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Cisco Small Business
SPA2102, SPA3102, SPA8000, SPA8800, PAP2T
Analog Telephone Adapters
ADMINISTRATION
GUIDE
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does not imply a partnership relationship between Cisco an d any other company. (1110R)
© 2011 Cisco Systems, Inc. All rights reserved. OL-17901-05
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Contents
Chapter 1: Introducing Cisco Small Business Analog Telephone Adapters 10
Comparison of ATA Devices 11
ATA Connectivity Requirements 14
PAP2T Connectivity 15
SPA2102 Connectivity 16
SPA3102 Connectivity 17
SPA8000 Connectivity 18
SPA8800 Connectivity 20
ATA Software Features 21
Voice Supported Codecs 21
SIP Proxy Redundancy 22
Other ATA Software Features 23
Chapter 2: Basic Administration and Configuration 31
Basic Services and Equipment Required 31
Downloading Firmware 32
Basic Installation and Configuration 32
Upgrading the Firmware for the ATA Device 32
Setting up Your ATA Device 33
Using the Administration Web Server 34
Connecting to the Administration Web Server 35
Setting Up the WAN Configuration for Your ATA Device 35
Registering to the Service Provider 37
Advanced Configurations 38
Upgrading, Rebooting, and Resyncing Your ATA Device 38
Upgrade URL 38
Resync URL 39
Reboot URL 39
Provisioning Your ATA Device 40
Provisioning Capabilities 40
Configuration Profile 41
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Chapter 3: Configuring Your System for ITSP Interoperability 42
Network Address Translation (NAT) and Voice over IP (VoIP) 42
NAT Mapping with Session Border Controller 43
NAT Mapping with SIP-ALG Router 43
Configuring NAT Mapping with a Static IP Address 43
Configuring NAT Mapping with STUN 45
Determining the Routers NAT Mechanism 47
Firewalls and SIP 48
Configuring SIP Timer Values 48
Chapter 4: Configuring Voice Services 49
Supported Codecs 49
Using a FAX Machine 50
Fax Troubleshooting 51
Managing Caller ID Service 53
Silence Suppression and Comfort Noise Generation 55
Configuring Dial Plans 56
About Dial Plans 56
Editing Dial Plans 65
Secure Call Implementation 66
Enabling Secure Calls 66
Secure Call Details 67
Using a Mini-Certificate 68
Generating a Mini Certificate 68
SIP Trunking and Hunt Groups on the SPA8000 70
About SIP Trunking 71
Setting the Trunk Group Call Capacity 73
Inbound Call Routing for a Trunk Group 73
Contact List for a Trunk Group 74
Outgoing Call Routing for a Trunk Group 76
Configuring a Trunk Group 77
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Trunk Group Management 78
Setting the Hunt Policy 80
Additional Notes About Trunk Groups 80
Chapter 5: Configuring Music on Hold 81
Using the Internal Music Source for Music On Hold 81
Using the Internal Music Source 81
Changing the Music File for the Internal Music Source 82
Configuring a Streaming Audio Server 83
About the Streaming Audio Server 83
Configuring the Streaming Audio Server 85
Using the IVR with an SAS Line 86
Chapter 6: Configuring the PSTN (FXO) Gateway on the SPA3102 87
Connecting to PSTN and VoIP Services 87
How VoIP-To-PSTN Calls Work 88
One-Stage Dialing (SPA3102 and SPA8800) 88
Two-Stage Dialing (SPA3102) 89
How PSTN-To-VoIP Calls Work 90
Terminating Gateway Calls 91
VoIP Outbound Call Routing (SPA3102) 92
Configuring VoIP Failover to PSTN 93
Sharing One VoIP Account Between the FXS and PSTN Lines (SPA3102)93
Other Options 94
PSTN Call to Ring Line 1 (SPA3102) 94
Symmetric RTP (SPA3102 and SPA8800) 94
Call Progress Tones (SPA3102 and SPA8800) 95
Call Scenarios 95
PSTN to VoIP Call with and Without Ring-Thru 96
VoIP to PSTN Call With and Without Authentication 96
Call Forwarding to PSTN Gateway (SPA3102 and SPA8800) 98
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Appendix A: ATA Routing Field Reference 100
Router Status page 100
Product Information section 101
System Status section 101
WAN Status page 102
Internet Connection Settings section 102
Static IP Settings section 103
PPPoE Settings section 103
Optional Settings section 104
MAC Clone Settings section 105
Remote Management section 105
QOS Settings section 105
VLAN Settings section 106
LAN Status page 106
Networking Service section 106
LAN Networking Settings section 107
Static DHCP Lease Settings section 107
Application page 107
Port Forwarding Settings section 108
DMZ Settings section 108
Miscellaneous Settings section 109
System Reserved Ports Range section 109
Appendix B: ATA Voice Field Reference 110
Info page 111
Product Information section 111
System Status section 112
Line Status section 112
System Information section (PAP2T) 115
PSTN Line Status section (SPA3102) 115
Trunk Status section (SPA8000) 118
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System page 119
System Configuration section 119
Internet Connection Type section (PAP2T) 120
Optional Network Configuration section (PAP2T) 120
Miscellaneous Settings section (not used with PAP2T) 121
SIP page 122
SIP Parameters section 122
SIP Timer Values (sec) section 124
Response Status Code Handling section 126
RTP Parameters section 127
SDP Payload Types section 129
NAT Support Parameters section 130
Trunking Parameters section (SPA8000) 133
Regional page 134
Call Progress Tones section 135
Distinctive Ring Patterns section 137
Distinctive Call Waiting Tone Patterns section 138
Distinctive Ring/CWT Pattern Names section 139
Ring and Call Waiting Tone Spec section 140
Control Timer Values (sec) section 141
Vertical Service Activation Codes section 143
Vertical Service Announcement Codes section (SPA2102, SPA8000) 149
Outbound Call Codec Selection Codes section 149
Miscellaneous section 151
Line page 155
Line Enable section 156
Streaming Audio Server (SAS) section 157
NAT Settings section 158
Network Settings section 159
SIP Settings section 160
Call Feature Settings section 163
Proxy and Registration section 164
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Subscriber Information section 166
Supplementary Service Subscription section 167
Audio Configuration section 169
Gateway Accounts section (SPA3102) 174
VoIP Fallback to PSTN section (SPA3102 and SPA8800) 175
Dial Plan section 175
FXS Port Polarity Configuration section 177
VoIP-to-PSTN Gateway Setup section (SPA8800) 177
FXO Timer Values (sec) section (SPA8800) 179
PSTN Disconnect Detection section (SPA8800) 180
International Control section (SPA8800) 183
Call Forward, Speed Dial, Supplementary Services, and Ring Settings
(SPA8000 and SPA8800) 184
Trunk Group page (SPA8000) 185
Line Enable section 185
Network Settings section 185
SIP Settings section 186
Subscriber Information section 189
Dial Plan section 191
NAT Settings section 191
Proxy and Registration section 192
PSTN Line page (SPA3102) 194
Line Enable section 194
NAT Settings section 195
Network Settings section 195
SIP Settings section 196
Proxy and Registration section 199
Subscriber Information section 201
Audio Configuration section 202
Dial Plans section 205
VoIP-To-PSTN Gateway Setup section 206
VoIP Users and Passwords (HTTP Authentication) section 208
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Ring Settings section 209
FXO (PSTN) Timer Values (sec) section 209
PSTN Disconnect Detection section 211
International Control (Settings) section 214
User page 216
Call Forward Settings section 217
Selective Call Forward Settings section 218
Speed Dial Settings section 218
Supplementary Service Settings section 219
Distinctive Ring Settings section 220
Ring Settings section 221
PSTN User page (SPA3102) 222
PSTN-To-VoIP Selective Call Forward Settings section 222
PSTN-To-VoIP Speed Dial Settings section 223
PSTN Ring Thru Line 1 Distinctive Ring Settings section 223
PSTN Ring Thru Line 1 Ring Settings section 223
Appendix C: Troubleshooting 224
Appendix D: Environmental Specifications 228
PAP2T 228
SPA2102 229
SPA3102 229
SPA8000 230
SPA8800 230
Appendix E: Where to Go From Here 231
Product Resources 231
Related Documentation 232
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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. The ATA can be used with an onsite call-control system such as the
SPA9000 Voice System or legacy PBX, or with an Internet-based call-control
system.
