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UCM6510 User Manual

1. cccccccccccsssseeeeeeeeeeeeeeeeeeeseeeeseeeeeceeeeeaeeeeeeeseeeeeeeeesseaeeeeesssaaeeeeeeeas 43 Figure 17 Configure Dynamic Defense esasenaiinenmssierie na ee nanei oe a EREE E EEE a 45 Figure 18 LDAP Server Configurations ccccccccsssescccceeeceseesssecceeeeeeaeeeseceeeeeseeeasseseeeeesssaeeseeeeeeeessaaaaeess 48 Figure 19 Default LDAP Phonebook DN 48 Figure 20 Default LDAP Phonebook Attributes ccccccccceccceceeeseeeeeeeeeeeceeeeeeeeceeseseaeeeeeesseaeeeeessaeeeeeesas 49 Figure 21 Add LDAP le ele ie 49 Figure 22 Edit LDAP en Ee ele 50 Figure 23 GXP2200 LDAP Phonebook Configuration ccccccccecccssesseeceeeeeeeeaeseeeeeeeeeeseeeeeeeeeeeessaanaeees 52 Figure 24 UCM6510 Email Gettmges 54 Figure 25 UCM6510 Email Settings Send Test Email 55 Figure 26 Set Time Manually 02 ccccccc cccs secccceeecccce scenes cuesesccessesneseeatecesssesnecteutsocsdseserectstececesereeeectstecenses 57 Figure 27 UCGMGS TO 280 CONG BE 60 Figure 28 Auto Provision Settings cccccccsseeecceeeeeseeeeeeceeeeeeeeeeeeeseeeessaeeeeeeesseeeeeeeesseeseeeeesseeeeeesessageeeeeeeas 61 Faure 29 UNO DISC OV EEE EE EEE EEE ER 62 Figure 30 Discovered Devices sn nnnunsrererereerrrrrsrrrrrrortnrrrrrnrtrrosrnrtrrtnrrrrostnttrranrtrrnsanrrrrnnrerennnnrren nnne 63 Figure 31 Assign Extension To Device 63 Figure 32 Create New Device nno1nnnnnnrnnenensrnrrrrsrnrrrrosrnrrrrsnnrrrrrsrnrrrrstnrrrrotnrrrrs
2. Table 42 Voicemail Email Settings l If enabled voicemails will be sent to user s Email address The default Attach Recordings to E Mail E setting is Yes Fill in the Subject and Message content to be used in the Email when sending to the user The template variables are e t TAB Tae ra Meter Garer VM_NAME Recipient first name and Aa name e VM_DUR The duration of the voicemail message e VM_MAILBOX The recipient s extension e VM_CALLERID The caller ID of the person who has left the message e VM_MSGNUM The number of messages in the mailbox e VM_DATE The date and time when the message is left Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 142 of 233 Andstream Innovative IP Voice amp Video Click on Load Default Settings button to view the default template as an example CONFIGURE VOICEMAIL GROUP The UCM6510 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 5600 Mame VMGroup1 Available Mailboxes Voicemail Group Mailboxes 5005 Warehouse 5000 John Doe 5006 Sales 5001 Stacy Green 5007 Tech Support 5002 Tom Lin 5008 Customer Service 5003 Ricky Chan 5009 RMAF AN
3. ccccceeeeeeeeeeeeeeeeeeeeeeeeeeeesaaeeeeeeeeeeeeeeaaeaeeeeeeeeessaaaneeeeeeseesaaas 104 Data Trunk VV PP eee ed 105 Data TUNK CONNOUrAUO EEN 106 DOD extension selection cccccccccseseeceecceeeeeeeceseeeceeecseaseeeesaaaseeeesseauseeeessuaaeeeeessaaeeeeessaaaes 115 Edt 86 DEE EE 115 Blacklist Configuration Harameiers A 122 Conference Invitation From web GU 127 Conference Recording EEE EEE EEE 130 Click On Prompt To Create IVR Prompt annnnnnnnneseeeensssserennnnrtrrtnereeensssssrrnnnnrrtrrrrreeeensssnnen 133 Record New IVR NE 133 Gel 91200 IVR gg 011 0 EEE EE EEE EE 134 Language Settings for Voice Prompt ernrvrrrrrnnnnnnnnnnnnnrrrrrrvvnnnvnnrnnnnnnnnnnnnsnnrrsrrevnnnnnnrnnrnnnnnnnnnn 136 Voice Prompt Package et 136 New Voice Prompt Language Added rrrrrrnnnnnnvnnnnnnnnnvnnrnnnnnrennrnnnnnnennnnnnnnsennnnnnsnsennnnnnnsennnnnnn 137 Voicemail nd LE 142 ie e Ti Go o ME 143 Kilvlet Edel le 145 RING Group ii let D te Lu WE 146 PACMAG WATE de ein GROUP EE EE EEE EEE 147 Page Intercom Group Settings cccccecccecccceecceeeeseeeceeeeeeseeeseeeeeeseeseeesuseeeeesssseaeeeseeeeesssaaseses 148 SEO EEE NN 149 Agom COIN Te Le EEE EE ican 151 EIU EXON TON OO EE EN EN Nr 153 Select Extension Group in Outbound Route 154 Edit PICKUP GA OUND RE EEE EE 155 Music On Hold Default Class 157 Configure Analog Trunk without Fax Detection rrrrrrrrrrrrrrnnnnnnnnnnrnnnrorrnnnnnnrrrrrnnnnnnnnnnnnnnnnnner 160 Configu
