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1. Figure 15 Auto Provision Setting Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 34 of 108 E istan innovative IP Voice amp Video 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 If the extension is assigned to existing device already this extension will be skipped and the next available extension will be used The default setting is 6000 Generate Random Password If enabled random password will be generated for the extension when it s created Otherwise default password will be used Default Password Specify default password for the extension if no random password is generated The default setting is admin Click on Save to start the discovery and provisioning process Reboot the device and the assigned extension will be registered after booting up MANUAL PROVISIONING DISCOVERY Users could manually discover the device by specifying the IP address or scanning the entire network Three methods are supported to scan the devices e PING e ARP e SIP MESSAGE OPTIONS Click on Auto Discover fill in the scan method and scan IP Then click on Save to start discovering
2. Limit the maximum amount of call numbers allowed for a single IP address If set to Yes call token is required If set to Auto it may lock out users who depend on backward compatibility when peer authentication credentials are shared between physical endpoints The default setting is Yes Other Settings SRTP FAX Detect Firmware Version 1 0 0 32 Enable SRTP for the call Enable to detect fax signal from the user trunk during the call and send the received fax to the Email address configured in this configuration page If no Email address can be found for the user send the received UCM6102 6104 6108 6116 USER MANUAL Page 43 of 108 Essen innova tive IP Voice 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 EDIT EXTENSION All the PBX extensions are listed under Web GUI gt PBX gt Basic Call Routes gt Extensions with Caller D Name Technology IP Port and registration status displayed Each extension has a checkbox to be selected and options for users to edit e Edit single extension Click on to start editing the extension The configuration options are listed in Table 14 Extensi
3. Allow Guest Calls Enables disables guest calls Overlap Dialing Support Enables disables dialing support Allow Transfer Enables disables all transfers unless enabled in peers or users initiated by the endpoint The Dial options t and T are not related to whether SIP transfers are allowed or not Enable DNS SRV Lookups on outbound calls MWI From When sending MWI NOTIFY requests this value will be used in the Enables disables DNS SRV lookups on calls Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 78 of 108 From Domain Auto Domain Allow External Domains SIP SETTINGS CODECX GE siren 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 Configures the domain in the From field of the SIP header It may be required by some providers for authentication When turned on the UCM6102 UCM6104 UCM6108 UCM61 16 will add local host name and local IP to domain list Allow requests for domains that are UCM6102 UCM6104 UCM6108 UCM61 16 not served by the The following codes are supported in UCM6102 UCM6104 UCM6108 UCM6116 for IAX Select the codecs Co from the right side list to the left side Click on AS amp to arrange the order e PCMU e PCMA e GSM e ILBC e G 722 e G 726 e ADPCM e LPC10 e G 729 e G 723 e H 263 e H 263p e H 264 SIP SETTINGS JITTER BUFFER Enable Ji
4. Configures codec negotiation priority to Caller Host Disabled or Regonly Configures ToS bit for preferred IP routing Configures frequency of trunk frames measured in milliseconds UCM6102 6104 6108 6116 USER MANUAL Page 77 of 108 Gos yam innovative 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 Limits the amount of call numbers allowed for a single IP address Max Nonvalidated Call Limits the amount of nonvalidated call numbers for all IP addresses Numbers combined Call Number Limits Limits the call numbers for a given IP range SIP SETTINGS The UCM6102 UCM6104 UCM6108 UCM6116 IAX Settings can be accessed via Web GUI gt PBX gt SIP Settings SIP SETTINGS GENERAL Realm For Digest Realms MUST be globally unique according to RFC 3261 Configure this Authentication value as your host or domain name The default setting is asterisk If a system name is configured in asterisk conf this value will be set to the configured system name UDP Port to Bind to The default setting is 5060 IP Address to Bind to The default setting is 0 0 0 0 which means binding to all addresses Domain Use comma to separate a list of domains that the UCM6102 UCM6104 UCM6108 UCM6116 will be responsible for
5. IAX Thread Count IAX Max Thread Count Auto Kill Authentication Debugging Codec Priority Type of Service Trunk Frequency Firmware Version 1 0 0 32 Gos yam innovative Enables the use of jitter buffer on the receiving side of a SIP channel Forces the use of jitter buffer on the receiving side of a SIP channel Configures drop count Configures the maximum time in milliseconds 0 for the buffer Configures the maximum number of interpolated frames the jitter buffer should return consecutively Jumps in the frame timestamps over where the jitter buffer is resynchronized This feature is useful to improve the quality of voice with big jumps in broken timestamps sent from exotic devices and programs The default setting is 1000 Configures the maximum number in milliseconds to pad the jitter buffer Configures the minimum number in milliseconds to pad the jitter buffer Configures the jitter shrink rate Minimum duration in seconds of registrations subscriptions The default setting is 60 Maximum duration in seconds of incoming registration subscriptions The default setting is 3600 Configures number of IAX threads Configures maximum number of IAX threads When set to yes the connection will be terminated if ACK for the NEW message Is not received in 2000ms Users could also specify number in milliseconds in addition to yes and no Enables disables IAX related debug output in log messages
6. e Hardware DSP based 128ms tail length carrier grade line echo cancellation LEC e Supports up to 60 concurrent calls and up to 32 conference attendees e Flexible dial plan call routing site peering call recording e Automated detection and provisioning of IP phones video phones ATA and other endpoints for easy deployment e Hardware encryption accelerator to ensure strongest security protection using SRIP TLS and HTTPS TECHNICAL SPECIFICATIONS Table 1 TECHNICAL SPECIFICATIONS Analog Telephone FXS Ports 2 ports PSTN Line FXO Ports e UCM6102 2 ports e UCM6104 4 ports e UCM6108 8 ports e UCM6116 16 ports Network Interfaces e UCM6108 UCM6116 Single 10M 100M 1000M RJ45 Ethernet port with integrated PoE Plug IEEE 802 3at 2009 e UCM6102 UCM6104 Dual 10M 100M 1000M RJ45 Ethernet ports with integrated PoE Plug IEEE 802 3at 2009 NAT Router Yes UCM6102 only Peripheral Ports USB SD LED Indicators Power Ready Network PSTN Line USB SD Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 11 of 108 LCD Display Reset Switch ndstream innovative IP Voice amp Video 128x32 graphic LCD with DOWN and OK button Yes Voice Video Capabilities Voice over Packet Capabilities Voice and Fax Codecs Video Codecs QoS DTMF Methods Provisioning Protocol and Plug and Play Network Protocols Disconnect Methods LEC with NLP Packetized Voice Protocol Unit 128ms tail length
7. ndstream Innovative IP Voice amp Video Grandstream Networks in inc User TA Grandstream Networks Inc www grandstream com E istan innovative IP Voice amp Video UCM6102 UCM6104 UCM6108 UCM6116 User Manual Index CHANGE LOG ccsccsccecceccncccccnscnscnccncceucnucnsensenecnsensensenecuuenusnecneceuenecnecneeneenes 9 FIRMWARE VERSION 1 0 0 32 0 0 c cccccccccccccccccecccececcaceceacececcaceccauecusueuususueausnecuaueneaueneauauecusuecnsnennanens 9 WELCOME e KN KN KN NENNEN NNN NENNEN KEN ENN ENNEN NENNEN NENNEN NNN ENNEN K NNN KN EN NENNEN NN NNN NNN Au 10 PRODUCT ONERVIEMN NK NK NENNEN KEN ENNEN REN NNN KN NENNEN NENNEN KN ENNEN NENNEN NENNEN KEN V 11 KEATUREHIGHTUIGHTIe n nananana anea A LLA A ERARA A AALA Anaan nanna annann 11 TECHNICAL GSbPECIEICATIONG 11 INSTALLATION uuu cece cceccnccsccccccccccccceccenencencenceneeneuneunennunaeuneuceneuncuneeneuneuneuas 14 EQUIPMENT PACKAGING lannnannnnnnnnnnannnnnnnnnnnnnnnn nnan nn nananana aana E L AELA A ALEAR AERAR ALAARA AADA n anana annaa 14 CONNECTING YOUR UChMeGiO2 CHMGiOoaA UChMGiOog cChMeiie 14 CONNECTING THE UChMerOoz 14 CONNECTING THE UCM6104 WEE 15 CONNECTING THE UCMerOog 16 CONNECTING THE UCM611 6 ricciccdeccsederteceecvebveciceceinseaddivececdbeestecdecveetexie destebveciecsennstvaivereneaseed 17 SAFETY COMPLIANCES e aaaaaaaaaanaannnaannnnnnnnnnnnannnnnnnnnnnnnann nananana annann naa LEA PAALA RALEA AALA A ALLAR annann enaner ae ne 18 VE EHS CH E EE
8. 0 packet loss round trip min avg max 13 600 14 536 19 300 ms Done Figure 53 PING Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 105 of 108 ndstream innovative IP Voice amp Video 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 Traceroute Ci Target Host www google com Start Output Result traceroute Dignostic run traceroute to www google com 74 125 224 179 6 kk k T ae 61 01 csw3_LosAngeles1 Level3 net 4 69 137 10 14 700 ms 33 675 ms 14 675 ms 6 ae 1 60 edge1 LosAngeles9 Level net 4 69 144 10 14 000 ms ae 4 90_edge1_LosAngeles9 Level3 net 4 69 144 202 17 900 ms 11 725 ms 9 GOOGLE INC edge1 LosAngeles9 Level net 4 53 226 6 20 625 ms 21 550 ms 14 600 ms 10 64 233 174 236 64 233 174 236 13 325 ms 19 450 ms 13 900 ms 11 72 14 236 11 72 14 236 11 15 675 ms 15 025 ms 15 275 ms 12 lax02s501 in f19 1e100 net 74 125 224 179 13 775 ms 11 925 ms Done Figure 54 Traceroute Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 106 of 108 E istan innovative IP Voice amp Video EXPERIENCING THE UCM6102 UCM6104 UCM6108 UCM6116 Please visit our website http www grandsiream 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 doc
9. 2 Screws Yes 6 CONNECTING YOUR UCM6102 UCM6104 UCM6108 UCM6116 CONNECTING THE UCM6102 Naviation Keys LED Indicators LAN LED WAN LED USB LED SDCardLED FXSLED FXO LED Figure 1 UCM6102 Front View Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 14 of 108 andstream innovative IP Voice amp Video SD Card Slot Reset WAN Port Ground USB Port DC 12V LAN Port 2x FXS Port 2x FXO Port 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 The LCD in the front will show its hardware information when the boot process is done 5 Once the UCM6102 is successfully connected to network the LED indicator for WAN 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 CONNECTING THE UCM6104 GH LCD Naviation 7 Keys d a K co LED C Indicators LAN 1 LED LAN2LED USB LED SDCardLED FXS LED FXO LED Figure 3 UCM6104 Front View Firmware Ve
10. 97 e Press 97 to access the voicemail box Agent Pause e Default Code 83 e Pause the agent in all call queues Agent Unpause e Default Code 84 e Unpause the agent in all call queues Paging Prefix e Default Code 81 e To page an extension enter the code followed by the extension number Intercom Prefix e Default Code 80 e To intercom an extension enter the code followed by the extension number Call Pickup e Default Code e To pick up a call for extension xxxx enter the code followed by the extension number xxxx INTERNAL OPTIONS RTP SETTINGS RTP Start RTP port starting address The default setting is 10000 RTP End RTP port ending address The default setting is 20000 Strict RTP Enables disables strict RTP protection When enabled RTP packets that do not come from the source of the RIP stream will be dropped The default setting is Disable RTP Checksums Enables Disables RTP Checksums The default setting is Disable INTERNAL OPTIONS HARDWARE CONFIG Analog Hardware Options Select Loop Start or Kewl Start for each FXS port and FXO port Tone Region Select country for default tones dial tone busy tone ring tone and etc Advanced Settings FXO Opermode Specify On Hook Speed Ringer Impedance Ringer Threshold Current Limiting TIP RING voltage adjustment Minimum Operational Loop Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 73 of 108 ACIM Override FXS Opermode T
11. Example 2 UCM61xx PSTN Lines gt gt 3 GXP Phone GXP Phone GXV Phone Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 38 of 108 E istan innovative IP Voice amp Video Figure 21 Provisioning Example 2 This is another typical setup In this setup the UCM6102 UCM6104 UCM6108 UCM61 16 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 But the UCM6102 UCM6104 UCM6108 UCM6116 will need help to get the phones to point itself to the UCM6102 UCM6104 UCM6108 UCM6116 first One possible solution could be as follows e Turn on DHCP Option 66 in the network where the phones are deployed and set the value as option tftp server name http s ucm_ip_address port zccg e All Grandstream phones have DHCP Option 66 turned on by default e Once the phone is provisioned with the DHCP Option 66 it will be redirected to the UCM6102 UCM6104 UCM6108 UCM6116 and send request for config file e When the phone requests cfgMAC xml from the UCM6102 UCM6104 UCM6108 UCM6116 the UCM6102 UCM6104 UCM6108 UCM6116 will add the phone to the provision list Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 39 of 108 CREATE NEW USER ndstream innovative IP Voice amp Vide
12. nnnononnnnnnannnnesnnnnnsssnrnnssnrrresrnnnresnnrrrossrrrersnrrrernnrrreennrrreserrrerenrerer 11 Table 2 UCM6102 UCM6104 EQUIPMENT PACKAGING ssnnsooonnneosnnnnussnnnnnssrrnnossrrreosrrrrerernrresenrrreserener 14 Table 3 UCM6108 UCM6116 EQUIPMENT PACKAGING cc ccesccccceessecceeeeeeeeeeeeeeseceeeeaeseeeeesanneeeetens 14 Table 4 LOD MENU OP TION LE 19 Table 5 UCM6102 UCM6104 LED INDICATOR 20 Table 6 UCM6108 UCM6116 LED INDICATORG 20 Table 7 Tea ER le 23 Table 8 UCM6102 NETWORK SETTINGS asonnnsnonnnnensnnnnnssnrnnoserrrnosrrrrursrnrrrssrrrroserrrersnnrrernrnrreennrrreserrnee 24 Table 9 Firewall Rule Getpnges 26 TOETO e RR OHV oe HIS EE 30 Table 112 IR ln e E EN Ee KR ien Le EN EN Table Kr Be E e Sein EE 35 Table 14 Extension Configuration Parameters sannnnsnnnnsennnsnnnnoanrnosnrnonnrrrnnnresnrrrsnrrrnnnrrrnnrrrenrrrenrrene 40 Table 15 Batch Add Extension Parameters cccccccccceeececeeceeseeceseeeeseuceseeeeeseucesseceesegeesseeesaeeesseeeesaes 42 Table 16 Analog Trunk Configuration Parametere 45 Table 17 VoIP Trunk Configuration Parametere 47 Table 18 Outbound Route Configuration Parameters cccccsecccccssececceeeeeeceeeeeceeseeeeeueeeeeeuseeesaeeeeesaaeees 49 Table 19 Inbound Route Configuration Parameters nnnn0annnnannnnannnnennnonnnrennnresnrnrsnrrrsnnrrennrrrsnrrrenrrene 51 Table 20 Conference Bridge Configuration Parametere 53 Table 217 IVR GOnuGQuration Para
13. 6104 6108 6116 USER MANUAL Page 36 of 108 andstream innovative IP Voice amp Video 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 Create New Device Mac Address IP Address Extension Version Model Cancel save 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 EXAMPLES Depending on the topology the discovery and provisioning can be done in different ways Example 1 Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 37 of 108 tream innovative IP Voice amp Video Router gt Remote Extension GXP Phone GXP Phone GXV Phone UCM61xx PSTN Lines External Storage for Unlimited Call Recording and future video recording 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
14. 6108 6116 USER MANUAL Page 17 of 108 E istan innovative IP Voice amp Video 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 Use the universal power adapter provided with the UCM6102 UCM6104 UCM6108 UCM6116 package only The manufacturer s warranty does not cover damages to the device caused by unsupported power adapters WARRANTY If the UCM6102 UCM6104 UCM6108 UCM6116 was purchased from a reseller please contact the company where the device was purchased for replacement repair or refund If the device was purchased directly from Grandstream contact the Grandstream Sales and Service Representative for a RMA Return Materials Authorization number before the product is returned Grandstream reserves the right to remedy warranty policy without prior notification A Warning Use the power adapter provided with the UCM6102 UCM6104 UCM6108 UCM6116 Do not use a different power adapter as this may damage the device This type of damage is not covered under warranty Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 U
