Cisco Systems PAP2T, SPA2102, SPA3102, WRP400, SPA8000 manual Two-Stage Dialing

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Configuring the PSTN (FXO) Gateway on the SPA3102

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How VoIP-To-PSTN Calls Work

 

 

 

 

 

 

 

 

 

 

NOTE If Authentication is disabled, a default dial plan is used for all unknown VoIP users.

Two-Stage Dialing

In two-stage dialing, the ATA device takes the FXO port off-hook but does not automatically dial any digits after accepting the call. To invoke two-stage dialing, the VoIP caller should INVITE the PSTN Line without the user-id in the Request-URI or with a user-id that matches exactly the <User ID n> 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 ATA device (as if no user-id is given) if one-stage dialing is disabled.

NOTE If Authentication is disabled, a default dial plan is assigned to all VoIP callers.

HTTP Digest Authentication can be also used for two-stage dialing, as in 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).

If PIN Authentication is enabled, the VoIP caller is prompted to enter a PIN number after the ATA device answers the call. The PIN number must end with a # key. The inter-PIN-digit timeout is 10 seconds (not configurable). Up to eight VoIP caller PIN numbers can be configured on the ATA device. A dial plan can be selected for each PIN number. If the caller enters a wrong PIN or the ATA device times out waiting for more PIN digits, the ATA device tears down the call immediately with a BYE request.

NOTE When the source address of the INVITE is 127.0.0.1, authentication is automatically disabled because this is a call by the local user. This applies to both one-stage and two-stage dialing.

