GAC2500 - Administration Guide

Protocols/Standards

SIP RFC3261, TCP/IP/UDP, RTP/RTCP, HTTP/HTTPS, ARP, ICMP, DNS (A record, SRV, NAPTR), DHCP, PPPoE, SSH, TFTP, NTP, STUN, SIMPLE, LLDP, LDAP, TR-069, 802.1x, TLSv1.1, TLSv1.2, SRTP, IPv6, OpenVPN (pending)

Voice Codec

Support for G.711µ/a, G.722, G.726, iLBC, Opus, G.722.1 and G.722.1c (pending), in-band and out-of-band DTMF (In audio, RFC2833, SIP INFO), G.729A/B, VAD, CNG, AEC, PLC, AJB, AGC

Platform Bridging

Bridge SIP calls with any Android™ VoIP apps such as Skype™, Skype for Business (Lync), Google Hangouts™ and more

Telephony Features

Hold, transfer, forward (unconditional/no-answer/busy/conditional), call park/pickup, 7-way audio conference, auto answer, downloadable XML phone book (up to 2000 entries), LDAP, call waiting, call history (up to 2000 entries), flexible dial plan, personalized music ringtones, server redundancy & fail-over.

Sample Applications

Skype™, Google Hangouts™, Skype for Business (Lync), Web browser, Facebook™, Twitter™, YouTube, Google calendar, mobile phone data import/export via Bluetooth, etc. API/SDK available for advanced custom application development

Application Deployment

Allows Android 4.4 compliant applications to be deployed in the device with provisioning control

QoS

Layer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS

Security

User and administrator level passwords, MD5 and MD5-sess based authentication, 256-bit AES encrypted configuration file, TLSv1.1, TLSv1.2, 128/256-bit SRTP/TLS, HTTPS, 802.1x media access control

Multi-Language

English, German, Italian, French, Spanish, Portuguese, Russian, Turkish, Polish, Chinese, Korean, Japanese, and more

Upgrade/

Provisioning

Firmware upgrade via TFTP / HTTP / HTTPS or local HTTP upload, mass provisioning using TR-069 or AES encrypted XML configuration file

Network Interface

1x auto-sensing Gigabit Ethernet port, integrated 2.4GHZ auto-band Wi-Fi (802.11a/b/g/n) and 4.0 Bluetooth.

HD Audio

3x microphones (12 ft. pickup range), 1x speaker (220-18,000 Hz, 15 ft. coverage range up to 86dB)

Auxiliary Ports

3.5mm audio interface, Micro-USB interface, RJ48 interface, reset pin

Graphic Display

4.3″ (800*480) capacitive touch screen LCD

Power & Green Energy Efficiency

Universal power adapter included:

Input: 100-240VAC 50-60Hz; Output: 12VDC, 2A (24W)

Temperature and Humidity

Operating: 32 – 104oF / 0 – 40oC

Storage: 14 – 140oF / -10 – 60oC

Humidity: 10% to 90% Non-condensing

Compliance

FCC: Part 15 (CFR 47) Class B; UL 60950 (power adapter)

CE：EN55022 Class B，EN55024，EN61000-3-2，EN61000-3-3，EN60950-1，EN62479，RoHS

RCM：AS/ACIF S004；AS/NZS CISPR22/24；AS/NZS 60950；AS/NZS 4268

GAC2500

• 1 x GAC2500 Main Case.
• 1 x 12V Power Adapter.
• 1 x Ethernet Cable
• 1 x Micro-USB Cable
• 1 x RJ48 CAT5
• 1 x Quick Installation Guide
• 1 x GPL License

Parameters

Descriptions

Paired device

Displays paired devices like Bluetooth remote control, cellphone or speaker.

Available devices

Displays all devices in the search range which have enabled Bluetooth.

Parameters

Descriptions

Account

Select account.

Account Activation

Indicates whether the account is active. If disabled, the GAC2500 will not send registration information to SIP server.

Account Name

Configures the name associated with account to be displayed on the upper left corner of LCD.

SIP Server

Specifies the URL or IP address, and port of the SIP server. This is provided by your VoIP service provider (ITSP).

SIP User ID

Configures user account information provided by your VoIP service provider (ITSP). It’s usually in the form of digits similar to phone number or actually a phone number.

SIP Authentication ID

Configures the SIP service subscriber’s Authenticate ID used for authentication. It can be identical to or different from the SIP User ID.

SIP Authentication Password

Configures the account password required for the phone to authenticate with the ITSP (SIP) server before the account can be registered. After saving, it will appear as hidden for security purpose.

Voice mail access number

Sets if the phone system allows users to access the voice messages by pressing the Voicemail icon on the GAC2500. For example, in UCM6xxx IPPBX, *97 could be used.

Display Name

Specifies the SIP server subscriber’s name (optional) that will be used for Caller ID display. The configured content will be included in the “From”, “Contact” and “P-Preferred-Identity” headers of SIP INVITE message.

Show account name only

Displays the account name; if it is checked, the user ID will be hidden and the account name will be displayed only.

Tel URI

Sets this field to “” User=Phone”, If the account is assigned a PSTN number. Then a “User=Phone” parameter will be attached to the Request-Line and TO” header in the SIP request to indicate E.164 number. If set to “Enabled”, “Tel” will be used instead of “SIP”. Default is “Disabled”.

Parameters

Descriptions

Server name

UCM Server name

Server URL

UCM Sever URL

Port

UCM server port.

Access mode

Access mode to the UCM, default is HTTP.

Parameters

Descriptions

Address Type

Allows users to configure the appropriate network settings on the device. Users could select “DHCP”, “Static IP” or “PPPoE”.

IP Address

Enters the IP address when static IP is used.

Subnet Mask

Enters the subnet mask when static IP is used.

Default Router

Enters the default router when static IP is used.

DNS Server 1

Enters the DNS Server 1 address when static IP is used.

DNS Server 2

Enters the DNS Server 2 address when static IP is used.

PPPoE Account ID

Enters the PPPoE account ID when PPPoE is used.

PPPoE Password

Enters the PPPoE password when PPPoE is used.

Parameters

Descriptions

IPv6 address type

Allows users to configure the appropriate network settings on the device. Users could select “DHCP” or “Static IP”.

Static IPv6 address

Enters the IP address when static IP is used.

IPv6 prefix length

Enters the IPv6 prefix length when static IP is used.

Parameters

Descriptions

Enable/Disable

Wi-Fi

Enable/disable Wi-Fi. Once enabled, the device will search for available Wi-Fi nearby automatically and display below.

Parameters

Descriptions

Name

Identify VPN network, fill in company name or server name you are connecting to.

Type

Defines VPN type. By default, it’s PPTP (Point to Point Tunneling Protocol).

Server Address

Fill in the VPN server URL or IP address.

PPP Encryption (MPPE)

Defines whether to use PPP encryption.

Show Advanced Options

Check to display more options below.

DNS Search Domain

Defines search domain.

DNS Server

Fill in DNS Server address.

Forwarding Routes

Fill in DNS. For example,10.0.0.0/8.

Parameters

Descriptions

HTTP/HTTPS Proxy Hostname

Configures the HTTP/HTTPS proxy URI of the network.

