Linksys ATA Administrator Users Guide Document Version 3.2 Corporate Headquarters Linksys 121 Theory Drive Irvine, CA 92617 USA http://www.linksys.
Linksys ATA Administrator Guide Copyright ©2007 Cisco Systems, Inc. All rights reserved.Specifications are subject to change without notice. Linksys is a registered trademark or trademark of Cisco Systems, Inc. and/or its affiliates in the U.S. and certain other countries. Other brands and product names are trademarks or registered trademarks of their respective holders.
CONTENTS Preface Document Audience i-xi Linksys Analog Telephone Adapters How This Document is Organized Document Conventions i-xii Related Documentation i-xiii Technical Support i-xi i-xii i-xiii i-xiii CHAPTER 1 Introducing Linksys Analog Telephone Adapters Overview 1-1 1-1 Ensuring Voice Quality 1-3 Audio Compression Algorithm Silence Suppression 1-4 Packet Loss 1-4 Network Jitter 1-4 Echo 1-4 Hardware Noise 1-5 End-to-End Delay 1-5 1-4 Feature Descriptions 1-5 SIP Proxy Redundancy 1-5
Contents Simple Traversal of UDP Through NAT SIP-NAT Interoperation 1-12 Where to Go From Here CHAPTER 2 Getting Started 1-12 1-13 2-1 Linksys Analog Telephone Adapters (ATAs) Caring for Your Hardware 2-2 AG310 2-2 PAP2T 2-3 RTP300 2-4 SPA1001 2-5 SPA2102 2-6 SPA3102 2-7 SPA8000 2-9 WRP400 2-10 WRTP54G 2-11 WRT54GP2 2-13 Establishing Connectivity 2-14 Bandwidth Requirements 2-14 Making the Physical Connections 2-1 2-15 Connecting the SPA8000 2-16 SPA8000 Architecture 2-16 Connectivity Requirement
Contents Provisioning Capabilities 3-4 Configuration Profile 3-4 Configuring a Dial Plan 3-5 Dial Plan Digit Sequences 3-5 Dial Plan Rules 3-6 Digit Sequence Syntax 3-6 Element Repetition 3-7 Sub-sequence Substitution 3-7 Inter-sequence Tones 3-7 Number Barring 3-7 Interdigit Timer Master Override 3-7 Local Timer Overrides 3-7 Pause 3-7 Dial Plan Examples 3-8 Dial Plan Timers 3-9 Interdigit Long Timer 3-9 Interdigit Short Timer 3-9 Dial Plans 3-9 Secure Call Implementation 3-10 Enabling Secure Calls 3-10 S
Contents Two-Stage Dialing 4-3 How PSTN-To-VoIP Calls Work 4-4 Terminating Gateway Calls 4-4 VoIP Outbound Call Routing 4-5 Configuring VoIP Failover to PSTN 4-7 Sharing One VoIP Account Between the FXS and PSTN Lines 4-7 Other Options 4-7 PSTN Call to Ring Line 1 4-8 Symmetric RTP 4-8 Call Progress Tones 4-8 Call Scenarios 4-8 PSTN to VoIP Call with and Without Ring-Thru 4-9 VoIP to PSTN Call with and Without Authentication 4-9 Using PIN Authentication 4-9 Using HTTP Digest Authentication 4-10 Witho
Contents SIP Timer Values (sec) 5-12 Response Status Code Handling RTP Parameters 5-14 SDP Payload Types 5-15 NAT Support Parameters 5-16 5-14 Regional Tab 5-19 Call Progress Tones 5-19 Distinctive Ring Patterns 5-21 Distinctive Call Waiting Tone Patterns 5-21 Distinctive Ring/CWT Pattern Names 5-22 Ring and Call Waiting Tone Spec 5-23 Control Timer Values (sec) 5-23 Vertical Service Activation Codes 5-24 Vertical Service Announcement Codes 5-28 Outbound Call Codec Selection Codes 5-28 Miscellaneous 5-30
Contents Dial Plans 5-57 VoIP-To-PSTN Gateway Setup 5-57 VoIP Users and Passwords (HTTP Authentication) Ring Settings 5-60 FXO (PSTN) Timer Values (sec) 5-60 PSTN Disconnect Detection 5-62 International Control (Settings) 5-63 5-59 User 1/2 Tab 5-65 Call Forward Settings 5-65 Selective Call Forward Settings 5-66 Speed Dial Settings 5-67 Supplementary Service Settings 5-67 Distinctive Ring Settings 5-68 Ring Settings 5-68 PSTN User Tab (SPA3102/AG310) 5-70 PSTN-To-VoIP Selective Call Forward Settings 5-70
Contents Call Hold C-7 Three-Way Calling C-7 Three-Way Ad-Hoc Conference Calling C-8 Call Return C-8 Automatic Call Back C-9 Call FWD—Unconditional C-9 Call FWD – Busy C-10 Call FWD—No Answer C-11 Anonymous Call Blocking C-11 Distinctive/Priority Ringing and Call Waiting Tone C-12 Speed Calling—Up to Eight Numbers or IP Addresses C-12 INDEX Linksys ATA Administrator Guide Document Version 3.
Contents Linksys ATA Administrator Guide x Document Version 3.
Preface This guide describes administration and use of the Linksys Analog Telephone Adapters (ATAs).
Preface How This Document is Organized • SPA3102—Voice adapter with router and PSTN connectivity • SPA8000—Voice adapter supporting up to eight FXS connections • AG310—ADSL2+ gateway with VoIP and PSTN connectivity • RTP300—IP router with two FXS ports • WRP400—Wireless-G IP router with FXS ports • WRTP54G—Wireless-G IP router with two FXS ports • WRT54GP2—Wireless-G IP router with two FXS ports How This Document is Organized This document is divided into the following chapters and appendices
Preface Related Documentation Related Documentation The following documentation provides additional information about features and functionality of Linksys ATAs: • AA Quick Guide • IVR Quick Guide • SPA Provisioning Guide The following documentation describes how to use other Linksys Voice System products: • SPA9000 Administrator Guide • LVS CTI Integration Guide • LVS Integration with ITSP Hosted Voicemail Guide • SPA900 Series IP Phones Administrator Guide • Linksys Voice over IP Product G
Preface Linksys ATA Administrator Guide xiv Document Version 3.
C H A P T E R 1 Introducing Linksys Analog Telephone Adapters This guide describes the administration and use of Linksys analog telephone adapters (ATAs).
Chapter 1 Introducing Linksys Analog Telephone Adapters Overview Figure 1-1 illustrates how the different ATAs provide voice connectivity in a VoIP network, including the SPA3102, which acts as a SIP-PSTN gateway.
Chapter 1 Introducing Linksys Analog Telephone Adapters Ensuring Voice Quality Each Linksys ATA 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. Linksys ATAs maintain the states of all the calls it terminates and makes the proper reaction to user input events (such as on/off hook or hook flash).
Chapter 1 Introducing Linksys Analog Telephone Adapters Ensuring Voice Quality Audio Compression Algorithm Speech signals are sampled, quantized, and compressed before they are packetized and transmitted to the other end. For IP Telephony, speech signals are usually sampled at 8000 samples per second with 12–16 bits per sample. The compression algorithm plays a large role in determining the voice quality of the reconstructed speech signal at the other end.
Chapter 1 Introducing Linksys Analog Telephone Adapters Feature Descriptions Hardware Noise Certain levels of noise can be coupled into the conversational audio signals because of the hardware design. The source can be ambient noise or 60 Hz noise from the power adaptor. The Linksys ATA hardware design minimizes noise coupling.
Chapter 1 Introducing Linksys Analog Telephone Adapters Feature Descriptions 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 Linksys ATA tries to contact the list of hosts in the order of their stated priority.
Chapter 1 Introducing Linksys Analog Telephone Adapters Feature Descriptions starts streaming audio to the calling party provided the FXS port is off-hook. If the FXS port is on-hook when the incoming call arrives, the Linksys ATA replies with a SIP 503 response code to indicate “Service Not Available.” If an incoming call is auto-answered, but later the FXS port becomes on-hook, the Linksys ATA does not terminate the call but continues to stream silence packets to the caller.
Chapter 1 Introducing Linksys Analog Telephone Adapters Feature Descriptions The Linksys ATA has a Network Jitter Level control setting for each line of service. The jitter level decides how aggressively the Linksys ATA 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. Other Features The following table summarizes other features provided by Linksys ATAs.
Chapter 1 Introducing Linksys Analog Telephone Adapters Technology Background Table 1-4 Linksys ATA Features Feature Description Signaling Hook Flash Event The Linksys ATA 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.
Chapter 1 Introducing Linksys Analog Telephone Adapters Technology Background Session Initiation Protocol Linksys ATAs are implemented using open standards, such as Session Initiation Protocol (SIP), allowing interoperation with all ITSPs supporting SIP. Figure 1-2 illustrates a SIP request for connection to another subscriber in the network. The requestor is called the user agent server (UAS), while the recipient is called the user agent client (UAC).
Chapter 1 Introducing Linksys Analog Telephone Adapters Technology Background A typical application of a NAT is to allow all the devices in a subscriber home network to access the Internet through a router with a single public IP address assigned by an ISP. The IP header of the packets sent from the private network to the public network is substituted by NAT with the public IP address and a port assigned by the router.
Chapter 1 Introducing Linksys Analog Telephone Adapters Technology Background With symmetric NAT all requests from the same internal IP address and port to a specific destination IP address and port are mapped to a unique external source IP address and port. If the same internal host sends a packet with the same source address and port to a different destination, a different mapping is used. Only an external host that receives a packet can send a UDP packet back to the internal host.
