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1. 43 Table 20 Extension Configuration Parameters cccccccssscceceeceeceseceeeeseeeeeeeeceseeeeeeaeseeeseaeeeeseaeeeesaaeees 48 Table 21 Batch Add Extension Parameters ccccccccccceececeeceeseeceseeeeceeeeseeeeeseeceseueeesaueessaeeeseeeesseeeesaes 50 Table 22 Analog Trunk Configuration Parameters cccccecccccssececcssececeeseeeeaeseeeseeseceeeuseeesenseeeseaeeeesaaeees 55 Table 23 PSTN Detection For Analog Trunk llseeesssesssssssssseseeeen nennen nennen nnne nnne nennen nnns 58 Table 24 VoIP Trunk Configuration Parameters 59 Table 25 Outbound Route Configuration Parameters sese 64 Table 26 Inbound Rule Configuration Parameters 66 Table 27 Conference Bridge Configuration 69 Table 28 Conference Caller IVR MCU vesisesssaniascavinetessaceestateandnnrntsaaanneeneseedaaansausiannedgeuaseeatabearinnssthananeentn 73 Table 29 IVR Configuration Parameters ccccccsscccccsscceeseceeceeeceeceneeeeeeeeeeseseeeceeneeessugeeeesaeeesaaeeeesaneees 75 Table 30 Voloemall SOPIDOQS 82 Table 31 Voicemail Email Settings 2 0 0 0 ccccccccccccesecceeeeceeeceseeceeseecesseeeeseeceseeeeeseucessaeeeseueesseeeseneesseeeesees 8
2. 128 NO LOCAL FIRMWARE SERVERS 130 Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 4 of 138 istan innovative IP Voice amp Video BACKUP E E 130 LOCAL BACI UP 130 NETWORICBACRUP 131 RESTORE CONFIGURATION FROM BACKUP FILE rere 132 CLEANER V eee ee ee 133 RESET AND REBOOT eee 134 135 THODSEESHOOTIN 135 ELHERNETUAP TUBE a 136 d eee 136 uH e 137 EXPERIENCING THE 6102 6104 6108 6116 138 Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 5 of 138 istan innovative IP Voice amp Video Table of Tables UCM61xx User Manual Table 1 Technical sass ta EN iw Fired da 12 Table 2 UCM6102 UCM61 04 Equipment Packaging ccccsscccccesecceeeseeeeeeeeeteeueeeeseseeesenseeesaaeeeesaaees 15 Table 3 UCM6108 UCM6116 Equipment Packaging cccccsscccccssececeseeeeaeeeeeseeseeeseuseeeeeueeesaaeeeesaaeees 15 Table4 Se Emm 21 Table 5 UCM6102 UCM6104 LED INDICATORS esssesessessseeeemennenmennnnnnn nnn nnn nnn nennen nnn 22 Table 6 UCM6108 U
3. PH 124 alll FSO OU RR 125 Call Report Entry With Audio Recording File 126 126 Network Upgrade TEES 127 oca UDI glo MR P 129 Upgrading Firmware File s cccccsssccccsseeecceeeeeesesseeeseaeeeseeeeessaeeeesauscessaaesesseaeeessseeeesaeeensas 129 FSO OIG o SERRE RR mm 129 MEET T Tm 131 NOWO dE ONES 132 Restore UCM61xx From Backup Fil ccccccccssseceeeeaeeeeeeeeseeeceeeseeeeeeeesaeeeeeeseaeaseeesssaaeeeees 133 NC ANON EMEN c 134 Reset and 135 STS O AO lt ERE 136 cs 137 T 137 UCM6102 6104 6108 6116 USER MANUAL Page 9 of 138 andstream innovative IP Voice amp Video CHANGE LOG This section documents significant changes from previous versions of the UCM61xx 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 1 22 e This ts the initial version Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 10 of 138 istan innovative IP Voice amp Video
4. Analog e SIP AX Username Display username for this trunk Port Hostname IP Display 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 web GUI gt PBX gt Basic Call Routes gt Analog Trunks e Click on 3 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 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 116 of 138 6000 6001 5002 5003 5004 5005 5006 6007 6008 6000 001 AmyB5002 Amy5003 004 John Doe Alex Chan IAX 6008 ahal lame Label Emily Green SIP User SIP User SIP User SIP User SIP User SIP User SIP User SIP User AX User stream innovative IP Voice amp Video 6009 6009 lessage Analog User FXS 1 gy Voice Mail Main Features g Dial Voice Mail Features Call Pickup Features Pageing Prefix Features 80 Intercom Prefix Features Total 25 Show 1 2 Jumpto ca ca Figure 47 Extension Status Table 53 Extension Status Display extension number including feature code The color indicator has the following definitions Green Free Blu
5. A new dialog window of voice prompt package list will be displayed Users can see the version number latest version available V S current installed version package size and options to upgrade or download the language Voice Prompt Package List English 0 9 1 0 hir 0 9 0 9 Deutsch 1 0 Francais 1 0 pl 1 0 7 Italiana 1 0 3 4M Espanol 1 0 3 7M Palski 1 01 4 2M Portugu s 1 0 3 2M Total 10 Show 1 2 Go Figure 33 Voice Prompt Package List Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 80 of 138 andstream innovative IP Voice amp Video m Click on to download the language to the UCM61xx The installation will be automatically started once the downloading is finished Language Settings Upload Voice Prompt Package D Choose Voice Prompt to Upload Upload Voice Prompt Package List 1 Language amp English Delete Cancel Check Prompt List A new language option will be displayed after successfully installed Users then could select it to apply in the UCM61xx system voice prompt or delete it from the UCM61 xx CUSTOMIZE AND UPLOAD VOICE PROMPT PACKAGE The UCM61xx provides interface from web GUI for users to customize their own voice prompts Users could directly upload the package from web GUI For detailed instructions on voice prompt customizing and uploading please refer to the link below http www grandstream com product
6. yam innovative The current system time on UCM61xx is displayed under Web GUI gt Status gt System Status To change the time settings on the UCM61xx to Web GUI gt Settings gt Time Settings NTP Server Enable DHCP Option 2 Enable DHCP Option 42 Time Zone Self Defined Time Zone Firmware Version 1 0 1 22 Table 18 Time Settings Specify the URL or IP address of the NTP server for the UCM61xx to synchronize the date and time The default NTP server is ntp ipvideotalk com If set to Yes the UCM61 xx is allowed to get provisioned for Time Zone from DHCP Option 2 in the local server automatically The default setting is Yes If set to Yes the UCM61xx is allowed to get provisioned for NTP server from DHCP Option 42 in the local server automatically This will override the manually configured NTP Server The default setting is Yes Select the proper time zone option so the UCM61xx can display correct time accordingly The default setting is GMT 05 00 Eastern Time If Self Defined Tome Zone is selected please specify the time zone parameters in Self Defined Time Zone field as described in below option 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 2 6 5 4 1 0 11 1 0 MTZ 6MDT 5 This indicates a time zone
7. Figure 30 Record New IVR Prompt e Specify the IVR file name e Select the format GSM or WAV for the IVR prompt file to be recorded e Select the extension to receive the call from the UCM61xx to record the IVR prompt e Click the Record button A request will be sent to the UCM61xx The UCM61xx will then call the extension for recording the IVR prompt from the phone e Pick up the call from the extension and start the recording following the voice prompt e The recorded file will be listed in the IVR Prompt web page Users could select to re record play or delete the recording UPLOAD IVR PROMPT If the user has a pre recorded IVR prompt file click on Upload IVR Prompt in Web GUI gt PBX gt Internal Options gt IVR Prompt page to upload the file to the UCM61xx The following are required for the IVR prompt file to be successfully uploaded and used by the UCM61 xx e PCM encoded e 16 bits e 8000HZ mono e In mp3 or format or raw ulaw alaw gsm file with ulaw or alaw suffix e File size under 5M Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 77 of 138 andstream innovative IP Voice amp Video Upload IVR Prompt Choose voice prompt to upload Sound file must be PCM encoded 16 bits at S000Hz mono with mp3 wav format or raw ulaw alaw gsm file with ulaw _alaw suffix The file size must be under 5M Choose file ta upload Upload Figure 31 Upload IVR Prompt
8. to add a new outbound route e Click on to edit the outbound route e Click on to delete the outbound route e Clickon to move the outbound route up down to arrange the priority of the outbound rule The outbound rule listed on the top has higher priority When the dialing pattern matches two or more outbound rules for example the same pattern is configured for 2 different trunks or dialing out 1000 matches pattern 1xxx for trunk 1 and pattern 100x for trunk 2 the one list on the top will be used Table 25 Outbound Route Configuration Parameters Configure the name of the calling rule e g local long_distance and Calling Rule Name etc Letters digits _ and are alllowed All patterns are prefixed with the _ e Special characters X Any Digit from 0 9 Z Any Digit from 1 9 N Any Digit from 2 9 Wildcard Match or more characters Pattern l Wildcard Match zero or more characters immediately Example 12345 9 Any digit from 1 to 9 Select privilege level for the outbound rule Privilege Level e Internal The lowest level required All users can use this rule e Local Users with Local National or International level are allowed Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 64 of 138 Pin Set ndstream innovative IP Voice amp Video to use this rule e National Users with National or International level are allowed to use this rule e Internation
9. 5003 Ricky Chan 5004 Front Desk 5005 Warehouse 5006 Sales 5008 Customer Service 5007 Tech Support 5009 Ring Group Options Ring Strategy Shinninn Ring in order Ring Timeout on Each Member 30 5 Enable Voicemail Voicemail Secret Email Address Firmware Version 1 0 1 22 Extension of Ring Group 4234199802 techsuppon ucmb1txx sample com Cancel Save Figure 37 Ring Group Configuration UCM6102 6104 6108 6116 USER MANUAL Page 86 of 138 andstream innovative IP Voice amp Video PAGING AND INTERCOM GROUP The UCM61xx paging and intercom can be used via feature code to a single extension or a paging intercom group This sections describes the configuration of paging intercom group under Web GUI gt PBX gt Call Features gt Paging Intercom CONFIGURE PAGING INTERCOM GROUP e Click on Create New Paging Intercom Group to add paging intercom group Edit Paging Intercom Group 5101 Extension 5101 Type 2 Way Intercom Paging Intercom Group Members Avaliable Users SIP 5005 Warehouse SIP 5000 John Dae SIP 5010 Shipping SIP 5001 Stacy Green SIP 5002 Tom Lin SIP 5003 Ricky Chan SIP 5004 Front Desk SIP ANNA Salas Cancel Save Figure 38 Page Intercom Group Table 33 Page Intercom Group Configuration Parameters Extension Configure page intercom group extension Type Select 2 way In
10. Intercom Prefix Blacklist Add Blacklist Remove Call Pickup on Ringing CALL RECORDING Default Code 98 Enter 98 and follow the voice prompt Or dial 98 followed by the extension and to access the entered extension s voicemail box Default Code 97 Press 97 to access the voicemail box Default Code 83 Pause the agent in all call queues Default Code 84 Unpause the agent in all call queues Default Code 81 To page an extension enter the code followed by the extension number Default Code 80 To intercom an extension enter the code followed by the extension number Default Code 40 To add a number to blacklist for inbound route dial 40 and follow the voice prompt to enter the number Default Code 41 To remove a number from current blacklist for inbound route dial 41 and follow the voice prompt to remove the number Default Code To pick up a call for extension xxxx enter the code followed by the extension number The UCM61xx allows users to record audio during the call Please follow the instructions below to record the call Make sure the feature code for Audio Mix Record is configured and enabled Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 97 of 138 stream e After establishing the call enter the Audio Mix Record feature code by default it s 3 followed by SEND to start recording e stop the recording
11. Configuration Partition Total 96MB Available 54 MB Used 40 MB USB Disk a Partition 1 Total 1 888MB Available 1 855 MB Used 31 MB Data Partition Total 3 168MB Available 3 071 MB Used 95 MB SD Card 1 Partition 1 Total 3 808MB Ayailable 3 455 MB Used 351 MB andstream innovative IP Voice amp Video Figure 52 System Status gt Storage Usage RESOURCE USAGE When configuring and managing the UCM61xx users could access resource usage information to estimate the current usage and allocate the resources accordingly Under Web GuUI gt Status gt System Status gt Resource Usage the current CPU usage and Memory usage are shown in the pie chart Resource Usage CPU Usage Memory Usage Figure 53 System Status gt Resource Usage Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 123 of 138 andstream innovative IP Voice amp Video 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 UCM61xx e Start Time Format 2013 03 27 16 47 03 e Call From Format John Doe lt 6012 gt Call To Format 6005 e Call Time Format 0 00 10 e Talk Time Format 0 00 10 e Status Format NO ANSWER BUSY ANSWERED or FAILED e Option Voice
12. Fill in the Subject and Message content to be used in the Email l when sending the Fax to the users Template Variables The template variables are Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 93 of 138 istan innovative IP Voice amp Video CALLERIDNUM Caller ID Number CALLERIDNAME Caller ID Name RECEIVEEXTEN The extension to receive the Fax FAXPAGES Number of pages in the Fax VM_DATE The date and time when the Fax is received e Click on to edit the Fax extension e Clickon to delete the Fax extension Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 94 of 138 andstream innovative IP Voice amp Video CALL FEATURES The UCM61xx supports call recording transfer call forward call park and other call features via feature code This section lists all the feature codes in the UCM61xx and describes how to use the call features FEATURE CODES Table 36 UCM61xx Feature Codes Feature Maps e Default code 1 e Enter the code during active call After hearing Transfer you will hear dial tone Enter the number to transfer to Then the user will be disconnected and transfer is completed e Options Disable Allow Caller Enable the feature code on caller side only Allow Callee Enable the feature code on callee side only Allow Both Enable the feature code on both caller and callee Blind Transfer e Default code 2 e Enter th
13. Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 91 of 138 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 upload music files The default Music On Hold class already has 5 audio files defined for users to use Manage Music On Hold Create New MOH Class Music On Hold Classes default e Delete Upload an 8 KHz Mono Music file file size under 5M Choose file to upload Upload List of Sound Files macroform cold_day wav macroform robot_dity wav macroform the simplicity wavy Manolo camp maorning coffee wav rena praject system wav Figure 41 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 Click on to select music file from local PC and click to start uploading The music file uploaded has to be 8 KHz Mono format with size smaller than 5M EE e Clickon to delete the sound file for the Music On Hold Class from the list of sound files Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 92 of 138 m Innovative FAX T 38 The UCM61xx supports T 30 T 38 Fax and Fax Pass through After receiving the Fax UCM61xx can convert it to PDF format and send it to the configured Emai
14. UCM6102 6104 6108 6116 USER MANUAL Page 56 of 138 Congestion Tone PSTN Detection PSTN DETECTION andstream innovative IP Voice amp Video Default value f1 480 50 f2 620 50 c 500 500 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 manually configure Busy Tone value Default value f1 480 50 f2 620 50 c 250 250 Click on Detect to detect the busy tone Polarity Reversal and Current Disconnect by PSTN Before the detecting please make sure there are more than one channel configured and working properly If the detection has busy tone the Tone Country option will be set as Custom The UCM61xx provides PSTN detection function to help users detect the busy tone Polarity Reversal and Current Disconnected by PSTN during analog trunk setup Select the analog trunk under Web GUI gt PBX gt Basic Call Routes gt Analog Trunks page first In the dialog window to edit the trunk click on Detect for PSTN Detection The following dialog will show for users to perform the detection Edit Analog Trunk trunk 2 i WL ort i i Detect model Auto Detect Source Channel to be 1 detected Destination Channel 1 Destination Number Not
15. WELCOME Thank you for purchasing Grandstream UCM6102 6104 6108 6116 UCM6102 6104 6108 6116 is an innovative IP PBX appliance designed for small to medium business Powered by an advanced hardware platform with robust system resources the UCM6102 6104 6108 6116 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 6104 6108 6116 offers quick setup deployment with ease and unrivaled reliability all at an unprecedented price point A Caution Changes or modifications to 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 6104 6108 6116 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 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 11 of 138 ndstream innovat
16. e Local Users with Local National or International level are allowed to use this rule Privilege Level e National Users with National or International level are allowed to use this rule e International The highest level required Only users with international level can use this rule Default Destination Select the default destination for the inbound call Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 66 of 138 ndstream innovative IP Voice amp Video Extension Extension s voicemail Call Queue Ring Group Voicemail Access Code Fax Operator Hangup Congestion By DID Time Condition Start Time Select the start time hour minute for the trunk to use the inbound rule End Time 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 Date the inbound rule Week Select the day in the week to use the inbound rule Select the default destination for the inbound call Destination e Extension Extension s voicemail Call Queue Ring Group Voicemail Access Code Fax Operator Hangup Congestion By DID DID Features If enabled external users can dial outbound calls by DID through Dial Trunk inbound trunks Select the DID destination Only the selected category can be reached by DID e User Extension This is selected by default DID Destination e Conference e Call Queue
17. ndstream Innovative IP Voice amp Video Gi o Networks Ve IP PBX Appliance User Manual 7 Grandstream Networks Inc www grandstream com sea innovative IP Voice amp Video UCM61xx User Manual Index CHANGE 1 10 FIRMWARE VERSION 1 0 1 22 10 WELCOME uuu ii ccccccccccnccncecceccecenccuncecencennuneuneeneuauununneneeununneneengenguneunenneneeneuneens 11 PRODUCT OVERVIEW ice cceccccccceccnccncenceccencencenceneenencuneeneuneuneennuneuneennenes 12 FEATURE HIGHT LIGHTS 2 0 0 ccc ccc cece cc ccccceccccecececceceaceceeeteceaueceaceneguaueesuuaususueausneausneaeauenesteneatsneas 12 TECHNICAL SPECIFICATIONS 12 INSTALLATION 15 EQUIPMENT PACKAGING ccccccccccecccccceccccececuccecuaceceaeeteceaueneacenegcauectsueaususesusneatsnentaneneseeneatanears 15 CONNECT YOUR UCM61 XX 0 0 ccc cece cc eccccecececcececcccececcececcuueausuacuataueausueuauenesuenesuauecesueaususueausueansneas 16 CONNECT THE UCM6 16 CONNECT THE UCMO 109 Rr RR PR Gg ES 17 CONNECT THE UCM6108 1 0 c cccccccceccsececeececscceteceeusneeuneuetsutntsuenususauanseenuseen
18. Network Upgrade Configuration Network Backup Configuration Cleaner Configuration UCM6102 6104 6108 6116 USER MANUAL istan innovative IP Voice amp Video Page 7 of 138 C istan innovative IP Voice amp Video Table of Figures UCM61xx User manual Figure 1 UCM6102 Front 16 Figure 2 UCIIG10Z2 Back VION EET Em 16 Figure 3 UCM6104 Front 17 Figure 4 UCM6104 Back VIGI seisein a a Ee a i 17 Figure 5 UCM6108 Front 18 Figure 6 UCIVIG108 Back ViGW E E 18 Figure 7 UCMG6116 Front d RT u 18 Figure 8 UCMG6116 Back E 18 Figure 9 UCM6116 Web GUI Login Page nnnm nnn nnne n nnn nnns nnns 23 Figure 10 UCM61xx Web GUI Language sssssesssssseeeennenennen nennen nnne 25 Figure 11 Create New Firewall Rule eessseeesessssseseeseeeeen nennen nnne nnn nnne nnn nnn nnn 32 Figure 12 LDAP Server nenne nnn nnne rnnt nnn nnne nnns 34 Figure 13 Default LDAP Phonebook in UCMO1XXx essseseesssssssesenneennnn nennen nnnm nn nn nnn nnns 35 Figure T4 Add EDAPPholebOOK tae
19. UCM6102 30 Be 30 SAHA O Be 30 Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 1 of 138 istan innovative IP Voice amp Video DVAAMICDERENSE 33 33 LDAP ee ee 34 LDAP SERVER CONFIGURATIONS eritis ttti ttim tente 34 LOAP PHONE BOOK aatem uua 35 LDAP CLIENT 36 1312 7 37 EMAI EIC c 38 ul zzwccc O 39 jn Od 6 Cot 41 enu 41 iners rq 41 MANUAL PROVISIONING m 44 aeo 44 ACS NMEN 45 CREATE NEW DEVIO E Mut S 45 urea A E eee eed 46 EXAMP EE 46 uz ION c 48 GREATE NEW USER 48 EATCHADD EXTENSION Oreore maT 50 EON EXTENSION 53 ee E M 55 ANALOG TRUNK eee 55 ANALOG TRUNK CONFIGURATION 55
20. identify the trunk when listed in outbound rules inbound rules and etc Configure the IP address or URL for the VoIP provider server of the trunk If enabled the trunk CID will not be overridden by extension s CID when the extension has CID configured The default setting is Configure the Caller ID This is the number that the trunk will try to use when making outbound calls For some providers it might not be possible to set the CallerlD with this option and this option will be ignored When making outgoing calls the following rules are used to determine which will be used if they exist CallerID configured for the extension will be looked up first e lf no CallerlD configured for the extension the CallerlD configured for the trunk will be used e the above two are missing the Global Outbound CID defined in Web GUI gt PBX gt Internal Options gt General will be used Configure the name of the caller to be displayed when the extension has no CallerlD Name configured Select audio and video codec for the VolP trunk The available codecs are GSM G 726 G 722 G 729 G 723 ILBC ADPOM LPC10 H 264 H 263 263 If enabled the UCM61 xx will regularly send SIP OPTIONS to the device to check if the device is still online The default setting is No Enable to detect Fax signal from the trunk during the call and send the received Fax to the default Email address i