Figure1 ATA Deployment without Onsite Call Control
1
V
V
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1
V
V
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 ATA Models table summarizes the ports and features provided by the ATA
devices described in this document.
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Introducing Cisco Small Business Analog Telephone Adapters
Comparison of ATA Devices
ATA Models
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Product
Name
PAP2T 2 1 2 Voice adapter with two FXS ports.
SPA2102 2 1 1 2 Voice adapter with router.
SPA3102 1 1 1 1 1 Voice adapter with router and PSTN
SPA8000 8 1 Maintenance
SPA8800 4 4 1 Maintenance
FXS
(Analog
Phone)
FXO
PSTN
RJ-45
Internet
(WAN)
RJ-45
Ethernet
(LAN)
only
only
Voice
Lines
8 Voice adapter with support for up to
8 Voice adapter with support for up to
Description
connectivity.
eight FXS devices. Supports SIP
Trunking for inbound call routing to
trunk groups. Also has a single multiport RJ-21 50-pin connector for
optional patch-panel connectivity.
four FXS phones and up to four FXO
PSTN lines. Also has a single multiport RJ-21 50-pin connector for
optional patch-panel connectivity.
NOTE Additional ATA models, Cisco SPA112, SPA122, and SPA232D are covered in
separate Quick Start Guides and administration guides. For links to all ATA
documentation, see www.cisco.com/go/smallbizvoicegateways.
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.
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Introducing Cisco Small Business Analog Telephone Adapters
Comparison of ATA Devices
The following figure illustrates how the different ATA devices provide voice
connectivity in a VoIP network.
Figure 3 How ATAs Provide Voice Connectivity
1
SPA2102
SPA3102
The PAP2T and the SPA8000 provide FXS ports to connect fax machines and
analog phones to IP telephone services.
The SPA3102 and the SPA8800 act as SIP-PSTN gateways.
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Introducing Cisco Small Business Analog Telephone Adapters
ATA Connectivity Requirements
NOTE For information about the WRP400, see the WRP400 Administration Guide.
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:
1
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.
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.
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Introducing Cisco Small Business Analog Telephone Adapters
ATA Connectivity Requirements
PAP2T Connectivity
As shown in the following figure, the PAP2T has two FXS ports (voice lines 1 and
2).
1
IP
IP
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.
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ATA Connectivity Requirements
SPA2102 Connectivity
As shown in the following illustration, the SPA2102 has two FXS ports (voice lines
1 and 2).
IP
IP
1
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.
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ATA Connectivity Requirements
SPA3102 Connectivity
As shown in the following figure, the SPA3102 has one FXS port (voice line 1).
IP
IP
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.
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ATA Connectivity Requirements
SPA8000 Connectivity
As shown in the following illustration, the SPA8000 consists of eight voice ports
1
IP
IP
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.
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ATA Connectivity Requirements
NOTE
You can use line 1 or line 2 to access the IVR functions.
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 70.
1
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ATA Connectivity Requirements
SPA8800 Connectivity
As shown in the following figure, the SPA8800 has four voice modules that each
provide 1 FXS port and 1 FXO port.
1
IP
IP
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.
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ATA Software Features
NOTE
The IVR functions are accessed by connecting an analog telephone to Phone 1
only.
For proper operation, the service provider should use an Outbound Proxy to
forward all voice traffic when the SPA8800 is located behind a router. If
necessary, explicit port ranges can be specified for SIP and RTP.
ATA Software Features
1
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 21
SIP Proxy Redundancy, on page 22
Other ATA Software Features, on page 23
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
codec is to be used for each line. G.711a and G.711u are always enabled.
You can configure your preferred codec in Configuration Utility. See SDP
Payload Types section, on page 129 and Audio Configuration section, on
page 169. See also Supported Codecs, on page 49 for a list of which codecs
are supported on each ATA device.
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Introducing Cisco Small Business Analog Telephone Adapters
ATA Software Features
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Codec (Voice Compression
Algorithm)
G.711 (A-law and m -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 When no static payload value is assigned per RFC 1890, the ATA device can
support dynamic payloads for G.726.
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.
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Introducing Cisco Small Business Analog Telephone Adapters
ATA Software Features
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.
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 page 100.
Other ATA Software Features
1
The following table summarizes other features provided by ATA devices.
Feature Description
Streaming Audio
Server
T.38 Fax Relay See Using a FAX Machine, on page 50.
Silence
Suppression
See Configuring a Streaming Audio Server, on
page 83.
See Silence Suppression and Comfort Noise
Generation, on page 55.
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ATA Software Features
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.)
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 (Line 1 or 2 tab) is set to yes
for that line (in that case FAX pass-through is the same as
Modem pass-through).
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.
International Caller
ID Delivery
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
page 110.
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 page 110.
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Introducing Cisco Small Business Analog Telephone Adapters
ATA Software Features
Feature Description
Secure Calls A user (if enabled by service provider or administrator)
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 66.
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 110 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 page 110.
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 page110.
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 page 110.
Call Progress Tone
Pass Through
Echo Cancellation Impedance mismatch between the telephone and the IP
Cisco Small Business ATA Administration Guide 25
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 page110.
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 page110.
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Introducing Cisco Small Business Analog Telephone Adapters
ATA Software Features
Feature Description
1
Signaling Hook
Flash Event
Configurable Dial
Plan with Interdigit
Timers
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-call-
control. 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 hook-
flash event to the softswitch:
Call Waiting Service (parameter set in the
Line tab)
Three Way Conference Service (parameter
set in the Line tab)
Three Way Call Service (parameter set
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 page 110.
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.
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 56 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 page110.
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ATA Software Features
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 auto-
answer equipment, which then knows when to disengage.
CPC is configured in the Regional, Line, and PSTN Line
tabs. See ATA Voice Field Reference, on page 110.
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 page 110.
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
page 110.
SIP Over TCP To guarantee state-oriented communications, you can
Cisco Small Business ATA Administration Guide 27
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 e-commerce. SIP over TCP is configured in
the Line tabs. See ATA Voice Field Reference, on
page 110.
