Chapter 1. Get Started

- Your Analog Telephone Adapter

- Install Your Cisco ATA

- ATA Voice Quality

Your Analog Telephone Adapter

The ATA 191 and ATA 192 analog telephone adapters are telephony-device-to-Ethernet adapters that allow regular analog phones to operate on IP-based telephony networks. Both models support two voice ports, each with an independent phone number. Both have an RJ-45 10/100BASE-T data port while the ATA 192 has an additional Ethernet port.

The ATA connects to the Internet through a broadband (DSL or cable) modem or router. The ATA can be used with an on-site call-control system or an Internet-based call-control system.

The 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. An ATA maintains the state of each call it terminates and reacts appropriately to user input events (such as on/off hook or hook flash). The ATAs use the Session Initiation Protocol (SIP) open standard so there is on/off hook or hook flash. The ATAs use the Session Initiation Protocol (SIP) open standard so there is little or no involvement by a “middle-man” server or media gateway controller. SIP allows inter-operation with all ITSPs that support SIP.

Cisco Analog Telephone Adapter

- ATA 191 and ATA 192 Top Panel

- ATA 191 and ATA 192 Back Panel

ATA 191 and ATA 192 Top Panel

The following figure shows the different LEDs and buttons found on the top of your ATA.

ATA 191 and ATA 192 Top Panel

ATA 191 and ATA 192 Top Panel Items

Item

Description

Power LED

Steady green: System booted up successfully and is ready for use.

Slow flashing green: System is booting up.

Fast flashing green three times, then repeats: System failed to boot up.

Fast flashing green: The LED behaviour occurs in the following situations:

- System detects a factory reset.

To perform a factory reset, press and hold the RESET button for about 10 seconds.

- A factory reset is performed successfully.

Off: Power is off.

Network LED

Flashing green: Data transmission or reception is in progress through the WAN port.

Off: No link.

Phone 1 LED

Phone 2 LED

Steady green: On hook.

Slow flashing green: Off hook.

Fast flashing green three times, then repeats: The analog device failed to register.

Fast flashing green: A factory reset is performed successfully.

Off: The port is not configured.

Problem Report Tool (PRT) Button

Press this button to create a problem report using the Problem Report Tool.

Note

This button is not a power button. When you press this button, a problem report is generated and uploaded to a server for the system administrator.

Problem Report Tool (PRT) LED

Flashing amber: The PRT is preparing the data for the problem report.

Fast Flashing amber: The PRT is sending the problem report log to the HTTP server.

Solid green for five seconds, then off: The PRT report was sent successfully.

Fast flashing green: A factory reset is performed successfully.

Flashing red: The PRT report failed. Press the PRT button once to cancel the flashing, then press again to trigger a new PRT.

- Problem Report Tool Button

Problem Report Tool Button

The Problem Report Tool (PRT) button is on the ATA top panel. Press the PRT button, and a log file is prepared and uploaded to the server for troubleshooting your network.

You can instruct your analog phone users to press the PRT button on the ATA device to start the PRT log file process.

One of the following must be completed to upload the PRT log file from the ATA:

- Set up the HTTP server to upload the PRT log file from the ATA.

- Configure the customer support upload URL to best suit your needs, and apply it to the ATA.

ATA 191 and ATA 192 Back Panel

The following figures shows the different ports and buttons found on the back of your ATA.

ATA 191 Back Panel

ATA 192—Back Panel

ATA 191 and ATA 192 Back Panel Items

Item

Description

RESET

To restart the ATA, use a paper clip or similar object to press this button briefly.

To restore the factory default settings, press and hold for about 10 seconds.

The LED behaviour for the factory reset:

1. After you press and hold the button for about 10 seconds, the Power LED is fast flashing green.

2. After the factory reset is performed successfully, all LEDs are fast flashing green for about 5 seconds.

PHONE 1

Use an RJ-11 phone cable to connect an analog phone or fax machine.

PHONE 2

Use an RJ-11 phone cable to connect a second analog phone or fax machine.

ETHERNET (ATA 192 only)

Use an Ethernet cable to connect your ATA to a device on your network, such as a computer.

NETWORK

Use an Ethernet cable to connect to the network.

DC 5V POWER

Use the power adapter that was provided to connect to a power source.

Install Your Cisco ATA

You can use either Category 3/5/5e/6 cabling for 10-Mbps connections, but you must use Category 5/5e/6 for 100-Mbps connections.

Procedure

Step 1

Connect the power supply to the Cisco DC Adapter port.

Step 2

Connect a straight-through Ethernet cable from the network to the network port on the ATA. Each ATA ships with one Ethernet cable in the box.

ATA Voice Quality

The ATA can be custom provisioned within a wide range of configuration parameters. The sections below describe the factors that contribute to voice quality.

- Supported Codecs

- SIP Proxy Redundancy

- Other ATA Voice Quality Features

Supported Codecs

The ATA supports the codecs listed below. You can use the default settings or configure the codec settings in the Audio Configuration section of the Line 1 and Line 2 Settings (PHONE 1 and PHONE 2) page.

Supported Codecs

Codec

Description

G.711 (A-law and mu-law)

Very low complexity codecs that support uncompressed 64 kbps digitized voice transmissions at one through ten 5 ms voice frames per packet. These codecs provide the highest narrow-band voice quality and uses the most bandwidth of anyof the available codecs.

G.726-32

Very low complexity codecs that support uncompressed 64 kbps digitized voice transmissions at one through ten 5 ms voice frames per packet. These codecs provide the highest narrow-band voice quality and uses the most bandwidth of anyof the available codecs.

G.729a

ITU G.729 voice coding algorithm used to compress digitized speech. G.729a is a reduced complexity version of G.729 requiring about half the processing power of G.729. The G.729 and G.729a bit streams are compatible and interoperable, but not identical.

SIP Proxy Redundancy

An average SIP Proxy Server can handle tens of thousands of subscribers. A backup server allows an active server to be temporarily switched out for maintenance. The ATA supports the use of backup servers to minimize or eliminate service disruption.

A simple way to support proxy redundancy is to specify a SIP Proxy Server. The ATA sends a DNS NAPTR or SRV query to the DNS server. If configured, the DNS server returns SRV records that contain a list of servers for the domain, with their hostnames, priority, listening ports, and so forth. The ATA tries to contact the servers in the order of the priority. The server with a lower number has a higher priority. Up to 6 NAPTR records and 12 SRV records are supported in a query. And an SRV record can associate with up to 10 A records.

When the ATA fails to communicate with the primary server, the ATA can failover to a lower-priority server. If configured, the ATA can restore the connection back to the primary. Failover and failback support switches between servers with different SIP transport protocols. The ATA doesn't perform failback to the primary server during an active call until the call ends and the failback conditions are met.

Example of Resource Records from the DNS Server

as1bsoft     3600    IN NAPTR 50   50  "s"  "SIPS+D2T"     ""  _sips._tcp.tlstest
3600    IN NAPTR 90   50  "s"  "SIP+D2T"      ""  _sip._tcp.tcptest
3600    IN NAPTR 100  50  "s"  "SIP+D2U"      ""  _sip._udp.udptest

_sips._tcp.tlstest  SRV 1 10 5061 srv1.sipurash.com.
SRV 2 10 5060 srv2.sipurash.com.
_sip._tcp.tcptest   SRV 1 10 5061 srv3.sipurash.com.
SRV 2 10 5060 srv4.sipurash.com.
_sip._udp.udptest   SRV 1 10 5061 srv5.sipurash.com.
SRV 2 10 5060 srv6.sipurash.com.

srv1     3600    IN    A   1.1.1.1
srv2     3600    IN    A   2.2.2.2
srv3     3600    IN    A   3.3.3.3
srv4     3600    IN    A   4.4.4.4
srv5     3600    IN    A   5.5.5.5
srv6     3600    IN    A   6.6.6.6

The following example shows the priority of the servers from the perspective of the ATA.

Priority       IP Address     SIP Protocol    Status
1st            1.1.1.1            TLS            UP
2nd            2.2.2.2            TLS            UP
3rd            3.3.3.3            TCP            UP
4th            4.4.4.4            TCP            UP
5th            5.5.5.5            UDP            UP
6th            6.6.6.6            UDP            UP

The ATA always sends SIP messages to the available address with the top priority and with the status UP in the list. In the example, the ATA sends all the SIP messages to the address 1.1.1.1. If the address 1.1.1.1 in the list is marked with the status DOWN, then the ATA communicates with 2.2.2.2 instead. The ATA can restore the connection back to 1.1.1.1 when the specified failback conditions are met. For more details about failover and failback, see SIP Proxy Failover and SIP Proxy Fallback.

- SIP Proxy Failover

- SIP Proxy Fallback

SIP Proxy Failover

The ATA performs a failover in any of these cases:

- The ATA sends SIP messages and doesn't get responses from the server.

- The server responds with a code that matches the specified code in Try Backup RSC.

- The ATA gets a TCP disconnection request.

We strongly recommend that you set the Auto Register When Failover to yes when SIP Transport is set to AUTO.

ATA Failover Behavior

When the ATA fails to communicate with the currently connected server, it refreshes the server list status. The unavailable server is marked with the status DOWN in the server list. The ATA tries to connect to the top-priority server with the status UP in the list.

In the following example, the addresses 1.1.1.1 and 2.2.2.2 aren’t available. The ATA sends SIP messages to 3.3.3.3, which has the top priority among the servers with the status UP.

Priority       IP Address     SIP Protocol    Status
1st            1.1.1.1            TLS          DOWN
2nd            2.2.2.2            TLS          DOWN
3rd            3.3.3.3            TCP          UP
4th            4.4.4.4            TCP          UP
5th            5.5.5.5            UDP          UP
6th            6.6.6.6            UDP          UP

In the following example, there are two SRV records from the DNS NAPTR response. For each SRV record, there are three A records (IP addresses).

Priority       IP Address     SIP Protocol    Server            Status
1st            1.1.1.1            UDP         SRV1              DOWN
2nd            1.1.1.2            UDP         SRV1              UP
3rd            1.1.1.3            UDP         SRV1              UP
4th            2.2.2.1            TLS         SRV2              UP
5th            2.2.2.2            TLS         SRV2              UP
6th            2.2.2.3            TLS         SRV2              UP

Let's assume that the ATA failed to connect to 1.1.1.1 and then registered to 1.1.1.2. When 1.1.1.2 goes down, ATA behavior depends on the setting of Proxy Fallback Intvl.

- When Proxy Fallback Intvl is set to 0, the ATA tries with the addresses in this order: 1.1.1.1, 1.1.1.3, 2.2.2.1, 2.2.2.2, 2.2.2.3.

- When Proxy Fallback Intvl is set to a value other than zero, the ATA tries with the addresses in this order: 1.1.1.3, 2.2.2.1, 2.2.2.2, 2.2.2.3.

SIP Proxy Fallback

The proxy fallback requires a value other than zero specified in the Proxy Fallback Intvl field under the Proxy and Registration section in the ATA administration web page. If you set this field to 0, the SIP proxy failback feature is disabled.

The time when the ATA triggers a failback depends on the ATA configuration and the SIP transport protocols in use.

To enable the ATA to perform failback between different SIP transport protocols, set SIP Transport to AUTO under the Proxy and Registration section from Voice > Line (n) of the ATA administration web page.

Failback from a UDP Connection

The failback from a UDP connection is triggered by SIP messages. In the following example, the ATA first failed to register to 1.1.1.1 (TLS) at the time T1 since there's no response from the server. When SIP Timer F expires, the ATA registers to 2.2.2.2 (UDP) at the time T2 (T2=T1+SIP Timer F). The current connection is on 2.2.2.2 via UDP.

Priority       IP Address     SIP Protocol    Status
1st            1.1.1.1            TLS          DOWN        T1 (Down time)
2nd            2.2.2.2            UDP          UP
3rd            3.3.3.3            TCP          UP

The ATA has the following configuration:

<Proxy_Fallback_Intvl_n_ ua="na">60</Proxy_Fallback_Intvl_n_>
<Register_Expires_n_ ua="na">3600</Register_Expires_n_>
<SIP_Timer_F ua="na">16</SIP_Timer_F>

where n is the extension number.

The ATA refreshes the registration at time T2 (T2=(3600-16)*78%). The ATA checks the address list for the availability of the IP addresses and the down time. If T2-T1 > = 60, the failed server 1.1.1.1 resumes back to UP and the list is updated to the following. The ATA sends SIP messages to 1.1.1.1.

Priority       IP Address     SIP Protocol    Status
1st            1.1.1.1            TLS           UP
2nd            2.2.2.2            UDP           UP
3rd            3.3.3.3            TCP           UP

Failback from a TCP or TLS Connection

The failback from a TCP or TLS connection is triggered by the parameter Proxy Fallback Intvl. In the following example, the ATA failed to register to 1.1.1.1 (UDP) at the time T1 and thus registered to 2.2.2.2 (TCP). The current connection is on 2.2.2.2 via TCP.

Priority       IP Address     SIP Protocol    Status
1st            1.1.1.1            UDP          DOWN        T1 (Down time)
2nd            2.2.2.2            TCP          UP
3rd            3.3.3.3            TLS          UP

The ATA has the following configuration:

<Proxy_Fallback_Intvl_n_ ua="na">60</Proxy_Fallback_Intvl_n_>
<Register_Expires_n_ ua="na">3600</Register_Expires_n_>
<SIP_Timer_F ua="na">16</SIP_Timer_F>

where n is the extension number.

The proxy fallback interval (60 seconds) counts down from T1. The ATA triggers proxy failback at the time of T1+60. If you set the proxy fallback interval to 0 in this example, the ATA keeps the connection on 2.2.2.2.

Other ATA Voice Quality Features

Silence Suppression and Comfort Noise Generation

Voice Activity Detection (VAD) with Silence Suppression reduces the bandwidth needed for a single call, making it possible for your network to support more calls overall. VAD distinguishes between speech and non-speech signals, and Silence Suppression removes the natural silences that occur in a conversation. The IP bandwidth is used only to transmit speech.

Comfort Noise Generation provides white noise when nobody is talking so you know that your call is still connected.

Modem and Fax Pass-Through

The following applies to modem and fax pass-through:

- Modem pass-through mode can be triggered by predialing the Vertical Service Activation Code for the Modem Line Toggle Code. You can configure this setting in the Vertical Service Activation Codes section of the Regional page.

- A CED/CNG tone or an NSE event triggers FAX pass-through mode.

- Echo canceller is automatically disabled for Modem passthrough mode.

- Echo canceller is disabled for FAX pass-through if FAX Disable ECAN (Line 1 or 2 tab) is set to “Yes” for that line. In this case, FAX pass-through is the same as Modem pass-through.

- Call waiting and silence suppression are automatically disabled for both FAX and Modem pass-through. Out-of-band DTMF transmission is disabled during modem or fax passthrough.

Adaptive Jitter Buffer

The ATA can buffer incoming voice packets to minimize the impact of variable network delays. This process is known as jitter buffering. The size of the jitter buffer adjusts to changing network conditions. The ATA has a Network Jitter Level control setting for each line of service. The jitter level determines how aggressively the 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. You can use the default settings or configure this feature in the Network Settings section of the "Voice Settings Configuration" chapter.

Adjustable Audio Frames Per Packet

This feature allows you to set the number of audio frames contained in one RTP packet. Packets can be adjusted to contain from 1 to 10 audio frames. Increasing the number of packets decreases the bandwidth utilized, but it also increases delay and may affect voice quality. You can configure this setting in the RTP Parameters section of the SIP page.

DTMF Relay

The ATA may relay DTMF digits as out-of-band events to preserve the fidelity of the digits. This action enhances the reliability of DTMF transmission required by many IVR applications such as dial-up banking and airline information. You can configure this setting in the RTP Parameters section of the SIP page.

Call Progress Tones

The ATA has configurable call progress tones. Call progress tones are generated locally on the ATA and alert you to a call's status. Parameters for each type of tone, such as a dial tone, may include frequency and amplitude of each component, and cadence information. You can keep the default settings or configure these tones in the Call Progress Tones section of the Regional page.

Call Progress Tone Pass Through

This feature allows you to hear the call progress tones (such as ringing) that are generated from the far-end network.

Echo Cancellation

Impedance mismatch between the phone and the IP Telephony gateway phone port can lead to near-end echo. The ATA has a near-end echo canceller that compensates for impedance mismatch. The ATA also implements an echo suppressor with Comfort Noise Generator (CNG) so that any residual echo is not noticeable. This feature is enabled by default. You can configure this setting in the Audio Configuration of the Line 1 and Line 2 Settings (PHONE 1 and PHONE 2) page.

Hook Flash Events

The ATA signals hook flash events to the proxy during a connected call. This feature can be used to provide advanced mid-call services with third-party-call control.

- Depending on your service provider, you may need to disable Call Waiting Service, Three Way Conference Service, or Three Way Call Service. These three features could prevent the signaling of a hook flash event to the softswitch. You can configure these settings in the Supplementary Service Subscription section of the Line 1 and Line 2 Settings (PHONE 1 and PHONE 2) page.

- The Hook Flash setting determines the time period required for hook flash detection. It is in the Control Timer Values section of the SIP page.

Configurable Dial Plan with Interdigit Timers

The ATA has three configurable interdigit timers:

- The initial timeout—signals that a phone is taken off hook.

- A long timeout—signals the end of a dialed string.

- A short timeout—signals that more digits are expected.

Polarity Control

The ATA allows the polarity to be set when a call is connected and when a call is disconnected. This feature is required to support some pay phone system and answering machines. You can configure these settings in the FXS Port Polarity Configuration section of the Line 1 and Line 2 Settings (PHONE 1 and PHONE 2) page.

Calling Party Control

Calling Party Control (CPC) momentarily removes the voltage between the tip and the ring signals, signaling that the calling party has hung up. This feature is useful for auto-answer equipment. You can configure these settings in the Control Timer Values section of the Regional page.

Encryption of SIP Messages using SIP over TLS

You can enable SIP over Transport Layer Security (TLS) to encrypt the SIP messages between the service provider and your business. SIP over TLS relies on the TLS protocol to encrypt the signaling messages. You can configure the SIP Transport parameter in the SIP Settings section of the Line 1 and Line 2 Settings (PHONE 1 and PHONE 2) page.

Secure Calling Using SRTP

Voice packets are encrypted by using Secure Real-Time Transport Protocol (SRTP). This function is implemented on a standards basis (RFC4568). Secure call service (Secure Call Serv) is enabled by default. It is located in the Supplementary Service Subscription section of the Line 1 and Line 2 Settings (PHONE 1 and PHONE 2) page. When this service is enabled, you can activate secure calling by pressing the star (*) key before dialing a phone number. You can also enable the Secure Call Setting to encrypt all calls from a phone.

DNS NAPTR Support

The Line 1 and Line 2 (PHONE 1 and PHONE2) can select the SIP transport protocol (TPC, UDP, or TLS) automatically based on the Name Authority Pointer (NAPTR) records on the DNS server. Typically, the line uses the protocol with the highest priority in the records.

Chapter 2. New and Changed Information

- New and Changed for Firmware Release 11.2(4)

- New and Changed for Firmware Release 11.2(3)

- New and Changed for Firmware Release 11.2(2)

- New and Changed for Firmware Release 11.2(1)

New and Changed for Firmware Release 11.2(4)

Revision

New and Changed

Updated the FAX Enable T38 parameter for the feature Outbound Fax Refinement

Audio Configuration

Updated the topics for the feature OPTIONS Support in NAT Keep Alive Messages

NAT Settings

Added the missing topic for the feature Log Management for Users

Crash Dump

Updated the topics for the feature Default Web Access Protocol Changes to HTTPS

Access the Phone Web Interface

Cisco ATA 192 Web Access Fields

Cisco ATA 191 Web Access Fields

Cisco ATA 192 Remote Access Fields

Troubleshoot Your Fax

Updated the topics for the feature Increased Maximum Number of A Records for SRV Record

SIP Proxy Redundancy

New and Changed for Firmware Release 11.2(3)

Revision

New and Changed

Added the topic for the HTTP Proxy Support feature

HTTP Proxy (ATA 191 and 192)

Updated the topics to add the value range of the parameters Call Back Delay, VMWI Refresh Intvl, DTMF Playback Length, DTMF Playback Level, and DTMF Twist

Control Timer Values (Sec)

Miscellaneous

New and Changed for Firmware Release 11.2(2)

Revision

New and Changed

Updated the parameter SIP Transport

SIP Settings

Updated the topic to add the new feature DNS NAPTR Support

Other ATA Voice Quality Features

Updated the topic to add more examples

SIP Proxy Redundancy

Added the topic for the SIP proxy redundancy feature

SIP Proxy Failover

SIP Proxy Fallback

Updated the topic to add the parameter Auto Register When Failover

Proxy and Registration

Updated the topic to mention the maximum number of NAPTR records and SRV records supported in a query

SIP Proxy Redundancy

Added a new task topic on how to secure a line by automatic SRTP/RTP selection

Set up a Secure Line

Updated the Call Feature Settings table with a new parameter Secure Call Option

Call Feature Settings

New and Changed for Firmware Release 11.2(1)

Revision

New and Changed

Updated the navigation path

Debug Log Module

Added the topics for the E911 feature

Emergency Calls

Emergency Call Support Background

Emergency Call Support Terminology

Configure the ATA to Make Emergency Calls

E911 Geolocation Configuration

Updated RTP Parameters table to add the Call Statistics parameter

RTP Parameters

Added a topic for the reporting End-of-Call Statistics feature

Call Statistics are Not Available in the Server

Added a task for the remote PRT generation feature

Generate a Problem Report Remotely

Updated the topic for the remote PRT generation feature

PRT Viewer

Added a topic how to address the issue when ATA 11.1.0MSR3-9 and older doesn't upgrade

ATA with Firmware Release 11.1.0MSR3-9 and Older Doesn't Upgrade

Chapter 3. Quick Setup for Voice over IP Service

- Set Up Voice over IP

Set Up Voice over IP

The Quick Setup page displays when you log in to the ATA web page for the first time. Use this page to connect your phone to your provider’s Voice over IP network.

Note

You need an Internet connection to link with your service provider’s network. With the default network settings, your ATA has Internet connectivity, if the WAN port connects to a port on your router.

Procedure

Step 1

For Line 1 and Line 2, enter the settings for the phone services used by the phones or fax connected to the PHONE1 and PHONE2 ports.

- Proxy: Enter the IP address of the service provider’s proxy server.

- Display Name: Enter the name or DN that you want to use to identify your account. This name typically is used as your Caller ID name.

- User ID: Enter the user ID that is required to log in to your Internet account.

- Password: Enter the password that is required to log in to your Internet account.

- Dial Plan in (Line section only): Keep the default settings (recommended) or edit the dial plan to suit your site.

Step 2

Click Submit to save your settings. The voice service restarts.

Step 3

To verify your progress, perform the following tasks:

1. Check if phone LED is a steady green, indicating that the phone has registered.

If the line is not registered, refresh the browser several times because it can take a few seconds for the registration to complete. Also verify that your Internet Settings, including DNS server settings, are configured according to the information from your ISP.