15. Extension Range Firmware Version 1 0 0 32 If enabled random password will be generated when the extension is created The default setting is enabled It is recommended to enable it for security purpose If set to Yes users could disable specified extension range The default extension range assignment are User Extension 6000 6299 Conference Extension 6300 6399 IVR Extension 7000 7100 Ring Group Extension 6400 6499 Queue Extensions 6500 6599 VoiceMail Group Extension 6600 6699 Note lt is recommended to keep the system assignment for PBX to work UCM6102 6104 6108 6116 USER MANUAL Page 70 of 108 andstream innovative IP Voice amp Video properly INTERNAL OPTIONS FEATURE CODES Blind Transfer Attended Transfer Disconnect Call Parking Audio Record Firmware Version 1 0 0 32 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 Default code 2 Enter the code during active call After hearing Transfer enter the number to transfer to and the user will be connected to this number Hang up the call to complete the attended transfer Options Neither Disable the feature code Caller Enable Enable the feature code on call
16. ID Channel Extension Timeout 6010 SIP 6010 00000050 701 96 6005 SIP 6005 00000052 702 113 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 The status page will Timeout dynamically update this timer from 120 seconds default to 0 When the timer reaches 0 the caller who parked the call will be called back Other operations are also available in parking lot status section e Click on Parking Lot the web page will redirect to feature codes page which can also be accessed via web GUI gt PBX gt Internal Options gt Feature Codes ZC e Clickon to refresh the parking lot status e Click on to expand the parking lot details e Click on to hide the parking details SYSTEM STATUS The UCM6102 UCM6104 UCM6108 UCM6116 system status can be accessed via Web GUI gt Status gt System Status which displays the following system information e General e Network e Storage Usage e Resource Usage Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 91 of 108 ndstream innovative IP Voice amp Video GENERAL Under Web GUI gt Status gt System Status gt General users could check the hardware and software information for the UCM6102 UCM6104 UCM6108 UCM6116 Please se
17. If enabled the message duration will be announced at the beginning of the voicemail message The default setting is disabled If enabled introduction of each message will be played when accessed from the voicemail application The default setting is enabled If enabled users can review the message before sending out The default setting is disabled CONFIGURING VOICEMAIL GROUP Voicemail Group can be configured under Web GUI gt PBX gt Call Features gt Voicemail Group In this page users could create voicemail group which contains members that will receive the voicemail if the voicemail group extension has voice messages Click on Create New Voicemail Mail Group to configure the group Create New Voice Mail Group Voicellail Group Extension 6600 H ame VM G rou D Voicemail Group Mailboxes Available Mailboxes 6001 Amy60017 a 6000 EXT6000 a 6006 Alex Chan S 6002 Amy6002 a 6007 Emily Green Kei 6003 m 3 a 6005 John Doe ad 6004 Amyeo0 lt 3 6008 AX 6008 Firmware Version 1 0 0 32 Sa wes SS SS se Figure 26 Voicemail Group UCM6102 6104 6108 6116 USER MANUAL Page 60 of 108 andstream e Enter the Voicemail Group Extension The voicemail messages left to this extension will be forwarded to all the voicemail group members e Configure the Name to identify the voicemail group Letters digits underscore and hyphen are allowed e Select available mailboxes from th
18. Settings IAX SETTINGS GENERAL Bind Port Bind Address IAX1 Compatibility No Checksums Delay Reject ADSI Music On Hold Interpret Music On Hold Suggest Language Bandwidth IAX SETTINGS CODECS Allows iax2 to listen to another port The default setting is 4569 Forces iax2 to bind to a specific address instead of all addresses The default setting is 0 0 0 0 Enables disables iax1 style compatibility Enables disables checksums Enables disables iax2 to delay reject of calls to avoid DOS Enables disables ADSI phone compatibility Specifies Music On Hold class Suggests Music On Hold for the channel Configures default language for the channel This can be used by prompts Configures allowed codecs for different bandwidth requirement The default setting is Low The following codes are supported in UCM6102 UCM6104 UCM6108 UCM6116 for IAX Select the codecs from the right side list to the left side Click on A SW E to arrange the order e PCMU e PCMA e GSM e ILBC e G 722 e G 726 e ADPCM e LPC10 e G 729 e G 723 Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 76 of 108 e H 263 e H 263p e H 264 IAX SETTINGS JITTER BUFFER Enable Jitter Buffer Force Jitter Buffer Drop Count MAX Jitter Buffer MAX Interpolation Frames Recync Threshold Max Excess Buffer Min Excess Buffer Jitter Shrink Rate IAX SETTINGS REGISTRATION Min Reg Expire Max Reg Expire
19. UCM6102 6104 6108 6116 USER MANUAL Page 103 of 108 E istan e User Configuration All the Extensions Trunks and Routing configurations as well as the local settings network settings upgrading setting and etc will be cleared e User Data All the data including voicemail recordings IVR Prompt Music on Hold CDR and backup files will be cleared e All All the configurations and data will be reset to factory default Reset amp Reboot Factory Reset Mode Type User Configuration EN User Configuration User Data All Figure 51 Reset and Reboot Reboot SYSLOG On the UCM6102 UCM6104 UCM6108 UCM6116 users could dump to syslog information to a remote server under Web GUI gt Maintenance gt Syslog Enter the syslog server hostname or IP address and select the module level for the syslog information TROUBLESHOOTING On the UCM6102 UCM6104 UCM6108 UCM6116 users could capture traces ping remote host and traceroute remote host for troubleshooting purpose under Web GUI gt Maintenance gt Troubleshooting ETHERNET CAPTURE The captured trace can be downloaded for analysis Also the instructions or result will be displayed in the web GUI output result Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 104 of 108 ndstream innovative IP Voice amp Video Ethernet Capture Interface Type LAN CO Capture Filter host 192 168 40 178 k Start H Download Output Result capture Dig
20. 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 Once backup is done the list of the backups will be displayed with date and time Users then can download restore or delete it from the UCM6102 UCM6104 UCM6108 UCM6116 or the external device Backup Configuration Create New Backup Config File d Voice File v Voicemail File Ml Voice Records v CDR d List of Previous Configuration Backups 1 backup_2013mar26_180249 18 02 51 Mar 26 2013 J Ep Figure 48 Local Backup NETWORK BACKUP Users could backup the voice records voice mails CDR FAX in a daily basis via SFTP protocol automatically under Web GUI gt Maintenance gt Backup gt Network Backup Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 101 of 108 Backup Configuration OO Enable Backup CO Account CO Password i Server Address CO Backup time Enable Backup Account Password Server Address Backup Time andstream innovative IP Voice amp Video 67 110 250 179 Figure 49 Network Backup Table 38 Network Backup Configuration Enable the auto backup function Enter the Account name on the SFTP backup server Enter the Password associate with the Account on the SFTP backup server Enter the SFTP server address Enter 0 23 to specify the backup hour of the day All the backup logs will be listed o
21. Voicemail recording files IVR file music on hold files and etc e USB disk USB disk will display if connected e SD Card SD Card will display if connected Storage Usage Configuration Partition Total 96MB Available 54 MB EZ Used 40 MB USB Disk a Partition 1 Total 1 888MB Available 1 855 MB PS Used 31 MB Data Partition Total 3 168MB Available 3 071 MB E Used 95 MB SD Card 1 Partition 1 Total 3 808MB Available 3 455 MB SZ Used 351 MB Figure 40 System Status gt Storage Usage RESOURCE USAGE When configuring and managing the UCM6102 UCM6104 UCM6108 UCM6116 users could access resource usage information to estimate the current usage and allocate the resources accordingly Under Web GUI gt Status gt System Status gt Resource Usage the current CPU usage and Memory usage are shown in the pie chart Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 93 of 108 andstream innovative IP Voice amp Video Resource Usage CPU Usage Memory Usage Figure 41 System Status gt 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 UCM61 16 e Start Time Format 2013 03 27 16 47 03 e Duration Format 0 00 10 e S
22. XML format configuration file The UCM6102 UCM6104 UCM6108 UCM61 16 provides a Plug and Play mechanism to auto provision the Grandstream SIP devices in a zero configuration manner which allows users to finish the installation with ease and start using the SIP devices in a managed way To provision a phone three steps are involved i e discovery assignment and provisioning The UCM6102 UCM6104 UCM6108 UCM61 16 is capable to accomplish the following configurations on the SIP end point device e Assign an extension to the phone e Set up config server download path for further provisioning purpose e Setup LDAP client side configurations to use the PBX default phonebook This section explains how zero config works on the UCM6102 UCM6104 UCM6108 UCM6116 The settings for this feature can be accessed via Web GUI gt PBX gt Basic Call Routes gt Zero Config AUTO PROVISIONING By default the Zero Config feature is enabled on the UCM6102 6104 6108 6116 for auto provisioning Three methods of auto provisioning are used see below e SIP SUBSCRIBE The UCM6102 UCM6104 UCM6108 UCM6116 can automatically discover the phones in the same network using PnP feature with multicast SUBSCRIBE NOTIFY All current Grandstream phones support PnP feature and will send SUBSCRIBE at boot up and in the process be discovered by the PBX with the same type PnP feature support On the phone side after the phone boots up it will send out multicast SUBSCRIBE me
23. _ e xX Any Digit from 0 9 e Z Any Digit from 1 9 e N Any Digit from 2 9 e Wildcard Match one or more characters e Il Wildcard Match zero or more characters immediately Example 12345 9 Any digit from 1 to 9 Select privilege level for the inbound rule e Local The lowest level required All users can use this rule e National Users with National level or International level are allowed to use this rule e International The highest level required Only users with international level can use this rule Select the default destination e Extension e Extension s voicemail e Call Queue e Conference Room e Operator e Hangup e Voicemail Dial Code e Congestion e Local Extension by DID Time Condition Start Time End Time Date Destination Firmware Version 1 0 0 32 Select the start time hour minute for the trunk to use the inbound rule Select the end time hour minute for the trunk to use the inbound rule Select By Week or By Day and specify the date for the trunk to use the inbound rule Select the destination when the inbound rule is used at the configured time range e Extension e Extension s voicemail UCM6102 6104 6108 6116 USER MANUAL Page 51 of 108 ndstream innovative IP Voice amp Video Call Queue Conference Room Operator Hangup Voicemail Dial Code Congestion Local Extension by DID BD DM a 140 E Dial Trunk If enabled users can dial outbound cal
24. address displayed on the UCM6102 UCM6104 UCM6108 UCM6116 LCD By default the protocol is HTTPS and the Port number is 8089 For example if the LCD shows 192 168 40 167 please enter the following in your web browser https 192 168 40 167 8089 Enter the administrator s login and password to access the Web Configuration Menu The default administrator s username and password is admin and admin WEB GUI CONFIGURATIONS There are four main sections in the Web GUI for users to view the PBX status configure and manage the PBX Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 21 of 108 E istan innovative IP Voice amp Video e Status Displays PBX status System Status and CDR e PBX To configure extensions call routes call features internal options IAX settings and SIP settings e Settings To configure network settings change password LDAP Server HTTP Server Email Settings and Time Settings e Maintenance To perform firmware upgrade backup configurations cleaner setup reset reboot syslog setup and troubleshooting SAVING AND APPLYING CHANGES After configuring the web GUI options in one page click on the Save button on the bottom of the page if displayed After saving all the changes make sure click on Apply Changes button on the top right corner to submit all the changes Follow the prompted message to reboot the device if it s required MAKING YOUR FIRST CALL Power up the UCM6
25. monitored hosts The default setting is 100 Enables disables SIP debugging Records SIP history Dumps SIP history at the end of SIP dialog Configures a specific context for SUBSCRIBE requests This setting is useful to limit subscriptions to local extensions Enables disables support for subscriptions Sends out NOTIFY on ringing status UCM6102 6104 6108 6116 USER MANUAL Page 84 of 108 stream innovative IP Voice amp Video 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 gt Status gt PBX Status Trunks Conference Rooms gt Unmonitored Grandstream SIP 192 168 40 140 0 24 E 6000 0 24 Unavailable Trunk1 Analog Ports 1 Extensions gt Interfaces Status Ail Analog Features IAX SIP d Pe Extension S tus ype LAN H 6000 Jessages 0 0 0 SIP User Oo 6001 jes 0 0 0 SIP User 1 6002 0 0 0 SIP User des 6003 yes 0 0 0 SIP User 7 2 5 3 1 53 45 6004 Messages 0 0 0 SIP User EXO 97 Voice Mail Main Features 3 r 5 5 wei mei fal fe 98 Dial Voice Mail Features e Call Pickup Features 81 Pageing Prefix Features A 80 Intercom Prefix Features Parking Lot 83 Agent Pause Features ses 804 Agent Unpause Features aS MHS EATEN UNESU 71 Call Forward Busy Activate Features
26. 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 Default code 77 Default code 78 Default Code 71 Enter the code and follow the voice prompt Or enter the code followed by the extension to forward the call Default Code 72 Default Code 73 Enter the code and follow the voice prompt Or enter the code followed by the extension to forward the call Default Code 74 Default Code 75 Enter the code and follow the voice prompt Or enter the code followed by the extension to forward the call Default Code 76 Default Setting 1000 This is the maximum timeout in milliseconds between the digits input of the feature code Default Extension 700 During an active call press to park the call Default Extension 701 720 These are the extensions where the calls will be parked i e parking lots that the parked calls can be retrieved Default setting 120 This is the timeout for the call to be parked After the timeout if the call is not retrieved the extension who parked the call will be called back Default Code 98 Enter 98 and follow the voice prompt Or dial 98 followed by the extension and to access the entered extension s Page 72 of 108 ndstream innovative IP Voice amp Video voicemail box Voice Mail Main e Default Code
