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Grandstream BT200

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About

Grandstream BT200Grandstream BudgeTone 200 VoIP phone

LCD display - monochrome

Grandstream's BudgeTone-200 SIP IP phone is the innovative IP telephone that offers a rich set of functionality and superb sound quality. It is fully compatible with SIP industry standard and can interoperate with many other SIP compliant devices and software on the market.

Here you can find all about Grandstream BT200, for example manual and price, firmware, review. You can also write a review.

 

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User reviews and opinions

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Comments to date: 3. Page 1 of 1. Average Rating:
ma102ma 10:05pm on Monday, September 6th, 2010 
Nice Inexpensive SIP VoIP Phone Station Set I bought two of these units for a little experiment - I have a crossover RJ45 cable so the two VoIP SIP ph...
vasudevan 12:15pm on Wednesday, September 1st, 2010 
Our company recently purchsed few BT200 Bluetooth printer adapters for our portable printers to enable Bluetooth function.
Schuluniform 1:58pm on Thursday, April 8th, 2010 
This is a awesome product that in my home office I can print thur my laptop in the different room.

Comments posted on www.ps2netdrivers.net are solely the views and opinions of the people posting them and do not necessarily reflect the views or opinions of us.

 

Documents

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Table 4: BT200 Technical Specifications Protocol Support Device Management Support SIP 2.0, TCP/UDP/IP, PPPoE, RTP/RTCP/SRTP, HTTP, ARP/RARP, ICMP, DNS, DHCP, NTP, TFTP. NAT-friendly remote software upgrade (via TFTP/HTTP) for deployed devices including behind firewall/NAT Auto/manual provisioning system, GUI Interface Support Layer 2 (802.1Q, VLAN, 802.1p) and Layer 3 QoS (ToS, DiffServ, MPLS) Full-duplex hands-free speakerphone, headset enabled Advanced Digital Signal Processing (DSP) Dynamic negotiation of codec and voice payload length Support for PCMU, PCMA, GSM, G.723.1, G.729A/B, G.726-32, G.722 (wide band), iLBC 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) Acoustic Echo Cancellation (AEC) with Acoustic Gain Control (AGC) for speakerphone mode, Support side tone Adaptive jitter buffer control (patent-pending) and packet delay & loss concealment Voice mail indicator, downloadable custom ring-tones, call hold, call transfer (attended/blind), call forward, call waiting, caller ID, mute, redial, call log, caller ID display or block and volume control 3-way-conference, off-hook auto dial, auto answer, early dial and speed dial Via keypad/LCD, Web browser, or secure (AES encrypted) central configuration file, manual or dynamic host configuration protocol (DHCP) network setup
BT200 User Manual Firmware 1.1.6.44 Page 8 of 33 Last Updated: 12/2008

Audio Features

Telephony Features

Network and Provisioning

Firmware Upgrades Advanced Server Features Security
Support NAT traversal using IETF STUN and Symmetric RTP Support for IEEE 802.1p/Q tagging (VLAN), Layer 3 TOS Support firmware upgrade via TFTP or HTTP, Support for Authenticating configuration file before accepting changes User specific URL for configuration file and firmware files Message waiting indication, support DNS SRV Look up and SIP Server Fail Over DIGEST authentication and encryption using MD5 and MD5-sess, SRTP, SIP over TLS (pending)
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Using the GXP SIP Enterprise Phone GETTING FAMILIAR WITH THE LCD
BT200 phone has a numeric LCD of 64mmx24mm size with backlight. This model has a small red LED status reminder. Here is the display when all segments illuminate: Figure 5: BT200 LCD
When the phone is in the normal idle state, the backlight is off. Whenever an event (call) occurs, the backlight will turn on automatically to bring the users attention. In addition, if Voice Mail configured and there is a VM waiting, the red LED will be blinking to remind user there is a Voice Mail in the Voice Mail server. Table 5: LCD Icons

LCD Icon Definitions

Connectivity Status / SIP Proxy/Server Icon: OFF IP address of Sip server is not found ON IP address of Sip server are located Blinking Ethernet link failure or the phone is not registered properly Phone Status Icon: OFF when the handset is on-hook ON when the handset is off-hook Speaker Phone Status Icon: FLASH when phone rings OFF when the speakerphone is off ON when the speakerphone is on Handset and Speakerphone/Headset Volume Icons: 0-7 scales to adjust handset / speakerphone volume
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Realtime Clock: Synchronized to Internet time server Time zone configurable via web browser AM/PM indicator Call Logs: 01-10 for CALLED history (dialed number) 01-10 for CALLERS history (Incoming caller ID)
Time Icon: AM for the morning PM for the afternoon
IP Address Separator Icons
Numerical Numbers and Characters: 0-9 *= #= A, b, C, c, d, E, F, G, g, H, h, I, L, n, O, o, P, q, r, S, t, U, u, Y
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GETTING FAMILIAR WITH KEYPAD
LCD Up/Down Keys Message Waiting Indicator Menu Button Called Message Callers Hold Transfer Conference Flash Standard Keypad

Speaker

Mute/Delete
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Table 6: BT200 Keypad Buttons Key Button 0 - 9, *, # Key Button Definitions
Digit, star and pound keys are usually used to make phone calls 1) Reduce handset, speakerphone/headset volume after off hook the phone via handset or speaker 2) Reduce ring tone volume when phone in IDLE and off hook to confirm the changed ring tone volume 3) Next menu item browsing when phone is in IDLE mode after MENU key pressed, off hook to interrupt and exit Enter keypad MENU mode when phone is in IDLE mode. It is also the ENTER key once entering MENU After off hook, press to display the dialed numbers. When number displayed, press the SEND key can make call using that displayed number After off hook, press to display the incoming Caller IDs. When number displayed, press the SEND key can make call using that displayed number Enter to retrieve voice mails from Voice Mail Portal or Server Temporarily hold the active call Transfer the active call to another party Establish 3-way conferencing call Flash event to switch between two lines Mute an active call; or Delete a key entry Also used to REJECT incoming call. Dial a new number inputted or Redial the number last dialed. After entering the phone number, pressing this key would force a call to go out immediately before timeout Enter hands-free mode 1) Increase handset, speakerphone/headset volume after off hook the phone via handset or speaker 2) Increase ring tone volume when phone in IDLE and off hook to confirm the changed ring tone volume 3) Next menu item browsing when phone is in IDLE mode after MENU key pressed, off hook to interrupt and exit

MENU CALLED CALLERS MESSAGE HOLD TRANSFER CONFERENCE FLASH MUTE/DEL SEND/(RE)DIAL SPEAKERPHONE
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MAKING PHONE CALLS

Handset, Speakerphone and Headset Mode
The regular Handset mode can be switched with either the Speaker mode (Hand free) or the Headset mode, however, whenever the Headset is plugged in, Speaker mode will be switched to the Headset mode automatically. To Switch between Handset and Speaker/Headset, simply press the Hook Flash in the Handset cradle or the Speaker button.