4. ve Innovative IP Voice amp Video Grandstream Networks Inc UCM6510 IP PBX User Manual This page intentionally left blank stream Innovative IP Voice amp Video UCM6510 IP PBX User Manual Index CHANGE LOG i avanananananannnnnnnnsnnnanananansnansnsnsnnsnanvannnnnsnnnnnnnnnnnnnnnnnnnnnnnnnnnnnn 12 FIRMWARE VERSION 10O02b nnana anaana anaa AADA A LAADA nnana anann an nanna 12 WELCOME uu NK NK ENNEN NN ENN KEN NN ENN NENNEN NENNEN NN ENNEN NK ENNEN NN KN NK ENN ENN NNN NN ENN 13 PRODUCT OVERVIEW uuu cccccccsccccccccncccnccccncecncenneuunueunaeeunueunueennenuneuennees 15 KEATUREHIGHTUIGHTeG nnan n anaana anaa LAADA ALAALA LAADAL Aana n nanena nnn nna 15 TECHNICAL SPECIFICATIONS 00 0 cccc ccc cccccccccecececececccuaceccceuececuaeaesuauececuauauauauenecuuaesusuenececnsnsesusneneas 16 INSTALLATION eu NNN KRAN NNN NNN NN ENN NNN REN NN ENN NNN NN ENN ENN NENNEN NN ENN KEN NN ENNEN 19 EQUIPMENT PACKAGING c cccccccceccccccccececececucccceceuececuaeaugustenecuauaesusuenenuavsesusneneneanaususnsneuanaesusnes 19 CONNECT YOUR UCMGbi aonana ALAA nAAL AAPA ALARDA An nannan nannan 19 CONNECT IHE UCMGS TO EENG 19 SAFETY COMPHANCES ee 21 WARRANTY cdeecedetercedieccasaentiaendsucaenenbenewssdliddeestwarhiienedicacaanmbiesneandkadnesetendsnanheaddicane2aneediidawantexedaendcncatarans 21 GETTING STARTED NNN NNN NN KN NN KN NK NNN KN NN NNN REN NN ENN KN NN EN NN ENN NNN NN ENNEN NA 23 USE THE LOD MENU WEE 23 USE THE LED INDICA
5. Account Select Account d None A Figure 32 Create New Device PROVISIONING After the successful discovery and assignment configuration on the UCM6510 the device will start downloading the config file and take the new configuration with the extension registered Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 64 of 233 clean Innovative IP Voice amp Video EXTENSIONS CREATE NEW USER CREATE NEW SIP EXTENSION To manually create new SIP user go to web GUI gt PBX gt Basic Call Routes gt Extensions Click on Create New User gt Create New SIP Extension and a new dialog window will show for users to fill in the extension information The configuration parameters are as follows Table 18 SIP Extension Configuration Parameters The extension number associated with the user This is the SIP UserlD Extension t for registration Configure the CallerID Number that would be applied for outgoing calls from this user CallerID Number Note The ability to manipulate your outbound Caller ID may be limited by your VolP 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 an outbound rule s privilege in order to make outbound calls using this rule If the outbound rule privileg
6. By default the UCM6510 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 UCM6510 to negotiate endpoint to endpoint media routing The default setting is No Select DIMF mode for the user to send DTMF The default setting is RFC2833 If Info is selected SIP INFO message will be used If DTMF Mode 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 Insecure required e 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 Enable Keep alive K keep the NAT port open The default setting is Yes Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 67 of 233 Keep alive Frequency Auth ID Andstream Configure the Keep alive interval in seconds to check if the host is up The default setting is 60 seconds Configure the authentication ID for the user If not configured the extension number will be used for authentication Other Setting
7. Firmware Version 1 0 0 25 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 VolP provider server of the trunk When enabled it can avoid overridden by extension s CID if the extension has CID configured The default setting is enabled 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 CallerID with this option and this option will be ignored When making outgoing calls the following rules are used to determine which CallerID will be used if they exist e The CallerID configured for the extension will be looked up first e f no CallerID configured for the extension the CallerID configured for the trunk will be used e If 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 CallerID Name configured Enter the username to register to the trunk from the provider Enter the password to register to the trunk from the provider This is the authentication ID for the UCM6510 to register to the trunk if required by the provider If not specified the CallerID name will be sued for authentication Select audio and video codec for the VoIP trunk T