This feature applies to SPA2102, SPA3102, and SPA8800.
Page 28
Introducing Cisco Small Business Analog Telephone Adapters
ATA Software Features
Feature Description
SIP Over TLS SPA2102, SPA3102, and SPA8800 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 widely-
deployed 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 page 110.
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 page 110.
1
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Introducing Cisco Small Business Analog Telephone Adapters
ATA Software Features
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 DelayRandom delay range (in
seconds) to add to the Register Retry Intvl parameter when
retrying a SIP REGISTER after a failure. The default is 0,
which disables this feature.
Reg Retry Long Random DelayRandom delay range (in
seconds) to add to the Register Retry Long Intvl parameter
when retrying a SIP REGISTER after a failure. The default is
0, which disables this feature.
Reg Retry Intvl CapThe maximum value to cap the
exponential back-off retry delay. The exponential back-off
retry delay starts with the setting found in the Register Retry
Intvl parameter and doubles it on every REGISTER retry after
a failure. In other words, the retry interval after a failure is
always set to the seconds configured in the Register Retry
Intvl parameter. If this feature is enabled, the Reg Retry
Random Delay setting is added on top of the exponential
back-off adjusted delay value. The default value is 0, which
disables the exponential back-off feature.
Register Retry is configured in the SIP tab. See ATA Voice
Field Reference, on page 110.
Cisco Small Business ATA Administration Guide 29
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Introducing Cisco Small Business Analog Telephone Adapters
ATA Software Features
Feature Description
1
DHCP Renewal on
Timeout
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.
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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 31
Downloading Firmware section on page 32
Basic Installation and Configuration section on page 32
Upgrading the Firmware for the ATA Device section on page 32
Setting up Your ATA Device section on page 33
2
Using the Administration Web Server section on page 34
Upgrading, Rebooting, and Resyncing Your ATA Device section on
page 38
Provisioning Your ATA Device section on page 40
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
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Basic Administration and Configuration
Downloading Firmware
Computer with Microsoft Windows XP or Windows Vista (for system
configuration)
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 by going to:
2
http://www.cisco.com/en/US/products/ps10024/
tsd_products_support_series_home.html
and clicking the link.
Basic Installation and Configuration
See your the Quick Installation Guide and the User Guide for 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. (You may not hear a dial tone. Continue to step b.)
b. Press on the keypad to access the IVR menu.
c. Press to determine the Internet (WAN) IP address.
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Basic Administration and Configuration
Setting up Your ATA Device
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.
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.
b. When the window appears, click .
c. In the next window that appears, enter the IP address of the ATA device, and
then click .
d. In the window, verify that the correct device information and
product number appear. Then click .
2
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 to close the confirmation message.
g. To verify the upgrade, point the web browser to the IP address of the ATA
device. Check the page. The field should
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.
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.
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Basic Administration and Configuration
Using the Administration Web Server
NOTE By default, there are no passwords assigned for either the Administrator account or
the User account.
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.
2
When browsing pages with the Administrator account privilege, you can switch to
User account privilege by clicking the link.
If the User account password is set, the browser prompts for authentication when
you click the link. From the User account, you can switch to the
Administrator account by clicking the link. Authentication is required
if the Administrator account password has been set.
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 35
Setting Up the WAN Configuration for Your ATA Device section on
page 35
Registering to the Service Provider section on page 37
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Basic Administration and Configuration
Using the Administration Web Server
Advanced Configurations section on page 38
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 device.
STEP 2 Determine the address of the ATA device.
a. Connect an analog telephone to the Phone 1 port of the ATA device. (You may
not hear a dial tone. Continue to step b.)
b. Press on the keypad to access the IVR menu.
2
c. Press to determine the Internet (WAN) IP address.
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 page appears. By default, the page is in Basic User mode. Log
on to the administrator view by clicking , near the top right corner of
the page. Then click .
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.
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.
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Basic Administration and Configuration
Using the Administration Web Server
STEP 3 Complete the WAN configuration for DHCP, static IP addressing, or PPPoE.
For DHCP:
a. Select DHCP from the drop-down menu.
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
d. Then enter the PCs MAC address in the field.
For Static IP Addressing:
a. Select Static IP from the drop-down menu.
b. In the Static IP Settings section, enter the IP address in the field, the
subnet mask in the field, and the default gateway IP address in the
field.
2
setting.
c. In the Optional Settings section, enter the DNS server address(es) in the
and optional fields.
For PPPoE:
a. Select PPPoE from the drop-down menu. This is the correct
setting for most DSL users.
b. Enter the values provided by the ITSP in the following fields:
PPPoE Login Name
PPPoE Login Password
PPPoE Service Name
STEP 4 Click . The ATA device reboots.
STEP 5 To verify your progress, click the tab and then click . Under
, confirm the , , ,
, and .
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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.
STEP 2 Click Voice tab > Line , where is the line number that you want to configure.
STEP 3 Enter the account information for your ITSP. The following is the minimum required
configuration to connect the ATA device to an ITSP:
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)
2
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
section of the page. Verify that the line is registered.
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.
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.
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Basic Administration and Configuration
Upgrading, Rebooting, and Resyncing Your ATA Device
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 ServerYou can enable an external music source for music
on hold. See the Configuring the Streaming Audio Server, on page 85 for
further information.
NAT SettingsYou 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 42 for further information.
Subscriber InformationYou can configure security parameters. See the
Secure Call Implementation, on page 66 for further information.
2
Dial PlanYou can configure a dial plan for a specific line. See the
Configuring Dial Plans, on page 56 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.
NOTE If the value of the parameter in the Provisioning page is No, you
cannot upgrade the ATA device even if the web page indicates otherwise.
The syntax of the Upgrade URL is as follows:
Both HTTP and TFTP are supported for the upgrade operation.
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Basic Administration and Configuration
Upgrading, Rebooting, and Resyncing Your ATA Device
If no is specified, TFTP is assumed. If no is specified, the
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)
The is typically the file name of the binary located in a
directory on the TFTP or HTTP server. If no is specified,
is assumed, as in the following example:
Resync URL
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:
2
NOTE The SPA resyncs only when it is idle.
If no parameter follows the Profile Rule setting from the Provisioning
page is used.
If no is specified, TFTP is assumed. If no is specified, the
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).
The profile-path is the path to the new profile with which to resync, for example:
Reboot URL
The Reboot URL lets you reboot the ATA device. The Reboot URL is as follows:
.
NOTE The ATA device reboots only when it is idle.
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Basic Administration and Configuration
Provisioning Your ATA Device
Provisioning Your ATA Device
This section describes the provisioning functionality of the ATA device. This
section includes the following topics:
Provisioning Capabilities section on page 40
Configuration Profile section on page 41
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.
2
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.
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.
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Basic Administration and Configuration
Provisioning Your ATA Device
Configuration Profile
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:
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.
2
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:
Refer to the SPA Provisioning Guide for further details.
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.
Cisco Small Business ATA Administration Guide 41
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3
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 42
Firewalls and SIP, on page 48
Configuring SIP Timer Values, on page 48
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 43
NAT Mapping with SIP-ALG Router, on page 43
Configuring NAT Mapping with a Static IP Address, on page 43
Configuring NAT Mapping with STUN, on page 45
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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
the Routers NAT Mechanism, on page 47.