HTTP/HTTPS Proxy Port

Configures the HTTP/HTTPS proxy port number of the network.

Bypass proxy for

Defines the specific URI that the GAC2500 can directly access to without HTTP/HTTPS proxy.

Parameters

Descriptions

LLDP

Enable or disable LLDP. The default setting is disabled.

Layer 3 QoS for SIP

This field defines the layer 3 QoS parameter for SIP packets. It is the value used for IP Precedence, Diff-Serv or MPLS. The Default value is 48.

Layer 3 QoS for Audio

This field defines the layer 3 QoS parameter for audio packets. It is the value used for IP Precedence, Diff-Serv or MPLS. The Default value is 48.

Layer 2 QoS 802.1q/VLAN Tag (Ethernet)

Assigns the VLAN Tag of the Layer 2 QoS packets for LAN port. Default value is 0.

Note: Please do not change the setting before understanding the VLAN’s settings or consulting the network administrator. Otherwise, the device might not be able to get the correct IP address.

Layer 2 QoS 802.1p Priority (Ethernet)

Assigns the priority value of the Layer 2 QoS packets. The default value is 0.

Layer 2 QoS 802.1q/VLAN Tag (Wi-Fi)

Assigns the VLAN Tag of the Layer 2 QoS packets for Wi-Fi. Default value is 0.

Layer 2 QoS 802.1p Priority (Wi-Fi)

Assigns the priority value of the Layer 2 QoS packets. The default value is 0.

802.1x mode

Allows the user to enable/disable 802.1x mode on the device. Configures 802.1x authentication when connecting to the authentication server.

The default setting is “Close”.

Identity

Enter the Identity information for the 802.1x mode.

MD5 Password

Enter the MD5 Password for the 802.1x mode.

CA Certificate

Upload the CA certificate for the 802.1x mode.

Client Certificate

Upload the client certificate for the 802.1x mode.

Private Key

Upload the private key for the 802.1x mode.

Parameters

Descriptions

Screen lock

Display paired devices like Bluetooth remote control, cellphone or speaker.

Owner info

Set up display owner info on lock screen.

Encrypt phone

Set up input digit PIN or password to unlock the GAC2500 each time you boot.

Display passwords

Check/uncheck to show/hide password as you type.

Device administrators

Add or remove device administrators.

Unknown sources

Check/uncheck to enable/disable permission to install applications that you obtain from web sites, email, or other locations other than GS Market and Google Play.

Verify apps

Disable or warn the users before apps installation.

Trusted credentials

Check/uncheck to allow/disallow applications to access secure certificates and other credentials.

Install from SD card

Install encrypted certificates from SD card.

Clear credentials

Clear credential storage of all contents and reset its password.

User Type

Username

Default Password

Accessible Web Pages

Normal User

user

123

• Call
• Contacts
• Account: Call Settings
• Advanced: MPK General Settings, MPK LCD Settings
• Maintenance: Network Settings, Time & Language, Security Settings, Device Manager
• Status: Account Status, Network Status, System Info

Administrator

admin

admin

All pages

Clear the old list

Click on the checkbox in order to clear old list.

Clear old history mode

Select either to clear all old history or clear only reserved local contacts.

Replace duplicate entries

Click the checkbox in order to replace duplicate entries.

Replace duplicate entries mode

Select how to replace duplicate entries, either by name or by number.

File encoding

Chose the encoding type of the file, following are available:

UTF-8

GBK

UTF-16

UTF-32

Big5

Big5-HKSCS

Shift-JIS

ISO8859-1

ISO8859-15

Windows-1251

EUC-KR

File type

Choose the file type from the drop down, following are available:

XML

VCard

CSV

Browse

Brows to import a contact file.

File encoding

Chose the encoding type of the file, following are available:

UTF-8

GBK

UTF-16

UTF-32

Big5

Big5-HKSCS

Shift-JIS

ISO8859-1

ISO8859-15

Windows-1251

EUC-KR

File type

Choose the file type to export to from the drop down, following are available:

XML

VCard

CSV

Export

Click Browse to select a directory where to export the contact file.

Clear the old list

Click on the checkbox in order to clear old list.

Clear old history mode

Select either to clear all old history or clear only reserved local contacts.

Replace duplicate entries

Click the checkbox in order to replace duplicate entries.

Replace duplicate entries mode

Select how to replace duplicate entries, either by name or by number.

Download mode

Choose the download mode from the drop down, available are:

Off

TFTP

HTTP

HTTPS

Note: if Off is selected remote download will be deactivated.

File encoding

Chose the encoding type of the file, following are available:

UTF-8

GBK

UTF-16

UTF-32

Big5

Big5-HKSCS

Shift-JIS

ISO8859-1

ISO8859-15

Windows-1251

EUC-KR

Download interval

Choose the download interval from the drop-down list, available are:

2 Hours

4 Hours

6 Hours

8 Hours

12 Hours

Username

Enter the username of the remote server.

Password

Enter the password of the remote server.

Download server

Enter the domain name, the IPv4 or the IPv6 address of the download server.

download

Click download to download the contact file.

Name

Enter last name and first name of the new contact.

Number

Select the account and enter the number of the new contact.

Email

Enter the email of the new contact.

Groups

Select the group where the set the new contact.

Name

Enter the name of the group.

Group Ringtone

Select a ringtone form the drop-down list.

Group Members

Select members of the group.

Number

Enter the number to black list.

Note(optional)

Add a note regarding that number.

DND mode

Select whether to activate or deactivate DND mode.

Block mode

Select to open or close the block mode.

Block options

Select a block option from the drop-down list, available are:

Block all

Block blacklist only

Block anonymous

Block anonymous and blacklist

Block notification

Set block notification to either open or close.

Time settings

Select time settings for the block mode.

Start time

Enter the start time of the block mode.

End time

Enter the end time of the block mode.

Conference name

Enter a conference name.

Members of conference

Select members of conference.

Recurring conference

Select in order to enable the conference repeat and how often to repeat the conference from the drop-down list.

Start date

Select the conference start date.

End date

Select whether to repeat the conference indefinitely or set an end date.

Conference date

If recurring conference is not selected, set the start date of the conference.

Time zone

Select the time zone.

Start time

Select the start time of the conference.

Conference duration (Min)

Select the conference duration in minutes.

Remind time

Select to send a reminder before the conference start.

Auto conference

Select to enable auto conference.

Auto answer for conference members

Select to auto answer for conference members.

Block when enter

Select to block when enter.

Block calls not from members in schedule

Select to block call not incoming from members.

Password

Enter the conference password.

Auto record

Select to auto record.

Connection Mode

Select which protocol will be used for LDAP searching, LDAP or LDAPS.

Server Address

Configures the URI of the LDAP (Lightweight Directory Access Protocol) server.

Port

Configures the LDAP server port. The default LDAP port number is 389.

Base DN

Determines the LDAP search base. This is the location in the directory where the search is requested to begin.

Example:

dc=grandstream, dc=com

ou=Boston, dc=grandstream, dc=com

User Name

Configures the bind “Username” for querying LDAP servers. Some LDAP servers allow anonymous binds in which case the setting can be left blank.

Password

Specifies the bind “Password” for querying LDAP servers. The field can be left blank if the LDAP server allows anonymous binds.