8. The agent has been logged out On the UCM6510 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 Dr e Click on to refresh the call queue status e Click on to expand the call queue detail e Click on to hide the call queue detail Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 196 of 233 ndstream Innovative IP Voice amp Video 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 Figure 87 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 O to refresh the conference room status e Click on to expand the conference room details e Click on to hide the conference room details INTERFACES STATUS This section displays interface connection status on the UCM6510 for USB
9. ssssssssssssttttttiiitrrrrssesssssnntenttttttnnnttrrrrrnrssnnnnnn nn nnnetrrrr rnrn 182 AK DENS Jr 183 NE a MEINERS C EE 183 IAX SETTINGSREGISTRATION EE 183 IAX SETTINGS STATIG DEFENSE ENEE 184 MVP SETTINGS 187 SIP SETTINGS GENERAL nsrensrrsssrensrrsarrnnsrrnarennsrensrenssrensrnaseensernasennsrenssennstensensseenseenasennseenssennsenn 187 PETN 55992299 188 PEN EAN 188 SIP SETTINGS TCP and E 189 Sege 190 Se le te 191 STATUS AND REPORTING ege 193 BEE 193 Te 193 DTS SEE 194 eebe 196 6 01731210 ee SEE EEE NE 197 NER NE 197 DIGITAL CHANNELS SI Sene 199 TENNE 200 SYSTEM STATUS EEE SE 201 SE 202 FUN 202 SOLE EE EEE 203 a100 0 EEE 203 SEE EE E 204 ALERT EVENTO LIST EEE EE 204 rg e 206 ALERT DN 207 Me 207 DOWNLOADED CDRA EEE 209 SEE SEERE 211 ENE 212 CDR API CONFIGURATION FILES ENEE 213 UPGRADING AND MAINTENANCE rennnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnnen 219 FAN 45 219 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 5 of 233 san Innovative IP Voice amp Video UPGRADING VIA NETWORK 219 UPGRADING VIA LOCAL ERE 220 NO LOCAL FIRMWARE SERVERS ENEE 222 Se TE 222 ee 223 DATA SYN EE EE et 223 RESTORE CONFIGURATION FROM BACKUP ELE 224 TN 225 RESET AND REBOOT EE 226 Ve 227 TENNE 228 ETHERNET Eve 228 PI 228 EE 229 PRI SS7 MFC R2 SIGNALING TRACE EE 229 Te Ee 230 EXPERIENCING THE UCM6510 SERIES IP PBX eennnnnnnneennnnnnnnnevnnnnnn 233 Firmware Version 1 0 0
10. Enter the DNS server 2 address for static IP settings Enter the user name to connect via PPPoE Enter the password to connect via PPPoE Assign the VLAN tag of the layer 2 QoS packets for WAN port The default value is O Assign the priority value of the layer 2 QoS packets for WAN port The default value is 0 LAN when Method is set to Route IP Address Subnet Mask DHCP Server Enable DNS Server 1 DNS Server 2 Allow IP Address From Allow IP Address To Default IP Lease Time 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 Enter the DHCP IP Pool starting address The default setting is 192 168 2 100 Enter the DHCP IP Pool ending address The default setting is 192 168 2 254 Enter the IP lease time in seconds The default setting is 43200 LAN when Method is set to Switch IP Method IP Address Subnet Mask Gateway IP Firmware Version 1 0 0 25 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 subnet mask address for static IP settings The default setting is 255 255 0 0 Enter the gateway IP address for stat
11. Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 216 of 233 ndstream Innovative IP Voice amp Video A Note e Disallowed characters in the caller callee startTime or endTime strings and non digit characters in the values of numRecords offset minDur or maxDur will result in no records returned the appropriate container header for the output format will be the only output If the format parameter is in error the CSV header will be used Error messages will appear in the Asterisk log along with errors stemming from failed database connections etc e Other errors which return no records include Multiple hyphens in an extension range e g caller 5300 5301 6300 Empty parameter value e g caller Extension values starting with comma or with consecutive commas e g caller 5300 5303 Unknown parameters e g caler 5300 or URI ending with amp Except for caller and callee multiple instances of the same parameter within the URI e g minDur 5 amp minDur 10 Example Output The following are examples of each of the output formats for the same data set CSV Acctld accountcode src dst dcontext clid channel dstchannel lastapp lastdata start answer end duration billsec disposition amaflags uniqueid userfield channel_ ext dstchannel evt service 62 5300 5301 from internal pn01 lt 5300 gt SIP 5300 00000000 SIP 5301 00000001 Dial SIP 5301 60 2013 12 03 11 46 40 2013 12 03 11 46 43 2013 12
