Grandstream Networks, Inc. UCM6102/UCM6104/UCM6108/UCM6116 All-in-one Hybrid IPPBX Appliance User Manual Grandstream Networks, Inc. www.grandstream.
UCM6102/UCM6104/UCM6108/UCM6116 User Manual Index CHANGE LOG ........................................................................................... 9 FIRMWARE VERSION 1.0.0.32 ............................................................................................................ 9 WELCOME ............................................................................................... 10 PRODUCT OVERVIEW ............................................................................
LDAP CLIENT CONFIGURATIONS ............................................................................................. 29 HTTP SERVER .................................................................................................................................... 30 EMAIL SETTINGS ............................................................................................................................... 31 TIME SETTINGS ................................................................................
RING GROUP........................................................................................... 62 CONFIGURING RING GROUP ........................................................................................................... 62 RING GROUP PARAMETERS ............................................................................................................ 62 PAGING AND INTERCOM GROUP ......................................................... 64 CALL QUEUE ........................................
STATUS AND REPORTING ..................................................................... 85 PBX STATUS ....................................................................................................................................... 85 TRUNKS ....................................................................................................................................... 85 EXTENSIONS ...............................................................................................................
Table of Tables UCM6102/UCM6104/UCM6108/UCM6116 User Manual Table 1: TECHNICAL SPECIFICATIONS ................................................................................................... 11 Table 2: UCM6102/UCM6104 EQUIPMENT PACKAGING ........................................................................ 14 Table 3: UCM6108/UCM6116 EQUIPMENT PACKAGING ......................................................................... 14 Table 4: LCD MENU OPTIONS ..............................................
Table 39: Cleaner Configuration ............................................................................................................... 103 Firmware Version 1.0.0.
Table of Figures UCM6102/UCM6104/UCM6108/UCM6116 User manual Figure 1: UCM6102 Front View................................................................................................................... 14 Figure 2: UCM6102 Back View ................................................................................................................... 15 Figure 3: UCM6104 Front View...................................................................................................................
Figure 40: System Status->Storage Usage ................................................................................................. 93 Figure 41: System Status->Resource Usage .............................................................................................. 94 Figure 42: CDR Filter .................................................................................................................................. 95 Figure 43: Call Report .....................................................
CHANGE LOG This section documents significant changes from previous versions of the UCM6102/UCM6104/UCM6108/UCM6116 user manuals. Only major new features or major document updates are listed here. Minor updates for corrections or editing are not documented here. FIRMWARE VERSION 1.0.0.32 This is the initial version. Firmware Version 1.0.0.
WELCOME Thank you for purchasing Grandstream UCM6102/UCM6104/UCM6108/UCM6116. UCM6102/UCM6104/UCM6108/UCM6116 is an innovative, all-in-one hybrid IP PBX appliance designed for small to medium business. Powered by an advanced hardware platform with robust system resources, the UCM6102/UCM6104/UCM6108/UCM6116 offers a highly versatile state-of-the-art Unified Communication (UC) solution for converged voice, video, data, fax and video surveillance application needs.
PRODUCT OVERVIEW FEATURE HIGHTLIGHTS 1GHz ARM Cortex A8 application processor, large memory (512MB DDR RAM, 4GB NAND Flash), and dedicated high performance multi-core DSP array for advanced voice processing. Integrated 2/4/8/16 PSTN trunk FXO ports, 2 analog telephone FXS ports, and up to 50 SIP trunk options. Gigabit network port with integrated PoE, USB, SD; integrated NAT router with advanced QoS support (UCM6102 only). Supports a wide range of popular voice codes (including G.
LCD Display 128x32 graphic LCD with DOWN and OK button Reset Switch Yes Voice/Video Capabilities Voice-over-Packet LEC with NLP Packetized Voice Protocol Unit, 128ms-tail-length carrier Capabilities grade Line Echo Cancellation, Dynamic Jitter Buffer, Modem detection & auto-switch to G.711 Voice and Fax Codecs G.711 A-law/U-law, G.722, G.723.1 5.3K/6.3K, G.726, G.729A/B, iLBC, GSM; T.38 Video Codecs H.264, H.263, H.
based on agent skills/availability busy level, in-queue announcement Customizable Auto Up to 5 layers of IVR (Interactive Voice Response) Attendant Concurrent Calls Conference Bridges UCM6102: Up to 30 simultaneous calls UCM6104: Up to 45 simultaneous calls UCM6108/UCM6116: Up to 60 simultaneous calls UCM6102/UCM6104: Up to 3 password-protected conference bridges allowing up to 25 simultaneous PSTN or IP participants UCM6108/UCM6116: Up to 6 password-protected conference bridges allo
INSTALLATION This section describes detailed information on installation, connection and warranty policy of the UCM6102/UCM6104/UCM6108/UCM6116.
Figure 2: UCM6102 Back View To set up the UCM6102, follow the steps below: 1. Connect one end of an RJ-45 Ethernet cable into the WAN port of the UCM6102; 2. Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch/hub; 3. Connect the 12V DC power adapter into the 12V DC power jack on the back of the UCM6102. Insert the main plug of the power adapter into a surge-protected power outlet; 4. Wait for the UCM6102 to boot up.