LDAP Name Attributes

Configures the “name” attributes of each record which are returned in the LDAP search result. This field allows the users to configure multiple space separated name attributes.

Example:

gn

cn sn description

LDAP Number Attributes

Configures the “number” attributes of each record which are returned in the LDAP search result. This field allows the users to configure multiple space separated number attributes.

Example:

telephoneNumber

telephoneNumber Mobile

LDAP Mail Attributes

Determines the “mail” attributes of each record which are returned in the LDAP search result.

Example:

mail

Search filed filter

Configures to filter according to which fields when LDAP search. The default setting is “Filter All”.

LDAP Name Filter

Configures the filter used for name lookups.

Examples:

(|(cn=%)(sn=%)) returns all records which has the “cn” or “sn” field starting with the entered prefix;

(!(sn=%)) returns all the records which do not have the “sn” field starting

with the entered prefix;

(&(cn=%) (telephoneNumber=*)) returns all the records with the “cn” field starting with the entered prefix and “telephoneNumber” field set.

LDAP Number Filter

Defines the filter used for number lookups.

Examples:

(|(telephoneNumber=%)(Mobile=%) returns all records which has the “telephoneNumber” or “Mobile” field starting with the entered prefix;

(&(telephoneNumber=%) (cn=*)) returns all the records with the “telephoneNumber” field starting with the entered prefix and “cn” field set.

LDAP Mail Filter

Determines the filter used for mail lookups.

Example:

(mail=%)

LDAP Displaying Name Attributes

Configures the entry information to be shown on phone’s LCD. Up to 3 fields can be displayed.

Example:

%cn %sn %telephoneNumber

Max Hits

Specifies the maximum number of results to be returned by the LDAP server, valid range from 0 to 4000. If set to 0, server will return all search results. Default setting is 50.

Search Timeout (ms)

Configures the interval (in seconds) for the server to process the request and client waits for server to return. The default setting is 4000 seconds.

LDAP Lookup For Outgoing Call

Sets the phone system to do the LDAP number searching when making outgoing calls. The default setting is “No”.

LDAP Lookup For Incoming Call

Sets the phone system to do LDAP number searching for incoming calls. The default setting is “No”.

LDAP Dialing Default Account

Configures the default account that being used when dialing LDAP contact. Users may choose the Account 1-6, the default setting is “Default”.

Outbound Proxy

Configures the IP address or the domain name of the primary outbound proxy, media gateway or session border controller. It’s used by the phone for firewall or NAT penetration in different network environments. If a symmetric NAT is detected, STUN will not work and only an outbound proxy can provide a solution

Secondary Outbound Proxy

Sets IP address or domain name of the secondary outbound proxy, media gateway or session border controller. The phone system will try to connect the Secondary outbound proxy only if the primary outbound proxy fails.

DNS Mode

Defines which DNS service will be used to lookup IP address for SIP server’s hostname. It can be selected from the dropdown list:

• A Record
• SRV
• NATPTR/SRV

If it needs DNS SRV resource, which DNS server response more than one result, it should be set to “SRV” or “NATPTR/SRV”.

The default setting is “A Record”.

DNS SRV Fail-over Mode

Configure the preferred IP mode for DNS SRV.

• If “default” is selected, first IP from query result will be applied.

• If “Saved one until DNS TTL” is selected, first IP will be applied before reaches DNS timeout.

• If “Saved one until no response” is selected, first IP will be applied even after DNS times out until it cannot response.

NAT Traversal

Specifies which NAT traversal mechanism will be enabled on the phone system. It can be selected from the dropdown list:

• NAT NO
• STUN
• Keep-alive
• UPnP
• Auto
• VPN

If the outbound proxy is configured and used, it can be set to “NAT NO”.

If set to “STUN” and STUN server is configured, the phone system will periodically send STUN message to the SUTN server to get the public IP address of its NAT environment and keep the NAT port open. STUN will not work if the NAT is symmetric type.

If set to “Keep-alive”, the phone system will send the STUN packets to maintain the connection that is first established during registration of the phone. The “Keep-alive” packets will fool the NAT device into keeping the connection open and this allows the host server to send SIP requests directly to the registered phone.

If it needs to use OpenVPN to connect host server, it needs to set it to “VPN”. If the firewall and the SIP device behind the firewall are both able to use UPNP, it can be set to “UPNP”. The both parties will negotiate to use which port to allow SIP through. The default setting is “Keep-alive”.

Proxy-Require

Adds the Proxy-Required header in the SIP message. It is used to indicate proxy-sensitive features that must be supported by the proxy. Do not configure this parameter unless this feature is supported on the SIP server.

SIP Registration

Allows the phone system to send SIP REGISTER messages to the proxy/server. The default setting is “Yes”.

Unregister before new registration

Controls whether to clear SIP user’s information by sending un-register request to the proxy server. When set to “All”, the un-registration is performed by sending a REGISTER message with “Contact” header set to * and Expires=0 parameters to the SIP server when the phone starts pre-registration after rebooting. If set to “Instance”, the phone only cleans the current SIP user’s info by sending REGISTER message with “Contact” header set to concerned SIP user’s info and Expires=0 parameters to the SIP server.

The default setting is “Instance”.

Register Expiration (m)

Configures the time period (in minutes) in which the phone refreshes its registration with the specified registrar. The default setting is 60.

The maximum value is 64800 (about 45 days).

Re-register before Expiration (s)

Specifies the time frequency (in seconds) that the phone sends re-registration request before the Register Expiration.

The default setting is 0. The range is from 0 to 64,800.

Wait Time Retry Registration (s)

Configures the time period (in seconds) in which the phone will retry the registration process in the event that is failed.

Local SIP Port

Defines the local SIP port used to listen and transmit. The default setting is 5060 for Account 1, 5062 for Account 2, 5064 for Account 3, 5066 for Account 4, 5068 for Account 5, and 5070 for Account 6.

SUBSCRIBE for MWI

Configures the phone system to subscribe voice message service. If it is set to “Yes”, the phone system will periodically send SIP SUBSCRIBE message for Message Waiting Indication service. GAC2500 phone system supports both synchronized and non-synchronized MWI.

The default setting is “No”.

Enable OPTIONS Keep Alive

Configures whether to use OPTIONS to query account registration status. If enabled, the device will send OPTIONS query message and the connection status with sever regularly to server.

The default setting is “No”.

OPTIONS Keep Alive Interval (s)

Sets the time interval the device sends the OPTIONS message to server. The default setting is 30 seconds, which means the device sends the OPTIONS message to server every 30 seconds.

The effective range is 1- 64800.

OPTIONS Keep Alive Maximum Tries

Configures the maximum times of sending OPTIONS message consistently from the phone to server. Phone will keep sending OPTIONS messages until it receives response from SIP server. The default setting is “3”, which means when the device sends OPTIONS message for 3 times, and SIP server does not respond this message, the phone will send RE-REGISTER message to register again. The valid range is 3-10.

Enable Session Timer

Allows the phone system to use the session timer, when set to “Yes”, it will be added in the SIP INVITE message to notify the server.