12. Innovative IP Voice amp Video DIGITAL TRUNK CONEIGURATION ennn nnneerr errr r rnn 102 DIRECT OUTWARD DIALING DOD VIA DIGITAL TRUNKS 103 DIGITAL TRUNK TROUBLESHOOT ING EE 103 BEE EE NE 105 VP TRUNG S ee 107 VOIP TRUNK CONEIGURATION nabab b bbb b etne Ennn nrrnrrrrn anans 107 DIRECT OUTWARD DIALING DOD VIA VOIP TRUNKS 114 CACC eg eee ee ee E rn Re eer eee oer ee A 117 ENE E 117 EIE Jr 119 INBOUND RULE CONFIGURATIONS ve tachenestiaie de cesteancraiatedeectase 119 BLACKLIST CONFIGURATIONS EEE 122 CONFERENCE BRIDGE Seen ee 125 CONFERENCE BRIDGE CONEIGURATIONS 125 JOIN A CONFERENCE CALL acces eee reece cee eee 127 INVITE OTHER PARTIES TO JOIN CONFERENGE ennenen 127 DURING THE CONFERENCE ENEE 128 RECORD 90 1 EEE EE 129 EE EE EE NE EE 131 CONFIGURE EEE EE 131 CREATE IVR OE EN 133 RECORD NEW IVR OE 133 UPLOAD IVR DROMDT ENEE 134 LANGUAGE SETTINGS FOR VOICE PROMPT cccccecsseesseeeeeeeees 135 DOWNLOAD AND INSTALL VOICE PROMPT PACKAGE 135 OG asap ts ses nes ae sc E epee ee ee paeeeeeeee 139 ENN 139 PS SO TE 140 VOICEMAIL EMAIL SETTINGS ENEE 141 CONFIGURE VOICEMAIL GROUP sssssssesssessssessseeesneesnecesneesnecesneesneesneeesneesneesesesneesensesneesneeenesee 143 HING GROUP seere 145 CONFIGURE RING GROUP 145 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 3 of 233 Gan Innovative IP Voic PAGING AND INTERCOM GROUP nnnunnnnnnnennnnnnnnnnnnnnnnennnnnnnnennnnnnnunnen 147 CONFIGURE PAGING
13. Innovative IP Voice amp Video Note This setting doesn t apply 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 RIP 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 Configure whether the Remote Party ID should be sent or not The default setting is No Configure whether the UCM6510 should generate inband ringing or not The default setting is Never e Yes The UCM6510 will send 180 Ringing followed by 183 Session Progress and in band audio e No The UCM6510 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 UCM6510 will send 180 Ringing as long as 2000K 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 UCM6510 If enabled compact SIP headers will be sent The default setting is No If enabled user phone will be added to URI that contains a valid phone number The default setting is No UCM6510 IP PBX User Manual Page 192 of 233 PBX STATUS andstream Innovative IP Voice amp Video STATUS AND REPORTING The UCM6510 monitors the status for Trunks Extensions Queue
14. Note When one type is selected you might not be able to dial another class of numbers For example if National is configured you won t be able to dial local or international numbers This setting is used to specify the type of the caller number The service provider will usually verify this Configure the prefix in PRI Local Dial Plan and PRI Dial Plan for each type Select the PRI Indication e outofoand Use RELEASE DISCONNECT or other messages with CAUSE to indicate call progress e g cause unassigned number or user busy e inband use in band tones to play busy or congestion signal to the other side This is the default setting The interval that restarts idle channels This setting is used to set up the ChannellD in SETUP message If enabled only the specified B channel can be used Otherwise select one of the channels in B channel If you need override the existing channels selection routine and force all PRI channels to be marked as exclusively selected please enable it If selected transmission of facility based ISDN supplementary services such as caller name from CPE over facility will be enabled Some switches AT amp T especially require network specific facility Currently the supported values are none sdn megacom tollfreemegacom accunet Table 30 Digital Hardware Configuration Parameters T1 J1 SS7 Clock Firmware Version 1 0 0 25 All E1 T1 J1 spans generate a clock signa
15. SD Card LAN WAN LAN PoE Heartbeat Power 1 Power 2 Digital FXS and FXO ports Table 67 Interface Status Indicators Disconnected Connected but not configured Connected and idle gt P Connected and in use FXS Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 197 of 233 a Connected but not configured SES Connected and idle amp Connected and in use SD Card SD Card plugged in SD Card unplugged d USB plugged in d USB unplugged LAN PoE i PoE is used WR L PoE is not used Power 1 2 m Power supply is working e Power supply is abnormal E H lt No power supply LAN WAN Heart Beat Not connected Digital Port T1 E1 J1 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual stream Innovative IP Voice amp Video Page 198 of 233 ndstream Innovative IP Voice amp Video Connected and working RED alarm there is physical wiring problem loss of connectivity or a framing line coding mismatch with the remote switch YELLOW alarm connected but the link is working only one way This means that the remote switch is not able to maintain sync with you or is not receiving your transmission The following example scenarios could trigger fi YELLOW alarm l The T1 port is connected with Ji connection 2 Incorrect cable is used 3 When using E1 one end is using CRC4 while the other end is not BLUE alarm th