2. Use an external phone to call the phone number assigned to you by your ISP. Verify that the phone rings and you have two-way audio on the call.

Chapter 4. Network Configuration

- Web-Based Configuration Utility

- Basic Setup

- Advanced Settings

- Application

Web-Based Configuration Utility

Your phone system administrator can allow you to view the phone statistics and modify some or all the parameters. This section describes the features of the phone that you can modify with the phone web user interface.

- Access the Phone Web Interface

- Allow Web Access to the ATA

Access the Phone Web Interface

If your service provider has disabled access to the configuration utility, contact the service provider before proceeding.

Procedure

Step 1

Ensure that the computer can communicate with the phone. No VPN in use.

Step 2

Start a web browser.

Step 3

Enter the IP address of the phone in your web browser address bar.

- User or Admin Access: https://<ip address>:<port>/, and then enter the username and password.

For example, https://10.64.84.147/

Allow Web Access to the ATA

To view the ATA parameters, enable the configuration profile. To make changes to any of the parameters, you must be able to change the configuration profile. Your system administrator might have disabled the option to make the ATA web user interface viewable or writable.

For more information, see the Cisco ATA 191 and 192 Multiplatform Firmware Provisioning Guide

Before you begin

Access the phone administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Click System.

Step 2

In the System Configuration section, set Enable Web Server to Yes.

Step 3

To update the configuration profile, click Submit All Changes after you modify the fields in the phone web user interface.

The phone reboots and the changes are applied.

Step 4

To clear all changes that you made during the current session (or after you last clicked Submit All Changes), click Undo All Changes. Values return to their previous settings.

Basic Setup

Use the Network Setup > Basic Setup pages to configure your Internet connection, local network settings (ATA 192 only), and your time settings.

- Network Service (ATA 192 Only)

- Basic Settings

- IPv4 Settings

- IPv6 Settings

- IPv4 LAN Settings (ATA 192 Only)

- IPv6 LAN Setting (ATA 192 Only)

- Time Settings

Network Service (ATA 192 Only)

Use the Network Setup > Basic Setup > Network Service page to configure the operating mode of the ATA 192.

After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

You can configure the ATA to operate in one of the following modes:

- NAT: Network Address Translation (NAT) allows multiple devices on a private network to share a public, routable IP address. In order for theVoice over IP service to co-exist with NAT, some form of NAT traversal is required either on the ATA or another network device. Use this option if your ATA connects to one network on the WAN port and to another network on the LAN port. This option is selected by default and is suitable for most deployments.

- Bridge: Bridged mode is used if the ATA is acting as a bridge device to another router. Choose this option if your ATA bridges a network to its LAN port (with connected devices also in the 10.0.0x range).

- Monitor Network Drop on Internet Port Only: This parameter is used for how to report the link state when both Ethernet(LAN) and Network(WAN) ports are connected.

- Off: If you unplug and plug the WAN cable, meanwhile the LAN port state is still UP, then the ATA doesn't perform any operation.

- On: If you unplug and then plug the WAN cable, then the ATA triggers a warm reboot even though the LAN port state is still UP. In this case, the ATA will try to register again.

Note

- The parameter takes effect only when you select the Bridge mode.

- The parameter doesn't take effect when only the WAN port is connected.

Basic Settings

Use the Network Setup > Basic Settings page to set up your basic network settings.

Basic Settings

Field

Description

Domain Name

The domain name, if specified by your ISP. Otherwise, leave the field blank.

Host Name

The name of the ATA. The default value is the model number. Your ISP may specify a host name to use.

Stack mode

Choose the stack mode for network; there are three modes can be set: IPv4 only, Pv6 Only, or Dual.

Signaling Preference

Choose the SIP packet preference, either IPv4 or IPv6.

Media Preference

Choose the RTP packet preference, either is IPv4 or IPv6.

IPv4 Settings

Use the Network Setup > Basic Setup > IPv4 Settings page to set up your IPv4 connection.

Enter the settings as described in the table. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Internet Connection Type

Field

Description

Connection Type

Specify the Internet addressing method that your ISP requires. Default setting: Automatic Configuration - DHCP

- Automatic Configuration - DHCP: Use this setting if your ISP dynamically provides an IP address. No additional settings are required on this page.

- Static IP: Use this setting if your ISP assigned a static IP address. Complete the fields that appear.

- PPPoE (DSL service): Some DSL-based ISPs use PPPoE (Point-to-Point Protocol over Ethernet) to establish Internet connections. If you are connected to the Internet through a DSL line, check with your ISP to see if they use PPPoE. Complete the fields that appear.

Static IP Settings

- Internet IP Address and Subnet Mask: Enter the IP address and subnet mask that was assigned to your account by your service provider. This address is seen by external users on the Internet.

- Default Gateway: Enter the Gateway IP Address that was provided by your ISP.

If needed, you can adjust the MTU and Optional Settings.

PPPoE Settings

- User Name and Password: Enter the user name and password that you use to log in to your ISP network through a PPPoE connection.

- Service Name: If provided by your ISP, enter the Service Name.

- Connect on Demand: You can configure the ATA to disconnect your Internet connection after a specified period of inactivity (Max Idle Time). If your Internet connection has timed out, this feature also enables the ATA to re-establish your connection when you attempt to access the Internet again. If you choose this option, also set the Max Idle Time.

- Keep Alive: This option keeps you connected to the Internet indefinitely, even when your connection sits idle. If you choose this option, also set the Redial Period, which is the interval at which the ATA verified Internet connectivity. The default period is 30 seconds.

If needed, you can adjust the MTU and Optional Settings.

MTU

The Maximum Transmission Unit (MTU) setting specifies the largest protocol data unit (in bytes) permitted for network transmission. Generally, a larger MTU means greater efficiency. However, a larger packet may cause delays for other traffic and is more likely to become corrupted. Usually, you keep the default setting to allow the ATA to choose the appropriate MTU. To specify the MTU, select Manual, and then enter the number of bytes.

Optional Settings

Field

Description

DNS Server Order

Choose the preferred method for choosing a DNS server.

- DHCP-Manual—The DNS server settings from the network server takes precedence, and your entries in the DNS fields are used only as a backup.

- Manual-DHCP—Your entries in the DNS fields take precedence, and the DNS server settings from the network server are used as a backup.

- Manual—Your entries in the DNS fields are used to choose a DNS server.

Primary DNS

Set the Primary DNS for IPv4.

Secondary DNS

Set the Secondary DNS for IPv4.

IPv6 Settings

Use the Network Setup > Basic Setup > IPv6 Settings page to set up your IPv6 connection.

Enter the settings as described in the table. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

IPv6 Settings

Field

Description

Internet Connection Type

Specify the Internet addressing method that your ISP requires. Default setting: Automatic Configuration - DHCP

Automatic Configuration - DHCP: Use this setting if your ISP dynamically provides an IP address. No additional settings are required on this page.

Static IP: Use this setting if your ISP assigned a static IP address. Complete the following fields:

- Internet IPv6 Address and Prefix Length—Enter the IPv6 address and prefix length that was assigned to your account by your service provider. The public sees this address.

- Default Gateway—Enter the Gateway IPv6 Address that was provided by your ISP.

PPPoE (DSL service): Some DSL-based ISPs use PPPoE (Point-to-Point Protocol over Ethernet) to establish Internet connections. If you are connected to the Internet through a DSL line, check with your ISP to see if they use PPPoE. Complete the following fields:

- User Name and Password—Enter the username and password that you use to log in to your ISP network through a PPPoE connection.

- Service Name—If provided by your ISP, enter the Service Name.

- Connect on Demand—You can configure the ATA to disconnect your Internet connection after a specified period of inactivity (Max Idle Time). If your Internet connection has timed out, this feature enables the ATA to automatically reconnect when you try to access the Internet again. If you choose this option, also set the Max Idle Time.

- Keep Alive—This option keeps you connected to the Internet indefinitely, even when your connection sits idle. If you choose this option, also set the Redial Period, which is the interval at which the ATA verified Internet connectivity. The default period is 30 seconds.

Optional Settings

Field

Description

DNS Server Order

Choose the preferred method for choosing a DNS server.

- DHCP-Manual—The DNS server settings from the network server takes precedence, and your entries in the DNS fields are used only as a backup.

- Manual-DHCP—Your entries in the DNS fields take precedence, and the DNS server settings from the network server are used as a backup.

- Manual—Your entries in the DNS fields are used to choose a DNS server.

Allow Auto Configuration.

Enable if you want to allow Auto Configuration.

Primary DNS

Set the Primary DNS for IPv6.

Secondary DNS

Set the Secondary DNS for IPv6.

IPv4 LAN Settings (ATA 192 Only)

Use the Network Setup > Basic Setup > IPv4 LAN Settings page to set the IP address and subnet mask for your local network. Also configure the settings for the built-in DHCP server (ATA 192 only).

After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Router IP

Enter the Local IP Address and Subnet Mask for your local network. The default setting is 192.168.15.1 with a subnet mask of 255.255.255.0.

DHCP Server Setting

Field

Description

DHCP Server

The ATA can use the built-in DHCP server to dynamically assign IP addresses to connected devices. Click Enabled to enable the DHCP server, or clickDisabled to disable this feature.

Default setting: Enable

IP Reservation

Click the Show DHCP Reservation button to view and manage the DHCP client list. Click the Hide DHCP Reservation button to hide the list. When the list is displayed, you can perform the following tasks:

- To reserve a static IP address for a current DHCP client: Check the box for the client in the Select Clients from DHCP Tables list. ClickAdd Clients. The selected clients are added to the Clients Already Reserved list. These clients have static IP addresses that do not change.

- To add a client that isn't in the Select Clients from DHCP Tables list: Type a name for the client in the Enter Client Name box. Enter an IP address for this client in the Assign IP Address box. Enter the MAC address in the following format:00:00:00:00:00:00. Click Add.

- To remove a client from the Clients Already Reserved list: Check the box for the client. Click Remove.

Default Gateway

Enter the IP address of the default gateway to be used by the DHCP clients.

Default setting: 192.168.15.1 (the IP address of the ETHERNET (LAN) interface)

Starting IP Address

Enter the first address in the range of addresses assigned dynamically by the DHCP server.

Default setting: 192.168.15.100

Maximum DHCP Users

Enter the maximum number of devices that can dynamically receive, or “lease,” DHCP addresses from the DHCP server.

Default setting: 50

IMPORTANT: Typically, the ATA can support up to five connected computers for business-related tasks such as web browsing and viewing email. The ATA is not designed to support streaming music, video, games, or other network traffic-intensive tasks.

Client Lease Time

Enter the number of minutes that a dynamically assigned IP address can be in use, or “leased.” After this time elapses, a client device has to request a DHCP lease renewal. Use 0 to represent 1 day, 9999 never expire.

Default setting: 0

Option 66

Provides provisioning server address information to hosts that request this option. Server information can be defined in one of three ways:

- None: The ATA uses its own TFTP server to source provisioning files, so it returns its own local IP address to the client.

- Remote TFTP Server: The ATA was configured by using this method, and received server information through Option 66 on its WAN interface. In response to client requests, it provides the remote TFTP server information.

- Manual TFTP Server: Allows the manual configuration of a configuration server address. This option is used to provide either an IP address or a fully qualified hostname. But the ATA also accepts and offers a full URL including protocol, path, and filename to meet the requirements of specific clients.

Default setting: None

TFTP Server

If you chose Manual TFTP Server for Option 66, enter the IP address, hostname, or URL of the TFTP server.

Default setting: blank

Option 67

Provides a configuration or bootstrap filename to hosts that request this option. This option is used with option 66 to allow a client to form an appropriate TFTP request for the file.

Default setting: blank

Option 159

Provides a configuration URL to clients that request this option. An option 159 URL defines the protocol and path information by using an IP address for clients that cannot use DNS. For example: https://10.1.1.1:888/configs/bootstrap.cfg

Default setting: blank

Option 160

Provides a configuration URL to clients that request this option. An option 160 URL defines the protocol and path information by using a fully qualified domain name for clients that can use DNS. For example: https://myconfigs.cisco.com:888/configs/bootstrap.cfg

Default setting: blank

DNS Proxy

When enabled, the DNS proxy relays DNS requests to the current public network DNS server. It also replies as a DNS resolver to the client device on the network. Click Enabled to enable this feature, or clickDisabled to disable it. If DNS proxy is disabled, then DHCP clients are offered DNS server information by using the Static DNS servers or by using the servers specified for the INTERNET (WAN) interface.

IPv6 LAN Setting (ATA 192 Only)

Use the Network Setup > Basic Setup > IPv6 LAN Settings page to set up your IPv6 LAN connection.

Enter the settings as described in the table. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Internet Connection Type

Field

Description

DHCP Server

Click Enabled to enable the DHCP server, or click Disabled to disable this feature.

Default setting: Enable

Address Assign Type

Choose the address assign type: SLAAC/DHCPv6.

DHCPv6 Delegation

Choose whether to support DHCPv6 delegation, if Yes, user can’t configure IPv6 Address Prefix.

IPv6 Address Prefix

Set the IPv6 address prefix for IPv6 LAN interface, the Prefix Length is fixed to 64.

IPv6 Address Length

Set the IPv6 address prefix length for IPv6 LAN interface.

Range:1-112

IPv6 Static DNS

Set the IPv6 Static DNS.

LAN IPv6 Address

Display the LAN IPv6 address information.

Time Settings

Use the Network Setup > Basic Setup > Time Settings page to set the system time for the ATA. By default, the system time is set automatically by using a Network Time Protocol (NTP) server. You can configure the system time manually. In addition, you can use this page to specify your time zone, enable Daylight Saving adjustments, and modify related settings.

After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Note

If the ATA doesn't receive the response from the NTP server, then the system time for the ATA uses the build date and time of the firmware.

User Manual

If you prefer to set the system manually, click User Manual and then enter the date and time.

Time Settings

Field

Description

Date

Enter the date in the following order: four-digit year, month, day.

Time

Enter the time in the following order: hour (from 1 to 24), minutes, and seconds.

Time Zone

To use a time server to establish the time settings, select Time Zone. Then complete the fields in this section.

Time Zone Settings

Field

Description

Time Zone.

Choose the time zone for the site where the ATA is in operation. Default setting: (GMT-08:00) Pacific Time (USA & Canada).

Adjust Clock for Daylight Saving Changes.

Check the box if you want to automatically adjust the time when Daylight Savings Time is in effect. Otherwise, uncheck the box.

Time Server Address.

To use the ATA’s default Network Time Protocol (NTP) server, select Auto from the drop-down list. If you want to specify the NTP server, select Manual, and then enter the NTP server address.

Default setting: Auto

Resync Timer

Enter the Resync timer interval value (in seconds). This timer controls how often the ATA resynchronizes with the NTP server.

Default setting: 3600 seconds

Auto Recovery After Reboot

Choose this option to allow the ATA to automatically reconnect to the time server after a system reboot.

Default setting: Disabled

Advanced Settings

Use the Network Setup > Advanced Settings pages to configure features including port flow control, MAC address cloning, VPN passthrough, and VLAN.

- Port Setting (ATA 192 Only)

- MAC Address Clone

- VPN Passthrough (ATA 192 Only)

- VLAN

- CDP and LLDP

Port Setting (ATA 192 Only)

Use the Network Setup > Advanced Settings > Port Setting page to set the ETHERNET (LAN) port attributes.

After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Port Settings

Field

Description

Flow Control

Flow control is a mechanism that temporarily stops the transmission of data on a port. For example, a device is transmitting data faster than some other part of the network can accept it. The overwhelmed network element halts the transmission of the sender for a specified time.

Choose Enabled to enable this feature, or choose Disabled to disable this feature.

Default setting: Enabled

Speed Duplex

Choose the duplex mode. You can select from Auto-negotiate, 10 Half, 10 Full, 100 Half and 100 Full. Cisco recommends that choosing Auto-negotiate to automatically select the appropriate mode for the traffic. Use caution with other settings. Problems can result if you choose a setting that is not appropriate for the network devices.

Default setting: Auto-negotiate

MAC Address Clone

A MAC address is a 12-digit code assigned to a unique piece of hardware for identification purposes. Some ISPs require that you register a MAC address in order to access the Internet. If you previously registered your account with another MAC address, it may be convenient to assign that MAC address to your ATA. You can use the Network Setup > Advanced Settings > MAC Address Clone page to assign a MAC address that you previously registered with your Service Provider.

After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

MAC Address Clone Settings

Field

Description

MAC Clone

Click Enabled to enable MAC address cloning, or click Disabled to disable this feature.

Default setting: Disabled.

MAC Address

Enter the MAC address that you want to assign to your ATA. If your computer’s MAC address is the address that you previously registered for your ISP account, click Clone Your PC’s MAC. Your computer’s MAC address appears in the MAC Address field.

Default setting: the current MAC address of your ATA

VPN Passthrough (ATA 192 Only)

Use the Network Setup > Advanced Settings > VPN Passthrough page to configure VPN passthrough for IPsec, PPTP, and L2TP protocols. Use this feature if there are devices behind the ATA that require an independent IPsec tunnel. For example, a device may need to use a VPN tunnel to connect to another router on the WAN.

By default, VPN Passthrough is enabled for IPsec, PPTP, and L2TP.

After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

VPN Passthrough Settings

Field

Description

IPsec Passthrough

Internet Protocol Security (IPsec) is a suite of protocols used to implement secure exchange of packets at the IP layer. Click Enabled to enable this feature, or click Disabled to disable it.

Default setting: Enabled

PPTP Passthrough

Point-to-Point Tunneling Protocol (PPTP) allows the Point-to-Point Protocol (PPP) to be tunneled through an IP network. To disable PPTP Passthrough, select Disabled.

Default setting: Enabled

L2TP Passthrough

Layer 2 Tunneling Protocol is the method used to enable Point-to-Point sessions using the Internet on the Layer 2 level. Click Enabled to enable this feature, or click Disabled to disable it.

Default setting: Enabled

VLAN

Use the Network Setup > Advanced Settings > VLAN page to assign a VLAN ID to your network. For example, your call control system may require a particular voice VLAN ID.

After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

VLAN Settings

Field

Description

Enable VLAN.

Click Enabled to enable a VLAN, or click Disabled to disable this feature.

Default setting: Disabled

VLAN ID

The VLAN ID can be any numeral from 1 to 4094. When VLAN is enabled, the default setting is 1.

CDP and LLDP

Device discovery protocols enable directly connected devices to discover information about each other. You may wish to enable these protocols to allow your network management system to learn about your ATA and endpoints. Use the Network Setup > Advanced Settings > CDP & LLDP page to specify the settings for Cisco Discovery Protocol (CDP) and the Link Layer Discovery Protocol (LLDP). When enabled, the ATA sends messages to a multicast address and listens to the messages sent by other devices using the protocol.

After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Application

Use the Network Setup > Application pages to support voice service and any servers that you host for public access.

- Quality of Service (QoS) (ATA 192 Only)

- Port Forwarding (ATA 192 Only)

- Manually Add Port Forwarding (ATA 192 Only)

- DMZ (ATA 192 Only)

- HTTP Proxy (ATA 191 and 192)

Quality of Service (QoS) (ATA 192 Only)

Use the Network Setup > Application > QoS page to set the upstream bandwidth to suit your broadband service. This feature is enabled by default and helps to ensure that voice is prioritized during periods of heavy network traffic.

After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

QoS Settings

Field

Description

QoS Policy

Click Always On to enable QoS settings always, or click On When Phone In Use to enable it only when there is voice traffic.

Default setting: On When Phone In Use

Upstream Bandwidth

Enter the maximum available upstream bandwidth value specified by your Internet Service Provider.

Default setting: 100000 kbps

Important: Do not overstate the upstream bandwidth that you receive from your service provider. Setting this value higher than the available service bandwidth can result in traffic being dropped arbitrarily in the service provider's network.

Port Forwarding (ATA 192 Only)

Use the Network Setup > Application > Port Forwarding page if you require access to specific ports from external devices.

List of Port Forwarding

To add a port forwarding rule, click Add Entry. To edit a port forwarding rule, select it in the list and then click the pencil icon. To remove a port forwarding rule, click the delete icon.

Port Forwarding Settings

Field

Description

Number

An identification number for the port forwarding rule.

Type

The type of rule: Single Port Forwarding or Port Range Forwarding.

Status

The status of the rule: Enabled or Disabled.

Application

The application that uses this rule to access a network resource.

Port Forwarding Details

To display the details, click an entry in the List of Port Forwarding.

Port Settings

Field

Description

External Port

The port that external clients use to set up this connection.

Internal Port

The port that the ATA uses when forwarding traffic to the internal server.

Protocol

The protocol that is used: TCP or UDP.

IP Address

The IP address of the internal server accessed by this rule.

Manually Add Port Forwarding (ATA 192 Only)

Use this page to enter the port forwarding settings for an application.

Enter the settings as described. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Port Forwarding Settings

Field

Description

Port Forwarding Type

Choose the type of port forwarding:

- Single Port Forwarding: Forwards traffic for a specified port to the same or an alternate port on the target server in the LAN.

- Port Range Forwarding: Forwards traffic to a range of ports to the same ports on the target server in the LAN. See the Internet application’s documentation for the required ports or ranges.

Application Name

For single port forwarding, choose a common application from the drop-down list (such as Telnet, or DNS).

To add an application that is not on the list, choose Add a new name, and then enter the name in the Enter a Name field.

Enter a Name

If you chose Port Range Forwarding, or if you chose Add a new name in the Application Name list for Single Port Forwarding, enter a name to identify the application.

External Port, Internal Port

For Single Port Forwarding, specify the ports to use. For simplicity, the internal and external port numbers are often the same. Different external port numbers could be used to differentiate traffic of the same application type intended for different servers, or for privacy by using non-standard ports.

- External port: For single port forwarding, enter the port number that external clients use to set up a connection with the internal server.

- Internal port: For single port forwarding, enter the port number that the ATA uses when forwarding traffic to the internal server.

The correct entries appear automatically if you choose a standard application from the Application Name list for Single Port Forwarding.

Start - End Port

For Port Range Forwarding, specify the range of ports to use. Valid values are from 1 to 65535.

Protocol

Select the protocols that can be forwarded: TCP, UDP, or TCP and UDP.

IP Address

Enter the IP address of the local server that receives forwarded traffic.

For correct forwarding of traffic, local servers must either be configured with a static IP address, or be assigned a reserved IP address through DHCP. Use the Interface Setup > LAN > DHCP Server page to reserve IP addresses.

Enabled

Check the box to enable this port forwarding rule, or uncheck the box to disable it.

Default setting: Disabled

DMZ (ATA 192 Only)

Use the Network Setup > Application > DMZ page if you want a local device exposed to the Internet for a special-purpose service.

The specified network device must have its DHCP client function disabled. It must also have a reserved IP address to ensure that it is reachable at the specified IP address.

Note

A Demilitarized Zone (DMZ) is similar to Port Range Forwarding. Both features allow Internet traffic to access a resource on your private network. However, Port Range Forwarding is more secure because it only opens the ports that you specify for an application. DMZ hosting opens all the ports of one device, exposing it to the Internet.

Enter the settings as described. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

DMZ Settings

Field

Description

Staus.

Click Enabled to enable this feature, or click Disabled to disable it.