27. server users could also upload the firmware to the UCM6102 UCM6104 UCM6108 UCM6116 directly via Web GUI Please follow the steps below to upload firmware locally e Download the latest UCM6102 UCM6104 UCM6108 UCM6116 firmware file from the following link and save it in your PC http www grandstream com support firmware e Log inthe Web GUI as administrator in the PC sl e Go to Web GUI gt Maintenance gt Upgrade upload the firmware file by clicking on and select the firmware file from your PC e Click on de to start upgrading Local Upgrade Gi Firmware File Path e Upgrade Figure 47 Local Upgrade e Wait until the upgrading process is successful and a window will be popped up Click on OK to reboot the UCM6102 UCM6104 UCM6108 UCM6116 and check the firmware version when it boots up Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 99 of 108 E istan innovative IP Voice amp Video A 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 Alternatively users can download a free TFTP or HI TP server and conduct a lo
28. the devices within the same network Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 35 of 108 Auto Discover stream innovative IP Voice amp Video The PBX can automatically discover the new identifiable devices by ARP or PING It can scan this network segment or one ip address if there are new devices CO Scan Method Gi Scan IP Ping 192 Cancel Save 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 1 000B823E1D8D 2 000B823E1D7C 3 OQ00B823E5E88 4 000B823E175D 5 000B823E1D7F ASSIGNMENT In the discovered list click on to assign an extension to the device 192 168 40 249 192 168 40 122 192 168 40 207 192 168 40 145 192 168 40 163 1 0 2 12 1 0 1 40 1 0 5 23 1 0 1 40 1 0 2 12 Grandstream GXP2200 Grandstream GXP2200 Grandstream GXP2124 Grandstream GXP2200 Grandstream GXP2200 Figure 17 Discovered Devices Edit Device 0O00B823E1D C Firmware Version 1 0 0 32 Mac Address IP Address Extension Version Model 000B823E1D7C 192 168 400 122 1 0 1 40 GxAP2200 Cancel save Figure 18 Assign Extension To Device Connected Connected Connected Connected Connected No d D No 2 D No Z D No 2 i No Z D UCM6102
29. 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 CHANGE PASSWORD After login the Web GUI for the first time it is highly recommended for users to change the default password admin to more complicated password for security purpose Follow the steps below to change the Web GUI access password e Go to Web GUI gt Settings gt Change Password page e Enter the old password first e Enter the new password and retype the new password to confirm The new password field has to be at least 5 characters e Click on Save and the user will be logged out e Once the web page comes back to the login page again enter the username admin and the new password to login LDAP SERVER The UCM6102 UCM6104 UCM6108 UCM6116 has an embedded LDAP server for users to manage corporate phonebook in a centralized way By default the LDAP server has generated the phonebook based on the created extensions already If users have the Grandstream phone provisioned by the UCM6102 UCM6104 UCM6108 UCM6116 the LDAP directory has been set up on the phone and can be used right away Or users could manually configure the LDAP client settings accordingly to manipulate the built in LDAP server on the PBX Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 27 of 108 andstream innovative IP Voice amp Video
30. 0 16383 Select Tone Country Custom to edit manually UCM6102 6104 6108 6116 USER MANUAL Page 46 of 108 ndstream innovative IP Voice amp Video Default value f1 480 f2 620 c 250 250 VOIP TRUNKS Go to Web GUI gt PBX gt Basic Call Routes gt VoIP Trunks to add and edit VoIP trunks e Click on Create New SIP IAX Trunk to add a new VolP trunk first Then click on S to configure more options for the VolP trunk e Clickon 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 Select the VoIP trunk type e Peer SIP trunk e Register SIP trunk e Peer IAX trunk e Register IAX trunk Provider Name A unique label to help you identify this trunk when listed in outbound rules incoming rules etc Host Name IP address or URL for your VoIP providers server Provider Name Configure the provider name for the VoIP trunk Host Name Configure the host name for the VoIP trunk Transport Configure the SIP transport protocol UDP TCP or TLS Caller ID Configure the Caller ID CallerlID Name The new name of caller to replace when extension was not set CallerID Name Codec Preference Select audio and video codec for the VoIP trunk The available codecs are PCMU PCMA GSM G 726 G 722 G 729 G 723 ILBC ADPCM LPC10 H 264 H 263 H 263p Enable Qualify If enabled the PBX will send SIP OPTIONS to c
31. 102 UCM6104 UCM6108 UCM6116 and your phone with network connected Then follow the steps below to make your first call 1 Log in the UCM6102 UCM6104 UCM6108 UCM6116 web GUI go to PBX gt Basic Call Routes gt Extensions 2 Click on Create New User to create a new extension You might need User ID Password and Voicemail Password information to register and use the extension later 3 Register the extension on your phone with the User ID Password information 4 When your phone is registered with the extension and ready dial 97 to access the voicemail box Enter the Voicemail Password and you will be prompted with the Voice Mail Main menu 5 You are successfully connected to the PBX system now Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 22 of 108 ndstream innovative IP Voice amp Video 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 1X SETTINGS After successfully connecting the UCM6102 UCM6104 UCM6108 UCM61 16 to the network for the first time users could login the Web GUI and go to Settings gt Network Settings to configure the network parameters for the device depending on the network environment The settings are similar for UCM6104 UCM6108 UCM6116 The UCM6102 supports both WAN
32. 108 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 The LCD in the front will show its hardware information when the boot process is done 5 Once the UCM6108 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 CONNECTING THE UCM6116 frre y Cee yl Noon GC d LZ 8 4 H 4 4 SD Card Slot 2x FXS Port 16x FXO Port LED Indicators LCD a Figure 7 UCM6116 Front View DC 12V Reset LAN Port Ground Figure 8 UCM6116 Back View To set up the UCM6116 follow the steps below 1 Connect one end of an RJ 45 Ethernet cable into the LAN port of the UCM6116 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 UCM6116 Insert the main plug of the power adapter into a surge protected power outlet 4 Wait for the UCM6116 to boot up The LCD in the front will show its hardware information when the boot process is done Firmware Version 1 0 0 32 UCM6102 6104
33. 16 User manual Figure T WCMG6102 Front VOW caie eE E 14 Figure 2 UCM6102 Back View 15 Foure o UCIIG1 04 Front ViGW eise E E E 15 Figure A UCM6104 Back View 16 Fiure an UC MOTOS Front WiC Weess E E E E E A 16 Figure 6 UCM6108 Back View 16 IQS 7 UCMOTIG Froni VOW senai ca tcsrerti ics exmanedcectvance E E AE 17 Figure 8 UCM6116 Back View 17 Figure 9 UCM6116 Web GUI LOGIN Page 21 Figure 10 Current Gervice ai naain du oiea A EET nDNA naai 25 Figure 11 Create New Firewall ue 26 Figure 12 LDAP Server CGonfouratons 28 Figure 13 Add New LDAP Phonebook A 29 Figure 14 GXP2200 LDAP Phonebook Contfguraton 30 Figure 19 Aulo Provision Een Le EE 34 Figure 16 RTE wiissivcavinesssvtsaesvedeissctwantuseteviuaed bheiaislicvavvandasdauwebandedsantwaunessiscdbwesbhuddiaie cannes dasdaunes ens 36 Figure RER 36 Figure 18 Assign Extension TO Device s nnsnneennnsenronrrsrrrrrrrrrrrornrrsrrrrnrrsrrronrrnrrorrrrtrtnnnrrsrernntnnnnen nenne 36 Figure 19 Create New Device 37 Figure 20 Provisioning Example A 38 Figure 21 Provisioning Example 2 scscencscnmanicnasdmnnnanaiauneattsbaived Gna Aa a aaa 39 Figure 22 Click On Prompt To Create IVR brommt 56 Figure 23 Record New IVR Promp EE 5 7 Figure 24 Language Settings For Voice brommt 58 Figure 25 Default Email Template 59 Figure 26 Voicemail GrOUD D 60 FOWE A RDO FOU WE 62 Figure 28 Ring Group Configuration cccccccceccccceeseceeeeeeeceeeeseeeeceeeeeeee
34. 18 GETTING STARTED uu icc ceccccceccnccnccncccccncencenucceneeneuneuneenennununueucenneneeneeneeas 19 USING THE LCD MENU 19 USING THE LED INDICATORG cece ccc ccccecccccceccceccacececcaueceaceueguaueccuuuususueausueaesnentanenesueneataneass 20 USING THE WEB GU 20 AGGESSING WEB GU EE 20 WEB GUI CONEIGURHATIONG 21 SAVING AND APPLYING CHANGE S ccccccccecccccceccececscceuececuscsetsnsenueeensenuaueeususuesnsnsnussensnsesas 22 MAKING YOUR FIRST CAUL 22 SYSTEM SETTINGS wii ieee ccccccnccncccccccncencecceccecceneeneeneuneenenceneeneunueneuneuneuneenens 23 NETWORK GEIIINGS aannnnnnnnn nnn nnnnnnnnnnnnannnnn nnna unnan annann nan AAA LAERE EALLA ALEAR APARA LP ARADA A AL Anana a nanana 23 LAN WAN 802 1X GEITIINGG 23 NETWORK SECURITY SETTINGS cccccccccceccccecccccnacectacecsucenscesuensuectsnentanenecnauestanetsauenscenssenas 25 CHANGE DAGGWOHD A ALDAE ALAE ELEA LEARE A ADRAR ALAA A LLARA aa ranan n nnn 27 ED Se e o AET E E E EE E E E A AE E tinted bene A EE EE 27 LDAP SERVER CONFIGURATIONS 0 cccccecccccceceececsccececencecstenscenteneensuentenseuensnenusnennsnsensnsesas 28 LDAP PAON EBOOK EE 28 Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 1 of 108 E istan innovative IP Voice amp Video LDAP CLIENT CONEIGURATIONS eee seeetttnrnrnirrrrnnn ennnen nee ete nern rr rrn 29 HTTP E eiescteeenrveceten end saseaenmeneaaghnecf scene ea veepseees cotendsdegdecnsesvedeanestetaveeatcnseeesseeueacteasss 30 SIE nde
35. 21 1002 2021 228 lt 2021 gt ANSWERED 2013 03 08 01 41 59 0 00 03 2023 81003 2023 228 lt 2023 gt FAILED 5 2013 03 08 01 42 16 0 00 02 2023 81002 2023 228 lt 2023 gt FAILED 6 2013 03 08 01 42 40 0 00 03 2023 1002 2023 228 lt 2023 gt ANSWERED if 2013 03 08 01 49 24 0 00 07 1002 2023 1002 214 lt 1002 gt ANSWERED 8 2013 03 08 01 49 43 0 00 06 1003 2022 1003 214 lt 1003 ANSWERED 9 2013 03 11 17 14 38 0 00 04 6002 7000 6002 lt 6002 FAILED 10 2013 03 11 17 15 09 0 00 08 6002 6004 6002 lt 6002 gt ANSWERED Figure 43 Call Report Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 95 of 108 andstream innovative IP Voice amp Video Users could perform the following operations on the call report e 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 e Download Records On the bottom of the page click on Download Records button to export the report in csv format e Delete All On the bottom of the page click on Delete All button to remove all the call report information e Play Download Delete Recording File per entry If the entry has audio recording file for the call the three icons on the most right column will be activated for users to select In the following picture the first row has audio recording file for the call Figure 44 Ca
36. 31 TIME SETTINGS ecscecseecssecssecsssecssscesseesnscesneesnseesseesnseessessnseesneesnscesseesnseesneesneesseesneeesseesneeessesneeeee 31 ghd N C EE 33 OVERVIEW e R ses c E E E eee 33 AUTO PROVISIONING A 33 MANUAL PROVISIONING cessssesseecssessssesssseesueesnseesseesnseesneesnscesneesnscesnessneeesseesnseesueesneeeseesneeeeeses 35 ee 35 e oe ee a ec ee cen ee AE E ee ae ete ene oe ee etree 36 EH 37 EXAMPLE E 37 EXTENSIONS eRNERnE pee ne eS pr nse ne ree er renee errr rerrereee ee emer errr cere 40 CREATE NEW USER EENS 40 Sieste wu Ee 42 FIT TNO EO 44 SEI LE 45 ANALOG TRUNKS sssesesestessettttttttttttrtseeet bbb b btt btt tt ttrt trara nab E EEEE E EEE EEEE EE EEE e ree anna 45 Gell 47 Hee 49 OUTBOUND Ciel Dn 49 lieft cl TuS 50 CONFERENCE 151 BC Sennen enn manne rere ene inner nr nese rere ae re nerene rer eee rrrrrrrrre 53 A r PAR EEE A E E E E 55 CONFIGURING IVR eee eeeccsticeaceese cece cog ttt ttt ttr setae ctes AAMEN EEEE EEE EEE Err rana AEAEE EEEE EE EEEn Errr rnrn anane a annene 55 CREATING IVR PROMEPT 56 RECORD NEW IVR PROMEPT nennen nee eee nern rn rrn 56 UPLOAD IVR PROMPT cares ccee te aetecn seep nctecetvatecet2secses bpasdeas sctgioer ested eeteseteneacttereeerepadaeee 57 VOCE PRONE E 58 Aale AT 59 CONFIGURING VOICE N A sarepeccesyeecena tee aectaseeeceteceseee coer bbb bbt btt tt ettr rrr tanran ababab Ente EEEn rr rrr nrnna anean tanne 59 CONFIGURING VOICEMAIL GROUP ssssssesssessssesssseesse
37. Client S Select 802 1X client certificate from local PC and then upload Certificate Table 8 UCM6102 NETWORK SETTINGS Settings gt Network Settings gt WAN IP Method Select DHCP Static IP or PPPoE The default setting is DHCP IP Address Enter the IP address for static IP settings Gateway IP Enter the gateway IP address for static IP settings Subnet Mask Enter the subnet mask address for static IP settings DNS Server 1 Enter the DNS server 1 address for static IP settings DNS Server 2 Enter the DNS server 2 address for static IP settings User Name Enter the user name to connect via PPPoE Password Enter the password to connect via PPPoE Preferred DNS Server Enter the preferred DNS server address Settings gt Network Settings gt LAN Mode Select LAN port mode as Router or Switch IP Address Enter the IP address assigned to the LAN port Subnet Mask Enter the subnet mask DHCP Server Enable Enable or disable DHCP server capability DNS Server 1 Enter DNS server address 1 DNS Server 2 Enter DNS server address 2 Allow IP Address Enter the IP Pool starting address From Allow IP Address To Enter the IP Pool ending address Default IP Lease Time Enter the IP lease time in seconds Settings gt Network Settings gt 802 1X 802 1X Mode Select 802 1X mode The default setting is Disable The supported 802 1X mode are e EAP MD5 Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Pa
38. Configure the Call Forward No Answer target number If not configured the Call Forward No Answer feature is deactivated Configure the Call Forward Busy target number If not configured the Call Forward Busy feature is deactivated Configure the number of seconds to ring the user before sending to the user s voicemail if enabled or hangup The default setting is 60 seconds Technology Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 40 of 108 SIP LAY Analog Station ndstream Check SIP if the user is using SIP or a SIP device Check IAX if the user is using IAX or a IAX device Select the port number if the user is attached on the analog port of the PBX NAT Call Reinvite DTMF Mode Insecure Enable Keep alive Keep alive Frequency 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 By default the PBX will route the media steams from SIP endpoints through itself If enabled the PBX will attempt to negotiate with the endpoints to route the media stream directly It is not always possible for the PBX to negotiate endpoint to endpoint media routing The default setting is No Select DIMF mode for the user to send DIMF The default setting is RFC2833 If Info is selected SIP INFO message will be used If Inband i
39. D The caller id of the person who left the message e VM_MSGNUM The message number in your mailbox e VM_DATE The date and time the message was left Click on Load Default button to view a default template as an example From Subject Message Max Greeting Firmware Version 1 0 0 32 pbhx yourcompany null New voicemail from HWVM_CALLERID for SV MAILBOX Hello SVM NAME you receiwwed a message lasting SWM_DUR at HVM_DATE from SViM_CALLERID This is message HVM_MSGNUMY in your voicemail Inbox Cancel Load Defaults Figure 25 Default Email Template Table 23 Voicemail Settings Configure the maximum number of seconds for the voicemail greeting The default setting is 60 seconds UCM6102 6104 6108 6116 USER MANUAL Page 59 of 108 Dial 0 For Operator Max Messages Per Folder Max Message Time Say Message Caller ID Say Message Duration Play Envelope Allow Users To Review andstream innovative IP Voice amp Video If enabled the caller can press O to exit the voicemail application and connect to the configured operator s extension Configure the maximum number of messages in users voicemail folders The default setting is 25 Configure the maximum length of the voicemail message in seconds The default setting is 2 minutes lf enabled the caller ID of the user leaving the message will be announced at the beginning of the voicemail message The default setting is disabled