16. billsec 3 disposition ANSWERED amaflags DOCUMENTATION uniqueid 1386100901 0 userfield EXT channel ext 5300 dstchannel ext 5301 service s Acctld 64 accountcode src 5300 dst 5301 dcontext from internal clid on01 lt 5300 gt channel SIP 5300 00000002 dstchannel SIP 5301 00000003 lastapp Dial lastdata SIP 5301 60 start 2013 12 03 14 02 23 answer 2013 12 03 14 02 27 end 2013 12 03 14 02 31 duration 8 billsec 4 disposition ANSWERED amaflags DOCUMENTATION uniqueid 1386100943 2 userfield EXT channel ext 5300 dstchannel evt 5301 service s Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 218 of 233 clean Innovative IP Voice amp Video UPGRADING AND MAINTENANCE UPGRADING The UCM6510 can be upgraded to a new firmware version remotely or locally This section describes how to upgrade your UCM6510 via network or local upload UPGRADING VIA NETWORK The UCM6510 can be upgraded via TFTP HTTP HTTPS by configuring the URL IP Address for the TFTP HTTP HTTPS server and selecting a download method Configure a valid URL for TFTP HTTP or HTTPS the server name can be FQDN or IP address Examples of valid URLs firmware grandstream com The upgrading configuration can be accessed via web GUI gt Maintenance gt Upgrade Upgrade Firmware Network Up
17. conference room and parking lot For example on the user s phone configure the parking lot number 701 as the BLF monitored number When there is a parked call on 701 the LED for this BLF key will light up in red meaning a call is parked against this parking lot Pressing this BLF key can pick up the call from this parking lot A Note e On the Grandstream GXP phones the MPK supports Call Park mode which is normally used to park the call by configuring the MPK number as call park feature code e g 700 Users could also use Call Park mode to monitor and pick up the call on this parking lot by configuring the MPK number as parking lot number e g 701 EVENT LIST Besides BLF users can also configure the phones to monitor event list By using event list local extensions on the same UCM6510 or remote extensions on the VOIP trunk can be monitored The event list settings is under web GUI gt Call Features gt Event List e Click on Create New Event List to add a new event list e Clickon to edit the event list configuration e Click on to delete the event list Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 165 of 233 Create New Event List GJ Local Extensions Available Extensions 800 801 i Remote Extensions Available Extensions URI 1 Selected Extensions Selected Extensions ndstream Innovative IP Voice amp Video F I URI Local Extensions Remote Extensions Spec
18. e User Extensions 1000 6299 User Extensions is referring to the extensions created under web UI gt PBX gt Basic Call Routes gt Extensions page e Pick Extensions 4000 4999 This refers to the extensions that can be manually picked from end device when being provisioned by the UCM6510 There are two related options in zero config page gt Auto Provision Settings Pick Extension Segment and Enable Pick Extension If Enable Pick Extension under zero config settings is selected the extension list defined in Pick Extension Segment will be sent out to the device after receiving the device s request This Pick Extension Segment should be a subset of the Pick Extensions range here This feature is for the GXP series phones that support selecting extension to be provisioned via phone s LCD e Auto Provision Extensions 5000 6299 UCM6510 IP PBX User Manual Page 180 of 233 clean Innovative IP Voice amp Video This sets the range for Zero Config Extension Segment which is the extensions can be assigned on the UCM6510 to provision the end device e Conference Extensions 6300 6399 e Ring Group Extensions 6400 6499 e Queue Extensions 6500 6599 e Voicemail Group Extensions 6600 6699 e IVR Extensions 7000 7100 e Fax Extensions 7200 8200 INTERNAL OPTIONS JITTER BUFFER Table 52 Internal Options Jitter Buffer SIP Jitter Buffer Select to enable jitter buffer on the sending side of the SIP channel The default sett
