HT813 - Administration Guide

• 2 SIP profiles through 1 FXS port and 1 FXO port
• Dual 100 Mbps LAN and WAN ports
• 3-way voice conferencing
• Wide range of caller ID formats
• Advanced telephony features, including call transfer, call forward, call-waiting, do not disturb, message waiting indication, multi-language prompts, flexible dial plan and more
• T.38 Fax for creating Fax-over-IP
• TLS and SRTP security encryption technology to protect calls and accounts
• Automated provisioning options include TR-069 and XML config files
• Strong AES encryption with security certificate per unit
• Failover SIP server automatically switches to secondary server if main server loses connection
• Use with Grandstream’s UCM series of IP PBXs for Zero Configuration provisioning
• Lifeline support (FXS port will be hard-relayed to FXO port) in case of power outage.

Interfaces

Telephone Interfaces

One (1) RJ11 FXS port, One (1) RJ11 FXO PSTN line port with lifeline support

Network Interface

Two (2) 10/100 Mbps ports (RJ45) with integrated NAT router

LED Indicators

POWER, FXO, FXS, WAN, LAN.

Factory Reset Button

Yes

Voice, Fax, Modem

Telephony Features

Caller ID display or block, call waiting, flash, blind or attended transfer, forward, hold, do not disturb, 3-way conference

Voice Codecs

G.711 with Annex I (PLC) and Annex II (VAD/CNG), G.723.1, G.729A/B, G.726, iLBC, OPUS, dynamic jitter buffer, advanced line echo cancellation

Fax over IP

T.38 compliant Group 3 Fax Relay up to 14.4kpbs and auto-switch to G.711 for Fax Pass-through

Short/Long Haul Ring Load

3 REN: Up to 1km on 24 AWG

Caller ID

Bellcore Type 1 & 2, ETSI, BT, NTT, and DTMF-based CID

Disconnect Methods

Busy Tone, Polarity Reversal/Wink, Loop Current

Signaling

Network Protocols

TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, FTP/FTPS, ARP/RARP, ICMP, DNS, DDNS, DHCP, NTP, TFTP, SSH, Telnet, STUN, SIP (RFC3261), SIP over TCP/TLS, SRTP, TR-069

QoS

Layer 2 (802.1Q VLAN, SIP/RTP 802.1p) and Layer 3 (ToS, Diffserv, MPLS).

DTMF Methods

In-audio, RFC2833 and/or SIP INFO

Provisioning and Control

HTTP, HTTPS, SSH, FTP, FTPS, Telnet, SSH, TFTP, TR-069, secure and automated provisioning using AES encryption, syslog

Security

Media

SRTP

Control

TLS/SIPS/HTTPS

Management

Syslog support, SSH, Telnet remote management using web browser.

Physical

Universal Power Supply

Input: 100-240VAC, 50/60Hz

Output: 12V/0.5A

Environmental

Operational: 32° – 104°F or 0º – 40ºC

Storage: 14° – 140°F or -10º – 60ºC

Humidity: 10 – 90% Non-condensing

Dimensions and Weight

130.5 x 90.5 x 29 mm (L x W x D)

Weight: 0.142Kg

Compliance

Compliance

FCC/CE/C-TICK/ITU-K.21

FXS

Connects the analog phone / fax machine to the ATA using an

RJ-11 telephone cable.

FXO

FXO telephone port (PSTN Port) 1x PSTN pass-through and life line port.

Connects the ATA to your router, switch or modem using an Ethernet RJ45 network cable.

Connects the ATA to your PC or switch using an Ethernet RJ45 network cable.

DC 12V

Connects the ATA to PSU (Output: 12V/0.5A )

Reset

Factory reset button. Press for 7 seconds to reset factory default.

settings. Quick press will only reboot the unit.

Power LED

The Power LED lights up when the HT813 is powered on and it flashes when the HT813 is booting up.

WAN LED

The WAN LED lights up when the HT813 is connected to your network through the WAN port.

LAN LED

The LAN LED lights up when the HT813 is connected to your network through the LAN port.

FXS/FXO LED

The FXS LEDs indicate status of the respective FXS/FXO port-phone on the back panel

• OFF – Unregistered
• ON (Solid Blue) – Registered and Available
• Blinking every 500 ms – Off-Hook / Busy
• Slow blinking – FXS LEDs indicates voicemail

Main Menu

“Enter a Menu Option”

Press “*” for the next menu option

Press “#” to return to the main menu

Enter 01-05, 07,10, 12-17,47 or 99 menu options

01

“DHCP Mode”,

“Static IP Mode”

“PPPoE Mode“

Press “9” to toggle the selection

If using “Static IP Mode”, configure the IP address information using menus 02 to 05.

If using “Dynamic IP Mode”, all IP address information comes from the DHCP server automatically after reboot.

If using “PPPoE Mode”, configure PPPoE Username and Password from web GUI to get IP from your ISP.

02

“IP Address “ + IP address

The current WAN IP address is announced

If using “Static IP Mode”, enter 12-digit new IP address. You need to reboot your HT813 for the new IP address to take Effect.

03

“Subnet “ + IP address

Same as menu 02

04

“Gateway “ + IP address

Same as menu 02

05

“DNS Server “ + IP address

Same as menu 02

07

Preferred Vocoder

Press “9” to move to the next selection in the list:

• PCM U / PCM A
• iLBC
• G-726
• G-723
• G-729
• OPUS

10

“MAC Address”

Announces the MAC address of the unit.

Note: The device has two MAC addresses. One for the WAN port and one for the LAN port. The device MAC address announced is the address of LAN port.

12

WAN Port Web Access

Press “9” to toggle between Auto / Enabled / Disabled.

Default is Auto.

13

Firmware Server IP Address

Announces current Firmware Server IP address.

Enter 12-digit new IP address.

14

Configuration Server IP Address

Announces current Config Server Path IP address. Enter 12-digit new IP address.

15

Upgrade Protocol

Upgrade protocol for firmware and configuration update. Press “9” to toggle between TFTP / HTTP / FTP / FTPS or HTTPS.

Default is HTTPS.

16

Firmware Version

Announces Firmware version information.

17

Firmware Upgrade

Firmware upgrade mode. Press “9” to toggle among the following three options:

• Always check
• Check when pre/suffix changes
• Never upgrade

47

“Direct IP Calling”

Enter the target IP address to make a direct IP call, after dial tone. (See “Make a Direct IP Call”.)

86

Voice Mail

Access to your voice mails messages.

99

“RESET”

Press “9” to reboot the device

Enter MAC address to restore factory default setting

(See RESTORE FACTORY DEFAULT SETTINGS section)

“Invalid Entry”

Automatically returns to main menu

“Device not registered”

This prompt will be played immediately after off hook If the device is not registered and the option “Outgoing Call without Registration” is in NO

End User Level

123

Only Status and Basic Settings

Administrator Level

admin

All pages

Viewer Level

viewer

Only checking. Not allowed to modify content.

Account Active

Activates / Deactivates the accounts. The FXS port configuration will not change if disabled, although the port will not be operational, in this state, there will be no dial tone when picking up the analog phone and making/receiving calls will not be possible.

Primary SIP Server

Configures SIP server IP address or domain name provided by VoIP service provider. This is the primary SIP server used to send/receive SIP messages from/to HT813.

Failover SIP Server

Specifies failover SIP server IP address or domain name provided by VoIP service provider.

This server will be used if the primary SIP server becomes unavailable.

Prefer Primary SIP Server

Selects to prefer primary SIP server. The account will register to primary Server if registration with Failover server expires.

Default is No.

Outbound Proxy

Specifies IP address or domain name of outbound Proxy, or media gateway, or session border controller. Used by HT813 for firewall or NAT penetration in different network environments. If symmetric NAT is detected, STUN will not work and only outbound proxy can correct the problem.

