This document is contains links to Grandstream GUI Interfaces. Please remember to download these examples http://www.grandstream.com/user_manuals/GUI/GUI_HT488.rar for your reference. This document is subject to change without notice. The latest electronic version of this user manual is available for download from the following location: http://www.grandstream.com/user_manuals/HT4488_User_Manual.pdf Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for any purpose without the express written permission of Grandstream Networks, Inc. is not permitted.
HT488 User Manual Firmware 1.0.3.86 Page 4 of 31 Last Updated: 7/2007

INSTALLATION

EQUIPMENT PACKAGING
The HT488 ATA package contains: One HT488 Main Case One Universal Power Adaptor One Ethernet Cable

CONNECTING YOUR ATA

HT488 Analog Telephone Adaptor is an all-in-one VoIP integrated device designed to be a total solution for networks providing VoIP services. All of the HT488 VoIP features and functions are available via a regular analog telephone.
FIGURE 1: CONNECTING THE HT488
RJ45 10M Ethernet LAN - WAN

RJ11 FXS Port (Phone)

+5V/1200mA

BUTTON RED LED GREEN LED

RJ11 FXO Port (Phone Line)
TABLE 1: DEFINITIONS OF THE HT488 CONNECTORS +5V/1.2A LAN Port (RJ-45) WAN Port (RJ-45) PHONE (RJ-11) LINE (RJ-11) BUTTON Power adapter connection Connect the LAN port with an Ethernet cable to your PC. Connect to the internal LAN network or router. FXS port to be connected to analog phones / fax machines. FXO port should be connected to the PSTN line Button and two colors led indicator.
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FIVE EASY STEPS TO INSTALL THE HT488
The HT488 is designed for easy configuration and easy installation. Detailed configuration instructions are located in the CONFIGURATION section. 1. Connect a standard touch-tone analog telephone to the PHONE port. 2. Insert a standard RJ11 telephone cable into the LINE port and connect the other end of the telephone cable to a wall jack. 3. Insert the Ethernet cable into the WAN port of HT488 and connect the other end of the Ethernet cable to an uplink port (a router or a modem, etc). 4. Connect a PC to the LAN port of HT488 if HT488 is used as a router. 5. Insert the power adapter into the HT488 and connect it to a wall outlet.

Call Handling Features Network and Provisioning Fax over IP Security
TABLE 3: HT488 HARDWARE SPECIFICATION LAN interface WAN interface FXS port FXO port (PSTN Port) Button LED Universal Switching Power Adaptor Dimension Weight Temperature Humidity Compliance 1xRJ45 10Mbps 1xRJ45 10Mbps 1 x FXS 1x PSTN pass-through and life line port 1 Dual color (green/red) Input: 100-240VAC 50-60 Hz / Output: +5VDC, 1200mA UL certified 70mm (W) x 130mm (D) x 27mm (H) 0.6lbs (0.3kg) 40 - 130F / 5 45C 10% - 90% (non-condensing)
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BASIC OPERATIONS

GET FAMILIAR WITH VOICE PROMPT
HT488 has a stored voice prompt menu for quick browsing and simple configuration. The voice prompt menu and the LED button is designed for the FXS port ONLY. Press the button or *** from the analog phone to enter the IVR menu. TABLE 4: HT488 IVR MENU DEFINITIONS
Menu Main Menu Voice Prompt Enter a Menu Option Options Press * for the next menu option Press # to return to the main menu Enter 01-05, 07,12-17,47 or 99 menu options 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. The current WAN IP address is announced If using Static IP Mode, enter 12 digit new IP address. Same as menu 02 Same as menu 02 Same as menu 02 Press 9 to move to the next selection in the list: PCM U / PCM A G.723 G.729 G.726 iLBC Press 9 to toggle between enable / disable Announces current Firmware Server IP address. Enter 12 digit new IP address. IP Announces current Config Server Path IP address. Enter 12 digit new IP address. Upgrade protocol for firmware and configuration update. Press 9 to toggle between TFTP / HTTP Firmware version information. Firmware upgrade mode. Press 9 to toggle among the following three options: - always check - check when pre/suffix changes - never upgrade Enter a 12 digit IP address to make a direct IP call, after dial tone. (See Make a Direct IP Call.) Press 9 to reboot the device; or Enter encoded MAC address to restore factory default setting (See Restoring Factory Settings) Automatically returns to main menu
DHCP Mode, Static IP Mode
IP Address + IP address Subnet + IP address Gateway + IP address DNS Server + IP address Preferred Vocoder
WAN Port Web Access Firmware Server IP Address Configuration Address Upgrade Protocol Firmware Version Firmware Upgrade Server

Direct IP Calling RESET

Invalid Entry
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NOTE:

* shifts down to the next menu option # returns to the main menu 9 functions as the ENTER key in many cases to confirm an option All entered digit sequences have known lengths - 2 digits for menu option and 12 digits for IP address. For IP address, add 0 before the digits if the digits are less than 3 (like 192.168.0.26 should be key in like 192168000026, no dot needed while input). Once all of the digits are collected, the input will be processed. Key entry can not be deleted but the phone may prompt error once it is detected

MAKE PHONE CALLS

CALLING PHONE OR EXTENSION NUMBERS There are currently two methods to make an extension number call: a) Dial the numbers directly and wait for 4 (default) seconds. b) Dial the numbers directly, and press # (assuming that use # as dial key is selected in web configuration). Examples: To dial another extension on the same proxy, such as 1008, simply pick up the attached phone, dial 1008 and then press the # or wait for 4 seconds. To dial a PSTN number such as 6266667890, you might need to enter in some prefix number followed by the phone number. Please check with your VoIP service provider to get the information. If you phone is assigned with a PSTN-like number such as 6265556789, most likely you just follow the rule to dial 16266667890 as if you were calling from a regular analog phone of North America, then followed by pressing # or wait for 4 seconds. DIRECT IP CALLS Direct IP calling allows two parties, that is, a HT with an analog phone and another VoIP Device, to talk to each other in an ad hoc fashion without a SIP proxy. This kind of VoIP calls can be made between two parties if: Both HT488 and other VoIP Device(i.e., another HT ATA or Budgetone SIP phone or other VoIP unit) have public IP addresses, or Both HT488 and other VoIP Device are on the same LAN using private IP addresses, or Both HT488 and other VoIP Device can be connected through a router using public or private IP addresses (with necessary port forwarding or DMZ).
To make a direct IP call, first pick up the analog phone or turn on the speakerphone on the analog phone, then access the voice menu prompt by dial *** or press the button on the HT488, and dials 47 to access the direct IP call menu. User will hear a voice prompt Direct IP Calling and a dial tone. Enter a 12-digit target IP address to make a call. Destination ports can be specified by using *4 (encoding for :) followed by the port number. Examples: If the target IP address is 192.168.0.10, the dialing convention is Voice Prompt with option 47, then followed by pressing the # key if it is configured as a send key or wait for more than 5 seconds. If the target IP address/port is 192.168.1.20:5062, then the dialing convention is: Voice Prompt with option 47, then 192168001020*45062 followed by pressing the # key if it is configured as a send key or wait for 4 seconds.

