioning Guide for further details.
ATA Administration Guide 45
Basic Administration and Configuration
Provisioning Your ATA Device
The names of parameters in XML profiles can generally be inferred from the ATA
configuration Web pages, by substituting underscores (_) for spaces and other
control characters. Further, to distinguish between Lines 1, 2, 3, and 4,
corresponding parameter names are augmented by the strings _1_, _2_, _3_, and
_4_. For example, Line 1 Proxy is named Proxy_1_ in XML profiles.
Parameters in the case of source text files for the SPC tool are similarly named,
except that to differentiate Line 1, 2, 3, and 4, the appended strings ([1], [2], [3], or
[4]) are used. For example, the Line 1 Proxy is named Proxy[1] in source text
profiles for input to the SPC.
2
ATA Administration Guide 46
Configuring Your System for ITSP Interoperability
This chapter provides configuration details to help you to ensure that your
infrastructure properly supports voice services.
• “Network Address Translation (NAT) and Voice over IP (VoIP),” on page 47
• “Firewalls and SIP,” on page 53
3
• “Configuring SIP Timer Values,” on page 53
Network Address Translation (NAT) and Voice over IP (VoIP)
NAT is a function that allows multiple devices to share the same public, routable, IP
address to establish connections over the Internet. NAT is present in many
broadband access devices to translate public and private IP addresses. To enable
VoIP to co-exist with NAT, some form of NAT traversal is required.
Some ITSPs provide NAT traversal, but some do not. If your ITSP does not provide
NAT traversal, you have several options.
• “NAT Mapping with Session Border Controller,” on page 48
• “NAT Mapping with SIP-ALG Router,” on page 48
• “Configuring NAT Mapping with a Static IP Address,” on page 48
• “Configuring NAT Mapping with STUN,” on page 50
ATA Administration Guide 47
Configuring Your System for ITSP Interoperability
Network Address Translation (NAT) and Voice over IP (VoIP)
NAT Mapping with Session Border Controller
It is strongly recommended that you choose an ITSP that supports NAT mapping
through a Session Border Controller. With NAT mapping provided by the ITSP, you
have more choices in selecting a router.
NAT Mapping with SIP-ALG Router
If the ITSP network does not provide a Session Border Controller functionality, you
can achieve NAT mapping by using a router that has a SIP ALG (Application Layer
Gateway). The WRV200 router is recommended for this purpose, although any
router with a SIP-ALG can be used. By using a SIP-ALG router, you have more
choices in selecting an ITSP.
3
Configuring NAT Mapping with a Static IP Address
If the ITSP network does not provide a Session Border Controller functionality, and
if other requirements are met, you can configure NAT mapping to ensure
interoperability with the ITSP.
Requirements:
• You must have an external (public) IP address that is static.
• The NAT mechanism used in the router must be symmetric. See “Determining
Whether the Router Uses Symmetric or Asymmetric NAT,” on page 52.
• The LAN switch must be configured to enable Spanning Tree Protocol and Port
Fast on the ports to which the SPA devices are connected.
NOTE Use NAT mapping only if the ITSP network does not provide a Session Border
Controller functionality.
STEP 1 Connect to the administration web server, and choose Admin access with
Advanced settings.
STEP 2 Click Voice tab > SIP.
ATA Administration Guide 48
Configuring Your System for ITSP Interoperability
Network Address Translation (NAT) and Voice over IP (VoIP)
STEP 3 Scroll down to the NAT Support Parameters section, and then enter the following
settings to support static mapping to your public IP address:
• Handle VIA received, Insert VIA received, Substitute VIA Addr: yes
• Handle VIA rport, Insert VIA rport, Send Resp To Src Port: yes
• EXT IP: Enter the public IP address for your router.
Voice tab > SIP: NAT Support Parameters
3
STEP 4
STEP 5 Scroll down to the NAT Settings section.
Click Voice tab > Line N, where N represents the line interface number.
• NAT Mapping Enable: Choose YES.
• NAT Keep Alive Enable: Choose YES (optional).
Voice tab > Line N > NAT Settings
STEP 6
Click Submit All Changes.
NOTE You also need to configure the firewall settings on your router to allow SIP
traffic. See “Firewalls and SIP,” on page 53.
ATA Administration Guide 49
Configuring Your System for ITSP Interoperability
Network Address Translation (NAT) and Voice over IP (VoIP)
Configuring NAT Mapping with STUN
If the ITSP network does not provide a Session Border Controller functionality, and
if other requirements are met, it is possible to use STUN as a mechanism to
discover the NAT mapping. This option is considered a practice of last resort and
should be used only if the other methods are unavailable.
Requirements:
• STUN is a viable option only if your router uses asymmetric NAT. See
“Determining Whether the Router Uses Symmetric or Asymmetric NAT,” on
page 52.
• You must have a computer running STUN server software.
• The LAN switch must be configured to enable Spanning Tree Protocol and Port
Fast on the ports to which the SPA devices are connected.
3
NOTE Use NAT mapping only if the ITSP network does not provide a Session Border
Controller functionality.
STEP 1 Connect to the administration web server, and choose Admin access with
Advanced settings.
STEP 2 Click Voice tab > SIP.
STEP 3 Scroll down to the NAT Support Parameters section, and then enter the following
settings to enable and support the STUN server settings:
• Handle VIA received: yes
• Handle VIA rport: yes
• Insert VIA received: yes
• Insert VIA rport: yes
• Substitute VIA Addr: yes
• Send Resp To Src Port: yes
• STUN Enable: Choose yes.
• STUN Server: Enter the IP address for your STUN server.
ATA Administration Guide 50
Configuring Your System for ITSP Interoperability
Network Address Translation (NAT) and Voice over IP (VoIP)
Voice tab > SIP > NAT Support Parameters
3
STEP 4
STEP 5 Scroll down to the NAT Settings section.
Click Voice tab > Line N, where N is the number of the line interface.
• NAT Mapping Enable: Choose yes.
• NAT Keep Alive Enable: Choose yes (optional).
Voice tab > Line N > NAT Settings
NOTE Your ITSP may require the SPA device to send NAT keep alive messages to
keep the NAT ports open permanently. Check with your ITSP to determine
the requirements.
STEP 6 Click Submit All Changes.
NOTE You also need to configure the firewall settings on your router to allow SIP
traffic. See “Firewalls and SIP,” on page 53.
ATA Administration Guide 51
Configuring Your System for ITSP Interoperability
Network Address Translation (NAT) and Voice over IP (VoIP)
Determining Whether the Router Uses Symmetric or Asymmetric NAT
STUN does not work on routers with symmetric NAT. With symmetric NAT, IP
addresses are mapped from one internal IP address and port to one external,
routable destination IP address and port. If another packet is sent from the same
source IP address and port to a different destination, then a different IP address
and port number combination is used. This method is restrictive because an
external host can send a packet to a particular port on the internal host only if the
internal host first sent a packet from that port to the external host.
NOTE This procedure assumes that a syslog server is configured and is ready to receive
syslog messages.
3
STEP 1 Make sure you do not have firewall running on your PC that could block the syslog
port (port 514 by default).
STEP 2 Connect to the administration web server, and choose Admin access with
Advanced settings.
STEP 3 To enable debugging, complete the following tasks:
a. Click Voice tab > System.
b. In the Debug Server field, enter the IP address of your syslog server. This
address and port number must be reachable from the SPA9000.
c. From the Debug level drop-down list, choose 3.
STEP 4 To collect information about the type of NAT your router is using, complete the
following tasks:
a. Click Voice tab > SIP.
b. Scroll down to the NAT Support Parameters section.
c. From the STUN Test Enable field, choose yes.
STEP 5 To enable SIP signalling, complete the following task:
N
a. Click Voice tab > Line
b. In the SIP Settings section, choose full from the SIP Debug Option field.
ATA Administration Guide 52
, where N represents the line interface number.
Configuring Your System for ITSP Interoperability
Firewalls and SIP
STEP 6 Click Submit All Changes.
STEP 7 View the syslog messages to determine whether your network uses symmetric
NAT. Look for a warning header in the REGISTER messages, such as Warning: 399
spa "Full Cone NAT Detected.”
Firewalls and SIP
To enable SIP requests and responses to be exchanged with the SIP proxy at the
ITSP, you must ensure that your firewall allows both SIP and RTP unimpeded
access to the Internet.
• Make sure that the following ports are not blocked:
3
• SIP ports—UDP port 5060 through 5063, which are used for the ITSP line
interfaces
• RTP ports—16384 to 16482
• Also disable SPI (Stateful Packet Inspection) if this function exists on your
firewall.
Configuring SIP Timer Values
The default timer values should be adequate in most circumstances. However, you
can adjust the SIP timer values as needed to ensure interoperability with your
ISTP. For example, if SIP requests are returned with an “invalid certificate”
message, you may need to enter a longer SIP T1 retry value.
To view the default settings or to make changes, open the Voice > SIP page, and
scroll down to the SIP Timer Values section. For field descriptions, see ”SIP Timer
Values (sec) section,” on page135 of Appendix B.