Backup Outbound Proxy

Configures the backup outbound proxy to be used when the “Outbound Proxy” registration fails. By default, this field is left empty.

Prefer Primary Outbound Proxy

If the user configures this option to “Yes”, when registration expires, the device will re-register via primary outbound proxy.

By default, this option is disabled.

Allow DHCP Option 120 (override SIP Server)

Configures the HT813 to collect SIP server address from DHCP option 120.

Default is No.

SIP Transport

Selects transport protocol for SIP packets; UDP or TCP or TLS. Please make sure your SIP Server or network environment supports SIP over the selected transport method. Default is UDP.

SIP URI Scheme When Using TLS

Specifies if “sip” or “sips” will be used when TLS/TCP is selected for SIP Transport. The default setting is “sips”.

Use Actual Ephemeral Port in Contact with TCP/TLS

Controls the port information in the Via header and Contact header. If set to “No”, these port numbers will use the permanent listening port on the phone. Otherwise, it will use the ephemeral port for the connection. The default setting is “No”.

NAT Traversal

Indicates type of NAT for each account. This parameter configures whether the NAT traversal mechanism is activated. Users could select the mechanism from No, Keep-alive, STUN, UPnP. Default setting is No.

SIP User ID

Defines user account information provided by VoIP service provider (ITSP). Usually in the form of digit similar to phone number or actually a phone number.

Authenticate ID

Determines account authenticate ID provided by VoIP service provider (ITSP). Can be identical to or different from “SIP user ID”.

Authenticate Password

Specifies account password provided by VoIP service provider (ITSP) to register to SIP servers.

Name

Chooses a name to be associated to user.

DNS Mode

Selects DNS mode to use for the client to look up server. Default is A Record.

• A Record: resolves IP Address of target according to domain name.
• SRV: DNS SRV resource records indicate how to find services for various protocols.
• NAPTR/SRV: Naming Authority Pointer according to RFC 2915.

DNS SRV use Registered IP

When this option is set to “Yes”, when the HT is registered on second SRV and makes an outbound call, it will try the second SRV (registered IP) first. By default, this option is disabled and the DNS SRV will use first SRV instead of the registered IP.

Tel URI

Indicates E.164 number in “From” header by adding “User=Phone” parameter or using “Tel:” in SIP packets, if the HT813 has an assigned PSTN Number.

• Disabled: Use “SIP User ID” information in the Request-Line and “From” header.
• User=Phone: “User=Phone” parameter will be attached to the Request-Line and “From” header in the SIP request to indicate the E.164 number. If set to “Enable”.
• Enabled: “Tel:” will be used instead of “sip:” in the SIP request.

Please consult your carrier before changing this parameter.

Default is Disabled.

SIP Registration

Controls whether the HT813 needs to send REGISTER messages to the proxy server. Default setting is Yes.

Unregister on Reboot

Controls whether to clear SIP user’s information by sending un-register request to the proxy server. The un-registration is performed by sending a REGISTER message with Contact set to * and Expires=0 parameters to the SIP server. This will unregister the SIP account under the concerned FXS page. Default is No.

Outgoing Call Without Registration

Enables the ability to place outgoing calls even if the account is not registered (if allowed by ITSP); device will not be able to receive incoming calls. Default is No.

Register Expiration

Refreshes registration periodically with specified SIP proxy (in minutes). Maximum interval is 65535 minutes (about 45 days). Default is 60 minutes (or 1 hour).

Reregister before Expiration

Sends re-register request after specific time (in seconds) to renew registration before the previous registration expires.

SIP Registration Failure Retry Wait Time

Sends re-register request after specific time (in seconds) when registration process fails. Maximum interval is 3600 seconds (1 hour). Default is 20 seconds.

SIP Registration Failure Retry Wait Time upon 403 Forbidden

Sends re-register request after specific time (in seconds) when registration process fails with error 403 Forbidden. Maximum interval is 3600 seconds (1 hour).

Default is 1200 seconds.

Enable SIP OPTIONS Keep Alive

Enables SIP OPTIONS to track account registration status so the phone adapter will send periodic OPTIONS message to server to track the connection status with the server. Default setting is No.

SIP OPTIONS Keep Alive Interval

Configures the time interval when the phone adapter sends OPTIONS message to SIP server. The default setting is 30 seconds, which means the phone adapter will send an OPTIONS message to the server every 30 seconds. The default range is 1-64800.

SIP OPTIONS Keep Alive Max Lost

Defines the Number of max lost packets for SIP OPTIONS Keep Alive before re-registration. Between 3-10, default is 3.

Layer 3 QoS

Defines Diff-Serv values for SIP and RTP. Defaults are:

SIP DSCP: 26

RTP DSCP: 46

Local SIP Port

Defines local port to use by the HT813 for listening and transmitting SIP packets. Default value for FXS is 5060.

Local RTP Port

Defines the local RTP-RTCP port pair the HT813 will listen and transmit. It is the HT813 RTP port for channel 0. The default value for FXS port is 5004.

Use Random SIP Port

Controls whether to use configured or random SIP ports. This is usually necessary when multiple HT813 are behind the same NAT. Default is No.

Use Random RTP Port

Controls whether to use configured or random RTP ports. This is usually necessary when multiple HT813 are behind the same NAT. Default is No.

Enable RTCP

Allows users to enable RTCP. Default setting is Yes.

Hold Target Before Refer

Allows user to hold the phone call before referring it. If set to No, the call will not be hold before referred. Default is Yes.

Refer-To Use Target Contact

Includes target’s “Contact” header information in “Refer-To” header when using attended transfer. Default is No. 

Transfer on Conference Hang-up

If set to “Yes”, when the phone hangs up as the conference initiator, the conference call will be transferred to the other parties so that other parties will remain in the conference call. Default setting is No.

Disable Bellcore Style 3-Way Conference

Gives the users the possibility of making conference calls by pressing “Flash” key, when it is enabled by dialing *23 +second callee number. Default is No

Remove OBP from Route Header

Support SIP Instance ID

Includes “SIP Instance ID” attribute to “Contact” header in REGISTER request as defined in IETF SIP outbound draft. Default is No.

Validate Incoming SIP Messages

Validates incoming messages. Default is No.

Check SIP User ID for Incoming INVITE

Checks SIP User ID in the Request URI of incoming INVITE; if it does not match the HT813 SIP User ID, the call will be rejected. Direct IP calling will also be disabled. Default is No.

Authenticate Incoming INVITE

Challenges the incoming INVITE for authentication with SIP 401 Unauthorized message. Default is No.

Authenticate server certificate domain

Configures whether to validate the domain certificate when download the firmware/config file. If it is set to “Yes”, the phone will download the firmware/config file only from the legitimate server. The default setting is “No“.

Authenticate server certificate chain

Configures whether to validate the server certificate when download the firmware/config file. If it is set to “Yes”, the phone will download the firmware/config file only from the legitimate server. The default setting is “No“.

Trusted CA Certificates

Uses the certificate for Authentication if “Check Domain Certificates” is set to “Yes” under “Account”🡪“SIP Settings”.

Allow Incoming SIP Messages from SIP Proxy Only

Checks SIP address of the Request URI in the incoming SIP message; if it does not match the SIP server address of the account, the call will be rejected. Default is No.

Use Privacy Header

Determines if the “Privacy header” will be presented in the SIP INVITE message and if it includes the caller info in this header. If set to Default, it will add Privacy header unless special feature is Telkom SA or CBCOM. Default is Default.

Use P-Preferred-Identity Header

Specifies if the P-Preferred-Identity Header will be presented in the SIP INVITE message. If set to “default”, the P-Preferred-Identity Header will be omitted in SIP INVITE message when Telkom SA or CBCOM is active. If set to “Yes”, the P-Preferred-Identity Header will always be presented. If set to “No”, it will be omitted.

