Grandstream Networks, Inc.
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HT503 USER MANUAL INDEX GNU GPL INFORMATION .......................................................................... 4 CHANGE LOG ........................................................................................... 5 CHANGES FROM 1.0.13.3 USER MANUAL ........................................................................................ 5 CHANGES FROM 1.0.12.4 USER MANUAL ........................................................................................ 5 CHANGES FROM 1.0.12.1 USER MANUAL ..
ROUTE CALLS TO PSTN ................................................................................................................... 25 FORWARD CALLS TO PSTN.............................................................................................................. 25 FORWARD CALLS TO VOIP............................................................................................................... 25 ONE STAGE DIALING ....................................................................................
TABLE OF FIGURES HT503 User Manual Figure 1: CONNECTING THE HT503 ............................................................................................ 10 Figure 2: INTERCONNECTION DIAGRAM OF THE HT503 ......................................................... 11 Figure 3: UPLINK/DOWNLINK BANDWIDTH LIMITATION ........................................................... 36 TABLE OF TABLES HT503 User Manual Table 1: DEFINITIONS OF THE HT503 CONNECTORS ....................................................
GNU GPL INFORMATION HT503 firmware contains third-party software licensed under the GNU General Public License (GPL). Grandstream uses software under the specific terms of the GPL. Please see the GNU General Public License (GPL) for the exact terms and conditions of the license. Grandstream GNU GPL related source code can be downloaded from Grandstream web site from: http://www.grandstream.com/support/faq/gnu_gpl . FIRMWARE VERSION 1.0.14.
CHANGE LOG This section documents significant changes from previous versions of HT503 user manuals. Only major new features or major document updates are listed here. Minor updates for corrections or editing are not documented here. CHANGES FROM 1.0.13.3 USER MANUAL Added feature [Hold Target Before Refer] in profile settings, which allows user to hold or not hold the phone call before refer.
CHANGES FROM 1.0.11.3 USER MANUAL Added the options to enable/disable [Do Not Escape '#' as %23 in SIP URI] Added network whist/black list function on WAN port [White list for WAN side] [Black list for WAN side] Added settings for [PSTN Ring Timeout (sec)] CHANGES FROM 1.0.10.9 USER MANUAL Added the options to enable/disable [Use P-Preferred-Identity Header] and [Use Privacy Header] Added the option to enable/disable [Error! Reference source not found.] CHANGES FROM 1.0.7.
WELCOME Thank you for purchasing Grandstream’s HT503, the affordable, feature rich, Analog Telephone Adaptor/IAD. The HT503 combines a sleek design with the latest technology to offer more advanced telephony features and significantly better integrated router performance than its predecessor – the HT488. It is the second ATA/IAD in the HandyTone 50x series. The HT503 functions as a true 3-in-1 gateway for PSTN network, analog telephone FXS interface and IP network.
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/products/ht_series/ht503/documents/ht503_usermanual_english.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. FIRMWARE VERSION 1.0.14.
CONNECT YOUR HT503 EQUIPMENT PACKAGING The HT503 ATA package contains: One HT503 Main Case One Universal Power Adaptor One Ethernet Cable One HT503 Vertical Stand CONNECTING THE HT503 The HT503 is designed for easy configuration and easy installation. Configure the HT503 following the directions in the Configuration section of this manual. 1. Connect a standard touch-tone analog telephone to the PHONE port. 2.
HT503 Back View HT503 Front View Display LEDs (Green) RJ-45 Ports 10/100 Mbps Reset Power Supply (12V) RJ11 RJ11 FXS Port FXO Port FIGURE 1: CONNECTING THE HT503 The HT503 has one FXS port and one FXO port. The PHONE port next to the power supply is an FXS port. The LINE port on the back right of the HT503 is an FXO port. Both the FXS port and the FXO port can have a separate SIP account. This is a key feature of HT503 as it supports simultaneous calls on both the FXS port and FXO port.
TABLE 2: HT503 LED DEFINITIONS POWER LED Indicates Power. Remains ON when power is connected WAN LED Indicates LAN (or WAN) port activity LAN LED Indicates PC (or LAN) port activity PHONE/ LINE LED Indicates the status of the FXS and FXO ports on the back panel. Busy – ON (Solid Green) Available – OFF Slow blinking FXS LEDs indicates voicemail for that port. Note: Slow blinking of POWER, WAN, and LAN LEDs together indicate firmware upgrade/provisioning state.