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.
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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
STEP 4 Click Voice tab > Line , where N represents the line interface number.
3
STEP 5 Scroll down to the NAT Settings section.
NAT Mapping Enable: Choose YES.
NAT Keep Alive Enable: Choose YES (optional).
Page 45
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 the Routers NAT Mechanism, on page 47.
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.
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Configuring Your System for ITSP Interoperability
Network Address Translation (NAT) and Voice over IP (VoIP)
Voice tab > SIP > NAT Support Parameters
STEP 4 Click Voice tab > Line , where N is the number of the line interface.
STEP 5 Scroll down to the NAT Settings section.
NAT Mapping Enable: Choose yes.
NAT Keep Alive Enable: Choose yes (optional).
3
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 48.
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Configuring Your System for ITSP Interoperability
Network Address Translation (NAT) and Voice over IP (VoIP)
Determining the Routers NAT Mechanism
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.
STEP 1 Make sure you do not have firewall running on your PC that could block the syslog
port (port 514 by default).
3
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:
a. Click Voice tab > Line , where N represents the line interface number.
b. In the SIP Settings section, choose full from the SIP Debug Option field.
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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 portsUDP port 5060 through 5063, which are used for the ITSP line
interfaces
RTP ports16384 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 page 124 of Appendix B.
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Page 49
Configuring Voice Services
This chapter describes how to configure your ATA device to meet the customers
requirements for voice services.
Supported Codecs, on page 49
Using a FAX Machine, on page 50
Managing Caller ID Service, on page 53
Silence Suppression and Comfort Noise Generation, on page 55
Configuring Dial Plans, on page 56
Secure Call Implementation, on page 66
4
SIP Trunking and Hunt Groups on the SPA8000, on page 70
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
page 100.
PAP2T / SPA2102 / SPA3102 / SPA8000 / SPA8800
G.711u (configured by default)
G.711a
G.726-16
G.726-24
G.726-32
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Configuring Voice Services
Using a FAX Machine
G.726-40
G.729a
G.723
Using a FAX Machine
You can connect a fax machine to an FXS port on the SPA2102, SPA3102,
SPA8000, and SPA8800. T.38 Fax is supported on these devices, for fax
transmission over an IP network. The SPA2102 and SPA3102 support a single
connection, while the SPA8000 and SPA8800 support one connection for each
pair of FXS ports (1/2, 3/4, 5/6, and 7/8) for a maximum of four connections.
Follow this procedure to optimize fax completion rates.
4
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.
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
STEP 4 If you are using a Cisco media gateway for PSTN termination, disable T.38 (fax
relay) and enable fax using modem passthrough.
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Configuring Voice Services
Using a FAX Machine
For example:
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:
NOTE If a T.38 call cannot be set-up, then the call should automatically revert to
4
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 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
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Configuring Voice Services
Using a FAX Machine
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 C,
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
4
telephony service provider (ITSP) that supplied the equipment.
If you are an authorized Cisco partner, contact Cisco technical support.
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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 Tab 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 DTMFCID only. DTMF sent after DTAS (and no
polarity reversal) and before first ring.
ETSI DTMF With PRCID only. DTMF sent after
polarity reversal and DTAS and before first ring.
ETSI DTMF After RingCID only. DTMF sent after
first ring (no polarity reversal or DTAS).
ETSI FSKCID, 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.
Caller ID
FSK
Standard
Cisco Small Business ATA Administration Guide 53
Regional
DTMF (Denmark) With PRCID only. DTMF sent after
polarity reversal (and no DTAS) and before first ring.
The default is Bellcore(N.Amer, China).
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.23.
The default is bell 202.
This field is not found in the PAP2T.
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Configuring Voice Services
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).
a) Bellcore/ETSI Onhook Post-Ring FSK
First
Ring
b) ETSI Onhook Post-Ring DTMF
First
Ring
c) ETSI Onhook Pre-Ring FSK/DTMF
Polarity
Reversal
d) Bellcore Onhook FSK w/o Ring
OSI FSK
e) ETSI Onhook FSK w/o Ring
Polarity
Reversal
CAS
(DTAS)
CAS
(DTAS)
FSK
DTMF
DTMF/
FSK
FSK
First
Ring
f) Bellcore/ETSI Offhook FSK
Cisco Small Business ATA Administration Guide 54
CAS
(DTAS)
Wait For
ACK
FSK
Page 55
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 page 100.
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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 56
Editing Dial Plans, on page 65
About Dial Plans
4
This section provides information to help you understand how dial plans are
implemented.
Refer to the following topics:
Digit Sequences, on page 56
Digit Sequence Examples, on page 59
Acceptance and Transmission the Dialed Digits, on page 61
Dial Plan Timer (Off-Hook Timer), on page 62
Interdigit Long Timer (Incomplete Entry Timer), on page 63
Interdigit Short Timer (Complete Entry Timer), on page 63
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.
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Configuring Dial Plans
Digit Sequence Function
Enter any of these characters to represent a key
that the user must press on the phone keypad.
Enter to represent any character on the phone
keypad.
Enter characters within square brackets to create
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 to allow the
user to press any one digit from 2 through 9.
Numeric range with other characters
For example, you would enter
the user to press 3, 5, 6, 7, 8, or *.
4
to allow
(period)
Enter a period for element repetition. The dial plan
accepts 0 or more entries of the digit. For
example, allows users to enter 0, 01, 011,
0111, and so on.
For sequence substitution, use this format to
indicate that certain dialed digits are replaced by
other characters when the sequence is
transmitted. The digits can be zero or
more characters.
EXAMPLE 1:
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:
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
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Configuring Dial Plans
Digit Sequence Function
For an intersequence tone, enter a comma
(comma)
(exclamation point)
between digits to play an outside line dial tone
after a user-entered sequence.
EXAMPLE:
An outside line dial tone is sounded after the
user presses 9, and the tone continues until the
user presses 1.
For number barring, enter an exclamation point to
prohibit a dial sequence pattern.
EXAMPLE:
The system rejects any 11-digit sequence that
begins with 1900.
4
Enter an asterisk to allow the user to enter a 2digit star code.
For Interdigit Timer Master Override, enter to
reduce the short inter-digit timer to 0 seconds, or
enter L0 to reduce the long inter-digit timer to 0
seconds.
For a pause, enter followed by a number and a
space. The duration of the pause is the specified
number of seconds. This feature is typically used
for implementation of a hot line and warm line,
with 0 delay for the hot line and a non-zero delay
for a warm line.
EXAMPLE:
NOTE The SPA implicitly appends the vertical code sequences entered in the regional
parameter settings to the end of the dial plan. Likewise, if Enable_IP_Dialing is
enabled, then IP dialing is also accepted on the associated line.
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Configuring Dial Plans
4
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:
Extensions on your system
EXAMPLE: [1-8]xx
[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]xxx
Local dialing with seven-digit number
EXAMPLE: 9, xxxxxxx
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.
Local dialing with 3-digit area code and a 7-digit local number
EXAMPLE: 9, <:1>[2-9]xxxxxxxxx
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.