ATA Administration Guide

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Contents Administration Guide OL-17901-01 Basic Administration and Configuration Contents About This DocumentContents Configuring Your System for Itsp InteroperabilityConfiguring Voice Services Configuring the Pstn FXO Gateway on the SPA3102 Configuring Music on HoldCall Scenarios 105 Appendix a ATA Routing Field Reference 111Info 122 Appendix B ATA Voice Field Reference 121Line 165 Pstn Line page SPA3102 190 Appendix F Where to Go From Here 244 Purpose AudiencePreface Typographic Meaning Element FirmwareDocument Conventions Product Firmware VersionOrganization Chapter ContentsThese appendices provide information about other Press Enter Click Search ATA Administration Guide Introducing Cisco Small Business Analog Telephone Adapters Introducing Cisco Small Business Analog Telephone Adapters Comparison of ATA DevicesIntroducing Cisco Small Business Analog Telephone Adapters How ATAs Provide Voice Connectivity ATA Connectivity Requirements PAP2T Connectivity SPA2102 Connectivity SPA3102 Connectivity SPA8000 SPA8000 ConnectivityVoice Supported Codecs ATA Software FeaturesAudio codec SIP Proxy Redundancy Other ATA Software FeaturesLine tabs. See ATA Voice Field Reference, on page121 Dtmf ATA Voice Field Reference, on page121See Configuring Dial Plans, on page 61 for more Tabs. See ATA Voice Field Reference, on page121 Voice Field Reference, on page121 Register Retry is configured in the SIP tab. See ATA Voice Introducing Cisco Small Business Analog Telephone Adapters Basic Services and Equipment Required Basic Administration and ConfigurationDownloading Firmware Basic Installation and Configuration Basic Administration and ConfigurationUpgrading the Firmware for the ATA Device Setting up Your ATA Device Using the Administration Web Server Connecting to the Administration Web Server Setting Up the WAN Configuration for Your ATA DeviceClick Network tab WAN Setup For Dhcp For Static IP AddressingFor PPPoE Registering to the Service Provider Upgrading, Rebooting, and Resyncing Your ATA Device Advanced ConfigurationsUpgrade URL Resync URL Reboot URL Provisioning Your ATA DeviceProvisioning Capabilities Configuration Profile Basic Administration and Configuration Network Address Translation NAT and Voice over IP VoIP Configuring Your System for Itsp InteroperabilityNAT Mapping with SIP-ALG Router Configuring NAT Mapping with a Static IP AddressConfiguring Your System for Itsp Interoperability NAT Mapping with Session Border ControllerClick Submit All Changes Configuring NAT Mapping with Stun Configuring Your System for Itsp Interoperability Click Voice tab System Firewalls and SIP Configuring SIP Timer ValuesPAP2T / SPA2102 / SPA3102 / SPA8000 Supported CodecsConfiguring Voice Services Using a FAX Machine SPA2102, SPA3102 or SPA8000FAXPassthruMethod ReINVITE Fax Troubleshooting FSK Managing Caller ID ServiceDefault is BellcoreN.Amer, China Parameter Tab Description and ValueCAS Silence Suppression and Comfort Noise Generation About Dial Plans Configuring Dial Plans2 3 4 5 6 7 8 9 Digit Sequence FunctionDigit Sequence Examples Local dialing with seven-digit number Blocked number Terminating Event Processing Acceptance and Transmission the Dialed DigitsDial Plan Timer Off-Hook Timer Interdigit Long Timer Incomplete Entry Timer Syntax 2 sequence Ss Syntax 1 Ss, dial planClick Voice tab Regional Editing Dial PlansConfiguring Voice Services Enabling Secure Calls Secure Call ImplementationSecure Call Details Using a Mini-Certificate Genmc ca-key user-name user-id expire-date Generating a Mini CertificateExample SIP Trunking and Hunt Groups on the SPA8000 About SIP Trunking Itsp Logical Block Diagram of SIP TrunkingInbound Call Routing for a Trunk Group Setting the Trunk Group Call CapacitySyntax line,line,line…,hunt=hrule,cfwd=target Contact List for a Trunk Group3,4,5,6,7,8,hunt=re*1 Examples?,hunt=ra121,cfwd=14085550123 Outgoing Call Routing for a Trunk GroupConfiguring a Trunk Group Trunk Group Management Setting the Hunt Policy Additional Notes About Trunk Groups Using the Internal Music Source Using the Internal Music Source for Music On HoldChanging the Music File for the Internal Music Source Configuring Music on HoldAbout the Streaming Audio Server Configuring a Streaming Audio ServerConfiguring Music on Hold Configuring the Streaming Audio Server Using the IVR with an SAS Line Connecting to Pstn and VoIP Services Configuring the Pstn FXO Gateway on the SPA3102Configuring the Pstn FXO Gateway on the SPA3102 How VoIP-To-PSTN Calls WorkOne-Stage Dialing Authentication Parameters Web Description Values Two-Stage Dialing Parameters for Two-Stage Dialing Web Description Values How PSTN-To-VoIP Calls WorkParameter Web Description Values Terminating Gateway CallsIs Medium VoIP Outbound Call Routing Example Description Configuring VoIP Failover to PstnSharing One VoIP Account Between the FXS and Pstn Lines Parameter Web Description ValueYes Other Options Pstn Call to Ring LineSymmetric RTP Call Progress Tones Call ScenariosPstn to VoIP Call with and Without Ring-Thru VoIP to Pstn Call With and Without AuthenticationUsing Http Digest Authentication Without Authentication Forward-On-No-Answer to the Pstn Gateway Call Forwarding to Pstn GatewayForward to a Particular Pstn Number Router Status ATA Routing Field ReferenceProduct Information section System Status sectionATA Routing Field Reference Internet Connection Settings section WAN SetupPPPoE Settings section Static IP Settings