Default setting: Disabled

Private IP.

Specify the local IP address of the device that can be accessed through the DMZ.

HTTP Proxy (ATA 191 and 192)

Use the Network Setup > Application > HTTP Proxy page to set up a proxy server for the ATA to enhance security. A proxy server acts as a firewall between the ATA and Internet. After successful configuration, the ATA connects to Internet through the proxy server which protects the ATA from cyber attack.

You can set up a proxy server by either using an automatic configuration script or by manually configuring the host server (hostname or IP address) and port of the proxy server.

After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

When the ATA is configured with the HTTP proxy feature, the feature applies to all the applications that use the HTTP protocol. The applications include the following:

- Profile Rule B, C, and D

- Log Resync Request, Success, and Failure Msg

- Report Rule

- Upgrade Rule

- Custom CA URL

- E911 Request URL

- EDOS RC Server

- TR69 (ACS URL)

- PRT Upload URL

HTTP Proxy Settings

Field

Description

Proxy Mode

Choose the HTTP proxy mode:

- Auto: The ATA automatically retrieves a Proxy Auto-Configuration (PAC) file to select a proxy server. In this mode, you can determine whether to use Web Proxy Auto-Discovery (WPAD) protocol to retrieve a PAC file or manually enter a valid URL of the PAC file.

For details about the fields, see Use Auto Discovery (WPAD) and PAC URL.

- Manual: You must manually specify a server (hostname or IP address) and a port of a proxy server.

For details about the fields, see Proxy Host and Proxy Port.

- Off: You disable the HTTP proxy feature on the ATA.

Default setting: Off

Use Auto Discovery

Click Yes to use the Web Proxy Auto-Discovery (WPAD) protocol to retrieve a PAC file automatically. Click No to specify the URL of a PAC file manually.

For details about the field, see PAC URL.

Default setting: Yes

PAC URL

Specify the URL of a PAC file.

For example, http://proxy.department.branch.example.com

TFTP, HTTP, and HTTPS are supported.

The field must be configured when the Proxy Mode is Auto and Use Auto Discovery is No.

Default setting: Blank

Proxy Server Requires Authentication

Enter the authentication credentials (username and password) if the proxy server requires. This field is configured according to the actual behaviour of the proxy server.

For details about the fields, see Username and Password.

Default setting: No

Proxy Host

IP address or hostname of the proxy host server for the ATA to access. For example:

proxy.example.com

The scheme (http:// or https://) is not required.

Default setting: Blank

Proxy Port

Port number of the proxy host server. The value range is from 2 to 65535.

Default setting: 3128

Username

Username for a credential user on the proxy server.

If Proxy Mode is set to Manual and Proxy Server Requires Authentication is set to Yes, you must configure the field.

Default setting: Blank

Password

Password of the specified username for the proxy authentication purpose.

If Proxy Mode is set to Manual and Proxy Server Requires Authentication is set to Yes, you must configure the field.

Default setting: Blank

Chapter 5. Voice Settings Configuration

- Information

- System

- SIP

- Provisioning

- Regional

- Line 1 and Line 2 Settings (PHONE 1 and PHONE 2)

- User 1 and User 2

Information

Use the Voice > Information page to view information about the ATA voice application.

- Product Information

- System Status

- Line 1 and Line 2 Settings (PHONE 1 and PHONE 2)

- Custom CA Status

- Provision Status

Product Information

Product Information

Field

Description

Product Name

The product name of ATA.

Serial Number

The serial number of ATA.

Software Version

The software version of ATA.

Hardware Version

The hardware version of ATA.

MAC Address

The mac address of ATA.

Client Certificate

The client certificate of ATA.

Customization

The customization of ATA.

System Status

System Status Settings

Field

Description

Current Time

Current date and time of the system; for example, 10/3/2003 16:43:00.

Set the system time by using the Network Setup > Time Settings page.

Elapsed Time

Total time elapsed since the last reboot of the system; for example, 25 days and 18:12:36.

RTP Packets Sent

Total number of RTP packets sent, including redundant packets.

RTP Bytes Sent

Total number of RTP bytes sent.

RTP Packets Recv

Total number of RTP packets received, including redundant packets.

RTP Bytes Recv

Total number of RTP bytes received.

SIP Messages Sent

Total number of SIP messages sent, including retransmissions.

SIP Bytes Sent

Total number of bytes of SIP messages sent, including retransmissions.

SIP Messages Recv

Total number of SIP messages received, including retransmissions.

SIP Bytes Recv

Total number of bytes of SIP messages received, including retransmissions.

External IP

The External IP address used for NAT mapping.

Line 1 and Line 2 Settings (PHONE 1 and PHONE 2)

Use the Voice > Line 1 and Voice > Line 2 pages to configure the settings for calls through the PHONE 1 and PHONE 2 ports.

Enter the settings as described. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Note

In a configuration profile, the FXS parameters must include an appropriate numeral for identifying the port receiving the setting.

Custom CA Status

CA Status Settings

Field

Description

Custom CA Provisioning Status

The status of the latest custom CA (Certificate Authority) certificate download.

Custom CA Info

The successfully downloaded CA information, or “Not Installed” if no custom CA certificate was installed.

Default setting: Not Installed

Provision Status

Provision Status Settings

Field

Description

Provisioning Profile

Profile rule setting

Default setting: Empty

Provision Status

Indicate the status of last provisioning

Default setting: Empty

Provisioning Failure Reason

Reason for failure

Default setting: Empty

System

Use the Voice > System page to configure general voice system settings and to enable logging by using a syslog server. Logging can also be configured in the Administration > Logging pages.

- System Configuration

- Miscellaneous Settings

System Configuration

System Settings

Field

Description

Restricted Access Domains

Domain that Cisco IP phones responds to SIP messages only from the identified servers. Applicable to Line 1.

IVR Admin Passwd

Password for the administrator to manage the ATA by using the built-in IVR through a connected phone.

Network Startup Delay

The number of seconds of delay between restarting the voice module and initializing network interface.

Default setting: 3

Miscellaneous Settings

Miscellaneous Settings

Field

Description

DNS Query TTL Ignore

In DNS packages, the server suggests a TTL value to the client. If this parameter is set to Yes, the value from the server is ignored.

Default setting: No

SIP

Use the Voice > SIP page to configure SIP parameters and values.

Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Note

For a deeper understanding of these fields, refer to Request for Comments (RFC) 3261.

- SIP Parameters

- SIP Timer Values

- Response Status Code Handling

- RTP Parameters

- SDP Payload Types

- NAT Support Parameters

SIP Parameters

SIP Parameters Settings

Field

Description

Max Forward:

The maximum times a call can be forwarded. The valid range is from 1 to 255.

Default setting: 70

Max Redirection:

Number of times an invite can be redirected to avoid an infinite loop.

Default setting: 5.

Max Auth:

The maximum number of times (from 0 to 255) a request may be challenged.

Default setting: 2

SIP User Agent Name:

The User-Agent header used in outbound requests. If empty, the header is not included. Macro expansion of $A to $D corresponding to GPP_A to GPP_D allowed.

Default setting: $VERSION

SIP Server Name:

The server header used in responses to inbound responses.

Default setting: $VERSION

SIP Reg User Agent Name:

The User-Agent name to be used in a REGISTER request. If this value is not specified, the SIP User Agent Name parameter is also used for the REGISTER request.

Default setting: Blank

SIP Reg Starting Sequence Number:

Defines the SIP Reg message Sequence Number.

Default setting: Blank

SIP Accept Language:

Accept-Language header used. There is no default; this indicates that the ATA does not include this header. If empty, the header is not included.

Default setting: Blank

DTMF Relay MIME Type:

The MIME Type used in a SIP INFO message to signal a DTMF event.

Default setting: Application/dtmf-relay.

Hook Flash MIME Type:

The MIME Type used in a SIP INFO message to signal a hook flash event.

Default setting: Application/hook-flash.

Remove Last Reg:

Determines whether the ATA removes the last registration before submitting a new one, if the value is different. Select yes to remove the last registration, or select no to omit this step.

Default setting: No

Use Compact Header:

Determines if the ATA uses compact SIP headers in outbound SIP messages.

Select Yes to use compact SIP headers in outbound SIP messages.

Select No to use normal SIP headers.

If inbound SIP requests contain compact headers, the ATA reuses the same headers when generating the response regardless of the Use Compact Header parameter. If inbound SIP requests contain normal headers, the ATA substitutes those headers with compact headers as defined by RFC 261 when Use Compact Header is set to Yes.

Default setting: No

Escape Display Name:

Determines if the Display Name is private. Select Yes if you want the ATA to enclose the string configured in the Display Name in a pair of double quotes for out bound SIP messages. If the display name includes " or \, these will be escaped to \" and \\ within the double quotes. Otherwise, select No.

Default setting: No

RFC 2543 Call Hold:

Configures the type of call hold: a:sendonly or 0.0.0.0. Do not use the 0.0.0.0 syntax in a HOLD SDP; use the a:sendonly syntax.

Default setting: Yes

Mark All AVT Packets:

Select Yes if you want all AVT tone packets encoded for redundancy to have the marker bit set for each DTMF event.

Select No to have the marker bit set only for the first packet.

Default setting: Yes

AVT Packet Size:

Indicates the AVT Packet size according to value set in ptime or fixed 10ms.

Default setting: ptime

SIP TCP Port Min:

The lowest TCP port number that can be used for SIP sessions.

Default setting: 5060

SIP TCP Port Max:

The highest TCP port number that can be used for SIP sessions.

Default setting: 5080

CTI Enable:

Enables or disables the Computer Telephone Interface feature provided by some servers.

Default setting: no

Keep Referee When REFER Failed:

Set this parameter to Yes to configure the phone to handle NOTIFY sipfrag messages.

You can also configure this parameter in the configuration file:

<Keep_Referee_When_REFER_Failed ua="na">Yes

</Keep_Referee_When_REFER_Failed>

Caller ID Header:

Provides the option to take the caller ID from PAID-RPID-FROM,P-ASSERTEDIDENTITY, REMOTE-PARTY-ID, or FROM header.

Default setting: PAID-RPID-FROM

SIP Timer Values

SIP Timer Values Settings

Field

Description

SIP T1

RFC 3261 T1 value (round-trip time estimate), which can range from 0 to 64 seconds.

Default setting: 0.5

SIP T2

RFC 3261 T2 value (maximum retransmit interval for non-INVITE requests and INVITE responses), which can range from 0 to 64 seconds.

Default setting: 4

SIP T4

RFC 3261 T4 value (maximum duration a message remains in the network), which can range from 0 to 64 seconds.

Default setting: 5

SIP Timer B

INVITE time-out value, which can range from 0 to 64 seconds.

Default setting: 32

SIP Timer F

Non-INVITE time-out value, which can range from 0 to 64 seconds.

Default setting: 16

SIP Timer H

H INVITE final response, time-out value, which can range from 0 to 64 seconds.

Default setting: 32

SIP Timer D

ACK hang-around time, which can range from 0 to 64 seconds.

Default setting: 32

SIP Timer J

Non-INVITE response hang-around time, which can range from 0 to 64 seconds.

Default setting: 32

INVITE Expires

INVITE request Expires header value. If you enter 0, the Expires header is not included in the request. Range: 0–(2^31–1)

Default setting: 240

ReINVITE Expires

ReINVITE request Expires header value. If you enter 0, the Expires header is not included in the request. Range: 0–(2^31–1)

Default setting: 30

Reg Min Expires

Minimum registration expiration time allowed from the proxy in the Expires header or as a Contact header parameter. If the proxy returns a value less than this setting, the minimum value is used.

Default setting: 1

Reg Max Expires

Maximum registration expiration time allowed from the proxy in the Min-Expires header. If the value is larger than this setting, the maximum value is used.

Default setting: 7200

Reg Retry Intvl

Interval to wait before the ATA retries registration after failing during the last registration.

Default setting: 30

Reg Retry Long Intvl

When registration fails with a SIP response code that does not match Retry Reg RSC, the ATA waits for the specified length of time before retrying. If this interval is 0, the ATA stops trying. This value must be larger than the Reg Retry Intvl value, which cannot be 0.

Default setting: 1200

Reg Retry Random Delay

Random delay range (in seconds) to add to Register Retry Intvl when retrying REGISTER after a failure.

Default setting: 0 (disabled)

Reg Retry Long Random Delay

Random delay range (in seconds) to add to Register Retry Long Intvl when retrying REGISTER after a failure.

Default setting: 0 (disabled)

Reg Retry Intvl Cap

The maximum value to cap the exponential back-off retry delay (which starts at Register Retry Intvl and doubles on every REGISTER retry after a failure). The retry interval is always at Register Retry Intvl seconds after a failure. If this feature is enabled, Reg Retry Random Delay is added on top of the exponential back-off adjusted delay value.

Default setting: 0, which disables the exponential backoff feature.

Response Status Code Handling

Response Status Code Settings

Field

Description

SIT1 RSC

SIP response status code for the appropriate Special Information Tone (SIT). Reorder or Busy tone is played by default for all unsuccessful response status code for SIT 1 RSC through SIT 4 RSC.

Default setting: blank

SIT2 RSC

SIP response status code to INVITE on which to play the SIT2 Tone.

Default setting: blank

SIT3 RSC

SIP response status code to INVITE on which to play the SIT3 Tone.

Default setting: blank

SIT4 RSC

SIP response status code to INVITE on which to play the SIT4 Tone.

Default setting: blank

Try Backup RSC

SIP response code that retries a backup server for the current request.

Default setting: blank

Retry Reg RSC

Interval to wait before the ATA retries registration after failing during the last registration.

Default setting: blank

RTP Parameters

RTP Parameters

Field

Description

RTP Port Min

Minimum port number for RTP transmission and reception.

The RTP Port Min and RTP Port Max parameters should define a range that contains at least 4 even number ports, such as 100 to 106.

Default setting: 16384.

RTP Port Max

Maximum port number for RTP transmission and reception.

Default setting: 16482.

RTP Packet Size

Packet size in seconds, which can range from 0.01 to 0.16. Valid values must be a multiple of 0.01 seconds.

Default setting: 0.030

RTP Tx Packet Size Follows Remote SDP

Enable the Remote pair RTP Packet Size.

Default setting: Yes

Max RTP ICMP Err

Number of successive ICMP errors allowed when transmitting RTP packets to the peer before the ATA terminates the call. If value is set to 0, the ATA ignores the limit on ICMP errors.

Default setting: 0

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 ATA 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 extra BYE packet. Each SR except the last one contains exactly 1 RR (Receiver Report); the last SR carries no RR. The SDES contains CNAME, NAME, and TOOL identifiers. The CNAME is set to <User ID>@<Proxy>, NAME is set to <Display Name> (or Anonymous if user blocks caller ID), and TOOL is set to the Vendor/Hardware-platform-software-version. The NTP timestamp used in the SR is a snapshot of the local time for the ATA, not the time reported by an NTP server. If the ATA receives a RR from the peer, it attempts to compute the round-trip delay and show it as the Call Round Trip Delay value (ms) on the Information page.

Default setting: 0

No UDP Checksum

Select yes if you want the ATA to calculate the UDP header checksum for SIP messages. Otherwise, select no.

Default setting: no

Stats In BYE

Determines whether the ATA includes the P-RTP-Stat header or response in a BYE message. The header contains the RTP statistics of the current call. Select yes or no from the drop-down list.

Default setting: yes

The format of the P-RTP-Stat header is:

P-RTP-State: PS=<packets sent>,OS=<octets sent>,PR=<packets received>,OR=<octets received>,PL=<packets lost>,JI=<jitter in ms>,LA=<delay in ms>,DU=<call duration ins>,EN=<encoder>,DE=<decoder>.

Call Statistics

Specifies whether the phone sends end-of-call statistics within SIP messages when a call terminates or is put on hold.

You can select Yes to enable the phone to send end-of-call statistics in Session Initiation Protocol (SIP) messages (BYE and re-INVITE messages). The phone sends call statistics to the other party of the call when the call terminates or when the call is on hold. The statistics include:

- Real-time Transport Protocol (RTP) packets sent or received

- Total bytes sent or received

- Total number of lost packets

- Delay jitter

- Round-trip delay

- Call duration

The call statistics are sent as headers in SIP BYE messages and SIP BYE response messages (200 OK and re-INVITE during hold). For audio sessions, the headers are RTP-RxStat and RTP-TxStat.

- Rtp-Rxstat -> Data received

- Rtp-Txstat -> Data transmitted

Example of call statistics in a SIP BYE message:

From: xxxx

User-Agent: xxxx

Call-ID: xxxx

Rtp-Rxstat:Dur=13,Pkt=408,Oct=97680,LatePkt=8,LostPkt=0,AvgJit=0,VQMetrics="CCR=0.0017;ICR=0.0000;ICRmx=0.0077;CS=2;

SCS=0;VoRxCodec=PCMU;CID=4;VoPktSizeMs=30;VoPktLost=0;VoPktDis=1;VoOneWayDelayMs=281;maxJitter=12;MOScq=4.21;MOSlq=3.52;network=ethernet;hwType=CP-8865;rtpBitrate=60110;rtcpBitrate=0“

Rtp-Txstat: Dur=13,Pkt=417,Oct=100080,tvqMetrics="TxCodec=PCMU;rtpbitrate=61587;rtcpbitrate=0"
You can also use the Call_Statistics parameter in the phone configuration file to enable this feature.

<Call_Statistics ua="na">Yes</Call_Statistics>
Default: Yes

Options: Yes and No

SDP Payload Types

SDP Payloads

Field

Description

NSE Dynamic Payload

NSE dynamic payload type. The valid range is 96-127.

Default setting: 100

AVT Dynamic Payload

AVT dynamic payload type. The valid range is 96-127.

Default setting: 101

INFOREQ Dynamic Payload

INFOREQ dynamic payload type.

Default setting: blank

G726r32 Dynamic Payload

G726r32 dynamic payload type.

Default setting: 2

G729b Dynamic Payload

G.729b dynamic payload type. The valid range is 96-127.

Default setting: 99

EncapRTP Dynamic Payload

EncapRTP Dynamic Payload type.

Default setting: 112

RTP-Start-Loopback Dynamic Payload

RTP-Start-Loopback Dynamic Payload type.

Default setting: 113

RTP-Start-Loopback Codec

RTP-Start-Loopback Codec. Select one of the following: G711u, G711a, G726-32, G729a.

Default setting: G711u

NSE Codec Name

NSE codec name used in SDP.

Default setting: NSE

AVT Codec Name

AVT codec name used in SDP.

Default setting: telephone-event

G711u Codec Name

G.711u codec name used in SDP.

Default setting: PCMU

G711a Codec Name

G.711a codec name used in SDP.

Default setting: PCMA

G726r32 Codec Name

G.726-32 codec name used in SDP.

Default setting: G726-32

G729a Codec Name

G.729a codec name used in SDP.

Default setting: G729a

G729b Codec Name

G.729b codec name used in SDP.

Default setting: G729ab

EncapRTP Codec Name

EncapRTP codec name used in SDP.

Default setting: encaprtp

NAT Support Parameters

NAT Support Parameters

Field

Description

Handle VIA received.

If you select Yes, the ATA processes the received parameter in the VIA header. The server inserts this value in a response to any one of its requests. If you select No, the parameter is ignored.

Default setting: No

Handle VIA rport.

If you select Yes, the ATA processes the rport parameter in the VIA header. This value is inserted by the server in a response to any one of its requests. If you select No, the parameter is ignored.

Default setting: No

Insert VIA received.

Inserts the received parameter into the VIA header of SIP responses if the received-from IP and VIA sent-by IP values differ.

Select Yes or No from the drop-down menu.

Default setting: No

Insert VIA rport.

Inserts the parameter into the VIA header of SIP responses if the received-from IP and VIA sent-by IP values differ.

Select Yes or No from the drop-down menu.

Default setting: No

Substitute VIA Addr

Lets you use NAT-mapped IP:port values in the VIA header. Select yes or no from the drop-down menu.

Default setting: No

Send Resp To Src Port

Sends responses to the request source port instead of the VIA sent-by port.

Select Yes or No from the drop-down menu.

Default setting: No

STUN Enable

Enables the use of STUN to discover NAT mapping.

Select Yes or No from the drop-down menu.

Default setting: No

STUN Test Enable

If the STUN Enable feature is enabled and a valid STUN server is available, the ATA can perform a NAT-type discovery operation when it powers on. It contacts the configured STUN server, and the result of the discovery is reported in a Warning header in all subsequent REGISTER requests. If the ATA detects symmetric NAT or a symmetric firewall, NAT mapping is disabled.

Default setting: No

STUN Server

IP address or fully-qualified domain name of the STUN server to contact for NAT mapping discovery.

Default setting: blank

EXT IP

External IP address to substitute for the actual IP address of the ATA in all outgoing SIP messages. If 0.0.0.0 is specified, no IP address substitution is performed.

If this parameter is specified, the ATA assumes this IP address when generating SIP messages and SDP. However, the results of STUN and VIA received parameter processing supersede this statically configured value.

This option requires that you have (1) a static IP address from your Internet Service Provider and (2) an edge device with a symmetric NAT mechanism. If the ATA is the edge device, the second requirement is met.

Default setting: blank

EXT RTP Port Min

External port mapping number of the RTP Port Min. number. If this number 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.

Default setting: blank

NAT Keep Alive Intvl

Interval between NAT-mapping keep alive messages.

Default setting: 15

Redirect Keep Alive

Enables or disables NAT Redirect keep alive messages.

Default setting: No

Provisioning

Use the Voice > Provisioning page to configure profiles and parameters to configure the ATA from a remote server.

Enter the settings as described. After making changes, click Submit to save your settings, or clickCancel to redisplay the page with the saved settings.

- Configuration Profile

- Firmware Upgrade

- CA Settings

- General Purpose Parameters

Configuration Profile

Configuration Profile Settings

Field

Description

Provision Enable:

Controls all resync actions independently of firmware upgrade actions. Set to yes to enable remote provisioning.

Default setting: Yes

Resync On Reset:

Triggers a resync after every reboot except for reboots caused by parameter updates and firmware upgrades.

Default setting: Yes

Resync Random Delay:

The maximum value for a random time interval that the ATA waits before making its initial contact with the provisioning server. This delay is effective only on the initial configuration attempt following power-on or reset. The delay is a pseudo-random number between zero and this value.

This parameter is in units of 20 seconds; the default value of 2 represents 40 seconds. This feature is disabled when this parameter is set to zero.

This feature can be used to prevent an overload of the provisioning server when many devices power-on simultaneously.

Default setting: 2 (40 seconds)

Resync At (HHmm):

The time of day when the device tries to resync. The resync is performed each day. Used with the Resync At Random Delay.

Default setting: blank

Resync At Random Delay:

Used with the Resync At (HHmm) setting, this parameter sets a range of possible values for the resync delay. The system randomly chooses a value from this range and waits the specified number of seconds before attempting to resync. This feature is intended to prevent the network jam that would occur if all resynchronizing devices began the resync at the exact same time of day.

Default setting: 600

Resync Periodic:

The time interval between periodic resyncs with the provisioning server. The associated resync timer is active only after the first successful synchronization with the server. Setting this parameter to zero disables periodic resynchronization.