Make Calls using Numbers

There are FIVE ways to make phone calls: Pick up handset or press SPEAKERPHONE button, and then enter the phone numbers Press the SEND button directly to redial the number last called. Once pressed, the last dialed number will be displayed on the LCD as the corresponding DTMF tones are played out and an outgoing call is sent. Browse the CALLED/CALLER history and press the SEND/REDIAL button. Pick up the handset or press the speakerphone button, then press the CALLED/CALLERS button to browse thru the last 10 numbers dialed out. Once the desired number is identified and displayed on the LCD screen, press the SEND button and a new call to that displayed number will be sent out immediately.
Examples: To dial another extension on the same proxy, such as 1008, simply pick up handset or press speakerphone, dial 1008 and then press the SEND button. 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, followed by pressing the SEND button.
Make Calls using IP Address
Direct IP calling allows two parties, that is, a BT200 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 BT200 phone and other VoIP Device(i.e., another IP Phone or BT200 SIP phone or other VoIP unit) have public IP addresses, or Both BT200 phone and other VoIP Device are on the same LAN using private or public IP addresses, or Both BT200 phone and other VoIP Device can be connected through a router using public or private IP addresses (with necessary port forwarding or DMZ).
This model has the ability to dial an IP address under the same LAN segment by simply pressing the last octet in the IP address. In the Advanced Settings page there is an option "Use Quick IP-call mode", by default it is set to No. When this option is set to YES, and #XXX is dialed, where X is 0-9 and XXX <=255, phone will make direct IP call to aaa.bbb.ccc.XXX where aaa.bbb.ccc comes from the local IP address REGARDLESS of subnet mask.

Grandstream Networks, Inc. BT200 User Manual Firmware 1.1.6.44 Page 14 of 33 Last Updated: 12/2008
#XX or #X are also valid so leading 0 is not required (but OK). eg. 192.168.0.2 calling 192.168.0.3 just dial #3 follow by SEND or # 192.168.0.2 calling 192.168.0.23 just dial #23 follow by SEND or # 192.168.0.2 calling 192.168.0.123 just dial #123 follow by SEND or # 192.168.0.2 dial #3 and #03 and #003 has same effect --> call 192.168.0.3 Note:- If you have a SIP Server configured, Direct IP-IP call will still work. However, if you are using STUN, Direct IP-IP call will also use STUN. OR To make a direct IP to IP call, first off hook, then press MENU key, then enter a 12-digit target IP address to make the call. If port is not default 5060, 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 MENU_key followed by pressing the SEND key or wait for seconds in the No Key Entry Timeout. If the target IP address/port is 192.168.1.20:5062, then the dialing convention would be: MENU_key 192168001020*45062 followed by pressing the SEND key wait for seconds in the No Key Entry Timeout.
Quick IP Call Mode The BT200 also supports Quick IP call mode. This enables the phone to make direct IP-calls, using only the last few digits (last octet) of the target phones IP-number. This is possible only if both phones are in under the same LAN/VPN. This simulates a PBX function using the CMSA/CD without a SIP server. Controlled static IP usage is recommended. Setting up the phone to make Quick IP calls To enable Quick IP calls, the phone has to be setup first. This is done through the web-setup function. In the Advanced Settings page, set the "Use Quick IP-call mode to YES. When #xxx is dialed, where x is 0-9 and xxx <=255, a direct IP call to aaa.bbb.ccc.XXX is completed. aaa.bbb.ccc is from the local IP address regardless of subnet mask. The numbers #xx or #x are also valid. The leading 0 is not required (but OK). For example: 192.168.0.2 calling 192.168.0.3 -- dial #3 follow by SEND or # 192.168.0.2 calling 192.168.0.23 -- dial #23 follow by SEND or # 192.168.0.2 calling 192.168.0.123 -- dial #123 follow by SEND or # 192.168.0.2: dial #3 and #03 and #003 results in the same call -- call 192.168.0.3 NOTE: If you have a SIP Server configured, a Direct IP-IP still works. If you are using STUN, the Direct IP-IP call will also use STUN. Configure the Use Random Port to NO when completing Direct IP calls.

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Answer an Incoming Call

There are two ways to answer an incoming call: 1. 2. Pick up the handset to answer the call normally using handset, or Press the SPEAKERPHONE button to answer in speakerphone or headset mode

Call Hold

While in conversation, pressing the Hold button will put the remote end on hold. Pressing the Hold button again will release the previously Hold state and resume the bi-directional media.
Call Waiting and Call Flashing
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 then can press FLASH button to put the current call party on hold automatically and switch to the other call. Pressing flash button toggles between two active calls.

Call Transfer

Two transfer operations are supported. BLIND TRANSFER User can transfer an active call to a third party without announcement. User presses the TRANSFER button and if the other voice channel is available (i.e., there is no other active conversation besides the current one), user will hear a dial tone. User can then dial the third partys phone number followed by pressing SEND button.
Enable Call Feature has to be configured to Yes in web configuration page in order to make the features to work.
A can hold on to the phone and wait for one of the three following behaviors: A quick confirmation tone (temporarily using the call waiting indication tone) follows by a dial tone. This indicates the transfer has been successful. At this point, the user can either hang up or make another call. A quick busy tone followed by a restored call (On supported platforms only). This means the transfer has failed due to the failed response sent from server and the phone will try to recover the call. The busy tone is just to indicate to the transferor that the transfer has failed. Busy tone keeps playing. This means the phone has failed to receive the final response and decide to time out. Be advised that this does not indicate the transfer has been successful, nor does it indicate the transfer has failed.
ATTENDED TRANSFER User can transfer an active call to a third party with announcement. User presses the FLASH button and hears a dial tone, then dial the third partys phone number followed by pressing SEND button. If the call is answered, press TRANSFER to complete the
Grandstream Networks, Inc. BT200 User Manual Firmware 1.1.6.44 Page 16 of 33 Last Updated: 12/2008

transfer operation and hand up, if the call is not answered, pressing FLASH button to resume the original call.
When Attended Transfer failed, if A hangs up, the BudgeTone phone will ring user A back again to remind A that B is still on the call. A can pick up the phone to restore conversation with B.

Conference Call

BT200 phone supports 3-way conference. Assuming that call party A and B are in conversation. A wants to bring C in a conference: 1. 2. 3. 4.
A presses the CONFERENCE button to get a dial tone and put B on hold A dials Cs number then SEND key to make the call If C answers the call, then A presses CONFERENCE button to bring B, C in the conference. If C does not answer the call, A can press FLASH back to talk to B.
During the conference, if B or C drops the call, the remaining two parties can still talk. However, if A the conference initiator hangs up, all calls will be terminated.
Checking Message and Message Waiting Indication
When BudgeTone-200 is on-hook, pressing the MESSAGE button will trigger the phone to call the VM Server (VMS) configured for the Account. The MWI (Message Waiting Indicator) LED will flash in red color in three quarters of a second when voicemail server sends message waiting information to BudgeTone-200.