sectionDHCP, and DHCP/Manual Optional Settings sectionRemote Management section MAC Clone Settings sectionQOS Settings section On or On when Phone is Use defaultVlan ID LAN SetupVlan Settings section Networking Service sectionLAN Networking Settings section Static Dhcp Lease Settings sectionApplication DMZ Settings section Port Forwarding Settings sectionSystem Reserved Ports Range section Miscellaneous Settings sectionATA Voice Field Reference ATA Voice Field Reference InfoLine Status section Idle ATA Administration Guide 125 System Information section PAP2T Pstn Line Status section SPA3102Dhcp Pstn Disconnect Tone VoIP State May take one of the following values Trunk Status section SPA8000 System System Configuration sectionInternet Connection Type section PAP2T Optional Network Configuration section PAP2TTable, Manual/DHCP, and DHCP/Manual Miscellaneous Settings section not used with PAP2TSIP Parameters section SIPDefault is application/hook-flash Default is application/dtmf-relaySIP T4 SIP Timer Values sec sectionSIP T1 SIP T2ATA Administration Guide 136 SIT3 RSC Response Status Code Handling sectionSIT1 RSC SIT2 RSCSIT4 RSC RTP Parameters sectionATA Administration Guide 139 SDP Payload Types section NAT Support Parameters section EXT IP ATA Administration Guide 143 Trunking Parameters section SPA8000 Regional Call Progress Tones section Default is 914@-16,1371@-16,1777@-1620.274/0 Default is 440@-19,480@-19*1/1/1+2Default is 600@-16 1.25/.25/1 Default is 985@-16,1428@-16,1777@-1620.380/0Distinctive Ring Patterns section Distinctive Call Waiting Tone Patterns section Default is 30.3/.1,.3/.1,.1/9.1 Default is 30.1/.1, .3/.1, .1/9.3Distinctive Ring/CWT Pattern Names section Ring and Call Waiting Tone Spec section Control Timer Values sec sectionATA Administration Guide 152 Vertical Service Activation Codes section ATA Administration Guide 154 ATA Administration Guide 155 ATA Administration Guide 156 ATA Administration Guide 157 ATA Administration Guide 158 Outbound Call Codec Selection Codes section Vertical Service Announcement Codes section SPA2102, SPA8000ATA Administration Guide 160 Miscellaneous section 600, 900, 600+2.16uF, 900+2.16uF, 270+750150nF220+850120nF, 220+820115nF, or 200+600100nF ATA Administration Guide 162 ATA Administration Guide 163 Use, bell 202 or Caller ID Method Following choices are availableLine Streaming Audio Server SAS section Line Enable sectionNAT Settings section Network Settings section Or disableMedium, high, very high, or extremely high Field Description SIP Settings sectionSIP Guid Configured Proxy or Outbound Proxy if Use Outbound Call Feature Settings section FromURL Outbound Proxy and Use OB Proxy in Dialog parameters Proxy and Registration sectionSubscriber Information section Supplementary Service Subscription section ATA Administration Guide 176 ATA Administration Guide 177 Gateway Accounts section SPA3102 Audio Configuration sectionVoIP Fallback to Pstn section SPA3102 Dial Plan sectionDial Plan Entry Functionality 9xxxxxxS0xxxxxxxxxxxx Default is *xx3469110002-9xxxxxx1xxx2Trunk Group page SPA8000 FXS Port Polarity Configuration sectionVoice tab Trunk Group ATA Administration Guide 183 ATA Administration Guide 184 ATA Administration Guide 185 Inbound calls When the limit is reached, the Trunk SUA Number cfwd=14089993326 #0-9A-D ATA Administration Guide 189 Pstn Line page SPA3102 Voice tab Pstn Line ATA Administration Guide 192 ATA Administration Guide 193 This field is not available with the PAP2T. The Global Proxy SIP proxy server for all outbound requests Outbound Proxy parameter and Use OB Proxy in Dialog is ATA Administration Guide 197 G726-16,G726-24,G726-32,G726-40,G729a, or G723 Info ATA Administration Guide 200 Dial Plans section VoIP-To-PSTN Gateway Setup section Caller 1/2/3/4/5/6/7/8 PIN VoIP Users and Passwords Http Authentication section FXO Pstn Timer Values sec section Ring Settings sectionATA Administration Guide 206 Pstn Disconnect Detection section Detect Long Silence is yes Impedance ATA Administration Guide 210 International Control Settings section ATA Administration Guide 212 User Call Forward Settings section Speed Dial Settings section Selective Call Forward Settings sectionSupplementary Service Settings section Distinctive Ring Settings section Default is New VM Available PSTN-To-VoIP Selective Call Forward Settings section PSTN-To-VoIP Speed Dial Settings sectionPstn User page SPA3102 Only Pstn Ring Thru Line 1 Ring Settings section Pstn Ring Thru Line 1 Distinctive Ring Settings sectionRtqos Feature/XML Tag Parameters ExamplesWlqos Wlwmenoack NoRtsp IgmpUpnp QOS Priorityrule QospriorityruleWLBASICSET2 WLBASICSET1Wlbasicset BASICSET1Wlsecurity Wlsecurity WLSECURITYSET1WL SECURITYSET1 Landhcp LAN DhcpWantype Fail Pattern Internet Connection Type WAN DNS Singleport Forwarding You also should configure a Dhcp Routersyslo G Troubleshooting How do I debug my ATA device? Is there a syslog? TroubleshootingHow do I save my current configuration? How do I access the ATA device if I forget my password?ATA Administration Guide 237 ATA Administration Guide 238 Environmental Specifications Environmental Specifications SPA2102SPA3102 SPA8000 WRTP54G WRP400WRTP54G Resource Link Product ResourcesWhere to Go From Here Related DocumentationDocument Title Description Intended Audience Document Title Description Intended Audience Resource Location Additional InformationSupport Contacts
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PAP2T, SPA8000, SPA3102, WRP400, SPA2102 specifications