Default setting: 3600

Resync Error Retry Delay:

Resync retry interval (in seconds) applied if there is a resync failure. The ATA has an error retry timer that activates if the previous attempt to sync with the provisioning server fails. The ATA waits to contact the server again until the timer counts down to zero.

This parameter is the value that is initially loaded into the error retry timer. If this parameter is set to zero, the ATA immediately retries to sync with the provisioning server following a failed attempt.

Default setting: 3600

Forced Resync Delay:

Maximum delay (in seconds) that the ATA waits before performing a resync. The ATA does not resync while one of its lines is active. Because a resync can take several seconds, it is desirable to wait until the ATA has been idle for an extended period before resynchronizing. It allows you to make calls in succession without interruption.

The ATA has a timer that begins counting down when all lines become idle. This parameter is the initial value of the counter.

Resync events are delayed until this counter decrements to zero.

Default setting: 14400

Resync From SIP:

Enables a resync to be triggered with a SIP NOTIFY message.

Default setting: yes

Resync After Upgrade Attempt:

Triggers a resync after every firmware upgrade attempt.

Default setting: yes

Resync Trigger 1:

Resync Trigger 2:

Configurable resync trigger conditions. A resync is triggered when the logic equation in these parameters evaluates to TRUE.

Default setting: blank

Resync Fails On FNF:

Determines whether a file-not-found response from the provisioning server constitutes a successful or a failed resync. A failed resync activates the error resync timer.

Default setting: yes

Profile Rule:

This parameter is a profile script that evaluates to the provisioning resync command. The command is a TCP/IP operation and an associated URL. The TCP/IP operation can be TFTP, HTTP, or HTTPS.

If the command is not specified, TFTP is assumed, and the address of the TFTP server is obtained through DHCP option 66. In the URL, either the IP address or the FQDN of the server can be specified. The filename can have macros, such as $MA, which expands to the ATA MAC address.

Default setting: /spa$PSN.cfg

Profile Rule B:

Profile Rule C:

Profile Rule D:

Defines second, third, and fourth resync commands and associated profile URLs. These profile scripts are executed sequentially after the primary Profile Rule resync operation has completed. If a resync is triggered and Profile Rule is blank, Profile Rules B, C, and D are still evaluated and executed.

Default setting: blank

DHCP Option To Use:

DHCP Options, delimited by commas, retrieves firmware and profiles.

Default setting: 66.160.159.150

Transport Protocol:

Transport Protocol retrieves firmware and profiles. If none is selected, TFTP is assumed and the IP address of the TFTP server is obtained from the DHCP server.

Default setting: https

Log Resync Request Msg:

This parameter contains the message that is sent to the Syslog server at the start of a resync attempt.

Default setting: $PN $MAC -- Requesting resync $SCHEME://$SERVIP:$PORT$PATH

Log Resync Success Msg:

Syslog message issued upon successful completion of a resync attempt.

Default setting: $PN $MAC -- Successful resync $SCHEME://$SERVIP:$PORT$PATH

Log Resync Failure Msg:

Syslog message issued after a failed resync attempt.

Default setting: $PN $MAC -- Resync failed: $ERR

Report Rule:

The target URL to which configuration reports are sent. This parameter has the same syntax as the Profile_Rule parameter, and resolves to a TCP/IP command with an associated URL.

A configuration report is generated in response to an authenticated SIP NOTIFY message, with Event: report. The report is an XML file containing the name and value of all the device parameters.

This parameter may optionally contain an encryption key. For example:

[ --key $K ] tftp://ps.callhome.net/$MA/rep.xml.enc

Default setting: blank

Firmware Upgrade

Firmware Upgrade Settings

Field

Description

Upgrade Enable.

Determines if the firmware upgrade operations occur independently of resync actions.

Default setting: yes

Upgrade Error Retry Delay.

The upgrade retry interval (in seconds) applied if there is an upgrade failure. The ATA has a firmware upgrade error timer that activates after a failed firmware upgrade attempt. The timer is initialized with the value in this parameter. The next firmware upgrade attempt occurs when this timer counts down to zero.

Default setting: 3600

Downgrade Rev Limit.

Enforces a lower limit on the acceptable version number during a firmware upgrade or downgrade. The ATA does not complete a firmware upgrade operation unless the firmware version is greater than or equal to this parameter.

Default setting: blank

Upgrade Rule.

This parameter is a firmware upgrade script with the same syntax as Profile_Rule. Defines upgrade conditions and associated firmware URLs.

Default setting: blank

Log Upgrade Request Msg.

Syslog message issued at the start of a firmware upgrade attempt.

Default setting: $PN $MAC -- Requesting upgrade $SCHEME://$SERVIP:$PORT$PATH

Log Upgrade Success Msg.

Syslog message issued after a firmware upgrade attempt completes successfully.

Default setting: $PN $MAC -- Successful upgrade $SCHEME://$SERVIP:$PORT$PATH -- $ERR

Log Upgrade Failure Msg.

Syslog message issued after a failed firmware upgrade attempt.

Default setting: $PN $MAC -- Upgrade failed: $ERR

CA Settings

CA Settings

Field

Description

Custom CA URL

The URL of a file location for a custom Certificate Authority (CA) certificate. Either the IP address or the FQDN of the server can be specified. The file name can have macros, such as $MA, which expands to the ATA MAC address.

Default setting: blank

General Purpose Parameters

General Purpose Settings

Field

Description

GPP A to GPP P

General purpose provisioning parameters. These parameters can be used as variables in provisioning and upgrade rules. They are referenced by prepending the variable name with a ‘$’ character, such as $GPP_A.

Default setting: blank

Regional

Use the Voice > Regional page to localize your system with the appropriate regional settings.

Enter the settings as described. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

- Ring, Cadence, and Tone Scripts

- Call Progress Tones

- Distinctive Ring Patterns

- Distinctive Call Waiting Tone Patterns

- Distinctive Ring/CWT Pattern Names

- Ring and Call Waiting Tone Spec

- Control Timer Values (Sec)

- Vertical Service Activation Codes

- Vertical Service Announcement Codes

- Outbound Call Codec Selection Codes

- Miscellaneous

Ring, Cadence, and Tone Scripts

To define ring and tone patterns, the ATA uses the concept of scripts. In the next sections, you will find information about creating Cadence Scripts (CadScripts), Frequency Scripts (FreqScripts), and Tone Scripts (ToneScripts).

- CadScript

- FreqScript

- ToneScript

CadScript

A mini-script of up to 127 characters that specifies the cadence parameters of a signal.

Syntax: S1[;S2], where:

Si=Di(oni,1/offi,1[,oni,2/offi,2[,oni,3/offi,3[,oni,4/offi,4[,oni,5/offi,5,oni,6/offi,6]]]]]) and is known as a section, oni,j and offi,j are the on/off duration in seconds of a segment and i = 1 or 2, and j = 1 to 6. Di is the total duration of the section in seconds. All durations can have up to three decimal places to provide 1-ms resolution. The wildcard character “*” represents infinite duration. The segments within a section are played in order and repeated until the total duration is played.

Example 1: 60(2/4)

Number of Cadence Sections = 1

Cadence Section 1: Section Length = 60 s

Number of Segments = 1

Segment 1: On=2s, Off=4s

Total Ring Length = 60s

Example 2—Distinctive Ring (short,short,short,long): 60(.2/.2,.2/.2,.2/.2,1/4)

Number of Cadence Sections = 1

Cadence Section 1: Section Length = 60s

Number of Segments = 4

Segment 1: On=0.2s, Off=0.2s

Segment 2: On=0.2s, Off=0.2s

Segment 3: On=0.2s, Off=0.2s

Segment 4: On=1.0s, Off=4.0s

Total Ring Length = 60s

FreqScript

A mini-script of up to 127 characters that specifics the frequency and level parameters of a tone.

Syntax: F1@L1[,F2@L2[,F3@L3[,F4@L4[,F5@L5[,F6@L6]]]]

Where F1–F6 are frequency in Hz (unsigned integers only) and L1–L6 are corresponding levels in dBm (with up to 1 decimal place). White spaces are allowed before and after the comma, but they are not recommended.

Example 1—Call Waiting Tone: 440@-10

Number of Frequencies = 1

Frequency 1 = 440 Hz at –10 dBm

Example 2—Dial Tone: 350@-19,440@-19

Number of Frequencies = 2

Frequency 1 = 350 Hz at –19 dBm

Frequency 2 = 440 Hz at –19 dBm

ToneScript

A mini-script of up to 127 characters that specifies the frequency, level, and cadence parameters of a call progress tone. May contain up to 127 characters.

Syntax: ToneScript;Z1[;Z2].

The section Z1 is similar to the S1 section in a CadScript except that each on/off segment is followed by a frequency components parameter: Z1 = D1(oni,1/offi,1/fi,1[,oni,2/offi,2/fi,2[,oni,3/offi,3/fi,3[,oni,4/offi,4/fi,4[,oni,5/offi,5/fi,5[,oni,6/offi,6/fi,6]]]]]), where fi,j = n1[+n2]+n3[+n4[+n5[+n6]]]]] and 1 < nk < 6 indicates which of the frequency components given in the FreqScript are used in that segment; if more than one frequency component is used in a segment, the components are summed together.

Example 1—Dial tone: 350@-19,440@-19;10(*/0/1+2)

Number of Frequencies = 2

Frequency 1 = 350 Hz at –19 dBm

Frequency 2 = 440 Hz at –19 dBm

Number of Cadence Sections = 1

Cadence Section 1: Section Length = 10 s

Number of Segments = 1

Segment 1: On=forever, with Frequencies 1 and 2

Total Tone Length = 10s

Example 2—Stutter tone: 350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2)

Number of Frequencies = 2

Frequency 1 = 350 Hz at –19 dBm

Frequency 2 = 440 Hz at –19 dBm

Number of Cadence Sections = 2

Cadence Section 1: Section Length = 2s

Number of Segments = 1

Segment 1: On=0.1s, Off=0.1s with Frequencies 1 and 2

Cadence Section 2: Section Length = 10s

Number of Segments = 1

Segment 1: On=forever, with Frequencies 1 and 2

Total Tone Length = 12s

Call Progress Tones

Call Progress Settings

Field

Description

Dial Tone

Prompts you to enter a phone number. Reorder Tone is played automatically when Dial Tone or any of its alternatives times out.

Default setting: 350@-19,440@-19;10(*/0/1+2)

Second Dial Tone

Alternative to the Dial Tone when you dial a three-way call.

Default setting: 420@-19,520@-19;10(*/0/1+2)

Outside Dial Tone

Alternative to the Dial Tone. It prompts you to enter an external phone number, as opposed to an internal extension. A comma character in the dial plan triggers it.

Default setting: 420@-16;10(*/0/1)

Prompt Tone

Prompts you to enter a call forwarding phone number.

Default setting: 520@-19,620@-19;10(*/0/1+2)

Busy Tone

Played when a 486 RSC is received for an outbound call.

Default setting: 480@-19,620@-19;10(.5/.5/1+2)

Reorder Tone

Played when an outbound call has failed, or after the far end hangs up during an established call. Reorder Tone is played automatically when Dial Tone or any of its alternatives times out.

Default setting: 480@-19,620@-19;10(.25/.25/1+2)

Off Hook Warning Tone

Played when the caller has not properly placed the handset on the cradle. Off Hook Warning Tone is played when the Reorder Tone times out.

Default setting: 480@-10,620@0;10(.125/.125/1+2)

Ring Back Tone

Played during an outbound call when the far end is ringing.

Default setting: 440@-19,480@-19;*(2/4/1+2)

Ring Back 2 Tone

Your ATA plays this tone instead of Ring Back Tone if the called party replies with a SIP 182 response without SDP to its outbound INVITE request.

Default setting: the same as Ring Back Tone, except the cadence is 1s on and 1s off.

Default setting: 440@-19,480@-19;*(1/1/1+2)

Confirm Tone

Brief tone to notify you that the last input value has been accepted.

Default setting: 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.

Default setting: 985@-16,1428@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0)

SIT2 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.

Default setting: 914@-16,1371@-16,1777@-16;20(.274/0/1,.274/0/2,.380/0/3,0/4/0)

SIT3 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.

Default setting: 914@-16,1371@-16,1777@-16;20(.380/0/1,.380/0/2,.380/0/3,0/4/0)

SIT4 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.

Default setting: 985@-16,1371@-16,1777@-16;20(.380/0/1,.274/0/2,.380/0/3,0/4/0)

MWI Dial Tone

Played instead of the Dial Tone when there are unheard messages in the caller’s mailbox.

Default setting: 350@-19,440@-19;2(.1/.1/1+2);10(*/0/1+2)

Cfwd Dial Tone

Played when all calls are forwarded.

Default setting: 350@-19,440@-19;2(.2/.2/1+2);10(*/0/1+2)

Holding Tone

Informs the local caller that the far end has placed the call on hold.

Default setting: 600@-19;*(.1/.1/1,.1/.1/1,.1/9.5/1)

Conference Tone

Played to all parties when a three-way conference call is in progress.

Default setting: 350@-19;20(.1/.1/1,.1/9.7/1)

Secure Call Indication Tone

Played when a call has been successfully switched to secure mode. Play it for a short period - less than 30 seconds - and at a reduced level - less than 19 dBm - so it doesn't interfere with the call.

Default setting: 397@-19,507@-19;15(0/2/0,.2/.1/1,.1/2.1/2)

Feature Invocation Tone

Played when a feature is implemented.

Default setting: 350@-16;*(.1/.1/1)

Call Remind Tone

The holding tone is played on the Phone ports during the active call to remind you of the held call.

Default setting: blank

Distinctive Ring Patterns

Distinctive Ring Settings

Field

Description

Ring1 Cadence

Cadence script for distinctive ring 1.

Default setting: 60(2/4)

Ring2 Cadence

Cadence script for distinctive ring 2.

Default setting: 60(.8/.4,.8/4)

Ring3 Cadence

Cadence script for distinctive ring 3.

Default setting: 60(.4/.2,.4/.2,.8/4)

Ring4 Cadence

Cadence script for distinctive ring 4.

Default setting: 60(.3/.2,1/.2,.3/4)

Ring5 Cadence

Cadence script for distinctive ring 5.

Default setting: 1(.5/.5)

Ring6 Cadence

Cadence script for distinctive ring 6.

Default setting: 60(.2/.4,.2/.4,.2/4)

Ring7 Cadence

Cadence script for distinctive ring 7.

Default setting: 60(.4/.2,.4/.2,.4/4)

Ring8 Cadence

Cadence script for distinctive ring 8.

Default setting: 60(0.25/9.75)

Distinctive Call Waiting Tone Patterns

Distinctive Call Waiting Tones

Field

Description

CWT1 Cadence

Cadence script for distinctive CWT 1.

Default setting: *(.3/9.7)

CWT2 Cadence

Cadence script for distinctive CWT 2.

Default setting: 30(.1/.1, .1/9.7)

CWT3 Cadence

Cadence script for distinctive CWT 3.

Default setting: 30(.1/.1, .1/.1, .1/9.7)

CWT4 Cadence

Cadence script for distinctive CWT 4.

Default setting: 30(.1/.1, .3/.1, .1/9.3)

CWT5 Cadence

Cadence script for distinctive CWT 5.

Default setting: 1(.5/.5)

CWT6 Cadence

Cadence script for distinctive CWT 6.

Default setting: 30(.1/.1,.3/.2,.3/9.1)

CWT7 Cadence

Cadence script for distinctive CWT 7.

Default setting: 30(.3/.1,.3/.1,.1/9.1)

CWT8 Cadence

Cadence script for distinctive CWT 8.

Default setting: 2.3(.3/2)

Distinctive Ring/CWT Pattern Names

Distinctive Ring/CWT Patterns

Field

Description

Ring1 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 1 for the inbound call.

Default setting: Bellcore-r1

Ring2 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 2 for the inbound call.

Default setting: Bellcore-r2

Ring3 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 3 for the inbound call.

Default setting: Bellcore-r3

Ring4 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 4 for the inbound call.

Default setting: Bellcore-r4

Ring5 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 5 for the inbound call.

Default setting: Bellcore-r5

Ring6 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 6 for the inbound call.

Default setting: Bellcore-r6

Ring7 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 7 for the inbound call.

Default setting: Bellcore-r7

Ring8 Name

Name in an INVITE’s Alert-Info Header to pick distinctive ring/CWT 8 for the inbound call.

Default setting: Bellcore-r8

Ring and Call Waiting Tone Spec

IMPORTANT: Ring and Call Waiting tones do not work the same way on all phones. When setting ring tones, consider the following recommendations:

- Begin with the default Ring Waveform, Ring Frequency, and Ring Voltage.

- If your ring cadence doesn’t sound right, or your phone doesn’t ring, change the following settings:

- Ring Waveform: Sinusoid

- Ring Frequency: 25

- Ring Voltage: 80

Ring and Call Waiting Tones

Field

Description

Ring Waveform

Waveform for the ringing signal. Choices are Sinusoid or Trapezoid.

Default setting: Trapezoid

Ring Frequency

Frequency of the ringing signal. Valid values are 15–50 (Hz)

Default setting: 20

Ring Voltage

Ringing voltage. Choices are 30–90 (V)

Default setting: 85

CWT Frequency

Frequency script of the call waiting tone. All distinctive CWTs are based on this tone.

Default setting: 440@-10

Synchronized Ring

If this is set to yes, when the ATA is called, all lines ring at the same time (similar to a regular PSTN line). After one line answers, the others stop ringing.

Default setting: no

Control Timer Values (Sec)

Control Timer Values

Field

Description

Hook Flash Timer Min.

Minimum on-hook time before off-hook qualifies as hook flash. Less than this value, and the on-hook event is ignored. Range: 0.1–0.4 seconds.

Default setting: 0.1

Hook Flash Timer.

Max Maximum on-hook time before off-hook qualifies as hook flash. More than this value, and the on-hook event is treated as on hook (no hook-flash event).

Range: 0.4–1.6 seconds.

Default setting: 0.9

Callee On Hook Delay.

Phone must be on-hook for time before the ATA tears down the current inbound call. It does not apply to outbound calls.

Range: 0–255 seconds.

Default setting: 0

Reorder Delay.

Delay after far end hangs up before reorder tone is played. 0 = plays immediately, inf = never plays. Range: 0–255 seconds.

Default setting: 5.

Call Back Expires.

Expiration time in seconds of a call back activation. Range: 0–65535 seconds.

Default setting: 1800

Call Back Retry Intvl.

Call back retry interval in seconds. Range: 0–255 seconds.

Default setting: 30

Call Back Delay.

Delay after receiving the first SIP 18x response before declaring the remote end is ringing. If a busy response is received during this time, the ATA still considers the call as failed and keeps on retrying. Range: 0–65 seconds.

Default setting: 0.5

VMWI Refresh Intvl.

Interval between VMWI refresh to the device. Range: 0–65535 seconds.

Default setting: 0

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.

Default setting: 10

Interdigit Short Timer.

Short timeout between entering digits when dialing. The Interdigit_Short_Timer is used after any one digit, if at least one matching sequence is complete as dialed, but more dialed digits would match other as yet incomplete sequences. Range: 0–64 seconds.

Default setting: 3

CPC Delay.

Delay in seconds after caller hangs up when the ATA starts removing the tip-and-ring voltage to the attached equipment of the called party. The range is 0–255 seconds. This feature is generally used for answer supervision on the caller side to signal to the attached equipment when the call has been connected (remote end has answered) or disconnected (remote end has hung up) This feature should be disabled for the called party (in other words, by using the same polarity for connected and idle state) and the CPC feature should be used instead.

Without CPC enabled, reorder tone is played after a configurable delay. If CPC is enabled, dial tone will be played when tip-to-ring voltage is restored. Resolution is 1 second.

Default setting: 2

CPC Duration.

Duration in seconds for which the tip-to-ring voltage is removed after the caller hangs up. After that, tip-to-ring voltage is restored and the dial tone applies if the attached equipment is still off-hook. CPC is disabled if this value is set to 0. Range: 0 to 1.000 second. Resolution is 0.001 second.

Default setting: 0.5

Vertical Service Activation Codes

Vertical Service Activation Codes are automatically appended to the dial-plan. There is no need to include them in dial-plan, although no harm is done if they are included.

Vertical Service Activation Codes

Field

Description

Call Return Code.

Call Return Code This code calls the last caller.

Default setting: *69

Call Redial Code.

Redials the last number called.

Default setting: *07

Blind Transfer Code.

Begins a blind transfer of the active call to the extension specified after the activation code.

Default setting: *98

Call Back Act Code.

Starts a callback when the last outbound call is not busy.

Default setting: *66

Call Back Deact Code.

Cancels a callback.

Default setting: *86

Call Back Busy Act Code.

Starts a callback when the last outbound call is busy.

Default setting: *05

Cfwd All Act Code.

Forwards all calls to the extension specified after the activation code.

Default setting: *72

Cfwd All Deact Code.

Cancels call forwarding of all calls.

Default setting: *73

Cfwd Busy Act Code.

Forwards busy calls to the extension specified after the activation code.

Default setting: *90

Cfwd Busy Deact Code.

Cancels call forwarding of busy calls.

Default setting: *91

Cfwd No Ans Act Code.

Forwards no-answer calls to the extension specified after the activation code.

Default setting: *92

Cfwd No Ans Deact Code.

Cancels call forwarding of no-answer calls.

Default setting: *93

Cfwd Last Act Code.

Forwards the last inbound or outbound call to the number that you specify after entering the activation code.

Default setting: *63

Cfwd Last Deact Code.

Cancels call forwarding of the last inbound or outbound call.

Default setting: *83

Block Last Act Code.

Blocks the last inbound call.

Default setting: *60

Block Last Deact Code.

Cancels blocking of the last inbound call.

Default setting: *80

Accept Last Act Code.

Accepts the last outbound call. It lets the call ring through when Do Not Disturb or call forwarding of all calls are enabled.

Default setting: *64

Accept Last Deact Code.

Cancels the code to accept the last outbound call.

Default setting: *84

CW Act Code.

Enables call waiting on all calls.

Default setting: *56

CW Deact Code.

Disables call waiting on all calls.

Default setting: *57

CW Per Call Act Code.

Enables call waiting for the next call.

Default setting: *71

CW Per Call Deact Code.

Disables call waiting for the next call.

Default setting: *70

Block CID Act Code.

Blocks caller ID on all outbound calls.

Default setting: *67

Block CID Deact Code.

Removes caller ID blocking on all outbound calls.

Default setting: *68

Block CID Per Call Act Code.

Blocks caller ID on the next outbound call.

Default setting: *81

Block CID Per Call Deact Code.

Removes caller ID blocking on the next inbound call.

Default setting: *82

Block ANC Act Code.

Blocks all anonymous calls.

Default setting: *77

Block ANC Deact Code.

Removes blocking of all anonymous calls.

Default setting: *87

DND Act Code.

Enables the Do Not Disturb feature.

Default setting: *78

DND Deact Code.

Disables the Do Not Disturb feature.

Default setting: *79

CID Act Code.

Enables caller ID generation.