NOTE: When making a direct IP call, the Use Random Port should set to NO.
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CALL HOLD This function is applicable on FXS port for VoIP calls only. While in conversation, pressing the flash button on the attached analogue phone (if the phone has that button) will put the remote end on hold. Pressing the flash button again will release the previously held party and the bi-directional media will resume. If no flash button, then on-off hook quickly (hook flash) will do the same thing but also risk of losing call if the time is not short enough. CALL WAITING This function is applicable on FXS port for VoIP calls only. If call waiting feature is enabled, while the user is in a conversation, he will hear a special stutter tone if there is another incoming call. User can press the flash button to put the current call party on hold and switch to the other call. Pressing flash button toggles between two active calls. The HT488 also provides CWCID (call waiting caller ID) information which includes caller ID information in addition to the special stutter tone. The analog phone must support this feature for it to work on the HT488. Both call waiting functions (call waiting and CWCID) are activated and deactivated from the configuration pages menu. CALL TRANSFER Blind Transfer This function is applicable on FXS port for VoIP calls only. Assume that call party A and B are in conversation. A wants to Blind Transfer B to C: 1. A press FLASH on the analog phone to hear the dial tone. 2. Then A dials *87 then dials Cs number, and then # 3. A can hang up. NOTE: Enable Call Feature has to be set to Yes in web configuration page. Three situations can follow the transfer: 1. A quick confirmation tone (temporarily using the call waiting indication tone) followed by a dial tone. This indicates the transfer is successful (transferee has received a 200 OK from transfer target). At this point, A can either hang up or make another call. 2. A quick busy tone followed by a restored call (on supported platforms only). This means the transferee has received a 4xx response for the INVITE and we will try to recover the call. The busy tone is just to indicate to the transferor that the transfer has failed. 3. Busy tone keeps playing. This means we have failed to receive the second NOTIFY from the transferee and decided to time out. Note: this does not indicate the transfer has been successful, nor does it indicate the transfer has failed. When transferee is a client that does not support the second NOTIFY (such as our own earlier firmware), this will be the case. In bad network scenarios, this could also happen, although the transfer may have been completed successfully. Attended Transfer This function is applicable on FXS port for VoIP calls only. Assume that call party A and B are in conversation. A wants to Attend Transfer B to C: 1. A presses FLASH on the analog phone to get a dial tone 2. A then dial Cs number followed by #. 3. If C answers the call, A and C are in conversation. Then A can hang up to complete transfer. 4. If C does not answer the call, A can press flash back to talk to B.

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To make a VoIP-to-PSTN call: 1. Dial the FXO SIP account phone number to establish the VoIP session. The caller will hear the ring back tone once. Then the caller hears either a special continuous tone or a dial tone. The special continuous tone is played if the pin code is configured, or the dial tone otherwise. 2. Enter in the pin code that is configurable on the configuration page. The caller will hear the dial tone and get connected to the PSTN line if the pin code is valid, otherwise the continuous tone is played again to prompt caller to enter in the pin code again. The use may try up to 3 times to enter in pin code, if none is valid, HT488 will hang up. 3. After the caller hears dial tone from PSTN line, the caller can start dialing number to make calls. 4. The user can hit the # key to identify the end of the pin code or wait 4 seconds for a new dial tone and then dialing the PSTN number.
NOTE: Users can choose whether apply password protection for VoIP-to-PSTN calls or not. A PIN (Pin for PSTN calls) consists of up to 8 numeric digits can be configured through BASIC SETTINGS of the web configuration page. By default, there is no password protection, i.e. there is no authentication required for callers on the use of PSTN line through HT488. When a PIN is configured for VOIP-to-PSTN call flow, the VoIP device that calls into the HT488 FXO account needs to configure RFC2833 or SIP Info for DTMF digit transmission. Upon hearing the special continuous tone for PIN code input, if the caller dont enter any digit, HT488 will time out and hang up the call in 10 seconds. During any stage of DTMF digits input, a 4 seconds timeout is applied to serve as an end of PIN or destination number input. Users may also use the # key to indicate the end of an input. On the web configuration page, if the Forward to PSTN is configured, the second stage dialing is eliminated, i.e., after dialing into the FXO SIP account number, the PSTN number will be called automatically.
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PSTN-TO-VOIP CALLS This function is applicable on FXO port that functions as a bridge between VoIP and PSTN. The user can make VoIP calls remotely by dialing into FXO Line port on HT488. To make a PSTN-to-VoIP call: 1. Make an incoming call to the PSTN line on FXO port. The attached analog phone will ring for 4 times by default, this setting is configurable on the configuration page. 2. If no one picks up the phone on FXS port after 4 rings (default configuration), then the caller hears either a special continuous tone or a dial tone. The continuous tone is played if the pin code is configured, or the dial tone otherwise. 3. Enter in the pin code that is configurable on the configuration page. The caller will hear the dial tone and get bridged to VoIP if the pin code is valid, otherwise the continuous tone is played again to prompt caller to enter in the pin code again. The use may try up to 3 times to enter in pin code, if none is valid, HT488 will hang up. 4. The caller can dial a VoIP number followed by # (or wait for 4 seconds), the VoIP call will be initiated from the SIP account configured on the FXO port. NOTE: Users can choose whether apply password protection for PSTN-to-VoIP calls or not. A PIN (Pin to VoIP calls) consists of up to 8 numeric digits can be configured through BASIC SETTINGS of the web configuration page. By default, there is no password protection, i.e. there is no authentication required for callers on the use of VoIP SIP account on FXO port. Upon hearing the special continuous tone for PIN code input, if the caller dont enter any digit, HT488 will time out and hang up the call in 10 seconds. During any stage of DTMF digits input, a 4 seconds timeout is applied to serve as an end of PIN or destination number input. Users may also use the # key to indicate the end of an input. On the web configuration page, if the Forward to VoIP is configured, the second stage dialing is eliminated, i.e., after bridging to VoIP, the configured VoIP number will be called automatically.