Figure 4: UCM6104 Back View To set up the UCM6104, follow the steps below: 1. Connect one end of an RJ-45 Ethernet cable into the LAN 1 port of the UCM6104; 2. Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch/hub; 3. Connect the 12V DC power adapter into the 12V DC power jack on the back of the UCM6104. Insert the main plug of the power adapter into a surge-protected power outlet; 4. Wait for the UCM6104 to boot up.
To set up the UCM6108, follow the steps below: 1. Connect one end of an RJ-45 Ethernet cable into the LAN port of the UCM6108; 2. Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch/hub; 3. Connect the 12V DC power adapter into the 12V DC power jack on the back of the UCM6108. Insert the main plug of the power adapter into a surge-protected power outlet; 4. Wait for the UCM6108 to boot up.
5. Once the UCM6116 is successfully connected to network, the LED indicator for NETWORK in the front will be in solid green and the LCD shows up the IP address; 6. (Optional) Connect PSTN lines from the wall jack to the FXO ports; connect analog lines (phone and fax) to the FXS ports. SAFETY COMPLIANCES The UCM6102/UCM6104/UCM6108/UCM6116 complies with FCC/CE and various safety standards. The UCM6102/UCM6104/UCM6108/UCM6116 power adapter is compliant with the UL standard.
GETTING STARTED This section provides information about using the LCD menu, LED indicators and Web GUI of the UCM6102/UCM6104/UCM6108/UCM6116. The last section describes how to make your first call using the UCM6102/UCM6104/UCM6108/UCM6116 with your SIP phone. USING THE LCD MENU Default LCD Display By default, when the device is powered on, the LCD will show device model, hardware version and IP address. Menu Access Press "Down" or "OK" button to start browsing menu options.
Factory Menu LCD Test Patterns: Press "Down" button to test different LCD patterns Fan Mode: Auto or On USING THE LED INDICATORS The UCM6102/UCM6104/UCM6108/UCM6116 has LED indicators in the front and the following table shows the status definitions.
Figure 9: UCM6116 Web GUI Login Page To access the Web GUI: 1. Connect the computer to the same network as the UCM6102/UCM6104/UCM6108/UCM6116; 2. Ensure the device is properly powered up and shows its IP address on the LCD; 3. Open a Web browser on the computer and enter the web GUI URL in the following format: http(s)://IP-Address:Port where the IP-Address is the IP address displayed on the UCM6102/UCM6104/UCM6108/UCM6116 LCD. By default, the protocol is HTTPS and the Port number is 8089.
Status: Displays PBX status, System Status and CDR. PBX: To configure extensions, call routes, call features, internal options, IAX settings and SIP settings. Settings: To configure network settings, change password, LDAP Server, HTTP Server, Email Settings and Time Settings. Maintenance: To perform firmware upgrade, backup configurations, cleaner setup, reset/reboot, syslog setup and troubleshooting.
SYSTEM SETTINGS This section explains configurations for system-wide parameters on the UCM6102/UCM6104/UCM6108/UCM6116. Those parameters include Network Settings, Change Password, LDAP server, HTTP server, Email settings and Time Settings. NETWORK SETTINGS LAN/WAN/802.
EAP-TLS EAP-PEAPv0/MSCHAPv2 Identity Enter 802.1X mode identity information. MD5 Password Enter 802.1X mode MD5 password information. 802.1X Certificate Select 802.1X certificate from local PC and then upload. 802.1X Client Certificate Select 802.1X client certificate from local PC and then upload. Table 8: UCM6102 NETWORK SETTINGS Settings -> Network Settings -> WAN IP Method Select DHCP, Static IP, or PPPoE. The default setting is DHCP.
EAP-TLS EAP-PEAPv0/MSCHAPv2 Identity Enter 802.1X mode identity information. MD5 Password Enter 802.1X mode MD5 password information. 802.1X Certificate Select 802.1X certificate from local PC and then upload. 802.1X Client Certificate Select 802.1X client certificate from local PC and then upload. Settings -> Network Settings -> Port Forwarding WAN Port Specify the WAN port number. Up to 8 ports can be configured. LAN IP Specify the LAN IP address. Up to 8 IP address can be configured.
Ping Enable. Enable or disable ICMP response for Ping request. The default setting is Yes. SYN Flood. Enable to prevent SYN Flood denial-of-service attack to the device. Death-of-Ping. Enable to prevent Death-of-Ping attack to the device. Create New Rule. Click on "Create New Rule" button and a new window will pop up to specify rule options. Figure 11: Create New Firewall Rule Table 9: Firewall Rule Settings Rule Name Specify the Firewall rule name.
TFTP HTTP LDAP Check the box to display advanced options. Advance Source Enter the source IP address and port. Destination Enter the destination IP address and port. Click on "Apply" button to save the change and then submit by clicking on "Apply Changes". The new rule will then display at the bottom of the page. Users can select to edit the rule, or select to delete the rule.
To access LDAP Server settings, go to Web GUI->Settings->LDAP Server. LDAP SERVER CONFIGURATIONS Figure 12: LDAP Server Configurations LDAP PHONEBOOK Users could use the default phonebook, edit the default phonebook as well as add new phonebook on the LDAP server. The first phonebook with default phonebook dn "ou=pbx,dc=pbx,dc=com" displayed on the LDAP server page is for extensions in this PBX. Users cannot add or delete contacts directly.