21. if the queue has no agent anymore The default setting is Strict e Yes Callers will be disconnected from the queue if all agents are paused or invalid No Never disconnect the callers from the queue when the queue is empty e Strict Callers will be disconnected from the queue if all agents are paused invalid or unavailable Configure whether the callers can dial into a call queue if the queue has no agent The default setting is No e Yes Callers can always dial into a call queue e No Callers cannot dial into a queue if all agents are paused or invalid Strict Callers cannot dial into a queue if the agents are paused invalid or unavailable If enabled the configured PIN number is required for dynamic agent to log in 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 Configure the number of seconds before a new call can ring the queue after the last call on the agent is completed If set to 0 there will be no delay between calls to the queue Configure the maximum number of calls to be queued at once This number does not include calls that have been connected with agents It only includes calls not connected yet The default setting is 0 which means unlimited When the maximum value is reached the caller will be treated with busy tone followed by the next calling rule after attempting to enter the queu
22. m F 24 Default Password admin Cancel Save Figure 18 Auto Provision Settings Table 19 Auto Provision Settings Enable or disable the zero config feature on the PBX The default setting is disabled If enabled when the device is discovered the PBX will automatically Enable Zero Config Automatically Assign Extension assign an extension to the device The default setting is disabled Specify the starting extension to be created assigned If the extension is Starting Extension assigned to existing device already this extension will be skipped and the next available extension will be used The default setting is 6000 If enabled random password will be generated for the extension when Generate Random Password l l it s created Otherwise default password will be used Specify default password for the extension if no random password is Default Password generated The default setting is admin Click on Save and then reboot the phones to have the discovery and provisioning process started Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 43 of 138 andstream innovative IP Voice amp Video 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 ARP e SIP MESSAGE OPTIONS Click on Auto Discover fill in t
23. ndstream innovative IP Voice amp Video Once added users can select to edit the phonebook attributes and contact list see figure below or select to delete the phonebook Edit Phonebook ou people dc pbx dc com LDAP Attributes Contact List FOEDE et N AccountName L 4 P Ji 11 BW AccountName CallerlbDName Email FirstName Department MabileMumber HomeNumber Fax D D D D LastName o D D o Figure 15 Edit LDAP Phonebook LDAP CLIENT CONFIGURATIONS The configuration on LDAP client is similar when you use other LDAP servers Here we provide an example on how to configure the LDAP client on the SIP end points to use the default PBX phonebook Please follow the instructions in the LDAP Client Configurations section described below Suppose your server Base DN is dczGrandstream your extension number is 1000 and your LDAP entry password is 1000 configure your LDAP client as follows case insensitive Base DN dczGrandstream Root DN AccountName 1000 dc Grandstream Password 1000 Filter amp CalleriDNamez AccountName z Port 389 Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 36 of 138 andstream innovative IP Voice amp Video The following figure shows the configuration information on a Grandstream GXP2200 to successfully use the LDAP server as configured in Figure 12 LDAP Server Configurations Server Ad
24. 0 1 22 If enabled random password will be generated when the extension is created The default setting is Yes It is recommended to enable it for security purpose If set to Yes users could disable the extension range pre configured configured on the UCM61xx The default setting is No The default extension range assignment is e User Extension 5000 6299 e Conference Extension 6300 6399 IVR Extension 7000 7100 e Ring Group Extension 6400 6499 e Queue Extensions 6500 6599 e Group Extension 6600 6699 UCM6102 6104 6108 6116 USER MANUAL Page 99 of 138 ndstream innovative IP Voice amp Video Note It is recommended to keep the system assignment to avoid inappropriate usage and unnecessary issues INTERNAL OPTIONS RTP SETTINGS Table 38 Internal Options Jitter Buffer SIP Jitter Buffer Enable Jitter Buffer Force Jitter Buffer Log Frames Max Jitter Buffer Resync Threshold Implementation Select to enable jitter buffer on the sending side of the SIP channel The default setting is Select to force the use of jitter buffer on the receiving side of the SIP channel The default setting is Select to enable jitter buffer frame logging The default setting is Configure the maximum time in ms to buffer for Adaptive jitter buffer implementation or used as the jitter buffer size for Fixed jitter buffer implementation The default setting is 200 Confi
25. 47 of 138 andstream innovative IP Voice amp Video EXTENSIONS CREATE NEW USER 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 dialog window will show for users to fill in the extension information The configuration parameters are as follows Table 20 Extension Configuration Parameters Extension The extension number associated with the user Configure the CallerlD Name associated with the user Number letter CallerlD Name or space are allowed Configure the CallerlD Number that would be applied for outbound calls from this user CallerlD Number Note The ability to manipulate your outbound Caller ID may be limited by your VoIP provider Assign permission level to the user The available permissions are Internal Local National and International from the lowest level to the highest level The default setting is Internal Permission Note Users need to have the same level as or higher level than a outbound rule s privilege in order to make outbound calls from this rule Configure the password for the user A random secure password will be SIP IAX Password automatically generated It is recommended to use this password for security purpose Enable Voicemail Enable Voicemail for the user The default setting is Yes Configure Voicemail password digits only A random numeric Voicemail Password password is automatically generated It
26. Conference Recording 74 Figure 29 Click On Prompt To Create IVR Prompt cccccccccccsseceeeceeeeceeeeeeeeceeeeeeeeeesesaaeeeeeesaaaeeeseeaaaes 76 Figure 30 Hecord New IVR Promp rw GR HERR YER 77 Figure 31 Upload IVR 78 Figure 32 Language Settings For Voice Prompt sess 80 Figure 33 Voice Prompt Package 80 Figure 34 Voicemail Email SettinGS cccccccccssseccceseceeceeececseeeceeseeseeesuaeeeseeuseessaeeeeseeeeeseaeeessaueeessegeeseas 83 Figure 35 VOICE Inu E GrOUD 84 FOUE 0 RIDO TOUD 85 Figure 37 Ring Group Configuration 86 ROU o RR m 87 Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 8 of 138 Figure 39 Figure 40 Figure 41 Figure 42 Figure 43 Figure 44 Figure 45 Figure 46 Figure 47 Fig
27. Conference administrator can always invite other users without enabling this option If enabled the caller will be announced to all conference participants when there the caller joins the conference The default setting is No Note Quiet Mode and Announce Callers cannot be enabled at the same time If enabled the voice quality for conference call will be improved However this could cause voice delay and increase system resource usage The default setting is No If enabled no authentication will be required when joining the UCM6102 6104 6108 6116 USER MANUAL Page 70 of 138 andstream innovative IP Voice amp Video conference call The default setting is Yes Play Hold Music For First If enabled the UCM61xx will play Hold music to the first participant in Caller the conference until another user joins in The default setting is No Skip Authentication When If enabled the invitation from Web GUI for a conference bridge with Invite User via Trunk from Web password will skip the authentication for the invited users The default GUI setting is No JOIN A CONFERENCE CALL Users could dial the conference bridge extension to join the conference If password is required enter the password to join the conference as a normal user or enter the admin password to join the conference as administrator INVITE OTHER PARTIES TO JOIN CONFERENCE When using the UCM61xx conference bridge there are two ways to invite other
28. Configure the name of the caller to be displayed when the extension has no CallerlD Name configured Enter the username to register to the trunk from the provider Enter the password to register to the trunk from the provider Select audio and video codec for the VolP trunk The available codecs are PCMU GSM G 726 G 722 G 729 G 723 ILBC ADPCM LPC10 H 264 H 263 263 If enabled the UCM61 xx will regularly send SIP OPTIONS to the device to check if the device is still online The default setting is No Enable to detect Fax signal from the trunk during the call and send the received Fax to the default Email address in Fax setting page under web GUI gt PBX gt Internal Options gt Fax T 38 Note If enabled Fax Pass through cannot be used Enable SRTP for the VoIP trunk The default setting is No UCM6102 6104 6108 6116 USER MANUAL Page 63 of 138 innova tive IP Voice CALL ROUTES OUTBOUND ROUTES In the UCM61xx 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
29. Enable Select or HTTPS The default setting is HTTPS Specify port number to access the HTTP server The default port number is 8089 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 UCM61xx 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 Display Name Sender Firmware Version 1 0 1 22 Table 17 Email Settings Enable or disable TLS during transferring 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 Specify the display name in the FROM header in the Email Specify the sender s Email address For example pobx example mycompany com UCM6102 6104 6108 6116 USER MANUAL Page 38 of 138 TIME SETTINGS
30. HTTPS SSH Physical e Output 12VDC 1 5A Input 100 240VAC 50 60Hz e Operating 32 104 F 0 40 C 10 90 non condensing e Storage 14 140 F 10 60 C UCM6102 6104 226mm L x 155mm W x 34 5mm UCM6108 6116 440mm x 185mm x 44mm UCM6102 6104 Wall mount and Desktop UCM6108 6116 Rack mount and Desktop Additional Features Yes English Chinese Spanish French German Russian ltalian for Web Universal Power Supply Environmental Dimensions Mounting Multi language Support GUI Customizable IVR to support any language Caller ID Bellcore Telcordia ETSI FSK ETSI DTMF SIN 227 BT NTT Japan Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 13 of 138 Polarity Reversal Wink Call Center Customizable Auto Attendant Concurrent Calls Conference Bridges Call Features Compliance Firmware Version 1 0 1 22 Yes with enable disable option upon call establishment and termination Multiple configurable call queues automatic call distribution ACD 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 6116 Up to 60 simultaneous calls UCM6102 6104 Up to password protected conference bridges allowing up to 25 simultaneous PSTN or IP participants UCM6108 6116 Up to
31. Name to identify the voicemail group Letters digits _ and are allowed e Voicemail Group Mailboxes Select available mailboxes from the right list and add them to the left list The extensions need to have voicemail enabled to be listed in available mailboxes list Click on Save to finish the configuration Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 84 of 138 istan innovative IP Voice amp Video RING GROUP The UCM61xx supports ring group feature with different ring strategies applied to the ring group members This section describes the ring group configuration on the UCM61 xx CONFIGURE RING GROUP Ring group settings can be accessed via Web GUI gt PBX gt Call Features gt Ring Group Create New Ring Group Ring Group Name Members techsupport 6005 6006 6007 Figure 36 Ring Group e Click on Create New Ring Group to add ring group e Click on to edit the ring group The following table shows the ring group configuration parameters Clickon to delete the ring group Table 32 Ring Group Parameters Configure ring group name to identify the ring group Letters digits _ Ring Group Name and are allowed Extension Configure the ring group extension Select available users from the right side to the ring group member list on the left side Click 62 amp J to arrange the order Select the ring strategy Ring Group Members e Ring simultaneousl
32. USER MANUAL Page 29 of 138 istan innovative IP Voice amp Video 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 PORT FORWORDING UCM6102 ONLY The UCM6102 network interface supports router functions which provides users the ability to do port forwarding Please see port forwarding settings in the table below Table 11 UCM6102 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 LAN Port Specify the LAN port number Select protocol type UDP Only TCP Only or TCP UDP for the forwarding in Protocol Type ae the selected port The default setting is UDP Only FIREWALL The UCM61xx provides users firewall configurations to prevent certain malicious attack to the UCM61 xx system Users could configure to allow restrict or reject specific traffic through the device for security and bandwidth purpose To configure firewall settings in UCM61xx go to Web GUI gt Settings gt Firewall page STATIC DEFENSE Under Web GUI gt Settings gt Firewall gt Static Defense page users will see the following information e Current service information with port process and type e Typical firewall settings e Custom f
33. Voice amp Video Figure 51 Parking Lot Status Table 56 Parking Lot Status Caller ID Display the caller ID who parks the call Channel Display channel for the call park Extension Display the parking lot number where the call is parked retrieved Display 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 parks 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 e Click on 6 to refresh the parking lot status e Click on to expand the parking lot details Click on to hide the parking details SYSTEM STATUS The UCM61xx 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 GENERAL Under Web GUI gt Status gt System Status gt General users could check the hardware and software information for the UCM61xx Please see details in the following table Table 57 System Status gt General Status gt System Status gt General Model Product model Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 121 of 138 stream innovative IP Voice am
34. click on to delete the recording file the call record entry will not be deleted 2013 07 03 18 27 47 0 00 14 2013 07 03 17 55 04 0 00 16 Figure 56 Call Report Entry With Audio Recording File CDR Statistics is an additional feature on the UCM61xx 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 Figure 57 CDR Statistics Table 60 CDR Statistics Filter Criteria Trunk Type Select one of the following trunk type e SIP Calls e PSTN Calls Call Type Select one or more in the following checkboxes e Inbound calls e Outbound calls e Internal calls e External calls All calls Time Range e By month of the selected year e By week of the selected year e By day of the specified month for the year e By hour of the specified date range For example 2013 01 To 2013 03 Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 126 of 138 andstream innovative IP Voice amp Video UPGRADING AND MAINTENANCE UPGRADING The UCM61xx can be upgraded to a new firmware version remotely or locally This section describes how to upgrade your UCM61xx via network or local upload UPGRADING VIA NETWORK The UCM61xx can be upgraded via TFTP HTTP HTTPS by configuring the URL IP Address for the TFTP HTTP HTTPS
35. means binding to all addresses If enabled the UCM61xx allows unauthorized INVITE coming into the PBX and the call can be made The default setting is No Select to enable overlap dialing support The default setting is No If set to No all transfers initiated by the endpoint in the UCM61xx will be disabled unless enabled in peers or users The default setting is Yes Select to enables DNS SRV lookups on outbound calls from the UCM61xx The default setting is Yes When sending MWI NOTIFY requests this value will be used in the From header as the name field If no From User is configured the user field of the URI in the From header will be filled with this value Configure the domain for the UCM61xx Incoming INVITE and REFER messages can be matched against a list of allowed domains each of which can direct the call to a specific context if desired By default all domains are accepted and sent to the default context or the context associated with the user peer placing the call Register to non local domains will be automatically denied if a domain list is configured Up to 10 domains can be added Configure the domain in the From header of the SIP message It may be required by some providers for authentication If enabled the UCM61xx will add local host name and local IP to domain list The default setting is If enabled requests for external domains that are not served by the UCM6102 610
36. parties to join the conference e Invite from Web GUI For each conference bridge in UCM61xx Web GUI gt PBX gt Call Features gt Conference there is an icon for option Invite a participant Click on it and enter the number of the party you would like to invite Then click on Add A call will be sent to this number to join it into the conference Invitation Participants Extensian Figure 27 Conference Invitation From Web GUI Note When a user invite other parties to join a conference from Web GUI the user doesn t have to be in the conference bridge Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 71 of 138 andstream innovative IP Voice amp Video e Invite by dialing or 1 during conference call A conference participant can invite other parties to the conference by dialing from the phone during the conference call Please make sure option Enable User Invite is turned on for the conference bridge first Enter O or 1 during the conference call Follow the voice prompt to input the number of the party you would like to invite A call will be sent to this number to join it into the conference 0 If O is entered to invite other party once the invited party picks up the invitation call a permission will be asked to accept or reject the invitation before joining the conference 1 1 is entered to invite other party no permission will be required from the invited party Note Confere
37. record playing downloading deleting 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 View Report button to display the generated report Call Detail Report D Inbound calls Caller Number Caller Name D Outbound calls From Date Date D Internal calls 1 External calls Figure 54 CDR Filter Table 59 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 Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 124 of 138 stream innovative IP Voice amp Video External calls External calls are calls sent from one trunk to another trunk which are not sent to any internal extension Caller Number Enter the caller number to be filtered in the CDR report Caller Name Enter the caller name to be filtered in the CDR report From Date Specify From date and time to be filtered for the CDR report Click on the field and the calendar will show for users to select the exact date and time To Date Specify To date and time to be filtered for the CDR repo
38. repeat the prompt if the DTMF input is invalid When the loop ends it will go to the invalid destination if configured or hang up The default setting is 4 Select the event for each key pressing for 0 9 Timeout and Invalid The event options are e Extension e VoiceMail Conference Rooms e VoiceMail Group e IVR e Ring Group e Queues e Page Group IVR Prompt e Hangup CREATE IVR PROMPT To record new IVR prompt or upload IVR prompt to be used in IVR 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 Name Main i Extension Dial Other Extensions Dial Trunk Permission Internal Welcome Prompt None C Em Timeout 10 Figure 29 Click On Prompt To Create IVR Prompt Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 76 of 138 andstream Once the IVR prompt file is successfully added to the UCM61x lt x it will be added into the prompt list options for users to select in different IVR scenarios RECORD NEW IVR PROMPT In the UCM61xx 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 Record New IVR prompt File Name Welcome Prompt 1 Format WAM I Dial This User Extension to 6000 Record a New Voice Prompt
39. 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 Upgrade Via Firmware Server Path fw ipvideotalk com gs Firmware File Prefix Firmware File Suffix HTTP HTTPS User HTTRIHTTPS Password Cancel Figure 58 Network Upgrade Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 127 of 138 istan innovative IP Voice amp Video Table 61 Network Upgrade Configuration Upgrade Via Allow users to choose the firmware upgrade method HTTP or HTTPS Firmware Server Path Define the server path for the firmware server Firmware File Prefix If configured only the firmware with the matching encrypted prefix will be downloaded and flashed into the UCM61xx Firmware File Suffix If configured only the firmware with the matching encrypted postfix will be downloaded and flashed into the UCM61 xx HTTP HTTPS User Name The user name for the HTTP HTTPS server HTTP HTTPS Password The password for the HTTP HTTPS server Please follow the steps below to upgrade the firmware remotely e Enter the firmware server path under Web GUI gt Maintenance gt Upgrade e Click on Save Then reboot the device to start the upgrading proces