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PRODUCT OVERVIEW The HT503 is an affordable, high-quality, integrated IP telephony solution for both the residential customers and the ‘road-warriors’ who need advanced call features between traditional PSTN network and IP network. The HT503 enables IP connectivity for any phone or fax using the FXS port and a webbased GUI for easy configuration and installation.
Audio Features Advanced Digital Signal Processing (DSP) Dynamic negotiation of codec and voice payload length Support for G.723, G.729/E, G.711, G.726-40/32/24/16, iLBC, T.38 codecs In-band and out-of-band DTMF (in audio, RFC2833, SIP INFO) Silence Suppression, VAD (voice activity detection), CNG (comfort noise generation), ANG (automatic gain control) Adaptive jitter buffer control Packet delay & loss concealment (PLC) & G.
HARDWARE SPECIFICATION The table below lists the hardware specification of HT503. TABLE 4: HT503 HARDWARE SPECIFICATION LAN interface 1xRJ45 10/100 Mbps Port WAN interface 1xRJ45 10/100 Mbps Port FXS telephone port 1 x FXS (RJ11) FXO 1x PSTN pass-through and life line port telephone port (PSTN Port) LED Power, WAN, LAN, PHONE, and LINE (Green) Universal Switching Input: 100–240 VAC, 50-60 Hz Power Adaptor Output: 12VDC, 0.
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BASIC OPERATIONS UNDERSTANDING HT503 VOICE PROMPT HT503 has a built-in voice prompt menu for simple device configuration. The voice prompt menu is designed for the FXS port only. To enter the voice prompt menu, press *** from the analog phone connected to the FXS port.
13 Firmware Server IP Address Announces current Firmware Server IP address. Enter 12 digit new IP address. 14 15 Configuration Server IP Announces current Config Server Path IP address. Enter 12 digit Address new IP address. Upgrade Protocol Upgrade protocol for firmware and configuration update. Press “9” to toggle between TFTP / HTTP / HTTPS 16 Firmware Version Firmware version information. 17 Firmware Upgrade Firmware upgrade mode.
PLACING A PHONE CALL 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 the 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.
Using Star Code 1. Pick up the analog phone then dial “*47” 2. Enter the target IP address using same format as above. Note: NO dial tone will be played between step 1 and 2. Destination ports can be specified by using “*” (encoding for “:”) followed by the port number. Examples: a) If the target IP address is 192.168.0.160, the dialing convention is *47 or Voice Prompt with option 47, then 192*168*0*160. followed by pressing the “#” key if it is configured as a send key or wait 4 seconds.
CALL TRANSFER The HT503 supports both blind transfer and attended transfer. BLIND TRANSFER This function is applicable using the FXS port for VoIP calls only. Assume that parties A and B are in conversation. Party A wants to Blind Transfer Party B to C: 3. A presses FLASH on the analog phone to hear the dial tone. 4. Then A dials *87, then dials C’s number, and then presses # 5. A can hang up. NOTE: “Enable Call Feature” has to be set to “Yes” in web configuration page.
2. A dials C’s number then # (or wait for 4 seconds). 3. If C answers the call, then A presses FLASH to bring B, C in the conference. 4. If C does not answer the call, A can press FLASH back to talk to B. 5. If A presses FLASH during the conference, C will be dropped out. 6. If A hangs up, the conference will be terminated for all three parties when configuration “Transfer on Conference Hangup” is set to “No”.
VOIP-TO-PSTN CALLS This function is available using the FXO port. The FXO port functions as a bridge between the Internet and PSTN. The user can remotely use a PSTN line to initiate a call. 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.
PSTN-TO-VOIP CALLS This function is available using the FXO port. The FXO port functions as a bridge between the Internet and PSTN and enables calls to be passed from the PSTN network to VoIP. The user can make VoIP calls remotely by dialing into the FXO line port on HT503. To Make a PSTN-to-VoIP Call: 1. Make an incoming call to the PSTN line on FXO port. The phone will ring for 4 times by default (this setting is configurable on the FXO port configuration page). 2.
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 special rule using the dial plan parameter located under FXS Port configuration page.
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.