40. IENT CONFIGURATIONS To configure the LDAP client so the default PBX phonebook can be used follow the instructions in the LDAP Client Configuration section Suppose your server Base DN is dc Grandstream your extension number is 1000 and your LDAP entry password is 1000 configure your LDAP client as follows case insensitive Base DN dc Grandstream Root DN AccountName 1000 dc Grandstream Password 1000 Filter amp Caller DName AccountName Port 389 The following figure shows the configuration information on a GXP2200 to successfully use the LDAP server as configured in Figure 12 LDAP Server Configurations Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 29 of 108 server Address Port Base DN User Name Password LDAP Name Attributes LDAP Number Attributes LDAP Mail Attributes LDAP Name Filter LDAP Number Filter LDAP Mail Filter LDAP Displaying Name Attributes Max Hits search Timeout ms LDAP Lookup For Dial LDAP Lookup For Incoming Call ndstream innovative IP Voice amp Video 192 168 40 50 389 dc pbx dc com AccountName 605 dc pbx dc cc CallerlbName AccountName AccountName CallerlDName 7AccountName YoCallerIbName 50 0 Enable Enable Figure 14 GXP2200 LDAP Phonebook Configuration HTTP SERVER The UCM6102 UCM6104 UCM6108 UCM6116 embedded Web server responds to HTTP HTTPS GET POST requests Em
41. ISS Override PCMA Override Boost Ringer Fast Ringer Low Power Ring Detect MWI Mode Firmware Version 1 0 0 32 Gus ean innovative 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 Check to override AC Impedance Specify On Hook Speed Ringer Impedance Ringer Threshold Current Limiting TIP RING voltage adjustment Minimum Operational Loop 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 The default setting is FCC Check to override Two Wire Impedance Synthesis Specifies the codec to be used for analog lines North American users should choose PCMU All other countries unless otherwise known should be assumed to be PCMA If no user choice is specified the default is PCMU Configure whether normal ringing voltage 40V or maximum ringing voltage 89V f
42. No Parked Calls defined 72 Call Forward Busy Deactivate Features 73 Call Forward NoAnswer Activate Features First Prev Total 20 Show 1 2 Jumpto Queues Figure 33 Status gt PBX Status TRUNKS Users could see all the configured trunk status in this section Trunks TIA j Lt i Unmonitored Grandstream SIP 192 168 40 140 Unavailable Trunki Analog Ports 1 Figure 34 Trunk Status Table 28 Trunk Status Displays trunk status Status e For analog trunk the possible status are Available Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 85 of 108 Busy Unavailable Unknown Error e For SIP Peer trunk the possible status are Unreachable The hostname cannot be reached Goan innovative Unmonitored QUALIFY feature is not turned on to be monitored Reachable The hostname can be reached e For SIP Register trunk the possible status are Registered Unrecognized Trunk Trunks Displays trunk name Displays trunk Type Tute e Analog This trunk is an analog trunk Me e SIP This trunk is a SIP trunk e AX This trunk is an IAX trunk Username Displays username for this trunk Port Hostname IP Displays Port for analog trunk or Hostname IP for VoIP SIP IAX trunk Other operations are also available in trunk status section e Click on Trunks the web page will redirect to trunk configuration page which can also be accessed via web GUI gt PBX gt Basic Call Ro
43. Prompts Package Lists Language English Figure 24 Language Settings For Voice Prompt eS e 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 txt containing the language name for display in UTF8 e Click on t to start uploading e Voice Prompts Package Lists allows users to select the default language for voice prompt for internal calls inbound calls and outbound calls Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 58 of 108 CONFIGURING VOICEMAIL andstream innovative IP Voice amp Video VOICEMAIL General Voicemail settings can be accessed via Web GUI gt PBX gt Call Features gt 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 Attach Recordings to E Mail Template For Voicemail Emails Table 22 Email Settings For Voicemails If enabled voicemails will be sent to user s Email address as the configured template The default setting is enabled Fill in the From Subject and Message content The template variables are e t TAB e VM_NAME Recipient s firstname and lastname e VM_DUR The duration of the voicemail message e VM_MAILBOX The recipient s extension e VM_CALLERI
44. SER MANUAL Page 18 of 108 GE siren innova tive IP Voice 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 e Default LCD Display By default when the device is powered on the LCD will show device model hardware version and IP address e Menu Access Press Down or OK button to start browsing menu options e Menu Navigation Press the Down arrow key to browser different menu options Press the OK button to select an entry e Exit There is Back option in the menu Select it to go back to previous menu Also the LCD will come back to default display after being idle in menu for more than 20 seconds The following table shows the LCD menu options Table 4 LCD MENU OPTIONS View Events e Critical Events e Other Events Device Info e Hardware Hardware version number e Software Software version number e P N Part number e MAC MAC address e Uptime System up time Network Info e Mode DHCP Static IP or PPPoE e IP IP address e Subnet Mask Network Menu e LAN Mode Select LAN mode as DHCP Static IP or PPPoE Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 19 of 108 Goan innovative Factory Menu e LCD Test Patterns Press Down button to test different LCD
45. To access LDAP Server settings go to Web GUI gt Settings gt LDAP Server LDAP SERVER CONFIGURATIONS LDAP Server configurations Base DN dc pbx dc com Pbx dn ou pbx dc pbx dc com Root DN en admin dc pbx dc com Root Password seas Root Password Confirm siti Allow anonymous 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 The contacts information will need to be modified via Web GUI gt PBX gt Basic Call Routes Extensions first The default LDAP phonebook will then be updated automatically A new sibling phonebook of the default PBX phonebook can be added by clicking on Add under LDAP Phonebook section Once added users can select to edit the phonebook attributes and contact list see Figure below or select U to delete the phonebook Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 28 of 108 stream innovative IP Voice amp Video Edit Phonebook ou people dc pbx dc com LDAP Attributes Contact List AccountName CallerIDName Email FirstName Department MobileNumber HomeNumber D LastName D D Fax Cancel Figure 13 Add New LDAP Phonebook LDAP CL
46. bedded 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 httos 192 168 40 50 8089 Users could also change the access protocol and port as preferred under Web GUI gt Settings gt HTTP Server Table 10 HTTP Server Settings Redirect From Port 80 Enable or disable redirect from port 80 On the PBX the default access protocol is HTTPS and the default port number is 8089 When this option Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 30 of 108 Protocol Type Port Gouin innovative is enabled the access using HTTP with Port 80 will be redirected to HTTPS with Port 8089 The default setting is Enable Select HTTP or HTTPS The default setting is HTTPS 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 EMAIL SETTINGS The Email application on the UCM6102 UCM6104 UCM6108 UCM61 16 can be used to send out Emails to users with Fax e g Fax To Email Voicemail Voicemail To Email and other information as attachment The configuration parameters can be accessed via Web GUI gt Settings gt Email Settings TLS Enable Type Domain TIME SETTINGS Table 11 Email Settings Enable or disable TLS during transferrin
47. cal firmware upgrade A free windows version TFTP server is available for download from http support solarwinds net updates New customerFree cfm http tttpd32 jounin net Instructions for local firmware upgrade via TFTP 1 Unzip the firmware files and put all of them in the root directory of the TF IP server 2 Connect the PC running the TFTP server and the UCM6102 UCM6104 UCM6108 UCM61 16 device to the same LAN segment 3 Launch the TFTP server and go to the File menu gt Configure gt Security to change the TFTP server s default setting from Receive Only to Transmit Only for the firmware upgrade 4 Start the TFTP server and configure the TFTP server in the UCM6102 UCM6104 UCM6108 UCM6116 s web configuration interface 5 Configure the Firmware Server Path to the IP address of the PC 6 Update the changes and reboot the UCM6102 UCM6104 UCM6108 UCM6116 End users can also choose to download a free HTTP server from hittp nttod apache org or use Microsoft IIS web server BACKUP LOCAL BACKUP Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 100 of 108 tream innovative IP Voice amp Video Users could backup the configurations for restore purpose under Web GUI gt Maintenance gt Backup gt Local Backup Before creating new backup file select the backup option first e If the Config File is selected only the backup file will be saved in the flash of the device e f Voice File Voicemail File
48. carrier grade Line Echo Cancellation Dynamic Jitter Buffer Modem detection amp auto switch to G 711 G 711 A law U law G 722 G 723 1 5 3K 6 3K G 726 G 729A B iLBC GSM T 38 H 264 H 263 H 263 Layer 3 QoS Signaling and Control In Audio RFC2833 and SIP INFO TFTP HTTP HTTPS auto discovery and auto provisioning of Grandstream IP endpoints TCP UDP IP RTP RTCP ICMP ARP DNS DDNS DHCP NTP TFTP SSH HTTP HTTPS PPPoE SIP RFC3261 STUN SRTP TLS SIP Call Progress Tone Polarity Reversal Hook Flash Timing Loop Current Disconnect Busy Tone Media SRTP TLS HTTPS SSH Physical Universal Power Supply Environmental Dimensions Mounting e Output 12VDC 1 5A e Input 100 240VAG 50 60Hz e Operating 32 104 F 0 40 C 10 90 non condensing e Storage 14 140 F 10 60 C e UCM6102 UCM6104 226mm L x 155mm W x 34 5mm H e UCM6108 UCM6116 440mm L x 185mm W x 44mm H Wall mount and Desktop Additional Features Caller ID Polarity Reversal Wink Call Center Firmware Version 1 0 0 32 Bellcore Telcordia ETSI FSK ETSI DTMF SIN 227 BT NTT Japan pending Yes with enable disable option upon call establishment and termination Multiple configurable call queues automatic call distribution ACD UCM6102 6104 6108 6116 USER MANUAL Page 12 of 108 Customizable Attendant Concurrent Calls Conference Bridges Call Features Compliance F
49. ceeesseaeeeesseeseeeeesseaeeeessaaaeeeseseaaes 63 Figure 29 Pag intercom E te De c cutecencct esacscecanncsanaasiiwesiinnnsbeaddensiestaciiencaticstacaceedanctangaddewesisensbeaddeesisstactencatis 64 Figure 30 Page Intercom Group Gettngs 65 FUSS is CAN TE 66 Figure 32 Music On Hold Default Claes 68 FOUS eS a Uae ate ware sa cet erenone cca gee ce EE toaseea ent cessanecnoee E EA E AT 85 Figure 34 Trunk StatUS oe siertce fos setza sans soenn sessemanestectesaisentecaerepe airnean daaa soean seusennasoeeatenasensesverepeseesonaseneeeeeae 85 PIOUS oo EXCISION IS a E A A A 87 FOUE SO 0 21 eye E 88 Figure 37 Conference ROOM Gate 89 Figure 38 UCM6116 Interfaces Status c cc cccccccccsssececeeeeeeeeeeseeeeeeeeeeeeeceeesaeeeeeeeseeaeeeesseeeeeeeseaaaeeeeeeeaaes 90 FONC AE ING Lt Ta cect ters E E EAE 91 Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 7 of 108 Figure 40 Figure 41 Figure 42 Figure 43 Figure 44 Figure 45 Figure 46 Figure 47 Figure 48 Figure 49 Figure 50 Figure 51 Figure 52 Figure 53 Figure 54 Firmware Version 1 0 0 32 E istan innovative IP Voice amp Video System Status gt Storage Usage 2 cccccccsceceescdeeeccesccecececccceeeceececasecesceaseccesccsestensceeeseessceeesees 93 System Status gt Resource Usage 94 COUR FIE a A oe de tederecens 95 Ca FOO E 95 Call Report Entry With Audio Recording Ee 96 EBELE e 96 PEON UDO IO aaa css cesta sees aes
50. cesseesnecesseesnseesseesnecesseesneceseeesneeeeeeesneeeeeses 60 Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 2 of 108 E istan innovative IP Voice amp Video Segel swat ea dee eee acs ee cet eect ee ages 62 CONFIGURING RING GROUP sssccssssscessssesceesssessnecsseessessseesseessseusseaseeaesssesoseeatesstecateeateeateaaeesaes 62 RING GROUP PARAMETERS ttar saaan narrnrrrrrrrrrerrrrreererrerrereeeeet 62 PAGING AND INTERCOM GROUR ssseeRRREREREEEEEEEEEEREEEEREEEEEEEEEEEEEEEEE EN 64 Eege 66 CONFIGURING CALL OUEUE EE 66 CALL QUEUE IE Eege 66 LU bo ON TOLD o a E 68 FAA EO TE 69 ONE GINGA ce cere cnee cee eee ses tyes espera cect acee aces ecn esti eeeeean cee cteoct panproreeeeeneceeeects 69 INTERNAL OPTIONS irisse ieie 70 INTERNAL OPTIONSGENERAL EE 70 INTERNAL OPTIONS FEATURE CODES acters gees ec cee enero eee cheese 71 INTERNAL OPTIONS RTP SETTINGS cecsssssesssesssesssessnecsneesseesnssseescsneesneesneesnseaseseeseeeneeeneeeneee 73 INTERNAL OPTIONS HARDWARE CONEIOG 73 INTERNAL OPTIONS STUN MONITOR 75 SC a RU Berner nero errr re erent tere rnin ener een er enrerrreer ore 76 BEER 76 AN SETTINGSGENER AL ENEE 76 JAX SE MH CRS 6 0 ere 76 YES a MBI EE EE SS IAX SETTINGS REGISTRATION ENEE 77 LAN SETTINGSSECURITN nten n nnen ennnen enean n a 78 SIE ETTING erer E E E EEE ee es 78 SIP SETTINGS GENERAL ge 78 EEN 79 SIP SETTINGS JITTER BUEFER ENEE 79 SIP SET TINGS MISCELLANEOUS caeca
51. csxzedsesacevaeesoactarececesehagetansesabesinctacxsstendsttenttunedtesaeermeelacen 80 GENEE 80 SIP SETTINGS TLS AND TOP ucraniana in EE 81 TEN ET U 81 EE 83 EE 84 Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 3 of 108 E istan innovative IP Voice amp Video STATUS AND REPORTING ccssccsesssssssnssscesecsnseenssenenensensssennsnsenssensnes 85 PBX ENKE 85 ee 85 e 86 OOL UE eeh 88 CONFEAENGE ROOM Eege EEN 89 EE 89 PARKING LO EE 91 AHA EE RE 91 eege 92 e 92 EE 93 RESOURCE USAGE eege 93 ODR Call De ai ROO ea E EEEE E EEEE 94 UPGRADING AND MAINTENANCE snssnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnne 98 Rest Go cast teeecceceau ues escusieuessedeasestoesuecnedesnessooseessieosuves esshenolted ssaceosiqcossedeeoieecsaassecsstecdecesenaiteves 98 UPGRADING VIA NETWORK cccssssssseecccccecsseeeseeceecssseeseeccesesassessecesessaaeeeseeeesssasaneeeecesseasaeneeeees 98 UPGRADING VIALOCAL UPLOAD E 99 NO LOCAL FIRMWARE EE 100 Se RTE 100 EOC AA EEN 100 NE TWORR EE 101 CLEANER e E eee ee 102 RE E IR TR WE 103 aii O CEE E E E nee ee eer ee ee ee ee ee ee 104 PRO leien te 104 Be ANE EE 104 Seet 105 TIA ee 106 EXPERIENCING THE UCM6102 UCM6104 UCM6108 UCM6116 107 Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 4 of 108 E istan innovative IP Voice amp Video Table of Tables UCM6102 UCM6104 UCM6108 UCM6116 User Manual Table 1 TECHNICAL SPECIFICATIONS
52. deo Technology SIP AX Check SIP if the users are using SIP or a SIP device Check IAX if the users are using IAX or a IAX device NAT Call Reinvite DTMF Mode Insecure Enable Keep alive Keep alive Frequency 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 By default the PBX will route the media steams from SIP endpoints through itself If enabled the PBX will attempt to negotiate with the endpoints to route the media stream directly It is not always possible for the PBX to negotiate endpoint to endpoint media routing The default setting is No Select DIMF mode for the user to send DIMF The default setting is RFC2833 If Info is selected SIP INFO message will be used If Inband is selected 64 kbit codec PCMU and PCMA are required When Auto is selected RFC2833 will be used if offered otherwise Inband will be used e Port Allow matching peers by IP address without matching port number e Invite Authentication of incoming INVITE messages is not required e No Normal IP based matching and authenticated INVITES The default setting is Port If enabled keep the NAT session open The default setting is enabled Configure the number of seconds for the host to be up for Keep alive Max Call Numbers Require Call Token