Session Expiration (s)

Configures the phone system’s SIP session timer. It enables SIP sessions to be periodically “refreshed” via a SIP request (UPDATE, or re-INVITE). If there is no refresh via an UPDATE or re-INVITE message, the session will be terminated once the session interval expires. Session Expiration is the time (in seconds) where the session is considered timed out, provided no successful session refresh transaction occurs beforehand. The default setting is 180. The valid range is from 90 to 64800.

Min-SE (s)

Determines the minimum session expiration timer (in seconds) if the phone act as a timer refresher. The default setting is 90.

The valid range is from 90 to 64800.

UAC Specify Refresher

Sets which party will refresh the active session if the phone makes outbound calls. If it is set to “UAC” and the remote party does not support Refresher feature, the phone system will refresh the active session. If it is set to “UAS”, the remote party will refresh it. If it is set to “Omit”, the header will be omitted so that it can be selected by the negotiation mechanism. The default setting is “Omit”.

UAS Specify Refresher

Specifies which party will refresh the active session if the phone receives inbound calls. If it is set to “UAC”, the remote party will refresh the active session. If it is set to “UAS” and the remote party does not support refresh feature, the phone system will refresh it. The default setting is “UAC”.

Force INVITE

Sets the SIP message type for refresh the session. If it is set to “Yes”, the Session Timer will be refreshed by using the SIP INVITE message. Otherwise, the phone system will use the SIP UPDATE or SIP OPTIONS message. Default is “No”.

Caller Request Timer

Sets the caller party to act as refresher by force. If set to “Yes” and both party support session timers, the phone will enable the session timer feature when it makes outbound calls. The SIP INVITE will include the content “refresher=uac”. The default setting is “No”

Callee Request Timer

Sets the callee party to act as refresher by force. If set to “Yes” and the both parties support session timers, the phone will enable the session timer feature when it receives inbound calls. The SIP 200 OK will include the content “refresher=uas”. The default setting is “No”.

Force Timer

Configures the session timer feature on the phone system by force. If it is set to “Yes”, the phone will use the session timer even if the remote party does not support this feature. If it is set to “No”, the phone will enable the session timer only when the remote party supports this feature. To turn off the session timer, select “No”. The default setting is “No”.

Enable 100rel

Activates PRACK (Provisional Acknowledgment) method. PRACK improves the network reliability by adding an acknowledgement system to the provisional Responses (1xx). It is set to “Yes”, the phone system will response to the 1xx response from the remote party. Default is “No”.

Caller ID Display

Specifies which header tag will be used from the SIP INVITE message for the Caller ID display. If it is set to Auto, the phone system will use the one of the available headers in the priority hierarchy of P-Asserted Identify Header, Remote-Party-ID Header and FROM Header. If it is set to “From Header”, it will use the FROM header information for the Caller ID. If it is set to “Disabled”, all the incoming calls Caller ID will be displayed with “Unavailable”. Default setting is “Auto”.

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 it is set to “Default”, the Privacy Header will be omitted in INVITE when “Huawei IMS” special feature is active. If set to “Yes”, it will always be presented. If set to “No”, it will always be omitted. The default setting 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 “Huawei IMS” special feature is active. If set to “Yes”, the P-Preferred-Identity Header will always be presented. If set to “No”, it will be omitted. The default setting is “Default”.

SIP Transport

Determines which network protocol will be used to transport the SIP message. It can be selected from TCP/UDP/TLS. Default setting is “UDP”.

SIP URI Scheme When using TLS

Determines the port information in the Via header and Contact header of SIP message when the phone system use TCP or TLS. If set to No, these port numbers will use the permanent listening port on the phone. Otherwise, they will use the ephemeral port for the particular connection. The default setting is “No”.

Use Actual Ephemeral Port in Contact with TCP/TLS

Determines the port information in the Via header and Contact header of SIP message when the phone system use TCP or TLS. If set to No, these port numbers will use the permanent listening port on the phone. Otherwise, they will use the ephemeral port for the particular connection. The default setting is “No”.

Symmetric RTP

Configures if the phone system enables the symmetric RTP mechanism. If it is set to “Yes”, the phone system will use the same socket/port for sending and receiving the RTP messages. The default setting is “No”.

RTP IP Filer

Receives the RTP packets from the specified IP address and Port by communication protocol. If it is set to “IP Only”, the phone only receives the RTP packets from the specified IP address based on the communication protocol; If it is set to “IP and Port”, the phone will receive the RTP packets from the specified IP address with the specified port based on the communication protocol.

The default setting is “Disable”.

Support SIP Instance ID

Determines if the phone system will send SIP Instance ID. The SIP instance ID is used to uniquely identify the device. If set to “Yes”, the SIP Register message Contact header will include +sip.instance tag.

The default setting is “Yes”

Validate Incoming SIP Messages

Specifies if the phone system will check the incoming SIP messages caller ID and CSeq headers. If the message does not include the headers, it will be rejected.

The default setting is “No”.

Check SIP User ID for Incoming INVITE

Configures the phone system to check the SIP User ID in the Request URI of the SIP INVITE message from the remote party. If it doesn’t match the phone’s SIP User ID, the call will be rejected. The default setting is “No”.

Authenticate Incoming INVITE

Configures the phone system to authenticate the SIP INVITE message from the remote party. If set to “Yes”, the phone will challenge the incoming INVITE for authentication with SIP 401 Unauthorized response. The default setting is “No”.

Only Accept SIP Requests from Known Servers

Answers the SIP request from saved servers when set to “Yes”, only the SIP requests from saved servers will be accepted; and the SIP requests from the unregistered server will be rejected.

The default setting is “No”.

Allow SIP Reset

Configures whether to allow SIP Notification message to perform factory reset on the phone.

The default setting is “No”.

SIP T1 Timeout

Defines an estimate of the round-trip time of transactions between a client and server. If no response is received in T1, the figure will increase to 2*T1 and then 4*T1. The request re-transmit retries would continue until a maximum amount of time define by T2.

The default setting is 0.5 sec.

SIP T2 Interval

Specifies the maximum retransmit time of any SIP request messages (excluding the SIP INVITE message). The re-transmitting and doubling of T1 continues until it reaches the T2 value.

The default setting is 4 sec.

SIP Timer D Interval

Defines the amount of time that the server transaction can remain when unreliable response (3xx-6xx) received. The valid value is 0-64 seconds.

The default value is 0.

Remove OBP from Route

Configures the phone system to remove the outbound proxy URI from the Route header. This is used for the SIP Extension to notify the SIP server that the device is behind a NAT/Firewall. If it is set to “Yes”, it will remove the Route header from SIP requests.

The default setting is “No”.

Check Domain Certificates

Sets the phone system to check the domain certificates if TLS/TCP is used for SIP Transport.

The default setting is “No”.

Validate Certification Chain

Configures whether to validate certification chain, when TLS/TCP is configured for SIP Transport. If this is set to “Yes”, phone will validate server against the new certificate list. The default setting is “No”.

Auto-filling Pickup Feature Code

Sets if the phone system will add the configured feature code to retrieve parked calls. If it is set to “Yes”, the “Pickup Feature Code” configured will be automatically prepended on the phone’s dial pad when retrieving the parked call.

The default setting is “Yes”.