Default setting: *65

CID Deact Code.

Disables caller ID generation.

Default setting: *85

CWCID Act Code.

Enables call waiting, caller ID generation.

Default setting: *25

CWCID Deact Code.

Disables call waiting, caller ID generation.

Default setting: *45

Dist Ring Act Code.

Enables the distinctive ringing feature.

Default setting: *26

Dist Ring Deact Code.

Disables the distinctive ringing feature.

Default setting: *46

Speed Dial Act Code.

Assigns a speed dial number.

Default setting: *74

Paging Code.

Used for paging other clients in the group.

Default setting: *96

Secure All Call Act Code.

Makes all outbound calls secure.

Default setting: *16

Secure No Call Act Code.

Makes all outbound calls not secure.

Default setting: *17

Secure One Call Act Code.

Makes the next outbound call secure. (It is redundant if all outbound calls are secure by default).

Default setting: *18

Secure One Call Deact Code.

Makes the next outbound call not secure. (It is redundant if all outbound calls are not secure by default).

Default setting: *19

Conference Act Code.

If this code is specified, you must enter it before dialing the third party for a conference call. Enter the code for a conference call.

Default setting: blank

Attn-Xfer Act Code.

If the code is specified, you must enter it before dialing the third party for a call transfer. Enter the code for a call transfer.

Default setting: blank

Modem Line Toggle Code.

Toggles the line to a modem. Modem passthrough mode can be triggered only by pre-dialing this code.

Default setting: *99

FAX Line Toggle Code.

Toggles the line to a fax machine.

Default setting: #99

Media Loopback Code.

Use for media loopback.

Default setting: *03

Referral Services Codes.

These codes tell the ATA what to do when you place the active call on hold and is listening to the second dial tone. One or more *codes can be configured into this parameter, such as *98, or *97|*98|*123, and so on. The maximum length is 79 characters. This parameter applies when you place the active call on hold by pressing the hook flash button. Each *code (and the following valid target number according to current dial plan) triggers the ATA to perform a blind transfer to a target number that is prepended by the service *code.

For example, after you dial *98, the ATA plays the Prompt Tone while waiting for you to enter a target number (which is checked according to dial plan as in normal dialing). When a complete number is entered, the ATA sends a blind REFER to the holding party with the Refer-To target equal to *98 target_number. This feature allows the ATA to hand off a call to an application server to perform further processing, such as call park.

The *codes should not conflict with any of the other vertical service codes internally processed by the ATA. You can empty the corresponding *code that you do not want the ATA to process.

Default setting: blank

Feature Dial Services Codes.

These codes let the ATA know what to do when you are listening to the first or second dial tone.

One or more *codes can be configured into this parameter, such as *72, or *72|*74|*67|*82, and so on. The maximum length is 79 characters. This parameter applies when you have a dial tone (first or second dial tone).

After receiving dial tone, you enters the *code and the target number according to current dial plan. For example, after you dial *72, the ATA plays a special tone called a Prompt tone while awaiting you to enter a valid target number. When a complete number is entered, the ATA sends a INVITE to *72 target_number as in a normal call. This feature allows the proxy to process features like call forward (*72) or Block Caller ID (*67).

The *codes should not conflict with any of the other vertical service codes internally processed by the ATA. You can remove a corresponding *code that you do not want to the ATA to process.

You can add a parameter to indicate which tone plays after the *code is entered, such as *72‘c‘|*67‘p‘. Below is a list of allowed tone parameters (note the use of open quotes surrounding the parameter, without spaces).

- 'c' = <Cfwd Dial Tone>

- 'd' = <Dial Tone>

- 'm' = <MWI Dial Tone>

- 'o' = <Outside Dial Tone>

- 'p' = <Prompt Dial Tone>

- 's' = <Second Dial Tone>

- 'x' = No tones are placed, x is any digit not used above.

If no tone parameter is specified, the ATA plays Prompt tone by default.

If the *code is not to be followed by a phone number, such as *73 to cancel call forwarding, do not include this parameter. Instead, add the *code in the dial plan and the ATA send INVITE *73@..... as usual when you dial *73.

Default setting: blank

Vertical Service Announcement Codes

Vertical Service Announcement Codes

Field

Description

Service Annc Base Number

Base number for service announcements.

Default setting: blank

Service Annc Extension Codes

Extension codes for service announcements.

Default setting: blank

Outbound Call Codec Selection Codes

Outbound Call Codec Selection Codes

Field

Description

Prefer G711u Code.

Dial prefix to make G.711u the preferred codec for the call.

Default setting: *017110

Force G711u Code.

Dial prefix to make G.711u the only codec that can be used for the call.

Default setting: *027110

Prefer G711a Code.

Dial prefix to make G.711a the preferred codec for the call.

Default setting: *017111

Force G711a Code.

Dial prefix to make G.711a the only codec that can be used for the call.

Default setting: *027111

Prefer G726r32 Code.

Dial prefix to make G.726r32 the preferred codec for the call.

Default setting: *0172632

Force G726r32 Code.

Dial prefix to make G.726r32 the only codec that can be used for the call.

Default setting: *0272632

Prefer G729a Code.

Dial prefix to make G.729a the preferred codec for the call.

Default setting: *01729

Force G729a Code.

Dial prefix to make G.729a the only codec that can be used for the call.

Default setting: *02729

Miscellaneous

Miscellaneous Settings

Field

Description

FXS Port Impedance:

Sets the electrical impedance of the PHONE port.

Choices are:

- 600

- 900

- 600+2.16uF

- 900+2.16uF

- 220+850||120nF

- 220+820||115nF

- 200+600||100nF

Default setting: 600

FXS Port Input Gain:

Input gain in dB, up to three decimal places. The range is 6.000 to -12.000.

Default setting: -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 FXS Port Output Gain parameter.

Default setting: -3

DTMF Playback Level:

Local DTMF playback level in dBm, up to one decimal place. Range: -30–0.

Default setting: -16.0

DTMF Twist:

To gain difference between the two tone frequency. Range: 0–5.

Default setting: 2

DTMF Playback Length:

Local DTMF playback duration in milliseconds. Range: 0–65 seconds.

Default setting: 0.1

Detect ABCD:

To enable local detection of DTMF ABCD, select Yes. Otherwise, select No. Default setting: Yes

This setting has no effect if DTMF Tx Method is INFO; ABCD is always sent OOB regardless in this setting.

Playback ABCD:

To enable local playback of OOB DTMF ABCD, select Yes. Otherwise, select No. Default setting: Yes

Caller ID Method:

Your choices are:

- Bellcore (N.Amer,China): CID, CIDCW, and VMWI. FSK sent after first ring (same as ETSI FSK sent after first ring) (no polarity reversal or DTAS).

- DTMF (Finland, Sweden): CID only. DTMF sent after polarity reversal (and no DTAS) and before first ring.

- DTMF (Denmark): CID only. DTMF sent before first ring with no polarity reversal and no DTAS.

- ETSI DTMF: CID only. DTMF sent after DTAS (and no polarity reversal) and before first ring.

- ETSI DTMF With PR: CID only. DTMF sent after polarity reversal and DTAS and before first ring.

- ETSI DTMF After Ring: CID only. DTMF sent after first ring (no polarity reversal or DTAS).

- ETSI FSK: CID, CIDCW, and VMWI. FSK sent after DTAS (but no polarity reversal) and before first ring. Waits for ACK from a device after DTAS for CIDCW.

- ETSI FSK With PR (UK): CID, CIDCW, and VMWI. FSK is sent after polarity reversal and DTAS and before first ring. Waits for ACK from a device after DTAS for CIDCW. Polarity reversal is applied only if equipment is on hook.

- DTMF (Denmark) with PR: CID only. DTMF sent after polarity reversal (and no DTAS) and before first ring.

Default setting: Bellcore(N.Amer, China)

FXS Port Power Limit:

The choices are from 1 to 8. Default setting: 3

Caller ID FSK Standard:

The ATA supports bell 202 and v.23 standards for caller ID generation. Default setting: bell 202

Feature Invocation Method:

Select the method you want to use, Default, or Sweden default. Default setting: Default.

Line 1 and Line 2 Settings (PHONE 1 and PHONE 2)

Use the Voice > Line 1 and Voice > Line 2 pages to configure the settings for calls through the PHONE 1 and PHONE 2 ports.

Enter the settings as described. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Note

In a configuration profile, the FXS parameters must include an appropriate numeral for identifying the port receiving the setting.

- General

- Streaming Audio Server (SAS)

- NAT Settings

- Network Settings

- SIP Settings

- Set up a Secure Line

- Call Feature Settings

- E911 Geolocation Configuration

- Proxy and Registration

- Subscriber Information

- Supplementary Service Subscription

- Dial Plan

- FXS Port Polarity Configuration

General

General Settings

Field

Description

Line Enable

To enable this line for service, select yes. Otherwise, select no.

Default setting: yes

Streaming Audio Server (SAS)

Streaming Audio Server Settings

Field

Description

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.

Default setting: no

SAS DLG Refresh Intvl

A non-zero value is the interval in which the streaming audio server sends out session refresh (SIP re-INVITE) messages to determine if the connection is active. If the caller does not respond to the refresh message, the ATA ends this call with a SIP BYE message. The range is 0 to 255 seconds (0 means that the session refresh is disabled).

Default setting: 30

SAS Inbound RTP Sink

This parameter works around devices that do not play inbound RTP if the SAS line declares itself as a send-only device and tells the client not to stream out audio. This parameter is an FQDN or IP address of an RTP sink to be used by the SAS line in the SDP of its 200 response to inbound INVITE from a client. It appears in the c = line and the port number appears in the m = line of the SDP.

If this value is not specified or is equal to 0, then c = 0.0.0.0 and a=sendonly are used in the SDP to tell the SAS client not to send any RTP to this SAS line. If a non-zero value is specified, then a=sendrecv and the SAS client streams audio to the given address.

Special case: If the value is $IP, then the SAS line’s own IP address is used in the c = line and a=sendrecv. In that case, the SAS client streams RTP packets to the SAS line.

Default setting: blank

NAT Settings

NAT Settings

Field

Description

NAT Mapping Enable

To use externally mapped IP addresses and SIP/RTP ports in SIP messages, select yes. Otherwise, select no.

Default setting: no

NAT Keep Alive Enable

To send the configured NAT keep alive message periodically, select yes. Otherwise, select no.

Default setting: no

NAT Keep Alive Msg

Enter the keep alive message sent periodically to maintain the current NAT mapping.

Valid values are: $NOTIFY, $REGISTER, and $OPTIONS.

- $NOTIFY: A NOTIFY message is sent to keep NAT alive.

- $REGISTER: A REGISTER message without contact is sent.

- $OPTIONS: An OPTIONS message is sent.

Default setting: $NOTIFY

NAT Keep Alive Dest

Destination receiving the NAT keep alive messages. If the value is $PROXY, the messages are sent to the current proxy server or outbound proxy server.

Default setting: $PROXY

Network Settings

Network Settings

Field

Description

SIP ToS/DiffServ Value

TOS/DiffServ field value in UDP IP packets carrying a SIP message.

Default setting: 0x68

SIP CoS Value [0-7]

CoS value for SIP messages. Valid values are 0 through 7.

Default setting: 3

RTP ToS/DiffServ Value

ToS/DiffServ field value in UDP IP packets carrying RTP data.

Default setting: 0xb8

RTP CoS Value [0- 7]

CoS value for RTP data. Valid values are 0 through 7.

Default setting: 6

Network Jitter Level

Determines how jitter buffer size is adjusted by the ATA. Jitter buffer size is adjusted dynamically. The minimum jitter buffer size is 30 milliseconds or (10 milliseconds + current RTP frame size), whichever is larger, for all jitter level settings. However, the starting jitter buffer size value is larger for higher jitter levels. This setting controls the rate at which the jitter buffer size is adjusted to reach the minimum. Select the appropriate setting: low, medium, high, very high, or extremely high.

Default setting: high

Jitter Buffer Adjustment

Choose yes to enable or no to disable this feature.

Default setting: yes

SIP Settings

SIP Settings

Field

Description

SIP Transport

Select the protocol for SIP messages:

- UDP

- TCP

- TLS

- AUTO

The TCP choice provides “guaranteed delivery”, which assures that lost packets are retransmitted. TCP also guarantees that the SIP packages are received in the same order that they were sent. As a result, TCP overcomes the main disadvantages of UDP. In addition, for security reasons, most corporate firewalls block UDP ports. With TCP, new ports don't need to be opened or packets dropped for activities such as Internet browsing or e-commerce.

AUTO allows the ATA to select the appropriate protocol automatically, based on the NAPTR records on the DNS server.

SIP Port

Port number of the SIP message listening and transmission port.

Default setting: 5060 for PHONE1 and 5061 for PHONE2

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.

Default setting: No

EXT SIP Port

The external SIP port number.

Default setting: blank

Auth Resync-Reboot

If this feature is enabled, the ATA authenticates the sender when it receives the NOTIFY resync reboot (RFC 2617) message. To use this feature, select Yes. Otherwise, select No.

Default setting: Yes

SIP Proxy-Require

The SIP proxy can support a specific extension or behavior when it sees this header from the user agent. If this field is configured and the proxy does not support it, it responds with the message, unsupported. Enter the appropriate header in the field provided.

Default setting: blank

SIP Remote-Party-ID

To use the Remote-Party-ID header instead of the From header, select Yes. Otherwise, select No.

Default setting: Yes

SIP GUID

This feature limits the registration of SIP accounts. The Global Unique ID is generated for each line for each ATA. When it is enabled, the ATA adds a GUID header in the SIP request. The GUID is generated the first time the unit boots up and stays with the unit through rebooting and even factory reset.

Default setting: No

RTP Log Intvl

The interval for the RTP log.

Default setting: 0

Restrict Source IP.

If configured, the ATA drops all packets sent to its SIP Ports from an untrusted IP address. A source IP address is untrusted if it doesn't match the IP addresses resolved from the configured Proxy (or Outbound Proxy if Use Outbound Proxy is Yes).

Default setting: No

Referor Bye Delay.

The number of seconds to wait before sending a BYE to the referrer to terminate a stale call leg after a call transfer).

Default setting: 4

Refer Target Bye Delay.

The number of seconds to wait before sending a BYE to the refer target to terminate a stale call leg after a call transfer.

Default setting: 0

Referee Bye Delay.

The number of seconds to wait before sending a BYE to the referee to terminate a stale call leg after a call transfer.

Default setting: 0

Refer-To Target Contact.

To contact the refer-to target, select yes. Otherwise, select No.

Default setting: no

Sticky 183.

If this feature is enabled, the ATA 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.

Default setting: No

Auth INVITE.

When enabled, authorization is required for initial incoming INVITE requests from the SIP proxy.

Default setting: No

Reply 182 On Call Waiting.

When enabled, the ATA replies with a SIP182 response to the caller if it is already in a call and the line is off-hook. To use this feature, select Yes.

Default setting: No

Use Anonymous With RPID.

Determines whether the ATA uses “Anonymous” when Remote Party ID is requested in the SIP message.

Default setting: Yes

Use Local Addr In From.

Use the local ATA IP address in the SIP FROM message.

Default setting: No

Broadsoft ALTC.

Set whether the SIP is the Broadsoft ALTC.

Options are: Yes or No.

Default setting: No

Set up a Secure Line

You can configure a line to only accept secure calls. If the line is configured to only accept secure calls, then any calls the line makes will be secure.

Before you begin

- Access the phone adapter administration web page. See Access the Phone Web Interface.

- Enable Secure Call Serv under Supplementary Service Subscription section from Voice > Line(n).

- SIP transport with TLS can be set statically on the phone adapter administration web page or automatically with information in the DNS NAPTR records. If the SIP transport parameter is set for the ATA line as TLS, it only allows SRTP. If the SIP transport parameter is set to AUTO, the phone adapter performs a DNS query to get the transport method.

Procedure

Step 1

Select Voice > Line(n) , where n is the line number that represents PHONE 1 or PHONE 2.

Step 2

In the section Call Feature Settings , set the paramter Secure Call Option as described in Call Feature Settings.

Step 3

Click Submit.

Call Feature Settings

Call Feature Parameters

Field

Description

Blind Attn-Xfer Enable

Enables the ATA to perform an attended transfer operation by ending the active call leg and performing a blind transfer of the other call leg. If this feature is disabled, the ATA performs an attended transfer operation by referring the other call leg to the active call leg, while maintaining both call legs. To use this feature, select yes. Otherwise, select no.

Default setting: no

MOH Server

User ID or URL of the auto-answering streaming audio server. When only a user ID is specified, the current or outbound proxy is contacted. Music-on-hold is disabled if the MOH Server is not specified.

Default setting: blank

Xfer When Hangup Conf

Makes the ATA perform a transfer when a conference call has ended. Select yes or no from the drop-down menu.

Default setting: yes

Conference Bridge URL

This feature supports external conference bridging for n-way conference calls (n>2), instead of mixing audio locally. To use this feature, set this parameter to that of the server's name. For example: conf@mysefver.com:12345 or conf (which uses the Proxy value as the domain).

Default setting: blank

Conference Bridge Ports

Select the maximum number of conference call participants. The range is 3 to 10.

Default setting: 3

Enable IP Dialing.

Enable or disable IP dialing. If IP dialing is enabled, you can dial [userid@] a.b.c.d[:port], where ‘@’, ‘.’, and ‘:’ are dialed by entering *, user-id must be numeric and a, b, c, d must be between 0 and 255; the port must be larger than 255. If port is not given, 5060 is used. Port and User-Id are optional. If the user-id portion matches a pattern in the dial plan, then it is interpreted as a regular phone number according to the dial plan. The INVITE message, however, is still sent to the outbound proxy if it is enabled.

Default setting: no

Emergency Number

Comma separated list of emergency number patterns. If outbound call matches one of the patterns, the ATA disables hook flash event handling. The condition is restored to normal after the call ends. Blank signifies that there is no emergency number. Maximum number length is 63 characters.

Default setting: blank

Mailbox ID

Enter the ID number of the mailbox for this line.

Default setting: blank

Feature Key Sync

Allows the phone to synchronize with the call server. If Do Not Disturb or Call Forwarding settings are changed on the phone, the changes are also made on the server. If changes are made on the server, they are propagated to the phone.

Default setting: no

Secure Call Option

Configures a line to only accept secure calls. Select any of the options:

Optional: Retains the current secure call option for the phone adapter.

Strict: Allows SRTP only when SIP transport is set to TLS and if the ATA receives an unsecure call, the call fails. Allows RTP only when SIP transport is UDP/TCP and if the ATA receives an unsecure call, the call fails.

Default setting: Optional

E911 Geolocation Configuration

E911 Geolocation Configuration

Field

Description

Company UUID

The Universally Unique Identifier (UUID) assigned to the customer by the emergency call services provider.

For example:

19c8168c-a366-44b5-853c-960fcaa19592

Allowed values: Maximum identifier length is 128 characters.

Default setting: Blank

Primary Request URL

URL of the primary location server that provides the emergency call services.

The location server returns an HELD response to the phone with the requested location URI that is tied to the user phone IP address.

This parameter must be in the form of a valid HTTP or HTTPS URL.

Allowed values: A valid URL not exceeding 255 characters.

Default setting: Blank

Secondary Request URL

URL of the backup server to obtain the user's phone location.

If the primary request URL fails, ATA tries to send the secondary request URL to the emergency call services provider.

This parameter must be in the form of a valid HTTP or HTTPS URL.

Allowed values: A valid URL not exceeding 255 characters.

Default setting: Blank

Proxy and Registration

Proxy and Registration Parameters

Field

Description

Proxy

SIP proxy server for all outbound requests.

Default setting: blank

Outbound Proxy

SIP Outbound Proxy Server where all outbound requests are sent as the first hop.

Default setting: blank

Use Outbound Proxy

Enables the use of an Outbound Proxy. If set to no, the Outbound Proxy and Use OB Proxy in Dialog parameters are ignored.

Default setting: no

Use OB Proxy In Dialog

Whether to force SIP requests to be sent to the outbound proxy within a dialog. Ignored if the parameter Use Outbound Proxy is no, or the Outbound Proxy parameter is empty.

Default setting: yes

Register

Enable periodic registration with the Proxy parameter. This parameter is ignored if Proxy is not specified.

Default setting: yes

Make Call Without Reg

Allow making outbound calls without successful (dynamic) registration by the unit. If No, dial tone will not play unless registration is successful.

Default setting: no

Register Expires

Expires value in sec in a REGISTER request. The ATA will periodically renew registration shortly before the current registration expired. This parameter is ignored if the Register parameter is no. Range: 0 – (231 – 1) sec.

Default setting: 3600

Ans Call Without Reg

Allow answering inbound calls without successful (dynamic) registration by the unit.

Default setting: no

Use DNS SRV

Whether to use DNS SRV lookup for Proxy and Outbound Proxy.

Default setting: no

DNS SRV Auto Prefix

If enabled, the ATA will automatically prepend the Proxy or Outbound Proxy name with _sip._udp when performing a DNS SRV lookup on that name.

Default setting: no

Proxy Fallback Intvl

After failing over to a lower priority server, the ATA waits for the specified Proxy Fallback Interval, in seconds, before retrying the highest priority proxy (or outbound proxy) servers. This parameter is useful only if the primary and backup proxy server list is provided to the ATA via DNS SRV record lookup on the server name.

Using multiple DNS A records per server name does not allow the notion of priority, so all hosts will be considered at the same priority and the ATA will not attempt to fall back after a failover.

If the value is 0, the SIP proxy fallback feature is disabled.

Range: 0 –65535 sec

Default setting: 3600

Proxy Redundancy Method

The method that the ATA uses to create a list of proxies returned in the DNS SRV records. If you select Normal, the list will contain proxiesranked by weight and priority. If you select Based on SRV port, the ATA also inspects the port number based on 1st proxy’s port.

Default setting: Normal

Mailbox Subscribe URL

The URL or IP address of the voicemail server.

Default setting: blank

Mailbox Subscribe Expires

Sets subscription interval for voicemail message waiting indication. When this time period expires, the ATA sends another subscribe message to the voice mail server.

Default setting: 2147483647

Auto Register When Failover

Controls the fallback duration.

- no: the fallback happens immediately and automatically. If the Proxy Fallback Intvl is exceeded, all the new SIP messages go to the primary proxy.

- yes: the fallback happens only when current registration expires, which means only a REGISTER message can trigger fallback.

For example, when the value for Register Expires is 3600 seconds and Proxy Fallback Intvl is 600 seconds, the fallback is triggered 3600 seconds later and not 600 seconds later. When the value for Register Expires is 600 seconds and Proxy Fallback Intvl is 1000 seconds, the fallback is triggered at 1200 seconds. After successfully registering back to primary server, all the SIP messages go to primary server.

Default setting: yes

Subscriber Information

Subscriber Information Parameters

Field

Description

Display Name

Display name for caller ID.

Default setting: blank

User ID

User ID for this line.

Default setting: blank

Password

Password for this line.

Default setting: blank

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.

Default setting: no

Auth ID

Authentication ID for SIP authentication.