Local dialing with an automatically inserted 3-digit area code
EXAMPLE: 8,
<:1212>xxxxxxx
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Configuring Dial Plans
4
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:
9, 1 [2-9] xxxxxxxxx
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.
Blocked number
EXAMPLE:
9, 1 900 xxxxxxx !
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.
U.S. international dialing
EXAMPLE:
9, 011xxxxxx.
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:
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
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Configuring Dial Plans
4
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.
Terminating Event Processing
The dialed digits do not match
any sequence in the dial plan.
The dialed digits exactly match
one sequence in the dial plan.
A timeout occurs. The number is rejected if the dialed digits are
The user presses the # key or
the dial softkey on the phone
display.
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.
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.
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.
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Configuring Dial Plans
4
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: s
s: The number of seconds; if no number is entered after , the default timer of 5
seconds applies.
n: (optional): The number to transmit automatically when the timer expires; you
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: P9
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.
Create a hotline for all sequences on the System Dial Plan
EXAMPLE: P9<:23>
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.
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.
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Configuring Dial Plans
NOTE This section explains how to edit a timer as part of a dial plan. Alternatively, you can
4
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.
modify the Control Timer that controls the default interdigit timers for all calls. See
Resetting the Control Timers, on page 65.
Syntax for the Interdigit Long Timer
SYNTAX: :s,
s: The number of seconds; if no number is entered after , 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,
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.
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Configuring Dial Plans
4
Syntax for the Interdigit Short Timer
SYNTAX 1: :s,
Use this syntax to apply the new setting to the entire dial plan within the
parentheses.
SYNTAX 2: sequence
Use this syntax to apply the new setting to a particular dialing sequence.
s: The number of seconds; if no number is entered after , the default timer of 5
seconds applies.
Examples for the Interdigit Short Timer
Set the timer for the entire dial plan.
EXAMPLE: S:6,
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.
Set an instant timer for a particular sequence within the dial plan.
EXAMPLE: 9,8,1[2-9]xxxxxxxxxS0
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.
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Configuring Dial Plans
STEP 1 Start Internet Explorer, and then enter the IP address of the SPA9000. Click
STEP 1 Connect to the administration web server, and choose Admin access with
STEP 2 Click Voice tab > Line , where N represents the line interface number.
4
Editing Dial Plans
You can edit dial plans and can modify the control timers.
and then click .
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.
Advanced settings.
STEP 3 Scroll down to the Dial Plan section.
STEP 4 Enter the digit sequences in the Dial Plan field. For more information, see About
Dial Plans, on page 56.
STEP 5 Click Submit All Changes.
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 About Dial Plans, on page 56.
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.
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.
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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 66
Secure Call Details section on page 67
Using a Mini-Certificate section on page 68
Generating a Mini Certificate section on page 68
NOTE This is an advanced topic meant for experience installers. See also the
.
4
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.
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.
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Secure Call Implementation
4
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
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 callers 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.
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)
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Secure Call Implementation
4
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).
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
. The 512-bit private key is base64 encoded into the
parameter, which should be kept secret, like a password. ( and
are configured in the Line tabs.)
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.
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
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Secure Call Implementation
NOTE The partner sites require a logon.
4
The Mini Certificate Generator uses the following syntax:
ca-key user-name user-id expire-date
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:
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 and parameters that can
be provisioned.
EXAMPLE:
This example produces the following Mini Certificate and SRTP Private Key:
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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.
This section provides information about SIP trunking and explains how to
configure your trunk groups.
4
Refer to the following topics:
About SIP Trunking, on page 71
Setting the Trunk Group Call Capacity, on page 73
Inbound Call Routing for a Trunk Group, on page 73
Contact List for a Trunk Group, on page 74
Outgoing Call Routing for a Trunk Group, on page 76
Configuring a Trunk Group, on page 77
Additional Notes About Trunk Groups, on page 80
Setting the Hunt Policy, on page 80
Trunk Group Management, on page 78
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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.
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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.
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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 groups 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.
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.
4
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 77.
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 77)
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.
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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:
: The line numbers (1 - 8), or a wildcard * or ? to represent all lines.
4
The Trunk UA rings only trunk lines, that is, lines that are assigned to a trunk
group through the Voice tab > Line page, Trunk Group 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 80.
: The hunt order, ring interval, and maximum duration, in the
following format:
algo: The hunt order.
- : Restart. Hunting starts at the beginning of the list. If the first line does
not answer within the specified (see below), the hunt
proceeds through the lines in sequential order.
- : 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 (see below), the hunt
proceeds through the lines in sequential order.
- : Random order. The Trunk UA randomly chooses a line from the list. If
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below), the hunt proceeds randomly through the unchosen lines until
each line is tried.
- : All. The Trunk UA rings all the lines at the same time.
: 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.
: 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 is greater than , it represents the total time in seconds
to hunt.
- If is less than , 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 , or (but
it must be present and should be set to 1).
4
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:
Lines 1 through 8 are included ( ). The hunt starts at the
beginning of the list ( ). 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 ).
A wildcard character ( ) is used to represent all trunk lines. All lines ring
simultaneously ( ). If there is no answer after 30 seconds ( ), the call
is forwarded to the specified number ( .
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A wildcard character is used to represent all trunk lines. The Trunk UA
chooses lines in random order ( ). If a selected line does not answer
within 12 seconds ( ), the Trunk UA chooses another line at random. If there is
no answer after 1 cycle ( ), the call is forwarded to forwarded to the specified
number ( ).
A wildcard character is used to represent all trunk lines. The Trunk UA
chooses lines in random order ( ). 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 ( ).
Outgoing Call Routing for a Trunk Group
Outbound calls on a trunk line are handled as follows:
4
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:
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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
2 None
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 , where represents the number of the line interface.
b. In the Trunk Group 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.
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STEP 3 Enter the settings for each trunk group, as needed:
a. Click Voice tab > T , where n represents the trunk group number (T1 ... T4).
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 74.
e. Repeat this step for each trunk group that you need to configure.
4
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: spa8000-ip-addr . This page is available with the User
Login or the Admin Login.
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The Trunk Status 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:
4
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.
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.
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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 Trunking Parameters 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.
4
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.
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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 81
Configuring a Streaming Audio Server, on page 83
Using the Internal Music Source for Music On Hold
5
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 (
). 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 81
Changing the Music File for the Internal Music Source, on page 82
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 button on the phone keypad.
b. Press , and then scroll down to .
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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 , and then click .
STEP 4 Click the tab.
STEP 5 Scroll down to the section.
STEP 6 Enter the following value in the field:
STEP 7 Click .
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 ( ).
5
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, 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 > .
STEP 4 Scroll down to the section.
STEP 5 Enter the following URL in the Internal Music URL field:
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
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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 , and the converted music file is named , then you
would enter the following URL:
STEP 6 Click . The unit reboots. Then the unit downloads the file and
stores it in flash memory.
Configuring a Streaming Audio Server
This section describes how to use and configure a streaming audio server (SAS). It
includes the following topics:
5
About the Streaming Audio Server, on page 83
Configuring the Streaming Audio Server, on page 85
Using the IVR with an SAS Line, on page 86
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
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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.