Default setting is: Default.

Use P-Access-Network-Info Header

With this feature enabled, device will populate the WAN access node with IEEE802.11a, IEEE-802.11b in P-Access-Network-Info SIP header.

Use P-Emergency-Info Header

This feature support of IEEE-48-addr and IEEE-EUI-64 in SIP header for emergency calls.

SIP REGISTER Contact Header Uses

Specifies which address (LAN or WAN address) the device will detect to use it in SIP Register Contact Header. When set to LAN, Contact header will include local IP from ATA in REGISTER messages, while if set to WAN, host/port/contact will be updated from SIP 401/403/404/407 Via header “received”/”rport” parameters in REGISTER messages. Default is LAN Address.

Caller ID Fetch Order

Selects the Caller ID display order which need to be respected by the HT813. The available options are:

• Auto: When set to “Auto”, the HT813 will look for the caller ID in the order of P-Asserted Identity Header, Remote-Party-ID Header and From Header in the incoming SIP INVITE.
• Disabled: When set to “Disabled”, all incoming calls are displayed

with “Unavailable”.

• From Header: When set to “From Header”, the HT813 will use the FROM header to display the caller ID.

Allow SIP Factory Reset

This feature allows user to reset the devices directly through SIP Notify. Default is No.

SIP T1 Timeout

Defines T1 timeout value.

It is an estimate of the round-trip time between the client and server transactions.

For example, the HT813 will attempt to send a request to a SIP server.

The time it takes between sending out the request to the point of getting a response is the SIP T1 timer. If no response is received the timeout is increased to (2*T1) and then (4*T1). Request re-transmit retries would continue until a maximum amount of time defined by T2. Default is 0.5 seconds.

SIP T2 Interval

Identifies maximum retransmission interval for non-INVITE requests and INVITE responses. Retransmitting and doubling of T1 continues until it reaches T2 value. Default is 4 seconds.

SIP Timer D

Configure the SIP Timer D defined in RFC3261. 0 – 64 seconds. Default 0

DTMF Payload Type

Defines payload type for DTMF using RFC2833.

Preferred DTMF method (in order)

Sorts DTMF methods (in-audio, via RTP (RFC2833) or via SIP INFO) by priority.

Disable DTMF Negotiation

Uses above DTMF order without negotiation.

Default is No.

Generate Continuous RFC2833 Events

When enabled the RFC2833 events are generated until key is released. Default is No.

Send Hook Flash Event

Default is No. If set to yes, flash will be sent as DTMF event.

Flash Digit Control

When it set to YES it allows the user to perform some call setting when both channels are used while pressing:

• “Flash + 1” in order to hang up the current call and resume a call that was held.
• “Flash + 2” in order to hold the current call and resume a call that was held.
• “Flash + 3” in order to perform 3-way conference.
• “Flash + 4” in order to perform attended transfer.

Note: Please refer to the user guide for detailed steps to perform above operations.

Enable Call Features

Enables do not disturb, call forward and other call features via the local feature codes on the base. Otherwise, ITSP feature codes can be used.

Default is Yes.

Off Hook Auto Dial

Configures a user ID or extension number that is automatically dialed when off-hook. Only the user part of a SIP address needs to be entered. FXS port will automatically append the “@” and the host portion of the corresponding SIP address.

Off Hook Auto Dial Delay

Specifies the auto-dial delay after off hook.

Proxy-Require

Determines a SIP Extension to notify the SIP server that the HT813 is behind a NAT/Firewall.

Use NAT IP

Defines NAT IP address used in SIP/SDP messages. It should only be used if required by ITSP.

Use SIP User Agent Header

Configures the SIP User-Agent Header.

SIP User-Agent

Configures SIP User-Agent. If not configured, device will use the default User Agent Header. The value range is 1024 to Maximum String Length. Default value is Null.

Disable Call Waiting

Disables receiving a second incoming call when the line is engaged.

Default is No.

Disable Call Waiting Caller ID

Disables displaying caller ID when receiving a second incoming call. Default is No.

Disable Call Waiting Tone

Disables playing call waiting tone during active call when receiving a second incoming call. The CWCID will still be displayed. Default is No.

Disable Connected Line ID

Disables displaying the number of the person answering the phone. Default is No.

Disable Receiver Off Hook Tone

Enables / disables the warning to alert that the phone has been left off-hook for an extended period of time. Default is No.

Disable Reminder Ring for On-Hold Call

Enables playing the reminder ring. Default is No

Disable Visual MWI

Disables use of visual message waiting indicator when there is an unread voicemail message. Default is No.

Do Not Escape ‘#’ as %23 in SIP URI

Replaces # by %23 in some special situations. Default is No.

Disable Multiple m Line in SDP

Sends only one m line in SDP, regardless of how many m fields are in the incoming SDP. Default is No.

Ring Timeout

Stops ringing when incoming call if not answered within a specific period of time. When set to 0, There will be no ringing timeout. Default is 60 seconds.

Delayed Call Forward Wait Timeout

Forwards incoming call if not answered within a specific period of time when delayed call forward is activated locally (using *92 code). Default value is 20 seconds.

No Key Entry Timeout

Initiates the call within this time interval if no additional key entry during dialing stage. Default is 4 seconds.

Early Dial

Sends an early INVITE each time a key is pressed when a user dials a number. Otherwise, only one INVITE is sent after full number is dialed (user presses Dial Key or after “no key entry timeout” expires).

This option should be used only if there is a SIP proxy is configured and supporting 484 responses (Incomplete Address). Otherwise, the call will likely be rejected by the proxy (with a 404 Not Found error). Default is No.

This feature is NOT designed to work with and should NOT be enabled for direct IP-to-IP calling.

Dial Plan Prefix

Adds specified prefix to dialed number.

Use # as Dial Key

Treats “#” as the “Send” (or “Dial”) key. If set to “No”, this “#” key can be included as part of the dialed number.

Default is Yes.

Dial Plan

Dial Plan Rules:

1. Accept Digits: 1,2,3,4,5,6,7,8,9,0 , *, #, A,a,B,b,C,c,D,d
2. Grammar: x – any digit from 0-9;
   1. xx+ – at least 2 digits number;
   2. xx – exactly 2 digits number;
   3. ^ – exclude;
   4. . – wildcard, matches one or more characters
   5. [3-5] – any digit of 3, 4, or 5;
   6. [147] – any digit 1, 4, or 7;
   7. <2=011> – replace digit 2 with 011 when dialing
   8. < =1> – add a leading 1 to all numbers dialed, vice versa will remove a 1 from the number dialed
   9. | – or

• Example 1: {[369]11 | 1617xxxxxxx} –

Allow 311, 611, 911, and any 10-digit numbers of leading digits 1617

• Example 2: {^1900x+ | <=1617>xxxxxxx} –

Block any number with leading digits 1900 and add prefix 1617 for any dialed 7-digit numbers

• Example 3: {1xxx[2-9]xxxxxx | <2=011>x+} –

Allow any length of number with leading digit 2 and 10 digit-numbers of leading digit 1 and leading exchange number between 2 and 9; If leading digit is 2, replace leading digit 2 with 011 before dialing.

1. Default: Outgoing – {x+}

Example of a simple dial plan used in a Home/Office in the US:

{ ^1900x. | <=1617>[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx | 011[2-9]x. | [3469]11 }

Dial Plan

Explanation of example rule (reading from left to right):

• ^1900x. – prevents dialing any number started with 1900
• <=1617>[2-9]xxxxxx – allows dialing to local area code (617) numbers by dialing 7 numbers and 1617 area code will be added automatically
• 1[2-9]xx[2-9]xxxxxx – allows dialing to any US/Canada Number with 11-digit length
• 011[2-9]x. – allows international calls starting with 011
• [3469]11 – allow dialing special and emergency numbers 311, 411, 611 and 911

Note: In some cases, user wishes to dial strings such as *123 to activate voice mail or other application provided by service provider. In this case * should be predefined inside dial plan feature. As an example { *x+ } will allow to dial * followed by any length of numbers.