53. ds of incoming registration and subscription allowed by the PBX The default setting is 3600 Configure the minimum length of time in seconds of incoming registration and subscription allowed by the PBX The default setting is 60 Configure the Music On Hold class for the channel when being put on hold This is used when the Music On Hold class is not set on the channel in the dialplan and the peer channel placing the call on hold doesn t suggest the Music On Hold class neither The default Music On Hold class will be used if not specified Configure the Music On Hold class to suggest to the peer channel when placing the peer on hold It can be specified globally or per user per peer The default Music On Hold class will be used if not specified Configure to relax the DTMF handling The default setting is disabled Select DTMF mode to send DTMF The default setting is RFC2833 H Info is selected SIP INFO message will be used If Inband is selected 64 kbit codec PCMU and PCMA are required When Auto is selected RFC2833 will be used if offered otherwise Inband will be used If there is no RTP activity in the timeout in seconds configured in this option the call will be terminated The default setting is no timeout This UCM6102 6104 6108 6116 USER MANUAL Page 83 of 108 RTP Hold Timeout Trust Remote Party ID Send Remote Party ID Generate In Band Ringing Server User Agent Allow Nonlocal Redirect Add user phone
54. e with password will skip the authentication for the invited users Please be cautious to use this option The default setting is disabled UCM6102 6104 6108 6116 USER MANUAL Page 54 of 108 CONFIGURING IVR Gos yam innovative IVR IVR configurations can be accessed under Web GUI gt PBX gt Call Features gt IVR Users could create edit view and delete IVR e Click on Create New IVR to add a new IVR e Click on e Click on Name Extension Dial Other Extensions Dial Trunk Permission Welcome Prompt Timeout Timeout Prompt Invalid Prompt Timeout Repeat Loops Invalid Repeat Loops Key Press Event Firmware Version 1 0 0 32 to edit the IVR configuration to delete the IVR Table 21 IVR Configuration Parameters Configure the name of the IVR Letters digits underscore and hyphen are allowed Enter the extension number for users to access the IVR If enabled callers to the IVR can dial extensions other than the ones explicitly defined in the IVR The default setting is disabled If enabled users can use trunk with configured permission Select permission for users to use trunk for outgoing calls Select a audio file to play as the welcome prompt Click on Prompt to add additional audio file After playing the prompts in the IVR the PBX will wait a period of time to detect DTMF entry This period of time is the timeout in seconds The default setting is 10 seconds Select th
55. e Agent Login Settings f wi f TechSupport1 Linear Warehouse Ringall Sales Ringall TechSupport2 Least Recent Figure 31 Call Queue e Click on Create New Queue to add call queue e Clickon to edit the call queue e Clickon l to delete the call queue e Click on Agent Login Settings to configure Agent Login Extension Postfix and Agent Logout Extension Postfix For example if the call queue extension is 6500 Agent Login Extension Positix is and Agent Logout Extension Postfix is users could dial 6500 to login and dial 6500 to logout CALL QUEUE PARAMETERS Table 26 Call Queue Parameters Extension Configure the call queue extension Name Configure the name to identify the call queue Strategy Select the strategy for the call queue e RingAll Ring all available Agents simultaneously until one answers e Linear Ring agents in the specified order specified e LeastRecent Ring the agent being least recent called e FewesitCalls Ring the agent with the fewest completed calls e Random Ring a random agent e RRmemory Ring the agents in Round Robin scheduling with memory Music On Hold Select the Music On Hold class for the call queue Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 66 of 108 Leave When Empty John Empty Dynamic Login Password TimeOut Wrapup Time Max Len Report Hold Time Wait Time Static Agents Firmware Version 1 0 0 32 innovative Gos
56. e details in the following table Table 33 System Status gt General Status gt System Status gt 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 Base Base version Program Program version This is the main software release version Recovery Recovery version NETWORK Under Web GUI gt Status gt System Status gt Network users could check the network information for the UCM6102 UCM6104 UCM6108 UCM6116 Please see details in the following table Table 34 System Status gt Network Status gt System Status gt Network MAC Address Global unique ID of device in HEX format The MAC address can be found on the label coming with original box and on the label located on the bottom of the device IP Address IP address Gateway Default gateway address Subnet Mask Subnet mask address DNS DNS Server address Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 92 of 108 andstream innovative IP Voice amp Video STORAGE USAGE Users could access the storage usage information from Web GUI gt Status gt System Status gt Storage Usage It shows the available and used space for the following partitions e Configuration partition Asterisk server configuration files and service configuration files e Data partition
57. e prompt message to be played when timeout occurs Select the prompt message to be played when an invalid extension is pressed Select the repeat time if no DIMF input After the repeat time go to the timeout destination if configured otherwise hangup The default setting Is 4 Select the repeat time if the input is invalid After the repeat time go to the timeout destination if configured otherwise hangup The default setting is 4 Select the event for each key pressing for 0 9 UCM6102 6104 6108 6116 USER MANUAL Page 55 of 108 andstream innovative IP Voice amp Video 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 gt PBX gt Internal Options gt IVR Prompt page directly Create New IVR i Name Main Men i Extension Dial Other Extensions Dial Trunk Permission Internal Welcome Prompt None Timeout 10 Figure 22 Click On Prompt To Create IVR Prompt RECORD NEW IVR PROMPT In Web GUI gt PBX gt Internal Options gt IVR Prompt page click on Record New IVR Prompt and follow the steps below to record new IVR prompt Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 56 of 108 andstream i
58. e right list and add them to the left list e Click Save the finish the configuration Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 61 of 108 ndstream innovative IP Voice amp Video 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 gt PBX gt Call Features gt Ring Group Create New Ring Group techsupport 6005 6006 6007 Figure 27 Ring Group e Click on Create New Ring Group to add ring group e Click on to edit the ring group e Clickon to delete the ring group RING GROUP PARAMETERS Table 24 Ring Group Parameters Ring Group Name Configure ring group name to identify the ring group Letters digits underscore and hyphen are allowed Extension Configure the ring group extension Ring Group Members Select available users to the ring group member list Click on CO Gi Gei to arrange the order Ring Strategy Select the ring strategy e Ring All Simultaneously Ring all the members at the same time when there is incoming call to the ring group extension If any of the member answers the call it will stop ringing e Ring In Order Ring the members with the configured order one by one If the first member doesn t answer the call it will stop ringing the first member and start ringing t
59. ean Note Music On Hold classes can be managed from PBX gt Internal Options gt Music On Hold Configure whether forcing the caller to leave if the call queue has no agent e Yes Callers will be forced to leave the call queue if the call queue is empty e No Never force the callers to leave the call queue when the queue is empty e Strict Callers will be forced to leave the call queue if the agents are paused invalid or unavailable This is the default setting Configure whether the callers can join the call queue if the queue has no agent The default setting is No e Yes Callers can always join a call queue e No Callers cannot join the queue if the queue has no agent e Strict Callers cannot join a queue if the agents are paused in an invalid state or unavailable If enabled the configured PIN number is required for the agent to login The default setting is disabled Configure the number of seconds an agent will ring before the call goes to the next agent The default setting is 15 seconds How many seconds after the completion of a call an Agent will have before the Queue can ring them with a new call 0 means no delay How many calls can be queued at once This count does not include calls that have been connected with Agents it only includes calls that have not yet been connected Default is 0 which is no limit When the limit has been reached a caller will hear a busy tone and advance to the next calling
60. er side only Callee Enable Enable the feature code on callee side only Default code 0 Enter the code during active call to disconnect the call 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 Default code 72 Enter the code during active call to park the call 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 Default code 1 Enter the code to record the audio call Options Neither Disable the feature code UCM6102 6104 6108 6116 USER MANUAL Page 71 of 108 Audio Mix Record Do Not Disturb DND Active Do Not Disturb DND Deactive Call Forward Busy Active Call Forward Busy Deactive Call Forward No Answer Active Call Forward No Answer Deactive Call Forward Uncondition Active Call Forward Uncondiion Deactive Feature Digit Timeout Extension to Dial to Park a Call Extensions for Parked Calls Parked Call Timeout in secs Dial Voice Mail Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Gos ean innovative Caller Enable Enable the feature code on caller side only Callee Enable Enable the feature code on callee side only Default code 3 Enter the code to record the audio call and the PBX will mix the streams natively
61. et 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 BATCH ADD EXTENSIONS Under Web GUI gt PBX gt Basic Call Routes gt Extensions click on Batch Add Extensions to start adding extensions in batch Table 15 Batch Add Extension Parameters Start Extension The starting extension number Create Number The number of extensions to be added Permission Select permission for the user The available permissions are Internal Local National and International The default permission is Internal Enable Voicemail Enable Voicemail for the user The default setting is enabled SIP IAX Password Configure the SIP IAX password for the users e User Random Password A random secure password will be automatically generated It is recommended to use this password for security purpose e Use Extension as Password e Enter a password to be used Voicemail Password Configure Voicemail password digits only for the users e User Random Password A random password in digits will be automatically generated It is recommended to use this password for security purpose e Use Extension as Password e Enter a password to be used Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 42 of 108 ndstream innovative IP Voice amp Vi
62. f analog FXO port Range 13 5 dB to 12 0 dB 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 Enabling this option enabled Caller ID detection This options allows one to define the start of a Caller ID signal Ring to start when a ring is received or Polarity to start when a polarity reversal is started This is the periodic time in milliseconds that the PBX will use to check on a voltage drop in the line Default setting is 200ms This option defines the type of Caller ID signalling to use bell bell202 as used in the United States v23 as used in the UK v23_jp as used in Japan or dtmf as used in Denmark Sweden and Holland Select country for tone settings You can also select Custom and set the values manually Syntax f1 val level f2 val level c on1 off1 on2 off2 on3 off3 Frequencies are in Hz and cadence on and off are in ms Frequencies Range 0 4000 Busy Level Range 300 0 Cadence Range 0 16383 Select Tone Country Custom to edit manually Default value f1 480 f2 620 c 250 250 Syntax f1 val level f2 val level c on1 off1 on2 off2 on3 off3 Frequencies are in Hz and cadence on and off are in ms Frequencies Range 0 4000 Busy Level Range 300 0 Cadence Range
63. for the Firewall settings e Interface Select the interface LAN WAN For firewall settings Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 25 of 108 Create New Rule andstream innovative IP Voice amp Video 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 Click on Create New Rule button and a new window will pop up to specify rule options Rule Name Action Protocol Type Service Create new firewall rule Rule Name Action Protocol Type Service advance Figure 11 Create New Firewall Rule Table 9 Firewall Rule Settings Specify the Firewall rule name Select the action for the Firewall to perform e ACCEPT e REJECT e DROP Select the protocol for the traffic e TCP e UDP e Both Select the traffic type e IN If selected users will need specify the interface for the incoming packets e OUT Select the service type e FTP e SSH e Telnet Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 26 of 108 E istan innovative IP Voice amp Video e TFTP e HTTP e LDAP Check the box to display advanced options e Source Advance Enter the source IP address and port e Destination Enter the destination IP address and port Click on Apply button
64. g submitting your Email to other SMTP server The default setting is Yes Select Email type e MTA Mail Transfer Agent The Email will be sent from the configured domain When MTA is selected there is no need to set up SMTP server for it or no user login is required However the Emails sent from MTA might be considered as spam by the target SMTP server e Client Submit Emails to the SMTP server A SMTP server is required and users need login with correct credentials Specify the domain name to be used in the Email The current system time on UCM6102 UCM6104 UCM6108 UCM6116 can be checked under Web GUI gt Status gt System Status To change the time settings go to Web GUI gt Settings gt Time Settings NTP Server Enable DHCP Option 2 Firmware Version 1 0 0 32 Table 12 Time Settings Specify the URL or IP address of the NIP server for the PBX to synchronize the date and time The default NIP server is ntp ipvideotalk com If set to Yes the device is allowed to get provisioned for Time Zone UCM6102 6104 6108 6116 USER MANUAL Page 31 of 108 Enable DHCP Option 42 Time Zone Self Defined Time Zone Firmware Version 1 0 0 32 Gein innovative from DHCP Option 2 in the local server automatically The default setting is Yes 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 configu
65. ge 24 of 108 tream innovative IP Voice amp Video e EAP TLS e 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 Select 802 1X client certificate from local PC and then upload Certificate Settings gt Network Settings gt 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 LAN Port Specify the LAN port number Up to 8 ports can be configured Protocol Style Select protocol type for the forwarding in the selected port NETWORK SECURITY SETTINGS The UCM6102 UCM6104 UCM6108 UCM6116 provides users Firewall configurations to prevent certain malicious attack to the device system allow restrict or reject specific traffic through the device for security and bandwidth purpose Go to Web GUI gt Settings gt Network Settings gt Security page users will see the current service information with Port Process and Type as well as Firewall settings asterisk tepIPy4 slapd top lPy4 dropbear toplPw4 lighttpd top lPw4 udpiiPw4 opentftpd udp lPy4 udpliPw4 asterisk ud plPy zero_config udp Pw udp iPw4 asterisk ud plPwe asterisk udpilv zero_config udp iPw4 syslogd ud plPwd Figure 10 Current Service Users could configure the following options