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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:
a. Click Voice tab > FXS , where N represents the number of the FXS port
where you connected the cable from the external music source.
b. In the Subscriber Infomation section, enter the following settings:
5
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 .
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 .
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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.
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6
Configuring the PSTN (FXO) Gateway on the
SPA3102
This chapter describes how to configure the PSTN gateway on the SPA3102 and
the SPA8800.
Connecting to PSTN and VoIP Services section on page 87
How VoIP-To-PSTN Calls Work section on page 88
How PSTN-To-VoIP Calls Work section on page 90
Configuring VoIP Failover to PSTN section on page 93
Sharing One VoIP Account Between the FXS and PSTN Lines (SPA3102)
section on page 93
Other Options section on page 94
Call Scenarios section on page 95
Connecting to PSTN and VoIP Services
Both the SPA3102 and the SPA8800 allow your analog and IP phones to
participate in calls over the VoIP network and the PSTN. Both devices provide a
VoIP-To-PSTN calling function, which is referred to as a , as well as a
PSTN-To-VoIP calling function, which is referred to as a .
These ATAs function somewhat differently because they are designed to meet
different business needs.
The SPA3102 is designed to work independently as a PSTN gateway and
VoIP gateway, without relying on a PBX for call control.
- FXS Port: The SPA3102 has 1 FXS port that you can connect to a
standard analog telephone or fax machine. Configure the FXS settings
by using the SPA3102 Line page.
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How VoIP-To-PSTN Calls Work
- FXO Port: The SPA3102 has 1 FXO port that you can connect to the
PSTN. Configure the FXO settings by using the SPA3102 PSTN Line
page.
The SPA8800 is designed to work with your PBX as a PSTN gateway and a
VoIP gateway. Call control is provided by a standard PBX, an Asteriskbased IP PBX, the SPA9000 Voice System, or an Internet-based call control
system.
- FXS Port: The SPA8800 has 4 FXS ports that you can connect to
standard analog telephones or fax machines. Configure the FXS settings
by using the SPA8800 Phone 1-4 pages.
- FXO Port: The SPA8800 has 4 FXO port sthat you can connect to the
PSTN. Configure the FXO settings by using the SPA8800 Line 1-4 pages.
6
How VoIP-To-PSTN Calls Work
To obtain PSTN services through the SPA3102 or the SPA8800, the VoIP caller
establishes a connection with the PSTN Line by way of a standard SIP INVITE
request addressed to the PSTN Line.
One-Stage Dialing (SPA3102 and SPA8800)
One-stage dialing allows a call to be started over VoIP and then immediately get a
dial tone on the PSTN. When you take a phone off hook and dial a number, the call
is automatically routed to the VoIP or the PSTN, based on the dial plan.
On the SPA3102, you can enable HTTP Digest Authentication. In this case, the
SPA3102 challenges the INVITE with a 401 response if it does not have a valid
Authorization header. The Authorization header should include a < n>
parameter, where n refers to 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 <
parameter must match one of the VoIP accounts stored on the ATA device. You can
configure these settings on the Voice tab > PSTN Line page. For more information,
see VoIP Users and Passwords (HTTP Authentication) section, on page 208.
n>
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Configuring the PSTN (FXO) Gateway on the SPA3102
How VoIP-To-PSTN Calls Work
Two-Stage Dialing (SPA3102)
In two-stage dialing, the SPA3102 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 < n> of the PSTN Line. A different
user-id in the Request-URI is treated as a request for one-stage dialing if onestage dialing is enabled, or dropped by the SPA3102 (as if no user-id is given) if
one-stage dialing is disabled.
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).
You also can enable PIN authentication. In this case, the VoIP caller is prompted to
enter a PIN number after the SPA3102 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 SPA3102. A dial plan can
be selected for each PIN number. If the caller enters a wrong PIN or the SPA3102
times out waiting for more PIN digits, the SPA3102 tears down the call
immediately with a BYE request.
6
The call scenarios may involve the following types of callers:
VoIP callerSomeone who calls the ATA device via VoIP to obtain PSTN
service
VoIP userA VoIP caller that has a user account (user-id and password) on the
SPA3102
PSTN callerSomeone who calls the ATA device from the PSTN to obtain VoIP
service
VoIP callers can be authenticated by one of the following methods:
No AuthenticationAll callers are accepted for service.
PINCaller is prompted to enter a PIN right after the call is answered.
HTTP digestSIP INVITE must contain a valid authorization header.
PSTN callers can be authenticated by one of the following methods:
No authenticationAll callers are accepted for service.
PINCaller is prompted to enter a PIN right after the call is answered.
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How PSTN-To-VoIP Calls Work
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.
These settings can be configured on the SPA3102 PSTN Line page. See VoIP-
To-PSTN Gateway Setup section, on page 206.
How PSTN-To-VoIP Calls Work
For PSTN-to-VoIP calls, the basic PSTN-to-VoIP call flow is as follows:
1. When a PSTN call comes in to the ATA device and is unanswered (after a
configurable number of rings), then the ATA device takes the FXO port off hook.
6
2. The ATA device plays dial tone.
3. The PSTN caller enters the target telephone number. The collected digits are
processed by the default dial plan.
On the SPA3102, you can add PIN authentication to the basic flow:
1. When a PSTN call comes in to the ATA device and is unanswered (after a
configurable number of rings), then the ATA device takes the FXO port off hook.
2. The SPA3102 prompts the caller to enter the PIN number followed by the # key.
3. The SPA3102 compares the PIN to the configured PSTN PIN values.
If the PIN matches one of the configured PSTN PIN values, then the
SPA3102 plays dial tone. The caller enters the telephone number and the
collected digits are processed by the dial plan associated with the PIN
number. (These dial plans are configured on the Voice Voice tab > PSTN
Line page, Dial Plans section.)
If the PIN does not match one of the configured PSTN PIN values, then the
ATA device plays the reorder tone and then takes the FXO port on-hook.
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Configuring the PSTN (FXO) Gateway on the SPA3102
How PSTN-To-VoIP Calls Work
NOTE For information about configuring the timer values for the above scenarios, see
FXO (PSTN) Timer Values (sec) section, on page 209.
For information about configuring caller authentication on the SPA3102, see VoIP-
To-PSTN Gateway Setup section, on page 206.
Terminating Gateway Calls
There are two call legs in a PSTN gateway call: the PSTN call leg and the VoIP call
leg. A gateway call is terminated when either call leg is ended. When the call
terminates, the FXO port goes on-hook so the PSTN line is available for use. The
ATA device detects that the PSTN call leg is ended when one of the following
conditions occurs during a call:
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The PSTN Line voltage drops to a very low value (this occurs if the line is
disconnected from the PSTN service or if the PSTN switch provides a CPC
signal).
A polarity reversal or disconnect tone is detected at the FXO port.
There is no voice activity for a configurable period of time in either direction
at the FXO port.
When any of the above conditions occur, the ATA device takes the FXO port on
hook and sends a BYE request to end the VoIP call leg. On the other hand, when
the ATA device receives a SIP BYE from the VoIP during a call, it takes the FXO port
on hook to end the PSTN call leg.
In addition, the ATA device can also send a refresh signal periodically to the VoIP
call leg to determine whether the call leg is still up. If a refresh operation fails, the
ATA device ends both call legs.