SUBSCRIBE for MWI

Sends SUBSCRIBE periodically (depends on “Register Expiration” parameter) for message waiting indication. Default is No.

Send Anonymous

Sets “From”, “Privacy” and “P_Asserted_Identity” headers in outgoing INVITE message to “anonymous”, blocking caller ID. Default is No.

Anonymous Call Rejection

Rejects incoming calls with anonymous caller ID with “486 Busy here” message. Default is No.

Special Feature

Selects Soft switch vendors’ special requirements Example of vendors: Standard, Broadsoft, CBCOM, RNK, Huawei, China Mobile, ZTE IMS, PhonePower, TELKOM SA, Vonage, Metaswitch, CenturyLink, MTS. Default is Standard.

Enable Session Timer

Disable the session timer when this option is set to “No”. By default, this option is enabled.

Session Expiration

Enables SIP sessions to be periodically “refreshed” via a SIP request (UPDATE, or re-INVITE). When the session interval expires, if there is no refresh via an UPDATE or re-INVITE message, the session will be terminated. Session Expiration is the time (in seconds) at which the session is considered timed out if no successful session refresh transaction occurs beforehand. Default is 180 seconds.

Min-SE

Defines Minimum session expiration (in seconds). Default is 90 seconds.

Caller Request Timer

Uses session timer when making outbound calls if remote party supports it.

Default is No.

Callee Request Timer

Uses session timer when receiving inbound calls with session timer request.

Default is No.

Force Timer

Uses session timer even if the remote party does not support this feature. Selecting “No” will enable session timer only when the remote party supports it. Default is No.

To turn off Session Timer, select “No” for Caller and Callee Request Timer, and Force Timer.

UAC Specify Refresher

Specifies which end will act as refresher for outgoing calls.

• UAC: The handy tone acts as the refresher.
• UAS: Callee or proxy server act as the refresher.

Default is Omit.

UAS Specify Refresher

Specifies which end will act as refresher for incoming calls:

• UAS: The handy tone acts as the refresher.
• UAC: Callee or proxy server act as the refresher.

Default is Omit.

Force INVITE

Uses INVITE message to refresh the session timer.

Default is No.

Enable 100rel

Appends “100rel” attribute to the value of the required header of the initial signaling messages. Default is No.

Add Auth Header on Initial REGISTER

Adds “Authentication” header with blank “nonce” attribute in the initial SIP REGISTER request.

Default is No.

Conference URI

Allows users to manually configure the conference URL.

Use First Matching Vocoder in 200OK SDP

Includes only the first matching vocoder in its 200OK response, otherwise it will include all matching vocoders in same order received in INVITE.

Default is No.

Preferred Vocoder

Configures vocoders in a preference list (up to 7 preferred vocoders) that will be included with same order in SDP message. Vocoder types are G.711 A-/U-law, G.726-32, G.723, G.729, iLBC and OPUS.

Voice Frames per TX

Transmits a specific number of voice frames per packet. Default is 2; increases to 10/20/32/64 for G711/G726/G723/other codecs, respectively.

G723 Rate

Operates at specified encoding rate for G.723 vocoder. Available encoding rates are 6.3kbps or 5.3kbps. Default is 6.3kbps.

iLBC Frame Size

Specifies iLBC packet frame size (20ms or 30ms).

Default is 20ms.

Disable OPUS Stereo in SDP

Disables OPUS stereo in SDP.

Default is No.

iLBC Payload type

Determines payload type for iLBC. Valid range is between 96 and 127.

Default is 97.

OPUS Payload Type

Determines payload type for OPUS. Valid range is between 96 and 127. Default is 123.

VAD

Allows detecting the absence of audio and conserves bandwidth by preventing the transmission of “silent packets” over the network. Default is No.

Symmetric RTP

Changes the destination to send RTP packets to the source IP address and port of the inbound RTP packet last received by the device. Default is No.

Fax Mode

Specifies the fax mode: T.38 (Auto Detect) FoIP by default, or Pass-Through (must use codec PCMU/PCMA)

Re-INVITE after Fax Tone Detected

Allows the unit to send out the re-INVITE for T.38 or Fax Pass Through if a fax tone is detected. Default is Enabled

Jitter Buffer Type

Selects jitter buffer type (Fixed or Adaptive) based on network conditions.

Jitter Buffer Length

• High (initial 200ms, min 40ms, max 600ms) Note: not all vocoders can meet the high requirement.
• Medium (initial 100ms, min 20ms, max 200ms).
• Low (initial 50ms, min 10ms, max 100ms).

SRTP Mode

Selects SRTP mode to use (“Disabled”, “Enabled but not forced”, or “Enabled and forced”). Default is Disabled.

It uses SDP Security Description to exchange key. Please refer to

SDES: https://tools.ietf.org/html/rfc4568

SRTP: https://www.ietf.org/rfc/rfc3711.txt

Crypto Life Time

Adds crypto life time header to SRTP packets. Default is Yes.

SLIC Setting

Depends on standard phone type (and location). Available options:

• USA 1 (BELLCORE 600 ohms)
• USA 2 (BELLCORE 600 ohms + 2.16uF)
• AUSTRALIA
• CHINA CO
• CHINA PBX
• EUROPEAN CTR21
• GERMANY
• INDIA/NEW ZEALAND
• JAPAN CO
• JAPAN PBX
• STANDARD 900 ohms
• UK

Caller ID Scheme

Selects the caller ID scheme. Available options:

• Bellcore/Telcordia
• ETSI-FSK during ringing
• ETSI-FSK prior to ringing with DTAS
• ETSI-FSK prior to ringing with LR+DTAS
• ETSI-FSK prior to ringing with RP
• ETSI-DTMF during ringing
• ETSI-DTMF prior to ringing with DTAS
• ETSI-DTMF prior to ringing with LR+DTAS
• ETSI-DTMF prior to ringing with RP
• SIN 227 – BT
• NTT JAPAN
• DTMF Denmark prior to ringing with no DTAS no LR
• DTMF Denmark prior to ringing with LR
• DTMF Sweden/Finland prior to ringing with LR
• DTMF Brazil
• DTMF-FSK Brazil

DTMF Caller ID

Defines the start and stop tones as delimiters for the caller ID.

Start Tone and Stop Tone can be set to “Default”, “A”, “B”, “C”, “D” or “#”

Polarity Reversal

Reverses the polarity upon call establishment and termination. Default is No.

Loop Current Disconnect

Allows the traditional PBX used with HT813 to apply this method for signaling call termination. Method initiates short voltage drop on the line when remote (VoIP) side disconnects an active call. Default is No.

Play busy/reorder tone before Loop Current Disconnect

Allow user to configure if it will play busy/reorder tone before loop current disconnect upon call fail. Default is No.

Loop Current Disconnect Duration

Configures the duration of voltage drop described in topic above. HT813 supports a duration range from 100 to 10000 ms. Default value is 200.

Enable Pulse Dialing

Allow users to enable Pulse Dialing option under FXS Port. Default is No.

Pulse Dialing Standard

This feature allows users to use Swedish pulse dialing standard or New Zealand pulse dialing standard. Default is General Standard.

Enable Hook Flash

Enables the FLASH button to be used for terminating calls. Default is Yes.

Hook Flash Timing

Defines the time period when the cradle is pressed (Hook Flash) to simulate FLASH. To prevent unwanted activation of the Flash/Hold and automatic phone ring-back, adjust this time value. HT813 supports a range from 40 to 2000 ms.

Default values are 300 minimum and 1100 maximum.