66. geeee ob cee E E cb acee beseed soasetancocecoaes 98 BOGAN I Tel 99 Beie Ree NEE 101 Network BACKUD ER 102 OC ANON eege 103 Reset and FRC ee EE 104 SEENEN eet 105 A E E E E A EEE EE P EE 105 NA OONO e E E E EE 106 UCM6102 6104 6108 6116 USER MANUAL Page 8 of 108 E istan innovative IP Voice amp Video 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 e This ts the initial version Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 9 of 108 E istan innovative IP Voice amp Video WELCOME Thank you for purchasing Grandstream UCM6102 UCM6104 UCM6108 UCM61 16 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 Incorporating industry leading features and performance the UCM6102 UCM6104 UCM6108 UCM6116 offers quick setup deployment with ease and unrivaled reliability all at an unprecedented price point A Caution Changes or modifications t
67. haracters e Il Wildcard Match zero or more characters immediately Example 12345 9 Any digit from 1 to 9 Privilege Level Select privilege level for the outbound rule e Local The lowest level required All users can use this rule e National Users with National level or International level are allowed to use this rule e International The highest level required Only users with international level can use this rule Pin Set Configure the password for users to use this rule Send This Call Trough Trunk Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 49 of 108 E istan 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 Prepend Specify the digits to be prepended before the call is placed via the trunk Those digits will be prepended after the dialing number is stripped Use Failover Trunk Failover Trunk Failover trunks can be used to make sure that a call goes through an alternate route when the primary trunk is busy or down If Use Failover Trunk is enabled and Failover trunk is defined the calls that cannot be placed v
68. he second member Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 62 of 108 Seconds to Ring Each Member Enable Voicemail Secret Email Address Ring Group Name Extension ndstream innovative IP Voice amp Video 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 If enabled the ring group extension can use voicemail Configure the password to access the ring group voicemail Configure the Email address of the ring group extension techsupport 6400 Ring Group Members Avaliable Users 6005 John Doe 6006 Alex Chan 6007 Emily Green Ring Group Options Ring Strategy 6000 EXTe000 6001 Amyso01 6002 Amyso02 6003 Amys003 6004 bm Z sie sw g MI BR te SF So Ring in Order e i Seconds to Ring Each Member 30 i Enable Voicemail Secret Email Address Firmware Version 1 0 0 32 640038985 techsuppon imycompany com Figure 28 Ring Group Configuration UCM6102 6104 6108 6116 USER MANUAL Page 63 of 108 PAGING AND INTERCOM GROUP andstream innovative IP Voice amp Video Paging and intercom can be configured in group level under Web GUI gt PBX gt Call Features gt Paging Intercom e Click on Create New Page Intercom Group to add page intercom group Extension 6000 Type 2 Way Interc
69. heck if the device is still alive The default setting is disabled Enable FAX Detect Enable both CNG and T 38 detect The default setting is disabled Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 47 of 108 andstream innovative IP Voice amp Video SRTP Enable SRTP for the VoIP trunk The default setting is disabled Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 48 of 108 stream innovative IP Voice amp Video 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 gt PBX gt Basic Call Routes gt Outbound Routes to add and edit outbound rules e Click on Create New Outbound Rule to add a new outbound route e Clickon Io edit the outbound route e Clickon to delete the outbound route e Click on O to move the outbound route up down to arrange the sequence Table 18 Outbound Route Configuration Parameters Calling Rule Name Configure the calling rule name e g local Pattern e All patterns are prefixed with the _ e X Any Digit from 0 9 e Z Any Digit from 1 9 e N Any Digit from 2 9 e Wildcard Match one or more c
70. ia the regular trunk may have a secondary trunk to go through Example The user s primary trunk is a VoIP trunk and the user would like to use the PSTN when the VolP trunk is not available The PSTN trunk can be configured as the failover trunk of the VoIP trunk 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 Prepend Specify the digits to be prepended before the call is placed via the trunk Those digits will be prepended after the dialing number is stripped INBOUND ROUTES Inbound routes can be configured via Web GUI gt PBX gt Basic Call Routes gt Inbound Routes e Click on Create New Inbound Rule to add a new inbound route e Click on DID Features to configure DID features for the inbound route e Clickon to edit the inbound route Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 50 of 108 andstream innovative IP Voice amp Video e Click on ri to delete the inbound route Trunks DID Pattern Privilege Level Default Destination Table 19 Inbound Route Configuration Parameters Select the trunk to configure the inbound route e All patterns are prefixed with the
71. irmware Version 1 0 0 32 Auto Goan innovative I based on agent skills availability busy level in queue announcement Up to 5 layers of IVR Interactive Voice Response e UCM6102 Up to 30 simultaneous calls e UCM6104 Up to 45 simultaneous calls e UCM6108 UCM6116 Up to 60 simultaneous calls e UCM6102 UCM6104 Up to 3 password protected conference bridges allowing up to 25 simultaneous PSTN or IP participants e UCM6108 UCM6116 Up to 6 password protected conference bridges allowing up to 32 simultaneous PSTN or IP participants Call park call forward call transfer DND ring hunt group paging intercom and etc e FCC Part 15 CFR 47 Class B Part 68 e CE EN55022 Class B EN55024 EN61000 3 2 EN61000 3 3 EN60950 1 TBR21 RoHS e TICK AS NZS CISPR 22 Class B AS NZS CISPR 24 AS NZS 60950 AS ACIF S002 e ITU T K 21 Basic Level UL 60950 power adapter UCM6102 6104 6108 6116 USER MANUAL Page 13 of 108 E istan innovative IP Voice amp Video INSTALLATION This section describes detailed information on installation connection and warranty policy of the UCM6102 UCM6104 UCM6108 UCM6116 EQUIPMENT PACKAGING Table 2 UCM6102 UCM6104 EQUIPMENT PACKAGING Main Case Yes 1 Power Adaptor Yes 1 Ethernet Cable Yes 1 Quick Installation Guide Yes 1 Table 3 UCM6108 UCM6116 EQUIPMENT PACKAGING Main Case Yes 1 Power Adaptor Yes 1 Ethernet Cable Yes 1 Quick Installation Guide Yes 1 Wall Mount Yes
72. l be detected during talking and the detected Fax will be received and sent to user s Email If user s Email address is not configured Fax will be sent to the default Email address in Table 27 FAX T 38 Settings If the default Email address is empty Fax will not be received Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 69 of 108 ndstream innovative IP Voice amp Video INTERNAL OPTIONS The configuration for PBX internal options can be accessed via Web GUI gt PBX gt Internal Options INTERNAL OPTIONS GENERAL General Preferences Global OutBound CID Global OutBound CID Name Operator Extension Ring Timeout This is the default global CallerlD that is used for all outgoing calls when no other CallerlD is defined If ther User tab or VoIP Trunks tab does not have defined CallerID neither this Global OutBound CID will be used for Caller This is the global Caller Name that is used for all outgoing calls If this value is defined all outgoing calls will have a Callerld Name set to this value Usually this value could be your company name Leave this value blank if you would like to have the users CalleriD Name display on outbound calls The operator extension is the number dialed when users press 0 to exit Voicemail It s also available in IVR option Number of seconds to ring an extension before sending to the user s voicemail box Extension Preferences Enable Rand Password Disable
73. ll Report Entry With Audio Recording File Click on to olay the recording file click on gt to download the recording file in wav format click on U to delete the recording file the call record entry will not be deleted CDR Statistics is an additional feature on the UCM6102 UCM6104 UCM6108 UCM6116 which provides users a visual overview of the call report across the time frame Users can filter with different criteria to generate the statistics chart Inbound calls Outbound calls Internal calls All calls External calls Mar Apr May Jun Jul Figure 45 CDR Statistics Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 96 of 108 Deg innova tive IP Voice Table 36 CDR Statistics Filter Criteria Trunk Type Select one of the following trunk type All SIP Calls PSTN Calls Call Type Select one or more in the following checkboxes Time Range Firmware Version 1 0 0 32 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 UCM6102 6104 6108 6116 USER MANUAL Page 97 of 108 E istan innovative IP Voice amp Video UPGRADING AND MAINTENANCE UPGRADING UPGRADING VIA NETWORK The UCM6102 UCM6104 UCM6108 UCM6116 can be upgraded via TFTP HTTP HT TPS by configuring the URL IP Add
74. ls 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 Ring Group Page Intercom Group UCM6102 6104 6108 6116 USER MANUAL Page 52 of 108 E istan innovative IP Voice amp Video CONFERENCE BRIDGE Conference bridge configurations can be accessed under Web GUI gt PBX gt Call Features gt Conference Users could create edit view and delete conference bridges The conference room status and activity will show in the page as well e Click on Create New Conference Room to add a new conference bridge e Click on to edit the conference room e Click on to invite a user to the conference The user will receive the ring to join the conference e Click on to kick a participant in the conference This will hang up the conference call on the user e Click on to lock the conference room e Click on to delete the conference bridge Table 20 Conference Bridge Configuration Parameters Extension Configure the conference number for the users to dial and join the conference Password When configured the users who would like to join the conference call must enter this password before accessing the conference room Admin Password Configure the password to join the conference roo
75. m as administrator Enable Caller Menu When enabled conference participant could press the key to access the conference bridge menu The default setting is disabled Record conference When enabled the calls in this conference room will be recorded in a wav format file The default setting is disabled Quite Mode When enabled if there are users entering or leaving the conference room voice prompt or notification tone won t be played The default setting is disabled Wait For Admin When enabled the participants will not hear each other until the admin joins the conference room The default setting is disabled Enable User Invite When enabled users could press 0 to invite other users to join the conference The default setting is disabled Note Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 53 of 108 Announce Callers Play Hold Music For First Caller Skip Authentication When Invite User via Trunk from Web GUI Firmware Version 1 0 0 32 E istan innovative IP Voice amp Video Admin can always press 0 to invite other users to the conference When enabled announcement will be made to all conference participants when there is user joining in the conference The default setting is disabled When enabled the PBX will play Hold music to the first participant until another user joins the conference room The default setting is disabled When enabled the invitation from Web GUI for the conference bridg
76. meter scccswscotnssiencaecassachessnacedsanddavanseruneeiiaponnsaedeinieiagsateeaxnecoinaedeavansctants 55 Table 22 Email Settings For VOICEMAIIS 0224 lt 6 0 lt c0c0scccssecnceeeccesnaeseaeecesecateedenseseseeeecsceecesesnesdeeceseeeeeceesestes 59 Table 23 Voicemal SENGS E 59 Table 24 RING Group Parametere 62 Table 25 Page Intercom Group Harametere 64 Table 26 Call QUuGUE e nn 66 Table 27 FAX L38 te 69 TOGO FUN E 85 PAD er Os Ete FISION AS EEN 87 TOE O AJEN a on ee A E ee 88 Table 31 Interface Status ue e TEEN 90 Tape 2 Parking EE 91 Table 33 System Giatus General 92 Table EE E ET nd 92 Table 35 GDR Fiter Criteria xcsicecccsceancvsaccsisicadeience aesdarvacdeatanecasseedesdaeecvbacestsinadeteadd cpesiealendeadaeecatesedesdsneeenewss 95 Table 36 CDR Statistics Filter Crtenza ccc ccccccccccceececeeececeeceeseeceseeeeseeeeseeeeeseuceseueeeseeesseeetaneesseeeesees 97 Table 37 Network Upgrade Configuration cccccccccceeccccessceeceeeceeceeeceeeeeeeeeseaeceseeecessugeeeeeueeesaaeeeesaaeees 98 Table 38 Network Backup Configuration cccccccccccceseceeesceeceeeceeceeececeeeeceeseeeeesaeecessuseeeesaeeesaueeeesanes 102 Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 5 of 108 Table 39 Cleaner Configuration Firmware Version 1 0 0 32 andstream innovative IP Voice amp Video UCM6102 6104 6108 6116 USER MANUAL Page 6 of 108 E istan Table of Figures UCM6102 UCM6104 UCM6108 UCM61
77. n the bottom of the page CLEANER Users could configure to clean the Call Detail Report Voice Records Voice Mails FAX automatically under Web GUI gt Maintenance gt Cleaner Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 102 of 108 E istan innovative IP Voice amp Video CDR Cleaner i Enable CDR Cleaner Wi CO CDR Clean Time 1 CO Clean Interval 28 Voice Records Cleaner i Enable VR Cleaner Wl CO VR Clean Threshold 70 CO VR Clean Time 2 D VR Clean Interval 28 a r 7h my fwa mI gend Ge Save 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 VR Clean Time Enter 0 23 to specify the hour of the day to clean up Voice Records Clean Interval Enter 1 30 to specify the day of the month to clean up Voice Records All the cleaner logs will be listed on the bottom of the page RESET AND REBOOT Users could perform reset and reboot under Web GUI gt Maintenance gt Reset and Reboot To factory reset the device select the mode type first There are three different types for reset Firmware Version 1 0 0 32
78. ng P ke The agent has been logged out On the UCM6102 UCM6104 UCM6108 UCM6116 Service Level is defined as the percentage of high quality calls over all calls in the call queue where high quality call means calls answered within 10 seconds Other operations are also available in queue status section Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 88 of 108 tream e Click on Queues the web page will redirect to call queue configuration page which can also be accessed via web GUI gt PBX gt Call Features gt Call Queue e Click on ZC to refresh the call queue status e Click on to expand the call queue detail e Click on to hide the call queue detail 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 Conference Rooms gt 6300 3 Users En A 6000 0 37 LI 6005 0 36 Li 6007 0 16 6301 Notin Use Figure 37 Conference Room Status Other operations are also available in conference room status section e Click on Conference Rooms the web page will redirect to conference room configuration page which can also be accessed via web GUI gt PBX gt Call Features gt Conference e Click on CA to refresh the conference room status e Click on to expand the conference room details e Click on to hide the conference r
79. nnovative IP Voice amp Video Record New IVR prompt File Name Welcome Prompt 1 Format WAM Le Dial This User Extension to 6000 EB Record a New Voice Prompt Figure 23 Record New IVR Prompt e Specify the IVR file name e Select the format GSM or WAV for the IVR file e Select the extension which will be dialed for the user to start recording the voice prompt e Click the Record button A request will be sent to the PBX and the PBX will then call the extension for recording e Pick up the call from the extension and start the recording UPLOAD IVR PROMPT In Web GUI gt PBX gt Internal Options gt IVR Prompt page click on Upload IVR Prompt and choose a file to upload The requirement on the audio file is as follows e PCM encoded e 16 bits e 8000HZ mono e In mp3 or wav format or raw ulaw alaw gsm file with ulaw or alaw suffix e File size smaller than 5M pere Click on F7 to select audio file from local PC and click on to start uploading Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 57 of 108 ndstream innovative IP Voice amp Video VOICE PROMPT Language settings for voice prompt can be accessed under Web GUI gt PBX gt Internal Options gt Language Users could upload a voice prompt package and then select the language in the list with available language options Upload Voice Prompts Package iama i Choose Voice Prompt to ms Upload Upload Voice