Figure 13: Add New LDAP Phonebook LDAP CLIENT CONFIGURATIONS To configure the LDAP client so the default PBX phonebook can be used, follow the instructions in the LDAP Client Configuration section.
Figure 14: GXP2200 LDAP Phonebook Configuration HTTP SERVER The UCM6102/UCM6104/UCM6108/UCM6116 embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML pages allow the users to configure the PBX through a Web browser such as Microsoft’s IE, Mozilla Firefox and Google Chrome. By default, the PBX can be accessed via HTTPS using Port 8089 (e.g., https://192.168.40.50:8089). Users could also change the access protocol and port as preferred under Web GUI->Settings->HTTP Server.
is enabled, the access using HTTP with Port 80 will be redirected to HTTPS with Port 8089. The default setting is "Enable". Protocol Type Select HTTP or HTTPS. The default setting is "HTTPS". Port Specify port number to access the HTTP server. Once the change is saved, the web page will be redirected to the login page using the new URL. Enter the username and password to login again.
from DHCP Option 2 in the local server automatically. The default setting is "Yes". Enable DHCP Option 42 If set to "Yes", the device is allowed to get provisioned for NTP Server from DHCP Option 42 in the local server automatically. This will then override the NTP Server manually configured on the PBX. The default setting is "Yes". Time Zone Select the proper time zone option so the PBX can display correct date and time accordingly.
PROVISIONING OVERVIEW Grandstream SIP Devices can be configured via Web interface as well as via configuration file through TFTP or HTTP/HTTPS download. All Grandstream SIP devices support a proprietary binary format configuration file as well as XML format configuration file.
be sent to the phone with config server path URL in the NOTIFY message body. The phone will then use the path to download the config file generated in the UCM6102/UCM6104/UCM6108/UCM6116. DHCP OPTION 66 This method should be used on the UCM6102 because only the UCM6102 has WAN and LAN port with LAN port supporting the router function. When the phone restarts (by default DHCP Option 66 is turned on), it will send out a DHCP DISCOVER request.
Table 13: Auto Provision Setting Enable Zero Config Enable or disable the zero config feature on the PBX. The default setting is Yes. Automatically Assign Extension If enabled, when the device is discovered, the PBX will automatically assign an extension to the device. The default setting is disabled. Starting Extension Specify the starting extension to be created/assigned.
Figure 16: Auto Discover The following figure shows a list of discovered phones. The MAC address, IP Address, Extension (if assigned), Version, Vendor, Model, Connect Status, Create Config, Options (Edit/Delete) are displayed in the list. Figure 17: Discovered Devices ASSIGNMENT In the discovered list, click on to assign an extension to the device. Figure 18: Assign Extension To Device Firmware Version 1.0.0.
Users could also directly create a new device and assign the extension at one time. Click on "Create New Device" and the following window will be popped out. Fill in the MAC address or IP address, and then select the extension to assign to the device. Click on "Save" to add the device to the provision list. Figure 19: Create New Device PROVISIONING After the discovery and assignment, reboot the device. It will download the config file and get provisioned with the assigned extension registered.
Figure 20: Provisioning Example 1 The above figure shows a common setup among small businesses, where the UCM6102/UCM6104/UCM6108/UCM6116 is placed behind a company’s router or firewall. The phones are in the same network as the UCM6102/UCM6104/UCM6108/UCM6116 and can be discovered automatically by UCM6102/UCM6104/UCM6108/UCM6116 using the Zero Config feature. Example 2: Firmware Version 1.0.0.
Figure 21: Provisioning Example 2 This is another typical setup. In this setup, the UCM6102/UCM6104/UCM6108/UCM6116 is placed directly over the internet (outside from the network where the phones are deployed). Under this topology, the UCM6102/UCM6104/UCM6108/UCM6116 cannot reach the phones on its own and the typical auto discovery will not work. In this case, the phones can still be provisioned.
EXTENSIONS CREATE NEW USER To manually create new user, go to Web GUI->PBX->Basic/Call Routes->Extensions. Click on "Create New User" and a new window will show to fill in the details. The configuration parameters are as follows. Table 14: Extension Configuration Parameters General Extension The extension number associated with the user. CallerID Name Configure the CallerID Name associated with the user. Number, letter, or space are allowed.
SIP Check SIP if the user is using SIP or a SIP device. IAX Check IAX if the user is using IAX or a IAX device. Analog Station Select the port number if the user is attached on the analog port of the PBX. SIP Settings NAT Use NAT when the PBX is on a public IP communicating with devices hidden behind NAT (e.g., broadband router). If there is one-way audio issue, usually it's related to NAT configuration or Firewall's support of SIP and RTP ports.
Only local subnets Only the user in specific subnet can register using the extension. Up to three subnet can be specified. A specific IP Address. Only the device on the specific IP address can register using the extension. The default setting is "Allow all". Disable Password If set to Yes, when dialing out with outgoing rules, the user doesn't need enter the password. Codec Preference Select audio and video codec for the user. The available codecs are: PCMU, PCMA, GSM, G.726, G.722, G.729, G.