ATA Administration Guide 53
Configuring Voice Services
This chapter describes how to configure your ATA device to meet the customer’s
requirements for voice services.
• “Supported Codecs,” on page 54
• “Using a FAX Machine (SPA2102, SPA3102 or SPA8000),” on page 55
• “Managing Caller ID Service,” on page 58
4
• “Silence Suppression and Comfort Noise Generation,” on page 60
• “Configuring Dial Plans,” on page 61
• “Secure Call Implementation,” on page 72
• “SIP Trunking and Hunt Groups on the SPA8000,” on page 77
Supported Codecs
The following list shows the current supported codecs for each ATA device. If you
need to change the G711u codec which is configured by default, set your
preferred codecs in the FXS Line tab(s); Audio Configuration. You may set your
first, second, and third preferred codec. See “ATA Routing Field Reference,” on
page111.
PAP2T / SPA2102 / SPA3102 / SPA8000
• G.711u (configured by default)
• G.711a
• G.726-16
• G.726-24
• G.726-32
ATA Administration Guide 54
Configuring Voice Services
Using a FAX Machine (SPA2102, SPA3102 or SPA8000)
• G.726-40
• G.729a
• G.723
WRTP54G
• G.711u (configured by default)
• G.711a
• G.726-32
• G.729a
• G.723
WRP400
4
• G.711u (configured by default)
• G.711a
• G.726-32
• G.729a
Using a FAX Machine (SPA2102, SPA3102 or SPA8000)
Follow this procedure to optimize fax completion rates.
NOTE T.38 Fax is only supported on the SPA2102, SPA3102, and the SPA8000. The
SPA2102 and SPA3102 support a single connection, while the SPA8000 supports
one connection for each pair of ports (1/2, 3/4, 5/6, and 7/8) for a maximum of four
connections.
STEP 1 Upgrade the ATA firmware to the latest version
STEP 2 Ensure that you have enough bandwidth for uplink and downlink.
• For G.711 fallback, it is recommend to have approximately 100Kbps.
• For T.38, allocate at least 50 kbps.
ATA Administration Guide 55
Configuring Voice Services
Using a FAX Machine (SPA2102, SPA3102 or SPA8000)
STEP 3 To optimize G.711 fallback fax completion rates, set the following on the Line tab
of your ATA device:
• Network Jitter Buffer: very high
• Jitter buffer adjustment: disable
• Call Waiting: no
• 3 Way Calling: no
• Echo Canceller: no
• Silence suppression: no
• Preferred Codec: G.711
• Use pref. codec only: yes
4
STEP 4 If you are using a Cisco media gateway for PSTN termination, disable T.38 (fax
relay) and enable fax using modem passthrough.
For example:
modem passthrough nse payload-type 110 codec g711ulaw
fax rate disable
fax protocol pass-through g711ulaw
STEP 5 Enable T.38 fax on the SPA 2102 by configuring the following parameter on the
Line tab for the FXS port to which the FAX machine is connected:
FAX_Passthru_Method: ReINVITE
NOTE If a T.38 call cannot be set-up, then the call should automatically revert to
G.711 fallback.
STEP 6 If you are using a Cisco media gateway use the following settings:
Make sure the Cisco gateway is correctly configured for T.38 with the SPA dial
peer. For example:
fax protocol T38
fax rate voice
fax-relay ecm disable
fax nsf 000000
no vad
ATA Administration Guide 56
Configuring Voice Services
Using a FAX Machine (SPA2102, SPA3102 or SPA8000)
Fax Troubleshooting
If have problems sending or receiving faxes, complete the following steps:
STEP 1 Verify that your fax machine is set to a speed between 7200 and 14400.
STEP 2 Send a test fax in a controlled environment between two ATAs.
STEP 3 Determine the success rate.
STEP 4 Monitor the network and record the following statistics:
• Jitter
• Loss
• Delay
4
STEP 5 If faxes fail consistently, capture a copy of the web interface settings by selecting
Save As > Web page, complete from the administration web server page. You
can send this configuration file to Technical Support.
STEP 6 Enable and capture the debug log. For instructions, refer to Appendix D,
“Troubleshooting.”.
NOTE You may also capture data using a sniffer trace.
STEP 7 Identify the type of fax machine connected to the ATA device.
STEP 8 Contact technical support:
• If you are an end user of VoIP products, contact the reseller or Internet
telephony service provider (ITSP) that supplied the equipment.
• If you are an authorized Cisco partner, contact Cisco technical support.
ATA Administration Guide 57
Configuring Voice Services
Managing Caller ID Service
Managing Caller ID Service
The choice of caller ID (CID) method is dependent on your area/region. To
configure CID, use the following parameters:
Parameter Ta b Description and Value
4
Caller ID
Method
Regional The following choices are available:
• Bellcore (N.Amer,China)—CID, CIDCW, and VMWI.
FSK sent after first ring (same as ETSI FSK sent after
first ring) (no polarity reversal or DTAS).
• DTMF (Finland, Sweden)—CID only. DTMF sent after
polarity reversal (and no DTAS) and before first ring.
• DTMF (Denmark)—CID only. DTMF sentbefore first
ring with no polarity reversal and no DTAS.
• ETSI DTMF—CID only. DTMF sent after DTAS (and no
polarity reversal) and before first ring.
• ETSI DTMF With PR—CID only. DTMF sent after
polarity reversal and DTAS and before first ring.
• ETSI DTMF After Ring—CID only. DTMF sent after
first ring (no polarity reversal or DTAS).
• ETSI FSK—CID, CIDCW, and VMWI. FSK sent after
DTAS (but no polarity reversal) and before first ring.
Waits for ACK from CPE after DTAS for CIDCW.
• ETSI FSK With PR (UK)—CID, CIDCW, and VMWI.
FSK is sent after polarity reversal and DTAS and
before first ring. Waits for ACK from CPE after DTAS for
CIDCW. Polarity reversal is applied only if equipment
is on hook.
• DTMF (Denmark) With PR—CID only. DTMF sent after
polarity reversal (and no DTAS) and before first ring.
The default is Bellcore(N.Amer, China).
Caller ID
FSK
Standard
ATA Administration Guide 58
Regional
The ATA device supports bell 202 and v.23
standards for caller ID generation. Select the FSK
standard you want to use, bell 202 or v.2 3.
The default is bell 202.
This field is not found in the PAP2T.
Configuring Voice Services
Polarity
Reversal
First
Ring
CAS
(DTAS)
DTMF/
FSK
Polarity
Reversal
CAS
(DTAS)
FSK
CAS
(DTAS)
Wait For
ACK
FSK
First
Ring
FSK
OSI FSK
a) Bellcore/ETSI Onhook Post-Ring FSK
d) Bellcore Onhook FSK w/o Ring
f) Bellcore/ETSI Offhook FSK
c) ETSI Onhook Pre-Ring FSK/DTMF
e) ETSI Onhook FSK w/o Ring
DTMF
b) ETSI Onhook Post-Ring DTMF
First
Ring
Managing Caller ID Service
There are three types of Caller ID:
4
• On Hook Caller ID Associated with Ringing — This type of Caller ID is used
for incoming calls when the attached phone is on hook. See the following
figure (a) – (c). All CID methods can be applied for this type of CID.
• On Hook Caller ID Not Associated with Ringing — This feature is used to
send VMWI signal to the phone to turn the message waiting light on and off
(see Figure 1 (d) and (e)). This is available only for FSK-based CID methods:
(Bellcore, ETSI FSK, and ETSI FSK With PR).
• Off Hook Caller ID — This is used to delivery caller-id on incoming calls
when the attached phone is off hook (see the following figure). This can be
call waiting caller ID (CIDCW) or to notify the user that the far end party
identity has changed or updated (such as due to a call transfer). This is
available only for FSK-based CID methods: (Bellcore, ETSI FSK, and ETSI
FSK With PR).
ATA Administration Guide 59
Configuring Voice Services
Silence Suppression and Comfort Noise Generation
Silence Suppression and Comfort Noise Generation
Voice Activity Detection (VAD) with Silence Suppression is a means of increasing
the number of calls supported by the network by reducing the required bandwidth
for a single call. VAD uses a sophisticated algorithm to distinguish between
speech and non-speech signals. Based on the current and past statistics, the VAD
algorithm decides whether or not speech is present. If the VAD algorithm decides
speech is not present, the silence suppression and comfort noise generation is
activated. This is accomplished by removing and not transmitting the natural
silence that occurs in normal two-way connection. The IP bandwidth is used only
when someone is speaking. During the silent periods of a telephone call, additional
bandwidth is available for other voice calls or data traffic because the silence
packets are not being transmitted across the network.
Comfort Noise Generation provides artificially-generated background white noise
(sounds), designed to reassure callers that their calls are still connected during
silent periods. If Comfort Noise Generation is not used, the caller may think the call
has been disconnected because of the “dead silence” periods created by the VAD
and Silence Suppression feature.
4
Silence suppression is configured in the Line and PSTN Line tabs. See “ATA
Routing Field Reference,” on page111.