Chapter 1 Introducing Linksys Analog Telephone Adapters Where to Go From Here responds to a special NAT-Mapping-Discovery request by sending back a message to the source IP address/port of the request, where the message contains the source IP address/port of the original request. The Linksys ATA system can send this request when it first attempts to communicate with a SIP entity over the Internet. It then stores the mapping discovery results returned by the server.
Chapter 1 Introducing Linksys Analog Telephone Adapters Where to Go From Here • SPA900 Series IP Phones Administrator Guide • Linksys Voice over IP Product Guide: SIP CPE for Massive Scale Deployment Linksys ATA Administrator Guide 1-14 Document Version 3.
C H A P T E R 2 Getting Started This chapter provides a brief description of each Linksys ATA and describes the tools and utilities available for administration.
Chapter 2 Getting Started Linksys Analog Telephone Adapters (ATAs) • WRTP54G, page 2-11 • WRT54GP2, page 2-13 • RTP300, page 2-4 Caring for Your Hardware The Linksys ATA is an electronic device that should not be exposed to excessive heat, sun, cold or water. To clean the equipment, use a slightly moistened paper or cloth towel. Do not spray or pour cleaning solution directly onto the hardware unit.
Chapter 2 Getting Started Linksys Analog Telephone Adapters (ATAs) Table 1-5 AG310 Front Panel LED Function Power Steady green indicates power on and Internet connection. Flashing indicates not connected to the Internet, booting or firmware upgrade. Ethernet 1-4 Steady green indicates an active connection to the network. Flashing indicates traffic. Voice Status Steady green indicates that a voice call is currently in progress.
Chapter 2 Getting Started Linksys Analog Telephone Adapters (ATAs) Figure 1-5 PAP2T The following tables describe the LEDS on the front panel and the ports on the back panel of the device. Table 1-7 PAP2T Front Panel LED Function Phone 1/2 Steady green indicates active or registered connection to the ITSP through the Phone port. Flashing indicates device is in use or off hook. Ethernet Steady green indicates active connection. Flashing indicates traffic.
Chapter 2 Getting Started Linksys Analog Telephone Adapters (ATAs) Figure 1-6 RTP300 The following tables describe the LEDS on the front panel and the ports on the back panel of the device. Table 1-9 RTP300 Front Panel LED Function Ethernet 1-4 Steady green indicates an active connection to the network. Flashing indicates traffic. Phone 1/2 Steady green when telephone or fax machine has an active or registered connection to the ITSP through the Phone port.
Chapter 2 Getting Started Linksys Analog Telephone Adapters (ATAs) Figure 1-7 SPA1001 The following table describes the LEDS and ports on the back panel of the device. Table 1-11 SPA1001 Back Panel LED/Port Function Phone Connect to an analog telephone or fax machine with an RJ-11 cable. Power Connect to the 5-volt DC power adapter. Act Steady green indicates active or registered connection to the ITSP through the Phone port. Flashing indicates device is in use or off hook.
Chapter 2 Getting Started Linksys Analog Telephone Adapters (ATAs) Figure 1-8 SPA2102 The following tables describe the LEDS on the front panel and the ports on the back panel of the device. Table 1-12 SPA2102 Front Panel LED Function Power Steady green indicates power on and Internet connection. Flashing indicates not connected to the Internet, booting or firmware upgrade. Internet Steady green indicates active connection. Flashing indicates traffic.
Chapter 2 Getting Started Linksys Analog Telephone Adapters (ATAs) Figure 1-9 SPA3102 The following tables describe the LEDS on the front panel and the ports on the back panel of the device. Table 1-14 SPA3102 Front Panel LED Function Power Steady green indicates power on and Internet connection. Flashing indicates not connected to the Internet, booting or firmware upgrade. Internet Steady green indicates active connection. Flashing indicates traffic.
Chapter 2 Getting Started Linksys Analog Telephone Adapters (ATAs) SPA8000 The SPA8000 is an analog telephone adapter that supports connecting up to eight analog telephone devices, including up to four fax devices. The SPA8000 consists of four hardware modules, each of which is similar in functionality to the SPA2102. However, IP routing functionality is not provided by the SPA8000.
Chapter 2 Getting Started Linksys Analog Telephone Adapters (ATAs) Table 1-17 SPA8000 Back Panel Port Function AUX Ethernet maintenance port. Connect to a network device, such as a PC or a switch with an Ethernet cable for accessing the administration web server on the SPA8000. The attached device is assigned an IP address dynamically by the internal DHCP server provided by the SPA8000. Ethernet Connect to a switch, router or broadband (cable/DSL) modem for access to the Internet.
Chapter 2 Getting Started Linksys Analog Telephone Adapters (ATAs) The following tables describe the LEDS on the front panel and the ports on the back panel of the device Table 1-18 WRP400 Front Panel LED Function Ethernet 1-4 Steady green indicates an active connection to the network. Flashing indicates traffic. Wireless Steady green indicates an active connection to the network. Flashing indicates traffic.
Chapter 2 Getting Started Linksys Analog Telephone Adapters (ATAs) Figure 1-12 WRTP54G The following tables describe the LEDS on the front panel and the ports on the back panel of the device Table 1-20 WRTP54G Front Panel LED Function Ethernet 1-4 Steady green indicates an active connection to the network. Flashing indicates traffic. Wireless Steady green indicates an active connection to the network. Flashing indicates traffic.
Chapter 2 Getting Started Linksys Analog Telephone Adapters (ATAs) WRT54GP2 The WRP54GP2 provides an ATA with two FXS ports and a Wireless-G multiport IP router (see Figure 1-13). The WRP54GP2 provides connectivity to an analog telephone as well as Internet connectivity to a LAN with a built-in four-port switch. The IP router provides QoS support, an integrated firewall and VPN pass-through, and full routing with DHCP and NAT.
Chapter 2 Getting Started Establishing Connectivity Table 1-23 WRTP54G Back Panel Port Function Ethernet 1-3 Connect to local IP devices, such as PCs, using an Ethernet cable. Power Connect to the power supply. Establishing Connectivity This section describes how to connect the Linksys ATA hardware.
Chapter 2 Getting Started Establishing Connectivity http://www.erlang.com/calculator/lipb/ http://www.packetizer.com/voip/diagnostics/bandcalc.
Chapter 2 Getting Started Connecting the SPA8000 Connecting the SPA8000 The SPA8000 provides up to eight analog telephone connections, and is designed to function as a network endpoint.
Chapter 2 Getting Started Connecting the SPA8000 The secondary modules (Module 2, 3, and 4) obtain configuration and firmware upgrades from the primary module and are not managed directly. The primary module routes all the SIP and RTP traffic to and from the secondary modules over the SPA8000 internal LAN, which also includes the AUX maintenance port. The AUX maintenance port cannot be used to forward IP traffic.
Chapter 2 Getting Started Using the Interactive Voice Response Interface DHCP server, and the address and port is translated by the SPA8000 using Network Address Translation (NAT) and Port Address Translation (PAT). The packet must then be routed back to the internal network on the SPA8000 by the local router or the ISP router. Problems can occur with calls between phones connected to the SPA8000 when an outbound proxy or a router with hairpinning support is not available.
Chapter 2 Getting Started Using the Interactive Voice Response Interface To enter a period, use the star key (*). When entering a value, such as an IP address, to exit without entering any changes, press the * (star) key twice within half a second. Otherwise, the * is treated as a decimal point. After entering a value, such as an IP address, press the # (pound) key to indicate you have finished your selection. To save the new setting, press 1. To review the new setting, press 2.
Chapter 2 Getting Started Using the Interactive Voice Response Interface Table 1-25 IVR Options (continued) Set Network Mask 121 Enter value using numbers on the telephone key pad. Use the * (star) key when entering a decimal point. DHCP must be “Disabled,” otherwise you hear, “Invalid Option,” if you try to set this value. Requires password Check Static Gateway IP Address 130 None IVR announces the current gateway IP address of the Linksys ATA.
Chapter 2 Getting Started Using the Interactive Voice Response Interface Table 1-25 IVR Options (continued) User Factory Reset of Unit 877778 Enter 1 to confirm Enter *(star) to cancel operation Linksys ATA prompts for confirmation. After confirming, you hear “Option Successful.” Hang up. Unit reboots and all “User Changeable” configuration parameters are reset to factory default values. 73738 (RESET) Enter 1 to confirm Enter * (star) to cancel operation Linksys ATA prompts for confirmation.
Chapter 2 Getting Started Using the Administration Web Server For example, to input password test#@1234 by phone keypad, you need to press the following sequence of digits: 8378001234. 1. After entering a value, press the # (pound) key to indicate end of input. – To save value, press 1. – To review the value, press 2. – To re-enter the value, press 3. – To cancel the value entry and return to the main configuration menu, press *’ (star). Notes: – The final # key is not included in the password value.
Chapter 2 Getting Started Using the Administration Web Server Step 5 Click Admin and Advanced. The Administrator account name is admin, and the User account name is user. These account names cannot be changed. The system prompts for the Administrator account password if it has been set. If prompted, type the password provided by the ITSP and press Enter. Step 6 To view the status information for the phones, click PBX Status. Enter the appropriate login information.
Chapter 2 Getting Started Using the Administration Web Server Linksys ATA Administrator Guide 2-24 Document Version 3.
C H A P T E R 3 Configuring Linksys ATAs This chapter describes how to perform site-specific configuration required to use a Linksys ATA or to enable specific features.