Pickup Feature Code

Configures the pickup feature code for call park. If “Auto-filling Pickup Feature Code” is set to “Yes”, this pickup feature code will be automatically filled in on the phone’s dial pad when picking up the parked call.

Preferred Vocoder

Lists the available and enabled Video codecs for this account. Users can enable the specific video codecs by moving them to the selected box and set them with a priority order from top to bottom. This configuration will be included with the same preference order in the SIP SDP message.

Hide Vocoder

Hide vocoder information on call screen.

Codec Negotiation Priority

Configures the phone to use which codec sequence to negotiate as the callee. When set to “Caller”, the phone negotiates by SDP codec sequence from received SIP Invite; When set to “Callee”, the phone negotiates by audio codec sequence on the phone. The default setting is “Callee”.

Use First Matching Vocoder in 200OK SDP

Configures the phone to use the first matching codec in the 200OK message.

The default value is 0.

Enable Audio RED with FEC

Enables FEC for audio calls, default setting is “No”.

iLBC Frame Size

Sets the ILBC (Internet Low Bitrate Codec) frame size if ILBC is used. Users can select it from 20ms or 30ms. The default setting is 30ms.

2 ITU Payload

G726-32 payload type for ITU packing mode. Payload 2 is static and payload dynamic is dynamic.

G726-32 Dynamic PT

Specifies the G726-32 payload type, and the valid range is 96 to 127. The default setting is “126”.

Opus Payload Type

Defines the desired value (96-127) for the payload type of the Opus codec. The default value is 123.

G.722.1 Rate

Supports 24kbps or 32kbps, please confirm it with your service provider. The default setting is 24kbps.

G.722.1 Payload Type

Defines the payload, the current range is 100-126. Default setting is 104.

DTMF

Specifies the mechanism to transmit DTMF (Dual Tone Multi-Frequency) signals. There are 3 supported modes: in audio, RFC2833, or SIP INFO.

• In audio, which means DTMF is combined in the audio signal (not very reliable with low-bit-rate codecs);

• RFC2833, which means to specify DTMF with RTP packet. Users could know the packet is DTMF in the RTP header as well as the type of DTMF;

• SIP INFO, which uses SIP INFO to carry DTMF. The defect of this mode is that it’s easily to cause desynchronized of DTMF and media packet if the SIP and RTP messages are required to transmitted respectively.

The default setting is “RFC2833”.

DTMF Payload Type

Configures the RTP payload type that indicates the transmitted packet contains DTMF digits. The valid range is from 96 to 127. The default value is 101.

SRTP Mode

Sets if the phone system will enable the SRTP (Secured RTP) mode. It can be selected from dropdown list:

• Disable
• Enabled but not forced
• Enabled and forced

SRTP uses encryption and authentication to minimize the risk of denial of service. (DoS). If the server allows to use both RTP and SRTP, it should be configured as “Enabled but not forced”. Default setting is “Disable”.

SRTP Key Length

Configures all the AES (Advanced Encryption Standard) key size within SRTP. It can be selected from dropdown list:

• AES128&256 bit

• AES 128 bit

• AES 256 bit

If it is set to “AES 128&256 bit”, the phone system will provide both AES 128 and 256 cipher suites for SRTP. If set to “AES 128 bit”, it only provides 128-bit cipher suite; if set to “AES 256 bit”, it only provides 256-bit cipher suite.

The default setting is “AES128&256 bit”.

Enable SRTP key life time

Defines the SRTP key life time. When this option is set to be enabled, during the SRTP call, the SRTP key will be valid within 231 SIP packets, and phone will renew the SRTP key after this limitation. The default setting is “Yes”.

Silence Suppression

Enables the silence suppression/VAD feature. If it is set to “Yes”, when silence is detected, a 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. The default setting is “No”.

Voice Frames per TX

Configures the number of voice frames transmitted per packet. When configuring this, it should be noted that the “ptime” value for the SDP will change with different configurations here. This value is related to the codec used and the actual frames transmitted during the in-payload call. For end users, it is recommended to use the default setting, as incorrect settings may influence the audio quality. The default setting is 2.

RTCP Destination

Configures the server address. When there is a call, the RTCP package sent from the phone will also be sent to this address.

Server Address format must be IP:Port. The port range is 0-65535.

Note: The address should contain port number. A prompt message will be displayed if the port number is not specified.

BLF Call-pick Prefix

Configures the prefix prepended to the BLF extension when the phone picks up a call with BLF key. The default setting is ** for each account.

Event List URI

Configures the event list BLF URI on the phone to monitor the extensions in the list with multi-purpose keys. The server side has to support this feature. Users need to configure an event list BLF URI on the service side first (i.e., BLF1006@myserver.com) with a list of extensions included. On the phone, in this “event list BLF URI” field, fill in the URI without the domain (i.e., BLF1006).

Force BLF Call-pickup by Prefix

Uses the prefix for BLF Call-pickup. The default setting is “No”.

BLF List 

Displays the configured MPK keys. Users could change the keys order by selecting “Up”, “Down”, “Check All”, or “Delete

Key Mode

Assigns a function to the Multi-Purpose Key in the MPK app

The key mode options are:

• Speed Dial

Press to dial the extension configured in UserID field.

• Busy Lamp Field (BLF)

Monitor the extension status as configured in UserID field.

• Call Transfer

Transfer the current active call to the extension configured in UserID field.

• Call Intercom

Intercom/paging to the extension configured in UserID field.

• Speed Dial via Active Account

Similar to Speed Dial but it will dial based on the current active account. For example, if the phone is off-hook and account 2 is active, it will call the configured extension using account 2.

• Dial DTMF

Dial the DTMF digits filled in UserID field during a call.

• Call Park

Configure the call park feature code to park or pick up the call.

• Multicast Paging

Configure the IP address and port for establish a multicast paging call.

• Group Call

Set up to 6 numbers for an account to start a conference. Users need to add the member numbers into a list.

Account

Selects the SIP Account used for the multi-purpose keys.

Name

Configures the display name for the multi-purpose key.

User ID

Configures the User ID for the corresponding multi-purpose key mode.

Display Format

Configures the display format for the multi-purpose keys. Users could select display “Name”, “UserID”, or “UserID(Name)”. The “Name” is the name saved in GAC2500 Contacts. The default setting is “Name”.

Show Display Name from Server

If check, the name on server will replace the one user configured before. The default setting is “No”.

Preferred Internet Protocol

Selects one of the following IP protocol modes:

• IPv4 Only: Enforce IPv4 protocol only.
• IPv6 Only: Enforce IPv6 protocol only.
• Both, Prefer IPv4: Enable both IPv4 and IPv6 and prefer IPv4.
• Both, prefer IPv6: Enable both IPv4 and IPv6 and prefer IPv6.

Note: Make sure to reboot the phone for the changes to take effect.

Address Type

Allows users to configure the appropriate network settings on the phone. Users could select “DHCP”, “Static IP” or “PPPoE”.

DHCP VLAN Override

Host name (Option 12)

Specifies the name of the client.

This field is optional but may be required by some Internet Service Providers.

Vendor Class ID (Option 60)

Used by clients and servers to exchange vendor class ID.

IP Address

Enter the IP address when static IP is used.

Subnet Mask

Enter the Subnet Mask when static IP is used.