ROUTE CALLS TO PSTN The FXO port enables access to the PSTN network. By default, the HT503 is in VoIP mode at off-hook. If Route call to PSTN is configured, certain calls will be initiated from the FXO PSTN line port. This call feature is especially useful for emergency calls or local telephone calls. To use this feature, users need to specify a prefix or a telephone number in the Route call to PSTN in the BASIC SETTINGS web configuration page. If the dialed digits match the specified prefix, outbound calls will be initiated from PSTN line. Note: The route to PSTN feature is only applicable to a phone connected to the FXS Port. The configuration is done using the dial plan feature under the FXS tab. An example of the configuration is {L: 911x+} This shows that only calls that start with 911 are immediately forwarded to the PSTN line. All other numbers will not be routed to the PSTN. An normal # would be: {L: 617x+|x+} or {x+| L: 617x+} For example, if Route call to PSTN is configured as 626, all outgoing calls starting with 626 will be initiated from the PSTN line.
FORWARD CALLS TO PSTN Any VOIP call may be forwarded to a specified PSTN number if the call is not answered after a pre configured numbers of rings. By default Number of Rings parameter has value 4.
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For example, if the end-user has configured a cell phone number in the field Forward to PSTN under BASIC SETTINGS configuration page, all calls will be forwarded to the cell phone number after 4 rings. FORWARD CALLS TO VOIP By default, each incoming PSTN call is received over the FXS port. The end-user may forward such a call to any preconfigured VoIP extension, in case the call is not answered in a certain number of rings. The Default value of the parameter Number of Rings is 4. If during 4 rings, the incoming from the PSTN call is not answered, the call will be forwarded to another VoIP number previously configured in the field:Forward to VoIP. This parameter can also be found under BASIC SETTINGS configuration page. ONE STAGE DIALING This feature is applicable for VoIP to PSTN calls. Any VoIP extension may dial directly to a local PSTN number if the one-stage dialing feature is activated. This feature is configured under the FXO Configuration page and requires SIP Server configuration and support. The special dial plan feature must be activated in the SIP Server. An outbound call will be sent directly to the assigned FXO port account, where there the HT488 will initiate a call to the local CO. The RequestURI header in the INVITE message contains the phone number used to initiate the call to the local CO. FAX SUPPORT HT488 supports FAX in two modes: T.38 (Fax over IP) and fax pass through. T.38 is the preferred method because it is more reliable and works well in most network conditions. If the service provider supports T.38, please use this method by selecting Fax mode to be T.38 (default). If the service provider does not support T.38, pass-through mode may be used. To send or receive faxes in fax pass through mode, users must select all the Preferred Codecs to be PCMU/PCMA (G.711-u/a).

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CALL FEATURES

TABLE 5: HT488 CALL FEATURE DEFINITIONS Key
*23 *30 *31 *67 *82 *50 *51 *70 *71 *72

Call Features

3-Way Conferencing. Refer to above section for procedure to perform 3-Way Calling. Block Caller ID (for all subsequent calls) Send Caller ID (for all subsequent calls) Block Caller ID (per call). Dial *67 + number. No dial tone will be played in the middle. Send Caller ID (per call). Dial *82 + number. No dial tone will be played in the middle. Disable Call Waiting (for all-config change) Enable Call Waiting (for all-config change) Disable Call Waiting (Per Call) Enable Call Waiting (Per Call) Unconditional Call Forward. To use this feature, dial *72, wait for the dial tone. Then dial the forward number ended with #, wait for dial tone, hang up. Cancel Unconditional Call Forward To cancel Unconditional Call Forward, dial *73 and get the dial tone, then hang up. Blind Transfer Refer to section above for procedure to perform Blind Transfer. Busy Call Forward To use this feature, dial *90, wait for the dial tone. Then dial the forward number ended with #, wait for dial tone, hang up. Cancel Busy Call Forward To cancel Busy Call Forward, dial *91 and get the dial tone, then hang up Delayed Call Forward To use this feature, dial *92, wait for the dial tone. Then dial the forward number ended with #, wait for dial tone, hang up. Cancel Delayed Call Forward To cancel this Forward, dial *93 and get the dial tone, then hang up When in conversation, this action will switch to the new incoming call if there is a call waiting beep. When in conversation and there is no call waiting, this action will switch to a new channel for a new call.

*73 *87 *90

*91 *92

*93 Flash/Hook

LED Light Pattern Indication TABLE 6: HT488 LED DEFINITIONS
RED LED always indicates not normal status DHCP Failed or WAN No Cable HT488 fails to register Firmware Upgrading Device Malfunctions GREEN LED indicates normal working status Message Waiting Indication RINGING RINGING INTERVAL In Conversation Grandstream Networks, Inc. Button flashes every 2 seconds Button flashes at 1/10 second Button flashes every second Green light steady on HT488 User Manual Firmware 1.0.3.86 Page 16 of 31 Last Updated: 7/2007 Button flashes every 2 seconds (if DHCP is configured) Button flashes every 2 seconds (if SIP server is configured) Button flashes every 2 seconds Red light steady on

CONFIGURATION GUIDE

CONFIGURING HT488 THROUGH VOICE PROMPT
DHCP MODE Follow Table 3 with voice menu option 01 to enable HT488 to use DHCP. STATIC IP MODE Follow Table 3 with voice menu option 01 to enable HT488 to use STATIC IP mode, then use Option 02, 03, 04 to set up the HT488s IP, Subnet Mask, Gateway respectively. TFTP SERVER ADDRESS Follow Table 3 with voice menu option 06 to configure the IP address of the TFTP server. FIRMWARE SERVER IP ADDRESS Select voice menu option 13 to configure the IP address of the firmware server. CONFIGURATION SERVER IP ADDRESS Select voice menu option 14 to configure the IP address of the configuration server. UPGRADE PROTOCOL Select voice menu option 15 to choose firmware and configuration upgrade protocol. User can choose between TFTP and HTTP. FIRMWARE UPGRADE MODE Select voice menu option 17 to choose firmware upgrade mode among the following three options: 1) always check, 2) check when pre/suffix changes, and 3) never upgrade

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DHCP hostname DHCP domain DHCP vendor class ID Time Zone Daylight Savings Time
Device Mode Reply to ICMP on WAN Port WAN Side HTTP/Telnet Access Cloned WAN MAC Address: LAN Subnet Mask LAN DHCP Base IP: DMZ IP: Port Forwarding: Number of rings PSTN access code PIN for PSTN calls PIN for VoIP calls Route Call to PSTN Forward to PSTN Forward to VoIP FXO One Stage Dialing
04,01,7,02,00;10,-1,7,02,00;60 This means the daylight saving time starts from the first Sunday of April at 2AM and ends the last Sunday of October at 2AM. The saving is 60 minutes (1hour). This parameter controls whether the device is working in NAT router mode or Bridge mode. Save the setting and reboot prior to configuring the HT488. When set to Yes, the HT488 responds to the PING command from other computers, but is also made vulnerable to DOS attacks. Default is No. When set to Yes, the user can access the web configuration pages through the WAN port, instead of through the PC port. Warning: this configuration is less secure than the default option. Default is No. This allows the user to change/set a specific MAC address on the WAN interface. Note: Set in Hex format Sets the LAN subnet mask. Default value is 255.255.255.0 Base IP for the LAN port, which functions as default gateway for its LAN. Default value is 192.168.2.1 Forward all WAN IP traffic to a specific IP address if no matching port is used by HandyTone488 itself or in the defined port forwarding. Allow users to forward a matching (TCP/UDP) port to a specific LAN IP address with a specific (TCP/UDP) port. Default is 4. It specifies number of phone rings before a PSTN incoming call is bridged to VoIP The code to access the PSTN line. Default is *00. PIN code to bridge from VoIP to PSTN PIN code to bridge from PSTN to VoIP If the dialed digits match one of the specified prefix here, outbound calls will be initiated from PSTN line. This field is especially useful for emergency calls. Calls are unconditionally forwarded to the specified PSTN phone number for all incoming VoIP calls on FXO port. Calls are unconditionally forwarded to the specified VoIP phone number for all incoming PSTN calls. This configuration is applicable for VoIP to PSTN calls and indicates one or two stage dialing methods.
ADVANCED CONFIGURATION AND FXS/FXO PORTS PARAMETERS
To login to the Advanced Setting and FXS port configuration pages, administrator password is required. The default administrator password is admin. User can change the administrator password here. The password is case sensitive and the maximum length is 25 characters.
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TABLE 9: HT488 ADVANCED SETTINGS PAGE DEFINITIONS

