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Two-Stage Dialing (SPA3102), Configuring the PSTN FXO Gateway on the SPA3102

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Configuring the PSTN (FXO) Gateway on the SPA3102 How VoIP-To-PSTN Calls Work 6 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 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. The call scenarios may involve the following types of callers: • VoIP caller-Someone who calls the ATA device via VoIP to obtain PSTN service • VoIP user-A VoIP caller that has a user account (user-id and password) on the SPA3102 • PSTN caller-Someone who calls the ATA device from the PSTN to obtain VoIP service VoIP callers can be authenticated by one of the following methods: • No Authentication-All callers are accepted for service. • PIN-Caller is prompted to enter a PIN right after the call is answered. • HTTP digest-SIP INVITE must contain a valid authorization header. PSTN callers can be authenticated by one of the following methods: • No authentication-All callers are accepted for service. • PIN-Caller is prompted to enter a PIN right after the call is answered. Cisco Small Business ATA Administration Guide 94

Section	Page
About This Document	11
Introducing Cisco Small Business Analog Telephone Adapters	16
Comparison of ATA Devices	17
ATA Connectivity Requirements	20
PAP2T Connectivity	21
SPA2102 Connectivity	22
SPA3102 Connectivity	23
SPA8000 Connectivity	24
SPA8800 Connectivity	25
ATA Software Features	26
Voice Supported Codecs	26
SIP Proxy Redundancy	28
Other ATA Software Features	28
Basic Administration and Configuration	36
Basic Services and Equipment Required	36
Downloading Firmware	37
Basic Installation and Configuration	37
Upgrading the Firmware for the ATA Device	37
Setting up Your ATA Device	38
Using the Administration Web Server	39
Connecting to the Administration Web Server	40
Setting Up the WAN Configuration for Your ATA Device	40
Registering to the Service Provider	42
Advanced Configurations	43
Upgrading, Rebooting, and Resyncing Your ATA Device	43
Upgrade URL	43
Resync URL	44
Reboot URL	44
Provisioning Your ATA Device	45
Provisioning Capabilities	45
Configuration Profile	46
Configuring Your System for ITSP Interoperability	47
Network Address Translation (NAT) and Voice over IP (VoIP)	47
NAT Mapping with Session Border Controller	48
NAT Mapping with SIP-ALG Router	48
Configuring NAT Mapping with a Static IP Address	48
Configuring NAT Mapping with STUN	50
Determining the Router’s NAT Mechanism	52
Firewalls and SIP	53
Configuring SIP Timer Values	53
Configuring Voice Services	54
Supported Codecs	54
Using a FAX Machine	55
Fax Troubleshooting	56
Managing Caller ID Service	58
Silence Suppression and Comfort Noise Generation	60
Configuring Dial Plans	61
About Dial Plans	61
Editing Dial Plans	70
Secure Call Implementation	71
Enabling Secure Calls	71
Secure Call Details	72
Using a Mini-Certificate	73
Generating a Mini Certificate	73
SIP Trunking and Hunt Groups on the SPA8000	75
About SIP Trunking	76
Setting the Trunk Group Call Capacity	78
Inbound Call Routing for a Trunk Group	78
Contact List for a Trunk Group	79
Outgoing Call Routing for a Trunk Group	81
Configuring a Trunk Group	82
Trunk Group Management	83
Setting the Hunt Policy	85
Additional Notes About Trunk Groups	85
Configuring Music on Hold	86
Using the Internal Music Source for Music On Hold	86
Using the Internal Music Source	86
Changing the Music File for the Internal Music Source	87
Configuring a Streaming Audio Server	88
About the Streaming Audio Server	88
Configuring the Streaming Audio Server	90
Using the IVR with an SAS Line	91
Configuring the PSTN (FXO) Gateway on the SPA3102	92
Connecting to PSTN and VoIP Services	92
How VoIP-To-PSTN Calls Work	93
One-Stage Dialing (SPA3102 and SPA8800)	93
Two-Stage Dialing (SPA3102)	94
How PSTN-To-VoIP Calls Work	95
Terminating Gateway Calls	96
VoIP Outbound Call Routing (SPA3102)	97
Configuring VoIP Failover to PSTN	98
Sharing One VoIP Account Between the FXS and PSTN Lines (SPA3102)	98
Other Options	99
PSTN Call to Ring Line 1 (SPA3102)	99
Symmetric RTP (SPA3102 and SPA8800)	99
Call Progress Tones (SPA3102 and SPA8800)	100
Call Scenarios	100
PSTN to VoIP Call with and Without Ring-Thru	101
VoIP to PSTN Call With and Without Authentication	101
Call Forwarding to PSTN Gateway (SPA3102 and SPA8800)	103
ATA Routing Field Reference	105
Router Status page	105
Product Information section	106
System Status section	106
WAN Status page	107
Internet Connection Settings section	107
Static IP Settings section	108
PPPoE Settings section	108
Optional Settings section	109
MAC Clone Settings section	110
Remote Management section	110
QOS Settings section	110
VLAN Settings section	111
LAN Status page	111
Networking Service section	111
LAN Networking Settings section	112