CALL FEATURES TABLE 6: HT503 CALL FEATURE DEFINITIONS Key Call Features *02 Forcing a Codec (per call) *027110 (PCMU), *027111 (PCMA), *02723 (G723), *02729 (G729), *0272616 (G726-r16), *0272624 (G724-r24), *0272632 (G726-r32), *0272640 (G726-r40), *027201 (iLBC) *03 Disable LEC (pe call) Dial “*03” + ” number ”. No dial tone is played in the middle.
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 server as Re-Dial key. FIRMWARE VERSION 1.0.14.
CONFIGURATION GUIDE CONFIGURING HT503 THROUGH VOICE PROMPT DHCP MODE Follow Table 4 with voice menu option 01 to enable HT503 to use DHCP. STATIC IP MODE Follow Table 4 with voice menu option 01 to enable HT503 to use STATIC IP mode, then use option 02, 03, 04 to set up HT503’s IP, Subnet Mask, Gateway respectively. FIRMWARE SERVER IP ADDRESS Select voice menu option 13 to configure the IP address of the firmware server.
ACCESS THE WEB CONFIGURATION MENU The HT503 HTML configuration page can be accessed via LAN or WAN ports. • From the LAN port: 1. Directly connect a computer to the LAN port 2. Open a command window on the computer 3. Type in “ipconfig /release”, the IP address etc becomes 0 4. Type in “ipconfig /renew”, the computer gets an IP address in 192.168.2.x segment by default 5. Open a web browser, type in the default IP address of the LAN port. http://192.168.2.1. You will see the log in page of the device.
End User and administrator is “123” and “admin” respectively. Only an administrator can access the “ADVANCED SETTING”, “FXS PORT” and “FXO PORT” configuration pages. NOTE: If you cannot log into the configuration page by using the default password, please check with the VoIP service provider. It is most likely the VoIP service provider has provisioned the device and configured for you therefore the password has already been changed.
Product Model This field contains the product model info, such as HT503. Software Version Program: This is the main software release. This number is always used for firmware upgrade. Current release is 1.0.7.6 Boot and Loader are seldom changed. Bootloader: current version is 1.0.0.9. Core: current version 1.0.7.1 Base: current version is 1.0.7.6 CPE: current version is 1.0.1.19 System Uptime This shows system up time since last reboot.
• If Static IP mode is selected, the IP address, Subnet Mask, Default Router IP address, DNS Server 1 (mandatory), DNS Server 2 (optional) fields need to be configured. DHCP hostname This option specifies the name of the client. This field is optional but may be required by some Internet Service Providers. Default is blank. DHCP vendor class ID This option is used by clients and servers to exchange vendor-specific information. Default is blank. PPPoE account ID PPPoE username.
Range: 0 - 3600 NAT UDP Timeout NAT TCP idle timeout in seconds. Connection will be closed after preconfigured, timeout if not refreshed. Range: 0 – 3600, default is 300 Uplink Bandwidth The maximum uplink bandwidth permitted by the device. This function is disabled by default. The total bandwidth can be set as: 128K, 256K, 512K, 1M, 2M, 3M, 4M, 5M, 10M or 15M. The primary function of this setting is to limit the uplink bandwidth for the device internal system, signaling and NATed traffic.
- If black list exists and white list is empty, then ONLY these IP addresses are NOT ALLOWED SIP/RTP access. Black list for WAN side List the IP address or IP range in the White list. The same rules as white list. Cloned This allows the user to change/set a specific MAC address on the WAN interface. WAN MAC Address Note: Set in Hex format LAN DHCP Base IP Base IP for the LAN port, which functions as default gateway for its LAN. Default value is 192.168.2.
FIGURE 3: UPLINK/DOWNLINK BANDWIDTH LIMITATION Advanced User configuration includes not only the end user configuration, but also advanced configurations such as: SIP configuration, Codec selection, NAT Traversal Setting and other miscellaneous configuration. TABLE 9: ADVANCED SETTINGS Admin Password Administrator password. Only the administrator can configure the “Advanced Settings” page. Password field is purposely blanked for security reason after clicking update and saved.