The Syslog uses USER facility. In addition to standard Syslog payload, it contains the following components: GS_LOG: [device MAC address][error code] error message Here is an example: May 19 02:40:38 192.168.1.14 GS_LOG: [00:0b:82:00:a1:be][000] Ethernet link is up
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TABLE 10: HT488 FXS PORT SETTINGS PAGES DEFINITIONS SIP Server This field contains the URI string or the IP address (and port, if different from 5060) of the SIP proxy server. e.g., the following are some valid examples: sip.my-VoIP-provider.com, or sip:my-company-sip-server.com, or 192.168.1.200:5066 IP address or Domain name of Outbound Proxy, or Media Gateway, or Session Border Controller. Used by ATA for firewall or NAT penetration in different network environment. If symmetric NAT is detected, STUN will not work and ONLY Outbound Proxy will work. User account information, provided by VoIP service provider (ITSP), usually has the form of digit similar to phone number or actually a phone number. This field contains the user part of the SIP address for this phone. e.g., if the SIP address is sip:my_user_id@my_provider.com, then the SIP User ID is: my_user_id. Do NOT include the preceding sip: scheme or the host portion of the SIP address in this field. ID used for authentication, usually same as SIP user ID, but could be different and decided by ITSP. Password for ATA to register to (SIP) servers of ITSP. Purposely blank out once saved for security. Maximum length is 25. SIP service subscribers name which will be used for Caller ID display Default is No. If set to Yes the client will use DNS SRV to lookup for the SIP server. If Yes is set, a user=phone parameter will be attached to the From header in SIP request This parameter controls whether the HandyTone ATA needs to send REGISTER messages to the proxy server. The default setting is Yes. Default is No. If set to yes, the device will first send registration request to remove all previous bindings. Use only if proxy supports this remove bindings request. This parameter allows the user to specify the time frequency (in minutes) the HandyTone ATA refreshes its registration with the specified registrar. The default interval is 60 minutes (or 1 hour). The maximum interval is 65535 minutes (about 45 days). This parameter defines the local SIP port the HandyTone ATA will listen and transmit. The default value for FXS port is 5060. This parameter defines the local RTP-RTCP port pair the HandyTone ATA will listen and transmit. It is the base RTP port for channel 0. When configured, channel 0 will use this port _value for RTP and the port_value+1 for its RTCP; channel 1 will use port_value+2 for RTP and port_value+3 for its RTCP. The default value for FXS port is 5004. Default No. If set to Yes, the device will pick randomly-generated SIP and RTP ports. This is usually necessary when multiple HandyTone ATAs are behind the same NAT. In case sip server fail from some reason, the HT488 will try to re-register not according to standard SIP timers, but according to preconfigured interval. Recommended to leave default value 20sec. Designed by special request of big service providers. This parameter sets the payload type for DTMF using RFC2833 This parameter specify the mechanism to transmit DTMF digit. There are 3

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Voice Frames per TX

G723 Rate: iLBC frame size:
iLBC payload type: Silence Suppression
Fax Mode Early Dial Dial Plan Prefix Use # as Send Key
Subscribe for MWI: Send Anonymous Lock keypad update Refer-To Uses Target Contact. Special Features Onhook Threshold FXS Impedance

Caller ID Scheme

This defines payload type for iLBC. Default value is 97. The valid range is between 96 and 127. This controls the silence suppression/VAD feature of G723. If set to Yes, when a silence is detected, small quantity of VAD packets (instead of audio packets) will be sent during the period of no talking. If set to No, this feature is disabled. T.38 (Auto Detect) FoIP by default, or fax Pass-Through. Default is No. Use only if proxy supports 484 response Sets the prefix added to each dialed number This parameter allows users to configure the # key to be used as the Send (or Dial) key. If set to Yes, pressing this key will immediately trigger the sending of dialed string collected so far. If set to No, this # key will then be included as part of the dial string to be sent out. Default is No. When set to Yes a SUBSCRIBE for Message Waiting Indication will be sent periodically. If this parameter is set to Yes, user ID will be sent as anonymous, essentially blocking the Caller ID from displaying. If this parameter is set to Yes, the configuration update via keypad is disabled. Used for Attended transfer Feature. Default is NO. If set to YES, the ReferTo header uses the transferred targets Contact header information. Default is Standard. Choose the selection to meet some special requirements from Soft Switch vendors like Nortel, Broadsoft, etc. Default setting is 800ms. If the flash event is longer than the settings, it is processed as on-hook event. Selects the impedance of the analog telephone connected to the Phone port. The following information may be useful for end user configuration: 600Ohm North America 270+750Ohm || 150nF Most of Europe 220+820Ohm || 120nF Australia, New Zealand 220+820Ohm || 115nF Austria, Bulgaria, Germany, Slovakia, South Africa 370+620Ohm || 310nF UK, India Select the Caller ID Scheme to suit the standard of different area. Bellcore (North America) CID - Canada DTMF (Brazil) DTMF (Sweden) DTMF (Denmark) ETSI-DTMF (Finland, Sweden) ETSI-FSK (France, Germany, Norway, Taiwan, UK-CCA)
Onhook Voltage Polarity Reversal Volume Amplification
Select the onhook voltage to suit the analog phone. Select Polarity Reversal to adapt some call charge/billing system. Default is No. Handset volume adjustment. RX is for receiving volume, TX is for transmission volume. Default values are 0dB for both parameters. +6dB generates the highest volume and -6dB generates the lowest volume.
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TABLE 11: HT488 FXO PORT SETTINGS PAGES DEFINITIONS Local SIP port Local RTP port PSTN AC Termination Enable PSTN Disconnect Tone Detection The default value for FXO port is 5062. The default value for FXO port is 5008. Selects the impedance of the analog PSTN line connected to the Line port. If set to Yes, a special tone is used as the disconnect signal. This must be preconfigured for the device to recognize the signal. In case call has been established through the FXO port and remote side has disconnected active call first, the FXO port will wait for this pre-configured signal to disconnect the VoIP call clear the line. This configuration should be configured by the VoIP service provider. Some country use single frequency tone to signal PSTN disconnection, some country use double frequency tone. This setting can be configured to suit the telephone companys standard in different country. Terminate call after long silence detected. Default setting is 60 sec, max 65536 The Default value is Yes. 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 HT488 will search for a power drop for a preconfigured time frame to disconnect such calls from a VoIP extension. Given value in milliseconds. 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 less than 500ms and the call will be considered as terminated when measured voltage drop period will be equal or more then 500ms. This is useful to prevent random call drops in some low quality PSTN lines.