19. strong password will be enforced for the password created on the UCM6510 The default setting is enabled Strong Password Rules 1 Password for voicemail voicemail group outbound route DISA call queue and conference requires non repetitive and non sequential digits with a minimum length of 4 digits Repetitive digits pattern UCM6510 IP PBX User Manual Page 179 of 233 Enable Random Password Disable Extension Range Extension Ranges Firmware Version 1 0 0 25 Deg Innovative IP Voice amp Video such as 0000 1111 1234 2345 and etc or common digits pattern such as 111222 321321 and etc are not allowed to be configured as password 2 Password for extension registration web GUI admin login LDAP and LDAP sync requires alphanumeric characters containing at least two categories of the following with a minimum length of 4 characters Numeric digits Lowercase alphabet characters Uppercase alphabet characters Special characters 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 UCM6510 The default setting is No The default extension range assignment is Note It is recommended to keep the system assignment to avoid inappropriate usage and unnecessary issues The default extension range assignment Is
20. 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 CallerID Name configured Select audio and video codec for the VolP trunk The available codecs are PCMU PCMA GSM AAL2 G 726 32 G 726 G 722 G 729 G 723 ILBC ADPCM H 264 H 263 H 263p If enabled the UCM6510 will regularly send SIP OPTIONS to the device to check if the device is still online The default setting is No When Enable Qualify option is set to Yes configure the timeout in ms for the Qualify SIP message If no response is received within the timeout the device is considered offline The default setting is 1000ms When Enable Qualify option is set to Yes configure the interval in seconds of the SIP OPTIONS message sent to the device to check if the device is still online The default setting is 60 seconds 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 UCM6510 IP PBX User Manual Page 112 of 233 san Innovative IP Voice amp Video Register IAX Trunk Configuration Parameters Provider Name Host Name Keep Trunk CID Caller ID CalleriID Name Username Password Auth ID Codec Preference Enable Qualify Qualify Timeout Qualify Frequency
21. 192 168 40 0 Gateway 192 168 69 1 i gt 192 168 40 126 392 169 03 131 ag f LAN 1 192 168 69 9 a GXP1160 EES Network 192 168 69 0 UCM6510 Network settings method Dual mode LAN 1 connected to 192 168 69 0 network GXP2130 and GXP1160 LAN 2 connected to 192 168 40 0 network Call each other Figure 12 UCM6510 Static Route Sample The network topology of the above diagram is as below e Network 192 168 69 0 has IP phones registered to UCM6510 LAN 1 address e Network 192 168 40 0 has IP phones registered to UCM6510 LAN 2 address e Network 192 168 66 0 has IP phones registered to UCM6510 via VPN e Network 192 168 40 0 has VPN connection established with network 192 168 66 0 In this network by default the IP phones in network 192 168 69 0 are unable to call IP phones in network 192 168 66 0 when registered on different interfaces on the UCM6510 Therefore we need configure a static route on the UCM6510 so that the phones in isolated networks can make calls between each other Create New Static Route G Destination 192 168 66 0 G Netmask 255 255 255 0 G Gateway 192 168 40 3 G Interface LAN2 Figure 13 UCM6510 Static Route Configuration Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 39 of 233 E ten Innovative IP Voice amp Video PORT FORWORDING The UCM6510 network interface supports router functions which provides users the ability to do port forwarding If t