ATA Administration Guide 60
Configuring Voice Services
Configuring Dial Plans
Configuring Dial Plans
Dial plans determine how the digits are interpreted and transmitted. They also
determine whether the dialed number is accepted or rejected. You can use a dial
plan to facilitate dialing or to block certain types of calls such as long distance or
international.
This section includes information that you need to understand dial plans, as well as
procedures for configuring your own dial plans. This section includes the following
topics:
• “About Dial Plans,” on page 61
• “Editing Dial Plans,” on page 70
4
About Dial Plans
This section provides information to help you understand how dial plans are
implemented.
Refer to the following topics:
• “Digit Sequences,” on page 61
• “Digit Sequence Examples,” on page 63
• “Acceptance and Transmission the Dialed Digits,” on page 66
• “Dial Plan Timer (Off-Hook Timer),” on page 67
• “Interdigit Long Timer (Incomplete Entry Timer),” on page 68
• “Interdigit Short Timer (Complete Entry Timer),” on page 68
Digit Sequences
A dial plan contains a series of digit sequences, separated by the | character. The
entire collection of sequences is enclosed within parentheses. Each digit
sequence within the dial plan consists of a series of elements, which are
individually matched to the keys that the user presses.
NOTE White space is ignored, but may be used for readability.
ATA Administration Guide 61
Configuring Voice Services
Configuring Dial Plans
Digit Sequence Function
4
0 1 2 3 4 5 6 7 8 9 0
* #
x Enter x to represent any character on the phone
[sequence] Enter characters within square brackets to create
Enter any of these characters to represent a key
that the user must press on the phone keypad.
keypad.
a list of accepted key presses. The user can press
any one of the keys in the list.
• Numeric range
For example, you would enter
user to press any one digit from 2 through 9.
[2-9] to allow the
• Numeric range with other characters
For example, you would enter
the user to press 3, 5, 6, 7, 8, or *.
.
(period)
<dialed:substituted> Use this format to indicate that certain dialed
Enter a period for element repetition. The dial plan
accepts 0 or more entries of the digit. For
example, 01. allows users to enter 0, 01, 011,
0111, and so on.
digits are replaced by other characters when the
sequence is transmitted. The dialed digits can
be zero or more characters.
[35-8*] to allow
EXAMPLE 1: <8:1650>xxxxxxx
When the user presses 8 followed by a sevendigit number, the system automatically replaces
the dialed 8 with 1650. If the user dials
85550112, the system transmits 16505550112.
EXAMPLE 2: <:1>xxxxxxxxxx
In this example, no digits are replaced. When the
user enters a 10-digit string of numbers, the
number 1 is added at the beginning of the
sequence. If the user dials 9725550112, the
system transmits 19725550112
ATA Administration Guide 62
Configuring Voice Services
Configuring Dial Plans
Digit Sequence Function
4
,
(comma)
!
(exclamation point)
*xx
S0 or L0
Enter a comma between digits to play an “outside
line” dial tone after a user-entered sequence.
EXAMPLE: 9, 1xxxxxxxxxx
An “outside line” dial tone is sounded after the
user presses 9, and the tone continues until the
user presses 1.
Enter an exclamation point to prohibit a dial
sequence pattern.
EXAMPLE: 1900xxxxxxx!
The system rejects any 11-digit sequence that
begins with 1900.
Enter an asterisk to allow the user to enter a 2digit star code.
Enter S0 to reduce the short inter-digit timer to 0
seconds, or enter L0 to reduce the long inter-digit
timer to 0 seconds.
Digit Sequence Examples
The following examples show digit sequences that you can enter in a dial plan.
In a complete dial plan entry, sequences are separated by a pipe character (|), and
the entire set of sequences is enclosed within parentheses.
EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8,
<:1212>xxxxxxx | 9, 1 [2-9] xxxxxx xxx | 9, 1 900 xxxxxx x ! |
9, 011xxxxxx. | 0 | [49]11 )
• Extensions on your system
EXAMPLE: (
<:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx !
| 9, 011xxxxxx. | 0 | [49]11 )
[1-8]xx Allows a user dial any three-digit number that starts with the digits 1
through 8. If your system uses four-digit extensions, you would instead enter
the following string:
[1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8,
[1-8]xx x
ATA Administration Guide 63
Configuring Voice Services
Configuring Dial Plans
• Local dialing with seven-digit number
4
EXAMPLE: ( [1-8]xx |
<:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx !
| 9, 011xxxxxx. | 0 | [49]111)
9, xxxxxxx After a user presses 9, an external dial tone sounds. The user can
enter any seven-digit number, as in a local call.
9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8,
• Local dialing with 3-digit area code and a 7-digit local number
EXAMPLE: ( [1-8]xx | 9, xxxxxxx |
<:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx !
| 9, 011xxxxxx. | 0 | [49]11 )
9, <:1>[2-9]xxxxxxxxx This example is useful where a local area code is required.
After a user presses 9, an external dial tone sounds. The user must enter a 10digit number that begins with a digit 2 through 9. The system automatically
inserts the 1 prefix before transmitting the number to the carrier.
9, <:1>[2-9]xxxxxxxxx | 8,
• Local dialing with an automatically inserted 3-digit area code
EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2- 9]xxxxxxxxx |
<:1212>xxxxxxx
011xxxxxx. | 0 | [49]11 )
| 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9,
8,
8, <:1212>xxxxxxx This is example is useful where a local area code is required
by the carrier but the majority of calls go to one area code. After the user
presses 8, an external dial tone sounds. The user can enter any seven-digit
number. The system automatically inserts the 1 prefix and the 212 area code
before transmitting the number to the carrier.
• U.S. long distance dialing
EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx |
8, <:1212>xxxxxxx |
011xxxxxx. | 0 | [49]11 )
9, 1 [2-9] xxxxxxxxx After the user presses 9, an external dial tone sounds. The
user can enter any 11-digit number that starts with 1 and is followed by a digit
2 through 9.
9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxx xx ! | 9,
ATA Administration Guide 64
Configuring Voice Services
Configuring Dial Plans
• Blocked number
• U.S. international dialing
4
EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx |
8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx |
9, 011xxxxxx. | 0 | [49]11 )
9, 1 900 xxxxxxx ! This digit sequence is useful if you want to prevent users from
dialing numbers that are associated with high tolls or inappropriate content,
such as 1-900 numbers in the U.S.. After the user press 9, an external dial tone
sounds. If the user enters an 11-digit number that starts with the digits 1900,
the call is rejected.
EXAMPLE: ( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx |
8, <:1212>xxxxxxx | 9, 1 [ 2-9] xxxxxxxxx | 9, 1 900 xx xxxxx
9, 011xxxxxx. | 0 | [49]11 )
! |
9, 1 900 xxxxxxx ! |
9, 011xxxxxx. After the user presses 9, an external dial tone sounds. The user
can enter any number that starts with 011, as in an international call from the
U.S.
• Informational numbers
EXAMPLE: ( [1-8]xx | 9, xxxxxxx |
<:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx !
| 9, 011xxxxxx. |
0 | [49]11 This example includes two digit sequences, separated by the pipe
character. The first sequence allows a user to dial 0 for an operator. The second
sequence allows the user to enter 411 for local information or 911 for
emergency services.
0 | [49]11 )
9, <:1>[2-9]xxxx xxxxx | 8,
ATA Administration Guide 65
Configuring Voice Services
Configuring Dial Plans
Acceptance and Transmission the Dialed Digits
When a user dials a series of digits, each sequence in the dial plan is tested as a
possible match. The matching sequences form a set of candidate digit sequences.
As more digits are entered by the user, the set of candidates diminishes until only
one or none are valid. When a terminating event occurs, the SPA9000 either
accepts the user-dialed sequence and initiates a call, or else rejects the sequence
as invalid. The user hears the reorder (fast busy) tone if the dialed sequence is
invalid.
The following table explains how terminating events are processed.
Ter mi nat in g Ev en t Processing
4
The dialed digits do not match
any sequence in the dial plan.
The dialed digits exactly match
one sequence in the dial plan.
The number is rejected.
• If the sequence is allowed by the dial plan, the
number is accepted and is transmitted
according to the dial plan.
• If the sequence is blocked by the dial plan, the
number is rejected.
A timeout occurs. The number is rejected if the dialed digits are
not matched to a digit sequence in the dial
plan within the time specified by the
applicable interdigit timer.
• The Interdigit Long Timer applies when the
dialed digits do not match any digit sequence
in the dial plan. The default value is 10
seconds.
• The Interdigit Short Timer applies when the
dialed digits match one or more candidate
sequences in the dial plan. The default value is
3 seconds.
The user presses the # key or
the dial softkey on the phone
display.
• If the sequence is complete and is allowed by
the dial plan, the number is accepted and is
transmitted according to the dial plan.
• If the sequence is incomplete or is blocked by
the dial plan, the number is rejected.
ATA Administration Guide 66
Configuring Voice Services
Configuring Dial Plans
Dial Plan Timer (Off-Hook Timer)
You can think of the Dial Plan Timer as “the off-hook timer.” This timer starts
counting when the phone goes off hook. If no digits are dialed within the specified
number of seconds, the timer expires and the null entry is evaluated. Unless you
have a special dial plan string to allow a null entry, the call is rejected. The default
value is 5.