Technology SIP Check SIP if the users are using SIP or a SIP device. IAX Check IAX if the users are using IAX or a IAX device. SIP Settings NAT Use NAT when the PBX is on a public IP communicating with devices hidden behind NAT (e.g., broadband router). If there is one-way audio issue, usually it's related to NAT configuration or Firewall's support of SIP and RTP ports. Call Reinvite By default, the PBX will route the media steams from SIP endpoints through itself.
fax to the default Email address in FAX setting page. Note: If enabled, FAX cannot use Passthrough. Disable Password If set to Yes, when dialing out with outgoing rules, the user doesn't need enter the password. Codec Preference Select audio and video codec for the user. The available codecs are: PCMU, PCMA, GSM, G.726, G.722, G.729, G.723, ILBC, ADPCM, LPC10, H.264, H.263, H.263p.
TRUNKS ANALOG TRUNKS Go to Web GUI->PBX->Basic/Call Routes->Analog Trunks to add and edit analog trunks. Click on "Create New Analog Trunk" to add a new analog trunk. Click on to edit the analog trunk. Click on to delete the analog trunk. The analog trunk options are listed in the table below. Table 16: Analog Trunk Configuration Parameters Channels Trunk Name Select the channel for the analog trunk.
hangup polarity switch. (default: 600ms)", RX Gain Gain for the receive channel of analog FXO port. Range: -13.5 (dB) to + 12.0 (dB). TX Gain Gain for the transmit channel of analog FXO port. Range: -13.5 (dB) to + 12.0 (dB). Ring Timeout unit: milisecond . Trunk (FXO) devices must have a timeout to determine if there was a hangup before the line was answered. This value can be tweaked to shorten how long it takes before asterisk considers a non-ringing line to have hungup.
Default value: f1=480,f2=620,c=250/250 VOIP TRUNKS Go to Web GUI->PBX->Basic/Call Routes->VoIP Trunks to add and edit VoIP trunks. Click on "Create New SIP/IAX Trunk" to add a new VoIP trunk first. Then click on to configure more options for the VoIP trunk. Click on to delete the VoIP trunk. The VoIP trunk options are listed in the table below. Table 17: VoIP Trunk Configuration Parameters Create New SIP/IAX Trunk Type Provider Name Select the VoIP trunk type.
SRTP Firmware Version 1.0.0.32 Enable SRTP for the VoIP trunk. The default setting is disabled.
CALL ROUTES OUTBOUND ROUTES An outgoing calling rule pairs an extension pattern with a trunk used to dial the pattern. This allows different patterns to be dialed through different trunks (e.g., "Local" 7-digit dials through a FXO while "Long distance" 10-digit dials through a low-cost SIP trunk). Users can also set up a failover trunk to be used when the primary trunk fails. Go to Web GUI->PBX->Basic/Call Routes->Outbound Routes to add and edit outbound rules.
Use Trunk Select the trunk for this outbound rule. Strip Allows the user to specify the number of digits that will be stripped from the beginning of the dialed string before the call is placed via the selected trunk. Example: The users will dial 9 as the first digit of a long distance calls. However, 9 should not be sent out via analog lines and the PSTN line. In this case, 1 digit should be stripped before the call is placed.
Click on to delete the inbound route. Table 19: Inbound Route Configuration Parameters Trunks Select the trunk to configure the inbound route. DID Pattern All patterns are prefixed with the "_". X: Any Digit from 0-9. Z: Any Digit from 1-9. N: Any Digit from 2-9. ".": Wildcard. Match one or more characters. "!": Wildcard. Match zero or more characters immediately. Example: [12345-9]: Any digit from 1 to 9. Privilege Level Select privilege level for the inbound rule.
Call Queue Conference Room Operator Hangup Voicemail Dial Code Congestion Local Extension by DID DID Features Dial Trunk If enabled, users can dial outbound calls by DID through inbound trunks. The privilege level can be set according to the corresponding inbound rules. DID Destination Select the DID destination. Only the selected category can be reached by DID. Firmware Version 1.0.0.32 User Extension. This is selected by default. Conference. Call Queue.
CONFERENCE BRIDGE Conference bridge configurations can be accessed under Web GUI->PBX->Call Features->Conference. Users could create, edit, view and delete conference bridges. The conference room status and activity will show in the page as well. Click on "Create New Conference Room" to add a new conference bridge. Click on to edit the conference room. Click on to invite a user to the conference. The user will receive the ring to join the conference.
Admin can always press 0 to invite other users to the conference. Announce Callers When enabled, announcement will be made to all conference participants when there is user joining in the conference. The default setting is disabled. Play Hold Music For First When enabled, the PBX will play Hold music to the first participant until Caller another user joins the conference room. The default setting is disabled.
IVR CONFIGURING IVR IVR configurations can be accessed under Web GUI->PBX->Call Features->IVR. Users could create, edit, view and delete IVR. Click on "Create New IVR" to add a new IVR. Click on to edit the IVR configuration. Click on to delete the IVR. Table 21: IVR Configuration Parameters Name Configure the name of the IVR. Letters, digits, underscore and hyphen are allowed. Extension Enter the extension number for users to access the IVR.