80. nostic run Package capturing Done Click on Download to download the captured packages 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 CO Target Host www google com Output Result 64 bytes trom 4 125 224 7 9 seg ttl 53 time 13_500 ms 64 bytes from 774 125 224 179 seq 2 ttl 53 time 19 300 ms 64 bytes from 774 125 224 179 seg 3 ttl 53 time 13 600 ms 64 bytes from 774 125 224 179 seq 4 ttl 55 time 13 625 ms 64 bytes from 774 125 224 179 seg 5 ttl 53 time 13 950 ms 64 bytes from 774 125 224 179 seqg 6 ttl 53 time 14 125 ms 64 bytes from 74 125 224 179 seq ttl 53 time 1 425 ms 64 bytes from 74 125 224 179 seg s ttl 53 time 13 d00 ms 64 bytes from 774 125 224 179 seg 9 ttl 53 time 13 875 ms 64 bytes from 774 125 224 179 seq 10 ttl 53 time 14 100 ms 64 bytes from 74 125 224 179 seq 11 ttl 53 time 14 175 ms 64 bytes from 74 125 224 179 seqg 12 ttl 53 time 14 025 ms 64 bytes from 74 125 224 179 seg 13 ttl 53 time 14 150 ms 64 bytes from 774 125 224 179 seq 14 ttl 53 time 13 900 ms www google com ping statistics 15 packets transmitted 15 packets received
81. o EXTENSIONS To manually create new user go to Web GUI gt PBX gt Basic Call Routes gt 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 Extension CallerlD Name CalleriID Number Permission SIP IAX Password Enable Voicemail Voicemail Password Email Address Call Forward Unconditional Call Forward No Answer Call Forward Busy Ring Timeout The extension number associated with the user Configure the Caller Name associated with the user Number letter or space are allowed Configure the CallerID Number that would be applied for outbound calls from this user Note The ability to manipulate your outbound Caller ID may be limited by your VoIP provider Select permission for the user The available permissions are Internal Local National and International The default permission is Internal Configure the password for the user A random secure password will be automatically generated It is recommended to use this password for security purpose Enable Voicemail for the user The default setting is enabled Configure Voicemail password digits only A random numeric password is automatically generated Fill in the Email address for the user Configure the Call Forward Unconditional target number If not configured the Call Forward Unconditional feature is deactivated
82. o support Generate Manager Events Generates manager events when SIP UA performs events e g hold Reject NonMatching Invites When rejecting an incoming INVITE or REGISTER request always reject with 401 Unauthorized instead of notifying the requester that if there is a matching user or peer for the request NonStandard G 726 Support If the peer negotiates G726 32 audio use AAL2 packing order instead of RFC3551 packing order this is required for Sipura and Grandstream ATAs SIP SETTINGS SESSTION TIMER Session Timers e Originate always request and run session timers e Accept Run session timers only when requested by other UA e Refuse Do not run session timers The default setting is Accept Session Expires The maximum session refresh interval in seconds The default setting is 1800 Min SE The minimum session refresh interval in seconds The default setting is 90 Session Refresher Selects the session refresher to be UAC or UAS The default setting is UAC Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 80 of 108 SIP SETTINGS TLS AND TCP TCP Enable TCP Bindaddr TLS Enable TLS Bindaddr TLS Self Signed CA TLS Cert TLS CA Cert TLS CA List SIP SETTINGS NAT External Address Firmware Version 1 0 0 32 Gus yam innovative Enables disables server for incoming TCP connections The default setting is No IP address for TCP server to bind to 0 0 0 0 binds to all in
83. o this product not expressly approved by Grandstream or operation of this product in any way other than as detailed by this User Manual could void your manufacturer warranty A Warning Please do not use a different power adaptor with the UCM6102 UCM6104 UCM6108 UCM6116 as it may cause damage to the products and void the manufacturer warranty This document is subject to change without notice The latest electronic version of this user manual is available for download here http www grandstream com support Reproduction or transmittal of the entire or any part in any form or by any means electronic or print for any purpose without the express written permission of Grandstream Networks Inc is not permitted Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 10 of 108 ndstream innovative IP Voice amp Video PRODUCT OVERVIEW FEATURE HIGHTLIGHTS e 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 e Integrated 2 4 8 16 PSTN trunk FXO ports 2 analog telephone FXS ports and up to 50 SIP trunk options e Gigabit network port with integrated PoE USB SD integrated NAT router with advanced QoS support UCM6102 only e Supports a wide range of popular voice codes including G 711 A law U law G 722 G 723 G 726 G 729A B iLBC GSM video codec including H 264 H 263 H 263 and Fax T 38
84. om Le Page Intercom Group Members Avaliable Users SIPMS006 Alex Chan 8 SIPS000 EXT6000 SIPS00T Emily Green SIPM6001 Amye001 SIP 6005 John Doe SIP6002 Amyo002 l SIP 6003 Amy6003 SIP 6004 Amy PCPS eS SS MB E Se dw Sg Cancel 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 l l l Select available users from the right list to the left Members e Clickon toedit the page intercom group e Clickon I to delete the page intercom group e Click on Paging Intercom Group Settings to edit Alert Info Header see figure below To edit page intercom feature code click on Feature Codes in the following figure Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 64 of 108 stream innovative IP Voice amp Video Paging intercom Group Settings Settings for Paging amp Intercom CO Alert Info Header Intercom Settings For Paging Intercom Feature Code Please go to Feature Codes page for setting paging intercom feature code Figure 30 Page Intercom Group Settings Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 65 of 108 ndstream innovative IP Voice amp Video CALL QUEUE CONFIGURING CALL QUEUE Call queue settings can be accessed via Web GUI gt PBX gt Call Features gt Call Queue Create New Queu
85. on Configuration Parameters e Reboot the user Click on to send NOTIFY reboot event to the device with the extension registered e Delete single extension Click on to delete the extension e Modify selected extensions Select the checkbox for the extension s Then click on Modify Selected Extensions to edit the extensions in a batch The configuration options are listed in Table 15 Batch Add Extension Parameters e Delete selected extensions Select the checkbox for the extension s Then click on Delete Selected Extensions to delete the extension s Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 44 of 108 andstream innovative IP Voice amp Video TRUNKS ANALOG TRUNKS Go to Web GUI gt PBX gt Basic Call Routes gt Analog Trunks to add and edit analog trunks e Click on Create New Analog Trunk to add a new analog trunk e Clickon to edit the analog trunk e Click on to delete the analog trunk The analog trunk options are listed in the table below Table 16 Analog Trunk Configuration Parameters Channels Select the channel for the analog trunk e UCM6102 2 channels UCM6104 4 channels UCM6108 8 channels UCM6116 16 channels Trunk Name A unique label to identify the trunk when listed in outbound rules incoming rules and etc Advanced Options Busy Detection Busy Detection is used to detect far end hangup or for detecting busy signal Enable to turn this featu
86. oom details INTERFACES STATUS This section displays interface port connection status on the UCM6102 UCM6104 UCM6108 UCM61 16 The following example shows the interface status for UCM6116 with USB SD card LAN port FXS1 connected Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 89 of 108 andstream innovative IP Voice amp Video Interfaces Status gt SD Card Figure 38 UCM6116 Interfaces Status Table 31 Interface Status Indicators d USB connected Ke 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 Other operations are also available in interface status section e Click on Interfaces Status the web page will redirect to hardware configuration page which can also be accessed via web GUI gt PBX gt Internal Options gt Hardware Config e Click on CA to refresh the interface status e Click on to expand the interface details e Click on to hide the interface details Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 90 of 108 andstream innovative IP Voice amp Video 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 Parking Lot gt Caller
87. or analog phones attached to the FXS port is required The default setting is Normal Configure to increase the ringing speed to 25HZ This option can be used with Low Power option The default setting is Normal Configure the peak voltage during Fast Ringer operation This option is used with Fast Ringer option The default setting is Normal Configure ring detection If set to Full Wave false ring detection will be prevented for lines where Caller ID is sent before the first ring and proceeded by a polarity reversal as in UK The default setting is Standard Configure the type of Message Waiting Indicator detection on trunk FXO interfaces e None No detection e FSK Frequency Shift Key detection e NEON Neon MWI detection The default setting is None UCM6102 6104 6108 6116 USER MANUAL Page 74 of 108 E istan innovative IP Voice amp Video INTERNAL OPTIONS STUN MONITOR STUN Server STUN Refresh Firmware Version 1 0 0 32 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 Number of seconds between STUN Refreshes The default setting is 30 seconds UCM6102 6104 6108 6116 USER MANUAL Page 75 of 108 TAX SETTINGS E istan innovative IP Voice amp Video PBX SETTINGS The UCM6102 UCM6104 UCM6108 UCM6116 IAX Settings can be accessed via Web GUI gt PBX gt IAX
88. ource Format 6012 e Destination Format 6005 e Caller ID Format John Doe lt 6012 gt e Disposition Format NO ANSWER BUSY ANSWERED or FAILED Users could filter the call report by specifying the date range and criteria depending on how the users would like to include the logs to the report Then click on the View Report button to display the generated report Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 94 of 108 stream innovative IP Voice amp Video CO Inbound calls i Outbound calls CO Internal calls i External calls Source Destination stant Time E From F To Duration E From EI To Seconds View Report 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 External calls External calls are calls sent from one trunk to another trunk which are not sent to any internal extension The call report will display as the following figure shows 1 2013 03 08 01 37 03 0 00 02 1003 1002 1003 214 lt 1003 gt NO ANSWER 2 2013 03 08 01 37 11 0 00 03 2022 2003 2022 228 lt 2022 gt FAILED 3 2013 03 08 01 37 40 0 00 03 20
89. patterns e 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 Table 5 UCM6102 UCM6104 LED INDICATORS LED Status LAN WAN FXS FXO USB SD Card SH Solid Connected Flashing Data Transferring OFF Not Connected Table 6 UCM6108 UCM6116 LED INDICATORS LED LED Status NETWORK SH Solid Connected OFF Not Connected ACT Line FXO Phone FXS USB SD Card IB Solid Connected SS Flashing Data Transferring OFF Not Connected USING THE WEB GUI ACCESSING WEB GUI The UCM6102 UCM6104 UCM6108 UCM6116 embedded Web server responds to HTTP HTTPS GET POST requests Embedded HTML pages allow users to configure the device through a Web browser such as Microsoft s IE Mozilla Firefox Google Chrome and etc Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 20 of 108 tream innovative IP Voice amp Video Gain UCM6116 Hybrid PBX ES Password Copyright Grandstream Networks Inc 2013 All Rights Reserved Figure 9 UCM6116 Web GUI Login Page To access the Web GUI Connect the computer to the same network as the UCM6102 UCM6104 UCM6108 UCM6116 Ensure the device is properly powered up and shows its IP address on the LCD 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
90. port and LAN port with Router or Switch mode function configurable on the LAN port Select each tab in the Network Settings page to configure LAN settings WAN settings UCM6102 only and 802 1X Please refer to the following tables for the network configuration parameters on UCM6104 UCM6108 UCM6116 and UCM6102 respectively Table 7 NETWORK SETTINGS Settings gt Network Settings gt LAN IP Method Select DHCP Static IP or PPPoE The default setting is DHCP IP Address Enter the IP address for static IP settings Gateway IP Enter the gateway IP address for static IP settings Subnet Mask Enter the subnet mask address for static IP settings DNS Server 1 Enter the DNS server 1 address for static IP settings DNS Server 2 Enter the DNS server 2 address for static IP settings User Name Enter the user name to connect via PPPoE Password Enter the password to connect via PPPoE Preferred DNS Server Enter the preferred DNS server address Settings gt Network Settings gt 802 1X 802 1X Mode Select 802 1X mode The default setting is Disable The supported 802 1X mode are e EAP MD5 Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 23 of 108 ndstream innovative IP Voice amp Video e EAP TLS e 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
91. re on Busy Count If Busy Detection is enabled it is also possible to specify how many busy tones to wait for before hanging up The default is 4 but better results can be achieved if set to 6 or even 8 Mind that the higher the number the more time that will be needed to hangup a channel but lowers the probability that you will get random hangups Congestion Detection Congestion detection is used to detect far end congestion signal Enable to turn this feature on Congestion Count If congestion detection is enabled it is also possible to specify how many congestion tones to wait for The default setting is 2 Enable Polarity Reversal If this option is enabled the reception of a polarity reversal will mark when a outgoing call is answered by the remote party in some countries a polarity reversal is used to signal the disconnect of a phone line the call will be considered hung up on a polarity reversal Polarity On Answer Delay minimal time period ms between the answer polarity switch and Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 45 of 108 RX Gain TX Gain Ring Timeout Use CallerlD Caller ID Start Current Disconnect Threshold ms CID Signalling Tone Country Busy Tone Congestion Tone Firmware Version 1 0 0 32 Gos pam hangup polarity switch default 600ms Gain for the receive channel of analog FXO port Range 13 5 dB to 12 0 dB Gain for the transmit channel o
92. red on the PBX The default setting is Yes Select the proper time zone option so the PBX can display correct date and time accordingly If Automatic is selected the PBX will obtain the time zone information according to the detected IP location If Self Defined Time Zone is selected in Time Zone option users will need define their own time zone following the format below The syntax is std offset dst offset start time end time Default is set to MTZ 6MDT 5 M4 1 0 M11 1 0 MTZ 6MDT 5 This indicates a time zone with 6 hours offset with 1 hour ahead which is U S central time If it is positive if the local time zone is west of the Prime Meridian A K A International or Greenwich Meridian and negative if it is east M4 1 0 M11 1 0 The 1st number indicates Month 1 2 3 12 for Jan Feb Dec The 2nd number indicates the nth iteration of the weekday 1st Sunday 3rd Tuesday The 3rd number indicates weekday 0 1 2 6 for Sun Mon Tues Gan Therefore this example is the DST which starts from the First Sunday of April to the 1st Sunday of November UCM6102 6104 6108 6116 USER MANUAL Page 32 of 108 E istan innovative IP Voice amp Video 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