40. 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 The default syslog level for all modules is error which is recommended in your UCM61xx settings because it can be helpful to locate the issues when errors happen Some typical modules for UCM61xx functions are as follows and users can turn on notic and verb levels besides error level pbx This module is related to general PBX functions chan sip This module is related to SIP calls chan dahdi This module is related to analog calls FXO F XS app meetme This module is related to conference bridge TROUBLESHOOTING On the UCM61xx users could capture traces ping remote host and traceroute remote host for troubleshooting purpose under Web GUI Maintenance Troubleshooting Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 135 of 138 andstream innovative IP Voice amp Video ETHERNET CAPTURE The captured trace can be downloaded for analysis Also the instructions or result will be displayed in the web GUI output result Ethernet Capture 1 Interface Type LAN Gi Capture Filter hast 192 168 40 178 gt Start Stop e Download Output Result capture Dignastic run Package capturing Done Click on Download to download the captured packages Figure 67 Ethernet Capture The output result
41. the computer to the same network as the UCM61 xx 2 Ensure the device is properly powered up and shows its IP address on the LCD Open Web browser on the computer and enter the web GUI URL in the following format http s IIP Address Port where the P Address is the IP address displayed on the UCM61xx 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 Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 23 of 138 andstream 4 Enter the administrator s login and password to access the Web Configuration Menu The default administrator s username and password is admin and admin It is highly recommended to change the default password after login for the first time WEB GUI CONFIGURATIONS There are four main sections in the Web GUI for users to view the PBX status configure and manage the PBX e Status Displays PBX status System Status and CDR e PBX To configure extensions trunks call routes zero config for auto provisioning call features internal options settings and SIP settings e Settings To configure network settings firewall 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 WEB GUI LA
42. the default password admin to a more complicated password for security purpose Follow the steps below to change the Web GUI access password 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 Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 33 of 138 andstream 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 UCM61xx has an embedded LDAP server for users to manage corporate phonebook in a centralized manner By default the LDAP server has generated the phonebook based on the created extensions already If users have the Grandstream phone provisioned by the UCM61xx the LDAP directory has been set up on the phone and can be used right away Also users could manually configure the LDAP client settings accordingly to manipulate the built in LDAP server on the PBX To access LDAP Server settings go to Web GUI gt Settings gt LDAP Server LDAP SERVER CONFIGURATIONS The following figure shows the default LDAP server configurations on the UCM61 xx LDAP Server configurations Base DN dc pbx dc com Pbx dn ou pbx dc pbx dc com Root DN cn admin dc pbx dc com Root Password sili Root Password Confirm Allow ano
43. with integrated PoE Plug IEEE 802 3at 2009 UCM6102 6104 Dual 10M 100M 1000M RJ45 Ethernet ports with integrated PoE Plug IEEE 802 3at 2009 PSTN Line FXO Ports Network Interfaces NAT Router Yes UCM6102 only Peripheral Ports USB SD LED Indicators Power Ready Network PSTN Line USB SD Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 12 of 138 ndstream innovative IP Voice amp Video LCD Display 128x32 graphic LCD with DOWN and OK button Reset Switch Yes Capabilities LEC with NLP Packetized Voice Protocol Unit 128ms tail length carrier grade Line Echo Cancellation Dynamic Jitter Buffer Modem detection Voice over Packet Capabilities and auto switch to G 711 2 711 A law U law G 722 G723 1 5 3K 6 3K 86 726 G 729A B iLBC Voice Fax Codecs GSM T 38 Video Codecs H 264 H 263 H 263 QoS Layer 3 QoS Signaling Control DTMF Methods In Audio RFC2833 and SIP INFO 5 auto discovery and auto provisioning Grandstream IP endpoints via ZeroConfig DHCP Option 66 multicast SIP SUBSCRIBE mDNS TCP UDP IP RTP RTCP ICMP ARP DNS DDNS DHCP NTP TFTP SSH HTTP HTTPS PPPoE SIP RFC3261 STUN SRTP TLS Provisioning Protocol and Plug and Play Network Protocols Call Progress Tone Polarity Reversal Hook Flash Timing Loop Current Disconnect Methods Disconnect Busy Tone Media SRTP TLS
44. with 6 hours offset and 1 hour ahead for DST which is U S central time If it is positive the local time zone is west of the Prime Meridian A K A International or Greenwich Meridian If it is negative the local time zone 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 Normally 1 2 3 4 are used If 5 is used it means the UCM6102 6104 6108 6116 USER MANUAL Page 39 of 138 Firmware Version 1 0 1 22 istan innovative IP Voice amp Video last iteration of the weekday The 3rd number indicates weekday 0 1 2 6 for Sun Mon Tues Sat 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 40 of 138 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 HTTP HTTPS download All Grandstream SIP devices support a proprietary binary format configuration file and XML format configuration file The UCM61xx provides a Plug and Play mechanism to auto provision the Grandstream SIP devices in a zero configuration manner by generating XML config file and having the phone to download it This allows users to finish the installation with ease and start using the SIP dev
45. with the SIP User ID SIP server and SIP Password information The SIP server address is the UCM61xx IP address 4 When your phone is registered with the extension dial 97 to access the voicemail box Enter the Voicemail Password once you hear Password voice prompt 5 Once successfully logged in you will be prompted with the Voice Mail Main menu 6 You are successfully connected to the PBX system now Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 25 of 138 C istan innovative IP Voice amp Video SYSTEM SETTINGS This section explains configurations for system wide parameters on the UCM61xx Those parameters include Network Settings Firewall Change Password LDAP server HTTP server Email settings and Time Settings NETWORK SETTINGS After successfully connecting the UCM61 xx 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 The network setting options are similar for UCM6108 and UCM6116 Additional network functions and settings are available for UCM6102 and UCM6104 e UCM6102 supports Router Switch Dual mode functions UCM6104 supports Switch Dual mode functions In this section all the available network setting options are listed for each model Select each tab in web GUI gt Settings gt Network Settings page to configure LAN settings WAN settings UCM6102 only 802 1X and Port Forwardi
46. 1 The above figure shows a common setup among small businesses where the UCM61xx is placed behind a company s router or firewall The phones are in the same network as the UCM61xx and can be discovered automatically by UCM61 xx using the Zero Config feature Example 2 Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 46 of 138 ndstream innovative IP Voice amp Video UCM61xx GXP Phone GXP Phone GXV Phone Figure 24 Provisioning Example 2 This is another typical setup In this setup the UCM61xx is placed directly over the internet outside from the network where the phones are deployed Under this topology the UCM61xx 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 UCM61 xx will need help to get the phones to point itself to the UCM61 xx 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 zccgi 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 UCM61xx and send request for the XML config file e When the phone requests cfgMAC xml from the UCM61xx the UCM61xx will add the phone to the provision list Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page
47. 1 Enable VR Cleaner 1 VR Clean Threshold D VR Clean Time 1 VR Clean Interval Enable CDR Cleaner CDR Clean Time Clean Interval Enable VR Cleaner VR Clean Threshold VR Clean Time Clean Interval Figure 65 Cleaner Table 63 Cleaner Configuration Enable the CDR Cleaner function Enter 0 23 to specify the hour of the day to clean up CDR Enter 1 30 to specify the day of the month to clean up CDR Enter the Voice Records Cleaner function Specify the Voice Records threshold from 0 to 99 by using local storage status in percentage Enter 0 23 to specify the hour of the day to clean up Voice Records 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 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 134 of 138 andstream User Data All the data including voicemail recordings IVR Prompt Music on Hold CDR and backup files will be cleared e All the configurations and data will be reset to factory default Reset amp Reboot Factory Reset Mode Type User Config User Configuration User Data All Reboot Figure 66 Reset and Reboot SYSLOG On the UCM61xx users could dump the
48. 3 RNO GOUD miii tc Om 85 Table 33 Page Intercom Group Configuration 87 Table 34 Call Queue Configuration 89 FAAN Se INS wicca NETT UT 93 Table 36 DUCMO TOC Feature 4 4 4 8428 95 Internal Opions General 99 Table 38 Internal Options Jitter eene nnne nennen 100 Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 6 of 138 Table 39 Table 40 Table 41 Table 42 Table 43 Table 44 Table 45 Table 46 Table 47 Table 48 Table 49 Table 50 Table 51 Table 52 Table 53 Table 54 Table 55 Table 56 Table 57 Table 58 Table 59 Table 60 Table 61 Table 62 Table 63 Firmware Version 1 0 1 22 Internal Options RTP Settings Internal Options Hardware Config Internal Options STUN Monitor Settings General Settings Registration Settings Static Defense SIP Settings General SIP Settings Misc SIP Settings Session Timer SIP Settings TCP and TLS SIP Settings NAT SIP Settings TOS SIP Settings Debug Extension Status Interface Status Indicators Parking Lot Status System Status gt General System Status gt Network CDR Filter Criteria CDR Statistics Filter Criteria
49. 3 Figure 62 Local Backup NETWORK BACKUP Besides local backup users could backup the voice records voice mails CDR FAX in a daily basis to a remote server via SFTP protocol automatically under Web GUI gt Maintenance gt Backup gt Network Backup Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 131 of 138 andstream innovative IP Voice amp Video Manage Configuration Network Backups Backup Configuration o Enable Backup Account D Password D Server Address ucmb51xx backup com o Backup Time 1 Figure 63 Network Backup Table 62 Network Backup Configuration Enable Backup Enable the auto backup function The default setting is Account Enter the Account name on the SFTP backup server Password Enter the Password associate with the Account on the SFTP backup server Server Address Enter the SFTP server address Backup Time Enter 0 23 to specify the backup hour of the day Before saving the configuration users could click on Test Connection The UCM61xx will then try connecting the server to make sure the server is up and accessible for the UCM61 xx Save the changes and all the backup logs will be listed on the web page RESTORE CONFIGURATION FROM BACKUP FILE To restore the configuration on the UCM61xx from a backup file users could go to Web GUI gt Maintenance gt Backup gt Local Backup Alist of previous configuration backups is displayed the web pa
50. 4 6108 6116 USER MANUAL Page 108 of 138 SIP SETTINGS CODECS Essen innova tive IP Voice UCM61xx will be allowed The default setting is Yes The following codecs are supported in UCM61xx for SIP Select the codecs from the right side list to the left FD F ALIM JP Al side Click 62 to arrange the order as appeared in the SDP of the SIP message PCMU e PCMA e GSM G 722 G 726 e ADPCM LPC10 9 729 G 723 e H 263 263 e H 264 SIP SETTINGS MISC Register Timeout Register Attempts Video Max Bit Rate kb s Support for SIP Video Generate Manager Events Reject Non Matching Invites Firmware Version 1 0 1 22 Table 46 SIP Settings Misc Configure the register retry timeout in seconds The default setting is 20 Configure the number of registration attempts before the UCM61xx gives up The default setting is 0 which means the UCM61xx will keep trying until the server side accepts the registration request Configure the maximum bit rate in kb s for video calls The default setting is 384 Select to enable video support in SIP calls The default setting is Yes If enabled the UCM61xx will generate manager events when SIP UA performs events e g Hold The default setting is No If enabled when rejecting an incoming INVITE or REGISTER request the UCM61xx will always reject with 401 Unauthorized instead of notifying the requester whether there is
51. 6 of 138 andstream innovative IP Voice amp Video Figure 68 PING 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 1 Target Hast www google com Start Stop Output Result traceroute Dignostic run traceroute to www google com 74 125 224 179 7 ockock T ae 81 81 csw3 LosAngeles 1 Level3 net 4 639 137 10 14 700 ms 33 675 ms 14 675 ms 8 ae 1 60_edgel LosAngeles9_Level3 net 4 689 144 10 14 000 ms ae 4 30 edge1 LosAngeles9 Level3 net 4 659 144 202 17 900 ms 11 725 ms 9 GOOGLE INC edge1 LosAngeles93 Level3 net 4 53 228 6 20 625 ms 21 550 ms 14 600 ms 10 64 233 174 238 64 233 174 238 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 lax02s01 in f19 1e100 net 74 125 224 179 13 775 ms 11 925 ms Done Figure 69 Traceroute Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 137 of 138 istan innovative IP Voice amp Video EXPERIENCING THE UCM6102 6104 6108 6116 Please visit our website http www grandstream com to receive the most up to date updates on firmware releases additional features FAQs documentation and news on new products We encourage you to browse our product related documentation FAQs and User and Developer Forum for answers to your general questions If you have purchased our products t
52. 6 password protected conference bridges allowing up to 32 simultaneous PSTN or IP participants Call park call forward call transfer ring hunt group paging intercom and etc e FCC Part 15 CFR 47 Class B Part 68 CE EN55022 Class B 55024 EN61000 3 2 EN61000 3 3 EN60950 1 TBR21 RoHS e A TICK AS NZS CISPR 22 Class B AS NZS CISPR 24 AS NZS 60950 AS ACIF 5002 adITU T K 21 Basic Level e UL60950 power adapter UCM6102 6104 6108 6116 USER MANUAL Page 14 of 138 istan innovative IP Voice amp Video INSTALLATION Before deploying and configuring the UCM61xx the device needs to be properly powered up and connected to network This section describes detailed information on installation connection and warranty policy of the UCM61xx 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 2 Screws Yes 6 Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 15 of 138 E istan innovative IP Voice amp Video CONNECT YOUR UCM61XX CONNECT THE UCM6102 LCD Navigation 7 Keys LED Indicators LAN LED WAN LED USB LED 50 CardLED FXS LED FXO LED Figure 1 UCM6102 Front View SD Card Slot Rese
53. 61xx The TLS Enable wu default setting is No Configure the IP address for TLS server to bind to 0 0 0 0 means TLS Bind Address binding to all interfaces The port number is optional If not specified 5061 will be used Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 110 of 138 m Innovative 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 draft ietf sip domain certs Select the TLS protocol for outbound client connections The default setting is TLSv1 TLS Client Protocol This is the CA certificate if the TLS server being connected to requires self signed certificate including server s public key This file will be renames as TLS ca automatically TLS Self Signed CA Note The size of the uploaded ca file must be under 2MB This is the Certificate file pem format only used for TLS connections It contains private key for client and signed certificate for the server This file will be renamed as TLS pem automatically TLS Cert Note The size of the uploaded certificate file must be under 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 TLS CA Cer the accessed servers Note The size of t
54. 7 46 UTC 03 00 120 04 KB meeime confrec 6300 1372205127 347 wav 2013 06 25 21 05 56 UTC 03 00 62 86 KB meetme confrec 6300 1372867161 40 wav 2013 07 03 13 10 29 UTC 03 00 10 17 MB meetme confrec 6300 1372864546 12 wav 2013 07 03 12 16 01 UTC 03 00 35 67 KB meetme conf rec 5300 1372866438 36 wav 2013 07 03 12 47 47 UTC 03 00 322 85 meetme conft ec 5300 1372204987 337 wav 2013 06 25 21 03 30 UTC 03 00 315 98 KB meetme conft rec 5300 1372864583 17 wav 2013 07 03 12 16 36 UTC 03 00 55 67 KB meetme conft rec 6300 1370385024 7 1 wav 2013 06 04 19 35 28 UTC 03 00 4 22 MB a ullum e mim e Figure 28 Conference Recording Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 74 of 138 CONFIGURE IVR yam innovative IVR IVR configurations can be accessed under the UCM61xx Web GUI gt PBX gt Call Features gt IVR Users could create edit view and delete an IVR e Click on Create New IVR to add a new IVR e Click on to edit the IVR configuration e Click on to delete the IVR Name Extension Dial Other Extensions Dial Trunk Permission Welcome Prompt Timeout Timeout Prompt Invalid Prompt Firmware Version 1 0 1 22 Table 29 IVR Configuration Parameters Configure the name of the IVR Letters digits are allowed Enter the extension number for users to access the IVH If enabled all callers to the IVR can dial other ext
55. CM61 xx and check the firmware version after it boots up AN Note Please do not interrupt or power cycle the UCM61 xx during upgrading process Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 129 of 138 istan innovative IP Voice amp Video 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 HTTP server and conduct a local firmware upgrade A free windows version TFTP server is available for download from http www solarwinds com products freetools free tftp server aspx http tftpd32 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 TFTP server 2 Connect the PC running the TFTP server and the UCM61 xx 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 UCM61xx web configuration interface 5 Configure the Firmware Server Path to the IP address of the PC 6 Update the changes and re
56. CM6116 LED INDICATORG ccccccescceceeeeseeeeeeeeseeeeceeeeeceeeeaeseceessaeaeceeetenseeeeeeas 22 Table 7 UCM6102 Network SettingsS gt Basic Settings cccccccccccseecceeseeeeseeeeeseeseeeceeseeeesaeeeeseaeeessaaeees 26 Table 8 UCM6104 Network Settings Basic Settings cccccccccccsseccceseeeeseececeeeeeceeeeeeeeegeeeseaeeeesaaeees 28 Table 9 UCM6108 UCM6116 Network Settings gt Basic SettinGS ccccccccccsssseceeeseseeeeeeseeeeeeseeeeeeeeeas 29 Table 10 UCM61xx Network gt 802 1 29 Table 11 UCM6102 Network Settings gt Port 30 Table 12 UCM61xx Firewall gt Static Defense gt Current 31 Table 13 Typical Firewall Settings ccccccccccsccccesceceeeeceeeeeseeeeeseceeseeesacecseeeesecessueeeseueesseeesensesseeeesens 31 Table T4 Firewall Rule Sells 32 Table 15 Firewall Dynamic Defense cccccccccceeccceececeeeceseeceeseecesseeeseeeseeeeeseucessueeeseueesaueeeseeeesseeesaes 33 Table PIIP SOIVer Oe HS eese di sateen M taU MUNDO MM IMMUNE 38 Table 17 Email Seting TT T sac EEE I reao 38 TOES NE 39 Table 19 Auto Provision
57. Click on L to select audio file from local PC and click on to start uploading Once uploaded the file will appear in the IVR Prompt web page Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 78 of 138 istan innovative IP Voice amp Video LANGUAGE SETTINGS FOR VOICE PROMPT The UCM61xx supports multiple languages in web GUI as well as system voice prompt The following languages are currently supported in system voice prompt English Chinese German French Arabic Italian Spanish Polish Portuguese Russian English and Chinese voice prompts are built in with the UCM61 xx already The other languages provided by Grandstream can be downloaded and installed from the UCM61xx web GUI directly Additionally users could customize their own voice prompts package them and upload to the UCM61 xx Language settings for voice prompt can be accessed under Web GUI 2PBX Internal Options gt Language DOWNLOAD AND INSTALL VOICE PROMPT PACKAGE To download and install voice prompt package in different languages from UCM61xx web GUI click on Check Prompt List button Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 79 of 138 tream innovative IP Voice amp Video Language Settings Upload Voice Prompt Package D Choose Voice Prompt to Upload Upload Voice Prompt Package List Language ig English 5PH Check Prompt List Figure 32 Language Settings For Voice Prompt