Syntax for the Dial Plan Timer
SYNTAX: (P s<:n> | dial plan )
• s: The number of seconds; if no number is entered after P, the default timer of 5
• n: (optional): The number to transmit automatically when the timer expires; you
4
seconds applies.
can enter an extension number or a DID number. No wildcard characters are
allowed because the number will be transmitted as shown. If you omit the
number substitution, <:n>, then the user hears a reorder (fast busy) tone after
the specified number of seconds.
Examples for the Dial Plan Timer
• Allow more time for users to start dialing after taking a phone off hook.
EXAMPLE: (
| 9,8,011xx. | 9,8,xx.|[1-8]xx )
P9 After taking a phone off hook, a user has 9 seconds to begin dialing. If no
digits are pressed within 9 seconds, the user hears a reorder (fast busy) tone.
By setting a longer timer, you allow more time for users to enter the digits.
P9 | (9,8<:1408>[2-9]xxxxx x | 9,8,1[2-9]xxxxxxxxx
• Create a hotline for all sequences on the System Dial Plan
EXAMPLE: (
9]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)
P9<:23> After taking the phone off hook, a user has 9 seconds to begin dialing. If
no digits are pressed within 9 seconds, the call is transmitted automatically to
extension 23.
P9<:23> | (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-
• Create a hotline on a line button for an extension
EXAMPLE:
( P0 <:1000>)
With the timer set to 0 seconds, the call is transmitted automatically to the
specified extension when the phone goes off hook. Enter this sequence in the
Phone Dial Plan for Ext 2 or higher on a client station.
ATA Administration Guide 67
Configuring Voice Services
Configuring Dial Plans
Interdigit Long Timer (Incomplete Entry Timer)
You can think of this timer as the “incomplete entry” timer. This timer measures the
interval between dialed digits. It applies as long as the dialed digits do not match
any digit sequences in the dial plan. Unless the user enters another digit within the
specified number of seconds, the entry is evaluated as incomplete, and the call is
rejected. The default value is 10 seconds.
NOTE This section explains how to edit a timer as part of a dial plan. Alternatively, you can
modify the Control Timer that controls the default interdigit timers for all calls. See
“Resetting the Control Timers,” on page 70.
Syntax for the Interdigit Long Timer
4
SYNTAX: L:s, ( dial plan )
• s: The number of seconds; if no number is entered after L:, the default timer of
5 seconds applies.
• Note that the timer sequence appears to the left of the initial parenthesis for the
dial plan.
Example for the Interdigit Long Timer
EXAMPLE: L:15, (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx |
9,8,011xx. | 9,8,xx.|[1-8 ]xx)
L:15, This dial plan allows the user to pause for up to 15 seconds between digits
before the Interdigit Long Timer expires. This setting is especially helpful to users
such as sales people, who are reading the numbers from business cards and other
printed materials while dialing.
Interdigit Short Timer (Complete Entry Timer)
You can think of this timer as the “complete entry” timer. This timer measures the
interval between dialed digits. It applies when the dialed digits match at least one
digit sequence in the dial plan. Unless the user enters another digit within the
specified number of seconds, the entry is evaluated. If it is valid, the call proceeds.
If it is invalid, the call is rejected. The default value is 3 seconds.
ATA Administration Guide 68
Configuring Voice Services
Configuring Dial Plans
Syntax for the Interdigit Short Timer
• SYNTAX 1: S:s, ( dial p lan )
• SYNTAX 2: sequence Ss
Examples for the Interdigit Short Timer
• Set the timer for the entire dial plan.
4
Use this syntax to apply the new setting to the entire dial plan within the
parentheses.
Use this syntax to apply the new setting to a particular dialing sequence.
s: The number of seconds; if no number is entered after S, the default timer of 5
seconds applies.
EXAMPLE:
9,8,011xx. | 9,8,xx.|[1-8 ]xx)
S:6, While entering a number with the phone off hook, a user can pause for up
to 15 seconds between digits before the Interdigit Short Timer expires. This
setting is especially helpful to users such as sales people, who are reading the
numbers from business cards and other printed materials while dialing.
S:6, (9,8<:1408>[2-9]xxxx xx | 9,8,1[2-9]xxxxxxxxx |
• Set an instant timer for a particular sequence within the dial plan.
EXAMPLE: (9,8<:1408>[2-9]xxxxxx |
9,8,011xx. | 9,8,xx.|[1-8 ]xx)
9,8,1[2-9]xxxxxxxxxS0 With the timer set to 0, the call is transmitted automatically
when the user dials the final digit in the sequence.
9,8,1[2-9]xxxxxxxxxS0 |
ATA Administration Guide 69
Configuring Voice Services
Configuring Dial Plans
Editing Dial Plans
You can edit dial plans and can modify the control timers.
STEP 1 Start Internet Explorer, and then enter the IP address of the SPA9000. Click Admin
Login and then click Advanced.
Entering the Line Interface Dial Plan
This dial plan is used to strip steering digits from a dialed number before it is
transmitted out to the carrier.
STEP 1 Connect to the administration web server, and choose Admin access with
Advanced settings.
4
STEP 2 Click Voice tab > Line
STEP 3 Scroll down to the Dial Plan section.
STEP 4 Enter the digit sequences in the Dial Plan field. For more information, see “A bo ut
Dial Plans,” on page 61.
STEP 5 Click Submit All Changes.
N
, where N represents the line interface number.
Resetting the Control Timers
You can use the following procedure to reset the default timer settings for all calls.
NOTE If you need to edit a timer setting only for a particular digit sequence or type of call,
you can edit the dial plan. See “Abou t Dia l Plans, ” on page 61.
STEP 1 Connect to the administration web server, and choose Admin access with
Advanced settings.
STEP 2 Click Voice tab > Regional.
STEP 3 Scroll down to the Control Timer Values section.
ATA Administration Guide 70
Configuring Voice Services
Configuring Dial Plans
STEP 4 Enter the desired values in the Interdigit Long Timer field and the Interdigit Short
Timer field. Refer to the definitions at the beginning of this section.
4
ATA Administration Guide 71
Configuring Voice Services
Secure Call Implementation
Secure Call Implementation
This section describes secure call implementation with the ATA device . It includes
the following topics:
• ”Enabling Secure Calls” section on page 72
• ”Secure Call Details” section on page 73
• ”Using a Mini-Certificate” section on page 74
• ”Generating a Mini Certificate” section on page 75
4
NOTE This is an advanced topic meant for experience installers. See also the
Provisioning Guide
.
Enabling Secure Calls
A secure call is established in two stages. The first stage is no different from
normal call setup. The second stage starts after the call is established in the
normal way with both sides ready to stream RTP packets.
In the second stage, the two parties exchange information to determine if the
current call can switch over to the secure mode. The information is transported by
base64 encoding embedded in the message body of SIP INFO requests, and
responses using a proprietary format. If the second stage is successful, the ATA
device plays a special Secure Call Indication Tone for a short time to indicate to
both parties that the call is secured and that RTP traffic in both directions is being
encrypted.
If the user has a phone that supports call waiting caller ID (CIDCW) and that
service is enabled, the CID will be updated with the information extracted from the
Mini-Certificate received from the remote party. The Name field of the CID will be
prepended with a ‘$’ symbol. Both parties can verify the name and number to
ensure the identity of the remote party.
LV S
ATA Administration Guide 72
Configuring Voice Services
Secure Call Implementation
The signing agent is implicit and must be the same for all ATAs that communicate
securely with each other. The public key of the signing agent is pre-configured into
the ATA device by the administrator and is used by the ATA device to verify the
Mini-Certificate of its peer. The Mini-Certificate is valid if it has not expired, and it
has a valid signature.
The ATA device can be configured so that, by default, all outbound calls are either
secure or not secure. If secure by default, the user has the option to disable
security when making a call by dialing *19 before dialing the target number. If not
secure by default, the user can make a secure outbound call by dialing *18 before
dialing the target number. However, the user cannot force inbound calls to be
secure or not secure; that depends on whether the caller has security enabled or
not.
The ATA device will not switch to secure mode if the CID of the called party from
its Mini-Certificate does not agree with the user-id used in making the outbound
call. The ATA device performs this check after receiving the Mini-Certificate of the
called party
4
Secure Call Details
Looking at the second stage of setting up a secure call in greater detail, this stage
can be further divided into two steps.
STEP 1 The caller sends a “Caller Hello” message (base64 encoded and embedded in the
message body of a SIP INFO request) to the called party with the following
information:
• Message ID (4B)
• Version and flags (4B)
• SSRC of the encrypted stream (4B)
• Mini-Certificate (252B)
Upon receiving the Caller Hello, the called party responds with a Callee Hello
message (base64 encoded and embedded in the message body of a SIP
response to the caller’s INFO request) with similar information, if the Caller Hello
message is valid. The caller then examines the Callee Hello and proceeds to the
next step if the message is valid.