Chapter 3 Configuring Linksys ATAs Initial Configuration If you use a cable modem, you may need to configure the MAC Clone Settings. (Contact your ISP for more information.) b. If your service uses a specific PC MAC address, then select yes from the Enable MAC Clone Service setting. c. Then enter the PC’s MAC address in the Cloned MAC Address field. If you are using static IP addressing, complete the following steps: a. Select Static IP from the Connection Type drop-down menu. b.
Chapter 3 Configuring Linksys ATAs Web Interface URLs Web Interface URLs The Linksys ATA web interface supports several functions through special URLs: • Upgrade • Reboot • Resync Administrator account privilege is needed for these functions. Upgrade URL The Upgrade URL lets you upgrade the Linksys ATA to the firmware specified by the URL, which can identify either a TFTP or HTTP server.
Chapter 3 Configuring Linksys ATAs Provisioning Reboot URL The Reboot URL lets you reboot the Linksys ATA. Note The Linksys ATA reboots only when it is idle. The Reboot URL is http://spa-ip-addr/admin/reboot. Provisioning This section describes the provisioning functionality of the Linksys ATA.
Chapter 3 Configuring Linksys ATAs Configuring a Dial Plan Yes Binary format profiles contain Linksys ATA parameter values and user access permissions for the parameters. By convention, the profile uses the extension .cfg (for example, spa2102.cfg). The Linksys Profile Compiler (SPC) tool compiles a plain-text file containing parameter-value pairs into a properly formatted and encrypted .cfg file.
Chapter 3 Configuring Linksys ATAs Configuring a Dial Plan • Only one candidate sequence remains, and it has been matched completely—The number is accepted and transmitted after any transformations indicated by the dial plan, unless the sequence is barred by the dial plan, in which case the number is rejected. • A timeout occurs—The digit sequence is accepted and transmitted as dialed if incomplete, or transformed as per the dial plan if complete.
Chapter 3 Configuring Linksys ATAs Configuring a Dial Plan – Ranges can be combined with other keys: e.g. [235-8*] means 2 or 3 or 5 or 6 or 7 or 8 or *. Element Repetition Any element can be repeated zero or more times by appending a period (.) to the element. Thus, “01.” matches “0”, “01”, “011”, “0111”, … and so on.
Chapter 3 Configuring Linksys ATAs Configuring a Dial Plan This syntax allows for the implementation of Hot-Line and Warm-Line services. To achieve this, one sequence in the plan must start with a pause, with a 0 delay for a Hot Line, and a non-zero delay for a Warm Line. Implicit Sequences The Linksys ATA implicitly appends the vertical code sequences entered in the administration web server Regional parameter settings to the end of the dial plan for both Line 1 and Line 2.
Chapter 3 Configuring Linksys ATAs Configuring a Dial Plan Dial Plan Timers The dial plan functionality is regulated by the following configurable parameters: • Interdigit_Long_Timer • Interdigit_Short_Timer • Dial_Plan ([1] and [2]) Interdigit Long Timer ParName Interdigit_Long_Timer Default 10 The specifies the default maximum time (in seconds) allowed between dialed digits, when no candidate digit sequence is as yet complete (see the discussion of the Dial_Plan parame
Chapter 3 Configuring Linksys ATAs Secure Call Implementation Secure Call Implementation This section describes secure call implementation with a Linksys ATA. It includes the following topics: • Enabling Secure Calls, page 3-10 • Secure Call Details, page 3-10 • Using a Mini-Certificate, page 3-11 • Generating a Mini-Certificate, page 3-11 Enabling Secure Calls A secure call is established in two stages. The first stage is no different from normal call setup.
Chapter 3 Configuring Linksys ATAs Secure Call Implementation • Mini-Certificate (252B) Upon receiving the Caller Hello, the called party responds with a Callee Hello message (base64 encoded and embedded in the message body of a SIP response to the caller’s INFO request) with similar information, if the Caller Hello message is valid. The caller then examines the Callee Hello and proceeds to the next step if the message is valid. 2.
Chapter 3 Configuring Linksys ATAs Configuring a Streaming Audio Server 9CC9aYU1X5lJuU+EBZmi3AmcqE9U1LxEOGwopaGyGOh3VyhKgi6JaVtQZt87PiJINKW8XQj3B9Qqe3VgYx WCQNa335YCnDsenASeBxuMIEaBCYd1l1fVEodJZOGwXwfAde0MhcbD0kj7LVlzcsTyk2TZYTccnZ75TuTj j13qvYs= 5nEtOrkCa84/mEwl3D9tSvVLyliwQ+u/Hd+C8u5SNk7hsAUZaA9TqH8Iw0J/IqSrsf6scsmundY5j7Z5mK 5J9uBxSB8t8vamFGD0pF4zhNtbrVvIXKI9kmp4vph1C5jzO9gDfs3MF+zjyYrVUFdM+pXtDBxmM+fGUfrp AuXb7/k= • user-name is the name of the subscriber, such as “Joe Smith”.
Chapter 3 Configuring Linksys ATAs Configuring a Streaming Audio Server Using a Streaming Audio Server The SAS feature lets you use attach an audio source to one of the Linksys ATA FXS ports (Phone 1 or Phone 2 on the PAP2T) and use it as a streaming audio source device. If the Linksys ATA has multiple FXS ports, either or both of the associated lines (Line 1 and Line 2 on the PAP2T) can be configured as an SAS server.
Chapter 3 Configuring Linksys ATAs Configuring a Streaming Audio Server If the Linksys ATA boots and finds that the SAS line is on-hook, it will not remove battery from the line so that IVR may be used. But if the Linksys ATA boots up and finds that the SAS line is off-hook, it will remove battery from the line because no audio session is in progress. Example SAS with MOH Figure 1-15 MOH Application with a Linksys ATA Line Configured as an SAS SPA2: IP=192.168.2.
Chapter 3 Configuring Linksys ATAs Using a FAX Machine with the SPA2102 or SPA8000 SAS Enable[1] = no MOH Server [1] = 1002@192.168.2.100:5061 or 1002@127.0.0.1:5061 SAS Enable[2] = yes On Linksys ATA 2: SAS MOH SAS MOH Enable[1] = no Server [1] = 1002@192.168.2.100:5061 Enable[2] = no Server [2] = 1002@192.168.2.100:5061 Configuring the Streaming Audio Server The following provides step-by-step procedures for implementing an SAS with an external music source.
Chapter 3 Configuring Linksys ATAs Using a FAX Machine with the SPA2102 or SPA8000 Step 1 Upgrade the ATA firmware to the latest version Step 2 Ensure that you have enough bandwidth for uplink and downlink. Step 3 Step 4 • For G.711 fallback, it is recommend to have ~100Kbps. • For T.38, allocate at least 50 kbps. To optimize G.
Chapter 3 Configuring Linksys ATAs Managing Caller ID Service • Loss • Delay Step 4 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. Step 5 Enable and capture the debug log. For instructions, refer to Item 4. in “Troubleshooting and Configuration FAQ” at the end of this chapter. Step 6 Identify the type of Fax machine connected to the ATA. Step 7 Contact technical support.
Chapter 3 Configuring Linksys ATAs Troubleshooting and Configuration FAQ Figure 1-17 Linksys ATA Caller ID Delivery Architecture a) Bellcore/ETSI Onhook Post-Ring FSK First Ring FSK b) ETSI Onhook Post-Ring DTMF First Ring DTMF c) ETSI Onhook Pre-Ring FSK/DTMF Polarity Reversal CAS (DTAS) DTMF/ FSK First Ring d) Bellcore Onhook FSK w/o Ring OSI FSK e) ETSI Onhook FSK w/o Ring Polarity Reversal CAS (DTAS) FSK f) Bellcore/ETSI Offhook FSK CAS (DTAS) Wait For ACK FSK Troubleshooting and Conf
Chapter 3 Configuring Linksys ATAs Troubleshooting and Configuration FAQ B. Press CTRL + F5. This is a hard refresh, which forces Windows Explorer to load new webpages, not cached ones. C. Click Tools. Click Internet Options. Click the Security tab. Click the Default level button. Make sure the security level is Medium or lower. Then click the OK button. 3. How do I save my current SPA configuration? Currently, the only way is to do HTTPGET from an HTTP client, from which you get the entire HTML page.
Chapter 3 Configuring Linksys ATAs Troubleshooting and Configuration FAQ C. Make sure you are not blocking the UDP PORT 5060,5061 and port for UDP packets in the range of 16384-16482. Also, disable “SPI” if this feature is provided by your firewall. Identify the SIP server to which the Linksys ATA is registering, if it supports NAT, using the parameter. D. Add a STUN server to allow traversal of UDP packets through the NAT device.
C H A P T E R 4 Configuring the PSTN Gateway (FXO) This chapter describes how to configure the PSTN gateway provided by Analog Telephone Adapters (ATAs) with one or more FXO ports, which includes the AG310 and SPA3102.
Chapter 4 Configuring the PSTN Gateway (FXO) How VoIP-To-PSTN Calls Work Line 1 can be configured with a regular VoIP account and can be used in the same way as the Line 1 of any Linksys ATA. With the SPA3102 and AG310, a second VoIP account can be configured to support PSTN gateway calls exclusively. A different SIP port should be assigned to Line 1 and the PSTN Line. The same VoIP account may be used for both Line 1 and the PSTN Line if a different SIP port is assigned to each.