On the SPA3102, these settings can be configured on the Voice tab > PSTN Line
page. For more information, see PSTN Disconnect Detection section, on
page 211.
On the SPA8800, these settings can be configured on the Voice tab > Phone page.
For more information, see FXS Port Polarity Configuration section, on
page 177.
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Configuring the PSTN (FXO) Gateway on the SPA3102
VoIP Outbound Call Routing (SPA3102)
VoIP Outbound Call Routing (SPA3102)
On the SPA3102, calls made from Line 1 are routed through the configured Line 1
service provider, by default. You can override this behavior by IP dialing, through
which the calls can be routed to any IP address that the user enters. The SPA3102
allows flexible call routing with four sets of gateway parameters and configurable
dial plans. These settings are configured on the SPA3102 Voice tab > Line page.
See Gateway Accounts section (SPA3102), on page 174.
NOTE The PBX handles outbound call routing for the SPA8800.
You can specify Gateways 1 to 4 in a dial plan by using the identifiers gw1, gw2,
gw3, or gw4. Also, gw0 represents the internal PSTN gateway via the FXO port.
You can specify in the dial plan to use gw ( = 0,1,2,3,4) when making certain calls.
In general, you can specify any gateway address in the dial plan. In addition, three
parameters are added that can be used with call routing:
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usr: User-id used for authentication with the given gateway
pwd: Password used for authentication with the given gateway
nat: Enable or disable NAT mapping when calling the gateway
The following table lists some examples.
Example Description
Dial 9 to start outside dial tone, followed by one or
more digits, and route the call to Gateway 1.
Route 911 and 311 calls to the local PSTN gateway
Dial 8 to start outside dial tone, prepend 1408
followed by seven digits, and route the call to
pstn.cisco.com:5061, with user-id = joe, and pwd =
bell_pwd, and enable NAT mapping
Dial 8 to start outside dial tone, prepend 1408
followed by seven digits, and route the call to
Gateway 2, but use the given port, user-id, and
password, and no pstn.cisco.com:5061, and with
user-id = Alex Bell and pwd = bell_pwd, and
disable NAT mapping
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Configuring the PSTN (FXO) Gateway on the SPA3102
Configuring VoIP Failover to PSTN
You can set up multiple PSTN gateways at different locations and configure Line 1
to use a different gateway when dialing specific numbers.
Configuring VoIP Failover to PSTN
When power is disconnected from the ATA, the FXS port is connected to the FXO
port. In this case, the telephone attached to the FXS port is electrically connected
to the PSTN service via the FXO port. When power is applied to the ATA device,
the FXS port is disconnected from the FXO port. However, if the PSTN line is in use
when the power is applied to the ATA device, the relay is not flipped until the
PSTN line is released. This feature ensures that the ATA device does not interrupt
any call in progress on the PSTN line.
When Line 1 VoIP service is down (because of registration failure or loss of
network link), the ATA device can be configured to automatically route all outbound
calls to the internal gateway using the parameter listed below.
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For SPA3102, you can configure this setting on the Voice tab > Line page. For
SPA8800, you can configure this setting on the Voice tab > Phone page. For more
information, see VoIP Fallback to PSTN section (SPA3102 and SPA8800), on
page 175.
Sharing One VoIP Account Between the FXS and PSTN Lines
(SPA3102)
On the SPA3102, both the FXS (Line 1) and FXO (PSTN Line) can receive incoming
calls for a single VoIP account if they are different ports. Consider the following
points:
If the service provider allows multiple registration contacts and
simultaneous ringing, both lines can register periodically with the service
provider. In this case, both lines receive inbound calls to this VoIP account.
The PSTN Line should be configured with a sufficiently long answer delay
before the call is automatically answered to allow for the function of the
PSTN gateway.
If the service provider does not allow more than one register contact, the
PSTN Line should not register. In this case, only Line 1 rings on the inbound
call to this VoIP account because it is the only line registered with the
service provider.
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Configuring the PSTN (FXO) Gateway on the SPA3102
Other Options
Line 1 can have the call forwarded to the PSTN Line after a few seconds
using the Call-Forward-On-No-Answer feature with gw0 as the forward
destination. Similarly, Line 1 can apply Call-Forward-All, Call-Forward-OnBusy, and Call-Forward-Selective feature, and direct the caller to the PSTNGateway.
Only PIN authentication is allowed when a VoIP caller is forwarded to the
PSTN-gateway from Line 1. If HTTP Authentication is used, the caller is not
authenticated.
When using the Forward-To-GW0 feature, you can forward the caller to a
specific PSTN number, using the syntax < >@gw0 in the
forward destination. When using this with Call-Forward-Selective, you can
develop some interesting applications. For example, you can forward all
callers with 408 area code to 14081234567, or all callers with 800 area
code to 18005558355 (This is the number for Tell Me). When this syntax is
used, authentication is not used and the target PSTN number is
automatically dialed by the ATA device after the caller is forwarded to gw0.
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Other Options
This section describes other options provided by the SPA3102 and the SPA8800:
PSTN Call to Ring Line 1 (SPA3102) section on page 94
Symmetric RTP (SPA3102 and SPA8800) section on page 94
Call Progress Tones (SPA3102 and SPA8800) section on page 95
PSTN Call to Ring Line 1 (SPA3102)
On the SPA3102, this feature allows a PSTN caller to ring Line 1. When the PSTN
line rings, the PSTN Line makes a local VoIP call to Line 1. If Line 1 is busy, it stops.
After a given number of rings, the VoIP gateway picks up the call.
Symmetric RTP (SPA3102 and SPA8800)
The parameter is used to send audio RTP to the source IP and
port of the inbound RTP packets. This facilitates NAT traversal. See SIP
Parameters section, on page 122.
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Call Scenarios
Call Progress Tones (SPA3102 and SPA8800)
The SPA3102 and the SPA8800 have configurable call progress tones. Call
progress tones are generated locally on the ATA, 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 the following specifications:
Number of frequency components
Frequency and amplitude of each component
Cadence information
When one VoIP account is shared between the FXS and PSTN Lines, the following
parameters are recommended to be set. For more information, see the Regional
page in the ATA Voice Field Reference, on page110.
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Call Scenarios
Call Progress Tone Description
VoIP PIN Tone This tone is played to prompt a VoIP caller to enter a PIN number.
PSTN PIN Tone This tone is played to prompt a PSTN caller to enter a PIN
number.
Outside Dial Tone During two-stage PSTN-gateway dialing and with a dial plan
assigned, the ATA device collects digits from the VoIP caller and
processes the number using the dial plan. The ATA device plays
the to prompt the VoIP caller to enter the
PSTN number. This tone should be specified to sound different
from the PSTN dial tone.