Via TFTP This is the IP address of the configured TFTP server. If this is configured, the HT503 retrieves the new configuration file or new code image from the specified TFTP server at boot time. After 5 attempts, the system will timeout and will start the boot process using the existing code image in the Flash memory. If a TFTP server is configured and a new code image is retrieved, the new downloaded image is saved into the Flash memory.
will be downloaded. Allow DHCP Option 66 If set to “Yes”, configuration and upgrade server information can be obtained using or 160 to override server DHCP option 66 or option 160 from DHCP server located in customer’s environment. Automatic Upgrade Choose “Yes” to enable automatic upgrade and provisioning. When set to No, HT503 will only do upgrade once at boot up. When “Check every day” or “Check every week” is checked, user can specify “Hour of the day(0-23)” or “Day of the week(0-6)”.
sounds. ON is the period of ringing (“On time” in ‘ms’) while OFF is the period of silence. In order to set a continuous tone, OFF should be zero. Otherwise it will ring ON ms and a pause of OFF ms and then repeat the pattern. Example for North America Dial Plan: f1=350@-13,f2=440@-13,c=0/0; Syntax: f1=freq@vol, f2=freq@vol, c=on1/off1-on2/off2-on3/off3; (Note: freq: 0 - 4000Hz; vol: -30 - 0dBm) Note : Maximum supported cadences is 3 Prompt Tone Access Key pattern to get Prompt Tone. Maximum 20 digits.
inbound and outbound calls (INFO level) registration status change (INFO level) negotiated codec (INFO level) Ethernet link up (INFO level) SLIC chip exception (WARNING and ERROR levels) memory exception (ERROR level) 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 Ex. May 19 02:40:38 192.168.1.
SIP User ID 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.
Local SIP port This parameter defines the local SIP port the HT503 will listen and transmit. The default value for FXS port is 5060. Local RTP port This parameter defines the local RTP port pair used by the HandyTone ATA. The default value for FXS port is 5004. Use Random Port Default is 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. Hold Target Before Refer Default is Yes.
(in listed order) (RFC2833) and via Sip Info. The user can configure DTMF method in a priority list. Disable DTMF Default is No. If set to yes, use above DTMF order without negotiation Negotiation Send Flash Event Default is No. If set to yes, flash will be sent as DTMF event. Enable Call Features Default is Yes.
Disable Visual MWI If set to “YES”, the MWI information will not be transferred to the analog phone connected to the FXS port. Do Not Escape '#' as %23 If set to “Yes”, device will use ‘#’ instead of %23 in the send URI. in SIP URI Disable Multiple m line in SDP Ring Timeout Default is No. If set to Yes, device will send only one m line in SDP, regardless how many m field in the incoming SDP. Sets the time in which an incoming call will stop ringing when not picked up.
Block any number of 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. 3. 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.
occurs beforehand. The default value is 180 seconds. Min-SE The minimum session expiration (in seconds). The default value is 90 seconds. Caller Request Timer If selecting “Yes” the phone will use session timer when it makes outbound calls if remote party supports session timer. Callee Request Timer If selecting “Yes” the phone will use session timer when it receives inbound calls with session timer request.
associated with the first vocoder in the above vocoder Preference List or the actual used payload type negotiated between the 2 conversation parties at run time.
SRTP Mode Secure RTP protocol used for media transmission over VoIP. Disabled by default. Other modes are: enabled but not forced & enabled and forced. Crypto Life Time Default is Enabled. Allows user to enable or disable Crypto life time when using SRTP. SLIC Setting Dependent on standard phone type (and location).
TABLE 11: FXO PORT SETTINGS Account Active When set to “Yes” the FXO port is activated. SIP Server SIP Server’s IP address or Domain name provided by VoIP Service Provider. Failover SIP Server This Field contains the URL or the IP address of a second SIP server, this one will be used in case the device loses the connection with the first server. Prefer Primary SIP Default is no.
URI format, then this option needs to be selected. SIP Registration Controls whether the HT503 needs to send REGISTER messages to the proxy server. The default setting is Yes. Unregister on Reboot Default is No. If set to Yes, the SIP user’s registration information will be cleared on reboot. Outgoing Call Without Default is No. If set to “Yes,” user can place outgoing calls even when not registered (if Registration allowed by ITSP) but is unable to receive incoming calls.
Disable DTMF Default is No. If set to yes, use above DTMF order without negotiation Negotiation Proxy Require SIP Extension to notify SIP server that the unit is behind a NAT/Firewall. Use NAT IP NAT IP address used in SIP/SDP message. Default is blank. Use SIP User-Agent Used to replace SIP User-Agent Header (No Default) Header Ring Timeout Sets the time in which an incoming from PSTN call will stop ringing when not picked up. Early Dial Default is No. Use only if proxy supports 484 response.