Static DHCP Lease Settings section	112
Application page	112
Port Forwarding Settings section	113
DMZ Settings section	113
Miscellaneous Settings section	114
System Reserved Ports Range section	114
ATA Voice Field Reference	115
Info page	116
Product Information section	116
System Status section	117
Line Status section	117
System Information section (PAP2T)	120
PSTN Line Status section (SPA3102)	120
Trunk Status section (SPA8000)	123
System page	124
System Configuration section	124
Internet Connection Type section (PAP2T)	125
Optional Network Configuration section (PAP2T)	125
Miscellaneous Settings section (not used with PAP2T)	126
SIP page	127
SIP Parameters section	127
SIP Timer Values (sec) section	129
Response Status Code Handling section	131
RTP Parameters section	132
SDP Payload Types section	134
NAT Support Parameters section	135
Trunking Parameters section (SPA8000)	138
Regional page	139
Call Progress Tones section	140
Distinctive Ring Patterns section	142
Distinctive Call Waiting Tone Patterns section	143
Distinctive Ring/CWT Pattern Names section	144
Ring and Call Waiting Tone Spec section	145
Control Timer Values (sec) section	146
Vertical Service Activation Codes section	148
Vertical Service Announcement Codes section (SPA2102, SPA8000)	154
Outbound Call Codec Selection Codes section	154
Miscellaneous section	156
Line page	160
Line Enable section	161
Streaming Audio Server (SAS) section	162
NAT Settings section	163
Network Settings section	164
SIP Settings section	165
Call Feature Settings section	168
Proxy and Registration section	169
Subscriber Information section	171
Supplementary Service Subscription section	172
Audio Configuration section	174
Gateway Accounts section (SPA3102)	179
VoIP Fallback to PSTN section (SPA3102 and SPA8800)	180
Dial Plan section	180
FXS Port Polarity Configuration section	182
VoIP-to-PSTN Gateway Setup section (SPA8800)	183
PSTN-To-VoIP Gateway Setup section (SPA8800)	183
FXO Timer Values (sec) section (SPA8800)	184
PSTN Disconnect Detection section (SPA8800)	185
International Control section (SPA8800)	188
Call Forward, Speed Dial, Supplementary Services, and Ring Settings (SPA8000 and SPA8800)	189
Trunk Group page (SPA8000)	190
Line Enable section	190
Network Settings section	190
SIP Settings section	191
Subscriber Information section	194
Dial Plan section	196
NAT Settings section	196
Proxy and Registration section	197
PSTN Line page (SPA3102)	198
Line Enable section	199
NAT Settings section	199
Network Settings section	200
SIP Settings section	201
Proxy and Registration section	204
Subscriber Information section	205
Audio Configuration section	206
Dial Plans section	210
VoIP-To-PSTN Gateway Setup section	210
VoIP Users and Passwords (HTTP Authentication) section	212
Ring Settings section	213
FXO (PSTN) Timer Values (sec) section	213
PSTN Disconnect Detection section	215
International Control (Settings) section	218
User page	220
Call Forward Settings section	221
Selective Call Forward Settings section	222
Speed Dial Settings section	222
Supplementary Service Settings section	223
Distinctive Ring Settings section	224
Ring Settings section	225
PSTN User page (SPA3102)	226
PSTN-To-VoIP Selective Call Forward Settings section	226
PSTN-To-VoIP Speed Dial Settings section	227
PSTN Ring Thru Line 1 Distinctive Ring Settings section	227
PSTN Ring Thru Line 1 Ring Settings section	227
Troubleshooting	229
Environmental Specifications	233
PAP2T	233
SPA2102	234
SPA3102	234
SPA8000	235
SPA8800	235
WRTP54G	236
Where to Go From Here	237
Product Resources	237

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Configuring the PSTN (FXO) Gateway on the SPA3102

How VoIP-To-PSTN Calls Work

Cisco Small Business ATA Administration Guide

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 <

User ID

> of the PSTN Line. A different

user-id in the Request-URI is treated as a request for one-stage dialing if one-

stage 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 one-

stage 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.

The call scenarios may involve the following types of callers:

•

VoIP caller—Someone who calls the ATA device via VoIP to obtain PSTN

service

•

VoIP user—A VoIP caller that has a user account (user-id and password) on the

SPA3102

•

PSTN caller—Someone who calls the ATA device from the PSTN to obtain VoIP

service

VoIP callers can be authenticated by one of the following methods:

•

No Authentication—All callers are accepted for service.

•

PIN—Caller is prompted to enter a PIN right after the call is answered.

•

HTTP digest—SIP INVITE must contain a valid authorization header.

PSTN callers can be authenticated by one of the following methods:

•

No authentication—All callers are accepted for service.

•

PIN—Caller is prompted to enter a PIN right after the call is answered.