58. Increase the volume of the conference call 7 Increase your volume More options 1 List all users currently in the conference call 2 Kick all non Administrator participants from the conference call 3 Mute Unmute all non Administrator participants from the conference call 4 Record the conference call 8 Exit the caller menu and return to the conference 1 Mute unmute yourself Decrease the volume of the conference call Decrease your volume Increase the volume of the conference call Increase your volume N O Ff Exit the caller menu and return to the conference RECORD CONFERENCE The UCM61xx allows users to record the conference call and retrieve the recording from web GUI gt PBX gt Call Features gt Conference To record the conference call when the conference bridge is in idle enable Record Conference from the conference bridge configuration dialog Save the setting and apply the change When the conference call starts the call will be automatically recorded in wav format Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 73 of 138 ndstream innovative IP Voice amp Video The recording files will be listed as below once available Users could click on to download the recording or click on delete the recording 6300 1372865271 25 2013 07 03 12 39 38 UTC 03 00 10 61 MB meetme confrec 6300 1372451238 6 wav 2013 06 28 17 2
59. NGUAGES Currently the UCM61xx web GUI supports the following languages English Chinese Spanish French Portuguese Russian Italian Polish German Users can select the displayed language in web GUI login page or at the upper right of the web GUI after logging Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 24 of 138 andstream innovative IP Voice amp Video Username English Password aitt EX Espa a English 4 ll France English EZ Espa a B France italiano Portugu s italiano Figure 10 UCM61xx Web GUI Language SAVE AND APPLY CHANGES Click on Save button after configuring the web GUI options in one page After saving all the changes make sure click on Apply Changes button on the upper right of the web page to submit all the changes If the change requires reboot to take effect a prompted message will pop up for you to reboot the device MAKE YOUR FIRST CALL Power up the UCM61xx and your SIP end point phone and connect both to network Then follow the steps below to make your first call 1 Login the UCM61xx web GUI go to PBX gt Basic Call Routes gt Extensions 2 Click on Create New User to create a new extension You will need User ID Password and Voicemail Password information to register and use the extension later 3 Register the extension on your phone
60. PS FIN TEC ION ririn n REA E E EE 57 VOIPTR NKS E E TE De S E E E eee ee 58 CALL ROUTES 64 OUTBOUND ROUT ES 64 POUT BS 9 66 INBOUND RULE CONFIGURATIONS 66 BLACKLIST CONFIGURATIONS 68 CONFERENCE BHIDOE 69 CONFERENCE BRIDGE CONFIGURATIONS 69 JOIN A CONFERENCE die 71 INVITE OTHER PARTIES JOIN CONFERENCE eser 71 Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 2 of 138 istan innovative IP Voice amp Video DURING THE CONFERENCE 72 RECORD 0N 0 ttt 73 MERE TL Tm 75 GONFIG RE AA 75 CREATE IVR 76 RECORD NEW IVR PROMPT ttt ttt ttt ttt 77 UPLOAD IVR PROMPT ttt ttt 77 LANGUAGE SETTINGS FOR VOICE PROMPT 79 DOWNLOAD AND INSTALL VOICE PROMPT PACKAGE 79 CUSTOMIZE AND UPLOAD VOICE PROMPT PACKAGE 81 VOICEMAIL 82 82 VOICEMAIL EMAIL SETTINGS ccccscsessssessssessssesssueesssessssetssessaressssesssteessseessnetssestesestaneetaseessneessnee
61. States of America USA Select country to set the On Hook Speed Ringer Impedance Ringer Threshold Current Limiting TIP RING voltage adjustment Minimum FXS Opermode Operational Loop Current and AC Impedance as predefined for your country s analog line characteristics The default setting is United States of America USA Select the impedance value for Two Wire Impedance Synthesis TISS TISS Override override Select the codec to be used for analog lines North American users should choose PCMU All other countries unless already Known should be assumed to be PCMA The default setting is PCMU PCMA Override Note This option requires system reboot to take effect Configure whether normal ringing voltage 40V or maximum ringing Boost Ringer voltage 89V for analog phones attached to the FXS port is required The default setting is Normal Fast Ringer Configure to increase the ringing speed to 25HZ This option can be Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 103 of 138 Low Power Ring Detect MWI Mode used with Low Power option The default setting is Normal Configure the peak voltage up to 50V during Fast Ringer operation This option is used with Fast Ringer The default setting is Normal 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 defaul
62. TTINGS GENER L IUE 105 DOCSETTINGS COC eee ese 105 SETTINGS REGISTRATION ccscecsesssesceesseessesseesseecasessessessneeasecaseaseeseeaneeatecateatesstesnteaeeeaseeees 106 OCS ETT INGS SAT CD EN Ses 107 ee Nac 108 SIP SETTINGS GENERAL 108 SIP SE TINGS OG eonebmso titu 109 scc 109 SIP SETTINGS SESSION TIMER uertice 110 SIPSETTINGS TCP and TES 110 SIP SETTINGS TOP and TIS aant eene 111 BIPSEPTINOSITOIS Ue 113 SIP SETTINGS BEBUG 114 STATUS AND HEPORUINGOL riui ineuntis eee 115 deci 115 TRONK cT 115 qid c 116 QUEUE 118 CONFERENCE ROOMS uiae tni Etats eluted iU MEM 119 INTERFACES STAFUS dium 119 acquc 120 cft aa M 121 121 IUe A O 122 STORAGE U AE 122 zisceiezic c 123 CDR CALL DETAIL REPORT epum 124 UPGRADING AND 127 Bcc 127 UPGRADING NETWORK essent tette tette ttt ttt 127 UPGRADING VIA LOCAL UPLOAD
63. The analog hardware FXS port and FXO port on the UCM61xx will be listed in this page Click on to edit signaling preference for FXS port or configure ACIM settings for FXO port Select Loop Start or Kewl Start for each FXS port And then click on Update to save the change Edit Analog Ports Signalling Preference Part 1 Loop Start I Port2 Kewl Start Figure 43 FXS Ports Signaling Preference For FXO port users could manually enter the ACIM settings by selecting the value from dropdown list for each port Or users could click on Detect for the UCM61xx to automatically detect the ACIM value The detecting value will be automatically filled into the settings Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 102 of 138 andstream innovative IP Voice amp Video ACIM Setting D ACIM Detection 1 6000 E Port 2 600 C Figure 44 FXO Ports ACIM Settings Table 40 Internal Options Hardware Config Select country to set the default tones for dial tone busy tone ring tone Tone Region and etc to be sent from the FXS port The default setting is United States of America USA Advanced Settings Select country to set the On Hook Speed Ringer Impedance Ringer Threshold Current Limiting TIP RING voltage adjustment Minimum FXO Opermode Operational Loop Current and AC Impedance as predefined for your country s analog line characteristics The default setting is United
64. The web GUI gives users access to all the configurations and options for UCM61xx setup This section provides step by step instructions on how to use the LCD menu LED indicators and Web GUI of the UCM61xx Once the basic settings are done users could start making calls from UCM61xx extension registered on a SIP phone as described at the end of this section USE THE LCD MENU e Default LCD Display By default when the device is powered up the LCD will show device model e g UCM6116 hardware version e g V1 5A and IP address Press Down button and the system time will be displayed as well e Menu Access Press OK button to start browsing menu options Please see menu options in Table 4 LCD Menu Options e Menu Navigation Press the Down arrow key to browser different menu options Press the OK button to select an entry Exit If Back option is available in the menu select it to go back to the previous menu For Device Info Network Info and Web Info which do not have Back option simply press the OK button to go back to the previous menu Also the LCD will display default idle screen after staying in menu option for 15 seconds e LCD Backlight The LCD backlight will be on upon key pressing The backlight will go off after the LCD stays in idle for 30 seconds The following table shows the LCD menu options Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 20 of 138 View Events Dev
65. a matching user or peer for the request This reduces the ability of an attacker to scan for valid SIP UCM6102 6104 6108 6116 USER MANUAL Page 109 of 138 yam innovative I usernames The default setting is No If enabled when the peer negotiates G726 32 audio the UCM61xx will Non Standard G 726 Support use AAL2 packing order instead of 551 packing order AAL2 G726 32 The default setting is SIP SETTINGS SESSION TIMER Table 47 SIP Settings Session Timer Select the session timer mode The default setting is Accept The options are Originate Always request run session timer Session Timers e Accept Run session timer only when requested by other UA e Refuse Do not run session timer Configure the maximum session refresh interval in seconds The Session Expires ME default setting is 1800 Min SE Configure the minimum session refresh interval in seconds The default setting is 90 Select the session refresher to be UAC or UAS The default setting is UAC Session Refresher SIP SETTINGS TCP and TLS Table 48 SIP Settings TCP and TLS Configure to allow incoming TCP connections with the UCM61xx The TCP Enable default setting is No Configure the IP address for TCP server to bind to 0 0 0 0 means TCP Bind Address binding to all interfaces The port number is optional If not specified 5060 will be used Configure to allow incoming TLS connections with the UCM
66. accessed via Web GUI gt PBX gt Call Features gt Call Queue e Click on Create New Queue to add call queue Agent Login Settings Call Queue Name Strategy 6500 TechSupport1 Linear Warehouse Ringall Sales Ringall TechSupport2 Least Recent Figure 40 Call Queue e Click on to edit the call queue The call queue configuration parameters are listed in the table below Table 34 Call Queue Configuration Parameters Extension Configure the call queue extension Name Configure the call queue name to identify the call queue Select the strategy for the call queue e Ring All Ring all available Agents simultaneously until one answers e Linear Ring agents in the specified order e Least Recent Ring the agent who has been called the least recently e Fewest Calls Ring the agent with the fewest completed calls e Random Ring a random agent e Round Robin Ring the agents in Round Robin scheduling with memory Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 89 of 138 Music Hold Leave When Empty Dial in Empty Queue Dynamic Login Password Time Out Wrapup Time Max Queue Length Firmware Version 1 0 1 22 m The default setting is Ring All Select the Music On Hold class for the call queue Note Music On Hold classes can be managed from Web GuUI gt PBX Internal Options gt Music On Hold Configure whether the callers will be disconnected from the queue or not
67. al The highest level required Only users with international level can use this rule Configure the password for users to use this rule when making outbound calls Send This Call Trough Trunk Use Trunk Strip Prepend Use Failover Trunk Select the trunk for this outbound rule 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 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 Failover Trunk Strip Prepend Firmware Version 1 0 1 22 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 via 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 VoIP trunk is not available The PSTN trunk can be configured as the failover trunk of the VoIP trunk Allows the user to specify the number of digits that will be stripped from the beginning o
68. analog lines phone and Fax to the FXS ports SAFETY COMPLIANCES The UCM61xx complies with FCC CE and various safety standards The UCM61xx power adapter is compliant with the UL standard Use the universal power adapter provided with the UCM61xx package only The manufacturer s warranty does not cover damages to the device caused by unsupported power adapters WARRANTY If the UCM61xx 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 our Technical Support Team 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 UCM61xx 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 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 19 of 138 istan innovative IP Voice amp Video GETTING STARTED The UCM61 xx provides LCD interface LED indication and web GUI configuration interface e The LCD displays hardware software and network information of the UCM61xx Users could also navigate in the LCD menu for device information and basic network configuration The LED indication at the front of the device provides interface connection and activity status e
69. arty during the conference call Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 69 of 138 Enable Caller Menu Record Conference Quiet Mode Wait For Admin Enable User Invite Announce Callers Enable Jitter Buffer Public Mode Firmware Version 1 0 1 22 innovative m Note If Public Mode is enabled the password is not required to join the conference bridge thus this field is invalid If enabled conference participant could press the key to access the conference bridge menu The default setting is No If enabled the calls in this conference bridge will be recorded automatically in a wav format file All the recording files will be displayed and can be downloaded in the conference web page The default setting is No If enabled if there are users joining or leaving the conference voice prompt or notification tone won t be played The default setting is No Note Quiet Mode and Announce Callers cannot be enabled at the same time If enabled the participants will not hear each other until the conference administrator joins the conference The default setting is No Note If Quiet Mode is enabled the voice prompt for Wait For Admin will not be announced If enabled users could press O to invite other users with the users permission or press 1 to invite other users without the user s permission to join the conference The default setting is No Note
70. audit eost 35 Figure 15 Edit LDAP PhornebDOOK dann taii shseezstdsaenoedasdistesnedaddsddanteacsadsdnesshseeieidsaeenedasdieeernenes 36 Figure 16 GXP2200 LDAP Phonebook Configuration cccccccescccceeeeseceeeeeeeeceeeseeeeeeeseaaeeeeessaeaeeesesaeaes 37 Figure 17 UCM61xx Zero Config 42 Figure 18 Auto Provision SETING S tp Fe 43 Figure 19 AVIO DISCOV ED 44 Figure 20 44 Figure 21 Assign Extension To De VICE cccccsscccessccceseecseeeccececsaeeeceueessaueecsuecesseeessueessaesessueesssatensees 45 Figure 22 Greate NEW De VIC 45 Figure 23 Provisioning Example 1 ccccccccsecceseseeeeseeeceeeeeseeeeneucessaeeeseaeeesaseessueeesaueesseeessaeeesaueeeseneesanees 46 Figure 24 Provisioning Example 2 ccccccccseccccssccceseecceeecseeeecsucessaueeseeesseeecsueessauecssueessaesessueessentessuees 47 Figure 25 PSTN Detection For Analog Trunnk cccccccseececceeeeeeeeeseeeseeeeeseeeeeesseceeesaaeeeessaeeeesaeeeeeseneeneas 57 Figure 26 Blacklist Configuration Parameters cccccccsecceceeeeceeeeeeeeseeeeeseeeeeesseeeeeeseeeessaeeessaeeeeseeneeeeas 68 Figure 27 Conference Invitation From Web GUI sseeesssssessssssseeeeneeeennen enne nnne nnns 71 Figure 28
71. boot the UCM61xx End users can also choose to download a free HTTP server from http httpd apache org or use Microsoft IIS web server BACKUP The UCM61xx configuration can be backed up locally or via network The backup file will be used to restore the configuration on UCM61xx when necessary LOCAL BACKUP 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 If the Config File is selected only the backup file will be saved in the flash of the UCM61 xx f Voice File Voicemail File Voice Records or CDR is selected external storage devices USB Flash drive or SD Card will be required because the backup file might be too large Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 130 of 138 tream innovative IP Voice amp Video Click on Create New Backup button to start backup Once the backup is done the list of the backups will displayed with date and time in the web page Users can download restore or delete U it from the UCM61 xx internal storage or the external device Manage Configuration Backups Backup Configuration Create New Backup Upload Backup File Config File Voice File Voicemail File Voice Records CDR List of Previous Configuration Backups backup 2013may14 232900 23 29 55 May 14 201
72. d 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 Can Reinvite 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 Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 51 of 138 DIMF Mode Insecure Enable Keep alive Keep alive Frequency ndstream innovative IP Voice amp Video 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 peers matching by IP address without matching port number e Very Allow peers matching by IP address without matching port number Also authentication of incoming INVITE messages is not required No Normal IP based peers matching and authentication of incoming INVITE The default setting is Port If enabled empty SDP packet will be sent to the SIP server periodically to keep the NAT port The default setting is Yes Configure the number of seconds for the host to be up for Kee
73. d rule to make outbound calls The default setting is No Select audio and video codec for the extension The available codecs Codec Preference are PCMU PCMA GSM G 726 G 722 G 729 G 723 ILBC ADPCM LPC10 H 264 H 263 and H 263p EDIT EXTENSION All the UCM61xx extensions are listed under Web GUI gt PBX gt Basic Call Routes gt Extensions with SIP status Extension CallerlD Name Technology IP and Port Each extension has a checkbox for users to select and options Edit Reboot Delete e SIP Status Users can see the following icon for each extension to indicate the SIP status Green Free a Blue Ringing Yellow In Use Grey Unavailable e Edit single extension Click on to start editing the extension The configuration options are listed in Table 20 Extension Configuration Parameters e Reboot the user Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 53 of 138 andstream innovative IP Voice amp Video Click on to send NOTIFY reboot event to the device with the extension registered To successfully reboot the user with the extension registered Zero Config needs to be enabled on the UCM61 xx web GUI gt PBX gt Basic Call Routes gt Zero Config gt Auto Provisioning Settings e Delete single extension Click on to delete the extension Or select the checkbox of the extension and then click on Delete Selected Extensions e Modify selected extensions Select the checkb
74. d to override the From Header For example trunk ucm61xx provider com is the From Domain in From Header sip 1234567 9trunk ucm61xx provider com Configure the actual user name of the extension This can be used to override the From Header There are cases where there is a single ID for registration single trunk with multiple DIDs For example 1234567 is the From User in From Header sip 1234567 9trunk ucm61xx provider com Select to enable outbound proxy in this trunk The default setting is No When outbound proxy support is enabled enter the IP address or URL of the outbound proxy for Register SIP Trunk type If enabled the UCM61 xx will regularly send SIP OPTIONS to the device to check if the device is still online The default setting is No Enable to detect Fax signal from the trunk during the call and send the received Fax to the default Email address in Fax setting page under web GUI gt PBX gt Internal Options gt Fax T 38 Note If enabled Fax Pass through cannot be used Enable SRTP for the VoIP trunk The default setting is Peer IAX Trunk Configuration Parameters Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 61 of 138 Provider Host Name Keep Trunk CID Caller ID CallerlD Name Codec Preference Enable Quality Fax Detection istan innovative IP Voice amp Video Configure the provider name for the VoIP trunk This is a unique label to
75. 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 duration in seconds of incoming outgoing registration The default setting is 120 Configure the maximum duration in seconds of incoming registration and subscription allowed by the UCM61xx The default setting is 3600 Configure the minimum duration in seconds of incoming registration and subscription allowed by the UCM61xx 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 and the peer channel placing the call on hold doesn t have Music On Hold Suggest Configure the Music On Hold class to suggest to the peer channel when placing the peer on hold Select to enable relaxed DTMF handling The default setting is No Select DTMF mode to send DTMF 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 The default setting is RFC2833 During an active call if there is no RTP activity within the timeout in seconds the call will be terminated The default setting is no timeout Note This setting doesn t a
76. dress Port Base DN User Name Password LDAP 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 192 168 40 50 389 dc pbx dc com AccountName 6805 dc pbx dc cc CallerlIDName AccountName AccountName CallerlDName 9oAccountName 50 0 Enable Enable Figure 16 GXP2200 LDAP Phonebook Configuration HTTP SERVER The UCM61xx embedded web server responds to HTTP HTTPS GET POST requests Embedded HTML pages allow the users to configure the PBX through a Web browser such as Microsoft s IE Mozilla Firefox and Google Chrome By default the PBX can be accessed via HTTPS using Port 8089 e g https 192 168 40 50 8089 Users could also change the access protocol and port as preferred under Web GUI gt Settings gt HTTP Server Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 37 of 138 Redirect From Port 80 Protocol Type Port yam innovative Table 16 HTTP Server Settings 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 is enabled the access using HTTP with Port 80 will be redirected to HTTPS with Port 8089 The default setting is