Chapter 4 Configuring the PSTN Gateway (FXO) How VoIP-To-PSTN Calls Work Table 1-27 VoIP User Account Information Note Parameter Group Description Values User ID 1/2/3/4/5/6/7/ 8 PSTN Line The username value. 31-character string Password 1/2/3/4/5/6/7/ 8 PSTN Line The password value. 31-character string User 1/2/3/4/5/6/7/ 8 DP PSTN Line Specifies the dial plan to be used for this VoIP Choice of 0-8 user.
Chapter 4 Configuring the PSTN Gateway (FXO) How PSTN-To-VoIP Calls Work Table 1-28 Two-Stage Dialing Parameter Group Description Values VoIP Caller 1/2/3/4/5/6/7/8 PIN PSTN Line The PIN for VoIP Caller 1, 2, 3, 4, 5, 6, 7, or 8. 31-character string VoIP Caller 1/2/3/4/5/6/7/8 DP PSTN Line Specifies which dial plan to be used for this VoIP caller. If 0, dial plan processing is disabled; the given target number is dialed to the PSTN as is.
Chapter 4 Configuring the PSTN Gateway (FXO) How PSTN-To-VoIP Calls Work Table 1-29 Terminating Gateway Call Parameters Parameter Group Description Values Detect CPC: PSTN Line If yes, SPA3102 detects CPC as a disconnect signal. Default = Yes Yes or No Detect Long Silence: PSTN Line If yes, SPA3102 detects prolonged silence period as a disconnect signal.
Chapter 4 Configuring the PSTN Gateway (FXO) How PSTN-To-VoIP Calls Work Table 1-30 VoIP Outbound Call Routing Parameters Parameters Group Description Values Gateway 1 Line 1 Fully qualified domain name (or IP address) of a gateway. If the port number is not specified, 5060 is assumed. Default value is [blank] Domain name or IP address GW1 Nat Line 1 Whether to enable NAT mapping when using Gateway 1. Default is “no”.
Chapter 4 Configuring the PSTN Gateway (FXO) Configuring VoIP Failover to PSTN Configuring VoIP Failover to PSTN When power is disconnected from the SPA3102, 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 SPA3102, the FXS port is disconnected from the FXO port.
Chapter 4 Configuring the PSTN Gateway (FXO) Call Scenarios • Symmetric RTP, page 4-8 • Call Progress Tones, page 4-8 PSTN Call to Ring Line 1 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 Symmetric RTP is used to send audio RTP to the source IP and port of the inbound RTP packets. This facilitates NAT traversal.
Chapter 4 Configuring the PSTN Gateway (FXO) Call Scenarios • PSTN to VoIP Call with and Without Ring-Thru, page 4-9 • VoIP to PSTN Call with and Without Authentication, page 4-9 • Call Forwarding to PSTN Gateway, page 4-10 • User Dialing 9 to Access PSTN-Gateway for Local Calls, page 4-11 • Using the PSTN-Gateway for 311 and 911 Calls, page 4-12 • Auto-Fallback to the PSTN-Gateway, page 4-12 PSTN to VoIP Call with and Without Ring-Thru The PSTN caller calls the PSTN line connected to the FXO
Chapter 4 Configuring the PSTN Gateway (FXO) Call Scenarios If the PSTN Line is busy (off-hook, ringing, or PSTN line not connected) when the VoIP caller calls, the SPA3102 replies with 503. If the PIN number is invalid or entered after the VoIP call leg is connected, the SPA3102 plays the reorder tone to the VoIP caller and eventually ends the call when the reorder tone times out.
Chapter 4 Configuring the PSTN Gateway (FXO) Call Scenarios 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 set to six seconds. 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.
Chapter 4 Configuring the PSTN Gateway (FXO) Call Scenarios Using the PSTN-Gateway for 311 and 911 Calls To implement this scenario, add the rule “[39]11<:@gw0>” to Line 1. When the user dials 311 or 911, the call is routed to the PSTN gateway. Note If the PSTN Line is busy after the user dials 311 or 911, the call still fails. For true life-line supports, therefore, the PSTN Line cannot be shared. Auto-Fallback to the PSTN-Gateway To implement this scenario, enable .
C H A P T E R 5 Linksys ATA Field Reference This chapter describes the fields within each section of the following administration web server pages: Note • Info Tab, page 5-2 • System Tab, page 5-8 • SIP Tab, page 5-11 • Regional Tab, page 5-19 • Line Tab, page 5-33 • PSTN Line Tab, page 5-49 • User 1/2 Tab, page 5-65 • PSTN User Tab (SPA3102/AG310), page 5-70 Througout this chapter, references to the SPA3102 also apply to the AG310.
Chapter 5 Linksys ATA Field Reference Info Tab Info Tab This section describes the fields for the following headings on the Info tab: Note • System Information (PAP2T), page 5-2 • System Status (VoIP), page 5-2 • Line 1/2 Status, page 5-3 • PSTN Line Status, page 5-5 The fields on this tab are read-only and cannot be edited. System Information (PAP2T) Field Description DHCP Indicates if DHCP is enabled. Current IP Displays the current IP address assigned to the Linksys IP phone.
Chapter 5 Linksys ATA Field Reference Info Tab Field Description Current Time Current date and time of the system; for example, 10/3/2003 16:43:00. Broadcast Pkts Dropped Total number of broadcast packets received but not processed. Broadcast Bytes Dropped Total number of broadcast bytes received but not processed. RTP Packets Sent Total number of RTP packets sent (including redundant packets). RTP Bytes Sent Total number of RTP packets received (including redundant packets).
Chapter 5 Linksys ATA Field Reference Info Tab Message Waiting Indicates whether you have new voicemail waiting: Yes or No. This is updated when voicemail notification is received. You can also manually modify it to clear or set the flag. Setting this value to Yes can activate stutter tone and VMWI signal. This parameter is stored in long term memory and survives after reboot or power cycle. Call Back Active Indicates whether a call back request is in progress: Yes or No.
Chapter 5 Linksys ATA Field Reference Info Tab PSTN Line Status Note References to the SPA3102 also apply to the AG310. Field Description (PSTN) Hook State Hook state of the FXO port. Either On or Off. (PSTN) Line Voltage The voltage existing on the PSTN line. (PSTN) Loop Current The current (milliamperes) existing on the local loop. Registration State Indicates if the line has registered with the SIP proxy. Last Registration At Last date and time the line was registered.
Chapter 5 Linksys ATA Field Reference Info Tab Call Type May take one of the following values: • PSTN Gateway Call = VoIP-To-PSTN Call • VoIP Gateway Call = PSTN-To-VoIP Call • PSTN To Line 1 = PSTN call ring through and answered by Line 1 • Line 1 Forward to PSTN Gateway = VoIP calls Line 1 then forwarded to PSTN GW • Line 1 Forward to PSTN Number =VoIP calls Line 1 then forwarded to PSTN number • Line 1 To PSTN Gateway • Line 1 Fallback To PSTN Gateway VoIP State Same as Line 1 Call 1.
Chapter 5 Linksys ATA Field Reference Info Tab VoIP Call Packets Lost Same as Line 1 Call 1. VoIP Call Packet Error Same as Line 1 Call 1. VoIP Call Mapped RTP Port Same as Line 1 Call 1. Linksys IP Phone Administrator Guide Document Version 3.
Chapter 5 Linksys ATA Field Reference System Tab System Tab This section describes the fields for the following headings on the System tab: • System Configuration, page 5-8 • Internet Connection Type (PAP2T), page 5-8 • Optional Network Configuration (PAP2T), page 5-9 • Miscellaneous Settings (Not in PAP2T), page 5-10 System Configuration Field Description Restricted Access Domains This feature is used when implementing software customization.
Chapter 5 Linksys ATA Field Reference System Tab Field Description Gateway The default gateway used by Linksys IP phone when DHCP is disabled. The default is 0.0.0.0. Optional Network Configuration (PAP2T) Field Description Host Name The host name of the Linksys IP phone. Domain The network domain of the Linksys IP phone. Primary DNS DNS server used by Linksys IP phone in addition to DHCP supplied DNS servers if DHCP is enabled; when DHCP is disabled, this is the primary DNS server.
Chapter 5 Linksys ATA Field Reference System Tab Miscellaneous Settings (Not in PAP2T) Field Description Syslog Server Specifies the IP address of the syslog server. Debug Server Specifies the IP address of the debug server, which logs debug information. The level of detailed output depends on the debug level parameter setting. Debug Level Determines the level of debug information that is generated. Select 0, 1, 2, or 3 from the drop-down menu.
Chapter 5 Linksys ATA Field Reference SIP Tab SIP Tab This section describes the fields for the following headings on the SIP tab: • SIP Parameters, page 5-11 • SIP Timer Values (sec), page 5-12 • Response Status Code Handling, page 5-14 • RTP Parameters, page 5-14 • SDP Payload Types, page 5-15 • NAT Support Parameters, page 5-16 SIP Parameters Field Description Max Forward SIP Max Forward value, which can range from 1 to 255. The default is 70.
Chapter 5 Linksys ATA Field Reference SIP Tab Use Compact Header Lets you use compact SIP headers in outbound SIP messages. Select yes or no from the drop-down menu. If set to yes, the Linksys IP phone uses compact SIP headers in outbound SIP messages. If set to no, the Linksys IP phone uses normal SIP headers. If inbound SIP requests contain compact headers, Linksys IP phone reuses the same compact headers when generating the response regardless the settings of the
Chapter 5 Linksys ATA Field Reference SIP Tab SIP Timer H INVITE final response, time-out value, which can range from 0 to 64 seconds. The default is 32. SIP Timer D ACK hang-around time, which can range from 0 to 64 seconds. The default is 32. SIP Timer J Non-INVITE response hang-around time, which can range from 0 to 64 seconds. The default is 32. INVITE Expires INVITE request Expires header value. If you enter 0, the Expires header is not included in the request. The default is 240.