ATA Administration Guide 73
Configuring Voice Services
Secure Call Implementation
STEP 2 The caller sends the “Caller Final” message to the called party with the following
information:
• Message ID (4B)
• Encrypted Master Key (16B or 128b)
• Encrypted Master Salt (16B or 128b)
Using a Mini-Certificate
The Master Key and Master Salt are encrypted with the public key from the called
party mini-certificate. The Master Key and Master Salt are used by both ends for
deriving session keys to encrypt subsequent RTP packets. The called party then
responds with a Callee Final message (which is an empty message).
4
The Mini-Certificate (MC) contains the following information:
• User Name (32B)
• User ID or Phone Number (16B)
• Expiration Date (12B)
• Public Key (512b or 64B)
• Signature (1024b or 512B)
The MC has a 512-bit public key used for establishing secure calls. The
administrator must provision each subscriber of the secure call service with an
MC and the corresponding 512-bit private key. The MC is signed with a 1024-bit
private key of the service provider, which acts as the CA of the MC. The 1024-bit
public key of the CA signing the MC must also be provisioned for each subscriber.
The CA public key is used to verify the MC received from the other end. If the MC
is invalid, the call will not switch to secure mode. The MC and the 1024-bit CA
public key are concatenated and base64 encoded into the single parameter
Certificate
parameter, which should be kept secret, like a password. (
SRTP Private Key
. The 512-bit private key is base64 encoded into the
are configured in the Line tabs.)
SRTP Private Key
Mini Certificate
and
Mini
Because the secure call establishment relies on exchange of information
embedded in message bodies of SIP INFO requests/responses, the service
provider must ensure that the network infrastructure allows the SIP INFO
messages to pass through with the message body unmodified.
ATA Administration Guide 74
Configuring Voice Services
Secure Call Implementation
Generating a Mini Certificate
Cisco provides a Mini Certificate Generator for the generation of mini certificates
and private keys. Partners can download the Mini Certificate Generator by going
to Cisco Partner Central, Voice & Conferencing page, Technical Resources section.
Use the following URL:
http://www.cisco.com/web/partners/sell/smb/products/
voice_and_conferencing.html#~vc_technical_resources
NOTE The partner sites require a logon.
The Mini Certificate Generator uses the following syntax:
gen_mc ca-key user-name user-id expire-date
4
Where:
• ca-key is a text file with the base64 encoded 1024-bit CA private/public key
pairs for signing/verifying the MC, such as the following:
9CC9aYU1X5lJuU+EBZmi3AmcqE9U1LxEOGwopaGyGOh3VyhKgi6JaVtQZt87PiJINKW8XQj3B9Qq
e3VgYxWCQNa335YCnDsenASeBxuMIEaBCYd1l1fVEodJZOGwXwfAde0MhcbD0kj7LVlzcsTyk2TZ
YTccnZ75TuTjj13qvYs=5nEtOrkCa84/mEwl3D9tSvVLyliwQ+u/
Hd+C8u5SNk7hsAUZaA9TqH8Iw0J/
IqSrsf6scsmundY5j7Z5mK5J9uBxSB8t8vamFGD0pF4zhNtbrVvIXKI9kmp4vph1C5jzO9gDfs3M
F+zjyYrVUFdM+pXtDBxmM+fGUfrpAuXb7/k=
• user-name is the name of the subscriber, such as “Joe Smith”. Maximum length
is 32 characters
• user-id is the User ID of the subscriber, which must match exactly the user-id
used in the INVITE when making the call, such as “14083331234”. The
maximum length is 16 characters.
• expire-date is the expiration date of the MC, such as “00:00:00 1/1/34”
(34=2034). Internally the date is encoded as a fixed 12B string: 000000010134
The tool generates the
be provisioned.
Mini Certificate
and
SRTP Private Key
parameters that can
ATA Administration Guide 75
Configuring Voice Services
Secure Call Implementation
EXAMPLE:
gen_mc ca_key “Joe Smith” 14085551234 “00:00:00 1/1/34”
This example produces the following Mini Certificate and SRTP Private Key:
<Mini Certificate>
Sm9lIFNtaXRoAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAxNDA4NTU1MTIzNAAAAAAAMDAwMDAwMDEw
MTM00OvJakde2vVMF3Rw4pPXL7lAgIagMpbLSAG2+++YlSqt198Cp9rP/
xMGFfoPmDKGx6JFtkQ5sxLcuwgxpxpxkeXvpZKlYlpsb28L4Rhg5qZA+Gqj1hDFCmG6dffZ9SJhx
ES767G0JIS+N8lQBLr0AuemotknSjjjOy8c+1lTCd2t44Mh0vmwNg4fDck2YdmTMBR516xJt4/
uQ/
LJQlni2kwqlm7scDvll5k232EvvvVtCK0AYa4eWd6fQOpiESCO9CC9aYU1X5lJuU+EBZmi3AmcqE
9U1LxEOGwopaGyGOh3VyhKgi6JaVtQZt87PiJINKW8XQj3B9Qqe3VgYxWCQNa335YCnDsenASeBx
uMIEaBCYd1l1fVEodJZOGwXwfAde0MhcbD0kj7LVlzcsTyk2TZYTccnZ75TuTjj13qvYs=
<SRTP Private Key>
b/DWc96X4YQraCnYzl5en1CIUhVQQqrvcr6Qd/8R52IEvJjOw/
e+Klm4XiiFEPaKmU8UbooxKG36SEdKusp0AQ==
4
ATA Administration Guide 76
Configuring Voice Services
SIP Trunking and Hunt Groups on the SPA8000
SIP Trunking and Hunt Groups on the SPA8000
The SPA8000 supports SIP Trunking, which allows you to connect a traditional
PBX to VoIP services. In this configuration, calls go through the ITSP rather than the
PSTN, yet the call routing functionality is similar to that of traditional PSTN lines.
You can configure up to four trunk groups for the purpose of inbound call routing
and outbound caller identification. You can configure a trunk number on the
SPA8000, such that an incoming call automatically rings the grouped lines
simultaneously or in a specified order. For outbound calls, SIP Trunking ensures
that all calls on a trunk line can be identified by the trunk number and a common
caller ID. This feature helps you to ensure that calls are directed to available lines
and that work groups such as sales teams can work together to answer calls. In
addition, teams can project a common identity when placing outbound calls on a
trunk.
4
This section provides information about SIP trunking and explains how to
configure your trunk groups.
Refer to the following topics:
• “About SIP Trunking,” on page 78
• “Setting the Trunk Group Call Capacity,” on page 80
• “Inbound Call Routing for a Trunk Group,” on page 80
• “Contact List for a Trunk Group,” on page 81
• “Outgoing Call Routing for a Trunk Group,” on page 83
• “Configuring a Trunk Group,” on page 84
• “Additional Notes About Trunk Groups,” on page 87
• “Setting the Hunt Policy,” on page 86
• “Trunk Group Management,” on page 85
ATA Administration Guide 77
Configuring Voice Services
InternetIntegrated
Access Device
SPA8000
ITSP
Fax
PBX System
Fax
PBX System
SIP Trunking and Hunt Groups on the SPA8000
About SIP Trunking
The SIP Trunking feature allows a traditional PBX to seamlessly migrate from
PSTN service to VoIP service over a broadband link. The SPA8000 offers up to
eight telephone lines to the PBX.
4
The SPA8000 offers four trunk groups, numbered T1, T2, T3, and T4. A SIP-based
voice service with an ITSP can be configured on each trunk group with a distinct
phone number. Each of the eight SPA8000 lines can be configured either as a
standalone line, as in a classic ATA FXS port, or as a trunk line that is associated
with a trunk group.
• Inbound calling: A trunk group offers a single number for callers to call into the
small business, with the capability to programmatically ring one or more trunk
lines.
• Outbound calling: When a PBX phone makes a call, the PBX selects one of the
available trunk lines. The trunk line assumes the Caller ID of the trunk group.
ATA Administration Guide 78
Configuring Voice Services
Phone 1
Phone 2
Phone 3
Phone 4
Phone 5
Phone 6
Phone 7
Phone 8
L1
L2
L3
L4
L5
L6
L7
L8
T1
T2
T3
T4
Internal
RTP Path
SIP Path
ITSP
Proxy
Server
SIP Trunking and Hunt Groups on the SPA8000
The following figure shows a simplified logical block diagram of the SPA8000 with
the SIP Trunking feature.
Figure1 Logical Block Diagram of SIP Trunking
4
• SIP Path: As a standalone line, the SIP User Agent (SIP UA) exchanges signaling
directly with the ITSP equipment. As a trunk line, the Line UA exchanges
signaling with the internal proxy server only. The Internal Proxy Server handles
all SIP signalling between both ends of the call, from call establishment to
termination.
• RTP Path: Whether the line is standalone or a member of a trunk group, the Line
UA exchanges RTP packets directly with the ITSP equipment.
ATA Administration Guide 79
Configuring Voice Services
SIP Trunking and Hunt Groups on the SPA8000
NOTE Although the figure shows only one ITSP account, each standalone line and each
Trunk Group can be configured with a different ITSP (with some limitations applied).
Setting the Trunk Group Call Capacity
The ITSP may set a limit to the number of calls that can be made on a trunk group.