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CONFIGURATION FILE DOWNLOAD
Grandstream SIP Device can be configured via Web Interface as well as via Configuration File through TFTP or HTTP. Config Server Path is the TFTP or HTTP server path for configuration file. It needs to be set to a valid URL, either in FQDN or IP address format. The Config Server Path can be same or different from the Firmware Server Path. A configuration parameter is associated with each particular field in the web configuration page. A parameter consists of a Capital letter P and 2 to 3 (Could be extended to 4 in the future) digit numeric numbers. i.e., P2 is associated with Admin Password in the ADVANCED SETTINGS page. For a detailed parameter list, please refer to the corresponding firmware release configuration template. When Grandstream Device boots up or reboots, it will issue request for configuration file named cfgxxxxxxxxxxxx, where xxxxxxxxxxxx is the MAC address of the device, i.e., cfg000b820102ab. The configuration file name should be in lower cases.
FIRMWARE AND CONFIGURATION FILE PREFIX AND POSTFIX
Firmware Prefix and Postfix allows device to download the firmware name with the matching Prefix and Postfix. This makes it the possible to store ALL of the firmware with different version in one single directory. Similarly, Config File Prefix and Postfix allows device to download the configuration file with the matching Prefix and Postfix. Thus multiple configuration files for the same device can be stored in one directory. In addition, when the field Check New Firmware only when F/W pre/suffix changes is set to Yes, the device will only issue firmware upgrade request if there are changes in the firmware Prefix or Postfix.
MANAGING FIRMWARE AND CONFIGURATION FILE DOWNLOAD
When Automatic Upgrade is set to Yes, Service Provider can use P193 (Auto Check Interval, in minutes, default and minimum is 60 minutes) to have the devices periodically check with either Firmware Server or Config Server, whenever they are defined. This allows the device periodically check if there are any new changes need to be taken on a scheduled time. By defining different intervals in P193 for different devices, Server Provider can spread the Firmware or Configuration File download in minutes to reduce the Firmware or Provisioning Server load at any given time.

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RESTORE FACTORY DEFAULT SETTING
WARNING! Restoring the Factory Default Setting will DELETE all configuration information of the phone. Please BACKUP or PRINT out all the settings before you approach to following steps. Grandstream will not take any responsibility if you lose all the parameters of setting and cannot connect to your VoIP service provider.

FACTORY RESET

IVR Command Reset default factory settings using the IVR Prompt (Table 5): 1. Dial *** for voice prompt. 2. Enter 99 and wait for reset voice prompt. 3. Enter the encoded MAC address (Look below on how to encode MAC address). 4. Wait 15 seconds and device will automatically reboot and restore factory settings. Encoding the MAC Address 1. Locate the MAC address of the device. It is the 12 digit HEX number on the bottom of the unit. 2. Key in the MAC address. Use the following mapping: 0-9: 0-9 a. A: 22 (press the 2 key twice, A will show on the LCD) b. B: 222 c. C: 2222 d. D: 33 (press the 3 key twice, D will show on the LCD) e. E: 333 f. F: 3333 For example: if the MAC address is 000b8200e395, it should be keyed in as 0002228200333395.
NOTE: 1. Factory Reset will be disabled if the Lock keypad update is set to Yes. 2. Please be aware by default the HT488 WAN side HTTP access is disabled. After a factory reset, the devices web configuration page can be accessed only from its LAN port. 3. If the HT488 was previously locked by your local service provider, pressing the RESET button will only restart the unit. The device will not return to factory default settings.
Page 31 of 31 Last Updated: 7/2007

BroadSoft Partner Configuration Guide
Grandstream HT286/386/486/488/496 Analog Telephone Adapter & BT100 SIP Phone
October 2007 Document Version 1.2

BroadWorks Guide

Copyright Notice
Copyright 2007 Grandstream Networks, Inc. All rights reserved. Any technical documentation that is made available by Grandstream Networks, Inc. is proprietary and confidential and is considered the copyrighted work of Grandstream Networks, Inc.

Trademarks

BroadWorks and BroadWorks Assistant are trademarks of BroadSoft, Inc. Microsoft, MSN, Windows, and the Windows logo are registered trademarks of Microsoft Corporation. Other product names mentioned in this manual may be trademarks or registered trademarks of their respective companies and are hereby acknowledged. This document is printed in the United States of America.
Document Revision History
Version 1.1 1.2 Reason for Change Introduced document for HT496 version 1.0.3.96 validation with BroadWorks R14.SP2 Added HT286/386/486/488/BT-100 models.
BROADSOFT PARTNER CONFIGURATION GUIDE GRANDSTREAM HT286/386/486/488/496/BT100
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2007 GRANDSTREAM. PROPRIETARY AND CONFIDENTIAL; DO NOT DUPLICATE, OR DISTRIBUTE.

Table of Contents

3 Overview.....5 BroadWorks Validation Package Support Level...6 Device Capabilities and Known Interoperability Issues...7 3.1 3.Capabilities....7 Interoperability Issues....9
BroadWorks Device Identity/Profile...10 BroadWorks Device Type....12 Configuration.....13 6.1 6.2 System Level Configuration.... 14 Subscriber Level Configuration Parameters... 15
Enhanced IP Phone Configuration....16
Appendix A: Sample HT-286 / HT-386 / HT-486 / HT-488 / HT-496 / BT-100 Configuration Files17 Appendix B: BroadWorks Validation Package Test Items..21 References.....23
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Overview