22. 25 UCM6510 IP PBX User Manual Page 6 of 233 E ten Innovative IP Voice amp Video Table of Tables UCM6510 IP PBX User Manual Tale 1 lechnical ee et Te ssireirsnsiinineiiiiomaa in naa EEN ADE EIEE EA EET 16 Table 2 UCM6510 Equipment Packagmng 19 Tabl e 3 LOD Menu t 24 Table 4 UCM6510 LED INDICATORS murrrnnnnnronrnnnnnnvrnrnnnnnsvnnnnnnnnsennvnnnnnsennvnnnnnsnnnnnnnnnrennnnnnnnnnnnnnnnnsennnnnnenee 25 Table 5 UCM6510 Network Settings gt Basic Settings cccccccccccsseceecceesceeceeeeeeeseeeceeceeceessaeeesseeeessaaes 31 Table 6 UCM6510 Network Gettmgs 20775 37 Table 7 UCM6510 Network Settings gt Static Routes rrrrrrrrnnnnnrrrrnnnnnvonrrnnnnnvennrnnnnnrennnnnnnnrenrnnnnnsrnnnnnnenee 38 Table 8 UCM6510 Network Settings gt Port Fonwardmg 40 Table 9 UCM6510 Firewall gt Static Defense gt Current Service ccccceeecceceeeececeeeceeceeeceesaeeeeeeseeeeessaees 42 Table 10 Typical Firewall Settings cccccccccccsscceceeseceeeeeceeceeceesseeecesseeceeseeaeeeseeaeceeseaeeessaeeesseeeessaaes 43 Table 11 Firewall Rule e EE 43 Table 12 UCM6510 Firewall Dynamic Detense 44 Table 13 PIPPI 46 Table 14 HTTP Server Gettngs 53 Taole do EMail 1 EEE EEE eee ae ete nee 53 Table 16 Auto Time Updating n00nnn00nnnnnannnnannnnnannnonnnnnsannnnnnrrnsnnnrrnnrrennnrrrnnrrnnnnrrrnnrrrnnnrrrnnrrronnrrrnnrnrnnnnrnne 56 Table 1 Auto Provision SENGS vasre dere 61 Table 18 SIP Extension Configuration Par
23. 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 Options Voice 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 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 207 of 233 ndstream Innovative IP Voice amp Video Call Detail Report Inbound calls Caller Number Caller Name Outbound calls From Date To Date Internal calls External calls Figure 99 CDR Filter Table 72 CDR Filter Criteria Inbound calls Inbound calls are calls originated from a non internal source like a VoIP trunk and sent to an internal extension Outbound calls Outbound calls are calls sent to a non internal source like a VoIP trunk from an internal extension Internal calls Internal calls are calls from one internal extension to another extension which are not sent over a trunk External calls External calls are calls sent from one trunk to another trunk which are not sent to any internal extension 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 s
24. 72 Configure Extension For Fax Machine 7 Goto web GUI gt PBX gt Basic Call Routes gt Inbound Routes page 8 Create an inbound route to use the Fax analog trunk Select the created extension for Fax machine in step 4 as the default destination Create New Inbound Rule Trunks AnalogTrunks FAX LINE GJ DID Pattern i Default Destination Extension Time Condition Click to add Time Condition Cancel Save Figure 73 Configure Inbound Rule for Fax Now the Fax configuration is done When there is an incoming Fax calling to the PSTN number for the FXO port it will send the Fax to the Fax machine Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 161 of 233 andstream Innovative IP Voice amp Video SAMPLE CONFIGURATION FOR FAX TO EMAIL The following instructions describes a sample configuration on how to use Fax to Email feature on the UCM6510 1 Connect PSTN line to the UCM6510 FXO port 2 Goto UCM6510 web GUI gt Internal Options gt Fax T 38 page Create a new Fax extension Create New Fax Extension Extension Name Fax G Email Address Faxtest ucm6510mycompany com Save Figure 74 Create Fax Extension 3 Go to UCM6510 web GUI gt Basic Call Routes gt Analog Trunks page Create a new analog trunk with FAX Detection set to No 4 Goto UCM6510 web GUI gt Basic Call Routes gt Inbound Routes page Create a new inbound route and set the default destination to the Fax extensi
25. DNS Server 1 0 0 0 0 DNS Server 2 Enter the DNS server 2 address for static IP settings User Name Enter the user name to connect via PPPoE Password Enter the password to connect via PPPoE Layer 2 QoS Assign the VLAN tag of the layer 2 QoS packets for LAN port The default value is 802 1Q VLAN Tag 0 Layer 2 QoS 802 1p Assign the priority value of the layer 2 QoS packets for LAN port The default Priority Value value is 0 e Method Route WAN port interface is used for uplink connection LAN port interface is used as a router Please see a sample diagram below Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 33 of 233 andstream Innovative IP Voice amp Video Method Route A 192 168 2 x 721825 m LAN Port 3 WAN Port DHCP Server 192 168 1 1 IP address 172 18 25 10 Figure 7 UCM6510 Network Interface Method Route e Method Switch WAN port interface is used for uplink connection LAN port interface is used as bridge for PC connection Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 34 of 233 fream Innovative IP Voice amp Video 2 LAN Internet e Ethernet 0 0 IP Address 192 168 40 1 I STEE EE Subnet mask 255 255 255 0 Sy PC 192 168 40 15 WAN Port IP address 192 168 40 10 Figure 8 UCM6510 Network Interface Method Switch e Method Dual Both WAN port and LAN port are used for uplink connection WAN port will be mapped to LAN 1 interface LAN port will be mapped