You can configure a trunk group’s call capacity parameter to meet the
requirements of the ITSP. Both incoming call and outgoing calls are counted
towards this limit. The call capacity has the following impact on call handling:
• Inbound calls: When the limit is reached, the Trunk UA replies 486 to the
caller.
4
• Outbound calls: When the limit is reached, the Line UA plays a fast busy
tone to the caller. Note that a trunk line can make an outgoing call only
through its own trunk. If that trunk reaches full capacity, it will not attempt to
failover to use other trunks.
You can configure this setting in the Voice tab > Trunk (T1 ... T4) page, Subscriber
Information section, Call Capacity field. For more information, see “Configuring a
Trunk Group,” on page 84.
Inbound Call Routing for a Trunk Group
An incoming call is handled as follows:
STEP 1 When an incoming call is detected by the Trunk UA, the UA first checks if there is
capacity to handle the call. If there is insufficient capacity, the UA rejects the call
with a 486 response.
STEP 2 If there is spare call capacity, the UA consults the Contact List to determine which
line or lines to ring (that is, for the proxy to send SIP INVITE to), and starts “hunting.”
(See “Configuring a Trunk Group,” on page 84)
STEP 3 When a line is selected to ring, one or more PBX phones may be alerted, according
to the PBX features and configuration.
STEP 4 The Caller ID of the external Caller is signaled by the Line UA out to the FXS port
using the configured Caller ID method (FSK, DTMF, etc.). The PBX must be able to
detect Caller ID signal in order for the proper Caller ID to show.
ATA Administration Guide 80
Configuring Voice Services
SIP Trunking and Hunt Groups on the SPA8000
STEP 5 If the call is picked up by the PBX, the Line UA replies 200 OK with SDP to the
internal Proxy. The Trunk UA in turn replies 200 OK to the ITSP and relay the Line
SDP in the 200 OK message also. If all goes well, the Line UA and the ITSP
equipment start exchanging RTP packets afterwards.
Contact List for a Trunk Group
The hunting process for incoming calls is controlled by the Contact List. The
Contact List specifies the lines to ring, the order in which to ring them, the duration
to ring one line before trying another line, and the maximum period to hunt. Below,
the syntax is described and examples are provided to help you to configure the
Contact List for each trunk group.
SYNTAX: line[,line[,line[…]]],hunt=hrule[,cfwd=target]
4
• line: The line numbers (1 - 8), or a wildcard * or ? to represent all lines.
• The Trunk UA rings only trunk lines, that is, lines that are assigned to a trunk
group through the Voice tab > Line page, Tr unk Gr oup field. The Trunk UA
does not ring any standalone lines that are included in the Contact List. The
Trunk UA rings any trunk line that is included in the list, even if it is not
assigned to the particular trunk group for this Contact List.
• You can instruct the SPA8000 to hunt only the phones that are on-hook,
through the Voice tab > SIP page, Trunking Parameters section, Hunt
Policy field. See “Setting the Hunt Policy,” on page 86.
• hunt=hrule: The hunt order, ring interval, and maximum duration, in the
following format: hunt =algo;interval;max
• algo: The hunt order.
- re: Restart. Hunting starts at the beginning of the list. If the first line does
not answer within the specified interval (see below), the hunt
proceeds through the lines in sequential order.
- ne: Next. The Trunk UA determines the line that was chosen in the
previous hunt, and hunting starts with the next line in the list. If that line
does not answer within the specified interval (see below), the hunt
proceeds through the lines in sequential order.
- ra: Random order. The Trunk UA randomly chooses a line from the list. If
the selected line does not answer within the specified interval (see
ATA Administration Guide 81
Configuring Voice Services
SIP Trunking and Hunt Groups on the SPA8000
below), the hunt proceeds randomly through the unchosen lines until
each line is tried.
- al: All. The Trunk UA rings all the lines at the same time.
• interval: The number of seconds to wait for one line to answer, before
choosing another line. If interval is *, the hunt is stopped at the first line that
starts ringing, and rings the line until it answers, or the caller hangs up, or the
line's ringer times out.
• max: The maximum duration of the hunt, either in seconds or cycles. When
this limit is reached, the call is rejected or is forwarded to the specified call
forward number (see below).
- If max is greater than interval, it represents the total time in seconds
to hunt.
- If max is less than interval, it represents the maximum number of
times to cycle through the hunt group. If max is 0, hunting continues
indefinitely until the caller either hangs up or the call is answered.
Exceptions: This value is ignored if algo = all, or interval = * (but
it must be present and should be set to 1).
4
• cfwd=target: If the call is unanswered and the maximum hunting duration
has been met, the call is forwarded to the specified number. When forwarding
the call, the SPA8000 sends a 302 response to the ITSP.
NOTE The call forward settings for the individual lines are ignored during hunting. Instead,
the cfwd settings in the Contact List are used.
EXAMPLES:
• 1,2,3,4,5,6,7,8,hunt=re;*;1
Lines 1 through 8 are included (1,2,3,4,5,6,7,8). The hunt starts at the
beginning of the list (hunt=re). When an available line is found, the call stays
with the line until the call is either answered, rejected, or cancelled by the caller
(* is entered for interval).
• ?,hunt=al;30;1,cfwd=14085550100
A wildcard character (?) is used to represent “all trunk lines.” All lines ring
simultaneously (hunt=al). If there is no answer after 30 seconds (30), the call
is forwarded to the specified number (cfwd=14085550100).
ATA Administration Guide 82
Configuring Voice Services
SIP Trunking and Hunt Groups on the SPA8000
• ?,hunt=ra;12;1,cfwd=14085550123
A wildcard character is used to represent “all trunk lines.” The Trunk UA
chooses lines in random order (hunt=ra). If a selected line does not answer
within 12 seconds (12), the Trunk UA chooses another line at random. If there is
no answer after 1 cycle (1), the call is forwarded to forwarded to the specified
number (cfwd=14085550123).
• ?,hunt=ra;*;1,cfwd=14085550155
A wildcard character is used to represent “all trunk lines.” The Trunk UA
chooses lines in random order (hunt=ra). The interval is *, meaning the hunt
stops when a selected line starts ringing, and will ring the line until it answers,
or the caller hangs up, or the line's ringer times out. If the ringer times out, the
call is automatically forwarded to the specified number (cfwd=14085550155).
Outgoing Call Routing for a Trunk Group
4
Outbound calls on a trunk line are handled as follows:
STEP 1 When a PBX phone selects an outside line, the PBX looks for an open line. If the
PBX finds an open line, it takes the line off hook and bridges the audio between
the PBX phone and the line. On detecting the off hook signal, the SPA8000 Line UA
plays dial tone and ready to collect digits from the PBX phone.
STEP 2 As the PBX phone user dials the number, the Line UA applies its dial plan to the
number. If the Line UA detects an invalid number, it rejects the all by playing
reorder tone, then howling tone, then silence. If a valid number is received, it sends
a SIP INVITE message to the internal Proxy.
STEP 3 The Proxy routes the call to the trunk group UA for the line, and the trunk group UA
will attempt to place the call to the ITSP if there is available capacity on the trunk. If
there is no call capacity left on the trunk, the internal Proxy will reject the INVITE
from the Line UA, which in turn terminates the call and plays reorder tone out to the
FXS port.
NOTE The SPA8000 will also apply the Trunk Dial Plan on the number before sending out
INVITE to the ITSP. This Trunk Dial Plan typically is redundant since the trunk should
trust the number sent by the Line UA. By default the trunk dial plan allows any nonempty number: ([*#0-9A-D][*#0-9A-D].)
ATA Administration Guide 83
Configuring Voice Services
SIP Trunking and Hunt Groups on the SPA8000
Configuring a Trunk Group
To configure a hunt group, you must first specify the trunk lines by assigning lines
to trunk groups. Then you enter the account information, specify the call capacity,
and configure the Contact List.
Before you begin this procedure, determine which lines you want to associate with
each trunk group that you are configuring. Refer to the following example:
Line Trunk Group
1, 3, 5 T1
4, 6, 8 T2
2None
4
STEP 1 Connect to the administration web server, and choose Admin access with
Advanced settings.
STEP 2 Assign each line to a trunk group, as needed:
a. Click Voice tab > L
b. In the Trunk Gr oup field, near the top of the line configuration page, choose a
trunk number or choose none for a standalone line (the default setting).
c. Repeat this step for each line that you want to add to a trunk group.
n
, where n represents the number of the line interface.
Voice > Ln > Trunk Group field
ATA Administration Guide 84
Configuring Voice Services
SIP Trunking and Hunt Groups on the SPA8000
STEP 3 Enter the settings for each trunk group, as needed:
a. Click Voice tab > T
b. Enter the account information in the Subscriber Information section.
• Display Name: The Caller ID that you want to use for outbound calls on this
line
• User ID: Your account number with the ITSP (usually the telephone number)
• Password: Your password for this ITSP account
c. In the Call Capacity field, enter the maximum number of concurrent calls
allowed by your ITSP, or leave the default setting, unlimited (16 calls).
d. In the Contact List field, modify the contact list as needed. See “Contact List for
a Trunk Group,” on page 81.
n
, where n represents the trunk group number (T1 ... T4).