Default Gateway

Enter the Default Gateway when static IP is used.

DNS Server 1

Enter the DNS Server 1 address when static IP is used.

DNS Server 2

Enter the DNS Server 2 address when static IP is used.

PPPoE Account ID

Enter the PPPoE account ID.

PPPoE Password

Enter the PPPoE Password.

Alternate DNS Server

Configures alternate DNS server.

Second Alternate DNS Server

Configures secondary alternate DNS server.

Enable LLDP

Enable / Disable LLDP.

LLDP TX Interval(s)

Configures the interval the phone sends LLDP-MED packet, the default setting is 30s.

Layer 3 QoS for SIP

Defines the Layer 3 QoS parameter for SIP packets. This value is used for IP Precedence, Diff-Serv or MPLS. Default value is 26.

Layer 3 QoS for Audio

Defines the Layer 3 QoS parameter for Audio packets. This value is used for IP Precedence, Diff-Serv or MPLS. Default value is 46.

Layer 2 QoS 802.1Q/VLAN Tag (Ethernet)

Assigns the VLAN Tag of the Layer 2 QoS packets for LAN port. The default value is 0. Note: Please do not change the setting before understanding the VLAN’s settings, or the phone can’t get the correct IP address.

Layer 2 QoS 802.1p Priority Value (Ethernet)

Assigns the priority value of the Layer2 QoS packets for LAN port. The default value is 0.

Layer 2 QoS 802.1Q/VLAN Tag (Wi-Fi)

Assigns the VLAN Tag of the Layer 2 QoS packets for Wi-Fi. The default value is 0. Note: Please do not change the setting before understanding the VLAN’s settings, or the phone can’t get the correct IP address.

Layer 2 QoS 802.1p Priority Value (Wi-Fi)

Assigns the priority value of the Layer2 QoS packets for Wi-Fi. The default value is 0.

HTTP/HTTPS User-Agent

This sets the user-agent for phonebook and screen saver

SIP User-Agent

Sets the user-agent for SIP.

802.1x mode

Enables and selects the 802.1x mode for the phone system. The supported 802.1x modes are:

• EAP-MD5
• EAP-TLS
• EAP-PEAP

The default setting is “Disable”.

802.1x Identity

Enters the identity information for the selected 802.1x mode.

(This setting will be displayed only if 802.1 X mode is enabled).

802.1x Secret 

Enters the secret for the 802.1x mode. This option will be appeared when 802.1x mode is EAP-MD5 or EAP-PEAP.

CA Certificate

Uploads the CA Certificate file to the phone. (This setting will be displayed only if the 802.1 X mode is enabled)

Client Certificate

Loads the Client Certificate file to the phone. (This setting will be displayed only if the 802.1 X TLS mode is enabled)

Private Key

Loads the private key file to the phone. (This setting will be displayed only if the 802.1 X TLS mode is enabled)

HTTP/HTTPS Proxy Hostname

Specifies the HTTP/HTTPS proxy hostname for the phone to send packets to.

HTTP/HTTPS Proxy Port 

Specifies the HTTP/HTTPS proxy port.

Bypass Proxy For

Defines the list of destination IP addresses where no proxy server is needed. Multiple IP addresses can be separated by “,”.

IPv6 address

Select either Statically configured or Auto configured address.

Static IPv6 address

Set the Static IPv6 address.

IPv6 prefix length

Enter the IPv6 prefix length, default value is 64.

DNS server 1

Enter DNS server 1 when static IPv6 is used.

DNS server 2

Enter DNS server 2 when static IPv6 is used.

Preferred DNS Server

Enter the preferred DNS server when static IPv6 is used.

Wi-Fi Basics

Wi-Fi Function

Enables/disables the Wi-Fi function. The default setting is set to “No”.

ESSID

Permits to scan and select the available Wi-Fi networks within the range if the WiFi feature is enabled. Press “Scan” to scan for the available wireless network. The number in brackets represents the signal intensity.

Wi-Fi Security

ESSID

Display the selected ESSID name.

Password

Configures the selected ESSID password.

Security Mode for Hidden SSID

Defines the security mode used for the wireless network when the SSID is hidden.

Specify NTP Server Address

Defines the URL or IP address of the NTP server. The phone may obtain the current date and time information from the server. The default setting is “pool.ntp.org”.

DHCP Option 42 override NTP server

Defines whether DHCP Option 42 should override NTP server or not. When enabled, DHCP Option 42 will override the NTP server if it’s set up on the LAN. The default setting is “Yes”.

DHCP Option 2 to override Time Zone setting

Allows device to get provisioned for Time Zone from DHCP Option 2 in the local server automatically. If it set to “Yes”, the DHCP offer with Option 2 will override the phone system’s time zone setting. The default setting is “No”.

Time Zone

Specifies the local time zone for the phone. It covers the global time zones and user can select the specific one from the drop-down list.

Use 24-hour format 

Use 24-hour time display format.

Date Display Format

Configures the date display format on the LCD.

The following formats are supported:

Normal (M/DD/YYYY): 1/31/2012

YYYY/MM/DD: 2012/01/31

MM/DD/YYYY: 01/31/2012

DD/MM/YYYY: 31/01/2012

The default setting is DD/MM/YYYY.

Language

Configures the language on device.

Emergency Call Numbers

Configures emergency call numbers.

Disable SSH

Blocks any SSH access to the phone. The default setting is “No”.

Disable web GUI

If set to “Yes”, the GAC will not allow any web access to it. The default value is “No”.

Access Method

Determines which protocol will be used to access the phone ‘s Web GUI. It can be selected from HTTP and HTTPS. The default setting is HTTP.

Port 

Specifies which port to use to access the phone ‘s Web UI. By default, if HTTP, the port number will be 80; if HTTPS is selected, the port number will be 443.

Admin Password

Configures the administrator password for Web GUI. It is case sensitive with a maximum length of 32 characters. Default password is “admin”.

Confirm Admin Password

Confirms the new admin password.

User Password 

Configures the user password for Web GUI. It is case sensitive with a maximum length of 32 characters. The default password is “123”.

Confirm User Password 

Confirms the new admin password.

User Authority Management

If not checked, then use the system default user authority configuration; If checked, you can click on the “Configure Permission” button to select authority. The default setting is “No”.

Configuration via Keypad Menu

Configures access control for keypad Menu settings on the Settings interface of the phone.

• Unrestricted: configure all settings on the Settings interface;

• Basic Settings Only: The Advanced Settings option will not be displayed;

• Constraint Mode: users need to input admin user password to configure Wireless & Network and Advanced Settings.

Warning: If the admin password is lost while constraint mode is enabled, your device may become permanently unusable. Remember to be careful when using constraint mode to avoid irreversible damage.

Note: When access control for keypad is limited to “Basic Settings Only” or “Constraint Mode”, the Admin authentication will be mandatory to start Factory Reset process.

SIP TLS Certificate

Defines the SSL private key used for SIP over TLS. The SSL private key the user specified for TLS encryption should be X.509 format.

SIP TLS Private Key

Defines the SSL private key used for SIP over TLS. The SSL private key the user specified for TLS encryption should be X.509 format.

SIP TLS Private Key Password

Defines the SSL private key password used for SIP over TLS.