The event options are: Extension VoiceMail Conference Rooms VoiceMail Group IVR Ring Group Queues Page Group IVR Prompt Hangup CREATING IVR PROMPT To Record New IVR Prompt or Upload IVR Prompt, click on "Prompt" next to the "Welcome Prompt" option and the users will be redirected to IVR Prompt page. Or users could go to Web GUI->PBX->Internal Options->IVR Prompt page directly.
Figure 23: Record New IVR Prompt Specify the IVR file name. Select the format (GSM or WAV) for the IVR file. Select the extension which will be dialed for the user to start recording the voice prompt. Click the "Record" button. A request will be sent to the PBX and the PBX will then call the extension for recording. Pick up the call from the extension and start the recording.
VOICE PROMPT Language settings for voice prompt can be accessed under Web GUI->PBX->Internal Options->Language. Users could upload a voice prompt package and then select the language in the list with available language options. Figure 24: Language Settings For Voice Prompt Click on to select a voice prompt package from local PC. The uploaded file must be smaller than 20 megabytes with package structure: [Package] [voice prompt dir]│[... dir]│[... files] info.
VOICEMAIL CONFIGURING VOICEMAIL General Voicemail settings can be accessed via Web GUI->PBX->Call Features->Voicemail. Users could configure the PBX to send the users Email with the voicemail as attachment. Click on "Email Settings For Voicemails" button to configure the Email attributes and content. Table 22: Email Settings For Voicemails Attach Recordings to E-Mail If enabled, voicemails will be sent to user's Email address as the configured template. The default setting is enabled.
Dial 0 For Operator If enabled, the caller can press 0 to exit the voicemail application and connect to the configured operator's extension. Max Messages Per Folder Configure the maximum number of messages in users' voicemail folders. The default setting is 25. Max Message Time Configure the maximum length of the voicemail message (in seconds). The default setting is 2 minutes.
Enter the Voicemail Group Extension. The voicemail messages left to this extension will be forwarded to all the voicemail group members. Configure the Name to identify the voicemail group. Letters, digits, underscore and hyphen are allowed. Select available mailboxes from the right list and add them to the left list. Click Save the finish the configuration. Firmware Version 1.0.0.
RING GROUP Users could create extension for ring group which contains members that will receive the call with specific ring strategy if the group extension has incoming calls. CONFIGURING RING GROUP Ring group settings can be accessed via Web GUI->PBX->Call Features->Ring Group. Figure 27: Ring Group Click on "Create New Ring Group" to add ring group. Click on to edit the ring group. Click on to delete the ring group.
Seconds to Ring Each Member Configure the number of seconds to ring each member. If set to 0, it will keep ringing (users could configure the ring timeout on the phone side as well). The default setting is 30 seconds. Enable Voicemail If enabled, the ring group extension can use voicemail. Secret Configure the password to access the ring group voicemail. Email Address Configure the Email address of the ring group extension. Figure 28: Ring Group Configuration Firmware Version 1.0.0.
PAGING AND INTERCOM GROUP Paging and intercom can be configured in group level under Web GUI->PBX->Call Features->Paging/Intercom. Click on "Create New Page/Intercom Group" to add page/intercom group. Figure 29: Page/Intercom Group Table 25: Page/Intercom Group Parameters Extension Configure page/intercom group extension. Type Select "2-way Intercom" or "1-way Page". Page/Intercom Group Members Select available users from the right list to the left.
Figure 30: Page/Intercom Group Settings Firmware Version 1.0.0.
CALL QUEUE CONFIGURING CALL QUEUE Call queue settings can be accessed via Web GUI->PBX->Call Features->Call Queue. Figure 31: Call Queue Click on "Create New Queue" to add call queue. Click on to edit the call queue. Click on to delete the call queue. Click on "Agent Login Settings" to configure Agent Login Extension Postfix and Agent Logout Extension Postfix.
Note: Music On Hold classes can be managed from PBX->Internal Options->Music On Hold. Leave When Empty Configure whether forcing the caller to leave if the call queue has no agent. Yes: Callers will be forced to leave the call queue if the call queue is empty. No: Never force the callers to leave the call queue when the queue is empty. Strict: Callers will be forced to leave the call queue if the agents are paused, invalid or unavailable. This is the default setting.
MUSIC ON HOLD Music On Hold settings can be accessed via Web GUI->PBX->Internal Options->Music On Hold. In this page, users could configure music on hold class and the music files. The "default" Music On Hold class already have 5 audio files defined for users to use. Figure 32: Music On Hold Default Class Click on "Create New MOH Class" to add a new Music On Hold class. Click on Click on to select music file from local PC and click on uploaded has to be 8 KHz Mono format with size less than 5M.
FAX/T.38 On the UCM6102/6104/6108/6116, the Fax extension can receive T.38 Fax to the specified Email address. Fax/T.38 settings can be accessed via Web GUI->PBX->Internal Options->FAX/T.38. CONFIGURING FAX/T.38 Click on "Create New Fax Extension". In the popped up window, fill the extension, name and Email address to send the received FAX to. Click on "Settings to Fax" to configure the following options. Table 27: FAX/T.