On Hook Timing

Specifies the on-hook time for an on-hook event to be validated. HT813 supports a range from 40 to 2000 ms. Default value is 400.

Gain

Adjusts the voice path volume.

• Rx is a gain level for signals transmitted by FXS

• Tx is a gain level for signals received by FXS.

Default = 0dB for both parameters. Loudest volume: +6dB Lowest volume: -6dB.

User can adjust volume of call using the Rx gain level parameter and the Tx gain level parameter located on the FXS port configuration page. If call volume is too low when using the FXS port (i.e. the ATA is at user site), adjust volume using the Rx gain level parameter under the FXS port configuration page. If voice volume is too low at the other end, user may increase the far end volume using the Tx gain level parameter under the FXS port configuration page.

Disable Line Echo Canceller (LEC)

Disables the LEC will per call base. Recommended for FAX/Data calls.

Default is No.

Disable Network Echo Suppressor

Disables the NEC will per call base. Recommended for FAX/Data calls.

Default is No.

Outgoing Call Duration Limit

Defines the call duration limit for the outgoing calls, Default is 0 (No limit).

Enable High Ring Power

Configures a high ringing voltage output for the FXS port of HT813.

RFC2833 Events Count

This feature allows users to customize the count of RFC2833 events. Supported range is 2-10. Default is 8.

RFC2833 End Events Count

This feature allows users to customize the count of RFC2833 end events. Supported range is 2-10. Default is 3.

Distinctive Ring Tone

Ring tones

Configures the ring tone cadence preferences. User has 10 choices. The configuration completed in Distinctive Ring Tones block in the same page, applies to ring tones cadences configured here.

Account Active

Activates / Deactivates the accounts. The FXO port configuration will not change if disabled, although the port will not be operational, in this state, there will be no dial tone when picking up the analog phone and making/receiving calls will not be possible.

Primary SIP Server

Configures SIP server IP address or domain name provided by VoIP service provider. This is the primary SIP server used to send/receive SIP messages from/to HT813.

Failover SIP Server

Specifies failover SIP server IP address or domain name provided by VoIP service provider. This server will be used if the primary SIP server becomes unavailable.

Prefer Primary SIP Server

Selects to prefer primary SIP server. The account will register to primary Server if registration with Failover server expires. Default is No.

Outbound Proxy

Specifies IP address or domain name of outbound Proxy, or media gateway, or session border controller. Used by HT813 for firewall or NAT penetration in different network environments. If symmetric NAT is detected, STUN will not work and only outbound proxy can correct the problem.

Backup Outbound Proxy

Configures the backup outbound proxy to be used when the “Outbound Proxy” registration fails. By default, this field is left empty.

Prefer Primary Outbound Proxy

If the user configures this option to “Yes”, when registration expires, the device will re-register via primary outbound proxy. By default, this option is disabled.

SIP Transport

Selects transport protocol for SIP packets; UDP or TCP or TLS. Please make sure your SIP Server or network environment supports SIP over the selected transport method. Default is UDP.

NAT Traversal

Indicates type of NAT for each account. This parameter configures whether the NAT traversal mechanism is activated. Users could select the mechanism from No, Keep-alive, STUN, UPnP. Default setting is No.

SIP User ID

Defines user account information provided by VoIP service provider (ITSP). Usually in the form of digit similar to phone number or actually a phone number.

Authenticate ID

Determines account authenticate ID provided by VoIP service provider (ITSP). Can be identical to or different from “SIP user ID”.

Authenticate Password

Specifies account password provided by VoIP service provider (ITSP) to register to SIP servers.

Name

Chooses a name to be associated to user.

DNS Mode

Selects DNS mode to use for the client to look up server. One mode can be chosen.

• A Record: resolves IP Address of target according to domain name.
• SRV: DNS SRV resource records indicate how to find services for various protocols.
• NAPTR/SRV: Naming Authority Pointer according to RFC 2915.

Default is A Record.

DNS SRV use Registered IP

When this option is set to “Yes”, when the HT is registered on second SRV and makes an outbound call, it will try the second SRV (registered IP) first.

By default, this option is disabled and the DNS SRV will use first SRV instead of the registered IP.

Tel URI

Indicates E.164 number in “From” header by adding “User=Phone” parameter or using “Tel:” in SIP packets, if the HT813 has an assigned PSTN Number.

• Disabled: Use “SIP User ID” information in the Request-Line and “From” header.
• User=Phone: “User=Phone” parameter will be attached to the Request-Line and “From” header in the SIP request to indicate the E.164 number. If set to “Enable”.
• Enabled: “Tel:” will be used instead of “sip:” in the SIP request.

Please consult your carrier before changing this parameter. Default is Disabled.

SIP Registration

Controls whether the HT813 needs to send REGISTER messages to the proxy server. Default setting is Yes.

Unregister on Reboot

Controls whether to clear SIP user’s information by sending un-register request to the proxy server. The un-registration is performed by sending a REGISTER message with Contact set to * and Expires=0 parameters to the SIP server. This will unregister the SIP account under the concerned FXO page. Default is No.

Outgoing Call Without Registration

Enables the ability to place outgoing calls even if the account is not registered (if allowed by ITSP); device will not be able to receive incoming calls. Default is No.

Register Expiration

Refreshes registration periodically with specified SIP proxy (in minutes). Maximum interval is 65535 minutes (about 45 days). Default is 60 minutes (or 1 hour).

Reregister Before Expiration

Sends re-register request after specific time (in seconds) to renew registration before the previous registration expires.

SIP Registration Failure Retry Wait Time

Sends re-register request after specific time (in seconds) when registration process fails. Maximum interval is 3600 seconds (1 hour). Default is 20 seconds.

SIP Registration Failure Retry Wait Time upon 403 Forbidden

Sends re-register request after specific time (in seconds) when registration process fails with error 403 Forbidden. Maximum interval is 3600 seconds (1 hour).

Default is 1220 seconds.

Enable SIP OPTIONS Keep Alive

Enables SIP OPTIONS to track account registration status so the phone adapter will send periodic OPTIONS message to server to track the connection status with the server. Default setting is No.

SIP OPTIONS Keep Alive Interval

Configures the time interval when the phone adapter sends OPTIONS message to SIP server. The default setting is 30 seconds, which means the phone adapter will send an OPTIONS message to the server every 30 seconds. The default range is 1-64800.

SIP OPTIONS Keep Alive Max Lost

Defines the Number of max lost packets for SIP OPTIONS Keep Alive before re-registration. Between 3-10, default is 3.

Layer 3 QoS

Defines Diff-Serv values for SIP and RTP. Defaults are:

SIP DSCP: 26

RTP DSCP: 46

Local SIP Port

Defines local port to use by the HT813 for listening and transmitting SIP packets. Default value for FXO port is 5062.

Local RTP Port

Defines the local RTP-RTCP port pair the HT813 will listen and transmit. It is the HT813 RTP port for channel 0. The default value for FXS port is 5012.

Use Random SIP Port

Controls whether to use configured or random SIP ports. This is usually necessary when multiple HT813 are behind the same NAT. Default is No.

Use Random RTP Port

Controls whether to use configured or random RTP ports. This is usually necessary when multiple HT813 are behind the same NAT. Default is No.

Enable RTCP

Allows users to enable RTCP. Default setting is Yes.

Remove OBP from Route Header

Removes outbound proxy info in “Route” header when sending SIP packets.

Default is No.

Support SIP Instance ID

Includes “SIP Instance ID” attribute to “Contact” header in REGISTER request as defined in IETF SIP outbound draft. Default is No.

Validate Incoming SIP Message

Validates incoming messages. Default is No.

Check SIP User ID for Incoming INVITE

Checks SIP User ID in the Request URI of incoming INVITE; if it does not match the HT813 SIP User ID, the call will be rejected. Direct IP calling will also be disabled. Default is No.