Mute and Delete

When in conversation with an ACTIVE LINE, pressing MUTE/DEL will mute the conversation, that is, you can hear the other party but the other party cannot hear you. Pressing the button again will resume the conversation. When dialing a number, press MUTE/DEL will delete the last entered digit. When receiving incoming call, press MUTE/DEL will Reject the call and forward to voice mail.
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CALL FEATURES

BT200 series phone supports a list of call features: Caller ID Block (or Anonymous Call), Disable/Enable Call Waiting, Call Forward on Busy, Delay, or Unconditional, etc. Following table shows the call features of BudgeTone-200 series phone. Table 7: BT200 Call Features Key
*30 *31 *67 *82 *50 *51 *70 *71 *72

Call Features

Block Caller ID (for all subsequent calls) Send Caller ID (for all subsequent calls) Block Caller ID (per call) Send Caller ID (per call) Disable Call Waiting (for all subsequent calls) Enable Call Waiting (for all subsequent calls) Disable Call Waiting (per Call) Enable Call Waiting (per Call) Unconditional Call Forward To use this feature, dial *72 and get the dial tone. Dial the forward number and # for a dial tone, then hang up. Cancel Unconditional Call Forward To cancel Unconditional Call Forward, dial *73 and get the dial tone, then hang up. Busy Call Forward To use this feature, dial *90 and get the dial tone. Dial the forward number and # for a dial tone, then 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 and get the dial tone. Dial the forward number and # for a dial tone, then hang up. Cancel Delayed Call Forward To cancel this Forward, dial *93 and get the dial tone, then hang up.

Others

CONFIGURATION VIA WEB BROWSER
The BT200 embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML pages allow a user to configure the IP phone through a Web browser such as Microsofts IE or Mozilla Firefox.
Access the Web Configuration Menu
To access the phones Web Configuration Menu Connect the computer to the same network as the phone1 Make sure the phone is turned on and shows its IP-address Start a Web-browser on your computer Enter the phones IP-address in the address bar of the browser2 Enter the administrators password to access the Web Configuration Menu3
The Web-enabled computer has to be connected to the same sub-network as the phone. This can easily be done by connecting the computer to the same hub or switch as the phone is connected to. In absence of a hub/switch (or free ports on the hub/switch), please connect the computer directly to the phone using the PC-port on the phone. If the phone is properly connected to a working Internet connection, the phone will display its IP address. This address has the format: xxx.xxx.xxx.xxx, where xxx stands for a number from 0-255. You will need this number to access the Web Configuration Menu. e.g. if the phone shows 192.168.0.60, please use http://192.168.0.60 in the address bar your browser. The default administrator password is admin; the default end-user password is 123.
NOTE: When changing any settings, always SUBMIT them by pressing the button on the bottom of the page. Reboot the phone to have the changes take effect. If, after having submitted some changes, more settings have to be changed, press the menu option needed.

Definitions

Grandstream Networks, Inc. BT200 User Manual Firmware 1.1.6.44 Page 20 of 33 Last Updated: 12/2008
This section will describe the options in the Web configuration user interface. As mentioned, a used can log in as an administrator or end-user. Functions available for the end-user are: Status: Displays the network status, account statuses, software version and MAC-address of the phone Basic: Basic preferences such as date and time settings, multi-purpose keys and LCD settings can be set here. Additional functions available to administrators are: Advanced Settings: To set advanced network settings, codec settings. Account: To configure each of the SIP accounts. Table 9: Device Configuration - Status MAC Address IP Address Product Model Part Number Software Version The device ID, in HEXADECIMAL format. This field shows IP address of BT200 This field contains the product model information. This field contains the product part number. System Up Time System Time Registered Program: This is the main software (firmware) release number, always used to identify the software (firmware) system of the phone. Boot: Booting code version number

Date Display Format

Choose one of the following formats: Year-Month-Day Month-Day-Year Day-Month-Year Choose to display Account Name or date on LCD. Default is No. This field lets user to choose whether to ring the phone Speaker when headset is connected.
Display Account Name instead of Date Mute Speaker Ringer
Advanced User configuration includes not only the end user configuration, but also advanced configuration such as SIP configuration, Codec selection, NAT Traversal Setting and other miscellaneous configuration.
Table 11: Advanced Settings Admin Password Administrator password. Only the administrator can access the Advanced Settings and Account Settings page. Password field is purposely blank for security reasons after clicking update and saved. The maximum password length is 25 characters.
BT200 User Manual Firmware 1.1.6.44 Page 22 of 33 Last Updated: 12/2008

G723 rate

Encoding rate for G723 codec. By default, 6.3kbps rate is set.
iLBC frame size iLBC packet frame size. Default is 20ms. For Asterisk PBX, 30ms might be required. iLBC payload type Silence Suppression Payload type for iLBC. Default value is 97. The valid range is between 96 and 127. This controls the silence suppression/VAD feature of the audio codec G.723 and G.729. If 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. This field contains the number of voice frames to be transmitted in a single Ethernet packet (be advised the IS limit is based on the maximum size of Ethernet packet is 1500 byte (or 120kbps)). When setting this value, be aware of the requested packet time (ptime, used in SDP message) is a result of configuring this parameter. This parameter is associated with the first codec in the above codec Preference List or the actual used payload type negotiated between the 2 conversation parties at run time. e.g., if the first codec is configured as G.723 and the Voice Frames per TX is set to 2, then the ptime value in the SDP message of an INVITE request will be 60ms because each G.723 voice frame contains 30ms of audio. Similarly, if this field is set to 2 and the first codec is G.729 or G.711 or G.726, then the ptime value in the SDP message of an INVITE request will be 20ms. If the configured voice frames per TX exceeds the maximum allowed value, the IP phone will use and save the maximum allowed value for the corresponding first codec choice. The maximum value for PCM is 10 (x10ms) frames; for G.726, it is 20 (x10ms) frames; for G.723, it is 32 (x30ms) frames; for G.729/G.728, 64 (x10ms) and 64 (x2.5ms) frames respectively. Please be careful when editing these parameters. Adjusting these parameters will also change the dynamic jitter buffer. The BT200 has a patent dynamic jitter buffer handling algorithm. The jitter buffer range is 20 ~ 200 ms. Grandstream recommends using the default settings provided. Grandstream does not recommend adjusting these parameters if you are an average user. Incorrect settings will affect the voice quality. Please refer to the Codec FAQ at http://www.grandstream.com/faqscodec.html for more technical detail. Layer 3 QoS Layer 2 QoS No Key Entry Timeout Use # as Dial Key This field defines the layer 3 QoS parameter. It is the value used for IP Precedence or Diff-Serv or MPLS. Default value is 48. This contains the value used for layer 2 VLAN tag. Default setting is blank. Default is 4 seconds. This parameter allows users to configure the # key as the Send (or Dial) key. If set to Yes, the # key will immediately send the call. In this case, this key is essentially equivalent to the Dial key. If set to No, the # key is included as part of the dial string.

Voice Frames per TX

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Local RTP port

This parameter defines the local RTP-RTCP port pair used to 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 is 5004. This parameter, when set to Yes, will force random generation of both the local SIP and RTP ports. This is usually necessary when multiple BT200s are behind the same NAT. Default is No. This parameter specifies how often the BT200 sends a blank UDP packet to the SIP server in order to keep the hole on the NAT open. Default is 20 seconds. NAT IP address used in SIP/SDP message. Default is blank. IP address or Domain name of the STUN server. STUN resolution result will display in the STATUS page of the Web UI. Default method is HTTP. Firmware upgrade may take up to 10 minutes depending on network environment. Do not interrupt the firmware upgrading process.
Use Random Port Keep-alive interval Use NAT IP STUN Server
Firmware Upgrade and Provisioning
Via TFTP Server This is the IP address of the configured TFTP server. If selected and it is non-zero or not blank, the BT200 will attempt to retrieve a new configuration file or new code image from the specified TFTP server at boot time. It will make up to 3 attempts before timeout and then it 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 will be verified and then saved into the Flash memory. Note: Grandstream strongly recommends that the user upgrade firmware locally in a LAN environment if using TFTP to upgrade. Please do NOT interrupt the TFTP upgrade process (especially the power supply) as this will damage the device. Via HTTP Server The HTTP server URL used for firmware upgrade and configuration via HTTP. For example: http://provisioning.mycompany.com:6688/Grandstream/1.1.6.44. Here :6688 is the specific TCP port that the HTTP server is using; omit if using default port 80. Note: If Auto Upgrade is set to No, BT200 will only perform HTTP download once at boot up.