Default setting: blank

Resident Online Number

This setting allows you to associate a "local" telephone number with this line using a valid Skype Online Number from Skype. Calls made to that number will ring your phone. Enter the number without spaces or special characters.

Default setting: blank

SIP URI

The parameter by which the user agent will identify itself for this line. If this field is blank, the actual URI used in the SIP signaling should be automatically formed as: sip:UserName@Domain

Where UserName is the username given for this line in the User ID, and Domain is the domain given for this profile in the User Agent Domain.

If the User Agent Domain is an empty string, then the IP address of the phone should be used for the domain.

If the URI field is not empty, but if a SIP or SIPS URI that contains no @ character, then the actual URI used in the SIP signaling should be automatically formed by appending this parameter with an @ character followed by the IP address of the device.

Supplementary Service Subscription

The ATA 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. A supplementary service should be disabled if a) the user has not subscribed for it, or b) the Service Provider intends to support similar service using other means than relying on the ATA.

Supplementary Service Subscription Settings

Field

Description

Call Waiting Serv

Enable Call Waiting Service.

Default setting: yes

Block CID Serv

Enable Block Caller ID Service.

Default setting: yes

Block ANC Serv

Enable Block Anonymous Calls Service

Default setting: yes

Dist Ring Serv

Enable Distinctive Ringing Service

Default setting: yes

Cfwd All Serv

Enable Call Forward All Service

Default setting: yes

Cfwd Busy Serv

Enable Call Forward Busy Service

Default setting: yes

Cfwd No Ans Serv

Enable Call Forward No Answer Service

Default setting: yes

Cfwd Sel Serv

Enable Call Forward Selective Service. Configure this service in the Selective Call Forward Settings section.

Default setting: yes

Cfwd Last Serv

Enable Forward Last Call Service

Default setting: yes

Block Last Serv

Enable Block Last Call Service

Default setting: yes

Accept Last Serv

Enable Accept Last Call Service

Default setting: yes

DND Serv

Enable Do Not Disturb Service

Default setting: yes

CID–Serv

Enable Caller ID Service

Default setting: yes

CWCID Serv

Enable Call Waiting Caller ID Service

Default setting: yes

Call Return Serv

Enable Call Return Service

Default setting: yes

Call Redial Serv

Enable Call Redial Service.

Default setting: yes

Call Back Serv

Enable Call Back Service.

Default setting: yes

Three Way Call Serv

Enable Three Way Calling Service. Three Way Calling is required for Three Way Conference and Attended Transfer.

Default setting: yes

Three Way Conf Serv

Enable Three Way Conference Service. Three Way Conference is required for Attended Transfer.

Default setting: yes

Attn Transfer Serv

Enable Attended Call Transfer Service. Three Way Conference is required for Attended Transfer.

Default setting: yes

Unattn Transfer Serv

Enable Unattended (Blind) Call Transfer Service.

Default setting: yes

MWI Serv

Enable MWI Service. MWI is available only if a Voice Mail Service is set-up in the deployment.

Default setting: yes

VMWI Serv

Enable VMWI Service (FSK)

Default setting: yes

Speed Dial Serv

Enable Speed Dial Service.

Default setting: yes

Secure Call Serv

Secure Call Service. If this feature is enabled, a user can make a secure call by entering an activation code (*18 by default) before dialing the target number. Then audio traffic in both directions is encrypted for the duration of the call.

Default setting: yes

Star codes are set in Vertical Service Activation Codes. To enable secure calling by default, without requiring a star code, set the user’s Secure Call Setting to yes. See User 1 and User 2.

Referral Serv

Enable Referral Service. See the Referral Services Codes parameter in Vertical Service Activation Codes for more information.

Default setting: yes

Feature Dial Serv

Enable Feature Dial Service. See the Feature Dial Services Codes parameter in Vertical Service Activation Codes for more information.

Default setting: yes

Service Announcement Serv

Enable Service Announcement Service.

Default setting: no

Reuse CID Number As Name

Use the Caller ID number as the caller name.

Default settings: yes

CONFCID Serv

Enable Caller ID during conference call.

Default settings: yes

- Audio Configuration

Audio Configuration

Audio Configuration Settings

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

- G726-32

- G729a

Default setting: G711u.

Second Preferred Codec

If the first codec fails, then second preferred codec is tried.

Default setting: blank

Third Preferred Codec

If the second codec fails, then third preferred codec is tried.

Default setting: blank

Use Pref Codec Only

To use only the preferred codec for all calls, select yes. (The call fails if the far end does not support this codec.) Otherwise, select no.

Default setting: no

Codec Negotiation

When set to Default, the Cisco IP phone responds to an Invite with a 200 OK response advertising the preferred codec only. When set to List All, the Cisco IP phone responds listing all the codecs that the phone supports.

Default setting: Default

G729a Enable

To enable the use of the G.729a codec at 8 kbps, select yes. Otherwise, select no.

Default setting: yes

Silence Supp Enable

To enable silence suppression so that silent audio frames are not transmitted, select yes. Otherwise, select no.

Default setting: no

G726-32 Enable

To enable the use of the G.726 codec at 32 kbps, select yes. Otherwise, select no.

Default setting: yes

Silence Threshold

Select the appropriate setting for the threshold: high, medium, or low.

Default setting: medium

FAX V21 Detect Enable

To enable detection of V21 fax tones, select yes. Otherwise, select no.

Default setting: yes

Echo Canc Enable

To enable the use of the echo canceller, select yes. Otherwise, select no.

Default setting: yes

FAX CNG Detect Enable

To enable detection of the fax Calling Tone (CNG), select yes. Otherwise, select no.

Default setting: yes

FAX Passthru Codec

Select the codec for fax passthrough, G711u or G711a.

Default setting: G711u

FAX Codec Symmetric

To force the ATA to use a symmetric codec during fax passthrough, select yes. Otherwise, select no.

Default setting: yes

DTMF Process INFO

To use the DTMF process info feature, select yes. Otherwise, select no.

Default setting: yes

FAX Passthru Method

Select the fax passthrough method: None, NSE, or ReINVITE.

Default setting: NSE

DTMF Process AVT

To use the DTMF process AVT feature, select yes. Otherwise, select no.

Default setting: yes

FAX Process NSE

To use the fax process NSE feature, select yes. Otherwise, select no.

Default setting: yes

DTMF Tx Method

Select the method to transmit DTMF signals to the far end: InBand, AVT, INFO, or Auto. InBand sends DTMF by using the audio path. AVT sends DTMF as AVT events. INFO uses the SIP INFO method. Auto uses InBand or AVT based on the outcome of codec negotiation.

Default setting: Auto

FAX Disable ECAN

If enabled, this feature automatically disables the echo canceller when a fax tone is detected. To use this feature, select yes. Otherwise, select no.

Default setting: no

DTMF Tx Mode

DTMF Detection Tx Mode is available for SIP information and AVT.

Options are: Strict or Normal.

Default setting: Strict for which the following are true:

- • A DTMF digit requires an extra hold time after detection.

- • The DTMF level threshold is raised to -20 dBm.

The minimum and maximum duration thresholds are:

- strict mode for AVT and SIP: the value set in DTMF Tx Strict Hold Off Time

- normal mode for AVT: 40 ms

- normal mode for SIP: 50 ms

DTMF Tx Strict Hold Off Time

This parameter is in effect only when DTMF Tx Mode is set to strict, and when DTMF Tx Method is not set to inband; that is, either AVT or INFO. The value can be set as low as 40 ms. There is no maximum limit. A larger value will reduce the chance of talk-off (beeping) during conversation, at the expense of reduced performance of DTMF detection, which is needed for interactive voice response systems (IVR).

Default setting: 70 ms

FAX Enable T38

To enable the use of ITU-T T.38 standard for FAX Relay, select yes. Otherwise select no.

- no: The ATA can parse only one "m=" line of the SDP packet.

If the ATA receives multiple "m=" lines contained in the SDP packet from a provider, an outbound FAX failure might occur. This issue typically occurs when the first "m=" line specifies an invalid port number "0" while the second "m=" line specifies a valid port.

To avoid this issue, set FAX Enable T38 to yes and FAX Passthru Method to ReINVITE.

In the aforementioned situation, the ATA can parse the second "m=" line successfully.

- yes: The ATA can parse the first two "m=" lines of the SDP packet. It ignores the other "m=" lines.

Default setting: no

Hook Flash Tx Method

Select the method for signaling hook flash events: None, AVT, or INFO. None does not signal hook flash events. AVT uses RFC2833 AVT (event = 16) INFO uses SIP INFO with the single line signal=hf in the message body. The MIME type for this message body is taken from the Hook Flash MIME Type setting.

Default setting: None

FAX T38 Redundancy

Select the appropriate number to indicate the number of previous packet payloads to repeat with each packet. Choose 0 for no payload redundancy. The higher the number, the larger the packet size and the more bandwidth consumed.

Default setting: 1

FAX T38 ECM Enable

Select yes to enable T.38 Error Correction Mode. Otherwise select no.

Default setting: yes

FAX Tone Detect Mode

This parameter has three possible values:

- caller or callee: The ATA will detect FAX tone whether it is callee or caller

- caller only: The ATA will detect FAX tone only if it is the caller

- callee only: The ATA will detect FAX tone only if it is the callee

Default setting: caller or callee.

Symmetric RTP

Enable symmetric RTP operation. If enabled, the ATA sends RTP packets to the source address and port of the last received valid inbound RTP packet. If disabled (or before the first RTP packet arrives) the ATA sends RTP to the destination as indicated in the inbound SDP.

Default setting: no

Fax T38 Return to Voice

When this feature is enabled, upon completion of the fax image transfer, the connection remains established and reverts to a voice call using the previously designated codec. Select yes to enable this feature, or select no to disable it.

Default setting: no

Modem Line

Enable an alternate method to make the modem call without Modem Line Toggle Code pre-dialing.

Default setting: no

RTP to Proxy in Remote Hold

Enable to send RTP to proxy when line is held by remote side.

Default setting: no

Dial Plan

The default dial plan script for the line is as follows:

(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxx|xxxxxxxxxxxx.)

Each parameter is separated by a semi-colon (;)

Example 1:

*1xxxxxxxxxx<:@fwdnat.pulver.com:5082;uid=jsmith;pwd=xy z

Example 2:

*1xxxxxxxxxx<:@fwd.pulver.com;nat;uid=jsmith;pwd=xyz

The syntax for a dial plan expression is described in the table below.

Dial Plan Settings

Dial Plan Entry

Functionality

*xx

Allow arbitrary 2 digit star code

[3469]11

Allow x11 sequences

0

Operator

00

Int’l Operator

[2-9]xxxxxx

US local number

1xxx[2-9]xxxxxx

US 1 + 10-digit long distance number

xxxxxxxxxxxx.

Everything else

FXS Port Polarity Configuration

FXS Port Polarity Settings

Field

Description

Idle Polarity

Polarity before a call is connected: Forward or Reverse.

Default setting: Forward

Caller Conn Polarity

Polarity after an outbound call is connected: Forward or Reverse.

Default setting: Forward.

Callee Conn Polarity

Polarity after an inbound call is connected: Forward or Reverse.

Default setting: Forward

User 1 and User 2

Use the Voice > User 1 and Voice > User2 pages to set the user preferences for the calls through the PHONE 1 and PHONE 2 ports.

Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

- Call Forward Settings

- Selective Call Forward Settings

- Speed Dial Settings

- Supplementary Service Settings

- Distinctive Ring Settings

- Ring Settings

Call Forward Settings

Call Forward Settings

Field

Description

Cfwd All Dest

Forward number for Call Forward All Service.

Default setting: blank

Cfwd Busy Dest

Forward number for Call Forward Busy Service. Same as Cfwd All Dest.

Default setting: blank

Cfwd No Ans Dest

Forward number for Call Forward No Answer Service. Same as Cfwd All Dest.

Default setting: blank

Cfwd No Ans Delay

Delay in sec before Call Forward No Answer triggers.

Default setting: 20

Selective Call Forward Settings

Selective Call Forward Settings

Field

Description

Cfwd Sel1-8 Caller

Caller number pattern to trigger Call Forward Selective service. When the caller’s phone number matches the entry, the call is forwarded to the corresponding Cfwd Selective Destination (Cfwd Sel1-8 Dest).

- • Use ? to match any single digit.

- • Use * to match any number of digits.

Example: 1408*, 1512???1234

In the above example, a call is forwarded to the corresponding destination if the caller ID either starts with 1408 or is an 11-digit numbering starting with 1512 and ending with 1234.

Default setting: blank

Cfwd Sel1-8 Dest

The destination for the corresponding Call Forward Selective caller pattern (Cfwd Sel1-8 Caller).

Default setting: blank

Cfwd Last Caller

The number of the last caller; this caller is actively forwarded to the Cfwd Last Dest via the Call Forward Last service. For more information, see Vertical Service Activation Codes.

Default setting: blank

Cfwd Last Dest

The destination for the Cfwd Last Caller.

Block Last Caller

The number of the last caller; this caller is blocked via the Block Last Caller Service. For more information, see Vertical Service Activation Codes.

Default setting: blank

Accept Last Caller

The number of the last caller; this caller is accepted via the Accept Last Caller Service. For more information, see Vertical Service Activation Codes.

Default setting: blank

Speed Dial Settings

Speed Dial Settings

Field

Description

Speed Dial 2-9

Target phone number (or URL) assigned to speed dial 2, 3, 4, 5, 6, 7, 8, or 9.

Default setting: blank

Supplementary Service Settings

Supplementary Service Settings

Field

Description

CW Setting

Call Waiting on/off for all calls.

Default setting: yes

Block CID

Setting Block Caller ID on/off for all calls.

Default setting: no

Block ANC

Setting Block Anonymous Calls on or off.

Default setting: no

DND

Setting DND on or off.

Default setting: no

CID Setting

Caller ID Generation on or off.

Default setting: yes

CWCID Setting

Call Waiting Caller ID Generation on or off.

Default setting: yes

Dist Ring

Setting Distinctive Ring on or off.

Default setting: yes

Secure Call Setting

If yes, all outbound calls are secure calls by default, without requiring the user to dial a star code first.

Default setting: no

- If Secure Call Setting is set to yes, all outbound calls are secure. However, a user can disable security for a call by dialing *19 before dialing the target number.

- If Secure Call Setting is set to No, the user can make a secure outbound call by dialing *18 before dialing the target number.

- A user cannot force inbound calls to be secure or not secure; that depends on whether the caller has security enabled or not.

Note

This setting is applicable only if Secure Call Serv is set to yes on the line interface. See Line 1 and Line 2 Settings (PHONE 1 and PHONE 2).

Message Waiting

Setting this value to yes can activate stutter tone and VMWI signal. This parameter is stored in long term memory and will survive after reboot or power cycle.

Default setting: no

Accept Media Loopback Request

Controls how to handle incoming requests for loopback operation.

- never—Never accepts loopback calls; replies 486 to the caller.

- automatic—Automatically accepts the call without ringing.

- manual—Rings the phone first, and the call must be picked up manually before loopback starts.

Default setting: Automatic

Media Loopback Mode

The loopback mode to assume locally when making call to request media loopback. Choices are: Source and Mirror.

Default setting: source

Note

If the ATA answers the call, the mode is determined by the caller.

Media Loopback Type

The loopback type to use when making call to request media loopback operation. Choices are Media and Packet.

Default setting: media

Note that if the ATA answers the call, then the loopback type is determined by the caller (the ATA always picks the first loopback type in the offer if it contains multiple type)

CONFCID Setting

Enables or disables the CONFCID.

Default setting: yes

Distinctive Ring Settings

Distinctive Ring Parameters

Field

Description

Ring1 - 8 Caller

Caller number pattern to play Distinctive Ring/CWT 1, 2, 3, 4, 5, 6, 7, or 8. Caller number patterns are matched from Ring 1 to Ring 8. The first match (not the closest match) will be used for alerting the subscriber. The distinctive rings are set on the Regional page. See Regional.

Default setting: blank

Ring Settings

Ring Parameters

Field

Description

Default Ring

Default ringing pattern, 1–8, for all callers.

Default setting: 1

Default CWT

Default CWT pattern, 1–8, for all callers.

Default setting: 1

Hold Reminder Ring

Ring pattern for reminder of a holding call when the phone is on-hook.

Default setting: 8

Call Back Ring

Ring pattern for call back notification.

Default setting: 7

Cfwd Ring Splash Len

Duration of ring splash when a call is forwarded (0 – 10.0s)

Default setting: 0

Cblk Ring Splash Len

Duration of ring splash when a call is blocked (0 – 10.0s)

Default setting: 0

VMWI Ring Policy

The parameter controls when a ring splash is played when a the VM server sends a SIP NOTIFY message to the ATA indicating the status of the subscriber’s mail box. Three settings are available.

Default setting: New VM Available

- New VM Available—Ring as long as there new voicemail messages.

- New VM Becomes Available—Ring at the point when the first new voicemail message is received.

- New VM Arrives—Ring when the number of new voicemail messages increases.

VMWI Ring Splash Len

Duration of ring splash when new messages arrive before the VMWI signal is applied (0 – 10.0s)

Default setting: 0

Ring On No New VM

If enabled, the ATA plays a ring splash when the voicemail server sends SIP NOTIFY message to the ATA indicating that there are no more unread voice mails. Some equipment requires a short ring to precede the FSK signal to turn off VMWI lamp.

Default setting: no

Chapter 6. Administration Settings

- Management

- Log

- Factory Defaults

- Firmware Upgrade

- Configuration Management

- Reboot

Management

Use the Management pages to manage web access to the ATA web page and to enable protocols for remote configuration and network management.

- Web Access Management

- TR-069

- SNMP

- User List (Password Management)

- Bonjour

- Reset Button

- SSH

Web Access Management

Use the Administration > Management > Web Access Management page to configure the settings for access to the administration of the ATA.

- Cisco ATA 191 Web Access Fields

- Cisco ATA 192 Web Access Fields

- Cisco ATA 192 Remote Access Fields

Cisco ATA 191 Web Access Fields

Access to the Cisco ATA 191 web page is enabled by default. Admin Access allows you to manage the configuration from a computer in your office network, and Web Utility Access allows you to connect from a computer on a different subnet or on the Internet.

To access the ATA web page, launch a web browser and enter the URL in the address bar. The URL must include the specified protocol, the WAN IP address of the ATA, and the specified port number. For example, with the HTTPS protocol, a WAN IP address of 203.0.113.50, and port 443, you would enter: https://203.0.113.50:443

Cisco ATA 191 Web Access Settings

Field

Description

Admin Access

This feature controls access to the ATA web page from devices that are connected via the ETHERNET (LAN) port.

Click Enabled to enable this feature, or click Disabled to disable it.

The default setting is Enabled. If you administer and configure the ATA from a computer that is connected to the LAN, this feature must be enabled.

Web Utility Access

Select the protocol to use for access to the ATA web page from a device on the WAN. Choose HTTP, HTTPS, or both entries. For secure Internet access, select HTTPS. The default value is HTTPS.

Remote Management Port

Enter the port number to use for access to the ATA web page from a device on the WAN side of the ATA. The default port number is 443 for HTTPS, 80 for HTTP.

Include the specified port when you enter the address in your web browser. For example, with the HTTPS protocol, a WAN IP address of 203.0.113.50, and the default Remote Management Port of 443, you would enter: https://203.0.113.50:443

Cisco ATA 192 Web Access Fields

Access to the Cisco ATA 192 web page is enabled by default. Admin Access allows you to manage the configuration from a computer in your office network, and Web Utility Access allows you to connect from a computer on a different subnet or on the Internet.

Cisco ATA 192 Web Access Management Settings

Field

Description

Admin Access

This feature controls access to the ATA web page from devices that are connected via the ETHERNET (LAN) port.

Click Enabled to enable this feature, or click Disabled to disable it.

The default setting is Enabled. If you administer and configure the ATA from a computer that is connected to the LAN, this feature must be enabled.

Web Utility Access

Select the protocol to use for access to the ATA web page from a device on the WAN. Choose HTTP and/or HTTPS. For secure Internet access, select HTTPS. The default value is HTTPS.

Cisco ATA 192 Remote Access Fields

In addition to the ATA web page access, the Cisco ATA 192 provides more features of Remote Management.

Remote Access Settings

Field

Description

Remote Management

Allows access to the ATA web page from a device that is on the WAN side of the ATA. For example, you could connect from another subnet in your office or from your home computer.

Click Enabled to enable this feature, or click Disabled to disable it.

The default setting is Disabled. The other fields in this section of the page are available only if you enable this feature. If you attempt to enable this feature while using the default administrator login credentials, you will be prompted to change the credentials. Click OK to acknowledge the warning message. Use the Administration > Management > User List page to change the administrator password. For more information, see User List (Password Management).

Web Utility Access

Select the protocol to use for access to the ATA web page from a device on the WAN side of the ATA. Choose HTTP and/or HTTPS.

The default value is HTTPS.

Include the specified protocol when you enter the address in your web browser. For example, with the HTTPS protocol, a WAN IP address of 203.0.113.50, and the default Remote Management Port of 443, you would enter: https://203.0.113.50:443

Remote Upgrade

If you enabled Remote Management, choose whether or not to allow firmware upgrades from a device on the WAN side of the ATA. Click Enabled to enable this feature, or click Disabled to disable it. The default value is Disabled.

You can change this setting only when your computer is connected to the configuration utility from the LAN.

Allowed Remote IP Address

You can use this feature to limit access to the ATA web page based on the IP address of a device. Choose Any IP Address to allow access from any external IP address. To specify an external IP address or range of IP addresses, select the second radio button and then enter the desired IP address or range. The default setting is Any IP Address.

Remote Management Port

Enter the port number to use for access to the ATA web page from a device on the WAN side of the ATA. The default port number is 443 for HTTPS, 80 for HTTP.

Include the specified port when you enter the address in your web browser. For example, with the HTTPS protocol, a WAN IP address of 203.0.113.50, and the default Remote Management Port of 443, you would enter: https://203.0.113.50:443

TR-069

Use the Administration > Management > TR-069 page to configure communication with an Auto-Configuration Server (ACS) via TR-069 CPE WAN Management Protocol (CWMP). TR-069 (Technical Report 069) provides a common platform to manage all voice devices and other customer-premises equipment (CPE) in large-scale deployments. It provides the communication between the CPE and the ACS.

Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

TR-069 Settings

Field

Description

Status

Click Enabled to enable remote provisioning, or click Disabled to disable this feature. The default setting is Disabled.

ACS URL

The URL for the ACS. The format should be http(s)://xxx.xxx.xxx.xxx:port or xxx.xxx.xxx.xxx:port. The xxx.xxx.xxx.xxx is the domain name or IP address of the ACS server.

Both the IP address and the port number are required.

ACS Username

The username for the ACS. The default username is the Organization Unit Identifier (OUI). This value is required and must match the username configured on the ACS.

ACS Password

The password for the ACS. This value is required and must match the password configured on the ACS.

Connection Request Port

The port to use for connection requests

Connection Request Username

The username for connection requests. This value must match the Connection Request Username configured on the ACS.

Connection Request Password

The password for connection requests. This value must match the Connection Request Password configured on the ACS.

Periodic Inform Interval

If Periodic Inform is enabled, the duration, in seconds, between CPE attempts to connect to the ACS. The default value is 86400 seconds.