Chapter 5 Linksys ATA Field Reference SIP Tab Response Status Code Handling Field Description SIT1 RSC SIP response status code for the appropriate Special Information Tone (SIT). For example, if you set the SIT1 RSC to 404, when the user makes a call and a failure code of 404 is returned, the SIT1 tone is played. Reorder or Busy Tone is played by default for all unsuccessful response status code for SIT 1 RSC through SIT 4 RSC.
Chapter 5 Linksys ATA Field Reference SIP Tab RTCP Tx Interval Interval for sending out RTCP sender reports on an active connection. It can range from 0 to 255 seconds. During an active connection, the Linksys IP phone can be programmed to send out compound RTCP packet on the connection. Each compound RTP packet except the last one contains a SR (Sender Report) and a SDES.(Source Description). The last RTCP packet contains an additional BYE packet.
Chapter 5 Linksys ATA Field Reference SIP Tab G726r40 Dynamic Payload G.726-40 dynamic payload type. The valid range is 96-127. The default is 96. G729b Dynamic Payload G.729b dynamic payload type. The valid range is 96-127. The default is 99. NSE Codec Name NSE codec name used in SDP. The default is NSE. AVT Codec Name AVT codec name used in SDP. The default is telephone-event. G711u Codec Name G.711u codec name used in SDP. The default is PCMU. G711a Codec Name G.711a codec name used in SDP.
Chapter 5 Linksys ATA Field Reference SIP Tab Handle VIA received If you select yes, the Linksys IP phone processes the received parameter in the VIA header (this is inserted by the server in a response to anyone of its requests). If you select no, the parameter is ignored. Select yes or no from the drop-down menu. The default is no.
Chapter 5 Linksys ATA Field Reference SIP Tab EXT RTP Port Min External port mapping number of the RTP Port Min. number. If this value is not zero, the RTP port number in all outgoing SIP messages is substituted for the corresponding port value in the external RTP port range. The default is 0. NAT Keep Alive Intvl Interval between NAT-mapping keep alive messages. The default is 15. Linksys IP Phone Administrator Guide 5-18 Document Version 3.
Chapter 5 Linksys ATA Field Reference Regional Tab Regional Tab This section describes the fields for the following headings on the Regional tab: • Call Progress Tones, page 5-19 • Distinctive Ring Patterns, page 5-21 • Distinctive Call Waiting Tone Patterns, page 5-21 • Distinctive Ring/CWT Pattern Names, page 5-22 • Ring and Call Waiting Tone Spec, page 5-23 • Control Timer Values (sec), page 5-23 • Vertical Service Activation Codes, page 5-24 • Vertical Service Announcement Codes, page 5
Chapter 5 Linksys ATA Field Reference Regional Tab Ring Back Tone Played during an outbound call when the far end is ringing. The default is 440@-19,480@-19;*(2/4/1+2). Confirm Tone Brief tone to notify the user that the last input value has been accepted. The default is 600@-16; 1(.25/.25/1). SIT1 Tone Alternative to the Reorder Tone played when an error occurs as a caller makes an outbound call. The RSC to trigger this tone is configurable on the SIP screen.
Chapter 5 Linksys ATA Field Reference Regional Tab Distinctive Ring Patterns Field Description Ring1 Cadence Cadence script for distinctive ring 1. The default is 60(2/4). Ring2 Cadence Cadence script for distinctive ring 2. The default is 60(.3/.2, 1/.2,.3/4. Ring3 Cadence Cadence script for distinctive ring 3. The default is 60(.8/.4,.8/4). Ring4 Cadence Cadence script for distinctive ring 4. The default is 60(.4/.2,.3/.2,.8/4). Ring5 Cadence Cadence script for distinctive ring 5.
Chapter 5 Linksys ATA Field Reference Regional Tab CWT6 Cadence Cadence script for distinctive CWT 6. The default is 30(.1/.1, .3/.1, .1/9.3). CWT7 Cadence Cadence script for distinctive CWT 7. The default is 30(.1/.1, .3/.1, .1/9.3). CWT8 Cadence Cadence script for distinctive CWT 8. The default is 2.3(..3/2). Distinctive Ring/CWT Pattern Names Field Description Ring1 Name Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 1 for the inbound call. The default is Bellcore-r1.
Chapter 5 Linksys ATA Field Reference Regional Tab Ring and Call Waiting Tone Spec Field Description Ring Waveform Waveform for the ringing signal. The default is Sinusoid. Ring Frequency Frequency of the ringing signal. Valid values are 10–100 (Hz). The default is 25. Ring Voltage Ringing voltage. 60–90 (V). The default is 70. CWT Frequency Frequency script of the call waiting tone. All distinctive CWTs are based on this tone. The default is 440@-10.
Chapter 5 Linksys ATA Field Reference Regional Tab VMWI Refresh Intvl Interval between VMWI refresh to the CPE. The default is 0.5. Interdigit Long Timer Long timeout between entering digits when dialing. The interdigit timer values are used as defaults when dialing. The Interdigit_Long_Timer is used after any one digit, if all valid matching sequences in the dial plan are incomplete as dialed. Range: 0–64 seconds. The default is 10.
Chapter 5 Linksys ATA Field Reference Regional Tab Call Back Act Code Starts a callback when the last outbound call is not busy. The default is *66. Call Back Deact Code Cancels a callback. The default is *86. Call Back Busy Act Code Starts a callback when the last outbound call is busy. (Not in PAP2T) The default is *05 Cfwd All Act Code Forwards all calls to the extension specified after the activation code. The default is *72. Cfwd All Deact Code Cancels call forwarding of all calls.
Chapter 5 Linksys ATA Field Reference Regional Tab CW Deact Code Disables call waiting on all calls. The default is *57. CW Per Call Act Code Enables call waiting for the next call. The default is *71. CW Per Call Deact Code Disables call waiting for the next call. The default is *70. Block CID Act Code Blocks caller ID on all outbound calls. The default is *67. Block CID Deact Code Removes caller ID blocking on all outbound calls. The default is *68.
Chapter 5 Linksys ATA Field Reference Regional Tab Speed Dial Act Code Assigns a speed dial number. The default is *74. Secure All Call Act Code Makes all outbound calls secure. The default is *16. Secure No Call Act Code Makes all outbound calls not secure. The default is *17. Secure One Call Act Code Makes the next outbound call secure. (It is redundant if all outbound calls are secure by default.) The default is *18. Secure One Call Deact Code Makes the next outbound call not secure.
Chapter 5 Linksys ATA Field Reference Regional Tab Feature Dial Services Codes These codes tell the Linksys IP phone what to do when the user is listening to the first or second dial tone. One or more *code can be configured into this parameter, such as *72, or *72|*74|*67|*82, etc. Max total length is 79 chars. This parameter applies when the user has a dial tone (first or second dial tone).
Chapter 5 Linksys ATA Field Reference Regional Tab Field Description Prefer G711u Code Makes this codec the preferred codec for the associated call. The default is *017110. Force G711u Code Makes this codec the only codec that can be used for the associated call. The default is *027110. Prefer G711a Code Makes this codec the preferred codec for the associated call. The default is *017111 Force G711a Code Makes this codec the only codec that can be used for the associated call.
Chapter 5 Linksys ATA Field Reference Regional Tab Miscellaneous Field Description Set Local Date (mm/dd) Sets the local date (mm stands for months and dd stands for days). The year is optional and uses two or four digits. Set Local Time (HH/mm) Sets the local time (hh stands for hours and mm stands for minutes). Seconds are optional. Time Zone Selects the number of hours to add to GMT to generate the local time for caller ID generation.
Chapter 5 Linksys ATA Field Reference Regional Tab FXS Port Input Gain Input gain in dB, up to three decimal places. The range is 6.000 to -12.000. The default is -3. FXS Port Output Gain Output gain in dB, up to three decimal places. The range is 6.000 to -12.000. The Call Progress Tones and DTMF playback level are not affected by the . The default is -3. DTMF Playback Level Local DTMF playback level in dBm, up to one decimal place. The default is -16.0.
Chapter 5 Linksys ATA Field Reference Regional Tab Feature Invocation Method Select the method you want to use, Default or Sweden default. (Not in PAP2T) The default is Default. More Echo Suppression Enable or disable more echo suppresion. The default is no. GR909 Test To use this test, select yes. Otherwise, keep the default, no. Linksys IP Phone Administrator Guide 5-32 Document Version 3.
Chapter 5 Linksys ATA Field Reference Line Tab Line Tab This section describes the fields for the following headings on the Line tabs: • Line Enable, page 5-33 • Streaming Audio Server (SAS), page 5-33 • NAT Settings, page 5-34 • Network Settings, page 5-35 • SIP Settings, page 5-35 • Call Feature Settings, page 5-38 • Proxy and Registration, page 5-38 • Subscriber Information, page 5-40 • Supplementary Service Subscription, page 5-40 • Audio Configuration, page 5-42 • VoIP Fallback t
Chapter 5 Linksys ATA Field Reference Line Tab SAS Enable To enable the use of the line as a streaming audio source, select yes. Otherwise, select no. If enabled, the line cannot be used for outgoing calls. Instead, it auto-answers incoming calls and streams audio RTP packets to the caller. The default is no.