This document describes the configuration procedures required for a Grandstream HT286 / HT-386 / HT-486 / HT-488 / HT-496 / BT-100 to make full use of the capabilities of BroadWorks. The HT-286 / HT-386 / HT-486 / HT-488 / HT-496 / BT-100 is one of the many access devices that interoperate with BroadWorks. The HT-286 / HT-386 / HT-486 / HT-488 / HT-496 / BT-100 uses the Session Initiation Protocol (SIP) to communicate with BroadWorks for call control. It also translates voice to audio packets for transmission across a packet network. This guide describes the specific configuration items that are important for use with BroadWorks. It does not describe the purpose and use of all configuration items on the HT-286 / HT-386 / HT-486 / HT-488 / HT-496 / BT-100. For those details, see the HT-286 / HT-386 / HT-486 / HT-488 / HT-496 / BT-100 User Manual supplied by Grandstream Error! Reference source not found.
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BroadWorks Validation Package Support Level
Devices are validated according to BroadWorks Validation Packages. Each package validates a subset of features or items. This section describes the devices support level for a BroadWorks Validation Package as well as the features or items in the package that are not supported. For specific issues, see section 3.2 Interoperability Issues. For a complete list of items validated per package, see Appendix B: BroadWorks Validation Package Test Items.
BroadWorks Package Basic Call BroadWorks Enhanced Services Support Level Full Partial Ring Splash Priority Alerting Priority Call Waiting Alternate Numbers Advanced Call Control DUT Services Call Control Partial Network Conference 3-Way Network Conference N-Way DUT Services Registration and Authentication DUT Services FAX DUT Services Busy Lamp Field Redundancy Partial Full None Partial Call Setup Failover Mid-Call Failover SBC/ALG Shared Call Appearance Feature Key Synchronization Video TCP Full None None None None BroadWorks Authentication Items Not Supported
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Device Capabilities and Known Interoperability Issues
This section describes the features supported by the HT-286 / HT-386 / HT-486 / HT-488 / HT-496 / BT-100, as well as BroadWorks interoperability issues and impact. The following table describes capabilities. Verified Revisions shows the results of partner (vendor) testing of a specific BroadWorks version with a specific partners device under test (DUT) version. Compatible Revisions indicates the maintenance versions that should interface properly with BroadWorks.

Capabilities

Device Type Lines and Appearances SIP ATA HT-286: 1 line HT-386: 2 lines HT-486: 1 line HT-486: 1 line HT-496: 2 lines BT-100: 1 line Speaker/Power/Bridge 1 appearance per line 1 appearance per line 1 appearance per line 1 appearance per line 1 appearance per line 1 appearance per line

HT-286: No/No/Yes HT-386: No/No/Yes HT-486: No/No/Yes HT-488: No/No/Yes HT-496: No/No/Yes BT-100: Yes/No/Yes Indicates speaker phone, in-line power, device bridge/conference.

Verified Revisions

BroadWorks Release: 14.SP2 Grandstream HT-286: 1.1.0.8 Grandstream HT-386: 1.0.3.96 Grandstream HT-486: 1.1.0.8 Grandstream HT-488: 1.0.3.96 Grandstream HT-496: 1.0.3.96 Grandstream BT-100: 1.1.0.8

Compatible Revisions

BroadWorks Release: 14.SP2 Grandstream HT-286: any maintenance version of 1.1.0.8 Grandstream HT-386: any maintenance version of 1.0.3.96 Grandstream HT-486: any maintenance version of 1.1.0.8 Grandstream HT-488: any maintenance version of 1.0.3.96 Grandstream HT496: any maintenance version of
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1.0.3.96 Grandstream BT-100: any maintenance version of 1.1.0.8 SIP Proxy FQDN DNS Lookup (A, SRV, NAPTR) Outbound Proxy Configurable Outbound Proxy FQDN DNS Lookup (A, SRV, NAPTR) BroadWorks Redundancy Enabled BroadWorks Shared Call Appearance BroadWorks Enhanced IP Phone Configuration Device Services Device Call Control (Device-Controlled or Flash INFObased) Codecs RFC 2833 DTMF T.38 Fax TCP TLS PCMU, PCMA, G723.1 G729A/B, iLBC, G726-32 Yes Yes No No No N/A No Call Forwarding, Call Transfer, 3-Way Conference, Call Waiting, Call Hold Device-Controlled Yes A, SRV A, SRV
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Interoperability Issues

This section lists the known interoperability issues between BroadWorks and partner release(s). For more information on issues related to the particular software release, see the partner release notes. None.
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BroadWorks Device Identity/Profile
BroadWorks configurable device identify/profile is introduced in BroadWorks Release 14.0. This section applies only to BroadWorks Release 14.0 and later. The following table identifies the required BroadWorks device identity/profile settings for interoperability between the HT496 and BroadWorks. For an explanation of the profile parameters, refer to the BroadWorks Device Inventory Guide [4]. For most of the parameters below, an X indicates the parameter function is supported and/or required. If the item is blank, it is not supported.
Grandstream HT-286 / HT-386 / HT-486 / HT-488 / HT-496 / BT-100 Identify/Device Profile
Signaling Address Type Number of Lines Ringback Tone/ Early Media Support Authentication Registration Capable Static Registration Capable E.164 Capable Trusted Authenticate REFER Authentication Override Video Capable RFC 3264 Hold Route Advance Wireless Integration PBX Integration Use Business Trunking Contact Forwarding Override Conference Device Music On Hold Device Auto Configuration Soft Client Web Based Configuration URL Auto Configuration Type Reset Event Enable Monitoring CPE System File Name ataSystem.txt/bt100System.txt http//:%BWIPADDRESS% 2 Config File checkSync X X Intelligent Proxy Addressing HT-286/HT-486/BT-100: 1 line HT-386/HT-488/HT-496: 2 lines Local Ringback No Early Media Enabled X

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Device File Format

%BWMACADDRESS%.txt
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BroadWorks Device Type

This section applies to BroadWorks Release 13.0 and before. The following BroadWorks device type should be used for Grandstream HT-286 / HT-386 / HT-486 / HT-488 / HT-496 / BT-100. Grandstream HT-286 Grandstream HT-386 Grandstream HT-486 Grandstream HT-488 Grandstream HT-496 Grandstream BT-100
The above device types require BroadWorks Release 13 MP<patch number>. If the BroadWorks system is currently at a lower patch level, either apply the necessary patch or use the following generic device type until the system can be patched accordingly. Generic SIP Smart (Proxy Addr)
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Configuration

Grandstream HT-286 / HT-386 / HT-486 / HT-488 / HT-496 / BT-100 can be configured via Web GUI Interface as well as via Configuration File through TFTP or HTTP. The HT-286 / HT-386 / HT-486 / HT-488 / HT-496 / BT-100 accepts configuration files in a proprietary binary format which is produced by the Grandstream configuration tool. The configuration tool creates a device configuration file based on a configuration template and the devices MAC address. All Grandstream products sample configuration templates are available on Grandstreams web site. A configuration template file contains all configuration parameters available for a product, with comments describing allowed values of the parameters. For a detailed parameter list, please refer to the corresponding firmware release configuration template [2]. When Grandstream device boots up or reboots, it will issue request for a binary configuration file named cfgxxxxxxxxxxxx, where xxxxxxxxxxxx is the MAC address of the device, i.e., cfg000b820102ab. The configuration file name should be in lower cases. Grandstream offers free Configuration File generator software in both Linux/Unix and Windows platform. Both Configuration File Generators can be downloaded from Grandstream official web site at http://www.grandstream.com/configurationtool.html [2]. The HT-286 / HT-386 / HT-486 / HT-488 / HT-496 / BT-100 can be configured with a configuration file using the Trivial File Transfer Protocol (TFTP) or through its embedded web server. The following examples describe how to set the parameters using a configuration file. This configuration description assumes the HT-286 / HT-386 / HT-486 / HT-488 / HT-496 / BT-100 will use the Dynamic Host Configuration Protocol (DHCP) to obtain an IP address and other network settings. The HT-286 / HT-386 / HT-486 / HT-488 / HT-496 / BT-100 should be configured to load the configuration file each time it resets or re-synchronizes. For detailed information on automated provisioning, please check the HT-286 / HT-386 / HT-486 / HT-488 / HT-496 / BT-100 User Manual Error! Reference source not found. The capabilities of the HT-286 / HT-386 / HT-486 / HT-488 / HT-496 / BT-100 have been verified for use with BroadWorks based on the settings described in the following table. For more information on the meaning, purpose, and applicability of the individual configuration items, see the HT-286 / HT-386 / HT-486 / HT-488 / HT-496 / BT-100 configuration template [2]. Configuration Files