INTERNAL OPTIONS The configuration for PBX internal options can be accessed via Web GUI->PBX->Internal Options. INTERNAL OPTIONS/GENERAL General Preferences Global OutBound CID This is the default global CallerID that is used for all outgoing calls when no other CallerID is defined. If ther "User" tab or "VoIP Trunks" tab does not have defined CallerID neither, this Global OutBound CID will be used for CallerID.
properly. INTERNAL OPTIONS/FEATURE CODES Call Feature Description Blind Transfer Default code: #1. Enter the code during active call. After hearing "Transfer", enter the number to transfer to. Then the user will be disconnected. Options - Neither: Disable the feature code. - Caller Enable: Enable the feature code on caller side only. - Callee Enable: Enable the feature code on callee side only. Attended Transfer Default code: *2. Enter the code during active call.
- Caller Enable: Enable the feature code on caller side only. - Callee Enable: Enable the feature code on callee side only. Audio Mix Record Default code: *3. Enter the code to record the audio call and the PBX will mix the streams natively on the fly as the call is in progress. Options - Neither: Disable the feature code. - Caller Enable: Enable the feature code on caller side only. - Callee Enable: Enable the feature code on callee side only. Do Not Disturb (DND) Active Default code: *77.
voicemail box. Voice Mail Main Agent Pause Agent Unpause Paging Prefix Default Code: *97. Press *97 to access the voicemail box. Default Code: *83. Pause the agent in all call queues. Default Code: *84. Unpause the agent in all call queues. Default Code: *81. To page an extension, enter the code followed by the extension number. Intercom Prefix Default Code: *80. To intercom an extension, enter the code followed by the extension number.
Current, and AC Impedance as predefined for your country's analog line characteristics. Select the country in the list. FCC is equivalent to United States. TBR21 is equivalent to Austria, Belgium, Denmark, Finland, France, Germany, Greece, Iceland, Ireland, Italy, Luxembourg, Netherlands, Norway, Portugal, Spain, Sweden, Switzerland, and the United Kingdom. If option is specified, FCC will be used by default. ACIM Override Check to override AC Impedance.
INTERNAL OPTIONS/STUN MONITOR STUN Server Configures the STUN server to query. Valid format: [(hostname | IP-address) [':' port] The default port number is 3478 if not specified. Leave this field blank to disable STUN. STUN Refresh Number of seconds between STUN Refreshes. The default setting is 30 seconds. Firmware Version 1.0.0.
PBX SETTINGS IAX SETTINGS The UCM6102/UCM6104/UCM6108/UCM6116 IAX Settings can be accessed via Web GUI->PBX->IAX Settings. IAX SETTINGS/GENERAL Bind Port Allows iax2 to listen to another port. The default setting is 4569. Bind Address Forces iax2 to bind to a specific address instead of all addresses. The default setting is 0.0.0.0. IAX1 Compatibility Enables/disables iax1 style compatibility. No Checksums Enables/disables checksums.
H.263 H.263p H.264 IAX SETTINGS/JITTER BUFFER Enable Jitter Buffer Enables the use of jitter buffer on the receiving side of a SIP channel. Force Jitter Buffer Forces the use of jitter buffer on the receiving side of a SIP channel. Drop Count Configures drop count. MAX Jitter Buffer Configures the maximum time (in milliseconds) 0 for the buffer. MAX Interpolation Frames Configures the maximum number of interpolated frames the jitter buffer should return consecutively.
Trunk Time Stamps Enables/disables attaching time stamps to trunk frames. IAX SETTINGS/SECURITY Call Token Optional A single IP address or a range of IP addresses for which call token validation is not required in the form 11.11.11.11 or 11.11.11.11/22.22.22.22. Max Call Numbers Max Limits the amount of call numbers allowed for a single IP address. Nonvalidated Call Limits the amount of nonvalidated call numbers for all IP addresses Numbers combined.
"From:" header as the \"name\" part. If no "fromuser" is configured, the \"user\" part of the URI in the "From:" header will be filled with this value as well. From Domain Configures the domain in the "From:" field of the SIP header. It may be required by some providers for authentication. Auto Domain When turned on, the UCM6102/UCM6104/UCM6108/UCM6116 will add local host name and local IP to domain list.
with big jumps in/broken timestamps sent from exotic devices and programs. The default setting is 1000. Implementation The Jitter buffer implementation used on the receiving side of a SIP channel. Users could select "Fixed" (with size always equals to jbmaxsize) or "Adaptive" (with variable size which is the new jb of IAX2). SIP SETTINGS/MISCELLANEOUS Register Register as a SIP user agent to a SIP proxy (provider).
SIP SETTINGS/TLS AND TCP TCP Enable Enables/disables server for incoming TCP connections. The default setting is "No". TCP Bindaddr IP address for TCP server to bind to (0.0.0.0: binds to all interfaces). The default port number is 5060 if not specified. TLS Enable Enables/disables server for incoming TLS (secure) connections. The default setting is "No". TLS Bindaddr IP address for TLS server to bind to (0.0.0.0: binds to all interfaces). The default port number is 5061 if not specified.
UCM6102/UCM6104/UCM6108/UCM6116 is behind NAT. If it's a hostname, it will only be looked up only. External Host Specifies an external host, which is similar to External Address except the hostname will be looked up every "External Refresh" interval and Asterisk will perform DNS queries periodically. External Refresh Configures the refresh interval for the external host.