77. e UCM6102 6104 6108 6116 USER MANUAL Page 90 of 138 andstream innovative IP Voice amp Video If enabled the UCM61xx will report to the agent the duration of time of Report Hold Time the before the caller is connected to the agent If enabled users will be disconnected after the configured number of seconds The default setting is No Wait Time Note It is recommended to configure Wait Time longer than the Wrapup Time Select the available agents from the available users on the right to the Static Agents l l static agents list on the left Click on to arrange the order to delete the call queue e Click on Agent Login Settings to configure Agent Login Extension Postfix and Agent Logout Extension Postfix Once configured users could log in the call queue as dynamic agent Agent Login Settings Agent Login Settings D Agent Login Extension Postfix D Agent Logout Extension Postfix Example For example if the call queue extension is 6500 Agent Login Extension Postfix is and Agent Logout Extension Postfix is users could dial 6500 to login to the call queue as dynamic agent and dial 6500 to logout from the call queue Dynamic agent doesn t need to be listed as static agent and can log in log out at any time e queue feature code Agent Pause and Agent Unpause can be configured under Web GUI gt PBX gt Internal Options gt Feature Codes
78. e Detection will keep the call up for about 1 minute If yau have selected Semi auto Detect please pick up the phone only after you are informed Figure 25 PSTN Detection For Analog Trunk Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 57 of 138 innovative IP Voice amp Video Table 23 PSTN Detection For Analog Trunk Select Auto Detect or Semi auto Detect for PSTN detection e Auto Detect Please make sure two or more channels are connected to the UCM61xx and in idle status before starting the detection During the detection one channel will be used as caller Source Channel and another channel will be used as callee Destination Channel The UCM61xx will control the call to be established and hang up between caller and callee to finish the detection Detect Model Semi auto Detect Semi auto detection requires answering or hanging up the call manually Please make sure one channel is connected to the UCM61xx and in idle status before starting the detection During the detection source channel will be used as caller and send the call to the configured Destination Number Users will then need follow the prompts in web GUI to help finish the detection The default setting is Auto Detect Source Channel Select the channel to be detected Destination Channel Select the channel to help detect when Auto Detect is used Configure the number to be called to help detect when Semi auto Desti
79. e Ringing Yellow In Use Grey Unavailable Extension Name Label Display name callerlD name or label for the extension Display message status for the extension Example 2 4 1 Description There are 2 urgent messages 4 messages in total and 1 message that has been already read Status Displays extension type e SIP User Type AX User e Analog User e Features Other operations are also available in extension status section 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 UCM6102 6104 6108 6116 USER MANUAL Page 117 of 138 Firmware Version 1 0 1 22 stream innovative IP Voice amp Video e Click on to refresh the extension status gt e Click on one of the tabs Al Anal Festes 88 to display the corresponding extensions accordingly e Click on to expand the status detail table Click on to hide the status detail table QUEUES Users could see all the configured call queue status in this section The following figure shows the call queue 6500 being in used E Service Level 510 096 within 0s Calls Completed 1 Calls Abandoned 1 Figure 48 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 def
80. e Ring Group Firmware Version 1 0 1 22 Page Intercom Group UCM6102 6104 6108 6116 USER MANUAL Page 67 of 138 andstream innovative IP Voice amp Video BLACKLIST CONFIGURATIONS In the UCM61xx Blacklist is supported in all inbound routes Users could enable the Blacklist feature manage the Blacklist by clicking on Blacklist Blacklist or all inbound routes D Blacklist Enable Blacklist Manage D Blacklist list 1234567 12345678 wm Total 2 D Add Blacklist Number C Figure 26 Blacklist Configuration Parameters e Select the checkbox for Blacklist Enable to turn on Blacklist feature for all inbound routes Blacklist is disabled by default e Enter a number in Add Blacklist Number field and then click to add to the list e To remove a number from the Blacklist select the number in Blacklist list and click on Note Users could also add a number to the Blacklist or remove a number from the Blacklist by dialing the feature code for Blacklist and Blacklist Remove from an extension The feature code can be configured under Web GUI 2PBX Internal Options gt Feature Codes Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 68 of 138 innova tive IP Voice CONFERENCE BRIDGE The UCM61xx supports conference bridge allowing multiple bridges used at the same time UCM6102 6104 supports up to 3 conference bridges allowing up to 25 simultaneo
81. e boots up it sends out SUBSCRIBE to a multicast IP address in the LAN The UCM61xx discovers and then sends a NOTIFY with the XML config file URL in the message body The phone will then use the path to download the config file generated in the UCM61xx and reboot again to take the new configuration 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 for example http 192 168 2 1 8089 zccgi The phone will then use the path to download the config file generated in the UCM61 xx e mDNS When the phone boots up it sends out mDNS query to get the TFTP server address The UCM61 xx will respond with its own address The phone will then send TFTP request to download the XML config file from the UCM61 xx Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 42 of 138 andstream innovative IP Voice amp Video To start the auto provisioning process under Web GUI gt PBX gt Basic Call Routes gt Zero Config click on Auto Provision Settings and fill in the auto provision information Auto Provision Settings Enable Zero Config D Automatically Assign Extension 1 start Extension 6000 1 Generate Random Password
82. e code during active call After hearing Transfer you will hear the dial tone Enter the number to transfer to and the user will be connected to this number Hang up the call to complete the attended transfer Attended Transfer e Options Disable Allow Caller Enable the feature code on caller side only Allow Callee Enable the feature code on callee side only Allow Both Enable the feature code on both caller and callee e Default code 72 e Enter the code during active call to park the call e Options Call Park Disable Allow Caller Enable the feature code on caller side only Allow Callee Enable the feature code on callee side only Allow Both Enable the feature code on both caller and Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 95 of 138 ndstream innovative IP Voice amp Video callee e Default code 3 e Enter the code followed by or SEND to start recording the audio call and the UCM61xx will mix the streams natively on the fly as the call is in progress e Options Disable Allow Caller Enable the feature code on caller side only Allow Callee Enable the feature code on callee side only Allow Both Enable the feature code on both caller and Audio Mix Record callee DND Call Forward Do Not Disturb DND Activate e Default code 77 Do Not Disturb DND Deactivate e Default code 78 e Default Code 90 Call Forward Busy Activate e Enter the code and follow the voice pr
83. egistration The default setting is 60 Configure the maximum period in seconds of registration The default setting is 3600 Configure the number of helper threads The default setting is 10 Configure the maximum number of IAX threads allowed The default setting is 100 If set to yes the connection will be terminated if ACK for the NEW message is not received within 2000ms Users could also specify number in milliseconds in addition to yes and no The default setting is yes If enabled authentication traffic in debugging will not show The default setting is No Configure codec negotiation priority The default setting is Regonly e Caller Consider the callers preferred order ahead of the host s e Host Consider the host s preferred order ahead of the caller s e Disabled Disable the consideration of codec preference all together e Regonly This is almost the same as Disabled except when the requested format is not available The call will only be accepted if the requested format is available Configure ToS bit for preferred IP routing Configure the frequency of trunk frames in milliseconds The default UCM6102 6104 6108 6116 USER MANUAL Page 106 of 138 Trunk Time Stamps siren innova tive IP Voice setting is 20 If enabled time stamps will be attached to trunk frames The default setting is No IAX SETTINGS STATIC DEFENSE Call Token Optional Max Call Numbers Max Nonvalida
84. en innova tive IP Voice number rule name action protocol type source destination and operation Users can click on to edit the rule or select to delete the rule DYNAMIC DEFENSE The UCM61xx supports firewall dynamic defense that can blacklist hosts dynamically It monitors the traffic coming into the UCM61xx and helps prevent massive connection attempts or brute force attacks to the device The blacklist can be created and updated by the UCM61 xx firewall which will then be displayed in the web page Please refer to the following table for dynamic defense options on UCM61 xx Table 15 Firewall Dynamic Defense Dynamic Defense Enable dynamic defense on UCM61 xx firewall Enable Configure the dynamic defense periodic time interval in minutes If the number of TCP connections from a host exceeds the connection threshold within this period this host will be added into Blacklist The valid value is between 1 to 59 when dynamic defense is turned on Periodical Time Blacklist Update Configure the blacklist update time interval in seconds The default setting is Interval 120 seconds Connection Configure the connection threshold Once the number of connections from Threshold the same host reaches the threshold it will be added into the blacklist Dynamic Defense un Configure the dynamic defense whitelist Whitelist CHANGE PASSWORD After login the Web GUI for the first time it is highly recommended for users to change
85. ensions The default setting is No If enabled all callers to the IVR is allowed to use trunk The permission must be configured for the users to use the trunk first The default setting is No Assign permission level for outbound calls The available permissions are Internal Local National and International from the lowest level to the highest level The default setting is Internal Note Users need to have the same level as or higher level than a outbound rule s privilege in order to make outbound calls from this rule Select an audio file to play as the welcome prompt for the IVR Click on Prompt to add additional audio file under web GUI Internal Options gt IVR Prompt After playing the prompts in the IVR the UCM61xx will wait for the DIMF entry within the timeout in seconds If no DTMF entry is detected within the timeout a timeout prompt will be played The default setting is 10 seconds Select the prompt message to be played when timeout occurs Select the prompt message to be played when an invalid extension is UCM6102 6104 6108 6116 USER MANUAL Page 75 of 138 andstream innovative IP Voice amp Video pressed Configure the number of times to repeat the prompt if no DTMF input is Timeout Repeat Loops detected When the loop ends it will go to the timeout destination if Invalid Repeat Loops Key Press Event configured or hang up The default setting is 4 Configure the number of times to
86. enter the Audio Mix Record feature code by default it s 3 followed by or SEND again Or the recording will be stopped once the call hangs up e The recording file can be retrieved under Web GUI gt Status gt CDR Click on to play the recording or click on to download the recording file View Report Click on the title of the column to sort by column Click on the row to display full record View 10 Start Time 7 1 2013 07 03 17 55 04 6000 5001 0 00 18 0 00 16 2 2013 07 03 17 54 32 6000 5001 0 00 19 0 00 18 3 2013 07 03 17 53 11 6000 6300 0 00 11 0 00 11 Figure 42 Download Recording File From CDR Page CALL PARK The UCM61 xx provides call park and call pickup features via feature code PARK A CALL There are two feature codes that can be used to park the call e Feature Maps gt Call Park Default code 72 During an active call press 72 and the call will be parked Parking lot number default range 701 to 720 will be announced after parking the call e Feature Misc gt Call Park Default code 700 During an active call initiate blind transfer default code 1 and then dial 700 to park the call Parking lot number default range 701 to 720 will be announced after parking the call RETRIEVE THE PARKED CALL To retrieve the parked call simply dial the parking lot number and the call will be established If a parked is not retrieved after the timeout the original extension who parks the call wi
87. er the DNS server 2 address for static IP settings Enter the user name to connect via PPPoE Enter the password to connect via PPPoE LAN when Method is set to Route IP Address Subnet Mask DHCP Server Enable DNS Server 1 DNS Server 2 Enter the IP address assigned to LAN port The default setting is 192 168 2 1 Enter the subnet mask The default setting is 255 255 255 0 Enable or disable DHCP server capability The default setting is Yes Enter DNS server address 1 The default setting is 8 8 8 8 Enter DNS server address 2 The default setting is 208 67 222 222 Allow IP Address From Enter the DHCP IP Pool starting address The default setting is 192 168 2 100 Allow IP Address To Enter the DHCP IP Pool ending address The default setting is 192 168 2 254 Default IP Lease Time Enter the IP lease time in seconds The default setting is 43200 LAN when Method is set Switch IP Method IP Address Gateway IP Subnet Mask DNS Server 1 DNS Server 2 User Name Password Select DHCP Static IP or PPPoE The default setting is DHCP Enter the IP address for static IP settings The default setting is 192 168 0 160 Enter the gateway IP address for static IP settings The default setting is 0 0 0 0 Enter the subnet mask address for static IP settings The default setting is 255 255 0 0 Enter the DNS server 1 address for static IP settings The default setting is 0 0 0 0 Enter the DNS server 2 add
88. even 8 Please note that the higher the number the more time is needed to hangup the channel However this might lower the probability to get random hangup Congestion detection is used to detect far end congestion signal The Congestion Detection ase default setting is ON l If Congestion Detection is enabled users can specify the number of Congestion Count OM congestion tones to wait for The default setting is 2 Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 55 of 138 Enable Polarity Reversal Polarity on Answer Delay Current Disconnect Threshold ms Ring Timeout RX Gain TX Gain Use CallerlD Fax Detection Caller ID Scheme Tone Country Busy Tone Firmware Version 1 0 1 22 m Innovative If enabled a polarity reversal will be marked as received when an outgoing call is answered by the remote party For some countries a polarity reversal is used for signaling the disconnection of a phone line and the call will be considered as hangup on a polarity reversal The default setting is No When FXO port answers the call FXS may send a Polarity Reversal If this interval is shorter than the value of Polarity on Answer Delay the Polarity Reversal will be ignored Otherwise the FXO will onhook to disconnect the call The default setting is 600ms This is the periodic time in ms that the UCM61 xx will use to check on a voltage drop in the line The default setti
89. f 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 Specify the digits to be prepended before the call is placed via the trunk UCM6102 6104 6108 6116 USER MANUAL Page 65 of 138 istan innovative IP Voice amp Video 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 Click on Blacklist do configure blacklist e Click on to edit the inbound route e Click on to delete the inbound route INBOUND RULE CONFIGURATIONS Table 26 Inbound Rule Configuration Parameters Trunks Select the trunk to configure the inbound rule e patterns are prefixed with e Special characters X Any Digit from 0 9 Z Any Digit from 1 9 N Any Digit from 2 9 DID Pattern Wildcard Match one or more characters I Wildcard Match zero or more characters immediately Example 12345 9 Any digit from 1 to 9 Select privilege level for the inbound rule e Internal The lowest level required All users can use this rule
90. ge Users could click on of the desired backup file and it will be restored to the UCM61xx Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 132 of 138 tream innovative IP Voice amp Video f users have other backup files on PC to restore on the UCM61xx click on Upload Backup File first and select it from local PC to upload on the UCM61xx Once the uploading is done this backup file will be displayed in the list of previous configuration backups for restore purpose Click on E to restore from the backup file Create New Backup li Upload Backup File Config File Voice File Voicemail File Voice Records CDR List of Previous Configuration Backups backup_2013may14_232900 23 29 55 May 14 2013 backup 2013mar26 180249 18 02 51 Mar 26 2013 Figure 64 Restore UCM61xx From Backup File Note e The uploaded backup file must be a tar file with no special characters like 4 amp space in the file name e he uploaded back file size must be under 10MB 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 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 133 of 138 CDR Cleaner andstream innovative IP Voice amp Video i Enable CDR Cleaner i CDR Clean Time 1 Clean Interval Voice Records Cleaner
91. gure the resync threshold for jitter buffer When the jitter buffer notices a significant change to delay that continues over a few frames it will resync assuming that the change in delay is caused by a time stamping mix up The threshold for noticing a change in delay is calculated as twice the measured jitter plus this resync threshold The default setting is 1000 Configure the jitter buffer implementation on the sending side of a SIP channel The default setting is Fixed Fixed The size is always equal to the value of Max Jitter Buffer e Adaptive The size is adjusted automatically and the maximum value equals to the value of Max Jitter Buffer Analog Jitter Buffer Enable Jitter Buffer Force Jitter Buffer Log Frames Max Jitter Buffer Firmware Version 1 0 1 22 Select to enable jitter buffer on the receiving side of the analog channel The default setting is Yes Select to force the use of jitter buffer on the receiving side of the SIP channel The default setting is Yes Select to enable jitter buffer frame logging The default setting is No Configure the maximum time in ms to buffer for Adaptive jitter buffer UCM6102 6104 6108 6116 USER MANUAL Page 100 of 138 C istan innovative IP Voice amp Video implementation or used as the jitter buffer size for Fixed jitter buffer implementation The default setting is 200 Configure the resync threshold for jitter buffer When the jitter buffer n
92. he scan method and scan IP The IP address segment will be automatically filled in based on the network mask detected on the UCM61xx If users need scan the entire network segment enter 255 for example 192 168 40 255 instead of a specific IP address Then click on Save to start discovering the devices within the same network Auto Discover The PBX can automatically discover the new devices by ARP or PING It can scan the entire network segment a single IP address Scan Method Ping D Scan IP 192 168 40 Cancel Save Figure 19 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 000B823E1D8D 192 168 40 249 Grandstream GXP2200 Connected 000B823E1D7C 192 168 40 122 Grandstream GXP2200 Connected 000B823E5E88 192 168 40 207 Grandstream GXP2124 Connected 000B823E175D 192 168 40 145 Grandstream GXP2200 Connected 000B823E1D7F 192 168 40 163 Grandstream GXP2200 Connected Figure 20 Discovered Devices Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 44 of 138 andstream innovative IP Voice amp Video ASSIGNMENT In the discovered list click on d to open the edit dialog to assign an extension to this device Edit Device 000B823E1D7C D Mac Address 000B823E1D7C 1 IP Address 182 168 40 122 1 Exte
93. he uploaded certificate file must be under 2MB TLS CA List Display a list of files under the CA Cert directory SIP SETTINGS TCP and TLS Table 49 SIP Settings NAT Configure a static address and port optional that will be used in External IP Address outbound SIP messages if the UCM61xx is behind NAT If it s a hostname it will only be looked up once Ext anes Specify an external host name which is similar to External Address xternal Hos except the host name will be looked up periodically based on the Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 111 of 138 External Refresh External TCP Port External TLS Port Local Network Address NAT Mode Allow RTP Reinvite Firmware Version 1 0 1 22 m Innovative External Refresh interval Configure the refresh interval for the external host if used The default setting is 10 Configure the externally mapped TCP port when the UCM61 xx is behind a static NAT or PAT Configures the externally mapped TLS port when UCM61xx is behind a static NAT or PAT Specify a list of network addresses that are considered inside of the NAT network Multiple entries are allowed If not configured the external IP address will not be set correctly A sample configuration could be as follows 192 168 0 0 16 This is a global NAT setting that will affects all peers and users The default setting is Force rport Use rport if the remote side re