Config Server Path Firmware File Prefix/Postfix
IP address or domain name of firmware server. Default is blank. If configured, BT200 will request the firmware file with the prefix/postfix. This setting is useful for ITSPs. End user should keep it blank. Default is blank. End user should keep it blank. Default is Yes. This allows device gets provisioned automatically.
Config File Prefix/Postfix Allow DHCP Option 66 to override server

Authenticate Conf File

Default is No. If set to Yes, configuration file would be authenticated before acceptance. End user should use default setting.
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Automatic Upgrade

This function is used by ITSP. End user should NOT touch these parameters. Default is No. Choose Yes to enable automatic HTTP upgrade and provisioning. In Check for upgrade every field, enter the number of minutes to check the HTTP server for firmware upgrade or configuration changes. When set to No, the phone will only perform HTTP upgrade and configuration check once at boot up.
Offhook Auto Dial DTMF Payload Type Onhook Threshold Syslog Server Syslog Level
To configure a User ID/extension to dial automatically when the phone is taken offhook. This parameter sets the payload type for DTMF using RFC2833. Default is 101. Time period when the cradle is pressed (Hook Flash) to simulate FLASH. To prevent unwanted activation of the Flash/Hold and automatic phone ring-back, adjust this time value. The IP address or URL of System log server. This feature is especially useful for ITSPs. Select the ATA to report the log level. Default is NONE. The level is one of DEBUG, INFO, WARNING or ERROR. Syslog messages are sent based on the following events: product model/version on boot up (INFO level) NAT related info (INFO level) sent or received SIP message (DEBUG level) SIP message summary (INFO level) 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 For example: May 19 02:40:38 192.168.1.14 GS_LOG: [00:0b:82:00:a1:be][000]. Ethernet link is up.

NTP server

This parameter defines the URI or IP address of the NTP (Network Time Protocol) serve. It is used to display the current date/time.
Distinctive Ring Caller ID must be configured. Select a Distinctive Ring Tone 1 through 3 for a particular Caller ID. The BT200 will ONLY use selected ring tones for particular Caller IDs. For all Tone other calls, the BT200 will use System Ring Tone. When selected and no Caller ID is configured, the selected ring tone will be used for all incoming calls. System Ring Tone System ring tone. Default is North American standard. Adjust system ring tone frequencies and cadences based on local telecom standard.
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Call Progress Tones

Using these settings, users can configure ring or tone frequencies based on parameters from local telecom. By default, they are set to North American standard. Frequencies should be configured with known values to avoid uncomfortable high pitch sounds. Syntax: f1=val,f2=val[,c=on1/off1[-on2/off2[-on3/off3]]]; (Frequencies are in Hz and cadence on and off are in 10ms) ON is the period of ringing (On time in ms) while OFF is the period of silence. In order to set a continuous ring, OFF should be zero. Otherwise it will ring ON ms and a pause of OFF ms and then repeat the pattern. Up to three cadences are supported.
Disable Call Waiting Disable Direct IP calls Use Quick IP Call Mode
Default is No. If set to Yes, the call waiting will be disabled. Default is No. If set to Yes, direct IP calls will be disabled. Dial an IP address under the same LAN/VPN segment by entering the last octet in the IP address. In the Advanced Settings page there is an option Use Quick IP-call mode. Default setting is No. When set to YES, and #XXX is dialed, where X is 0-9 and XXX <=255, phone will make direct IP call to aaa.bbb.ccc.XXX where aaa.bbb.ccc comes from the local IP address REGARDLESS of subnet mask. #XX or #X are also valid so leading 0 is not required (but OK). See Quick IP Call Mode for details.
Disable Conference Lock keypad update Disable DND Send Flash Event Headset TX gain(dB) Headset RX gain (dB)
Default is No. If set to Yes, conference calls will be disabled If set to Yes, the configuration changes via keypad are disabled. Default is No. If set to Yes, DND key on keypad will be disabled Default is No. If set to yes, flash will be sent as DTMF event. Transmission gain. Its a headset setting to control the voice intensity Receive gain. Its a headset setting to control the voice intensity.

SRTP Mode Special Feature
Enable SRTP mode based on selection. Default is No. Default is Standard. Choose the selection to meet special requirements from Soft Switch vendors.
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SAVING THE CONFIGURATION CHANGES
After the 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. Grandstream recommends reboot or power cycle the IP phone after saving changes.
REBOOTING THE PHONE REMOTELY
Press the Reboot button at the bottom of the configuration menu to reboot the phone remotely. The web browser will then display a message window to confirm that reboot is underway. Wait 30 seconds to log in again.
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Software Upgrade & Customization
Software (or firmware) upgrades are completed via either TFTP or HTTP. configuration settings are in the ADVANCED SETTINGS configuration page. The corresponding
FIRMWARE UPGRADE THROUGH TFTP/HTTP
To upgrade via TFTP or HTTP, select TFTP or HTTP upgrade method. Upgrade Server needs to be set to a valid URL of a HTTP server. Server name can be in either FQDN or IP address format. Here are examples of some valid URLs. firmware.mycompany.com:6688/Grandstream/1.1.6.44 168.75.215.189
There are two ways to set up the Upgrade Server to upgrade firmware: via Key Pad Menu and Web Configuration Interface.

Key Pad Menu

Only TFTP would be used when configure the phone using key pad. To configure the Upgrade Server via Key Pad Menu options, select [6]tFtP from the Main Menu, then press MENU button to enter this option. Enter the 12-digit IP address. Then, press MENU button to save. Must reboot the phone to take effect.
Web Configuration Interface
To configure the Upgrade Server via the Web configuration interface, open the web browser. Enter the BT200 IP address. Enter the admin password to access the web configuration interface. In the ADVANCED SETTINGS page, enter the Upgrade Servers IP address or FQDN in the Firmware Server Path field. Select TFTP or HTTP upgrade method. Update the change by clicking the Update button. Reboot or power cycle the phone to update the new firmware. During this stage, the LCD will display the firmware file downloading process. If a firmware upgrade fails for any reason (e.g., TFTP/HTTP server is not responding, there are no code image files available for upgrade, or checksum test fails, etc), the phone will stop the upgrading process and re-boot using the existing firmware/software. Firmware upgrades take around 60 seconds in a controlled LAN or 5-10 minutes over the Internet. Grandstream recommends completing firmware upgrades in a controlled LAN environment whenever possible.