Custom Certificate

This feature allows user to upload custom certificate. The priority of certificate is SIP TLS/Certificate/Private Key (Only SIP TLS used) > Custom Certificate > Individual Certificate > General Certificate.

Import Custom Certificate

Uploads certificate file from PC using BROWSE button. The certificate file should be “.pem”, “.crt”, and “.cer” or “.der” format. The Browse button will turn to “Processing” and back to “Browse” once the uploading finished. Note: Users can also delete uploaded custom certificate by clicking on “Delete”.

Import Trusted CA Certificates

Clicks the Browse to upload certificate file from PC. The certificate file should be “.pem”, “.crt”, “.cer”, or “.der” format. The Browse button will turn to “Processing” and back to “Browse“ once the upload finished.

Always send HTTP Basic Authentication Information

This setting is to enable/disable send HTTP basic authentication feature when the phone user wget package to download “cfg.xml” file. If set to “Yes”, the phone will always send HTTP with credentials. Otherwise, the phone only sends the HTTP with credentials only when the server requests.

The default is setting is “No”.

Validate Certificate Chain

Configures whether validate server certificate. If check, the phone will download firmware/config file from legal server. Default is setting is No.

mDNS Override Server

If set to “Use Type A” or “Use Type SRV, allows mDNS override config/firmware server setting on the phone if present. The default setting is “User Type A”.

Allow DHCP Option 43 and Option 66 to Override Server

It is used to set if the phone system allows the DHCP offer message to override the Config Server Path via the Option 66 header. The phone system supports both TFTP and HTTP method via Option 66. The default setting is “Yes”.

DHCP Option 120 Override SIP Server 

Override the Config Server Path via the Option 120 header. The default setting is “Yes”.

Firmware Upgrade Via

Allows users to choose the firmware upgrade method: TFTP, HTTP, HTTPS or Manual Upload.

The default setting is “HTTP”.

Firmware Server Path

Sets IP address or domain name of firmware server. The URL of the server that hosts the firmware release.

Default is fm.grandstream.com/gs.

Firmware HTTP/HTTPS User Name

Enters the user name for the firmware HTTP/HTTPS server.

Firmware HTTP/HTTPS Password

Enters the password for the firmware HTTP/HTTPS server.

Firmware File Prefix

This feature allows ITSP to load the firmware with specific prefix to the phone. Only the firmware with the matching encrypted prefix/postfix will be downloaded and flashed into the phone. This setting is for ITSP to configure.

Firmware File Postfix

This feature allows ITSP to load the firmware with specific postfix to the phone. Only the firmware with the matching encrypted prefix/postfix will be downloaded and flashed into the phone. This setting is for ITSP to configure.

Use Grandstream GAPS

When checked GAPS server will be used for provisioning redirection.

Config file Upgrade Via

Selects provisioning method: TFTP, HTTP or HTTPS. Default setting is “HTTPS”.

Config Server Path

Sets IP address or domain name of configuration server. The server hosts a copy of the configuration file to be installed on the GAC2500. Default is fm.grandstream.com/gs.

Config File HTTP/HTTPS User Name

If your HTTP/HTTPS config server has enabled user authentication mode, please fill in user name for authentication.

Config File HTTP/HTTPS Password

If your HTTP/HTTPS config server has enabled user authentication mode, please fill in password for authentication.

Config File Prefix

Checks if configuration files are with matching prefix before downloading them. This field enables user to store different configuration files in one directory on the provisioning server.

Config File Postfix

Checks if configuration files are with matching postfix before downloading them. This field enables user to store different configuration files in one directory on the provisioning server.

XML Config File Password

Decrypts XML configuration file when encrypted. The password used for encrypting the XML configuration file is using OpenSSL.

Authenticate Conf File

Sets the phone system to authenticate the configuration file before applying it. If it set to “Yes”, the configuration file must include value P1 with phone system’s administration password. If it is missed or does not match the password, the phone system will not apply it. The default setting is “No”

Download Device Configuration

Click to download the device configuration file to PC.

Upload Device Configuration

Click to upload the configuration file on PC to device.

Cust File Upgrade Via

Selects download method: TFTP, HTTP or HTTPS. Default is “HTTP”.

Cust Config Server Path

Sets IP address or domain name of the GUI customization file server. The server hosts a copy of the file to be installed on the phone. The Default setting is fm.grandstream.com/gs.

Cust File HTTP/HTTPS User Name

Enters the username for the firmware HTTP/HTTPS server.

Cust File HTTP/HTTPS Password

Enters the password for the firmware HTTP/HTTPS server.

Use Configurations of Config File Server

Retrieve and download customization file with the configuration of the config file.

Enable PNP Feature

Enables the PNP (Plug and Play) feature on the device. If it is enabled, the device will be set as a provision server to send SIP NOTIFY message including the provision URL to response the client phone’s SIP SUBSCRIBE request. This feature will be enabled if the PNP URL is configured. If this setting is enabled, the 3CX Auto provision will be disabled automatically.

PNP URL

Configures the URL to provision another client phone’s config server path. The URL will be included in the SIP NOTIFY message.

PNP(3CX) Auto Provision

Sets the phone system to broadcast the SIP SUBSCRIBE message during booting up to allow itself to be discovered and be configured by the SIP platform. The default setting is “Yes”.

Automatic Upgrade

It is used to set if and how often the phone system will check the server for new firmware and configuration file downloading. It can be selected from following options: (The default setting is “No”)

• “No”
• “Check Every Day”
• “Check Every Week”
• “Check at a period of time”

Enable Randomized Automatic Upgrade

If set to “Yes”, the phone will trigger the upgrade process automatically at a random time point within the time window configured without manual trigger.

Hour of the Day (0-23)

Defines at which hour of the day they phone system will check the HTTP/HTTPS/TFTP server for firmware upgrades or configuration files changes.

Day of the Week (0-6)

Defines which day of the week the phone system will check the HTTP/HTTPS/TFTP server for firmware upgrades or configuration files changes

Automatic Upgrade Check Interval (m)

Configures how the phone system will check the server for new firmware and configuration file downloading. It only valid if the user selects “Check at a period of time” in the “Automatic Upgrade”. The default setting is 10080 (namely 7 days).

Firmware Upgrade and Provisioning

Defines the rules for automatic upgrade and configuration file: Always Check at boot up, when Firmware prefix/suffix changes, Skip the Firmware Check.

Disable SIP NOTIFY Authentication

Device will not challenge NOTIFY with 401 when set to “Yes”.

Auto Reboot to Upgrade Without Prompt

If set to “Yes”, the phone will automatically start upgrading after downloading the firmware files. Otherwise, users would need confirm in the prompted message in LCD to start upgrading process. The default value is “Yes”.

Factory Reset 

It is used to reset the phone system to the default factory setting mode. If the “Clear the SD card” is checked, the SD card storage mounted to the phone will be format as well.

Call

Rename, download, and delete the recording that is saved in a call or a conference.

Normal

Rename, download, and delete the recording that is saved in Recorder app.

Syslog

Syslog Protocol

Choose whether to send syslog messages over UDP or secured SSL/TLS connection to Syslog Server. Default is UDP.