94. hows up the IP address Optional Connect PSTN lines from the wall jack to the FXO ports connect analog lines phone and Fax to the FXS ports l SD Card Slot 2xFXSPort 8xFXOPort LED Indicators LCD EU Figure 5 UCM6108 Front View DC 12v Reset LAN Port Ground Figure 6 UCM6108 Back View CONNECT THE UCM6116 1 r LhiFiag amp saacamga m L l SD Card Slot 2 FXS Port 16 Port LED Indicators LCD Figure 7 UCM6116 Front View DC 12v Reset LAN Port Ground Figure 8 UCM6116 Back View Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 18 of 138 istan innovative IP Voice amp Video To set up the UCM6116 follow the steps below 1 Connect one end of an 45 Ethernet cable into the LAN port of the UCM6116 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 the device hardware information when the boot process is done 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
95. hrough 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 6104 6108 6116 it will be sure to bring convenience and color to both your business and personal life Asterisk is a Registered Trademark of Digium Inc Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 138 of 138
96. ice Info Network Info Network Menu Factory Menu Firmware Version 1 0 1 22 m Table 4 LCD Menu Options e Critical Events e Other Events e Hardware Hardware version number e Software Software version number e P N Part number e WAN MAC WAN side MAC address UCM6102 only e LAN MAC LAN side MAC address e Uptime System up time For UCM6104 UCM6108 UCM6116 e LAN Mode DHCP Static IP or PPPoE e LAN IP IP address e LAN Subnet Mask For UCM6102 e WAN Mode DHCP Static IP or PPPoE e WAN IP IP address e WAN Subnet Mask e LAN IP IP address e LAN Subnet Mask For UCM6104 UCM6108 UCM6116 e LAN Mode Select LAN mode as DHCP Static IP or PPPoE For UCM6102 e WAN Mode Select WAN mode as DHCP Static IP or PPPoE e Reboot e Factory Reset e LCD Test Patterns Press Down button to test different LCD patterns When done press OK button to exit Fan Mode Select Auto or On e LED Test Patterns UCM6102 6104 6108 6116 USER MANUAL Page 21 of 138 yam innovative Select All On All Off or Blinking and check LED status e RTC Test Patterns Select 2022 02 22 22 22 or 2011 01 11 11 11 to start the RTC Real Time Clock test pattern Then check the system time from LCD idle screen by pressing DOWN button or from web GUI gt System Status gt General page Reboot the device manually after the RTC test is done e Hardware Testing Select Test SVIP to perform SVIP te
97. ices in a managed way To provision a phone three steps are involved i e discovery assignment and provisioning The UCM61 xx creates XML config file to the detected assigned Grandstream device and accomplishes the following configurations on the device after the provisioning An UCM61xx extension will be assigned and registered on the phone SlP related network settings such as NAT traversal and Use Random Port are configured on the phone e Call settings such as Dial Plan and Auto Answer DAP client configurations will be set up automatically on the phone to use the default LDAP directory generated in the UCM61xx LDAP server This section explains how zero config works on the UCM61xx 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 disabled on the UCM61xx for auto provisioning It can be turned on in Auto Provision Settings under Web GUI gt PBX gt Basic Call Routes gt Zero Config Three methods of auto provisioning are used Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 41 of 138 ndstream innovative IP Voice amp Video SIP End Device Discover Device Assign Extension to Device Create XML Contig File Download Config File Send Downloading URL to Device Reboot Cei Provisioned Figure 17 UCM61 xx Zero Config e SIP SUBSCRIBE When the phon
98. ide the conference room details INTERFACES STATUS This section displays interface port connection status on the UCM61xx The following example shows the interface status for UCM6116 with USB SD card LAN port and FXS1 connected Interfaces Status gt SD Figure 50 UCM6116 Interfaces Status Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 119 of 138 ndstream innovative IP Voice amp Video Table 55 Interface Status Indicators H USB connected bd USB disconnected 50 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 3 to refresh the interface status Click on to expand the interface details e Click on to hide the interface details PARKING LOT The UCM61xx supports call park using feature code When there is call being parked this section will display the parking lot status Parking Lot 5 SIP 6070 00000050 SIP 6005 00000052 Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 120 of 138 ndstream innovative IP
99. ill 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 UCM61xx 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 PCMU and PCMA are required When Auto is selected RFC2833 will be used if offered otherwise Inband will be used e Port Allow peers matching by IP address without matching port number e Very Allow peers matching by IP address without matching port number Also authentication of incoming INVITE messages is not required No Normal IP based peers matching and authentication of incoming INVITE The default setting is Port If enabled empty SDP packet will be sent to the SIP server periodically to keep the NAT port The default setting is Yes Configure the Keep alive interval in seconds to check if the host is up UCM6102 6104 6108 6116 USER MANUAL Page 49 of 138 ndstream innovative IP Voice amp Video Other Settings SRTP Enable SRTP for the call The default setting is disabled Enable to detect Fax signal from the user trunk during the call and send the received Fax to the Email address configured for this extension If no Email address can be found for the user se
100. ined as below Table 54 Agent Status The agent is available idle The agent is talking busy The agent is ringing The agent has been logged out On the UCM61xx 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 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 O to refresh the call queue status Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 118 of 138 stream e Click on to expand the call queue detail 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 call duration for each user and conference call Conference Rooms 5 6300 3 Users 6301 Notin Use Figure 49 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 to refresh the conference room status Click on to expand the conference room details Click on to h
101. ing that the change in delay is caused by a Resync Threshold time stamping mix up The threshold for noticing a change in delay is calculated as twice the measured jitter plus this resync threshold If the interval is longer than the resync threshold time resync the jitter buffer The default setting is 1000 Configure the maximum amount of excess jitter buffer in milliseconds Max Excess Buffer to pad to the jitter buffer before the jitter buffer is slowly shrunk to eliminate latency Configure the minimum amount of excess jitter buffer in milliseconds to Min Excess Buffer pad to the jitter buffer before the jitter buffer to slowly raised to eliminate latency Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 101 of 138 andstream innovative IP Voice amp Video Jitter Shrink Rate Configure the rate at which the jitter buffers are increased or decreased INTERNAL OPTIONS RTP SETTINGS Table 39 Internal Options RTP Settings RTP Start Configure the RTP port starting number The default setting is 10000 RTP End Configure the RTP port ending address The default setting is 20000 Configure to enable or disable strict RTP protection If enabled RTP Strict RTP packets that do not come from the source of the stream will be dropped The default setting is Disable Configure to enable or disable RTP Checksums on RIP traffic The RTP Checksums D default setting is Disable INTERNAL OPTIONS HARDWARE CONFIG
102. irewall settings The following table shows a sample current service status running on UCM61xx Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 30 of 138 7777 389 22 80 8089 69 9090 6060 2060 4569 9353 37435 innovative Table 12 UCM61xx Firewall gt Static Defense gt Current Service Protocol or Service asterisk slapd dropbear lighthttpd lighthttpd opentftpd asterisk zero config asterisk asterisk zero config syslogd tcp IPv4 tcp IPv4 tcp IPv4 tcp IPv4 tcp IPv4 udp IPv4 udp IPv4 udp IPv4 udp IPv4 udp IPv4 udp IPv4 udp IPv4 SIP LDAP SSH HTTP HTTPS SIP UCM61xx zero_config service SIP SIP UCM61xx zero_config service Syslog For typical firewall settings users could configure the following options on the UCM61 xx Ping Defense Enable SYN Flood Defense Enable Death of Ping Defense Enable Table 13 Typical Firewall Settings If enabled ICMP response will not be allowed for Ping request The default setting is disabled To enable or disable it click on the check box for the LAN or WAN UCM6102 only interface Enable to prevent SYN Flood denial of service attack to the device The default setting is disabled To enable or disable it click on the check box for the LAN or WAN UCM6102 only interface Enable to prevent Death of Ping attack to the device The default setting Is disabled To enable or disable i
103. is in 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 D Target Host www google com Output Result b4 bytes 4 125 224 1 9 seq 1 ttl 53 time 13 500 ms 64 bytes from 74 125 224 179 seq 2 ttl 53 time 19 300 ms 64 bytes from 74 125 224 179 seq 3 ttl 53 time 13 800 ms 64 bytes from 74 125 224 179 seq 4 ttl 53 time 13 825 ms 64 bytes from 74 125 224 179 seq 5 ttl 53 time 13 950 ms 64 bytes from 74 125 224 179 seq 6 ttl 53 time 14 125 ms 64 bytes from 74 125 224 179 seq ttl 53 time 17 425 ms 64 bytes from 74 125 224 178 seq 8 ttl 53 time 13 800 ms 64 bytes from 74 125 224 179 seq 8 ttl 53 time 13 675 ms 64 bytes from 74 125 224 178 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 seq 12 ttl 53 time 14 025 ms 64 bytes from 74 125 224 178 seq 13 ttl 53 time 14 150 ms 64 bytes from 74 125 224 179 seq 14 ttl 53 time 13 900 ms www google com ping statistics 15 packets transmitted 15 packets received 096 packet loss round trip min avg max 13 800 14 599 19 300 ms Done Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 13
104. is recommended to use the random generated password for security purpose Email Address Fill in the Email address for the user Configure the Call Forward Unconditional target number If not Call Forward Unconditional a configured the Forward Unconditional feature is deactivated Call Forward No Answer Configure the Call Forward No Answer target number If not configured Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 48 of 138 Call Forward Busy Ring Timeout Technology ndstream 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 the call is forwarded to voicemail voicemail is enabled or hang up voicemail is disabled SIP Analog Station Select SIP if the user is using SIP or a SIP device Select if the user is using or IAX device Select the FXS port if the user is attached on the analog port of the UCM61 xx NAT Can Reinvite DIMF Mode Insecure Enable Keep alive Keep alive Frequency Firmware Version 1 0 1 22 Use NAT when the UCM61xx 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 UCM61xx w
105. ister to the trunk from the provider when sername Register SIP Trunk or Register IAX Trunk type is selected Enter the password to register to the trunk from the provider when Password Register SIP Trunk or Register Trunk type is selected Enter the IP address or URL of the outbound proxy for Register SIP Outbound Proxy Trunk type Peer SIP Trunk Configuration Parameters Configure the provider name for the VoIP trunk This is a unique label to Provider Name identify the trunk when listed in outbound rules inbound rules and etc Configure the IP address or URL for the VoIP provider server of the Host Name Configure the SIP transport protocol to be used in this trunk The default setting is All UDP Primary e UDP Only e CP Only Transport LS Only All UDP Primary UDP is the primary transport protocol when the other SIP transport methods are available too All TCP Primary TCP is the primary transport protocol when all the other SIP transport methods are available too Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 59 of 138 Keep Trunk CID Caller ID CallerlD Name Codec Preference Enable Qualify Fax Detection SRTP istan Innovative IP Voice amp Video All TLS Primary TLS is the primary transport protocol when all the other SIP transport methods are available too If enabled the trunk CID will not be overridden by extensio
106. ive IP Voice amp Video PRODUCT OVERVIEW FEATURE HIGHTLIGHTS e 1GHz ARM Cortex 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 with lifeline capability in case of power outage and up to 50 SIP trunk options e Gigabit network port s 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 1 G 726 G 729A B iLBC GSM video codec including H 264 H 263 H 263 and Fax T 38 e Hardware DSP based 128ms tail length carrier grade line echo cancellation LEC e Supports up to 500 SIP endpoint registration 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 both with lifetime capability in case of power outage e UCM6102 2 ports UCM6104 4 ports e UCM6108 8 ports e UCM6116 16 ports e UCM6108 6116 Single 10M 100M 1000M RJ45 Ethernet port
107. l address To do this users could turn on Fax Detection for a specific VoIP trunk under UCM61xx web GUI gt PBX gt Basic Call Routes VoIP Trunks Or users can set up the extension for Fax and then configure PBX gt Call Features gt IVR gt Key Pressing Events to have the key pressing event go to the extension of the Fax Fax T 38 settings can be accessed Web GUI gt PBX gt Internal Options gt FAX T 38 CONFIGURE 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 Fax Settings to configure the Fax parameters Table 35 FAX T 38 Settings Enable Error Correction Mode ECM Configure to enable Error Correction Mode ECM for the Fax Configure the maximum transfer rate during the Fax rate negotiation Maximum Transfer Rate The possible values are 2400 4800 7200 9600 12000 and 14400 The default setting is 14400 Configure the minimum transfer rate during the Fax rate negotiation The Minimum Transfer Rate possible values are 2400 4800 7200 9600 12000 and 14000 The default setting is 2400 Configure the Email address to send the received Fax to if user s Email address cannot be found Default Email Address Note The extension s Email address or the Fax s default Email address needs to be configured in order to receive Fax from Email If neither of them is configured Fax will be not be received from Email
108. l message The default setting is No If enabled the message duration will be announced at the beginning of the voicemail message The default setting is No If enabled a brief introduction received time received from and etc of each message will be played when accessed from the voicemail application The default setting is Yes If enabled users can review the message following the IVR before sending the message out The default setting is No UCM6102 6104 6108 6116 USER MANUAL Page 82 of 138 andstream innovative IP Voice amp Video VOICEMAIL EMAIL SETTINGS The UCM61xx can be configured to send the voicemail as attachment to Email Click on Email Settings For Voicemails button to configure the Email attributes and content Voicemail Email Settings F Attach Recordings to E mail Template for Voicemail Emails Template Variables com p Subject New voicemail from CALLERID for MAILBOX Message Hello NAME you received a message lasting VM DUR at CALLERID This is message 3VM_MSGNUM in your voicemail Inbox Load Default Settings Figure 34 Voicemail Email Settings Table 31 Voicemail Email Settings If enabled voicemails will be sent to user s Email address The default Attach Recordings to E Mail EE setting is Yes Fill in the Subject and content to be used in the Email when sending to the users The
109. led and no checksums will be calculated checked on systems supporting this features The default setting is No If enabled the IAX2 will delay the rejection of calls to avoid DOS The default setting is No Select to enable ADSI phone compatibility The default setting is No Specify which Music On Hold class this channel would like to listen to when being put on hold This music class is only effective if this channel has no music class configured and the bridged channel putting the call on hold has no Music On Hold Suggest setting Specify which Music Hold class to suggest to the bridged channel when putting the call on hold Configure the bandwidth for IAX settings The default setting is Low The following codes are supported in UCM61xx for IAX Select the codecs from the right side list to the left Tm TA a y in side Click on 2 7 1 to arrange the order PCMU POMA e GSM e ILBC G 722 e G 726 Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 105 of 138 e ADPCM LPC10 G 729 G 723 e H 263 263 H 264 yam innovative IAX SETTINGS REGISTRATION Min Reg Expire Max Reg Expire IAX Thread Count IAX Max Thread Count Auto Kill Authentication Debugging Codec Priority Type of Service Trunk Frequency Firmware Version 1 0 1 22 Table 43 IAX Settings Registration Configure the minimum period in seconds of r
110. ll be called back Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 98 of 138 ndstream innovative IP Voice amp Video INTERNAL OPTIONS This section describes internal options that haven t been mentioned in previous sections yet The settings in this section can be applied globally to the UCM61xx including general configurations jitter buffer RTP settings hardware config and STUN monitor The options can be accessed Web GUI gt PBX gt Internal Options INTERNAL OPTIONS GENERAL Table 37 Internal Options General General Preferences Global OutBound CID Global OutBound CID Name Operator Extension Ring Timeout Configure the global CallerlD used for all outbound calls when no other CallerlD is defined with higher priority If no CallerlD is defined for extension or trunk the global outbound CID will be used as CallerlD Configure the global CallerlD Name used for all outbound calls If configured all outbound calls will have the CallerlD Name set to this name If not the extension s CallerlD Name will be used Specify the operator extension which will be dialed when users presses 0 to exit voicemail application The operator extension can also be used in IVR option Configure the number of seconds to ring an extension before the call goes to the user s voicemail box The default setting is 60 Extension Preferences Enable Random Password Disable Extension Range Firmware Version 1
111. n Fax setting page under web GUI gt PBX gt Internal Options gt Fax T 38 Note If enabled Fax Pass through cannot be used Register Trunk Configuration Parameters Provider Name Host Name Keep Trunk CID Firmware Version 1 0 1 22 Configure the provider name for the VoIP trunk This is a unique label to identify the trunk when listed in outbound rules inbound rules and etc Configure the IP address or URL for the VoIP provider server of the trunk If enabled the trunk CID will not be overridden by extension s CID when UCM6102 6104 6108 6116 USER MANUAL Page 62 of 138 Caller ID CallerlD Name Username Password Codec Preference Enable Qualify Fax Detection SRTP Firmware Version 1 0 1 22 yam the extension has CID configured The default setting is No Configure the Caller ID This is the number that the trunk will try to use when making outbound calls For some providers it might not be possible to set the CallerlD with this option and this option will be ignored When making outgoing calls the following rules are used to determine which will be used if they exist The CallerID configured for the extension will be looked up first e lf no CallerlD configured for the extension the CallerlD configured for the trunk will be used e the above two are missing the Global Outbound CID defined in Web GUI gt PBX gt Internal Options gt General will be used
112. n s CID when the extension has CID configured The default setting is No Configure the Caller ID This is the number that the trunk will try to use when making outbound calls For some providers it might not be possible to set the CallerlD with this option and this option will be ignored When making outgoing calls the following rules are used to determine which will be used if they exist e CGallerlD configured for the extension will be looked up first e lf no configured for the extension the CallerlD configured for the trunk will be used e the above two are missing the Global Outbound CID defined in Web GUI gt PBX gt Internal Options gt General will be used Configure the name of the caller to be displayed when the extension has no CallerlD Name configured Select audio and video codec for the VoIP trunk The available codecs GSM G 726 G 722 G 729 G 723 ADPCM LPC10 H 264 H 263 263 If enabled the UCM61 xx will regularly send SIP OPTIONS to the device to check if the device is still online The default setting is No Enable to detect Fax signal from the trunk during the call and send the received Fax to the default Email address in Fax setting page under web GUI gt PBX gt Internal Options gt Fax T 38 Note If enabled Fax Pass through cannot be used Enable SRTP for the VoIP trunk The default setting is Regis