Files Provided by Partner HT-286 / HT-386 / HT-486 / HT-488 / HT496 / BT-100 Configuration Template Example: sipp1_config_1.1.0.8.txt sipp2_config_1.0.3.96.txt cfgMAC Example: cfg000b82000000 System and Subscriber Binary configuration file generated from configuration template and an individual devices MAC address. Level System and Subscriber Description Contains configurable parameters that apply to an individual device in a deployment.
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System Level Configuration
This section describes system-wide configuration items that are generally required for each HT-286 / HT-386 / HT-486 / HT-488 / HT-496 / BT-100 to work with BroadWorks. Subscriber-specific settings are described in the next section. {THIS IS JUST AN EXAMPLE AND IS NOT INTENDED AS A COMPLETE OR SUFFICIENT TABLE OF PARAMETERS. EACH DEVICE WILL VARY AND MAY BE COMPLETELY DIFFERENT. IF YOUR DEVICE DOES NOT SUPPORT CONFIG FILES USE GUI CAPTURES OR OTHER METHODS TO DESCRIBE THE CONFIGURATION.}

Step Command Purpose

System Configuration Items Step 1 Set SIP Proxy/Domain. sipProxy = as.broadworks.net Set the HT-286 / HT-386 / HT486 / HT-488 / HT-496 / BT-100 SIP server to the Fully Qualified Domain Name (FQDN) for the BroadWorks Application Server cluster. The domain must match the domain configured for the BroadWorks subscribers line/port domain. Step 2 Set Outbound Proxy. outBoundProxy = sbc.broadworks.net Set the outbound proxy to the session border controller (SBC) if one is deployed between the HT-286 / HT-386 / HT-486 / HT488 / HT-496 / BT-100 and BroadWorks. If there are redundant SBCs, set it to the FQDN for the SBC cluster. Step 3 Enable DNS SRV lookup. Use_DNS_SRV = Yes ; Step 4 Set register mode. SIP Registration = Yes Step 5 Set SIP Timers. Register Expiration = 60 Step 6 Enable negotiated DTMF type. Send DTMF = via RTP (RFC2833) The default registration period is 60 minutes. Set the HT-286 / HT-386 / HT486 / HT-488 / HT-496 / BT-100 to enable inband or RFC 2833 negotiated DTMF. Disable feature access codes controlled by the device. Enable SIP register. Enable DNS SRV lookups.

Step 7

Disable local feature code services. Enable Call Features = No

Step 8

Enable BroadSoft mode. Special Feature = Broadsoft
Set the HT-286 / HT-386 / HT486 / HT-488 / HT-496 / BT-100 in BroadSoft mode.
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Subscriber Level Configuration Parameters
This section identifies the device-specific parameters, including registration and authentication. These settings must be unique across devices in order to be matched with the settings for a BroadWorks subscriber. Provisioning a subscriber to register with BroadWorks allows calls to terminate to the subscribers line. Registration requires that a unique address of record (AoR) is provisioned on BroadWorks and the phone; provisioning an AoR on BroadWorks consists of setting the line/port parameter to a unique value within the Application Server cluster. Grandstream HT-286 / HT-386 / HT-486 / HT-488 / HT-496 / BT-100 supports one SIP accounts (lines), two SIP accounts, one SIP account, two SIP accounts, two SIP accounts, and 1 SIP account, respectively. The follow example shows how to configure subscriber elements for an account.

Command

Purpose
Subscriber Configuration Items Step 1 Set Register User ID for each line. Example: SIP User_ID = "2405551111" ; Step 2 Enable SIP Authentication for each line. Example: Authenticate ID = "1111@as.mycompany.com" ; Authenticate Password = ; Step 3 "welcome" For each line, configure the name to be displayed on the device. If the Authentication service is configured on BroadWorks, these parameters must be configured to match the BroadWorks settings. The register user ID must correspond with the line/port setting on BroadWorks.
Configure display name for each line. Example: Name = Claire Smith ;
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Enhanced IP Phone Configuration
Enhanced IP Phone Configuration is a BroadWorks feature that enables automatic generation of device configuration files, given administrator-supplied templates. For more information on the Enhanced IP Phone Configuration feature, see the Enhanced IP Phone Configuration Guide [3]. The HT-286 / HT-386 / HT-486 / HT-488 / HT-496 / BT-100 uses only a phone-specific file to configure the device. However, the Enhanced IP Phone Configuration feature requires a system and a phone-specific file for each device. To satisfy the system file requirement, an empty file (ataDefault.txt, bt100Desfult.txt) will need to be uploaded to BroadWorks as the system file. This file will be sent to the FTP server but will not actually be used for the configuration file generation. The group template file is used to build the configuration files for the devices of this device type assigned to the group. For a sample HT-286 / HT-386 / HT-486 / HT-488 / HT-496 / BT-100 group template file, see Appendix A.

NOTE: BroadSoft does not manage or distribute template files for use with the Enhanced IP Phone Configuration feature. Obtain template files from Grandstream or use the configuration files obtained from Grandstream for the specific Grandstream firmware release to create template files appropriate for your installation. Since HT-286 / HT-386 / HT-486 / HT-488 / HT-496 / BT-100 require the configuration files be converted into a binary format, an extra step must be performed when using this feature. After the configuration files have been sent to the TFTP server and before the phone is reset, the configuration files must be converted to binary format using Grandstreams configuration tool. For detailed information on automated provisioning, please check the Grandstream configuration tool user guide [2].
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Appendix A: Sample HT-286 / HT-386 / HT-486 / HT-488 / HT-496 / BT-100 Configuration Files
NOTE: The following samples are examples and should be used as a reference only. DO NOT CUT AND PASTE THESE EXAMPLES TO GENERATE YOUR CONFIGURATION FILES. Use the configuration files obtained from Grandstream with the specific release to generate your configuration files.
System Default Parameters: NOTE: This is an example file and should be used for reference only.
# SIP Default Generic Configuration File #################################################### # Account 1 # # HT-286 / HT-486 / BT-100 supports Account 1 only # #################################################### # SIP Server P47 = as.broadworks.net # Outbound Proxy P48 = sbc.broadworks.net # Use DNS SRV. 0 - no, 1 - yes. P103 = 1 # SIP Registration. 0 - no, 1 - yes P31 = 1 # Register Expiration (in minutes. default 1 hour, max 45 days) P32 = 60 # Send DTMF. 8 - in audio, 1 - via RTP, 2 - via SIP INFO # 11 - In Audio & RTP & SIP INFO, 9 - In Audio & RTP # 10 - IN Audio & SIP INFO, 3 - RTP & SIP INFO P73 = 1 # Enable Call Features. 0 - no, 1 yes P191 = 0 # Special Feature. 102 BroadSoft P198 = 102 #################################################### # Account 2 # # # #################################################### # SIP Server P747 = as.broadworks.net