Authenticate Incoming INVITE

Challenges the incoming INVITE for authentication with SIP 401 Unauthorized message. Default is No.

Authenticate server certificate domain

Configures whether to validate the domain certificate when download the firmware/config file. If it is set to “Yes”, the phone will download the firmware/config file only from the legitimate server. The default setting is “No“.

Authenticate server certificate chain

Configures whether to validate the server certificate when download the firmware/config file. If it is set to “Yes”, the phone will download the firmware/config file only from the legitimate server. The default setting is “No“.

Trusted CA Certificates

Uses the certificate for Authentication if “Check Domain Certificates” is set to “Yes” under “Account”🡪“SIP Settings”.

Allow Incoming SIP Messages from SIP Proxy Only

Checks SIP address of the Request URI in the incoming SIP message; if it does not match the SIP server address of the account, the call will be rejected. Default is No.

Use Privacy Header

Determines if the “Privacy header” will be presented in the SIP INVITE message and if it includes the caller info in this header. If set to Default, it will add Privacy header unless special feature is Telkom SA or CBCOM. Default is Default.

Use P-Preferred-Identity Header

Specifies if the P-Preferred-Identity Header will be presented in the SIP INVITE message. If set to “default”, the P-Preferred-Identity Header will be omitted in SIP INVITE message when Telkom SA or CBCO is active. If set to “Yes”, the P-Preferred-Identity Header will always be presented. If set to “No”, it will be omitted.

Default setting is: Default.

Use P-Access-Network-Info Header

With this feature enabled, device will populate the WAN access node with IEEE802.11a, IEEE-802.11b in P-Access-Network-Info SIP header.

Use P-Emergency-Info Header

This feature support of IEEE-48-addr and IEEE-EUI-64 in SIP header for emergency calls.

SIP REGISTER Contact Header Uses

Specifies which address (LAN or WAN address) the device will detect to use it in SIP Register Contact Header. When set to LAN, Contact header will include local IP from ATA in REGISTER messages, while if set to WAN, host/port/contact will be updated from SIP 401/403/404/407 Via header “received”/”rport” parameters in REGISTER messages. Default is LAN Address.

Allow SIP Factory Reset

This feature allows user to reset the devices directly through SIP Notify. Default is No.

SIP T1 Timeout

Defines T1 timeout value.

It is an estimate of the round-trip time between the client and server transactions.

For example, the HT813 will attempt to send a request to a SIP server.

The time it takes between sending out the request to the point of getting a response is the SIP T1 timer. If no response is received the timeout is increased to (2*T1) and then (4*T1). Request re-transmit retries would continue until a maximum amount of time defined by T2. Default is 0.5 seconds.

SIP T2 Interval

Identifies maximum retransmission interval for non-INVITE requests and INVITE responses. Retransmitting and doubling of T1 continues until it reaches T2 value. Default is 4 seconds.

SIP Timer D

Configure the SIP Timer D defined in RFC3261. 0 – 64 seconds. Default 0

DTMF Payload Type

Defines payload type for DTMF using RFC2833.

Preferred DTMF method (in order)

Sorts DTMF methods (in-audio, via RTP (RFC2833) or via SIP INFO) by priority.

Disable DTMF Negotiation

Uses above DTMF order without negotiation.

Default is No.

Generate Continuous RFC2833 Events

When enabled the RFC2833 events are generated until key is released.

Default is No.

Flash Digit Control

When it set to YES it allows the user to perform some call setting when both channels are used while pressing:

• “Flash + 1” in order to hang up the current call and resume a call that was held.
• “Flash + 2” in order to hold the current call and resume a call that was held.
• “Flash + 3” in order to perform 3-way conference.
• “Flash + 4” in order to perform attended transfer.

Note: Please refer to the user guide for detailed steps to perform above operations.

Proxy-Require

Determines a SIP Extension to notify the SIP server that the HT813 is behind a NAT/Firewall.

Use NAT IP

Defines NAT IP address used in SIP/SDP messages. It should only be used if required by ITSP.

Use SIP User-Agent Header

Configures the SIP User-Agent Header.

SIP User-Agent

Configures SIP User-Agent. If not configured, device will use the default User Agent Header. The value range is 1024 to Maximum String Length. Default value is Null.

Do Not Escape ‘#’ as %23 in SIP URI

Replaces # by %23 in some special situations. Default is No.

Disable Multiple m Line in SDP

Sends only one m line in SDP, regardless of how many m fields are in the incoming SDP. Default is No.

Ring Timeout

Stops ringing when incoming call if not answered within a specific period of time. Default is 60 seconds. When configure the Ring Timeout to 0, will have no ring timeout.

Early Dial

Sends an early INVITE each time a key is pressed when a user dials a number. Otherwise, only one INVITE is sent after full number is dialed (user presses Dial Key or after “no key entry timeout” expires).

This option should be used only if there is a SIP proxy is configured and supporting 484 responses (Incomplete Address). Otherwise, the call will likely be rejected by the proxy (with a 404 Not Found error).

Default is No.

This feature is NOT designed to work with and should NOT be enabled for direct IP-to-IP calling.

Dial Plan Prefix

Adds specified prefix to dialed number.

Use # as Dial Key

Treats “#” as the “Send” (or “Dial”) key. If set to “No”, this “#” key can be included as part of the dialed number.

Default is Yes.

Dial Plan

Dial Plan Rules:

1. Accept Digits: 1,2,3,4,5,6,7,8,9,0 , *, #, A,a,B,b,C,c,D,d
2. Grammar: x – any digit from 0-9;
   1. xx+ – at least 2 digits number;
   2. xx – exactly 2 digits number;
   3. ^ – exclude;
   4. . – wildcard, matches one or more characters
   5. [3-5] – any digit of 3, 4, or 5;
   6. [147] – any digit 1, 4, or 7;
   7. <2=011> – replace digit 2 with 011 when dialing
   8. < =1> – add a leading 1 to all numbers dialed, vice versa will remove a 1 from the number dialed
   9. | – or

• Example 1: {[369]11 | 1617xxxxxxx} –

Allow 311, 611, 911, and any 10-digit numbers of leading digits 1617

• Example 2: {^1900x+ | <=1617>xxxxxxx} –

Block any number with leading digits 1900 and add prefix 1617 for any dialed 7-digit numbers

• Example 3: {1xxx[2-9]xxxxxx | <2=011>x+} –

Allow any length of number with leading digit 2 and 10 digit-numbers of leading digit 1 and leading exchange number between 2 and 9; If leading digit is 2, replace leading digit 2 with 011 before dialing.

1. Default: Outgoing – {x+}

Example of a simple dial plan used in a Home/Office in the US:

{ ^1900x. | <=1617>[2-9]xxxxxx | 1[2-9]xx[2-9]xxxxxx | 011[2-9]x. | [3469]11 }

Explanation of example rule (reading from left to right):

• ^1900x. – prevents dialing any number started with 1900
• <=1617>[2-9]xxxxxx – allows dialing to local area code (617) numbers by dialing 7 numbers and 1617 area code will be added automatically
• 1[2-9]xx[2-9]xxxxxx – allows dialing to any US/Canada Number with 11 digits length
• 011[2-9]x. – allows international calls starting with 011
• [3469]11 – allow dialing special and emergency numbers 311, 411, 611 and 911

Note: In some cases, user wishes to dial strings such as *123 to activate voice mail or other application provided by service provider. In this case * should be predefined inside dial plan feature. As an example { *x+ } will allow to dial * followed by any length of numbers.

SUBSCRIBE for MWI

Sends SUBSCRIBE periodically (depends on “Register Expiration” parameter) for message waiting indication. Default is No.

Anonymous Call Rejection

Rejects incoming calls with anonymous caller ID with “486 Busy here” message.