Chapter 5 Linksys ATA Field Reference Line Tab NAT Keep Alive Dest Destination that should receive NAT keep alive messages. If the value is $PROXY, the messages are sent to the current or outbound proxy. The default is $PROXY. Network Settings Field Description SIP ToS/DiffServ Value TOS/DiffServ field value in UDP IP packets carrying a SIP message. The default is 0x68. SIP CoS Value [0-7] CoS value for SIP messages. (Not in PAP2T) The default is 3.
Chapter 5 Linksys ATA Field Reference Line Tab SIP 100REL Enable To enable the support of 100REL SIP extension for reliable transmission of provisional responses (18x) and use of PRACK requests, select yes. Otherwise, select no. The default is no. EXT SIP Port The external SIP port number. Auth Resync-Reboot If this feature is enabled, the Linksys IP phone authenticates the sender when it receives the NOTIFY resync reboot (RFC 2617) message. To use this feature, select yes. Otherwise, select no.
Chapter 5 Linksys ATA Field Reference Line Tab SIP Debug Option SIP messages are received at or sent from the proxy listen port. This feature controls which SIP messages to log. Choices are as follows: • none—No logging. • 1-line—Logs the start-line only for all messages. • 1-line excl. OPT—Logs the start-line only for all messages except OPTIONS requests/responses. • 1-line excl. NTFY—Logs the start-line only for all messages except NOTIFY requests/responses. • 1-line excl.
Chapter 5 Linksys ATA Field Reference Line Tab Refer-To Target Contact To contact the refer-to target, select yes. Otherwise, select no. The default is no. Sticky 183 If this feature is enabled, the IP telephony ignores further 180 SIP responses after receiving the first 183 SIP response for an outbound INVITE. To enable this feature, select yes. Otherwise, select no. The default is no. Auth INVITE When enabled, authorization is required for initial incoming INVITE requests from the SIP proxy.
Chapter 5 Linksys ATA Field Reference Line Tab Field Description Use Outbound Proxy Enable the use of . If set to no, and
Chapter 5 Linksys ATA Field Reference Line Tab Field Description Voice Mail Server Enter the URL or IP address of the server. Mailbox Subscribe Expires <> Subscriber Information Field Description Display Name Display name for caller ID. User ID Extension number for this line. Password Password for this line. Use Auth ID To use the authentication ID and password for SIP authentication, select yes. Otherwise, select no to use the user ID and password. The default is no.
Chapter 5 Linksys ATA Field Reference Line Tab Field Description Call Waiting Serv Enable Call Waiting Service. The default is yes. Block CID Serv Enable Block Caller ID Service. The default is yes. Block ANC Serv Enable Block Anonymous Calls Service The default is yes. Dist Ring Serv Enable Distinctive Ringing Service The default is yes. Cfwd All Serv Enable Call Forward All Service The default is yes. Cfwd Busy Serv Enable Call Forward Busy Service The default is yes.
Chapter 5 Linksys ATA Field Reference Line Tab Field Description Call Back Serv Enable Call Back Service Three Way Call Serv Enable Three Way Calling Service. Three Way Calling is required for Three Way Conference and Attended Transfer. The default is yes. Three Way Conf Serv Enable Three Way Conference Service. Three Way Conference is required for Attended Transfer. The default is yes. Attn Transfer Serv Enable Attended Call Transfer Service.
Chapter 5 Linksys ATA Field Reference Line Tab Therefore it is important to disable the use of G.729a in order to guarantee the support of two simultaneous G.723/G.726 codec. Field Description Preferred Codec Preferred codec for all calls. (The actual codec used in a call still depends on the outcome of the codec negotiation protocol.) Select one of the following: G711u, G711a, G726-16, G726-24, G726-32, G726-40, G729a, or G723. The default is G711u.
Chapter 5 Linksys ATA Field Reference Line Tab G726-40 Enable To enable the use of the G.726 codec at 40 kbps, select yes. Otherwise, select no. The default is yes. FAX Passthru Codec Select the codec for fax passthrough, G711u or G711a. The default is G711u. DTMF Process INFO (Not in PAP2T) To use the DTMF process info feature, select yes. Otherwise, select no. The default is yes. FAX Codec Symmetric To force the Linksys IP phone to use a symmetric codec during fax passthrough, select yes.
Chapter 5 Linksys ATA Field Reference Line Tab FAX T38 Redundancy Select the appropriate number. The default is 1. Fax Tone Detect Mode If you want the Gateway to detect the fax tone whether the Gateway is a caller or callee, then select caller or callee. If you want the Gateway to detect the fax tone only if the Gateway is the caller, then select caller only. If you want the Gateway to detect the fax tone only if the Gateway is the callee, then select callee only.
Chapter 5 Linksys ATA Field Reference Line Tab VoIP Fallback to PSTN (SPA3102/AG310) Field Description Auto PSTN Fallback If enabled, the SPA will automatically route all calls to the PSTN gateway when the Line 1 proxy is down (registration failure or network link down). The default is yes. Dial Plan The default dial plan script for each line is as follows: (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxx|xxxxxxxxxxxx.).
Chapter 5 Linksys ATA Field Reference Line Tab Dial Plan Dial plan script for this line. The default(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.) The dial plan syntax is expanded in the SPA3102 to allow the designation of three parameters to be used with a specific gateway: • uid – the authentication user-id • pwd – the authentication password • nat – if this parameter is present, use NAT mapping Each parameter is separated by a semi-colon (;).
Chapter 5 Linksys ATA Field Reference Line Tab Linksys IP Phone Administrator Guide 5-48 Document Version 3.
Chapter 5 Linksys ATA Field Reference PSTN Line Tab PSTN Line Tab This section describes the fields for the following headings on the PSTN Line tab on the SPA3102 and AG310: • Line Enable, page 5-33 • NAT Settings, page 5-49 • Network Settings, page 5-50 • SIP Settings, page 5-50 • Proxy and Registration (SPA3102/AG310), page 5-53 • Subscriber Information (SPA3102/AG310), page 5-54 • Audio Configuration (SPA3102/AG310), page 5-55 • Dial Plans, page 5-57 • VoIP-To-PSTN Gateway Setup, page
Chapter 5 Linksys ATA Field Reference PSTN Line Tab NAT Keep Alive Msg Enter the keep alive message that should be sent periodically to maintain the current NAT mapping. If the value is $NOTIFY, a NOTIFY message is sent. If the value is $REGISTER, a REGISTER message without contact is sent. The default is $NOTIFY. NAT Keep Alive Dest Destination that should receive NAT keep alive messages. If the value is $PROXY, the messages are sent to the current or outbound proxy. The default is $PROXY.
Chapter 5 Linksys ATA Field Reference PSTN Line Tab SIP 100REL Enable To enable the support of 100REL SIP extension for reliable transmission of provisional responses (18x) and use of PRACK requests, select yes. Otherwise, select no. The default is no. EXT SIP Port The external SIP port number. Auth Resync-Reboot If this feature is enabled, the Linksys IP phone authenticates the sender when it receives the NOTIFY resync reboot (RFC 2617) message. To use this feature, select yes. Otherwise, select no.
Chapter 5 Linksys ATA Field Reference PSTN Line Tab SIP Debug Option SIP messages are received at or sent from the proxy listen port. This feature controls which SIP messages to log. Choices are as follows: • none—No logging. • 1-line—Logs the start-line only for all messages. • 1-line excl. OPT—Logs the start-line only for all messages except OPTIONS requests/responses. • 1-line excl. NTFY—Logs the start-line only for all messages except NOTIFY requests/responses. • 1-line excl.
Chapter 5 Linksys ATA Field Reference PSTN Line Tab Refer-To Target Contact To contact the refer-to target, select yes. Otherwise, select no. The default is no. Sticky 183 If this feature is enabled, the IP telephony ignores further 180 SIP responses after receiving the first 183 SIP response for an outbound INVITE. To enable this feature, select yes. Otherwise, select no. The default is no. Proxy and Registration (SPA3102/AG310) Field Description Proxy SIP proxy server for all outbound requests.
Chapter 5 Linksys ATA Field Reference PSTN Line Tab Field Description Proxy Fallback Intvl This parameter sets the delay (sec) after which the PAP2T will retry from the highest priority proxy (or outbound proxy) servers after it has failed over to a lower priority server. This parameter is useful only if the primary and backup proxy server list is provided to the PAP2T via DNS SRV record lookup on the server name.
Chapter 5 Linksys ATA Field Reference PSTN Line Tab Audio Configuration (SPA3102/AG310) A codec resource is considered as allocated if it has been included in the SDP codec list of an active call, even though it eventually may not be the one chosen for the connection. So, if the G.729a codec is enabled and included in the codec list, that resource is tied up until the end of the call whether or not the call actually uses G.729a. If the G729a resource is already allocated and since only one G.
Chapter 5 Linksys ATA Field Reference PSTN Line Tab FAX CED Detect Enable To enable detection of the fax Caller-Entered Digits (CED) tone, select yes. Otherwise, select no. The default is yes. G726-32 Enable To enable the use of the G726 codec at 32 kbps, select yes. Otherwise, select no. The default is yes. FAX CNG Detect Enable To enable detection of the fax Calling Tone (CNG), select yes. Otherwise, select no. The default is yes.
Chapter 5 Linksys ATA Field Reference PSTN Line Tab Release Unused Codec This feature allows the release of codecs not used after codec negotiation on the first call, so that other codecs can be used for the second line. To use this feature, select yes. Otherwise, select no. The default is yes. FAX Enable T38 To enable the use of the ITU-T T.38 standard for faxing, select yes. Otherwise, select no. The default is yes.