Managing Firmware and Configuration File Download
When Automatic Upgrade is set to Yes, a Service Provider can use P193 (Auto Check Interval, in minutes, default and minimum is 60 minutes) to have the devices periodically check for upgrades at prescheduled time intervals. By defining different intervals in P193 for different devices, a Server Provider can manage and reduce the Firmware or Provisioning Server load at any given time.
Page 32 of 33 Last Updated: 12/2008
Restore Factory Default Setting
WARNING: Restoring the Factory Default Setting will delete all configuration information of the phone. Please backup or print all the settings before you restoring factory default settings. Grandstream is not responsible for restoring lost parameters and cannot connect your device to your VoIP service provider. INSTRUCTIONS FOR RESTORATION: Step 1: Press the MENU button for Key Pad Menu options and press the Up button to see reset. Step 2: Enter the MAC address printed on the bottom of the sticker. Please 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 Example: if the MAC address is 000b8200e395, it should be key in as 0002228200333395. NOTE: If there are digits like 22 in the MAC, you need to type 2 then press -> right arrow key to move the cursor or wait for 4 seconds to continue to key in another 2. Step 3: Press the MENU button again. If the MAC address is correct, the phone will reboot. Wait for phone reboot and the LCD backlight finish flashing.
Page 33 of 33 Last Updated: 12/2008

doc1

Avaya Solution & Interoperability Test Lab
Application Notes for the Grandstream Telephones with Avaya Communication Manager 3.1.2 and Avaya SIP Enablement Services 3.1.1 Issue 0.1

Abstract

These Application Notes describe a solution comprised of Avaya Communication Manager 3.1.2, Avaya SIP Enablement Services (SES) 3.1.1, and Grandstream Networks SIP telephones. Grandstream GXP2000 and BT200 are SIP-based VoIP telephones. Grandstream GXP2000 is typically used in an enterprise or small business environment and BT200 is used by residential or SoHo users. During compliance testing, the Grandstream Telephones successfully registered with Avaya SES, placed and received calls to and from SIP and nonSIP telephones, and established conference calls. Information in these Application Notes has been obtained through compliance testing and additional technical discussions. Testing was conducted via the DeveloperConnection Program at the Avaya Solution and Interoperability Test Lab.

AT; Reviewed: RRR m/d/y

Solution & Interoperability Test Lab Application Notes 2006 Avaya Inc. All Rights Reserved.
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1. Introduction

These Application Notes describe a solution comprised of Avaya Communication Manager 3.1.2, Avaya SIP Enablement Services (SES) 3.1.1, and Grandstream Networks SIP telephones. Grandstream GXP2000 and BT200 are SIP-based VoIP telephones. Grandstream GXP2000 is typically used in an enterprise or small business environment and BT200 is used by residential or SoHo users. Grandstream GXP2000 telephone supports up to four lines, and on each line can bridge calls to establish a 3-party conference. Grandstream BT200 supports one line. Figure 1 illustrates a sample configuration consisting of an Avaya S8710 Media Servers, an Avaya G650 Media Gateway, an Avaya SIP Enablement Services (SES) server, and the Grandstream endpoints. Avaya Communication Manager is installed on the S8710 Media Servers. The solution described herein is also extensible to other Avaya Media Servers and Media Gateways. For completeness, Avaya 4600 Series SIP IP Telephones, Avaya 4600 Series H.323 IP Telephones, and Avaya 6400 and 8400 Series Digital Telephones, are included in Figure 1 to demonstrate calls between the SIP-based Grandstream Telephone and Avaya SIP, H.323, and digital telephones. The analog PSTN telephone is also included to demonstrate calls routed by Avaya Communication Manager between the Grandstream IP telephones and the PSTN. The Grandstream endpoint originates a call by sending a call request (SIP INVITE message) to the Avaya SES server. The Avaya SES server routes the call over a SIP trunk to Avaya Communication Manager for origination services. If the call is destined for another local SIP endpoint, such as another Grandstream telephone or an Avaya SIP telephone, then Avaya Communication Manager routes the call back over the SIP trunk to the Avaya SES server, which in turn delivers the call to the destination SIP telephone. Otherwise, Avaya Communication Manager routes the call to the PSTN, a local Avaya H.323, digital, or analog telephone, an adjunct, a vector, a hunt group, etc., depending on the destination number. For a call arriving to Avaya Communication Manager that is destined for the Grandstream SIP telephone, Avaya Communication Manager routes the call over the SIP trunk to the Avaya SES server, which in turn delivers the call to the Grandstream telephone.
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Figure 1: Sample configuration.
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2. Equipment and Software Validated
The following equipment and software/firmware were used for the sample configuration provided: Equipment Avaya S8710 Media Server Software/Firmware Avaya Communication Manager 3.1.2 (R013x.01.2.632.1) Avaya G650 Media Gateway TN2312BP IP Server Interface HW12 FW 31 TN799DP C-LAN Interface HW01 FW 17 TN2302AP IP Media Processor HW20 FW 112 Avaya SIP Enablement Services Server SES 3.1.1(R03.1.1-03.1.114.0) Avaya 4600 Series IP Telephones 2.3 (4602SW H.323) 2.5 (4625SW H.323) 2.2.3 (4610SW SIP) Avaya 6400 and 8400 Series Digital Telephones Grandstream Networks GXP2000 Telephone 1.1.2.26 Grandstream Networks BT200 Telephone 1.1.2.26 Analog Telephone -
3. Configure Avaya Communication Manager
This section describes the steps for configuring IP codec sets and associating SIP telephone numbers with off-PBX telephone stations in Avaya Communication Manager. The steps are performed from the Avaya Communication Manager System Access Terminal (SAT) interface. IP codec sets identify the codecs that may be used in calls involving VoIP telephones. An offPBX telephone is a phone that Avaya Communication Manager does not control, such as a cellular phone, a home telephone, or a SIP telephone. Avaya Communication Manager features and calling privileges, however, can be applied to an off-PBX telephone by associating a local, on-PBX, extension with the off-PBX telephone. This approach is taken for SIP Telephones that register with the Avaya SES server and intend to use Avaya Communication Manager for call origination and termination services. Specifically, an Administration WithOut Hardware (AWOH) on-PBX station is administered in Avaya Communication Manager and then associated with the telephone number of the SIP telephone. Similarly, on the Avaya SES server, the number of the SIP telephone is administratively associated with the extension of the on-PBX station. Throughout the rest of this document, on-PBX stations associated with SIP Telephones in such a manner will be referred to as Outboard Proxy SIP (OPS) stations.
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3.1. Capacity Verification
Step Description 1. Issue the command display system-parameters customer-options, and proceed to Page 2. Verify that the number of SIP trunks supported by the system is sufficient for the number of SIP trunks needed. If not, contact an authorized Avaya account representative to obtain additional licenses. Note: Each SIP call between two SIP endpoints (whether internal or external) requires two SIP trunks for the duration of the call. The license file installed on the system controls the maximum permitted.
display system-parameters customer-options OPTIONAL FEATURES IP PORT CAPACITIES Maximum Administered H.323 Trunks: Maximum Concurrently Registered IP Stations: Maximum Administered Remote Office Trunks: Maximum Concurrently Registered Remote Office Stations: Maximum Concurrently Registered IP eCons: Max Concur Registered Unauthenticated H.323 Stations: Maximum Video Capable H.323 Stations: Maximum Video Capable IP Softphones: Maximum Administered SIP Trunks: 200 Page 2 of 10