Note: Some devices do not support this (especially if one of them is behind NAT). SIP SETTINGS/ToS The following options are provided to configure SIP ToS on the UCM6102/UCM6104/UCM6108/UCM6116. ToS For Signaling Packets Configure the Type of Service for SIP packets. The default setting is None. ToS For RTP Audio Packets Configure the Type of Service for RTP audio packets. The default setting is None. ToS For RTP Video Packets Configure the Type of Service for RTP video packets.
setting doesn't apply to calls on hold. RTP Hold Timeout When the call is on hold, if there is no RTP activity in the timeout (in seconds) configured in this option, the call will be terminated. This value of RTP Hold Timeout should be larger than RTP Timeout. The default setting is no timeout. Trust Remote Party ID Configure whether the Remote-Party-ID should be trusted. The default setting is disabled. Send Remote Party ID Configure whether the Remote-Party-ID should be sent.
STATUS AND REPORTING PBX STATUS The UCM6102/UCM6104/UCM6108/UCM6116 monitors the status for Trunks, Extensions, Queues, Conference Rooms, Interfaces and Parking lot. It presents administrators the real time status in different sections under web GUI->Status->PBX Status. Figure 33: Status->PBX Status TRUNKS Users could see all the configured trunk status in this section. Figure 34: Trunk Status Table 28: Trunk Status Displays trunk status.
Busy Unavailable Unknown Error For SIP Peer trunk, the possible status are: Unreachable: The hostname cannot be reached. Unmonitored: QUALIFY feature is not turned on to be monitored. Reachable: The hostname can be reached. For SIP Register trunk, the possible status are: Registered Unrecognized Trunk Trunks Displays trunk name Displays trunk Type.
Figure 35: Extension Status Table 29: Extension Status Displays extension number (including feature code). The color indicator has the following definitions. Extension Name/Label Green: Free Blue: Ringing Yellow: In Use Grey: Unavailable Displays name (callerID name) or label (feature code function) for the extension. Displays message status for the extension. Status Example: 2/4/1 Description: There are 2 urgent messages, 4 messages in total and 1 message that has been read already.
Click on "Extensions", the web page will redirect to extension configuration page which can also be accessed via web GUI->PBX->Basic/Call Routes->Extensions. Click on Click on one of the tabs to refresh the extension status. to display the corresponding extensions accordingly. Click on [ + ] to expand the status detail table. Click on [ - ] to hide the status detail table. QUEUES Users could see all the configured call queue status in this section.
Click on "Queues", the web page will redirect to call queue configuration page which can also be accessed via web GUI->PBX->Call Features->Call Queue. Click on Click on [ + ] to expand the call queue detail. Click on [ - ] to hide the call queue detail. to refresh the call queue status. CONFERENCE ROOMS Users could see all the conference room status in this section. It shows all the configured conference rooms, current users and call duration for each user, as well as conference duration.
Figure 38: UCM6116 Interfaces Status Table 31: Interface Status Indicators USB connected. USB disconnected. SD Card connected. SD Card disconnected. LAN/WAN connected. LAN/WAN not configured. LAN/WAN disconnected. FXS/FXO connected. FXS/FXO waiting. FXS/FXO busy. FXS/FXO not configured. FXS/FXO disconnected.
PARKING LOT The UCM6102/UCM6104/UCM6108/UCM6116 supports call park using feature code. When there is call being parked, this section will display the parking lot status. Figure 39: Parking Lot Status Table 32: Parking Lot Status Caller ID Displays the caller ID who parks the call. Channel Displays channel for the call park. Extension Displays the parking lot number where the call is parked/retrieved. Displays timeout (in seconds) for the parked call.
GENERAL Under Web GUI->Status->System Status->General, users could check the hardware and software information for the UCM6102/UCM6104/UCM6108/UCM6116. Please see details in the following table. Table 33: System Status->General Status ->System Status -> General Model Product model. Part Number Product part number. System Time Current system time. Up Time System up time since the last reboot. Idle Time System idle time since the last reboot. Boot Boot version. Core Core version.
STORAGE USAGE Users could access the storage usage information from Web GUI->Status->System Status->Storage Usage. It shows the available and used space for the following partitions. Configuration partition Asterisk server configuration files and service configuration files. Data partition Voicemail, recording files, IVR file, music on hold files and etc. USB disk USB disk will display if connected. SD Card SD Card will display if connected.
Figure 41: System Status->Resource Usage CDR (Call Detail Report) A Call Detail Record (CDR) is a data record produced by telephone exchange activities or other telecommunications equipment documenting the details of a phone call that passed through the PBX. The CDR is composed of the following data fields on the UCM6102/UCM6104/UCM6108/UCM6116. Start Time. Format: 2013-03-27 16:47:03. Duration. Format: 0:00:10. Source. Format: 6012. Destination. Format: 6005. Caller ID.
Figure 42: CDR Filter Table 35: CDR Filter Criteria Inbound calls Inbound calls are calls originated from a non-internal source (like a VoIP trunk) and sent to an internal extension. Outbound calls Outbound calls are calls sent to a non-internal source (like a VoIP trunk) from an internal extension. Internal calls Internal calls are calls from one internal extension to another extension, which are not sent over a trunk.