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# Outbound Proxy P748 = sbc.broadworks.net # Use DNS SRV. 0 - no, 1 - yes. P702 = 1 # SIP Registration. 0 - no, 1 yes P731 = 1 # Register Expiration (in minutes. default 1 hour, max 45 days) P732 = 60 # Send # 11 # 10 P773 = DTMF. 8 - in audio, 1 - via RTP, 2 - via SIP INFO In Audio & RTP & SIP INFO, 9 - In Audio & RTP IN Audio & SIP INFO, 3 - RTP & SIP INFO 1
# Enable Call Features. 0 - no, 1 yes P751 = 0 # Special Feature. 102 BroadSoft P767 = 102
Phone-Specific Parameters: NOTE: This is an example file and should be used for reference only.
# SIP Device-specific Configuration File #################################################### # Account 1 # # HT-286 / HT-486 / BT-100 supports Account 1 only # #################################################### # SIP User ID P35 = 2405551111 # Authenticate ID P36 = 1111@as.mycompany.com # Authenticate password P34 = welcome # Display Name P3 = Claire Smith #################################################### # Account 2 # # # #################################################### # SIP User ID P735 = # Authenticate ID P736 =
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# Authenticate Password P734 = # Display Name P703 =
Group Template File: ATA-GroupTemplate.txt /BT100- GroupTemplate.txt NOTE: This is an example file and should be used for reference only.
#################################################### # SIP Default Generic Configuration Parameters # ####################################################
#################################################### # Account 1 # # HT-286 / HT-486 / BT-100 supports Account 1 only # #################################################### # SIP Server P47 = as.broadworks.net # Outbound Proxy P48 = sbc.broadworks.net # Use DNS SRV. 0 - no, 1 - yes. P103 = 1 # SIP Registration. 0 - no, 1 - yes P31 = 1 # Register Expiration (in minutes. default 1 hour, max 45 days) P32 = 60 # Send DTMF. 8 - in audio, 1 - via RTP, 2 - via SIP INFO # 11 - In Audio & RTP & SIP INFO, 9 - In Audio & RTP # 10 - IN Audio & SIP INFO, 3 - RTP & SIP INFO P73 = 1 # Enable Call Features. 0 - no, 1 yes P191 = 0 # Special Feature. 102 BroadSoft P198 = 102 #################################################### # Account 2 # # # #################################################### # SIP Server P747 = as.broadworks.net # Outbound Proxy
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P748 = sbc.broadworks.net # Use DNS SRV. 0 - no, 1 - yes. P702 = 1 # SIP Registration. 0 - no, 1 yes P731 = 1 # Register Expiration (in minutes. default 1 hour, max 45 days) P732 = 60 # Send # 11 # 10 P773 = DTMF. 8 - in audio, 1 - via RTP, 2 - via SIP INFO In Audio & RTP & SIP INFO, 9 - In Audio & RTP IN Audio & SIP INFO, 3 - RTP & SIP INFO 1

# Enable Call Features. 0 - no, 1 yes P751 = 0 # Special Feature. 102 BroadSoft P767 = 102 #################################################### # SIP Device-specific Configuration Parameters # #################################################### #################################################### # Account 1 # # HT-286 / HT-486 / BT-100 supports Account 1 only # #################################################### # SIP User ID P35 = %BWLINEPORT-1% # Authenticate ID P36 = %BWAUTHUSER-1% # Authenticate password P34 = %BWAUTHPASSWORD-1% # Display Name (John Doe) P3 = %BWCLID-1% #################################################### # Account 2 # # # #################################################### # SIP User ID P735 = %BWLINEPORT-2% # Authenticate ID P736 = %BWAUTHUSER-2% # Authenticate Password P734 = %BWAUTHPASSWORD-2% # Display Name (John Doe) P703 = %BWCLID-2%
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Appendix B: BroadWorks Validation Package Test Items
The following table describes the items tested in each BroadWorks Validation Package.
BroadWorks Validation Package Basic Call Items Supported Basic Call Origination / Termination Call Failure Codes Session Audit Ringback Dial Plan Inband DTMF RFC 2833/Negotiation DTMF Relay Codec Renegotiation BroadWorks Enhanced Services Basic CommPilot Call Manager Functions Voice Messaging Audio MWI Voice Messaging Visual MWI Ring Splash Priority Alerting Priority Call Waiting Alternate Numbers Advanced Call Control Anonymous Call Remote Restart Call Park Retrieve Answer with Hold DUT Services Call Control Call Waiting Call Hold Blind Transfer Attended Transfer 3-Way Call Network Conference 3-Way Network Conference N-Way DUT Services Registration and Authentication Authenticated Registration Maximum Registration Minimum Registration Rejected Registration Authenticated Origination Authenticated Re-INVITE Authenticated REFER BroadWorks Authentication
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BroadWorks Validation Package DUT Services - FAX
Items Supported FAX Passthrough FAX T38
DUT Services Busy Lamp Field Redundancy
Basic BLF DNS Registration Failover Call Setup Failover Mid-Call Failover

SBC/ALG

Registration Call Origination Call Termination

Shared Call Appearance

Line Seize Line Lamp Management Line Hold/Retrieve Multiple Call Arrangement SCA Bridging
Feature Key Synchronization Video
Feature Key Synchronization Integrated Video Phone Video Services Video Add-On
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References

[1] Grandstream Networks, Inc. 2007. HT-286 / HT-386 / HT-486 / HT-488 / HT-496 / BT100 User Manual. Download at: http://www.grandstream.com/user_manuals/ AND GUI Interfaces: http://www.grandstream.com/user_manuals/GUI/. [2] Grandstream Networks, Inc. 2007. Grandstream Configuration Tool and Templates. Available from http://www.grandstream.com/configurationtool.html. [3] BroadSoft, Inc. 2006. BroadWorks Enhanced IP Phone Configuration Guide, Release 14.0. Available from BroadSoft at www.broadsoft.com/boulevard. [4] BroadSoft, Inc. 2006. BroadWorks Device Inventory Guide, Release 14.0. Available from BroadSoft at www.broadsoft.com/boulevard. [5] BroadSoft, Inc. 2006. BroadWorks Redundancy Guide, Release 14.0. Available from BroadSoft at www.broadsoft.com/boulevard.
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