This section describes some typical scenarios where the ATA device can be
applied. Some terms are introduced in the first few sections and reused in later
sections. This section includes the following topics:
PSTN to VoIP Call with and Without Ring-Thru section on page 96
VoIP to PSTN Call With and Without Authentication section on page 96
Call Forwarding to PSTN Gateway (SPA3102 and SPA8800) section on
page 98
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Call Scenarios
PSTN to VoIP Call with and Without Ring-Thru
The PSTN caller calls the PSTN line connected to the FXO port. Ring-Thru is
disabled. After the call rings for a delay equal to the value in ,
the VoIP gateway answers the call and prompts the PSTN caller to enter a PIN
number (assuming PIN authentication is enabled). After a valid PIN is entered, the
caller is prompted to dial the VoIP number. A dial plan is selected according to the
PIN number entered by the caller. If authentication is disabled, the default PSTN
dial plan is used. Note than the dial plan choice cannot be 0 for a PSTN caller.
NOTE A in terms of Caller ID (ANI) patterns can be configured into the
ATA device to automatically grant access to the PSTN caller without entering the
PIN. In this case, the default PSTN dial plan is also used.
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The same scenario can be implemented using Ring-Thru. When the PSTN line
rings, Line 1 rings also. This feature is called . If Line1 is picked up before
the VoIP gateway auto-answers, it is connected to the PSTN call. Line 1 hears a
call waiting tone if it is already connected to another call.
VoIP to PSTN Call With and Without Authentication
This section describes three scenarios with and without authentication and
includes the following topics:
Using PIN Authentication (SPA3102) section on page 96
Using HTTP Digest Authentication (SPA3102) section on page 97
Without Authentication (SPA3102 and SPA8800) section on page 98
Using PIN Authentication (SPA3102)
This scenario assumes that the PSTN Line has a different VoIP account than the
Line 1 account. The VoIP caller calls the FXO number, which auto-answers after
. The ATA device then prompts the VoIP caller for a PIN. When a
valid PIN is entered, the ATA plays the and prompts the caller to
dial the PSTN number.
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Configuring the PSTN (FXO) Gateway on the SPA3102
Call Scenarios
The number dialed is processed by the dial plan corresponding to the VoIP caller.
If the dial plan choice is 0, no dial plan is needed and the user hears the PSTN dial
tone right after the PIN is entered. If the dial plan choice is not 0, the final number
returned from the dial plan after the complete number is dialed by the caller is
dialed to the PSTN. The caller does not hear the PSTN dial tone (except for a little
leakage before the first digit of the final number is auto-dialed by the ATA device).
If the PSTN Line is busy (off-hook, ringing, or PSTN line not connected) when the
VoIP caller calls, the ATA device replies with 503. If the PIN number is invalid or
entered after the VoIP call leg is connected, the ATA device plays the reorder tone
to the VoIP caller and eventually ends the call when the reorder tone times out.
NOTE If is specified and the VoIP caller ID does not match any of the
given patterns, the ATA device rejects the call with a 403. This rule applies
regardless of the authentication method, even when the source IP address of the
INVITE request is in the .
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Using HTTP Digest Authentication (SPA3102)
The same scenario can be implemented with HTTP digest authentication when
the calling device supports the configuration of a auth-ID and password to access
the ATA device PSTN gateway. When the VoIP caller calls the PSTN Line, the ATA
device challenges the INVITE request with a 401 response. The calling device
should then provide the correct credentials in a subsequent retry of the INVITE,
computed with the auth-ID and password using MD5.
If the credentials are correct, the target number specified in the user-id field of the
INVITE Request-URI is processed by the dial plan corresponding to the VoIP user
(assuming the dial plan choice is not 0). The final number is then auto-dialed by the
ATA device.
If the credentials are incorrect, the ATA device challenges the INVITE again. If the
auth-ID does not exist in the ATA device configuration, the ATA device replies 403
to the INVITE. If the target number is invalid according to the corresponding dial
plan, the ATA device also replies 403 to the INVITE. Again, if the PSTN Line is busy
at the time of the call, the ATA device replies 503.
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Configuring the PSTN (FXO) Gateway on the SPA3102
Call Scenarios
NOTE HTTP Digest Authentication is one way to perform one-stage dialing of a VoIP-To-
PSTN call. The other way is with no authentication require. However, if the target
number is not specified in the Request-URI or the number matches the account
user-id of the PSTN Line, the call reverts to two-stage dialing.
Without Authentication (SPA3102 and SPA8800)
This scenario can also be implemented without authentication, using one-stage or
two-stage dialing, as in the HTTP Authentication case. The default VoIP caller dial
plan is used in this scenario. Authentication is performed when the method is none
or when the source IP address of the inbound INVITE matches one of the
patterns.
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Call Forwarding to PSTN Gateway (SPA3102 and SPA8800)
This section describes a number of scenarios that forward calls to the PSTN
gateway. It includes the following topics:
Forward-On-No-Answer to the PSTN Gateway section on page 98
Forward-All to the PSTN gateway section on page 99
Forward to a Particular PSTN Number section on page 99
Forward-On-Busy to PSTN Gateway or Number section on page 99
On the SPA3102, you can configure Call Forward settings on the User page. On
SPA8800, the same parameters are set on the Phone page. For field descriptions,
see Call Forward Settings section.
Forward-On-No-Answer to the PSTN Gateway
In this scenario, Line 1 is configured to to the PSTN Gateway.
The scenario is implemented by setting User 1 to forward to gw0 on no answer,
with
The caller calls Line 1 and if Line 1 is not picked up after six seconds, the PSTN
Line picks up the call and the call reverts to a PSTN-Gateway call, as described
above. In this case, HTTP authentication is not allowed because Line 1 does not
authenticate inbound INVITE requests. If you need to authenticate the VoIP caller in
this case, you must select the PIN authentication method, or else the caller is
authenticated.
set to six seconds.
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Call Scenarios
NOTE If the PSTN Line is busy at the moment of the forward, it does not answer the VoIP
call. The call forward rule is ignored and Line 1 continues to ring.
Forward-All to the PSTN gateway
In this scenario, Line 1 is configured with parameter to the PSTN
gateway.This scenario is the same the previous case, except the FXO picks up the
Line 1 call immediately.
If the PSTN Line is busy at the moment of the call, the PSTN Line does not pick up
the call, the call forward rule is ignored, and Line 1 continues to ring.
Forward to a Particular PSTN Number
6
In this scenario, the forward destination is set to < >. This is
the same as in the previous examples, except that the ATA device automatically
dials the given target number on the PSTN line right after it answers the VoIP call
leg. This is a special case of one-stage dialing where the target number is
specified in the configuration. The caller is not authenticated in this case
regardless of the authentication method. However, the caller is still limited by the
parameter
Forward-On-Busy to PSTN Gateway or Number
This scenario is similar to the previous cases of call forwarding to gw0, but this
applies when Line 1 is active.
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ATA Routing Field Reference
This chapter describes the settings that you can configure under the Router and
Network tabs in the administration web server pages.
NOTE This information applies to the SPA2102, SPA3102, SPA8000, and SPA8800
models. To configure router settings for the PAP2T, see the user guide for the router.
After you click the Router tab on the SPA2102, SPA3102, or the Network tab on
the SPA8000 and SPA8800, you can choose the following pages:
Router Status page, on page 100
A
WAN Status page, on page 102
LAN Status page, on page 106
Application page, on page 107
NOTE Not all fields listed may be applicable to your ATA device or your setup.
Router Status page
You can use the Router tab > Status page to view information about the Router.
The Status page includes the following sections:
Product Information section, on page101
System Status section, on page 101
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