4
e. Repeat this step for each trunk group that you need to configure.
STEP 4 Click Submit All Changes.
Trunk Group Management
You can check the status of the trunks by clicking the Trunk Status link, which
appears both at the top right corner of the web page and at the lower left corner.
You also can connect directly to the Trunk Status Page by entering the following
URL: http://spa8000-ip-addr/status. This page is available with the User
Login or the Admin Login.
Trunk Status page
ATA Administration Guide 85
Configuring Voice Services
SIP Trunking and Hunt Groups on the SPA8000
The Trunk S tat us page shows all calls that are currently active on each trunk
group.
This page shows a snapshot of the trunk activity. You can refresh the data at any
time by clicking the Refresh button on the web browser toolbar. The page shows
the following information:
• External: The called number
• Station: The SPA8000 line that is in use for this call
• Direction: The direction of the call, either Outbound or Inbound
• State: The state of the call
• Calling: An outbound call was initiated but is not ringing at the other end.
• Proceeding: The outbound call is ringing at the other end.
4
• Ringing: An inbound call is ringing.
• Connected: The call is connected.
• Duration: The duration of the call
In the case of a hung call, you can select the check box for the call and then click
the Delete button to cancel the call.
Setting the Hunt Policy
You can configure the SPA8000 so that the hunt rule applies to all phone or only to
the phones that are on hook.
STEP 1 Connect to the administration web server, and choose Admin access with
Advanced settings.
STEP 2 Click Voice tab > SIP.
STEP 3 Scroll down to the Trunk ing Pa ramet ers section.
STEP 4 In the Hunt Policy field, choose the desired option:
• onhook only: The hunt includes only the phones that are on hook.
• any state: The hunt includes all phones regardless of the state.
STEP 5 Click Submit All Changes.
ATA Administration Guide 86
Configuring Voice Services
SIP Trunking and Hunt Groups on the SPA8000
Additional Notes About Trunk Groups
This section includes information about other topics that may be of interest when
you are configuring trunk groups:
• Voice mail: There is no individual mail box for a trunk line. For example, if lines 1,
2, 3, and 4 belong the trunk group T1, then the four lines implicitly share the
same voice mail box from the ITSP. When there is new voice mail waiting in the
trunk mail box, the UAs for all four lines will be notified by the ITSP via the
internal Proxy, and all four lines will show the message waiting indicator, such
as by playing stutter dial tone, if enabled by the administrator.
• Supplementary features: Supplementary features are offered at the line level
only, not at the trunk level. Via the PBX, the phone user can trigger/control
supplementary service and settings by signaling to the line port or configuring
the line parameters. For more information, refer to the Appendix B, “ATA Voice
Field Reference.”
4
ATA Administration Guide 87
Configuring Music on Hold
This chapter explains how to configure Music on Hold using either a music file or
streaming audio.
This chapter includes the following topics:
• “Using the Internal Music Source for Music On Hold,” on page 88
• “Configuring a Streaming Audio Server,” on page 90
5
Using the Internal Music Source for Music On Hold
An internal music source with the user ID imusic is available. It plays an internally
stored music file repeatedly. The unit ships with a default music file (
Amor
). You can override this file by downloading a new file into the unit by using
TFTP.
Refer to the following topics:
• “Using the Internal Music Source,” on page 88
• “Changing the Music File for the Internal Music Source,” on page 89
Using the Internal Music Source
To use the internal music source, simply identify imusic as the MOH server for each
IP phone.
STEP 1 Use the phone menu to find the IP address of the phone:
a. Press the Setup button on the phone keypad.
Romance de
b. Press 9 - Network, and then scroll down to 2- Current IP Address.
ATA Administration Guide 88
Configuring Music on Hold
Using the Internal Music Source for Music On Hold
STEP 2 Start Internet Explorer, and then enter the IP address of the telephone. The
Telephone Configuration page appears in a separate browser window.
STEP 3 Click Admin Login, and then click Advanced.
STEP 4 Click the Ext 1 tab.
5
STEP 5 Scroll down to the
STEP 6 Enter the following value in the
STEP 7 Click Submit All Changes.
STEP 8 To verify, place a test call to the extension. When the call is answered and put on
hold, the caller should hear the default music file (
Call Feature Settings
MOH Server
section.
field: imusic
Romance de Amor
).
Changing the Music File for the Internal Music Source
The following resources are required to change the music file for the internal music
source:
• TFTP server software
• The IP address of the administration computer that is connected to the
SPA9000
• A music source in G.711u format, sampled at 8000 samples/sec with no file
header, up to 65.5 seconds in length, with no header information
STEP 1 Before you begin, make sure that you have TFTP server software running on your
computer.
STEP 2 Start Internet Explorer, connect to the administration web server, and choose
Admin access with Advanced settings.
STEP 3 Click Voice tab > SIP.
STEP 4 Scroll down to the
STEP 5 Enter the following URL in the Internal Music URL field:
tftp://server_IPaddress:portpath
Internal Music Source Parameters
section.
• server_IPaddress: The local IP address of the computer you are using as the
TFTP server
• port: The port number used by the TFTP server (default 69)
ATA Administration Guide 89
Configuring Music on Hold
Configuring a Streaming Audio Server
• path: The location and name of a music file in the correct format
• For example, if the computer local IP address is 192.168.0.5, the directory is
named
would enter the following URL: tftp://192.168.0.5:69/musicdir/
jazzmusic.dat
STEP 6 Click Submit All Changes. The unit reboots. Then the unit downloads the file and
stores it in flash memory.
musicdir
, and the converted music file is named
Configuring a Streaming Audio Server
This section describes how to use and configure a streaming audio server (SAS). It
includes the following topics:
jazzmusic.dat
5
, then you
• “About the Streaming Audio Server,” on page 90
• “Configuring the Streaming Audio Server,” on page 92
• “Using the IVR with an SAS Line,” on page 93
About the Streaming Audio Server
The Streaming Audio Server (SAS) feature lets you attach an audio source to an
FXS port and use it as a streaming audio source device. If the unit has multiple FXS
ports, either or both of the associated lines can be configured as an SAS server.
Use a media signal adapter or “music coupler” to connect an Ethernet cable from a
media source to the FXS port. For example, the MC-9700 Music Coupler has been
tested with ATA devices and is available at the following URL:
www.neogadgets.com/cart/
cart.php?target=product&product_id=17&substring=music+coupler
ATA Administration Guide 90
Configuring Music on Hold
Configuring a Streaming Audio Server
After you complete the required configuration, the FXS port is ready to stream
audio. The functionality depends on the hook state of the FXS port:
• If the FXS port is off hook, an incoming call is answered automatically and
audio is streamed to the calling party.
NOTE Each SAS server can maintain up to five simultaneous calls. If the
• If the FXS port is on-hook when the incoming call arrives, a SIP 503 response
code is transmitted to indicate “Service Not Available.”
• If an incoming call is auto-answered, but later the FXS port changes to on-hook,
the call is not terminated but continues to stream silence packets to the caller.
5
second line on the unit is disabled, then the SAS line can maintain up
to 10 simultaneous calls. Further incoming calls receive a busy signal
(SIP 486 Response).
• The SAS line can be set up to refresh each streaming audio session
periodically using a SIP re-INVITE message, which detects if the connection to
the caller is down. If the caller does not respond to the refresh message, the
SAS line terminates the call so that the streaming resource can be used for
other callers.
Additional information:
• The SAS line does not ring for incoming calls even if the attached equipment is
on-hook.
• If no calls are in session, battery is removed from tip-and-ring of the FXS port.
Some audio source devices have an LED to indicate the battery status. This can
be used as a visual indication as to whether audio streaming is in progress.
• Call Forwarding, Call Screening, Call Blocking, DND, and Caller-ID Delivery
features are not available on an SAS line.
ATA Administration Guide 91
Configuring Music on Hold
Configuring a Streaming Audio Server
Configuring the Streaming Audio Server
Use the following procedure to configure an SAS with an external music source.
STEP 1 Connect an RJ-11 adapter between the music source (a CD player or iPod, for
example) and an FXS port.
STEP 2 Start Internet Explorer, connect to the administration web server, and choose
Admin access with Advanced settings.
STEP 3 Configure the FXS port:
5
a. Click Voice tab > FXS
where you connected the cable from the external music source.
b. In the Subscriber Infomation section, enter the following settings:
N
, where N represents the number of the FXS port
• Display Name: Enter an extension number of name for the FXS 1 port, such
as Receptionist Area Fax Machine.
• User ID: Enter a three- to four-digit extension number that is not is use by
another extension.
c. In the Streaming Audio Server (SAS) section, choose yes from the SAS
Enable drop-down list.
STEP 4 Click Submit All Changes.
STEP 5 Configure each phone to use this audio source as the MOH server:
a. Click the PBX Status link to view the list of phones.
b. In the list, find the phone that you want to configure, and then click the hyperlink
in the IP Address column. The Telephone Configuration page appears in a
separate window.
c. Click the Ext 1 tab.
d. Scroll down to the Call Feature Settings section.
e. In the MOH Server field, enter the extension number that you assigned to the
FXS port for the streaming audio server.
f. Click Submit All Changes.