USED 153

Maximum Number of DS1 Boards with Echo Cancellation: Maximum TN2501 VAL Boards: Maximum G250/G350/G700 VAL Sources: Maximum TN2602 Boards with 80 VoIP Channels: Maximum TN2602 Boards with 320 VoIP Channels: Maximum Number of Expanded Meet-me Conference Ports: (NOTE: You must logoff & login to effect the permission changes.)
Enter the display system-parameters customer-options command. Verify that there are sufficient Maximum Off-PBX Telephones OPS licenses. If not, contact an authorized Avaya account representative to obtain additional licenses.
display system-parameters customer-options OPTIONAL FEATURES G3 Version: V13 Location: 1 Platform: 8 Page 1 of 10
RFA System ID (SID): 1 RFA Module ID (MID): 1 USED 50 0
Platform Maximum Ports: Maximum Stations: Maximum XMOBILE Stations: Maximum Off-PBX Telephones - EC500: Maximum Off-PBX Telephones OPS: Maximum Off-PBX Telephones - SCCAN:
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3.2. IP Codec Set

This section describes the steps for administering codec set in Avaya Communication Manager. This codec set is used in the IP Network Region for communications between Avaya Communication Manager and Avaya SES. Step Description Enter the change ip-codec-set <c> command, where c is a number between 1 and 7, 1. inclusive. IP codec sets are specified in the IP Network Region forms to define which codecs may be used within and between network regions. For the compliance testing G.711MU and G.729AB were used. Note: Media encryption for SIP calls is currently not supported in Avaya Communication Manger, Avaya SIP telephones, and Grandstream SIP telephones.
change ip-codec-set 2 IP Codec Set Codec Set: 2 Audio Codec 1: G.711MU 2: G.729AB 3: 4: 5: 6: 7: Silence Suppression n n Frames Per Pkt Packet Size(ms) Page 1 of 2
Media Encryption 1: none 2: 3:
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3.3. IP Network Region

This section describes the steps for administering an IP Network Region in Avaya Communication Manager for communication between Avaya Communication Manager and Avaya SIP Enablement Services. Step Description 1. Enter the change ip-network-region <n> command, where n is a number between 1 and 250, inclusive and administer settings as per below. Codec Set Set to Codec Set as provisioned in Section 3.1. Authoritative Domain Set to the same value as SIP Domain on Avaya SIP Enablement Services Section 4, step 2. Inter-region IP-IP Direct Audio Set to yes to allow direct IP-to-IP audio connectivity between endpoints registered to Avaya Communication Manager or Avaya SES.

change ip-network-region 2 IP NETWORK REGION Region: 2 Location: Authoritative Domain: devconnect.com Name: MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes Codec Set: 2 Inter-region IP-IP Direct Audio: yes UDP Port Min: 2048 IP Audio Hairpinning? y UDP Port Max: 65535 DIFFSERV/TOS PARAMETERS RTCP Reporting Enabled? y Call Control PHB Value: 46 RTCP MONITOR SERVER PARAMETERS Audio PHB Value: 46 Use Default Server Parameters? y Video PHB Value: 26 802.1P/Q PARAMETERS Call Control 802.1p Priority: 6 Audio 802.1p Priority: 6 Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS H.323 IP ENDPOINTS RSVP Enabled? n H.323 Link Bounce Recovery? y Idle Traffic Interval (sec): 20 Keep-Alive Interval (sec): 5 Keep-Alive Count: 5 Page 1 of 19
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Step Description 2. Proceed to Page 3 of the IP NETWORK REGION form and enable inter-region connectivity between regions as per below. For purpose of these application notes, src rgn 2 and dst rgn 2 use codec set 2 as configured in Section 3.1.
Page 3 of 19 Inter Network Region Connection Management src rgn 2 dst rgn 15 codec set direct Total WAN WAN-BW-limits y :NoLimit Video Dyn WAN-BW-limits Intervening-regions CAC IGAR n

3.4. IP Node Names

This section describes the steps for setting IP node name for Avaya SES in Avaya Communication Manager. Step Description 1. Issue the command change node-names ip; and administer settings as per below. Add a node name for Avaya SES along with the IP address. Note: Verify that node-names are configured for the C-LAN and MEDPRO boards. change node-names ip Page 1
IP NODE NAMES Name CLAN-1A06 MEDPRO-1A13 SES IP Address 192.45.100.147 192.45.103.148 192.45.52.160
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3.5. SIP Signaling

This section describes the steps for administering a signaling group in Avaya Communication Manager for communication between Avaya Communication Manager and Avaya SIP Enablement Services. Step Description 1. Issue the command add signaling-group <s>, where s is an unallocated Signaling Group; and administer settings as per below. Group Type Set to sip. Transport Method Set to tls. Far-end Listen Port Set to 5061(default) Near-end Node Name - Set to CLAN IP Address as displayed in Section 3.3. Far-end Node Name - Set to IP Address of SES configured in Section 3.3. Far-end Network Region - Set to the region configured in Section 3.2. Far-end Domain - Set to the Authoritative Domain configured in Section 3.2, Step 1.

add signaling-group 10 SIGNALING GROUP Group Number: 10 Group Type: sip Transport Method: tls Page 1 of 5
Near-end Node Name: CLAN-1A06 Near-end Listen Port: 5061 Far-end Domain:devconnect.com
Far-end Node Name: SES Far-end Listen Port: 5061 Far-end Network Region: 2
Bypass If IP Threshold Exceeded? n DTMF over IP: rtp-payload Session Establishment Timer(min): 120 Direct IP-IP Audio Connections? y IP Audio Hairpinning? n
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3.6. SIP Trunking

This section describes the steps for administering a trunk group in Avaya Communication Manager for communication between Avaya Communication Manager and Avaya SES. Step Description 1. Issue the command add trunk-group <t>, where t is an unallocated Trunk Group; and administer settings as per below. Group Type Set to same value as Group Type configured in Section 3.4. TAC (Trunk Access Code) Set to any number with 1-4 digits;* and # may be used as first digit only. Signaling Group Set to same value as Group Number configured in Section 3.4. Number of Members Set to a value between 0 and 255. Group Name - Set a trunk group name. Note: Each SIP call between two SIP endpoints (whether internal or external) requires two SIP trunks for the duration of the call. The license file installed on the system controls the maximum permitted.
add trunk-group 10 TRUNK GROUP Group Number: Group Name: Direction: Dial Access? Queue Length: Service Type: 10 Group Type: SIP-SES-DevCon1 COR: two-way Outgoing Display? n 0 tie Auth Code? sip 1 n CDR Reports: y TAC: 110 Page 1 of 21
Night Service: n Signaling Group: 10 Number of Members: 150
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3.7. SIP Stations

This section describes the steps for administering OPS stations in Avaya Communication Manager and associating the OPS station extensions with the telephone numbers of the Grandstream SIP telephones. Step Description 1. Enter the add station <s> command, where <s> is an available extension in the dial plan, to administer an OPS station. On Page 1 of the station form configure the following: Type Set to 6408D+. Port Set to X. Name Set station name.
add station 54007 STATION Extension: Type: Port: Name: 54007 6408D+ X GXP2000 Lock Messages? n Security Code: Coverage Path 1: Coverage Path 2: Hunt-to Station: BCC: TN: COR: COS: Page 1 of 4
STATION OPTIONS Loss Group: Data Module? Speakerphone: Display Language:

2 n 2-way english

Personalized Ringing Pattern: 1 Message Lamp Ext: 54007 Mute Button Enabled? y

Media Complex Ext: IP SoftPhone? n
Enter the change off-pbx-telephone station-mapping <s> command, where <s> is the extension of the OPS station configured in Step 3. On Page 1 of the off-pbx-telephone station-mapping form, configure the following: Station Extension Set the extension of the OPS station. Application Set to OPS. Phone Number Enter the number that the Grandstream IP telephone will use for registration and call termination. In the example below, the Phone Number is the same as the OPS Station Extension, but is not required to be the same. Trunk Selection Set to the trunk configured in Section 3.5. Configuration Set Set to 1, which during compliance testing used the default values of the off-pbx-telephone configuration-set form.
change off-pbx-telephone station-mapping 54007 Page STATIONS WITH OFF-PBX TELEPHONE INTEGRATION Station Extension 54007 Application OPS Dial Phone Number Prefix - 54007 Trunk Selection of 2

Configuration Set 1

Repeat Steps 1 and 2 as necessary to administer additional OPS stations and associations for Grandstream SIP Telephones.
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4. Configure Avaya SIP Enablement Services
This section describes the steps for creating SIP user accounts in Avaya SIP Enablement Services (SES) and associating the SIP users with an Avaya Communication Manager OPS station extension. The Grandstream Telephone will register with Avaya SES using the SIP user accounts. This section assumes that the necessary Avaya SES configuration steps for establishing a SIP trunk with Avaya Communication Manager have been completed. Step Description 1. Open a web browser, enter http://<IP address of Avaya SES server>/admin for the URL, and log in with the appropriate credentials. Click on the Launch Administration Web Interface link upon successful login. From the Administration Web Interface: 2. Click the + sign to expand the options under Server Configuration. Click System Properties. Verify the SIP Domain matches the Authoritative Domain configured for the IP NETWORK REGION on Avaya Communication Manager in Section 3.2.
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Step Description 3. To enable secure SIP trunking between Avaya SES and Avaya Communication Manager, add a Media Server corresponding to Avaya Communication Manager from the Administration Web Interface: Click the + sign to expand the options under Media Servers. Click Add.

At the Add Media Server Interface page, provision SIP Trunk parameters as follows for connectivity to Avaya Communications Manager: SIP Trunk Link Type - Set to same value as Transport Method in Section 3.4. SIP Trunk IP Address - Set to same value as CLAN address in Section 3.3. Click the Add button when finished and hit the Continue button on the confirmation page [not shown].
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Step Description 5. In the left pane of the SES Administration Web Interface, expand Users and click on Add.
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Step Description 6. At the Add User page, configure the following: Primary Handle Enter the phone number of the Grandstream telephone. The number must match the phone number entered in Section 3.6 Step 3. Password and Confirm Password Specify a password that the Grandstream IP telephone must use to successfully register with Avaya SES. Host Select the IP address or FQDN of the Avaya SES server. First Name and Last Name Enter descriptive names. Check the Add Media Server Extension checkbox. Click on Add. Click Continue on the next page [not shown].
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Step Description 7. At the Add Media Server Extension screen, configure the following: Extension Set it to the corresponding Avaya Communication Manager OPS station configured in Section 3.6 Step 3. Media Server Set to the Media Server where this OPS station is configured. Click on Add. Click Continue on the next page [not shown]. Note: Media Server was previously configured on SES
Repeat Steps as necessary to configure SIP users for additional Grandstream SIP Telephones.
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Step Description 9. Click on Update at the bottom of the left pane.
10. Click on Continue at the bottom of the right pane.
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5. Configure the Grandstream Telephone
This section describes the steps for configuring the Grandstream Telephone. This section assumes that the Grandstream Telephones IP address is already configured. Step Description 1. Open a web browser, enter http://a.b.c.d for the URL, where a.b.c.d is the IP address of the Grandstream Telephone. Enter the password and click on Login button to proceed to the next screen. Note: Following steps are performed to configure GXP2000 but the configuration for BT200 endpoint is exactly same except BT200 has only one ACCOUNT to be configured in its pull-down menu.

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Step Description 2. At BASIC SETTINGS screen, configure the following: IP Address Set the IP address if required. Subnet Mask Set the subnet mask. Default Router Set the default router. Click Update to modify the values.
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Step Description 3. At ADVANCED SETTINGS screen, configure the following: Layer 3 QoS Set to the desired value. For compliance testing, we set it to 34 and 48. 802.1p priority value Set to the desired value between 0 and 7. For compliance testing, we set it 0 and 6. Click Update to modify the values.
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Step Description 4. At ACCOUNT1 screen, configure the following: Account Name Set to the Primary Handle configured in Section 4, Step 6. SIP Server Set to the SIP Domain configured in Section 4, Step 2. Outbound Proxy Set to the Avaya SES server IP address. SIP User ID Set to the Primary Handle configured in Section 4, Step 6. Authenticate ID Set to the User Id configured in Section 4, Step 6. Authenticate Password Set to the Password configured in Section 4, Step 6. Name Any String for identification purposes. Turn off speaker on remote disconnect Set the value to Yes. Click Update to modify the values.
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6. Interoperability Compliance Testing
The focus of the interoperability compliance testing was primarily on verifying call establishment on the Grandstream Telephones. Grandstream SIP telephone operations such as dialing methods (manual, re-dial, and phone book), hold, mute, and conference, and Grandstream SIP telephone interactions with Avaya SIP Enablement Services (SES), Avaya Communication Manager, and Avaya SIP, H.323, and digital telephones.
6.1. General Test Approach
The general test approach was to place calls to and from the Grandstream SIP telephone and exercise basic telephone operations on the Grandstream Telephone. The main objectives were to verify that: The Grandstream Telephone successfully registers with Avaya SES. The Grandstream Telephone successfully establishes calls with Avaya SIP, H.323, and digital telephones attached to Avaya SES or Avaya Communication Manager. The Grandstream Telephone successfully establishes calls with PSTN telephones through Avaya Communication Manager. The Grandstream Telephone successfully handles concurrent calls on its two lines. The Grandstream Telephone successfully negotiates the right codec. The Grandstream Telephone successfully shuffles for VOIP calls. The Grandstream Telephone successfully transmits DTMF during a call. The Grandstream Telephone successfully handles layer-3 (DiffServ) QoS for Audio. The Grandstream Telephone successfully handles layer-2 (802.1p) QoS for Audio.