The Cisco Systems SPA2102 is a versatile Voice over Internet Protocol (VoIP) adapter that serves as a bridge between traditional telephony systems and modern IP networks. Designed primarily for small to medium businesses, the SPA2102 is highly regarded for its reliability, ease of use, and rich feature set. This device allows users to make and receive phone calls over the internet while maintaining the ability to connect traditional analog phones.

One of the standout features of the SPA2102 is its dual-port capability. The device includes two FXS ports, allowing users to connect two separate analog telephones. This makes it an ideal choice for businesses that want to retain their existing telephony infrastructure while transitioning to a VoIP system. The ability to utilize two telephone lines simultaneously provides flexibility and convenience, especially for users in a busy office environment.

The SPA2102 leverages Session Initiation Protocol (SIP) technology, which is widely recognized for its robustness and interoperability. The support for SIP ensures that the SPA2102 can work seamlessly with various VoIP service providers, offering users a broad range of options for their telecommunication needs. In addition to SIP, the device supports various codecs, including G.711, G.726, and G.729, allowing for flexible audio quality settings and bandwidth management.

Security is also a critical aspect of the SPA2102. It incorporates advanced encryption methods, such as Secure Real-time Transport Protocol (SRTP) and Transport Layer Security (TLS), to protect voice communications from potential eavesdropping. This focus on security makes the SPA2102 a reliable choice for businesses concerned about the confidentiality of their conversations.

The device is easy to configure and manage, thanks to its web-based interface. This allows administrators to quickly set up the adapter, manage network settings, and troubleshoot any issues that may arise. Furthermore, the SPA2102 supports Quality of Service (QoS) features, ensuring that voice traffic is prioritized over other types of network traffic, which enhances call quality and reliability.

Overall, the Cisco SPA2102 is a powerful, user-friendly VoIP adapter that combines traditional telephony with modern IP technology. Its dual-port capability, support for SIP, extensive security features, and ease of configuration make it a solid choice for businesses looking to innovate their communication systems while minimizing disruption. Whether used in a small office or a larger organizational setting, the SPA2102 continues to be a reliable component of VoIP solutions.