Users could perform the following operations on the call report. Sort Click on the header of the column to sort by this category. For example, clicking on "Start Time" one time to sort the report according to start time. Clicking on "Start Time" again to reverse the order. Download Records On the bottom of the page, click on "Download Records" button to export the report in .csv format. Delete All On the bottom of the page, click on "Delete All" button to remove all the call report information.
Table 36: CDR Statistics Filter Criteria Trunk Type Call Type Time Range Select one of the following trunk type. All SIP Calls PSTN Calls Select one or more in the following checkboxes. Inbound calls Outbound calls Internal calls External calls All calls By month (of the selected year). By week (of the selected year). By day (of the specified month for the year). By hour (of the specified date). By range. For example, 2013-01 To 2013-03.
UPGRADING AND MAINTENANCE UPGRADING UPGRADING VIA NETWORK The UCM6102/UCM6104/UCM6108/UCM6116 can be upgraded via TFTP/HTTP/HTTPS by configuring the URL/IP Address for the TFTP/HTTP/HTTPS server and selecting a download method. Configure a valid URL for TFTP, HTTP or HTTPS; the server name can be FQDN or IP address. Examples of valid URLs: firmware.grandstream.com The upgrading configuration can be accessed via Web GUI->Maintenance->Upgrade.
Firmware File Prefix If configured, only the firmware with the matching encrypted prefix will be downloaded and flashed into the UCM6102/UCM6104/UCM6108/ UCM6116. Firmware File Suffix If configured, only the firmware with the matching encrypted postfix will be downloaded and flashed into the UCM6102/UCM6104/UCM6108/ UCM6116. HTTP/HTTPS User Name The user name for the HTTP/HTTPS server. HTTP/HTTPS Password The password for the HTTP/HTTPS server. Click on "Save" and "Apply Changes".
Note: Please do not interrupt or power cycle the UCM6102/UCM6104/UCM6108/UCM6116 when the upgrading process is on. NO LOCAL FIRMWARE SERVERS For users that would like to use remote upgrading without a local TFTP server, Grandstream offers a NAT-friendly HTTP server. This enables users to download the latest software upgrades for their devices via this server. Please refer to the webpage: http://www.grandstream.com/support/firmware.
Users could backup the configurations for restore purpose under Web GUI->Maintenance->Backup->Local Backup. Before creating new backup file, select the backup option first. If the Config-File is selected only, the backup file will be saved in the flash of the device. If Voice-File, Voicemail-File, Voice-Records or CDR is selected, external storage devices (USB Flash drive or SD Card) will be required because the backup file might be too large.
Figure 49: Network Backup Table 38: Network Backup Configuration Enable Backup Enable the auto backup function. Account Enter the Account name on the SFTP backup server. Password Enter the Password associate with the Account on the SFTP backup server. Server Address Enter the SFTP server address. Backup Time Enter 0-23 to specify the backup hour of the day. All the backup logs will be listed on the bottom of the page.
Figure 50: Cleaner Table 39: Cleaner Configuration Enable CDR Cleaner Enable the CDR Cleaner function. CDR Clean Time Enter 0-23 to specify the hour of the day to clean up CDR. Clean Interval Enter 1-30 to specify the day of the month to clean up CDR. Enable VR Cleaner Enter the Voice Records Cleaner function. VR Clean Threshold Specify the Voice Records threshold from 0 to 99 by using local storage status in percentage.
User Configuration: All the Extensions, Trunks and Routing configurations, as well as the local settings (network settings, upgrading setting and etc) will be cleared. User Data: All the data including voicemail, recordings, IVR Prompt, Music on Hold, CDR and backup files will be cleared. All: All the configurations and data will be reset to factory default.
Figure 52: Ethernet Capture The output result is in .pcap format. Therefore, users could specify the capture filter as used in general network traffic capture tool (host, src, dst, net, protocol, port, port range) before starting capturing the trace. PING Enter the target host in host name or IP address. Then press "Start" button. The output result will dynamically display in the window below. Figure 53: PING Firmware Version 1.0.0.
TRACEROUTE Enter the target host in host name or IP address. Then press "Start" button. The output result will dynamically display in the window below. Figure 54: Traceroute Firmware Version 1.0.0.
EXPERIENCING THE UCM6102/UCM6104/UCM6108/UCM6116 Please visit our website: http://www.grandstream.com to receive the most up- to-date updates on firmware releases, additional features, FAQs, documentation and news on new products. We encourage you to browse our product related documentation, FAQs and User and Developer Forum for answers to your general questions. If you have purchased our products through a Grandstream Certified Partner or Reseller, please contact them directly for immediate support.
Compliance FCC Notice This device complies with part15 of the FCC Rules. Operation is subject to the following two conditions: (1) This device may not cause harmful interference, and (2) this device must accept any interference received, including interference that may cause undesired operation. This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to part 15 of the FCC Rules.
Regulatory Information U.S. FCC Part 68 Statement This equipment complies with Part 68 of the FCC rules and the requirements adopted by the ACTA. The unit bears a label on the back which contains among other information a product identifier in the format US: GNIIS00BUCM6104. If requested, this number must be provided to the telephone company. This equipment uses the following standard jack types for network connection: RJ11C. This equipment contains an FCC compliant modular jack.