113. nation Number Detect is used Note e The PSTN detection process will keep the call up for about 1 minute lf Semi auto Detect is used please pick up the call only after informed from the web GUI prompt e Once the detection is successful the detected parameters Busy Tone Polarity Reversal and Current Disconnect by PSTN will be filled into the corresponding fields in the analog trunk configuration VOIP TRUNKS VoIP trunks can be configured in UCM61xx under Web GUI gt PBX gt Basic Call gt Trunks Once created the VoIP trunks will be listed with Provider Name Type Hostname IP Username and Options to edit and detect the trunk e Click on Create New SIP IAX Trunk to add a new VoIP trunk Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 58 of 138 ndstream innovative IP Voice amp Video e Click on p to configure detailed parameters for the VoIP trunk Clickon to delete the VoIP trunk The VoIP trunk options are listed in the table below Table 24 VoIP Trunk Configuration Parameters Create New SIP IAX Trunk Select the VoIP trunk type e Peer SIP Trunk Type e Register SIP Trunk e Peer IAX Trunk e Register Trunk Configure a unique label to identify this trunk when listed in outbound Provider Name rules inbound rules and etc Configure the IP address or URL for the VoIP providers server of the Host Name trunk U Enter the username to reg
114. nce administrator can always invite other parties from the phone during the call by entering O or 1 To join a conference bridge as administrator enter the admin password when joining the conference A conference bridge can have multiple administrators DURING THE CONFERENCE During the conference call users can manage the conference from web GUI or IVR e Manage the conference call from Web GUI Log in UCM61xx web GUI during the conference call the participants in each conference bridge will be listed 1 Click on to kick a participant from the conference 2 Click on to mute the participant 3 Click on to lock this conference bridge so that other users cannot join it anymore 4 Click on to invite other users into the conference bridge Note When there is participant in the conference the conference bridge configuration cannot be modified e Manage the conference from IVR If Enable Caller Menu is enabled conference participant can input to enter the IVR menu for the conference Please see options listed in the table below Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 72 of 138 tream innovative IP Voice amp Video Table 28 Conference Caller IVR Menu Conference Administrator IVR Menu 1 Mute unmute yourself 2 Lock unlock the conference bridge 3 Kick the last joined user from the conference 4 Decrease the volume of the conference call 5 Decrease your volume 6
115. nd the received Fax to the default Email address in Fax setting page under GUI gt PBX gt Internal Options gt Fax T 38 Fax Detection Note If enabled Fax Pass through cannot be used This option controls how the extension can be used on devices within different types of network Allow All Device in any network can register this extension e Local Subnet Only Only the user in specific subnet can register this extension Up to Strate d three subnet can be specified Specific IP Address Only the device on the specific IP address can register this extension The default setting is Allow All If enabled users will not need enter the PIN Set required by the Skip Trunk Auth outbound rule to make outbound calls The default setting is No Select audio and video codec for the extension The available codecs Codec Preference are PCMU PCMA GSM G 726 G 722 G 729 G 723 ILBC ADPCM LPC10 H 264 H 263 and H 263p BATCH ADD EXTENSIONS Under Web GUI gt PBX gt Basic Call Routes Extensions click on Batch Add Extensions to start adding extensions in batch Table 21 Batch Add Extension Parameters Configure the starting extension number of the batch of extensions to be added Start Extension Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 50 of 138 stream Create Number Specify the number of extensions to be added Assign permission level to the user The available pe
116. ng UCM6102 only BASIC SETTINGS Please refer to the following tables for basic network configuration parameters on UCM6102 UCM6104 and UCM6108 UCM61 16 respectively Table 7 UCM6102 Network Settings gt Basic Settings Select Route Switch or Dual mode on the network interface of UCM6102 The default setting is Route e Route WAN port interface will be used for uplink connection LAN port interface will Method be used to serve as router e Switch WAN port interface will be used for uplink connection LAN port interface will be used as bridge for PC connection e Dual Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 26 of 138 Preferred DNS Server ndstream innovative IP Voice amp Video Both ports can be used for uplink connection Users will need assign the default interface in option Default Interface Enter the preferred DNS server address WAN when Method is set to Route IP Method IP Address Gateway IP Subnet Mask DNS Server 1 DNS Server 2 User Name Password Select DHCP Static IP or PPPoE The default setting is DHCP Enter the IP address for static IP settings The default setting is 192 168 0 160 Enter the gateway IP address for static IP settings The default setting is 0 0 0 0 Enter the subnet mask address for static IP settings The default setting is 255 255 0 0 Enter the DNS server 1 address for static IP settings The default setting is 0 0 0 0 Ent
117. ng is 200ms Configure the ring timeout in ms Trunk FXO devices must have a timeout to determine if there was a hangup before the line is answered This value can be used to configure how long it takes before the UCM61xx considers a non ringing line with hangup activity Configure the RX gain for the receiving channel of analog FXO port The valid range is from 13 5 dB to 12 0 dB The default setting is O Configure the TX gain for the transmitting channel of analog FXO port The valid range is from 13 5 dB to 12 0 dB The default setting is 0 Configure to enable CallerlD detection The default setting is Yes Enable to detect Fax signal from the trunk during the call and send the received Fax to the default Email address in Fax setting page under web GUI gt PBX gt Internal Options gt Fax T 38 Note If enabled Fax Pass through cannot be used Select the Caller ID scheme for this trunk The default setting is Bellcore Telcordia Select the country for tone settings If Custom is selected users could manually configure the values for Busy Tone and Congestion Tone The default setting is United States of America USA 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 manually configure Busy Tone value
118. nks 3 Conference Rooms Unmonitored Grandstream SIP 192 168 40 140 Unavailable Trunk1 Analog Ports 1 gt Extensions 3 Interfaces Status USB SD Card T pit LAN SIP User e SIP User FXS SIP User ges SIP User Messages 0 0 0 SIP User Voice Mail Main Features Dial Voice Mail Features Call Pickup Features Pageing Prefix Features Intercom Prefix Features Parking Lot lt Agent Pause Features Agent Unpause Features Saner 1 Gnanne Extension Call Forward Busy Activate Features No Parked Calls defined Call Forward Busy Deactivate Features Call Forward NoAnswer Activate Features Total 20 Show 1 2 Jumpto zA GI Queues gt Figure 45 Status gt PBX Status TRUNKS Users could see all the configured trunk status in this section Trunks 3 Tfrunks Unm onitored Grandstream SIP 192 16840 140 Unavailable Trunk1 Analog Ports 1 Figure 46 Trunk Status Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 115 of 138 Innovative IP Voice amp Video Table 52 Trunk Status Display trunk status e Analog trunk status Available Busy Unavailable Unknown Error Status e SIP Peer trunk status Unreachable The hostname cannot be reached Unmonitored QUALIFY feature is not turned on to be monitored Reachable The hostname can be reached e SIP Register trunk status Registered Unrecognized Trunk Trunks Display trunk name Display trunk Type
119. nsion D Version 1 0 1 40 D Model GXP2200 Cancel Save Figure 21 Assign Extension To Device After saving the edit dialog the XML config file will be generated in the UCM61xx Reboot the phone to trigger the phone to download the config file CREATE NEW DEVICE Users could also directly create a new device and assign the extension before the device is discovered by the UCM61xx Once the device is plugged in it can then be discovered and get provisioned by the Click on Create New Device and the following dialog will show 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 None Version Model Cancel Save Figure 22 Create New Device Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 45 of 138 stream innovative IP Voice amp Video 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 a Remote Extension S GXP Phone Phone GXV Phone UCM61xx PSTN Lines External Storage for Unlimited Call Recording and future video recording Figure 23 Provisioning Example
120. nymous s Figure 12 LDAP Server Configurations The default phonebook list in this LDAP server can be viewed and edited by clicking on of this Phonebook the first phonebook under LDAP Phonebook Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 34 of 138 andstream innovative IP Voice amp Video Edit Phonebook ou pbx dc pbx dc com LDAP Attributes Contact List John Doe CallelbName John Doe stacy Green Email Tom Lin FirstName Ricky Chan LastName Front Desk Warehouse Department Sales MobileNumber Tech Support HomeNumber Customer Service Fax RMA shipping Test Figure 13 Default LDAP Phonebook in UCM61xx 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 gt 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 Add Phonebook D Phonebook Prefix i Phonebook DN Figure 14 Add LDAP Phonebook Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 35 of 138
121. o be LAN 1 or LAN 2 The default interface is LAN 1 Select DHCP Static IP or PPPoE The default setting is DHCP Enter the IP address for static IP settings The default setting is 192 168 0 160 Enter the gateway IP address for static IP settings The default setting is 0 0 0 0 Enter the subnet mask address for static IP settings The default setting is 255 255 0 0 Enter the DNS server 1 address for static IP settings The default setting is 0 0 0 0 Enter the DNS server 2 address for static IP settings Enter the user name to connect PPPoE Enter the password to connect via PPPoE Table 9 UCM6108 UCM6116 Network Settings gt Basic Settings Enter the preferred DNS server address Select DHCP Static IP or PPPoE The default setting is DHCP Enter the IP address for static IP settings Enter the gateway IP address for static IP settings Enter the subnet mask address for static IP settings Enter the DNS server 1 address for static IP settings Enter the DNS server 2 address for static IP settings Enter the user name to connect via Enter the password to connect via PPPoE The UCM61xx provides users 802 1X settings for LAN port and WAN port UCM6102 only 802 1X Mode Firmware Version 1 0 1 22 Table 10 UCM61 xx Network Settings gt 802 1X Select 802 1X mode The default setting is Disable The supported 802 1X mode are e EAP MD5 e EAP TLS e EAP PEAPvO MSCHAPv2 UCM6102 6104 6108 6116
122. ompt Or enter the code followed by the extension to forward the call Call Forward Busy Deactivate e Default Code 91 e Default Code 92 Call Forward No Answer Activate e Enter the code and follow the voice prompt Or enter the code followed by the extension to forward the call Call Forward No Answer e Default Code 93 Deactivate e Default Code 72 e Enter the code and follow the voice prompt Or enter the code followed by the extension to forward the call Call Forward Unconditional Activate Call Forward Unconditional e Default Code 73 Deactivate e Default Setting 1000 Feature Code Digits Timeout e Configure the maximum interval in milliseconds between the digits input to activate the feature code e Default Extension 700 Call Park e During an active call initiate blind transfer and then enter this code to park the call e Default Extension 701 720 Parked Lots e These are the extensions where the calls will be parked i e parking lots that the parked calls can be retrieved Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 96 of 138 Parking Timeout 5 andstream innovative IP Voice amp Video Default setting 300 This is the timeout allowed for a call to be parked After the timeout if the call is not picked up the extension who parks the call will be called back Feature Codes Voicemail Access Code My Voicemail Agent Pause Agent Unpause Paging Prefix
123. onnection LAN 2 port interface will be used as bridge for PC connection e Dual Both ports can be used for uplink connection Users will need assign the default interface in option Default Interface Enter the preferred DNS server address LAN when Method is set to Switch IP Method IP Address Gateway IP Subnet Mask DNS Server 1 DNS Server 2 User Name Password Firmware Version 1 0 1 22 Select DHCP Static IP or PPPoE The default setting is DHCP Enter the IP address for static IP settings The default setting is 192 168 0 160 Enter the gateway IP address for static IP settings The default setting is 0 0 0 0 Enter the subnet mask address for static IP settings The default setting is 255 255 0 0 Enter the DNS server 1 address for static IP settings The default setting is 0 0 0 0 Enter the DNS server 2 address for static IP settings Enter the user name to connect PPPoE Enter the password to connect via PPPoE UCM6102 6104 6108 6116 USER MANUAL Page 28 of 138 C istan innovative IP Voice amp Video LAN 1 LAN 2 when Method is set to Default Interface IP Method IP Address Gateway IP Subnet Mask DNS Server 1 DNS Server 2 User Name Password Preferred DNS Server IP Method IP Address Gateway IP Subnet Mask DNS Server 1 DNS Server 2 User Name Password 802 1X If Dual is selected as Method users will need assign the default interface t
124. otices a significant change to delay that continues over a few frames it will resync assuming that the change in delay is caused by a Resync Threshold time stamping mix up The threshold for noticing a change in delay is calculated as twice the measured jitter plus this resync threshold The default setting is 1000 Configure the jitter buffer implementation on the receiving side of a analog channel The default setting is Fixed e Fixed Implementation The size is always equal to the value of Max Jitter Buffer e Adaptive The size is adjusted automatically and the maximum value equals to the value of Max Jitter Buffer Jitter Buffer Enable Jitter Buffer Select to enable jitter buffer for IAX The default setting is Select to force the use of jitter buffer on all IAX connections The default setting is Force Jitter Buffer ISO ONU The drop count is the maximum number of voice packets to allow to drop rop Coun 2 out of 100 Usually the useful value is between 3 to 10 Configure the maximum time in ms to buffer in the jitter The default setting is 1000 Max Jitter Buffer Configure the number of interpolated frames the jitter buffer should Max Interpolation Frames e return consecutively The default setting is 10 Configure the resync threshold for jitter buffer When the jitter buffer notices a significant change to delay that continues over a few frames it will resync assum
125. ox for the extension s Then click on Modify Selected Extensions to edit the extensions in a batch The configuration options are listed in Table 21 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 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 54 of 138 ndstream 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 Clickon 1 to delete the analog trunk ANALOG TRUNK CONFIGURATION The analog trunk options are listed in the table below Table 22 Analog Trunk Configuration Parameters Select the channel for the analog trunk UCM6102 2 channels Channels e UCM6104 4 channels UCM6108 8 channels UCM6116 16 channels Specify a unique label to identify the trunk when listed in outbound rules Trunk Name l incoming rules and etc Advanced Options Busy Detection is used to detect far end hangup or for detecting busy Busy Detection TN signal The default setting is ON If Busy Detection is enabled users can specify the number of busy tones to be played before hanging up The default setting is 2 Better Busy Tone Count results might be achieved if set to 4 6 or
126. p Video 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 UCM61xx Please see details in the following table Table 58 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 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 This partition contains PBX system configuration files and service configuration files e Data partition Voicemail recording files IVR file music on hold files and etc e USB disk USB disk will display if connected e SD Card Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 122 of 138 SD Card will display if connected Storage Usage
127. p alive Max Call Numbers Require Call Token Other Settings Configure the maximum number of calls allow for each remote IP address If set to Yes call token is required If set to Auto it will lock out users who depend on backward compatibility when peer authentication credentials are shared between physical endpoints The default setting is Yes SRTP Fax Detection Strategy Firmware Version 1 0 1 22 Enable SRTP for the call The default setting is No Enable to detect Fax signal from the user trunk during the call and send the received Fax to the Email address configured for this extension If no Email address can be found for the user send the received Fax to the default Email address in Fax setting page under web GUI gt PBX gt Internal Options gt Fax T 38 Note If enabled Fax Pass through cannot be used This option controls how the extension can be used on devices within different types of network Allow All Device in any network can register this extension UCM6102 6104 6108 6116 USER MANUAL Page 52 of 138 Essen innova tive IP Voice e Local Subnet Only Only the user in specific subnet can register this extension Up to three subnet can be specified Specific IP Address Only the device on the specific IP address can register this extension The default setting is Allow All l If enabled users will not need enter the PIN Set required by the Skip Trunk Auth ee outboun
128. pply to calls on hold When the call is on hold if there is no RTP activity within the timeout in seconds 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 No UCM6102 6104 6108 6116 USER MANUAL Page 113 of 138 Send Remote Party ID Generate In Band Ringing Server User Agent Allow Non local Redirect Add user phone to URI Send Compact SIP Headers MW 1 Checking Interval Min SIP T1 Timeout SIP SETTINGS DEBUG Enable SIP Debugging Record SIP History Dump SIP History Subscribe Context Allow Subscribe Notify on Ringing Firmware Version 1 0 1 22 yam innovative Configure whether the Remote Party ID should be sent or not The default setting is No Configure whether the UCM61xx should generate inband ringing or not The default setting is Never e Yes The UCM61xx will send 180 Ringing followed by 183 Session Progress and in band audio e No The UCM61xx will send 180 Ringing if 183 Session Progress has not been sent yet If audio path is established already with 183 then send in band ringing e Never Whenever ringing occurs the UCM61xx will send 180 Ringing as long as 200 has not been set yet Inband ringing will not be generated even the end point device is not working properly Configure the user agent string for the UCM61
129. quires it e Force rport Force rport to always be e Yes Force rport to be always on and perform comedia RIP handling e Comedia Use rport if the remote side requires it and performs comedia RTP handling Note comedia RTP handling refers to the technique of sending RTP to the port where the other endpoint s RTP packets come from This can also be rephrased as connection oriented media If enabled the UCM61xx will try to redirect the RIP media stream audio to go directly from the caller to the callee The default setting is No e Yes e No NAT Allow media path redirection Reinvite but only when the peer is not be behind NAT The RTP core can detect 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 Note Some devices do not support this especially if one of them is behind NAT UCM6102 6104 6108 6116 USER MANUAL Page 112 of 138 SIP SETTINGS TOS ToS For SIP ToS For RTP Audio ToS For RTP Video Default Incoming Outgoing Registration Time Max Registration Subscription Time Min Registration Subscription Time Music On Hold Interpret Music On Hold Suggest Enable Relaxed DIMF Mode RTP Timeout RTP Hold Timeout Trust Remote Party ID Firmware Version 1 0 1 22 m Innovative I Table 50 SIP Settings TOS Configure the Type of Service for SIP packets The
130. ress for static IP settings Enter the user name to connect via PPPoE Enter the password to connect via PPPoE LAN 1 LAN 2 when Method is set to Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 27 of 138 Default Interface IP Method IP Address Gateway IP Subnet Mask DNS Server 1 DNS Server 2 User Name Password Method Preferred DNS Server istan innovative IP Voice amp Video If Dual is selected as Method users will need assign the default interface to be LAN 1 mapped to UCM6102 WAN port or LAN 2 mapped to UCM6102 LAN port and then configure network settings for LAN1 LAN2 The default interface is LAN 1 Select DHCP Static IP or PPPoE The default setting is DHCP Enter the IP address for static IP settings The default setting is 192 168 0 160 Enter the gateway IP address for static IP settings The default setting is 0 0 0 0 Enter the subnet mask address for static IP settings The default setting is 255 255 0 0 Enter the DNS server 1 address for static IP settings The default setting is 0 0 0 0 Enter the DNS server 2 address for static IP settings Enter the user name to connect Enter the password to connect via PPPoE Table 8 UCM6104 Network Settings gt Basic Settings Select Switch or Dual mode on the network interface of UCM6104 The default setting is Switch e Switch LAN 1 port interface will be used for uplink c