Notes:

• User can upload the CA file under Maintenance 🡪 Security settings.
• A reboot is required in order for the changes to take effect.

Syslog Server Address

Configures the URI, IPv4 or IPv6 which the phone system will send the syslog messages to. The default setting is “log.ipvideotalk.com”.

Note: The address Max Length is 128 characters.

Syslog Level

Selects the level of logging for syslog. The default setting is “None”.

There are 4 levels from the dropdown list: DEBUG, INFO, WARNING and ERROR. The following information will be included in the syslog packet:

• DEBUG (Sent or received SIP message,)
• INFO (Product model/version on boot up, NAT related info, SIP message summary, Inbound and outbound calls, Registration status change, negotiated codec, Ethernet link up)
• WARNING (SLIC chip exception)
• ERROR (SLIC chip exception, Memory exception)

Send SIP Log

Configures whether the SIP log will be included in the syslog messages. The default setting is “No”.

Syslog keyword filter

Configures syslog message to be filtered based on keywords provided. If multiple keywords are filled, it should be separated by “,”.

Note: No spaces allowed.

Logcat

Clear Log

Clears the log files saved in the phone system.

Log Tag

Configures the filter to display the specified process log file.

Log Priority

Selects the log priority to display. It can be selected from list below:

• Verbose (Default Setting)
• Debug
• Info
• Warn
• Error
• Fatal
• Silent (suppress all output)

Debug

One-Click Debugging

One-click Debugging

Capture the checked info in the debugging list, click “Start” to debug if including “Capture trace” item and click “Stop” to end. If “Capture Trace” is not checked “Start” button will change to “Capture”. All retrieved files will be generated to a package, and the last package will be overwritten, while the trace file will remain.

Debug Info Menu

Display a list of info items that can be debugged, currently supports Syslog, Logcat and Capture trace. The captured data can be viewed in “Debug information list”. The default is all selected.

Debug Info List

You can select the existing debugging info package or grab package. Click the “Delete” button on the right to delete the file.

View Debug Info

You can select the existing debugging info package or grab package. Click the “Delete” button on the right to delete the file.

Record

Record

Click to start capturing audio data, click the “Stop” button to end. To capture the audio data of the device can help to locate audio issues. The default is not enabled. You can record up to 1-minute audio data.

Recording List

Choose the existing audio file. Click the “Delete” button on the right to delete this file.

View Recording

Click on the “List” button to view. The captured audio data will be sorted by time. Click to download the data to the computer for analysis.

Core Dump

Enable Core Dump Generation

Configures whether to generate and save the core dump file when the program crashes. The default setting is “No”. This feature requires a reboot to take effect.

Core Dump List

Selects the existing core dump file in the drop-down box. Users could delete the file by pressing on “Delete” button.

View Core Dump

Press “List” button to view all existing core dump files. The files are listed in chronological order, users could click the file name to download it to local computer.

Traceroute

Target Host

Input the domain name or IP address and click “Start to show trout tracing result in page below.

Enable TR-069

Sets the phone system to enable the “CPE WAN Management Protocol” (TR-069). The default setting is “No”.

ACS URL

Specifies URL of TR-069 ACS (e.g., http://acs.mycompany.com), or IP address

ACS Username

Enters username to authenticate to ACS.

ACS Password

Enters password to authenticate to ACS.

Enable Periodic Inform

Sends periodic inform packets to ACS. Default is “No”.

Periodic Inform Interval (s)

Configures to sends periodic “Inform” packets to ACS based on specified interval.

Connection Request Username 

Enters user name for the ACS to connect to the phone

Connection Request Password

Enters password for the ACS to connect to the phone.

Connection Request Port 

Enters the port for the ACS to connect to the phone.

CPE Cert File

Uploads Cert File for the phone to connect to the ACS via SSL.

CPE Cert Key 

Uploads Cert Key for the phone to connect to the ACS via SSL.

Account

All the SIP accounts on the phone. Each account will show its status in this page.

Number

SIP User ID for the account.

SIP Server

URL or IP address, and port of the SIP server.

Status

Registration status for the SIP account.

MAC Address

Global unique ID of device, in HEX format. The MAC address will be used for provisioning and can be found on the label coming with original box and on the label located on the back of the device.

NAT Type

The type of NAT connection used by the phone.

IPv4

Address Type

The configured address type: DHCP, Static IP or PPPoE.

IPv4 Address

IP address of the phone.

Subnet Mask

Subnet mask of the phone.

Gateway

Default gateway of the phone.

DNS Server 1

DNS Server 1 of the phone.

DNS Server 2

DNS Server 2 of the phone.

IPv6

IPv6 Address Type

The configured address type: DHCP, Static IP.

IPv6 Address

IPv6 address of the device.

IPv6 DNS Server 1

IPv6 DNS Server 1 of the device.

IPv6 DNS Server 2

IPv6 DNS Server 2 of the device.

Product Model

Product model of the phone.

Hardware Revision

Hardware version number.

Part Number

Product part number.

System Version

Firmware version. This is the main software release version.

Recovery Version

Recovery image version.

Boot Version

Booting code version.

Kernel Version

Kernel version.

Android^TM Version

Android^TM OS version: 4.4.2.

System Up Time

System up time since the last reboot.

Code

Feature

*01

Select the preferred codec used for the call (One-time Only)

Dial *01+preferred codec+Phone/Ext. Number

• PCMU preferred codec：7110
• PCMA preferred codec：7111
• G726.32 preferred codec：72632
• G722 preferred codec：722
• ILBC preferred codec：7201

*02

Force the unique codec used for the call.

Dial *02+preferred codec+Phone/Ext. Number

*16

Force SRTP used for the call.

Dial *16

*17

Disable SRTP

Dial *17

*18

Enable SRTP (One-time Only)

Dial *18+Phone/Ext. Number

*19

Disable SRTP (One-time Only)

Dial *19+Phone/Ext. Number

*30

Anonymous Call (For all subsequent calls)

Dial *30

*31

Cancel Anonymous (For all subsequent calls)

Dial *31

*50

Disable Call Waiting (For all subsequent calls)

Dial *50

*51

Call Waiting (For all subsequent calls)

Dial *51

*67

Selective Anonymous Call (Current Call)

Dial *67+Phone/Ext. Number

Dial

*70

Disable Call Waiting (For all subsequent calls)

Dial *70+Phone/Ext. Number

Dial

*71

Enable Call Waiting (For all subsequent calls)

Dial *77+Phone/Ext. Number

Dial

*72

Unconditional Call Forward: Set up unconditional call forward

Dial *72 + Phone/Ext. Number.

Dial

*73

Cancel Unconditional Call Forward: Cancel unconditional call forward

Dial *73

*74

Enable paging mode directly when dialing up

Dial *74

*82

Selective Cancel Anonymous Call (Current Call)

Dial *82+Phone/Ext. Number

Dial

*90

Busy Call Forward: Set up busy call forward

Dial *90 + Phone/Ext. Number.

Dial

*91

Cancel Busy Call Forward: Cancel busy call forward

Dial*91

*92

Delayed Call Forward: Set up delayed call forward

Dial *92 + Phone/Ext. Number.

Dial

*93

Cancel Delayed Call Forward: Cancel busy call forward

Dial*93