Default is No.

Special Feature

Selects Soft switch vendors’ special requirements Example of vendors: Standard, Broadsoft, CBCOM, RNK, Huawei, China Mobile, ZTE IMS, PhonePower, TELKOM SA, Vonage, Metaswitch, CenturyLink, MTS. Default is Standard.

Session Expiration

Enables SIP sessions to be periodically “refreshed” via a SIP request (UPDATE, or re-INVITE). When the session interval expires, if there is no refresh via an UPDATE or re-INVITE message, the session will be terminated. Session Expiration is the time (in seconds) at which the session is considered timed out if no successful session refresh transaction occurs beforehand. Default is 180 seconds.

Min-SE

Defines Minimum session expiration (in seconds). Default is 90 seconds.

Caller Request Timer

Uses session timer when making outbound calls if remote party supports it.

Default is No.

Callee Request Timer

Uses session timer when receiving inbound calls with session timer request.

Default is No.

Force Timer

Uses session timer even if the remote party does not support this feature. Selecting “No” will enable session timer only when the remote party supports it. Default is No.

To turn off Session Timer, select “No” for Caller and Callee Request Timer, and Force Timer.

UAC Specify Refresher

Specifies which end will act as refresher for outgoing calls.

• UAC: The handy tone acts as the refresher.
• UAS: Callee or proxy server act as the refresher.

Default is Omit.

UAS Specify Refresher

Specifies which end will act as refresher for incoming calls:

• UAS: The handy tone acts as the refresher.
• UAC: Callee or proxy server act as the refresher.

Default is Omit.

Force INVITE

Always refresh with INVITE instead of UPDATE Default is No.

INVITE Ring-No-Answer Timeout (sec)

Between 5-300 seconds. Default 40 seconds.

Enable 100rel

Appends “100rel” attribute to the value of the required header of the initial signaling messages. Default is No.

Add Auth Header on Initial REGISTER

Adds “Authentication” header with blank “nonce” attribute in the initial SIP REGISTER request. Default is No.

Use First Matching Vocoder in 200OK SDP

Includes only the first matching vocoder in its 200OK response, otherwise it will include all matching vocoders in same order received in INVITE. Default is No.

Preferred Vocoder

Configures vocoders in a preference list (up to 7 preferred vocoders) that will be included with same order in SDP message. Vocoder types are G.711 A-/U-law, G.726-32, G.723, G.729, iLBC and OPUS.

Voice Frames per TX

Transmits a specific number of voice frames per packet. Default is 2; increases to 10/20/32/64 for G711/G726/G723/other codecs, respectively.

G723 Rate

Operates at specified encoding rate for G.723 vocoder. Available encoding rates are 6.3kbps or 5.3kbps. Default is 6.3kbps.

iLBC Frame Size

Specifies iLBC packet frame size (20ms or 30ms).

Default is 20ms.

Disable OPUS Stereo in SDP

Disables OPUS stereo in SDP. Default is No.

iLBC Payload type

Determines payload type for iLBC. Valid range is between 96 and 127. Default is 97.

OPUS Payload Type

Determines payload type for OPUS. Valid range is between 96 and 127. Default is 123.

VAD

Allows detecting the absence of audio and conserves bandwidth by preventing the transmission of “silent packets” over the network. Default is No.

Symmetric RTP

Changes the destination to send RTP packets to the source IP address and port of the inbound RTP packet last received by the device. Default is No.

Fax Mode

Specifies the fax mode: T.38 (Auto Detect) FoIP by default, or Pass-Through (must use codec PCMU/PCMA)

Re-Invite after Fax Tone Detected

Allows the unit to send out the re-INVITE for T.38 or Fax Pass Through if a fax tone is detected. Default is Enabled

Jitter Buffer Type

Selects jitter buffer type (Fixed or Adaptive) based on network conditions.

Jitter Buffer Length

• High (initial 200ms, min 40ms, max 600ms) Note: not all vocoders can meet the high requirement.
• Medium (initial 100ms, min 20ms, max 200ms).
• Low (initial 50ms, min 10ms, max 100ms).

SRTP Mode

Selects SRTP mode to use (“Disabled”, “Enabled but not forced”, or “Enabled and forced”). Default is Disabled.

It uses SDP Security Description to exchange key. Please refer to

SDES: https://tools.ietf.org/html/rfc4568

SRTP: https://www.ietf.org/rfc/rfc3711.txt

Crypto Life Time

Adds crypto life time header to SRTP packets. Default is Yes.

Caller ID Scheme

Selects the caller ID scheme. Available options:

• Bellcore/Telcordia
• ETSI-FSK during ringing
• ETSI-FSK prior to ringing with DTAS
• ETSI-FSK prior to ringing with LR+DTAS
• ETSI-FSK prior to ringing with RP
• ETSI-DTMF during ringing
• ETSI-DTMF prior to ringing with DTAS
• ETSI-DTMF prior to ringing with LR+DTAS
• ETSI-DTMF prior to ringing with RP
• SIN 227 – BT
• NTT JAPAN
• DTMF Denmark prior to ringing with no DTAS no LR
• DTMF Denmark prior to ringing with LR
• DTMF Sweden/Finland prior to ringing with LR
• DTMF Brazil
• DTMF-FSK Brazil

DTMF Caller ID

Defines the start and stop tones as delimiters for the caller ID.

Start Tone and Stop Tone can be set to “Default”, “A”, “B”, “C”, “D” or “#”

FSK Caller ID Minimum RX Level (dB)

An adjustable value for the Caller ID signal to help this device to recognize Caller ID from different networks. Range: -96 to -0dB. Default –40dB.

FSK Caller ID Seizure Bits

Range is from 0 to 800 bits. Default 70.

FSK Caller ID Mark Bits

Range is from 0 to 800 bits. Default 40.

Caller ID Transport Type

According to customer’s choice CID information will be transferred from PSTN network to VoIP network using following rules:

• Relay via SIP From – PSTN CID is in the SIP From field
• Relay via P-Asserted-Identity – SIP From field uses the pre-configured account user Id. PSTN CID is in the P-Asserted-Identity field
• Relay via P-Preferred-Identity – PSTN CID is in the P-Preferred-Identity field
• Send anonymous – SIP From field uses “anonymous”. PSTN CID is put in the P-Asserted-Identity field
• Disable – PSTN CID will not be sent. SIP From field uses the pre-configured account user ID

Send Hook Flash to PSTN

If Yes, hook flash will be sent to PSTN upon receiving flash event from RFC2833 or SIP INFO.

Hook Flash Duration (ms)

The time period when the cradle is pressed (Hook Flash) to simulate a FLASH. Adjust this time value to prevent unwanted activation of the Flash/Hold and automatic phone ring-back.

Gain

Voice path volume adjustment.

• RX is a gain level for signals transmitted by FXO (FXO-To-VoIP volume)

• TX is a gain level for signals received by FXO (FXO-To-PSTN volume).

Default = 0dB for both parameters. Loudest volume: +6dB; Lowest volume: -6dB.

User can adjust volume of call on either end using the Rx Gain Level parameter and the TX Gain Level parameter located on the FXO Port Configuration page. These parameters affect call volume ONLY for calls placed to/from PSTN and VoIP networks.

If call volume is too low when using VoIP extension, adjust volume using the Rx Gain Level parameter under the FXO Port Configuration page.

If voice volume is too low at the other end (PSTN side), user may increase the far end volume using the TX Gain Level parameter under the FXO Port Configuration page.

Disable Line Echo Canceller (LEC)

Disables the LEC will per call base. Recommended for FAX/Data calls. Default is No.

Disable Network Echo Suppressor

Disables the NEC will per call base. Recommended for FAX/Data calls. Default is No.

Outgoing Call Duration Limit

Defines the call duration limit for the outgoing calls, Default is 0 (No limit).