For serviceability testing, failures such as cable pulls and hardware resets were applied. For performance testing, a conference call involving two Grandstream Telephones and two Avaya telephones was formed as follows. A call was established between an Avaya telephone and a Grandstream Telephone. The Grandstream Telephone then used its second line to establish a call with another Grandstream Telephone, and bridged the two lines together, forming a 3-party conference. The second Grandstream Telephone then used its second line to establish a call with another Avaya telephone, and bridged its two lines together, effectively forming a 4-party conference.
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6.2. Test Results

The test objectives of Section 6.1 were verified. For serviceability testing, the Grandstream Telephones operated properly after recovering from failures such as cable disconnects, and resets of the Grandstream Telephones, the Avaya SES server, and Avaya Communication Manager. For performance testing, the conference call was successfully maintained for approximately two hours. Grandstream Telephones successfully shuffles to communicate directly with the other endpoint. Grandstream Telephones successfully negotiated the codec to be used. The following observations were made during testing: Grandstream Telephone does not support de-registration but when the telephone is rebooted, it automatically re-registers with Avaya SES. Grandstream Telephone does not support VLAN tagging. Grandstream Telephone cannot mute all parties if it initiates the conference. Only the first called party is muted. Grandstream Telephone fail to shuffle if both the endpoints are Grandstream telephones. A workaround is to configure both telephones to support the same set of codecs and these codecs should be unique. Grandstream Telephone terminates the call after a certain time when the call is muted or put on hold. Grandstream supports a configurable session timer which is incompatible with Avaya SIP implementation. A workaround is to make the session timer large enough for the SIP trunk configured in Avaya Communication Manager and Grandstream Telephone. Grandstream Telephone has a delay of about 5 seconds when the audio is muted/unmuted. Grandstream Telephone BT200 re-registers with Avaya SES under some extreme circumstances. Grandstream Telephone is not compatible with Avaya SES for Presence and IM implementation. Grandstream Networks expects to resolve the above observations in future releases. Contact Grandstream Networks (www.grandstream.com) for further updates.

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7. Verification Steps

The following steps may be used to verify the configuration: Verify that the Grandstream Telephones successfully register with the Avaya SES server by following the Users -> Registered Users links on the SES Administration Web Interface. Place calls to and from the Grandstream Telephone and verify that the calls are successfully established with two-way talk path. From the Avaya Communication Manager System Access Terminal (SAT) interface, use following commands to verify that the calls successfully shuffled between two SIP telephones:
Step Description 1. Check the ports which are active for the SIP trunk being used by using the following command: status trunk 10 Note down the members in active state. In our example, 10/2 and 10/6 are active.
Status trunk 10 TRUNK GROUP STATUS Member Port Service State Mtce Connected Ports Busy no no no no no no no no no no
0010/001 0010/002 0010/003 0010/004 0010/005 0010/006 0010/007 0010/008 0010/009 0010/010
T00046 T00047 T00048 T00049 T00050 T00051 T00052 T00053 T00054 T00055
in-service/idle in-service/active in-service/idle in-service/idle in-service/idle in-service/idle in-service/idle in-service/idle in-service/idle in-service/idle
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Step Description 2. Issue the following command for the ports in active state: status trunk 10/2 Note that the Near-end IP Addr and Far-end IP Addr for Audio are talking on the same port and Audio Connection Type is ip-direct. This signifies that the endpoints have shuffled and talking to each other directly.
status trunk 10/2 TRUNK STATUS Trunk Group/Member: 0010/002 Port: T00047 Signalling Group ID: Service State: in-service/active Maintenance Busy? No Page 1 of 2
Connected Ports: T0051 Port Signaling: 01A0617 G.711MU Audio: Video: Video Codec: Near-end IP Addr 192. 45.100.147 192. 45. 53.101 : Port : 5061 : 34008 Far-end IP Addr 192. 45. 52.160 192. 45. 53.102 : Port : 5061 : 34008
Authentication Type: None Audio Connection Type: ip-direct
Note on the second page of the status screen, it verifies that both endpoints are using the same codec g711u.
status trunk 10/2 SRC PORT TO DEST PORT TALKPATH src port: T00047 T00047:TX:192.45.53.101:34008/g711u/20ms T00051:TX:192.45.53.102:34008/g711u/20ms Page 2 of 2

Dst port: T00051

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8. Support

For technical support on Grandstream Networks telephones, consult the support pages at http://www.grandstream.com/y-services.htm or contact Grandstream Networks technical support at: Telephone: 1- (617) E-mail: Provide email address if available

9. Conclusion

These Application Notes describe a solution comprised of Avaya Communication Manager 3.1.2, Avaya SIP Enablement Services (SES) 3.1.1, and Grandstream Networks SIP telephones. Grandstream GXP2000 and BT200 are SIP-based VoIP telephones. Grandstream GXP2000 is typically used in an enterprise or small business environment and BT200 is used by residential or SoHo users. During compliance testing, the Grandstream Telephones successfully registered with Avaya SES, placed and received calls to and from SIP and non-SIP telephones, and established conference calls.
10. Additional References
Product documentation for Avaya products may be found at http://support.avaya.com/. [1] Administrator Guide for Avaya Communication Manager, Issue 2.1, May 2006, Document Number 03-300509 [2] Administration for Network Connectivity for Avaya Communication Manager, Issue 11, February 2006, Document Number 555-233-504 [3] SIP Support in Release 3.1 of Avaya Communication Manager, Issue 6, February 2006, Document Number 555-245-206 [4] Installing and Administering SIP Enablement Services R3.1.1, Issue 2.0, August 2006, Document Number 03-600768 Product documentation for Grandstream Networks products may be found at http://www.grandstream.com.
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2006 Avaya Inc. All Rights Reserved. Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by and are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks are the property of their respective owners. The information provided in these Application Notes is subject to change without notice. The configurations, technical data, and recommendations provided in these Application Notes are believed to be accurate and dependable, but are presented without express or implied warranty. Users are responsible for their application of any products specified in these Application Notes.
Please e-mail any questions or comments pertaining to these Application Notes along with the full title name and filename, located in the lower right corner, directly to the Avaya DeveloperConnection Program at devconnect@avaya.com.
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Technical specifications

General
Product TypeVoIP phone
Body MaterialABS plastic
Phone Features
Dialer TypeKeypad
Dialer LocationBase
Conference Call Capability3-way
IntercomYes
SpeakerphoneYes ( digital duplex )
Caller IDYes
Call WaitingYes
Call ForwardingYes
Call TransferYes
Call HoldYes
Function ButtonsMute button
Volume ControlYes
Ringer ControlYes
IndicatorsVoice message waiting indicator
IP Telephony
Main FeaturesIntegrated Ethernet switch, Power over Ethernet (PoE) support
VoIP ProtocolsSIP
Voice CodecsG.711, G.722, G.723, G.729
Quality of ServiceIEEE 802.1Q (VLAN), Differentiated Services (DiffServ), IEEE 802.1p, Type of Service (ToS)
IP Address AssignmentDHCP
Security128 bit AES
Network ProtocolsIP, TCP, TFTP, UDP, ARP, HTTP, DNS
Network Ports Qty2 x Ethernet 10Base-T/100Base-TX
Voice FeaturesEcho cancellation (ECN)
Network FeaturesNetwork Address Translation (NAT)
Display
TypeLCD display - monochrome
Display LocationBase
Line Qty8
Character Qty22
Display Resolution131 x 64 pixels
Miscellaneous
Connections1 x headset jack
Compliant StandardsCE, FCC
Power
TypePower adapter - external
Dimensions & Weight (Base)
Width8.3 in
Depth8.7 in
Height2.6 in
Weight2.9 lbs
Universal Product Identifiers
BrandGrandstream Networks
Part NumberBT200

 

Tags

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