Allow 311, 611, 911, and any 10 digit numbers of leading digits 1617 • Example 2: {^1900x+ | <=1617>xxxxxxx} – Block any number of 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. 6.
is 180 seconds. Min-SE The minimum session expiration (in seconds). The default value is 90 seconds. Caller Request Timer If selecting “Yes” the phone will use session timer when it makes outbound calls if remote party supports session timer. Callee Request Timer If selecting “Yes” the phone will use session timer when it receives inbound calls with session timer request. Force Timer If selecting “Yes” the phone will use session timer even if the remote party does not support this feature.
or G726, then the “ptime” value in the SDP message of an INVITE request will be 20ms (2 x10ms) If the configured voice frames per TX exceeds the maximum allowed value, the ATA will not accept it and will use and save the precedent configured allowed value for the corresponding first vocoder choice. G723 Rate: This defines the encoding rate for G723 vocoder. Default setting is 6.3kbps. iLBC Frame Size: This sets the iLBC size in 20ms or 30ms iLBC Payload Type: This defines payload type for iLBC.
Caller ID Transport Type According to customer’s choice CID information will be transferred from PSTN network to VoIP network using following rules: 1. via SIP from - PSTN CID is in the SIP From field 2. via P-Asserted-Identity - SIP From field uses the pre-configured account user Id. PSTN CID is in the P-Asserted-Identity field 3. via P-Preferred-Identity - PSTN CID is in the P-Preferred-Identity field 4. Send anonymous - SIP From field uses "anonymous".
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@-24,f2=620@-24,c=500/500;) Note : Maximum supported cadences is 3 AC Termination Model You can select the AC termination by Country or by Impedance. Country-Based 15 Countries are selectable in this version of the F/W.
SAVING THE CONFIGURATION CHANGES After user makes a change to the configuration, press the “Update” button in the Configuration Menu. The web browser will then display a message window to confirm saved changes, press “Apply” button to confirm. Grandstream recommends reboot or power cycle the IP phone after saving changes REBOOTING FROM REMOTE Press the “Reboot” button at the bottom of the configuration menu to reboot the phone remotely.
or with certain special provisioning settings. At boot-up, Grandstream devices by default point to Grandstream provisioning server GAPS, based on the unique MAC address of each device, GAPS provision the devices with redirection settings so that they will be redirected to customer’s TFTP or HTTP/HTTPS server for further provisioning. Grandstream also provides configuration tools (Windows and Linux/Unix version) to facilitate the task of generating device configuration files.
SOFTWARE UPGRADE Software upgrade can be done via TFTP, HTTP or HTTPS. The corresponding configuration settings are in the ADVANCED SETTINGS configuration page. FIRMWARE UPGRADE THROUGH TFTP/HTTP/HTTPS To upgrade via TFTP, HTTP or HTTPS, the “Firmware Upgrade and Provisioning upgrade via” field needs to be set to TFTP, HTTP or HTTPS, respectively. “Firmware Server Path” needs to be set to a valid URL of a TFTP or HTTP server; server name can be in either FQDN or IP address format.
environment if possible. For users who do not have a local firmware upgrade server, Grandstream provides a NAT-friendly HTTP server on the public Internet for firmware upgrade. Grandstream’s latest firmware is available http://www.grandstream.com/support/firmware. Oversea users are strongly recommended to download the binary files and upgrade firmware locally in a controlled LAN environment. Alternatively, user can download a free TFTP or HTTP server and conduct local firmware upgrade.
When a Grandstream device boots up or reboots, it will issue a request for a configuration file “cfgxxxxxxxxxxxx”, where “xxxxxxxxxxxx” is the MAC address of the device, i.e., “cfg000b820102ab”. In addition, device will also requests a XML configuration file “cfgxxxxxxxxxxxx.xml”. If the download of “cfgxxxxxxxxxxxx.xml” is not successful, the provision program will issue a request for a generic configuration file “cfg.xml”. Configuration file name should be in lower case letters.
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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 There are two (2) methods for resetting your unit: RESET BUTTON Reset default factory settings following these four (4) steps: 1.
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: 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 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.