FXO Termination

Enable Current Disconnect

This value should be used in case the PSTN provider uses line power drop to indicate call completion to the end point. In this case the HT813 will search for a power drop.

Current Disconnect Threshold (ms)

This is a preconfigured value of duration for a line power drop used by specific service providers. For example, for a configured value of 500ms the device will ignore any random voltage drops on the line if duration of such drop is less than 500ms and the call will NOT be considered as terminated. This is useful to prevent unnecessary call drops in some low quality PSTN lines. Default is 100 ms. Range from 50 to 800 ms.

Enable PSTN Disconnect Tone Detection

If set to Yes, arrived Busy Tone is used as the disconnect signal.

PSTN Disconnect Tone

In certain countries, the central office will send a special busy tone to indicate when a call is disconnected from the remote side. User can pre-configure this tone on the ATA. The user should know the frequency values and cadences of these tones.

Here is an example for the syntax for a busy tone in the U.S.A:

Syntax: f1=freq@vol, f2=freq@vol, c=on1/off1-on2/off2-on3/off3;

Note: freq: 0 – 4000Hz; vol: -30 – 0dBm

Default: Busy Tone – f1=480@-32, f2=620@-32, c=500/500;

Note: Maximum supported cadences are 3

Enable Polarity Reversal

This should be set to Yes only if the FXO lines are subscribed to PR service from PSTN Service provider. It is merely a PR detect feature. Default is No.

Note: If there is no PR service from provider on the FXO line, and this setting is

configured to Yes, calls will not be successful.

AC Termination Model

You can select the AC termination by Country or by Impedance.

Default is Country-based.

Country-Based

15 Countries are selectable in this version of the F/W.

• USA
• AUSTRIA
• AUSTRALIA/NEW ZEALAND
• BELGIUM
• CHINA
• FINLAND
• FRANCE
• GERMANY
• GREECE
• ITALY
• JAPAN
• NORWAY
• SPAIN
• SWEDEN
• UK

Default is USA

Impedance-Based

Select the Impedance used by the PSTN service provider.

• 600R – 600 ohms
• 600C – 600 ohms + 2.16uF
• 900R – 900 ohms
• 900C – 900 ohms + 2.16uF
• COMPLEX1 – 220 ohms + (820 ohms || 115nF)
• COMPLEX2 – 270 ohms + (750 ohms || 150nF)
• COMPLEX3 – 370 ohms + (620 ohms || 310nF)
• COMPLEX4 – 600R, 270 ohms + (750 ohms || 150nF)
• COMPLEX5 – 320 ohms + (1050 ohms || 230nF)
• COMPLEX6 – 350 ohms + (1000 ohms || 210nF)
• COMPLEX7 – 200 ohms + (680 ohms || 100nF)
• COMPLEX8 – 370 ohms + (820 ohms || 110nF)
• COMPLEX9 – 275 ohms + (780 ohms || 115nF)
• COMPLEX10 – 120 ohms + (820 ohms || 110nF)

Default is 600R – 600 ohms

Number of Rings

The FXO port will ring the number of times configured in this field before sending the call to the VoIP side. The supported range is 1-50. Default is 4.

PSTN Ring Thru FXS

If Yes, the phone connected to the FXS port will ring a configured number of times (see above). If not, the phone connected to the FXS port will not ring.

PSTN Ring Thru Delay (sec)

If the PSTN Ring Thru Delay is set to Yes, all incoming PSTN calls through FXO will ring the phone connected to the FXS port, after this delay or after caller id is detected (whichever comes first).

PSTN Ring Timeout (sec)

Range is 2-10 seconds. Default is 6 seconds. Option is used to detect PSTN hang up when FXO port is not answered.

PSTN Idle Wait Timeout between Outgoing Calls

Used to customize timeout value between PSTN outgoing calls. Range is 0-10 seconds. Default is 4 seconds.

Channel Dialing

DTMF Digit Length (ms)

Digit length and Dial Pause are port digit dialing configurations; FXO needs to dial out digits for VoIP to PSTN 1 stage calls, and unconditional call forward to PSTN, and route to PSTN. Digit Length is the play time for each digit.

Note: In order to receive the caller ID information, the delay should be set to a value larger than the delay required to complete the PSTN caller ID delivery.

DTMF Dial Pause (ms)

Dial pause is the time between 2 digits for the same scenario as explained above.

First Digit Timeout (sec)

Used for PSTN to VoIP calls. PSTN users need to enter the FIRST digit within the first digit timeout period. Otherwise the call will be dropped.

Inter-Digit Timeout

(sec)

When dialing from the PSTN to VoIP, subsequent digits must be input within the period of inter-digit timeout. Otherwise the dial plan thinks it is the end of the digit input.

Wait for Dial Tone

Wait for Dial tone is used for one stage VoIP to PSTN calls. If set to Yes, the device will first obtain a PSTN line and a dial tone from a central office. After obtaining the dial tone, the digits dialed will be sent to the central office.

Stage Method (1/2)

This configuration is applicable for VoIP to PSTN calls and indicates one or two stage dialing methods.

Min Delay Before Dial PSTN Number

The time to wait before HT813 initiates the call via PSTN line.

Default 500ms, range is from 50 to 65000ms.

*02

Forcing a Codec (per call) *027110 (PCMU), *027111 (PCMA), *02723 (G723), *02729 (G729), *027201 (iLBC).

*03

Disable LEC (per call) Dial “*03” +” number”.

No dial tone is played in the middle.

*16

Enable SRTP

*17

Disable SRTP

*30

Block Caller ID (for all subsequent calls)

*31

Send Caller ID (for all subsequent calls)

*47

Direct IP Calling. Dial “*47” + “IP address”. No dial tone is played in the middle.

*50

Disable Call Waiting (for all subsequent calls)

*51

Enable Call Waiting (for all subsequent calls)

*67

Block Caller ID (per call). Dial “*67” +” number”. No dial tone is played in the middle.

*82

Send Caller ID (per call). Dial “*82” +” number”. No dial tone is played in the middle.

*69

Call Return Service: Dial *69 and the phone will dial the last incoming phone number received.

*70

Disable Call Waiting (per call). Dial “*70” +” number”. No dial tone is played in the middle.

*71

Enable Call Waiting (per call). Dial “*71” +” number”. No dial tone is played in the middle

*72

Unconditional Call Forward: Dial “*72” and then the forwarding number followed by “#”. Wait for dial tone and hang up. (dial tone indicates successful forward)

*73

Cancel Unconditional Call Forward. To cancel “Unconditional Call Forward”, dial “*73”, wait for dial tone, then hang up.

*74

Enable Paging Call: Dial “*74” and then the destination phone number you want to page.

*78

Enable Do Not Disturb (DND): When enabled all incoming calls are rejected.

*79

Disable Do Not Disturb (DND): When disabled, incoming calls are accepted.

*87

Blind Transfer

*90

Busy Call Forward: Dial “*90” and then the forwarding number followed by “#”. Wait for dial tone then hang up.

*91

Cancel Busy Call Forward. To cancel “Busy Call Forward”, dial “*91”, wait for dial tone, then hang up.

*92

Delayed Call Forward. Dial “*92” and then the forwarding number followed by “#”. Wait for dial tone then hang up.

*93

Cancel Delayed Call Forward. To cancel Delayed Call Forward, dial “*93”, wait for dial tone, then hang up

Flash/
Hook

Toggles between active call and incoming call (call waiting tone). If not in conversation, flash/hook will switch to a new channel for a new call.

#

Pressing pound sign will serve as Re-Dial key.

Key

Mapping

0-9

0-9

A

22 (press the “2” key twice, “A” will show on the LCD)

B

222

C

2222

D

33 (press the “3” key twice, “D” will show on the LCD)

E

333

F

3333