Periodic Inform Enable

Click Enabled to enable CPE connection requests to the ACS, or click Disabled to disable this feature.

Request Download

If applied, ACS may call the Download RPC after it receives the request from the ATA.

SNMP

Use the Administration > Management > SNMP page to set up Simple Network Management Protocol (SNMP) for the ATA.

SNMP is a network protocol that allows network administrators to manage, monitor, and receive notifications of critical events as they occur on the network. The ATA supports SNMPv2 and SNMPv3.

It acts as an SNMP agent that replies to SNMP commands from SNMP Network Management Systems. It supports the standard SNMP get, next, and set commands. It also generates SNMP traps to notify the SNMP manager when configured alarm conditions occur. Examples include reboots, power cycles, and INTERNET (WAN) events.

Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

- SNMP Settings

- SNMPv3 Settings

- Trap Configuration

SNMP Settings

SNMP Parameters

Field

Description

Enabled, Disabled

Click Enabled to enable this feature, or click Disabled to disable it. The default setting is Disabled.

Trusted IPv4

Choose Any to allow access from any IPv4 address (not recommended).

Click Address to specify the IPv4 address and subnet mask of a single SNMP manager or trap agent that can access the ATA through SNMP.

Trusted IPv6

Choose Any to allow access from any IPv6 address (not recommended).

Click Address to specify the IPv6 address and prefix length of a single SNMP manager or trap agent that can access the ATA through SNMP.

Get/Trap Community

Enter a community string for authentication for SNMP GET commands. The default value is public.

Set Community

Enter a community string for authentication for SNMP SET commands. The default value is private.

SNMPv3 Settings

SNMPv3 Parameters

Field

Description

Enabled, Disabled

Click Enabled to enable this feature, or click Disabled to disable it. The default setting is Disabled.

R/W User

Enter the user name for SNMPv3 authentication. The default value is v3rwuser.

Auth-Protocol

Choose the SNMPv3 authentication protocol from the drop-down list (HMAC-MD5 or HMAC-SHA).

Auth-Password

Enter the authentication password.

PrivProtocol

Choose a privacy authentication protocol from the drop-down list (None or CBC-DES). If you select CBCDES, the privKey encrypts the data portion of the message that is being sent.

Privacy Password

Enter the key for the authentication protocol to use.

Trap Configuration

Trap Parameters

Field

Description

IP Address

The IP Address of the SNMP manager or trap agent.

Port

The SNMP trap port used by the SNMP manager or trap agent to receive the trap messages. Valid entries are 162 or 1025–65535. The default value is162.

SNMP Version

The SNMP version in use by the SNMP manager or trap agent. Choose a version from the list.

User List (Password Management)

Use the Administration > Management > User List page to manage the two user accounts for the ATA web page. The user-level account has access to modify a limited set of features.

For the IVR, you can configure these passwords on the System page.

- Update a Password

Update a Password

Procedure

Step 1

In the User List table, click the pencil icon for the account that you want to update.

Step 2

On the User Account page, enter the username and password, as described below.

- Username—Enter a username.

- Old Password (administrator account only)—Enter the existing password.

- New Password—Enter your new password. The password must contain 8 to 32 characters.

- Confirm New Password—Enter the new password again, to confirm.

Step 3

After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Bonjour

Use the Administration > Management > Bonjour page to enable or disable Bonjour. Bonjour is a service discovery protocol that locates network devices such as computers and servers on your LAN. It may be required by network management systems that you use. When this feature is enabled, the ATA periodically multicasts Bonjour service records to its entire local network to advertise its existence.

Click Enabled to enable this feature, or click Disabled to disable it. The default setting is Enabled.

After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Reset Button

Click Enabled to enable the reset button, or click Disabled to disable it. The default setting is Enabled.

After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

SSH

Use the Administration > Management > SSH to configure SSH related setting.

SSH Settings

Field

Description

User Name

Set SSH login user name.

Password

Set SSH login password.

SSH Access

Set the SSH access to enable or disable.

Log

The ATA allows you to record incoming, outgoing, and DHCP lists for various events that occur on your network. The Incoming Log displays a temporary list of the source IP addresses and destination port numbers for the incoming Internet traffic. The Outgoing Log displays a temporary list of the local IP addresses, destination URLs/IP addresses, and service/port numbers for the outgoing Internet traffic.

- Debug Log Module

- Debug Log Setting

- Debug Log Viewer

- Event Log Setting

- PRT Viewer

- PCM Viewer

- CSS Dump

- Crash Dump

Debug Log Module

Use the Administration > Log > Debug Log Module page to enable and configure logging.

- As a best practice, we recommend that you enable logging only when needed, and disable logging when you finish the investigation. Logging consumes resources and can impact system performance.

- In this page, you can select the modules which you want to see debug messages in all severity levels.

Debug Log Setting

If Debug Log Server is enabled on the Administration > Log > Debug Log Server page, the ATA will send the debug messages to one server.

Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Debug Log Setting

Field

Description

Debug Log Size

Enter the maximum size of the log file in kilobytes. Valid values are from 128 to 1024.

IPv4 Address

Enter the IPv4 address of the debug log server where the messages will be sent.

IPv6 Address

Enter the IPv6 address of the debug log server where the messages will be sent.

Port

Enter the port to use on the server. Valid values are from 1 to 65535.

Debug Log Viewer

If logging is enabled on the Administration > Log > Debug Log Viewer page, you can use the Log Viewer page view the logs online and to download the system log file to your computer. You can limit the contents of the log by choosing the types of entries to include and by specifying keywords.

For information about enabling and configuring logging, see Debug Log Module.

Debug Log Settings

Field

Description

Download Log

Click this button to download the contents of the log as a file on your computer. In the dialog box, you can open the file or save it. The file can be opened in a text editor such as Notepad.

Clear Log

Click this button to remove all entries from the log.

Filter

Enter a keyword to filter the log entries that appear in the viewer. The page will display only the entries that include the keyword.

Event Log Setting

Use the Administration > Log > Event Log Setting page to collect required event logs. Event log messages are sent via SYSLOG protocol using UDP transport type.

Use the Event Log Setting when troubleshooting. Four event categories are defined:

- DEV—Device information. A message is sent once device boot-up and network connectivity are ready.

- SYS—System-related information. A message is sent once while device boot-up and network connectivity are ready.

- CFG—Status of provision and configuration file change. A message is sent every time the provision service restarts due to configuration or network status change.

- REG—Registration status for each line. A message is sent every time registration status changes.

Enter the settings as described below. After making changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Event Log Settings

Field

Description

Address

Set the Event log server address.

Port

Set the Event log server port.

Default value: 514

Flag

Set the Event log flag, it’s a bitwise value. Setting list is as below:

- <Dev>:1(0x01)

- <SYS>:2(0x01<<1)

- <CFG>:4(0x01<<2)

- <REG>:8(0x01<<3)

Default value: 15 (All events)

PRT Viewer

Use the Administration > Log > PRT Viewer to generate and download Problem Report Tools (PRT) files.

To generate a problem report remotely, see Generate a Problem Report Remotely.

After making your changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Problem Report Tool Settings

Field

Description

PRT Upload URL

Set the PRT log upload URL

PRT Upload Method

Set the PRT log upload method, POST or PUT.

Default: POST

PRT Max Timer

Set the PRT max timer, valid range is 15-1440 minutes

Disabled: 0

Default: 0

Problem Report Tools Logs

List the PRT file which is generated by user on ATA.

Generate PRT

Click this button to generate and download the contents of the PRT as a file on your computer. In the dialog box, you can open the file or save it.

- Generate a Problem Report Remotely

Generate a Problem Report Remotely

You can initiate a problem report remotely. To do this, initiate a SIP-NOTIFY message from the server to the ATA, with the Event specified as prt-gen. and ATA response 200 OK to the server. Then

The workflow of the PRT generation is:

- The server sends NOTIFY to the ATA.

- ATA response 200 OK to the server, and sends PRT file to the upload server.

- The PRT upload server response 200 OK to the ATA.

Before you begin

- ATA registers successfully

- PRT Upload URL is configured

Procedure

Step 1

In the Administration > Log > PRT Viewer section, enter the PRT Upload URL parameter to specifiy the server to which you want to send the PRT. For example: http://10.74.133.94:9090.

The line is provisioned correctly with a valid SIP account.

Step 2

Click Submit All Changes.

PCM Viewer

Use the Administration > Log > PCM Viewer to download and view PCM.

The ATA allows you to capture the PCM log file while a user offhook to start a call.

After making your changes, click Submit to save your settings, or click Cancel to redisplay the page with the saved settings.

Log Viewer Settings

Field

Description

PCM Capture Enable

Enable or disable capture PCM.

Duration

Enter the PCM capture duration in seconds. The valid range is 20 to 300.

PCM File List

List the PCM file which is captured by user.

Click Refresh to refresh the PCM Memory Dump File. Click Download to download the dump file on your computer.

CSS Dump

Use the Administration > Log > CSS Dump page to set and download CSS dump file.

CSS Dump Settings

Field

Description

Auto Crash Dump

Set whether the ATA creates a crash dump file automatically when it occurs an error.

Click Enabled to enable the feature, click Disabled to disable it.

Default setting: Disabled

Manual Trigger Key(**##)

Set whether the user can manually trigger the creation of the CSS dump by pressing **## on the phone keypad.

Click Enabled to enable the feature, click Disabled to disable it.

Default setting: Disabled

CSS Dump List

List the CSS file which is captured by user.

Click Refresh to refresh the CSS Memory Dump File. Click Download to download the dump file on your computer.

Crash Dump

Use the Administration > Log > Crash Dump page to set and download crash dump file.

CSS Dump Settings

Field

Description

Runtime log to flash

Set whether the runtime log can be stored in the flash memory.

Click Enabled to enable the feature, click Disabled to disable it.

Default setting: Disabled

Crash Dump File

Display the captured crash dump file.

Click the file name to download it on your computer.

Click Refresh to refresh the crash dump file.

Factory Defaults

Use the Administration > Factory Defaults ATA web page to reset the ATA to the default configuration.

Alternatively, press and hold the RESET button for 10 seconds.

After the factory reset is performed successfully, all LEDs are fast flashing green.

All user-changeable non-default settings will be lost. This may include network and service provider data.

You can perform the following tasks:

- Restore Router Factory Defaults: Choose Yes to remove any custom data (router) settings that you have configured. The default settings will be restored when you click Submit.

- Restore Voice Factory Defaults: Choose Yes to remove any custom settings that you configured on the Voice pages of the ATA web page. The default settings will be restored when you click Submit.

Firmware Upgrade

Use the Administration > Firmware Upgrade page to upgrade the firmware on the ATA. It is not necessary to upgrade unless you are experiencing problems with the ATA or if the new firmware has a feature that you want to use.

Caution

Upgrading the firmware may take several minutes. Until the process is complete, DO NOT turn off the power, press the hardware reset button, or click the Back button in your current browser.

Before you begin

Before upgrading the firmware, download the firmware upgrade file for the ATA.

Procedure

Step 1

Click Browse and select the location of the upgrade file that you downloaded.

Step 2

Click the Upgrade button to upgrade the firmware.

- ATA with Firmware Release 11.1.0MSR3-9 and Older Doesn't Upgrade

ATA with Firmware Release 11.1.0MSR3-9 and Older Doesn't Upgrade

ATA with firmware version 11.1.0MSR3-9 and older fails to upgrade to the later releases if the server is implemented as HTTP chunk mode and the chunk-size is greater than 16K.

To bypass this upgrade failure, we recomment you to use this external HTTP server.

http://52.26.82.54/ata/ATA19x.11-1-0MPP0414-004.img

Configuration Management

Use the Administration > Config Management pages to backup and restore the configuration settings for the ATA.

- Backup Configuration

- Restore Configuration

Backup Configuration

Use the Administration > Config Management > Backup Configuration page to back up the ATA configuration settings to a file. You can then later restore these same settings to the ATA.

Click the Backup button to save the configuration information of the ATA. When the dialog box appears, choose a location where you want to save the .cfg file.

Tip: Rename the file with a name that includes the date and time when you did the backup.

Restore Configuration

Use the Administration > Config Management > Restore Configuration page to restore the ATA configuration settings from a previous backup. We recommend that you back up your current configuration settings before you restore a configuration.

Procedure

Step 1

Click Browse to locate the .cfg file on your computer.

Step 2

Click Restore to restore the settings from the selected file.

Reboot

Use the Administration > Reboot page to power cycle the ATA from the ATA web page. Another way to do it is by pressing the Reset > Reboot button.

Click the Reboot button to power cycle the ATA. When the warning message appears, read the information, and then click OK to reboot the ATA, or click Cancel to abandon the operation. The ATA and any connected devices will lose network connectivity during this operation.

Chapter 7. Status and Statistics

- System Information

- Interface Information

- Network Status

- Port Statistics (ATA 192 Only)

- Memory Information

- DHCP Server Information (ATA 192 Only)

System Information

Use the Status > System Information page to view information about the ATA and its current settings.

System Settings

Field

Description

Model

The model number and product description.

Product ID

The product ID of the ATA.

VID

The VID of the ATA

Serial Number

The serial number of the ATA.

Hardware Revision

The hardware version number.

Boot Version

The boot firmware version number.

Boot Partition

The boot partition of the ATA.

Firmware Version

The current firmware version.

Internet MAC Address

The MAC address of the WAN interface.

Host Name

The host name of the ATA.

Domain Name

The domain name of the ATA.

Current Time

Time that is set on the ATA.

Time Zone

Time zone that is set on the ATA.

Interface Information

Use the Status > Interface Information page to view information for the WAN interface (INTERNET port) and on ATA 192 only, the LAN interface (ETHERNET port).

IPv4 Interface List

Field

Description

Interface

The name of the interface: WAN or LAN (ATA 192 only).

Connect Type

The type of connection configured for the interface.

IP Address

The IPv4 address of the interface.

Subnet Mask

The subnet mask of the interface.

MAC Address

The MAC address of the interface.

IPv6 Interface List

Field

Description

Interface

The name of the interface: WAN or LAN (ATA 192 only).

Connect Type

The type of connection configured for the interface.

IP Address

The IPv6 address of the interface.

Prefix Length

The Prefix length of the interface.

MAC Address

The MAC address of the interface.

Port List (ATA 192 only)

Field

Description

Interface

The name of the interface: WAN or LAN.

TX (pkts)

The number of packets transmitted from this port.

RX (pkts)

The number of packets received by this port.

Status

The status of the port, showing whether the port is connected to a device or disconnected.

Clear TX & RX

Click this button to reset the count of TX and RX packets to zero.

Network Status

Use the Status > Network Status page to view information about the WAN interface (INTERNET port).

Basic Interface Detail

Field

Description

Link Status

The status of the INTERNET (WAN) interface, showing whether the port is connected or disconnected.

Host Name

The host name of the ATA.

Domain

The domain name of the ATA.

IPv4 Interface Detail

Field

Description

IP Address

The IPv4 address of the INTERNET (WAN) interface.

Subnet Mask

The subnet mask for the INTERNET (WAN) interface.

Gateway

The IPv4 address of the default gateway.

MTU Type

The method for setting the MTU: Auto or Manual

MTU Size

The largest protocol data unit (in bytes) permitted for network transmission

DNS 1-3 (if applicable)

IPv4 addresses for up to three DNS servers that are used for name resolution.

IPv6 Interface Detail

Field

Description

IP Address

The IPv6 address of the INTERNET (WAN) interface.

Prefix Length

The Prefix Length for the INTERNET (WAN) interface.

Gateway

The IPv6 address of the default gateway.

DNS 1-2 (if applicable)

IPv6 addresses for up to three DNS servers that are used for name resolution.

VLAN Information

Field

Description

CDP

The CDP status is enable or disable.

CDP VLAN ID

The CDP VLAN ID of the ATA.

IVR VLAN ID

The IVR VLAN ID of the ATA.

Active Vlan ID

The Active VLAN ID of the ATA.

Port Statistics (ATA 192 Only)

Use the Status > Port Statistics page to view information about the port activity on the WAN interface (INTERNET port) and the LAN interface (ETHERNET port).

Port Statistics Settings

Field

Description

Input (pkts)

The number of packets received by the port.

Output (pkts)

The number of packets transmitted by the port.

Input Errors

The number of receive errors for incoming traffic.

Input Broadcasts

The number of broadcast messages received by the interface.

Output Broadcasts

The number of broadcast messages sent by the interface.

Input Multicasts

The number of multicast messages received by the interface.

Output Multicasts

The number of multicast messages sent by the interface.

Memory Information

Use the Status > Memory Information page to view information about memory use.

Memory Information Interface Detail

Field

Description

MemTotal

The total memory of ATA.

MemFree

The free memory of ATA.

refresh

Refresh the latest memory information.

DHCP Server Information (ATA 192 Only)

Use the Status > DHCP Server Information page to view information about the DHCP server and clients.

IPv4 DHCP Pool Information

Field

Description

Client Name

The host name of the DHCP client.

IP Address

The IP address leased to the client.

MAC Address

The MAC address of the DHCP client.

Expires Time

The remaining time in the current DHCP lease, shown in HH:MM:SS (hours:minutes:seconds) format. The page is periodically updated with the new value as the timer counts down.

Interface

The interface through which the client is connected.

IPv6 DHCP Pool Information

Field

Description

Client Name

The host name of the DHCP client.

IP Address

The IP address leased to the client.

MAC Address

The MAC address of the DHCPv6 client.

Expires Time

The remaining time in the current DHCP lease, shown in HH:MM:SS (hours:minutes:seconds) format. The page is periodically updated with the new value as the timer counts down.

Interface

The interface through which the client is connected.

IPv4 DHCP Server Details

Field

Description

DHCP Server

The status of the DHCP server: Enabled or Disabled.

IP Address / Mask

The IP address and subnet mask for the ETHERNET (LAN) interface.

DNS Proxy

The setting for the DNS proxy service: Enabled or Disabled.

Maximum DHCP Users

The maximum number of clients that can lease an IP address from the DHCP server.

IP Address Range

The range of IP addresses that can be dynamically assigned by the DHCP server.

Client Lease Time

The maximum amount of time, in minutes, that a client can lease a dynamically assigned IP address.

Static DNS

The IP addresses of up to three DNS servers to be used by DHCP clients.

Option 66

The setting for Option 66, which provides provisioning server address information to hosts requesting this option. The ATA may be set to None (internal), Remote TFTP Server, or Manual TFTP Server.

TFTP Server

The IP address, hostname, or URL of the TFTP server used for provisioning.

Option 67

The configuration/bootstrap filename that is provided to hosts that request this option.

Option 159

The configuration URL that is provided to clients that request this option.

Option 160

The configuration URL that is provided to clients that request this option.

IPv6 DHCP Server Details

Field

Description

DHCPv6 Server

Display the DHCPv6 Server status.

Address Assign Type

Display the DHCPv6 Server Address assign type.

DHCPv6 Delegation

Display the DHCPv6 Server delegation is yes or no.

IPv6 Address Prefix

Display the DHCPv6 address prefix.

IPv6 Address Prefix Length

Display the DHCPv6 address prefix length.

IPv6 Static DNS

Display the DHCPv6 Static DNS.

IPv6 Active DNS1

Display the DHCPv6 Active DNS1.

IPv6 Active DNS2

Display the DHCPv6 Active DNS2.

IPv6 LAN Address

Display the DHCPv6 LAN address.

Chapter 8. Frequently Asked Questions

- I Can't Connect to the Internet Through the ATA

- I Upgraded my Firmware and the ATA doesn't Work Properly

- I Can't use the DSL Service to Connect Manually to the Internet

- There is no Dial Tone, and the Phone 1 or 2 LED is not Solid Green

- When I place an Internet Phone Call, the Audio Breaks Up

- When I Open a Web Browser, I am Prompted for a Username and Password. How can I Bypass this Prompt?

- The DSL Telephone Line Does Not Fit in the ATA WAN (Internet) Port.

- My Modem Doesn't Have an Ethernet Port

- The ATA Doesn't Have a Coaxial port for the Cable Connection

- Call Statistics are Not Available in the Server

I Can't Connect to the Internet Through the ATA

Procedure

Step 1

Make sure that the ATA is powered on. The Power/Sys LED should be solid green and not flashing.

If the Power LED is flashing, then power off all of your network devices, including the modem, the ATA, and the connected devices. Wait for 30 seconds. Then power on each device in the following order:

1. Cable or DSL modem

2. ATA

3. Connected Devices

Step 2

Check the cable connections. Ensure that the cable in the INTERNET (WAN) port is securely connected to the device that provides your Internet access, such as your modem or ADSL line. On the Cisco ATA 192, check the cable connection for the ETHERNET (LAN) port.

Step 3

Check the settings on the Network Setup > Internet Settings page. Verify that you entered the settings specified by your Internet Service provider.

I Upgraded my Firmware and the ATA doesn't Work Properly

If the ATA is not working properly after an upgrade, you may need to perform a factory reset. Use the Administration > Factory Defaults page to reset the ATA to the default configuration. Alternatively, press and hold the RESET button for 10 seconds. All user-changeable non-default settings will be lost. This may include network and service provider data.

I Can't use the DSL Service to Connect Manually to the Internet

After you have installed the ATA, it will automatically connect to your service provider’s network, so you no longer need to connect manually.

There is no Dial Tone, and the Phone 1 or 2 LED is not Solid Green

Procedure

Step 1

Make sure the telephone is connected to the appropriate port, PHONE 1 or 2.

Step 2

Disconnect the RJ-11 telephone cable from the PHONE port, and then reconnect it.

Step 3

Make sure your telephone is set to its tone setting (not pulse).

Step 4

Make sure your network has an active Internet connection.

Try to access the Internet, and check to see if the ATA WAN LED is flashing green. If you do not have a connection, then power off all of your network devices, including the modem, the ATA, and the computers. Wait 30 seconds. Then power on each device in the following order:

1. Cable or DSL modem

2. ATA

3. Computers and other devices

Step 5

Verify the settings on the Quick Setup page. Verify that you entered the account information and settings required by your service provider. On the Voice > Info page, Line 1 or Line 2 Status section, verify that the Registration State is registered. If the line is not registered, check with your ITSP to determine if additional settings are required.

When I place an Internet Phone Call, the Audio Breaks Up

Consider the following possible causes and solutions:

- Network activity—There may be heavy network activity, particularly if you are running a server or using a file sharing program. Try to limit network or Internet activity during Internet phone calls. For example, if you are running a file sharing program, files may be uploaded in the background even though you are not downloading any files, so make sure you exit the program before making Internet phone calls.

- Bandwidth—There may insufficient bandwidth available for your Internet phone call. You may want to test your bandwidth by using one of the bandwidth tests available online. If necessary, access your Internet phone service account and reduce the bandwidth requirements for your service. For more information, refer to the website of your ITSP.

When I Open a Web Browser, I am Prompted for a Username and Password. How can I Bypass this Prompt?

Launch the web browser and perform the following steps (these steps are specific to Internet Explorer but are similar for other browsers).

Procedure

Step 1

Select Tools > Internet Options.

Step 2

Click the Connections tab.