Chapter 5 Linksys ATA Field Reference PSTN Line Tab Field Description VoIP PIN Max Retry Number of trials to allow VoIP caller to enter a PIN number (used only if authentication method is set to PIN). The default is 3. One Stage Dialing Enable one-stage dialing (applicable if authentication method is none, or HTTP Digest, or caller is in the Access List). The default is yes.
Chapter 5 Linksys ATA Field Reference PSTN Line Tab Field Description Line 1 Fallback DP Index of the dial plan in the dial plan pool to be used when the VoIP Caller is calling from Line 1 of the same SPA3102 unit due to fallback to PSTN service when Line 1 VoIP service is down. Choose from {none, 1, 2, 3, 4, 5, 6, 7, 8}. The default is 1.
Chapter 5 Linksys ATA Field Reference PSTN Line Tab Field Description VoIP User 1/2/3/4/5/6/7/8 Password The password to be used with VoIP User 1. The user assumes the identity of VoIP User 1 must therefore compute the credentials using this password, or the INVITE will be challenged with a 401 response The default is blank.
Chapter 5 Linksys ATA Field Reference PSTN Line Tab Field Description PSTN Answer Delay Delay in seconds before auto-answering inbound PSTN calls after the PSTN starts ringing. The range is 0-255. The default is 16. VoIP PIN Digit Timeout Timeout to wait for the 1st or subsequent PIN digits from a VoIP caller. The range is 0-255. The default is 10. PSTN PIN Digit Timeout Timeout to wait for the 1st or subsequent PIN digits from a PSTN caller. The range is 0-255. The default is 10.
Chapter 5 Linksys ATA Field Reference PSTN Line Tab Field Description PSTN Hook Flash Len The length of the hook flash in seconds. During a PSTN-to-VoIP gateway call, the Linksys ATA processes the out-of-band hook flash signal sent from the VoIP peer through a hook-flash (momentary on-hook signal) on the FXO port. This allows the VoIP peer to initiate a three-way conference call and subsequent call transfer. The duration of the on-hook signal can be configured using this parameter. The default is 0.
Chapter 5 Linksys ATA Field Reference PSTN Line Tab Field Description (PSTN) Long Silence Duration This is minimum length of PSTN silence (or inactivity) in seconds to trigger a gateway call disconnection if is yes. The default is 30. Silence Threshold This parameter adjusts the sensitivity of PSTN silence detection. Choose from {very low, low, medium, high, very high}. The higher the setting, the easier to detect silence and hence easier to trigger a disconnection.
Chapter 5 Linksys ATA Field Reference PSTN Line Tab Field Description Ring Validation Time Specify the minimum signal duration required by the Gateway for recognition as a ring signal. The default is 256 ms. Tip/Ring Voltage Adjust Choices are {3.1, 3.2, 3.35, 3.5}. The default is 3.5. Operational Loop Current Min Choices for mA are: {10, 12, 14, 16). The default is 10. On-Hook Speed Choose from {Less than 0.5ms, 3ms (ETSI), 26ms (Australia)}. The default is Less than 0.5ms.
Chapter 5 Linksys ATA Field Reference User 1/2 Tab User 1/2 Tab This section describes the fields for the following headings on the User 1 and User 2 tabs: • Call Forward Settings, page 5-65 • Selective Call Forward Settings, page 5-66 • Speed Dial Settings, page 5-67 • Supplementary Service Settings, page 5-67 • Distinctive Ring Settings, page 5-68 • Ring Settings, page 5-68 User 1/2 refers to the subscriber of Line 1 or Line 2.
Chapter 5 Linksys ATA Field Reference User 1/2 Tab Field Description Cfwd No Ans Dest Forward number for Call Forward No Answer Service. Same as Cfwd All Dest. In addition to normal call forward destination as used in the other ATAs, on the SPA3102, you can specify the following additional parameters: • gw0 – forward the caller to use the PSTN gateway • @gw0 – forward to caller to the PSTN number (dialed automatically by the SPA through the PSTN gateway) The default is blank.
Chapter 5 Linksys ATA Field Reference User 1/2 Tab Speed Dial Settings Field Description Speed Dial 2/3/4/5/6/7/8/9 Target phone number (or URL) assigned to speed dial 2, 3, 4, 5, 6, 7, 8, or 9. The default is blank. Supplementary Service Settings The Linksys IP phone provides native support of a large set of enhanced or supplementary services. All of these services are optional. The parameters listed in the following table are used to enable or disable a specific supplementary service.
Chapter 5 Linksys ATA Field Reference User 1/2 Tab Field Description Accept Media Loopback Request Determines how the media loopback request is enabled. Choose automatic, never, or manual. The default is automatic. Media Loopback Mode Determines the media loopback mode. Choose source or mirror. Media Loopback Type Determines the media loopback type. Choose media or packet. Distinctive Ring Settings Caller number patterns are matched from Ring 1 to Ring 8.
Chapter 5 Linksys ATA Field Reference User 1/2 Tab Field Description VMWI Ring Splash Len Duration of ring splash when new messages arrive before the VMWI signal is applied (0 – 10.0s). The default is .5. VMWI Ring Policy The parameter controls when a ring splash is played when a the VM server sends a SIP NOTIFY message to the Linksys IP phone indicating the status of the subscriber’s mail box.
Chapter 5 Linksys ATA Field Reference PSTN User Tab (SPA3102/AG310) PSTN User Tab (SPA3102/AG310) This section describes the fields for the following headings on the PSTN User tab on the SPA3102: Note • PSTN-To-VoIP Selective Call Forward Settings, page 5-70 • PSTN-To-VoIP Speed Dial Settings, page 5-70 • PSTN Ring Thru Line 1 Distinctive Ring Settings, page 5-70 • PSTN Ring Thru Line 1 Ring Settings, page 5-71 • PSTN/VoIP Caller Commands via DTMF, page 5-71 References in this section to the
Chapter 5 Linksys ATA Field Reference PSTN User Tab (SPA3102/AG310) PSTN Ring Thru Line 1 Ring Settings Field Description Default Ring The default ring to be used to ring through Line 1. Choose from {1,2,3,4,5,6,7,8,Follow Line 1}. If Follow Line 1 is selected, the ring to be used is determined by Line 1’s distinctive ring settings. The default is 1. PSTN/VoIP Caller Commands via DTMF Field Description **# Disconnect the PSTN line (SPA will take the FXO port on-hook).
Chapter 5 Linksys ATA Field Reference PSTN User Tab (SPA3102/AG310) Linksys IP Phone Administrator Guide 5-72 Document Version 3.
A P P E N D I X A Acronyms ACS Auto-Configuration Server A/D Analog To Digital Converter ANC Anonymous Call B2BUA Back to Back User Agent Bool Boolean Values.
Appendix A FXS Foreign eXchange Station GW Gateway ITU International Telecommunication Union HTML Hypertext Markup Language HTTP Hypertext Transfer Protocol HTTPS HTTP over SSL ICMP Internet Control Message Protocol IGMP Internet Group Management Protocol ILEC Incumbent Local Exchange Carrier IP Internet Protocol ISP Internet Service Provider ITSP IP Telephony Service Provider IVR Interactive Voice Response LAN Local Area Network LBR Low Bit Rate LBRC Low Bit Rate Codec MC
Appendix A Acronyms SDP Session Description Protocol SDRAM Synchronous DRAM sec seconds SIP Session Initiation Protocol SLA Shared line appearance SLIC Subscriber Line Interface Circuit SP Service Provider SPA Linksys Phone Adaptor SSL Secure Socket Layer TFTP Trivial File Transfer Protocol TCP Transmission Control Protocol UA User Agent uC Micro-controller UDP User Datagram Protocol URL Uniform Resource Locator VM Voicemail VMWI Visual Message Waiting Indication/Indicator
Appendix A Acronyms Linksys ATA Administrator Guide A-4 Document Version 3.
A P P E N D I X B Glossary ACD (Automatic Call Distribution)—A switching system designed to allocate incoming calls to certain positions or agents in the order received and to hold calls not ready to be handled (often with a recorded announcement). Area code—A 3-digit code used in North America to identify a specific geographic telephone location. The first digit can be any number between 2 and 9. The second and third digits can be any number.
Appendix B Glossary Dedicated access—Customers have direct access to the long-distance provider via a special circuit (T1 or private lines). The circuit is hardwired from the customer site to the POP and does not pass through the LEC switch. The dial tone is provided from the long-distance carrier. Dedicated Access Line (DAL)—Provided by the local exchange carrier. An access line from the customer’s telephone equipment directly to the long-distance company’s switch or POP.
A P P E N D I X C User Guidelines This appendix provides documentation for the use of the SPA products. It includes the following sections: • Basic Services, page C-1 • Enhanced Services, page C-2 The SPA can be configured to the custom requirements of the service provider, so that from the subscriber point of view, the service behaves exactly as the service provider wishes, with varying degrees of control left with the end user.
Appendix C User Guidelines Enhanced Services Receiving a Phone Call Service description The SPA can receive calls from the PSTN or other IP Telephony subscribers User action required to activate or use When the telephone rings, pick up the handset and begin talking. Expected call and network behavior Each subscriber is assigned an E.164 ID (phone number) so that they may be reached from wired or wireless callers on the PSTN or IP network.
Appendix C User Guidelines Enhanced Services Expected call and network behavior Caller ID is sent to the distant party for this call only. Users must repeat this process at the start of each call. User action required to deactivate or end No action required. This service is only in effect for the duration of the current call.