131. rmissions are Internal Local National and International from the lowest level to the highest level The default setting is Internal Permission Note Users need to have the same level as or higher level than a outbound rule s privilege in order to make outbound calls from this rule Enable Voicemail Enable Voicemail for the user The default setting is Yes Configure the SIP IAX password for the users Three options are available to create password for the batch of extensions e User Random Password SIP IAX 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 on all the extensions in the batch Configure Voicemail password digits only for the users e User Random Password A random password in digits will be automatically generated It is Voicemail Password recommended to use this password for security purpose e Use Extension as Password e Enter a password to be used on all the extensions in the batch Configure the number of seconds to ring the user before the call is Ring Timeout forwarded to voicemail voicemail is enabled or hang up voicemail is disabled Technology SIP Select SIP if the users are using SIP or a SIP device Select if the users are using IAX or IAX device Use NAT when the PBX is on a public IP communicating with devices hidden behin
132. rt Click on the field and the calendar will show for users to select the exact date and time The call report will display as the following figure shows Start Time 7 2013 07 03 18 29 26 2013 07 03 18 29 00 2013 07 03 18 28 51 2013 07 03 18 28 38 2013 07 03 18 28 31 2013 07 03 18 28 06 2013 07 03 18 27 47 2013 07 03 17 55 04 2013 07 03 17 54 32 0660006606666 2013 07 03 17 53 11 Delete All Download Records First J Prev Last Figure 55 Call Report 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 will sort the report according to start time Clicking on Start Time again will 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 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 second entry has audio recording file for the call Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 125 of 138 andstream innovative IP Voice amp Video Click on to play the recording file click on to download the recording file in wav format
133. s e Please be patient during the upgrading process Once done a reboot message will be displayed in the LCD e Manually reboot the UCM61xx when it s appropriate to avoid immediate service interruption After it boots up log in the web GUI to check the firmware version UPGRADING VIA LOCAL UPLOAD If there is no server users could also upload the firmware to the UCM61xx directly via Web GUI Please follow the steps below to upload firmware locally e Download the latest UCM61 xx firmware file from the following link and save it in your PC http www grandstream com support firmware Login Web GUI as administrator in the PC eS e Go to Web GUI gt Maintenance gt Upgrade upload the firmware file by clicking on select the firmware file from your PC The default firmware file name is ucm6100fw bin Local Upgrade D Firmware File Path ucmbT100fw bin Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 128 of 138 innovative IP Voice amp Video Figure 59 Local Upgrade e Click on o to start upgrading Dt Upgrading Firmware files Figure 60 Upgrading Firmware Files Wait until the upgrading process is successful and a window will be popped up in the Web GUI Prompt information Device successfully upgraded Do you want to restart the device now to make the changes take effect Figure 61 Reboot UCM61xx e Click on OK to reboot the U
134. s 83 CONFIGURE VOICEMAIL GROUP cssssessssessssessssesssseesssessseesssestssessisestssessstesssnetsaneesseseaseeesseaee 84 RING GROUP miM t eee arcsec 85 CONFIGURE RING ttt ttt tte tte ttti 85 PAGING AND INTERCOM GROUP nani nnne 87 CONFIGURE PAGING INTERCOM GROUP tette ttti 87 E e U 89 CONFIGURE CALL 89 HOLD Sener 92 gi 93 CONFIGURE FAXI T 38 sscssssessssessssessssessssessssesssserssesssssessuetssessssestssessasestsnssssesssneesaneesieseasseeseneaee 93 CALE FE eS 95 FEATURE CODES tette ttt ttt tte ttt tte ttt 95 BECORDING 97 98 PARK ATA 98 RETRIEVE THE PARKED CALL 98 INTERNAL OPTIONS wees neces 99 INTERNAL OPTIONS GENERAL c sscesssssesssessseesssssessessssstssesssseessnesssneessteessseesseeessesessesesseseaseeen 99 Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 3 of 138 istan innovative IP Voice amp Video INTERNAL OPTIONS RTP SETTINGS 100 INTERNAL OPTIONS RTP SETTINGS 102 INTERNAL OPTIONS HARDWARE 6 102 INTERNAL OPTIONS STUN 104 m 105 AX SE
135. s ucm series ucm61xx documents ucm61xx voiceprompt customiz Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 81 of 138 CONFIGURE VOICEMAIL yam innovative VOICEMAIL If the voicemail is enabled for UCM61 xx extensions the configurations of the voicemail can be globally set up and managed under Web GUI gt PBX gt Call Features gt Voicemail Max Greeting Dial 0 For Operator Max Messages Per Folder Max Message Time Announce Message Caller ID Announce Message Duration Play Envelope Allow Users To Review Firmware Version 1 0 1 22 Table 30 Voicemail Settings Configure the maximum number of seconds for the voicemail greeting The default setting is 60 seconds If enabled the caller can press 0 to exit the voicemail application and connect to the configured operator s extension The operator extension can be configured under web GUI gt PBX gt Internal Options gt General Configure the maximum number of messages per folder in users voicemail The valid range 10 to 1000 The default setting is 50 Select the maximum duration of the voicemail message The message will not be recorded if the duration exceeds the max message time The default setting is 15 minutes The available options are 1 minute e 2minutes 5 minutes 15 minutes 30 minutes e Unlimited If enabled the caller ID of the user who has left the message will be announced at the beginning of the voicemai
136. snenususnennsneensnsnnens 17 CONNEC T THE UJOMDBT1B ce cra ee cts eect de ee 18 SAFETY COMPLIANCES c cccccccccccceccccececcccecescucuccecucesuucusuacucueuuausuauuauaneauanecuauacesuucususcucesuuausuess 19 B 19 GETTING STARTED 20 USE THE LCD 20 USE THE LED 22 USE THE WEB 23 ACCESS WEB 23 WEB GUI CONFIGURATIONS ccccccccceccccecccccececcetensccenscueusceetsnsnususatsnscenteneensnecusnsnusnsnennsnsnnns 24 WEB GUI 24 SAVE AND APPLY CHANGES 25 MAKE YOUR FIRST GALL a cece cece cece cecececcccececececcaueceacececcaueccauecesusuuausueausnscuanetesesneataueesuenestsneass 25 SYSTEM 26 NETWORK SETTINGS ccccccccececccceccccececcacececceuecaueceacenegeaueccaueaususuuctsueaususctansnesueneataueesueneataneass 26 PASO cat 26 29 PORT FORWORDING
137. st on the device This is mainly for factory testing purpose which verifies the hardware connection inside the device The diagnostic result will display in the LCD after the test is done e Protocol Web access protocol HTTP or HTTPS By default it s HTTPS Web Info e Port Web access port number By default it s 8089 USE THE LED INDICATORS The UCM61xx has LED indicators in the front to display connection status The following table shows the status definitions Table 5 UCM6102 UCM6104 LED INDICATORS LED Indicator LED Status LAN WAN Solid Connected USB Flashing Data Transferring SD OFF Not Connected FXS Phone Fax FXO Telco Line Table 6 UCM6108 UCM6116 LED INDICATORS LED LED Status Solid Connected NETWORK OFF Not Connected ACT USB SD Solid Connected Flashing Data Transferring MN Phone FXS OFF Not Connected Line FXO Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 22 of 138 dstream innovative IP Voice amp Video USE THE WEB GUI ACCESS WEB GUI The UCM61xx 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 UCM6116 IP PBX Appliance Copyright Grandstream Networks Inc 2013 All Rights Reserved Figure 9 UCM6116 Web GUI Login Page To access the Web GUI 1 Connect
138. t click on the check box for the LAN or WAN UCM6102 only interface Under Custom Firewall Settings users could create new rules to accept reject or drop certain traffic going through the UCM61xx To create new rule click on Create New Rule button and a new window will pop up for users to specify rule options Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 31 of 138 andstream innovative IP Voice amp Video Create new firewall rule Rule Name 1 Action 1 1 Service Figure 11 Create New Firewall Rule Table 14 Firewall Rule Settings Rule Name Specify the Firewall rule name to identify the firewall rule Select the action for the Firewall to perform e ACCEPT Action e REJECT e DROP Select the traffic type IN Type If selected users will need specify the network interface LAN or WAN for UCM6102 only for the incoming traffic e OUT Select the service type FTP e SSH e Telnet Service Id e HTTP LDAP e Custom If selected users will need specify Source IP and port Destination IP and port and Protocol TCP UDP or Both for the service Save the change and click on Apply button Then submit the configuration by clicking on Apply Changes on the upper right of the web page The new rule will be listed at the bottom of the page with sequence Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 32 of 138 sir
139. t WAN Port Ground USB Pot DC 12W LAN Pot 2 Pot 2x FXO Port Figure 2 UCM6102 Back View To set up the UCM6102 follow the steps below 1 Connect 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 the device 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 Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 16 of 138 istan innovative IP Voice amp Video CONNECT THE UCM6104 d C lis i e ls P EN LCD Navigation Cl Keys LAN 1 LED LAN2LED USE LED D FXS LED FXO LED Figure 3 UCM6104 Front View SD Card Slot Reset Ground USB Pot DC 12v 2x LAN Pot 2xFXS Port 4 Port Figure 4 UCM6104 Back View To set up the UCM6104 follow the steps below 1 Connect one end of an RJ 45 Ethernet cable into
140. t setting is Standard Configure the type of Message Waiting Indicator detection on trunk FXO interfaces The default setting is None e None No detection FSK Frequency Shift Key detection e NEON Neon MWI detection INTERNAL OPTIONS STUN MONITOR STUN Server STUN Refresh Firmware Version 1 0 1 22 Table 41 Internal Options STUN Monitor Configures the IP address or URL of the STUN server to query If not specified STUN is disabled The default setting is stun ipvideotalk com Valid format hostname IP address port The default port number is 3478 if not specified Configure the number of seconds between STUN Refreshes The default setting is 30 seconds UCM6102 6104 6108 6116 USER MANUAL Page 104 of 138 innova tive IP Voice IAX SETTINGS The UCM61 xx IAX global settings can be accessed Web GUI gt PBX gt IAX Settings IAX SETTINGS GENERAL Bind Port Bind Address IAX1 Compatibility No Checksums Delay Reject ADSI Music On Hold Interpret Music On Hold Suggest Bandwidth IAX SETTINGS CODECS Table 42 IAX Settings General Configure the port number that the 2 will be allowed to listen to The default setting is 4569 Configure the address that the IAX2 will be forced to bind to The default setting is 0 0 0 0 which means all addresses Select to configure IAX1 compatibility The default setting is No If selected UDP checksums will be disab
141. ted Call Numbers Call Number Limits IP or IP Range Firmware Version 1 0 1 22 Table 44 IAX Settings Static Defense Enter a single IP address or a range of IP addresses for which call token validation is not required For example 11 11 11 11 11 11 11 11 22 22 22 22 Configure the maximum number of calls allowed for a single IP address Configure the maximum number of unvalidated calls for all IP addresses Configure to limit the number of calls for a give IP address of IP range Enter the IP address or a range of IP addresses to be considered for call number limits For example 11 11 11 11 11 11 11 11 22 22 22 22 UCM6102 6104 6108 6116 USER MANUAL Page 107 of 138 innovative SIP SETTINGS The UCM61xx SIP global settings can be accessed Web GUI gt PBX gt SIP Settings SIP SETTINGS GENERAL Realm For Digest Authentication Bind UDP Port Bind IP Address Allow Guest Calls Overlap Dialing Support Allow Transfer Enable DNS SRV Lookups on Outbound Calls MWI From Domain From Domain Auto Domain Allow External Domains Firmware Version 1 0 1 22 Table 45 SIP Settings General Configure the host name or domain name for the UCM61xx Realms MUST be globally unique according to RFC3261 The default setting is Grandstream Configure the UDP port used for SIP The default setting is 5060 Configure the IP address to bind to The default setting is 0 0 0 0 which
142. template variables are e t TAB VM first name and EE name VM_DUR The duration of the voicemail message The recipient s extension VM_CALLERID The caller ID of the person who has left the message VM_MSGNUM The number of messages in the mailbox e The date and time when the message is left Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 83 of 138 andstream innovative IP Voice amp Video Click on Load Default Settings button to view the default template as an example CONFIGURE VOICEMAIL GROUP The UCM61 xx supports voicemail group and all the extensions added in the group will receive the voicemail to the group extension The voicemail group can be configured under Web GUI gt PBX gt Call Features gt Voicemail Group Click on Create New Voicemail Group to configure the group Create New Voicemail Group Voicemail Group Extension 6600 Name 4 Voicemail Group Mailboxes Available Mailboxes 5000 John Doe 5005 Warehouse 5001 Stacy Green 5006 Sales 5002 Tom Lin 5007 Tech Support 5003 Ricky Chan 5008 Customer Service 5008 Anan Figure 35 Voicemail Group e Voicemail Group Extension Enter the Voicemail Group Extension The voicemail messages left to this extension will be forwarded to all the voicemail group members e Name Configure the
143. ter SIP Trunk Configuration Parameters Provider Name Host Name Transport Firmware Version 1 0 1 22 Configure the provider name for the VoIP trunk This is a unique label to identify the trunk when listed in outbound rules inbound rules and etc Configure the IP address or URL for the VoIP provider server of the trunk Configure the SIP transport protocol to be used in this trunk The default setting is All UDP Primary UCM6102 6104 6108 6116 USER MANUAL Page 60 of 138 Username Password Codec Preference From Domain From User Outbound Proxy Support Outbound Proxy Enable Quality Fax Detection SRTP istan innovative IP Voice amp Video e UDP Only e TCP Only e TLS Only All UDP Primary UDP is the primary transport protocol when the other SIP transport methods are available too All TCP Primary TCP is the primary transport protocol when all the other SIP transport methods are available too All TLS Primary TLS is the primary transport protocol when all the other SIP transport methods are available too Enter the username to register to the trunk from the provider Enter the password to register to the trunk from the provider 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 ADPOM LPC10 H 264 H 263 263 Configure the actual domain name where the extension comes from This can be use
144. tercom or 1 way Page Intercom Group Select available users from the right side to the paging intercom group Members member list on the left e to edit the page intercom group e Clickon to delete the page intercom group Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 87 of 138 Firmware Version 1 0 1 22 andstream innovative IP Voice amp Video Click on Paging Intercom Group Settings to edit Alert Info Header This header will be included in the SIP INVITE message sent to the callee in paging intercom call Paging Intercom Group Settings Settings for Paging amp Intercom 7 Alert Info Header Intercom Settings For Paging Intercom Feature Code Please to Feature Codes page for setting paging intercam feature code Cancel Figure 39 Page Intercom Group Settings The UCM61xx has pre configured paging intercom feature code To edit page intercom feature code click on Feature Codes in the Paging Intercom Group Settings dialog Or users could go to Web GUI gt PBX gt Internal Options gt Feature Codes directly UCM6102 6104 6108 6116 USER MANUAL Page 88 of 138 andstream innovative IP Voice amp Video CALL QUEUE The UCM61xx supports call queue by using static agents or dynamic agents This sections describes the configuration of call queue under Web GUI gt PBX gt Call Features gt Call Queue CONFIGURE CALL QUEUE Call queue settings can be
145. 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 the device 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 CONNECT THE UCM6108 To set up the UCM6108 follow the steps below Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 17 of 138 andstream Connect one end of an RJ 45 Ethernet cable into the LAN port of the UCM6108 Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch hub 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 Wait for the UCM6108 to boot up The LCD in the front will show the device hardware information when the boot process is done Once the UCM6108 is successfully connected to network the LED indicator for NETWORK in the front will be in solid green and the LCD s
146. ure 48 Figure 49 Figure 50 Figure 51 Figure 52 Figure 53 Figure 54 Figure 55 Figure 56 Figure 57 Figure 58 Figure 59 Figure 60 Figure 61 Figure 62 Figure 63 Figure 64 Figure 65 Figure 66 Figure 67 Figure 68 Figure 69 Firmware Version 1 0 1 22 istan innovative IP Voice amp Video Page Intercom Group Seting 88 GUEST 89 On Hold tc 92 Download Recording File From CDR esses nennen nennen nnns 98 FXS Ports Signaling Preference lesen nennen nennen nnn nennen nnn nnns 102 FXO POMS ACIM UT T 103 elus PEA SIUS ONU REI C DEM SIRE E ias 115 TUNK RR EI 115 STIS ION E di 117 cucus IURE M 118 ciu de irat 119 6116 Interfaces Status 119 M NIRE 121 System Status Storage Usage ccccccsssccccccsseeceeeseeeeeeeeeeeeeeeeseeeeeeeeseaeeeeesaeaeeeeeeseaeeeeeeaegess 123 System Status Resource Usage ccceccccccaesseceeeceeeeeeeeaeeeeeceeseeeeeeeeseaueeeeeaeaueeeessaaueeeesaeaass 123 COR FINO EE
147. us PSTN or IP participants UCM6108 6116 supports up to 6 conference bridges allowing up to 32 simultaneous PSTN or IP participants The conference bridge configurations can be accessed under Web GUI gt PBX gt Call Features gt Conference In this page users could create edit view invite manage the participants and delete conference bridges The conference bridge status and conference call recordings if recording is enabled will be displayed in this web page as well CONFERENCE BRIDGE CONFIGURATIONS Click on Create New Conference Room to add a new conference bridge e Click on to edit the conference bridge e Click on to delete the conference bridge Table 27 Conference Bridge Configuration Parameters Configure the conference number for the users to dial into the Extension conference When configured the users who would like to join the conference call must enter this password before accessing the conference bridge Password Note If Public Mode is enabled the password is not required to join the conference bridge thus this field is invalid Configure the password to join the conference bridge as administrator Conference administrator can manage the conference call via IVR if Enable Caller Menu is enabled as as invite other parties to join Admin Password E M l the conference by dialing permission required from the invited party or 1 permission not required from the invited p
148. xx If enabled 302 or REDIRECT is allowed to non local SIP address The default setting is If enabled user phone will be added to URI that contains a valid phone number The default setting is If enabled compact SIP headers will be sent The default setting is Configure the default interval in seconds for checking MWI status of peer s voicemail The default setting is 10 Configure the minimum roundtrip time in milliseconds for the SIP messages sent to the monitored hosts The default setting is 100 Table 51 SIP Settings Debug Select to enable SIP debugging The default setting is No Select to enable recording SIP history The default setting is Select to enable dump SIP history at the end of SIP dialogue The default setting is Configure a specific context for SUBSCRIBE requests This setting is useful to limit subscriptions to local extensions The default setting is from internal Configure to allow subscriptions The default setting is Yes Configure to send out NOTIFY on ringing state The default setting is Yes UCM6102 6104 6108 6116 USER MANUAL Page 114 of 138 stream innovative IP Voice amp Video STATUS AND REPORTING PBX STATUS The UCM61xx 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 Tru
149. y 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 Ring Strategy will stop ringing e Ring in order Ring the members with the order configured in ring group list If the first member doesn t answer the call it will stop ringing the first member and start ringing the second member Firmware Version 1 0 1 22 UCM6102 6104 6108 6116 USER MANUAL Page 85 of 138 Ring Timeout on Each Member Enable Voicemail Secret Email Address Edit Ring Group 6400 Ring Group Extension istan innovative IP Voice amp Video Configure the number of seconds to ring each member If set to 0 it will keep ringing The default setting is 30 seconds Note The actual ring timeout might be overridden by users if the phone has ring timeout settings as well If enabled users could select to use the ring group extension as the voicemail or select another extension s voicemail box as the ring group voicemail Configure the password to access the ring group extension s voicemail Configure the Email address of the ring group extension s voicemail If Attach Recordings to E mail is enabled from Web GUI gt PBX gt Voicemail gt Voicemail Email Settings the voicemail can be sent to the ring group s Email address as attachment Techsupport 6400 Ring Group Members Avaliable Users 5000 John Doe 5001 Stacy Green 5002 Tom Lin

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