Step 3

Select Never dial a connection.

Step 4

Click OK.

The DSL Telephone Line Does Not Fit in the ATA WAN (Internet) Port.

The ATA does not replace your modem. You need your DSL modem in order to use the ATA. Connect your telephone line to the DSL modem.

My Modem Doesn't Have an Ethernet Port

If your modem does not have an Ethernet port, then it is a modem for traditional dial-up service. To use the ATA, you need a cable/DSL modem and a high-speed Internet connection.

The ATA Doesn't Have a Coaxial port for the Cable Connection

The ATA does not replace your modem. You need your cable modem in order to use the ATA. Connect your cable connection to the cable modem.

Call Statistics are Not Available in the Server

When Call Statistics are not available in the server, check the following:.

- Ensure the Call Statistics parameter is set to Yes in the web based configuration utility of ATA19x. You can check this parameter from Voice > SIP > RTP Parameters

- In the configuration file, the parameter must have value:

<Call_Statistics ua="na">Yes</Call_Statistics>

Chapter 9. IVR for Administration

- Use IVR for Administration

- IVR Tips

- IVR Actions

Use IVR for Administration

An IVR system is available to help you to configure and manage your ATA. Use a telephone keypad to select options and to make your entries.

Procedure

Step 1

Connect an analog phone to a PHONE port of the ATA.

Step 2

Press the star (*) key four times: ****

Step 3

When challenged for a password:

- Log in as an administrator.

- Log in as the PHONE port's user.

Step 4

Enter the code for the desired action.

IVR Tips

- Enter the numbers slowly, listening for the audio confirmation before entering the next number.

- After you select an option, press the # (pound) key.

- To exit the menu, hang up the telephone or enter 3948# to exit.

- After entering a value, such as an IP address, press the # (pound) key to indicate that you have finished your selection. Then proceed as needed:

- To save a setting, press 1.

- To review a setting, press 2.

- To re-enter a setting, press 3.

- To cancel your entry and return to the main menu, press * (star).

- While entering a value, you can cancel the changes by pressing the * (star) key twice within half a second. Be sure to press the key quickly, or the * will be treated as a decimal point entry.

- If the menu is inactive for more than one minute, the IVR times out. You will need to re-enter the IVR menu by pressing the star key four times: ****. Your settings take effect after you hang up the telephone or exit the IVR. The ATA may reboot at this time.

- To enter the decimal points in an IP address, press the * (star) key.

For example, to enter the IP address 191.168.1.105, perform the following tasks:

- Press these keys: 191*168*1*105

- Press the # (pound) key to indicate that you have finished entering the IP address.

- Press 1 to save the IP address or press the * (star) key to cancel your entry and return to the main menu.

IVR Actions

IVR Settings

IVR Action

Menu Option

Choices and Instructions

Enter IVR Menu

****

Check Internet Addressing Method

100

Check Internet6 Addressing Method

600

Set Internet Addressing Method

101

0—DHCP

1—Static IP

2—PPoE

Check Stack Mode

102

0—IPv4

1—IPv6

2—Dual

Set Stack Mode

103

0—IPv4

1—IPv6

2—Dual

Set Internet6 Addressing Method

601

0—DHCP

1—Static IP

2—PPPoE

Check IPv6 Auto Configuration

607

0—Disable

1—Enable

Set IPv6 Auto Configuration

606

0—Disable

1—Enable

Check Internet IP Address (INTERNET port)

110

Check Internet6 Address (INTERNET port)

610

Set Static IP Address (INTERNET port)

111

Enter the IP address by using numbers on the telephone key pad. Use the * (star) key to a decimal point.

Note

This option is available only after you choose Static IP as the Internet Connection Type, through option 101.

Set Static IPv6 Address (INTERNET port)

611

Available only in static IPv6 mode

Check Network Mask

120

Check IPv6 Prefix length

620

Set Network Mask

121

To enter the value, press numbers on the telephone key pad. Press the * (star) key to enter a decimal point.

Note

This option is available only after you choose Static IP as the Internet Connection Type, through option 101.

Set Static IPv6 Prefix length

621

Available only in static IPv6 mode

Check Gateway IP Address

130

Check Gateway IPv6 Address

630

Set Gateway IP Address

131

To enter the value, press numbers on the telephone key pad. Press the * (star) key to enter a decimal point.

Note

This option is available only after you choose Static IP as the Internet Connection Type, through option 101.

Set Gateway IPv6 Address

631

Available only in static IPv6 mode

Check MAC Address

140

Check Firmware Version

150

Check Primary DNS Server Setting

160

Check Primary IPv6 DNS Server Setting

660

Set Primary DNS Server

161

To enter the value, press numbers on the telephone key pad. Press the * (star) key to enter a decimal point.

Note

This option is available only after you choose Static IP as the Internet Connection Type, through option 101.

Set Primary IPv6 DNS Server

661

Check INTRNET web server port

170

ATA 192 only: Check LAN IP address (ETHERNET port)

210

Announce Line 1 SIP Transport

1910

Set Line 1 SIP Transport

1911

0—UDP

1—TCP

2—TLS

Check Line 2 SIP Transport

1920

Set Line 2 SIP Transport

1921

0—UDP

1—TCP

2—TLS

Exit IVR

3948

(Spells EXIT on the phone keypad)

Reboot of Voice System

732668

(Spells REBOOT on the phone keypad)

After you hear “Option successful,” hang up the phone. The ATA reboots.

Note

This action is equivalent to Pressing and immediately releasing the RESET button.

Factory Reset of Unit

Warning

All non-default settings will be lost. This includes network and service provider data.

73738

(Spells RESET on the phone keypad)

When prompted, press 1 to confirm, or press * (star) to cancel. After you hear “Option successful,” hang up the phone. The ATA reboots.

Note

This action is equivalent to Pressing and holding the RESET button for 10 seconds.

User Factory Reset of Unit

Warning

All user-changeable non-default settings will be lost. This may include network and service provider data.

877778

When prompted, press 1 to confirm, or press * (star) to cancel. After you hear “Option successful,” hang up the phone. The ATA reboots.

Chapter 10. Advanced Options for Phone Services

- Optimize Fax Completion Rates

- Dial Plan Configuration

- Emergency Calls

Optimize Fax Completion Rates

Issues can occur with fax transmissions over IP networks, even with the T.38 standard. Use the following task to help avoid any issues.

Procedure

Step 1

Ensure that you have enough bandwidth for the uplink and the downlink.

- For G.711 fallback, we recommend approximately 100 kbps.

- For T.38, allocate at least 50 kbps.

Step 2

Click Voice in the menu bar, and then click Line 1 or Line 2 in the navigation tree.

Step 3

In the Network Settings section, enter the following settings:

- Network Jitter Level—very high.

- Jitter Buffer Adjustment—no.

Step 4

In the Supplementary Service Subscription section, enter the following settings:

- Call Waiting Serv—no.

- Three Way Call Serv—no.

Step 5

In the Audio Configuration section, enter the following settings to support T.38 fax:

- Preferred Codec—G.711u (USA) or G.711a (rest of the world).

- Use pref. codec only—Yes.

- Silence Supp Enable—No.

- Echo Canc Enable—No.

- FAX Passthru Method—ReINVITE.

Step 6

Click Submit to save your settings or click Cancel to abandon the unsaved settings.

Step 7

If you are using a Cisco media gateway for PSTN termination, disable T.38 (fax relay) and enable fax using modem passthrough.

For example:

modem passthrough nse payload-type 110 codec g711ulaw

fax rate disable

fax protocol pass-through g711ulaw

Note

If a T.38 call cannot be set up, then the call automatically reverts to G.711 fallback.

Step 8

If you are using a Cisco media gateway, make sure that the Cisco gateway is correctly configured for T.38 with the dial peer.

For example:

fax protocol T38

fax rate voice

fax-relay ecm disable

fax nsf 000000

no vad

- Troubleshoot Your Fax

Troubleshoot Your Fax

If you have problems sending or receiving faxes, complete the following steps:

Procedure

Step 1

Verify that your fax machine is set to a speed between 7200 and 14400.

Step 2

Send a test fax in a controlled environment between two ATAs.

Step 3

Determine the success rate.

Step 4

Monitor the network and record the statistics for jitter, loss, and delay.

Step 5

If faxes fail consistently, capture a copy of the configuration. You can then send this file to Technical Support.

1. In your web browser, enter the path for the configuration file:

https://<ATA_Local_IP_Address>/admin/config.xml&xuser=

<admin_user>&xpassword=<admin_password>

2. On the File menu, choose Save As, and save the file with a filename such as MyConfiguration.xml.

Step 6

To enable logging, go to the Voice > System page, and set the IP address of your syslog or debug server. Set the Debug Level to 3. For more information, see System.

Note

You can also capture data using a sniffer trace.

Step 7

Identify the type of fax machine connected to the ATA.

Step 8

Contact technical support:

- If you are an user of VoIP products, contact the reseller or service provider that supplied the equipment.

- If you are an authorized Cisco partner, contact Cisco technical support. For contact options, see https://www.cisco.com/go/sbc.

Dial Plan Configuration

Dial plans determine how dialed digits are interpreted and transmitted. They also determine whether the dialed number is accepted or rejected. You can use a dial plan to facilitate dialing or to block certain types of calls such as long distance or international.

To edit a dial plan, click Voice on the menu bar, and then click Line 1 or Line 2 in the navigation tree. Scroll down to the Dial Plan section, and then enter the digit sequences in the Dial Plan field.

- Digit Sequences

- Acceptance and Transmission of the Dialed Digits

- Dial Plan Timer (Off-Hook Timer)

- Interdigit Long Timer (Incomplete Entry Timer)

- Interdigit Short Timer (Complete Entry Timer)

- Reset the Control Timers

Digit Sequences

A dial plan contains a series of digit sequences, separated by the pipe character: | .

The entire collection of sequences is enclosed within parentheses. Each digit sequence within the dial plan includes a series of elements, which are individually matched to the keys that the user presses.

Note

White space is ignored, but may be used for readability.

Digit Sequences

Digit Sequence

Function

0 1 2 3 4 5 6 7 8 9 0 * #

Enter any of these characters to represent a key that the user must press on the phone keypad.

x

Enter x to represent any character on the phone keypad.

[sequence]

Enter characters within square brackets to create a list of accepted key presses. The user can press any one of the keys in the list.

- Numeric range: For example, you would enter [2-9] to allow the user to press any one digit from 2 through 9.

- Numeric range with other characters: For example, you would enter [35-8*] to allow the user to press 3, 5, 6, 7, 8, or *.

. (period)

Enter a period for element repetition. The dial plan accepts zero or more entries of the digit. For example, 01. allows users to enter 0, 01, 011, 0111, and so on.

<dialed:substituted>

Use this format to indicate that certain dialed digits are replaced by other characters when the sequence is transmitted. The dialed digits can be zero or more characters.

EXAMPLE 1: <8:1650>xxxxxxx

When the user presses 8 followed by a seven digit number, the system automatically replaces the dialed 8 with 1650. If the user dials 85550112, the system transmits 16505550112.

EXAMPLE 2: <:1>xxxxxxxxxx

In this example, no digits are replaced. When the user enters a 10-digit string of numbers, the number 1 is added at the beginning of the sequence. If the user dials 9725550112, the system transmits 19725550112.

, (comma)

Enter a comma between digits to play an “outside line” dial tone after a user-entered sequence.

EXAMPLE: 9, 1xxxxxxxxxx

An “outside line” dial tone is sounded after the user presses 9, and the tone continues until the user presses 1.

! (exclamation point)

Enter an exclamation point to prohibit a dial sequence pattern.

EXAMPLE: 1900xxxxxxx!

The system rejects any 11-digit sequence that begins with 1900.

*xx

Enter an asterisk to allow the user to enter a 2-digit star code.

S0 or L0

Enter S0 to reduce the short inter-digit timer to 0 seconds, or enter L0 to reduce the long inter-digit timer to 0 seconds.

Digit Sequence Examples

The following examples show digit sequences that you can enter in a dial plan.

In a complete dial plan entry, sequences are separated by a pipe character (|), and the entire set of sequences is enclosed within parentheses.

EXAMPLE: ([1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

- Extensions on your system

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

[1-8]xx Allows a user dial any three-digit number that starts with the digits 1 through 8. If your system uses four-digit extensions, you would instead enter the following string: [1-8]xxx.

- Local dialing with seven-digit number

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]111)

9, xxxxxxx After a user presses 9, an external dial tone sounds. The user can then dial any seven-digit number, as in a local call.

- Local dialing with 3-digit area code and a 7-digit local number

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8,<:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx !| 9, 011xxxxxx. | 0 | [49]11 )

9, <:1>[2-9]xxxxxxxxx This example is useful where a local area code is required. After a user presses 9, an external dial tone sounds. The user must enter a 10-digit number that begins with a digit 2 through 9. The system automatically inserts the 1 prefix before transmitting the number to the carrier.

- Local dialing with an automatically inserted 3-digit area code

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

8, <:1212>xxxxxxx This is example is useful where a local area code is required by the carrier but the majority of calls go to one area code. After the user presses 8, an external dial tone sounds. The user can enter any seven-digit number. The system automatically inserts the 1 prefix and the 212 area code before transmitting the number to the carrier.

- U.S. long-distance dialing

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx |8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9,011xxxxxx. | 0 | [49]11 )

9, 1 [2-9] xxxxxxxxx After the user presses 9, an external dial tone sounds. The user can enter any 11-digit number that starts with 1 and is followed by a digit 2 through 9.

- Blocked number

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx |8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! |9, 011xxxxxx. | 0 | [49]11 )

9, 1 900 xxxxxxx ! This digit sequence is useful if you want to prevent users from dialing numbers that are associated with high tolls or inappropriate content, such as 1-900 numbers in the United States. After the user press 9, an external dial tone sounds. If the user enters an 11-digit number that starts with the digits 1900, the call is rejected.

- U.S. international dialing

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

9, 011xxxxxx. After the user presses 9, an external dial tone sounds. The user can enter any number that starts with 011, as in an international call from the United States.

- Informational numbers

( [1-8]xx | 9, xxxxxxx | 9, <:1>[2-9]xxxxxxxxx | 8, <:1212>xxxxxxx | 9, 1 [2-9] xxxxxxxxx | 9, 1 900 xxxxxxx ! | 9, 011xxxxxx. | 0 | [49]11 )

0 | [49]11 This example includes two digit sequences, separated by the pipe character. The first sequence allows a user to dial 0 for an operator. The second sequence allows the user to enter 411 for local information or 911 for emergency services.

Acceptance and Transmission of the Dialed Digits

When you dial a series of digits, each sequence in the dial plan is tested as a possible match. The matching sequences form a set of candidate digit sequences. As more digits are entered, the set of candidates diminishes until only one or none are valid. When a terminating event occurs, the ATA either accepts the dialed sequence and initiates a call, or else rejects the sequence as invalid. You hear the reorder (fast busy) tone if the dialed sequence is invalid.

The following table explains how terminating events are processed.

Terminating Events

Terminating Event

Processing

The dialed digits do not match any sequence in the dial plan.

The number is rejected.

The dialed digits exactly match one sequence in the dial plan.

- If the sequence is allowed by the dial plan, the number is accepted and is transmitted according to the dial plan.

- If the sequence is blocked by the dial plan, the number is rejected.

A timeout occurs.

The number is rejected if the dialed digits don''t matched a digit sequence in the dial plan within the time specified by the interdigit timer.

- The Interdigit Long Timer applies when the dialed digits do not match any digit sequence in the dial plan. Default setting: 10 seconds

- The Interdigit Short Timer applies when the dialed digits match one or more candidate sequences in the dial plan. Default setting: 3 seconds

You press the # key.

- If the sequence is complete and permitted by the dial plan, the number is accepted and is transmitted according to the dial plan.

- If the sequence is incomplete or is blocked by the dial plan, the number is rejected.

Dial Plan Timer (Off-Hook Timer)

You can think of the Dial Plan Timer as the "off-hook timer." This timer starts counting when the phone goes off hook. If no digits are dialed within the specified number of seconds, the timer expires and the null entry is evaluated. Unless you have a special dial plan string to allow a null entry, the call is rejected. Default setting: 5

Syntax for the Dial Plan Timer

(Ps<:n> | dial plan )

- s: The number of seconds; if no number is entered after P, the default timer of 5 seconds applies.

- n: (optional): The number to transmit automatically when the timer expires; you can enter a valid number. No wildcard characters are allowed because the number will be transmitted as shown. If you omit the number substitution, <:n>, then the user hears a reorder (fast busy) tone after the specified number of seconds.

Examples for the Dial Plan Timer

- Allow more time for users to start dialing after taking a phone off hook.

(P9 | (9,8<:1408>[2-9]xxxxxx | 9,8,1[2 9]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)

P9 After taking a phone off hook, a user has 9 seconds to begin dialing. If no digits are pressed within 9 seconds, the user hears a reorder (fast busy) tone. By setting a longer timer, you allow more time for users to enter the digits.

xx This code allows the entry of one or more digits. Do not use a single x, allowing 0 or more digits. This setting will produce unwanted results especially if you are deploying timers.

- Create a hotline for all sequences on the System Dial Plan

(P9<:23> | (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)

P9<:23> After taking the phone off hook, a user has 9 seconds to begin dialing. If no digits are pressed within 9 seconds, the call is transmitted automatically to extension 23.

- Create a hotline on a line button for an extension

(P0 <:1000>)

With the timer set to 0 seconds, the call is transmitted automatically to the specified extension when the phone goes off hook.

Interdigit Long Timer (Incomplete Entry Timer)

You can think of this timer as the “incomplete entry” timer. This timer measures the interval between dialed digits. It applies as long as the dialed digits do not match any digit sequences in the dial plan. Unless the user enters another digit within the specified number of seconds, the entry is evaluated as incomplete, and the call is rejected. Default setting: 10 seconds

This section explains how to edit a timer as part of a dial plan. Alternatively, you can modify the Control Timer that controls the default interdigit timers for all calls. See Reset the Control Timers.

Syntax for the Interdigit Long Timer

L:s, ( dial plan )

s: The number of seconds; if no number is entered after L:, the default timer of 5 seconds applies. The timer sequence appears to the left of the initial parenthesis for the dial plan.

Example for the Interdigit Long Timer

L:15, (9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)

L:15, This dial plan allows the user to pause for up to 15 seconds between digits before the Interdigit Long Timer expires.

Interdigit Short Timer (Complete Entry Timer)

You can think of this timer as the “complete entry” timer. This timer measures the interval between dialed digits. It applies when the dialed digits match at least one digit sequence in the dial plan. Unless the user enters another digit within the specified number of seconds, the entry is evaluated. If it is valid, the call proceeds. If it is invalid, the call is rejected. Default setting: 3 seconds

Syntax for the Interdigit Short Timer

SYNTAX 1: S:s, ( dial plan )

Use this syntax to apply the new setting to the entire dial plan within the parentheses.

SYNTAX 2: sequence Ss

Use this syntax to apply the new setting to a particular dialing sequence.

s: The number of seconds; if no number is entered after S, the default timer of 5 seconds applies.

Examples for the Interdigit Short Timer

Set the timer for the entire dial plan.

S:6,(9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxx | 9,8,011xx. | 9,8,xx.|[1-8]xx)

S:6, While entering a number with the phone off hook, a user can pause for up to 15 seconds between digits before the Interdigit Short Timer expires.

Set an instant timer for a particular sequence within the dial plan.

(9,8<:1408>[2-9]xxxxxx | 9,8,1[2-9]xxxxxxxxxS0 | 9,8,011xx. | 9,8,xx.|[1-8]xx)

9,8,1[2-9]xxxxxxxxxS0 With the timer set to 0, the call is transmitted automatically when the user dials the final digit in the sequence.

Reset the Control Timers

You can use the following procedure to reset the default timer settings for all calls.

To edit a timer setting only for a particular digit sequence or type of call, you can edit the dial plan. See Digit Sequences.

Procedure

Step 1

Log in to the ATA web page. If prompted, enter the administrative login provided by the Service Provider.

Step 2

Under the Voice menu, click Regional.

Step 3

In the Control Timer Values section, enter the desired values in the Interdigit Long Timer field and the Interdigit Short Timer field. See the definitions at the beginning of this section.

Emergency Calls

- Emergency Call Support Background

- Emergency Call Support Terminology

- Configure the ATA to Make Emergency Calls

Emergency Call Support Background

Emergency call service providers can register an ATA's location for each IP-based phone in a company. The location information server (LIS) transfers the emergency response location (ERL) to the ATA. The ATA stores its location during registration, after the ATA restarts. The location entry can specify the street address, building number, floor, room, and other office location information.

When you place an emergency call, the ATA transfers the location to the call server. The call server forwards the call and the location to the emergency call service provider. The emergency call service provider forwards the call and a unique call-back number (ELIN) to the emergency services. The emergency service or public safety answering point (PSAP) receives the ATA's location. The PSAP also receives a number to call you back, if the call disconnects.

See Emergency Call Support Terminology for the terms used to describe emergency calls from the phone.

The phone requests new location information for the following activities:

- You register the ATA with the call server.

- You or the user restarts the ATA and the ATA was previously registered with the call server.

- You change the network interface used in the SIP registration.

- You change the IP address of the ATA.

If both of the location servers do not send a location response, the phone resends the location request every two minutes.

Emergency Call Support Terminology

The following terms describe emergency call support for the ATA.

- Emergency Location ID Number (ELIN)–A number used to represent one or more ATA lines that locate the person who dialed emergency services.

- Emergency Response Location (ERL)–A logical location that groups a set of ATA lines.

- HTTP Enabled Location Delivery (HELD)–An encrypted protocol that obtains the PIDF-LO location for the ATA from a location information server (LIS).

- Location Information Server (LIS)–A server that responds to a SIP-based ATA HELD request and provides the ATA location using a HELD XML response.

- Emergency Call Service Provider–The company that responds to an ATA HELD request with the ATA's location. When you make an emergency call (which carries the ATA's location), a call server routes the call to this company. The emergency call service provider adds an ELIN and routes the call to the emergency services (PSAP). If the call is disconnected, the PSAP uses the ELIN to reconnect with the ATA used to make the emergency call.

- Public Safety Answering Point (PSAP)–Any emergency service (for example, fire, police, or ambulance) joined to the Emergency Services IP Network.

- Universally Unique Identifier (UUID)–A 128-bit number used to uniquely identify a company using emergency call support.

Configure the ATA to Make Emergency Calls

Before you begin

- Obtain the E911 Geolocation Configuration URLs and the company identifier for the ATA from your emergency call services provider. You can use the same Geolocation URLs and company identifier for line 1 and line 2 (PHONE 1 and PHONE 2).

- Access the phone adapter administration web page. See Access the Phone Web Interface.

Procedure

Step 1

Select Voice > Line n, where n is the line number that represents PHONE 1 or PHONE 2.

Step 2

In the section Call Feature Settings, set the parameter Emergency Number as described in Call Feature Settings.

Step 3

In the section E911 Geolocation Configuration, set the parameters Company UUID, Primary Request URL, and Secondary Request URL as described in E911 Geolocation Configuration.

Step 4

Click Submit.

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