93. ress 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 gt Maintenance gt Upgrade Upgrade Firmware Network Upgrade CO Upgrade Via HTTP CO Firmware Server Path fw ipvideotalk com gs Firmware File Prefix Firmware File Suffix HTTP HTTPS User Name Ooo oO g HTTP HTTPS Password Figure 46 Network Upgrade Table 37 Network Upgrade Configuration Upgrade Via Allow users to choose the firmware upgrade method TFTP HTTP or HTTPS Firmware Server Path Define the server path for the firmware server Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 98 of 108 E istan innovative IP Voice amp Video 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 Then reboot the device to start the upgrading process UPGRADING VIA LOCAL UPLOAD lf there is no HITPAFTP
94. rsion 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 15 of 108 andstream innovative IP Voice amp Video SD Card Slot Reset Ground USB Port DC 12V 2xLANPort 2xFXSPort 4xFXO Port Figure 4 UCM6104 Back View To set up the UCM61 04 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 The LCD in the front will show its hardware information when the boot process is done 5 Once the UCM6104 is successfully connected to network the LED indicator for LAN 1 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 CONNECTING THE UCM6108 Naviation SD Card Slot 2xFXS Port 8xFXOPort LED Indicators LCD Keys Figure 5 UCM6108 Front View DC 12V Reset LAN Port Ground Figure 6 UCM6108 Back View Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 16 of 108 E istan innovative IP Voice amp Video 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 UCM6
95. rt External TLS Port Local Network Address NAT Mode Allow RTP Reinvite Firmware Version 1 0 0 32 Gus ean innovative UCM6102 UCM6104 UCM6108 UCM6116 is behind NAT If it s a hostname it will only be looked up only 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 Configures the refresh interval for the external host Configures the externally mapped TCP port when the UCM6102 UCM6104 UCM6108 UCM6116 is behind a static NAT or PAT Configures the externally mapped TLS port when UCM6102 UCM6104 UCM6108 UCM6116 is behind a static NAT or PAT The default value is 5061 A list of network addresses that are considered inside of the NAT network Multiple entries are allowed e g a reasonable set could be as follows 192 168 0 0 255 255 0 0 This is a global NAT setting that will affects all peers and users e No Use rport if the remote side requires it e Force rport Force rport to always be on This is the default setting e Yes Force rport to always be on and perform comedia RIP handling e Comedia Use rport if the remote side requires it and perform comedia RTP handling Note comedia RIP handling refers to the technique of sending RIP to the port where the other endpoint s RTP comes from This can also be rephrased as connection oriented media When turned on
96. rule after attempting to enter the queue If enabled the PBX will report to the agent the duration of time of the call before connected to the agent If enabled users will be disconnected after the configured number of seconds The default setting is disabled Note It is recommended to configure Wait Time longer than the Wrapup Time Select the available agents from the right list to the left Click on C3 Gi E to arrange the order UCM6102 6104 6108 6116 USER MANUAL Page 67 of 108 tream innovative IP Voice amp Video MUSIC ON HOLD Music On Hold settings can be accessed via Web GUI gt PBX gt Internal Options gt 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 Create New MOH class Music On Hold Classes default Le CH Delete Upload an amp KHz Mono Music file size less then 5M i Choose file to Upload mal OG Upload List of Sound Files macroform cold_day wavy i macroform robot_dity waw IP macroform the_simplicity waw i manolo_camp morning_cotfee waw D reno_project system waw i Figure 32 Music On Hold Default Class e Click on Create New MOH Class to add a new Music On Hold class e Click on to delete the selected Music On Hold class aa e Click on _ to select music file from local PC and click on to start uploading The music file uploaded has
97. s selected 64 kbit codec PCMU and PCMA are required When Auto is selected RFC2833 will be used if offered otherwise Inband will be used e Port Allow matching peers by IP address without matching port number e Invite Authentication of incoming INVITE messages is not required e No Normal IP based matching and authenticated INVITES The default setting is Port If enabled keep the NAT session open The default setting is enabled Configure the number of seconds for the host to be up for Keep alive Other Settings SRTP FAX Detect Strategy Firmware Version 1 0 0 32 Enable SRTP for the call Enable to detect fax signal from the user trunk during the call and send the received fax to the Email address configured in this configuration page If no Email address can be found for the user send the received fax to the default Email address in FAX setting page Note If enabled FAX cannot use Passthrough This option controls how the extension can be used on the device e Allow all Device in any network can register using the extension UCM6102 6104 6108 6116 USER MANUAL Page 41 of 108 E istan innovative IP Voice amp Video e Only local subnets Only the user in specific subnet can register using the extension Up to three subnet can be specified e 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 s
98. sages in total and 1 Status message that has been read already Displays extension type e SIP User Type e AX User e Analog User e Features Other operations are also available in extension status section Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 87 of 108 stream e Click on Extensions the web page will redirect to extension configuration page which can also be accessed via web GUI gt PBX gt Basic Call Routes gt Extensions e Click on CA to refresh the extension status e Click on one of the tabs All Analog Festus IAN SIP to display the corresponding extensions accordingly e Click on to expand the status detail table e Click on to hide the status detail table QUEUES Users could see all the configured call queue status in this section The following figure shows the call queue 6500 being in used near 4 j F m oo SS Ss Ss E g Mi 7 F f Wd ly Srn wT Enn i ry 4 4 ry 7 L E E me Kal em mm DU EXT6 Wi 0 44 Oe Ope a eT oe TaT a tar J J service Level SL 0 0 within Os Calls Completed 1 Calls Abandoned 1 Figure 36 Queue Status The current call status caller ID duration agent status service level calls summary completed abandoned are shown for the call queue The agent status is defined as below Table 30 Agent Status The agent is available idle The agent is talking busy The agent is ringi
99. ssage When the UCM6102 UCM6104 UCM6108 UCM6116 receives the SUBSCRIBE a SIP NOTIFY message will Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 33 of 108 andstream 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 UCM61 16 e 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 The UCM6102 receives it and returns DHCP OFFER with the config server path URL in Option 66 The phone will then use the path to download the config file generated in the UCM6102 UCM6104 UCM6108 UCM61 16 e mDNS The mDNS process is similar to the SIP SUBSCRIBE To start the auto provisioning process under Web GUI gt PBX gt Basic Call Routes gt Zero Config click on Auto Provision Setting and fill in the auto provision information Auto Provision Setting SERGE SEN E am ell m ze ge ll a zm ge ee oe ae a es evices boot it will send the subscribe broadcast the severwi Bel E Ba rupgamg gea ges S Ae ell ef eum ue Gl Int thenreturn it a url of contig tile Enable Zero Config Automatically Assign Extension Start Extension 6000 Generate Random Password Default Password admin Cancel
100. terfaces The default port number is 5060 if not specified Enables disables server for incoming TLS secure connections The default setting is No 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 Note The IP address must match the common name hostname in the certificate Please do not bind a TLS socket to multiple IP addresses For details on how to construct a certificate for SIP please refer to the following document http tools ietf org html draftt iett sip domain certs This is the CA certificate is the TLS server being connected to requires self signed certificate including server s public key This file will be renamed as asterisk ca automatically Note The size of your ca file can t be larger than 2MB This is the Certificate file pem format only used for TLS connections This file will be renamed as asterisk pem automatically Note The size of your certificate can t be larger than 2MB This file must be named with the CA subject name hash value It contains CA s Certificate Authority public key which is used to verify the accessed servers Note The size of your certificate can t be larger than 2MB The list of files under the CA Cert directory A static address and port that will be in outbound SIP messages if the UCM6102 6104 6108 6116 USER MANUAL Page 81 of 108 External Host External Refresh External TCP Po
101. the UCM6102 UCM6104 UCM6108 UCM6116 will try to redirect the RIP media stream audio to go directly from the caller to the callee e Yes Enables RTP Reinvite e NoNAT Allows media path redirection reinvite but only when the peer is not be behind NAT The RTP core can determine if the peer is behind NAT or not based on the IP address where the media comes from e Update use UPDATE for media path redirection instead of INVITE UCM6102 6104 6108 6116 USER MANUAL Page 82 of 108 SIP SETTINGS ToS Gos pam innovative Note Some devices do not support this especially if one of them is behind NAT The following options are provided to configure SIP ToS on the UCM6102 UCM6104 UCM6108 UCM61 16 ToS For Signaling Packets ToS For RTP Audio Packets ToS For RTP Video Packets Default Incoming Outgoing Registration Time Max Registration Subscription Time Min Time Registration Subscription Music On Hold Interpret Music On Hold Suggest Enable Relaxed DTMF DTMF Mode RTP Timeout Firmware Version 1 0 0 32 Configure the Type of Service for SIP packets The default setting is None Configure the Type of Service for RTP audio packets The default setting is None Configure the Type of Service for RTP video packets The default setting is None Configure the default length of time in seconds of incoming outgoing registration The default setting is 120 Configure the maximum length of time in secon
102. to URI Send Compact SIP Headers Time Between MWI Checks Min Roundtrip Time T1 Time SIP SETTINGS DEBUG Enable SIP Debugging Record SIP History Dump SIP History Subscribe Context Allow Subscribe Notify on Ringing Firmware Version 1 0 0 32 Gos yam setting doesn t apply to calls on hold 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 Configure whether the Remote Party ID should be trusted The default setting is disabled Configure whether the Remote Party ID should be sent The default setting is disabled Configure whether the PBX should generate inband ringing If Never is selected inband ringing will not be generated even the end point device is not working properly The default setting is Never Configure to replace the user agent string If enabled 302 or Redirect is allowed to non local SIP address The default setting is disabled If enabled user phone will be added to URI that contains a valid phone number The default setting is disabled If enabled compact SIP headers will be sent The default setting is disabled Configure the default time in seconds between Mailbox checks for peers The default setting is 10 Configure the minimum roundtrip time in milliseconds of the messages sent to the
103. to be 8 KHz Mono format with size less than 5M ASL e Clickon 1 to delete the sound file for the selected Music On Hold file Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 68 of 108 E istan innovative IP Voice amp Video 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 gt PBX gt Internal Options gt FAX T 38 CONFIGURING FAX T 38 e Click on Create New Fax Extension In the popped up window fill the extension name and Email address to send the received FAX to e Click on Settings to Fax to configure the following options Table 27 FAX T 38 Settings Enable Error Correction Mode Enable ECM for the Fax ECM Maximum Transfer Rate Configure the maximum transfer rate during the Fax rate negotiation The possible values are 2400 4800 7200 9600 12000 and 14400 The default setting is 14400 Minimum Transfer Rate Configure the minimum transfer rate during the Fax rate negotiation The possible values are 2400 4800 7200 9600 12000 and 14000 The default setting is 2400 Default Email Address Configure the Email address to send the received Fax to if user s Email address cannot be found e Click on to edit the Fax extension e Click on to delete the Fax extension Note Users could also use Fax2Mail if FaxDetect option is enabled for the user and VoIP trunk The Fax signal wil
104. tter Buffer Force Jitter Buffer Log Frames Max Jitter Buffer Resync Threshold Firmware Version 1 0 0 32 Enables disables the use of jitter buffer on the receiving side of a SIP channel Forces the use of jitter buffer on the receiving side of a SIP channel Enable disables jitter buffer frame logging Configures max length of the jitter buffer in milliseconds Jumps in the frame timestamps over where the jitter buffer is resynchronized This feature is useful to improve the quality of voice UCM6102 6104 6108 6116 USER MANUAL Page 79 of 108 Geese 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 jomaxsize 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 Register Timeout The interval in seconds for the UCM6102 UCM6104 UCM6108 UCM6116 to retry registration The default setting is 20 Register Attempts Number of registration attempts before the UCM6102 UCM6104 UCM6108 UCM6116 gives up The default setting is O keep trying until the server side accepts the registration request Video Max Bitrate kb s Maximum bitrate kb s for video calls The default setting is 384 Support for SIP Video Enables disables SIP vide
105. umentation 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 Our technical support staff is trained and ready to answer all of your questions Contact a technical support member or submit a trouble ticket online to receive in depth support Thank you again for purchasing Grandstream UCM6102 UCM6104 UCM6108 UCM6116 it will be sure to bring convenience and color to both your business and personal life Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 107 of 108
106. utes gt Analog Trunks e Click on CA to refresh the trunk status e Click on to expand the status detail table e Click on to hide the status detail table EXTENSIONS Users could see all the configured extension status in this section Firmware Version 1 0 0 32 UCM6102 6104 6108 6116 USER MANUAL Page 86 of 108 stream innovative IP Voice amp Video Extensions gt 6000 EXT6O00 Messages 0 0 0 SIF User 6001 Amy6b001 He 3 Doro SIP User 6002 AmybOo Mes es 0 0 0 SIP User 6003 Amy6b003 Mes jes O O 0 SIP User 6004 Amyboo4 Mes 5 O O 0 SIP User 6005 John Doe les s Drar SIP User 6006 Alex Chan Messages 0 1 0 SIP User DUU Emily Green Messages 0 0 0 SIP User 6008 IAX 6008 Mes 3 O O 0 AX User 6009 FXS 6009 essages O 0 0 Analog User FAS 1 H Voice Mail Main Features W Dial Voice Mail Features on Call Pickup Features KA Pageing Prefix Features DU Intercom Prefix Features Total 25 Show 4 2 Jumpto Prev Figure 35 Extension Status Table 29 Extension Status Displays extension number including feature code The color indicator has the following definitions e Green Free a Blue Ringing gt Yellow In Use Grey Unavailable Extension Displays name calleriD name or label feature code function for the Name Label extension Displays message status for the extension Example 2 4 1 Description There are 2 urgent messages 4 mes

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