Appendix C User Guidelines Enhanced Services Disable or Cancel Call Waiting Service description The SPA supports disabling of call waiting permanently or on a per-call basis. User action required to activate or use To temporarily disable Call Waiting (for the length of one call) do the following before placing a call: 1. Lift Receiver 2. Press *70 3. Listen for dial tone, then dial the number you want to call. Call Waiting is now disabled for the duration of this call only.
Appendix C User Guidelines Enhanced Services Call-Waiting with Caller ID Service description When the user is on the phone and has Call Waiting active, the new caller Caller ID information is displayed on the user phone display screen at the same time the user is hearing the Call Waiting beeps/tones. User action required to activate or use The telephone equipment connected to the SPA must support Call-Waiting with Caller ID.
Appendix C User Guidelines Enhanced Services Attendant Call Transfer Service description Attendant Call Transfer lets a customer use their touchtone phone to send a call to any other phone, inside or outside their business, including a wireless phones. User action required to activate or use While in a call with the party to be transferred: 1. Press the switch hook or flash button on the phone to place the party on hold 2. Listen for three short tones followed by dial tone 3.
Appendix C User Guidelines Enhanced Services Expected call and network behavior When the user presses the switch hook or flash button, the transferee is placed on hold. When the user successfully dials the transfer number, the transferee automatically calls the dialed number. User action required to deactivate or end Not applicable. Call Hold Service description Call Hold lets you put a caller on hold for an unlimited period of time. It is especially useful on phones without the hold button.
Appendix C User Guidelines Enhanced Services Three-Way Ad-Hoc Conference Calling Service description This feature allows the user to conference up to two other numbers on the same line to create a three-way call. User action required to activate or use If you are already on a call and wish to add a third party: 1. Press the switch hook or flash button 2. Listen for dial tone 3. Dial the third party normally 4.
Appendix C User Guidelines Enhanced Services Automatic Call Back Service description User action required to activate or use Expected call and network behavior This feature allows the user to place a call to the last number they tried to reach whether the call was answered, unanswered, or busy, by dialing an activation code. 1. Pick up the receiver 2. Listen for dial tone 3. Press *07 If the number called is idle, the call rings through and completes normally.
Appendix C User Guidelines Enhanced Services Expected call and network behavior User action required to deactivate or end This feature allows a user the option to divert (forward) all calls to their telephone number to any number using the touchtone keypad of their telephone or web browser interface. This service is activated or deactivated from the phone being forwarded or the web browser interface. 1. Lift the receiver 2. Listen for dial tone 3.
Appendix C User Guidelines Enhanced Services Call FWD—No Answer Service description User action required to activate or use Calls are forwarded to the designated forwarding number after a configurable time period elapses while the SPA is ringing and does not answer. 1. Lift the receiver 2. Listen for dial tone 3. Press *92Listen for dial tone and enter the telephone number you are forwarding your call to. Activation is confirmed with three short bursts of tone and your forwarding is activated.
Appendix C User Guidelines Enhanced Services Distinctive/Priority Ringing and Call Waiting Tone Service description User action required to activate or use Expected call and network behavior The SPA supports a number of ringing and call waiting tone patterns to be played when incoming calls arrive. The choice of alerting pattern to use is carried in the incoming SIP INVITE message inserted by the SIP proxy server (or other intermediate application server in the service provider domain). 1.
Appendix C User Guidelines Enhanced Services Expected call and network behavior 1. Pick up the receiver 2. Listen for dial tone 3. Press single digit code assigned to the stored number (2-9) 4. Press # to signal dialing complete The number is automatically dialed normally. User action required to deactivate or end None Linksys ATA Administrator Guide Document Version 3.
Appendix C User Guidelines Enhanced Services Linksys ATA Administrator Guide C-14 Document Version 3.
INDEX Auth ID Symbols 5-40, 5-54 Auth Resync-Reboot 5-36, 5-51 **# command 5-71 AVT Codec Name **1 command 5-71 AVT Dynamic Payload Numerics 404 Forbidden 5-16 5-15 B 3-18 bandwidth budget binary format 2-14 3-4 Blind Attn-Xfer Enable A Blind Transfer Code Accept Last Act Code 5-25 Accept Last Deact Code accessing IVR 5-66, 5-70 5-25 Accept Last Serv parameter 5-41 Block CID Act Code 2-23 2-23 Block CID Deact Code 5-26 Block CID Setting parameter 2-20 Block Last Act Code 2-2
Index Call Return Code C 5-24 Call Return Serv parameter Call 1 Bytes Recv 5-4 Call 1 Bytes Sent Call Type parameter 5-4 Call 1 Callback 5-41 5-6 Call Waiting Serv parameter 5-4 Call 1 Decode Latency 5-4 candidate sequences 3-5 caring for hardware 2-2 5-41 Call 1 Decoder 5-4 Cblk Ring Splash Len parameter Call 1 Duration 5-4 Cfwd All Act Code Call 1 Encoder 5-4 Cfwd All Deact Code Call 1 FAX 5-4 Call 1 Jitter 5-4 5-25 5-25 Cfwd All Dest parameter 5-65 Cfwd All Serv param
Index multicast address network mask CWT Frequency 2-20 2-19 primary DNS server 2-20 static gateway IP address WAN IP address D 2-20 Daylight Saving Time Rule 2-19 CID_Serv parameter CID Act Code debugging 5-41 CID Setting parameter Debug Server 5-67 Conference Act Code Conference Tone 5-27 5-20 configuration 5-9 5-10 Debug Server parameter 5-9 Default CWT parameter 5-68 Default Ring parameter 5-68, 5-71 Default VoIP Caller DP parameter 3-19 configuration profile Confirm Tone
Index DND Setting parameter DNS Query Mode EXT SIP Port parameter 5-67 5-9 DNS server check set 5-36, 5-51 F 2-20 Factory 2-20 DNS Server Order parameter Domain parameter 5-39, 5-53 5-31 DTMF Playback Level DTMF Process AVT 5-31 2-21 FAX CED Detect Enable 5-43, 5-56 FAX CNG Detect Enable 5-43, 5-56 FAX Disable ECAN 5-44, 5-56 DTMF Relay MIME Type 2-21 FAX Codec Symmetric 5-44, 5-56 DTMF Process INFO resetting user 5-2, 5-9 DTMF Playback Length DTMF Tx Method factory defaults
Index G Hook Flash Timer Min 5-23 Hook Flash Tx Method 5-44, 5-56 G.
Index MOH Server L 5-38 multicast address LAN IP address check check 2-20 set Last Called Number 5-4 Last Called VoIP Number parameter 5-5 5-20 MWI Serv parameter 5-42 5-5 5-4 Last PSTN Caller parameter N 5-5 Last PSTN Disconnect Reason parameter Last Registration At 5-5 5-3, 5-5 NAT Keep Alive Dest 5-35, 5-50 NAT Keep Alive Enable Last VoIP Caller parameter 5-5 Line 1 Fallback DP parameter 5-59 Line 1 VoIP Caller DP parameter Line Enable 2-20 MWI Dial Tone Last Called PSTN
Index PSTN Hook Flash Len parameter P PSTN Peer Name parameter parameters Password 3-5 PSTN PIN Tone parameter PSTN Ring Timeout parameter pause, in dial plans 3-7 plain-text file profile 3-5 5-31 5-6 PSTN Tone parameter 5-6 5-29 Prefer G711u Code 5-29 Prefer G723 Code Prefer G726r16 Code 5-29 Prefer G726r24 Code 5-29 Prefer G726r32 Code 5-29 Prefer G726r40 Code 5-29 Preferred Codec quick-reference for IVR 5-29 Primary DNS parameter Real-Time Protocol reboot 5-2, 5-9 primary
Index Remove Last Reg Reorder Delay RTCP Tx Interval 5-11 RTP Bytes Recv parameter 5-23 Reorder Tone RTP CoS Value 3-7 resetting Restrict Source IP Resync URL 5-14 Ring1 Cadence 5-14 RTP Port Min 5-14 RTP ToS/DiffServ Value dial plans 5-35, 5-50 3-6 S 5-21 SAS DLG Refresh Intvl 5-22 SAS Enable 5-21 Second Dial Tone 5-22 Secure Call Serv parameter 5-21 Secure No Call Act Code 5-20 5-42 5-64 secure provisioning Ring Frequency Min parameter 5-64 Send Resp To Src Port 5-17
Index network mask 2-20 primary DNS server 2-20 static gateway IP address static IP addressing 2-20 2-19 Set Local Date (mm/dd) 5-14 SIT3 Tone 5-20 SIT4 RSC 5-14 SIT4 Tone 5-20 SPA To PSTN Gain parameter 5-30 Set Local Time (HH/mm) SIT3 RSC specifies 5-30 5-63 3-9 short interdigit timer 3-7 Speed Dial 2/3/4/5/6/7/8/9 parameter Silence Supp Enable 5-43, 5-55 Speed Dial Act Code Silence Threshold Silence Threshold parameter SIP 100REL Enable SIP Bytes Sent parameter SIP GUID
Index Tip/Ring Voltage Adjust parameter troubleshooting 5-64 5-6 VoIP Caller 1/2/3/4/5/6/7/8 DP parameter 3-18 Try Backup RSC VoIP Call Encoder parameter 5-59 VoIP Caller 1/2/3/4/5/6/7/8 PIN parameter 5-14 5-59 VoIP Caller Authentication Method parameter VoIP Caller ID Pattern parameter U Unattn Transfer Serv parameter 5-42 upgrade remote 3-4 3-3 URLs VoIP Call Jitter parameter 5-6 VoIP Call Packet Error parameter 5-7 VoIP Call Packets Lost parameter 5-7 VoIP Call Packets Recv parame