ATA Administration Guide 92
Configuring Music on Hold
Configuring a Streaming Audio Server
g. Close the window for the Telephone Configuration page.
h. Repeat this step to configure each phone, as needed.
Using the IVR with an SAS Line
The IVR can still be used on an SAS line, but the user needs to follow the following
steps:
STEP 1 Power off the ATA device.
STEP 2 Connect a phone to the port and make sure the phone is on-hook.
STEP 3 Power on the ATA device.
5
STEP 4 Pick up handset and press * * * * to invoke IVR in the usual way.
If the ATA device boots and finds that the SAS line is on-hook, it does not remove
battery from the line so that IVR may be used. But if the ATA device boots up and
finds that the SAS line is off-hook, it removes battery from the line because no
audio session is in progress.
ATA Administration Guide 93
6
Configuring the PSTN (FXO) Gateway on the SPA3102
This chapter describes how to configure the PSTN gateway on the SPA3102.
• ”Connecting to PSTN and VoIP Services” section on page 94
• ”How VoIP-To-PSTN Calls Work” section on page 95
• ”How PSTN-To-VoIP Calls Work” section on page 98
• ”Configuring VoIP Failover to PSTN” section on page102
• ”Sharing One VoIP Account Between the FXS and PSTN Lines” section on
page103
• ”Other Options” section on page104
• ”Call Scenarios” section on page105
Connecting to PSTN and VoIP Services
The SPA3102 has the following ports for connection to telephony devices:
• FXS port (Phone)—Connect to a standard analog telephone or fax machine,
configured by using the Line page.
• FXO port (Line)—Connect to a standard telephone wall jack for connectivity to
the PSTN, configured using the PSTN Line page.
Line 1 does not provide a gateway because it provides only VoIP service. The
VoIP-To-PSTN calling function is referred to as a
VoIP calling function is referred to as a
VoIP gateway
PSTN gateway
.
, and the PSTN-To-
Note the following definitions:
• VoIP caller—One who calls the ATA device via VoIP to obtain PSTN service
ATA Administration Guide 94
Configuring the PSTN (FXO) Gateway on the SPA3102
How VoIP-To-PSTN Calls Work
• VoIP user—VoIP caller that has a user account (user-id and password) on the
ATA d ev ic e
• PSTN caller—One who calls the ATA device from the PSTN to obtain VoIP
service
Line 1 can be configured with a regular VoIP account and can be used in the same
way as the Line 1 of any ATA device.
A second VoIP account can be configured to support PSTN gateway calls
exclusively. A different SIP port should be assigned to Line 1 and the PSTN Line.
The same VoIP account may be used for both Line 1 and the PSTN Line if a
different SIP port is assigned to each.
VoIP callers can be authenticated by one of the following methods:
• No Authentication—All callers are accepted for service.
6
• PIN—Caller is prompted to enter a PIN right after the call is answered.
• HTTP digest—SIP INVITE must contain a valid authorization header.
PSTN callers can be authenticated by one of the following methods:
• No authentication—All callers are accepted for service.
• PIN—Caller is prompted to enter a PIN right after the call is answered.
How VoIP-To-PSTN Calls Work
To obtain PSTN services through the SPA3102, the VoIP caller establishes a
connection with the PSTN Line by way of a standard SIP INVITE request
addressed to the PSTN Line. The PSTN Line can be configured to support onestage and two-stage dialing as described in the following sections.
One-Stage Dialing
One-stage dialing allows a call to be started over VoIP and then immediately get a
dial tone on the PSTN.
To use one-stage dialing, the Request-URI of the INVITE to the PSTN Line should
have the form <
number dialed by the VoIP caller, and <
of the SPA3102, such as 10.0.0.100:5061.
ATA Administration Guide 95
Dialed-Number
>@<
SPA-Address
>, where <
SPA-Address
Dialed-Number
> is a valid address and port
> is the
Configuring the PSTN (FXO) Gateway on the SPA3102
How VoIP-To-PSTN Calls Work
If the FXO port is currently in use (off-hook) or the PSTN line is being used by
another extension, the ATA device replies to the INVITE with a 503 response.
Otherwise, it compares the <
PSTN Line. If they are the same, the ATA device interprets this as a request for
two-stage dialing (see the ”Two-Stage Dialing” section on page 97). If they are
different, the ATA device processes the <
corresponding <
If dial plan processing fails, the ATA device replies with a 403 response.
Otherwise, it replies with a 200 and at the same time takes the FXO port off hook
and dials the target number returned after processing the dial plan.
Dial Plan
Dialed-Number
>.
> with the
User ID
Dialed-Number
6
parameter of the
> using the
NOTE If the
be disabled for the PSTN Line.
If HTTP Digest Authentication is enabled, the ATA device challenges the INVITE
with a 401 response if it does not have a valid Authorization header. The
Authorization header should include a <
one of eight VoIP user accounts that can be configured on the ATA device. The
credentials are computed based on the corresponding password using Message
Digest 5 (MD5). The <
stored on the ATA device. Each VoIP user account contains the information listed
below.
Table 1 Authentication Parameters
Parameter Web
User ID 1/2/
3/4/5/6/7/8
User ID parameter
User ID
Page
PSTN
Line
on the PSTN Line is blank, the
User ID
n> parameter must match one of the VoIP accounts
Description Values
The username value. 31-character string
n> parameter, where n refers to
Register
parameter should
Password 1/
2/3/4/5/6/7/
8
User 1/2/3/
4/5/6/7/8
DP
ATA Administration Guide 96
PSTN
Line
PSTN
Line
The password value. 31-character string
Specifies the dial plan to be used for
this VoIP user. If 0, dial plan
processing is disabled; the given
target number is dialed to the PSTN
as is.
Choice of 0-8
Configuring the PSTN (FXO) Gateway on the SPA3102
How VoIP-To-PSTN Calls Work
NOTE If Authentication is disabled, a default dial plan is used for all unknown VoIP users.
Two-Stage Dialing
In two-stage dialing, the ATA device takes the FXO port off-hook but does not
automatically dial any digits after accepting the call. To invoke two-stage dialing,
the VoIP caller should INVITE the PSTN Line without the user-id in the Request-URI
or with a user-id that matches exactly the <
user-id in the Request-URI is treated as a request for one-stage dialing if onestage dialing is enabled, or dropped by the ATA device (as if no user-id is given) if
one-stage dialing is disabled.
User ID
6
n> of the PSTN Line. A different
NOTE If Authentication is disabled, a default dial plan is assigned to all VoIP callers.
HTTP Digest Authentication can be also used for two-stage dialing, as in onestage dialing. If using HTTP Digest Authentication or Authentication is disabled, the
VoIP caller should hear the PSTN dial tone right after the call is answered (by a SIP
200 response).
If PIN Authentication is enabled, the VoIP caller is prompted to enter a PIN number
after the ATA device answers the call. The PIN number must end with a # key. The
inter-PIN-digit timeout is 10 seconds (not configurable). Up to eight VoIP caller PIN
numbers can be configured on the ATA device. A dial plan can be selected for
each PIN number. If the caller enters a wrong PIN or the ATA device times out
waiting for more PIN digits, the ATA device tears down the call immediately with a
BYE request.
NOTE When the source address of the INVITE is 127.0.0.1, authentication is automatically
disabled because this is a call by the local user. This applies to both one-stage and
two-stage dialing.
ATA Administration Guide 97
Configuring the PSTN (FXO) Gateway on the SPA3102
How PSTN-To-VoIP Calls Work
Table 2 Parameters for Two-Stage Dialing
6
Parameter Web
Page
VoIP Caller 1/2/
3/4/5/6/7/8 PIN
VoIP Caller 1/2/
3/4/5/6/7/8 DP
PSTN
Line
PSTN
Line
Description Values
The PIN for VoIP Caller 1, 2, 3, 4, 5, 6, 7,
or 8.
Specifies which dial plan to be used
for this VoIP caller. If 0, dial plan
processing is disabled; the given
target number is dialed to the PSTN
as is.
How PSTN-To-VoIP Calls Work
PSTN-To-VoIP calls can be made with two-stage dialing only. The only
authentication method available is the PIN method.
The ATA device takes the FXO port off hook after a configurable number of rings. If
PIN Authentication is enabled, it prompts the caller to enter the PIN number
followed by a # key. The Inter-PIN-digit timeout is set at 10 seconds. Up to eight
PSTN PIN numbers can be configured in the ATA device. If the given PIN does not
match any of the PSTN PIN values, the ATA device plays the reorder tone to the
FXO port for up to 10 seconds, and then takes the FXO port on-hook. If the given
PIN matches one of PSTN PIN values, the ATA device plays dial tone to the FXO
port and is ready to accept digits for the target VoIP number from the PSTN caller.
The collected digits are processed by the dial plan associated with the PIN
number.
31-character string
Choice of 1 to 8
NOTE If Authentication is disabled, a default dial plan is used for all PSTN callers.
ATA Administration Guide 98
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