26. IP PBX User Manual Page 56 of 233 dstream Innovative IP Voice amp Video SET TIME MANUALLY To manually set the time on the UCM6510 go to Web GUI gt Settings gt Time Settings gt Set Time Manually The format is YYYY MM DD HH MI SS set Time Manually G Current Time Figure 26 Set Time Manually A Note Manually setup time will take effect immediately after saving and applying change in the web Ul If users would like to reboot the UCM6510 and keep the manually setup time setting please make sure Remote NTP Server Enable DHCP Option 2 and Enable DHCP Option 42 options under Web GUI gt Settings gt Time Settings gt Time Auto Updating page are unchecked or set to empty Otherwise time auto updating settings in this page will take effect after reboot Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 57 of 233 Deg Innovative IP Voice amp Video NTP SERVER The UCM6510 can be used as a NIP server for the NTP clients to synchronize their time with To configure the UCM6510 as the NTP server set Enable NTP server to Yes under web GUI gt Settings gt Time Settings gt NTP Server On the client side point the NTP server address to the UCM6510 IP address or host name to use the UCM6510 as the NTP server Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 58 of 233 E ten Innovative IP Voice amp Video PROVISIONING OVERVIEW Grandstream SIP Devices can be configured via web interf
27. If not configured the Call Forward No Answer feature is deactivated The default setting is deactivated Configure the Call Forward Busy target number so that the incoming call to this extension will be forwarded to the target number if the call is rejected or the extension is in talking busy status If not configured the Call Forward Busy feature is deactivated The default setting 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 If not specified the default ring timeout is 60 seconds on the UCM6510 which can be configured in the global ring timeout setting under web GUI gt Internal Options General Preference The valid range is between 5 seconds and 600 seconds Note If the end point also has a ring timeout configured the actual ring timeout used is the shortest time set by either device Enable automatic recording for the calls using this extension The default setting is disabled The recording files will be saved in external storage if plugged in and can be accessed under web GUI gt CDR gt Recording Files When user dials voicemail code the password verification IVR is skipped If enabled this would allow one button voicemail access By default this option is disabled This mode can be used for devices that support hot desking feature For example the GXP21xx series phones support hot desking feature by t
28. PBX User Manual Page 282 of 233 ndstream Innovative IP Voice amp Video EXPERIENCING THE UCM6510 SERIES IP PBX Please visit our website http Wwww grandstream com to receive the most up to date updates on firmware releases additional features FAQs documentation and news on new products We encourage you to browse our product related documentation FAQs and User and Developer Forum for answers to your general questions If you have purchased our products through a Grandstream Certified Partner or Reseller please contact them directly for immediate support 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 UCM6510 IP PBX appliance 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 0 25 UCM6510 IP PBX User Manual Page 233 of 233
29. Prompt Invalid Prompt Response Timeout Repeat Loops Invalid Repeat Loops Language Key Press Event Firmware Version 1 0 0 25 E ten Innovative IP Voice amp Video 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 pressed Configure the number of times to repeat the prompt if no DTMF input is detected When the loop ends it will go to the timeout destination if configured or hang up The default setting is 3 Configure the number of times to 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 3 Select the voice prompt language to be used for this IVR The default setting is Default which is the selected voice prompt language under web GUI gt PBX gt Internal Options gt Language The dropdown list shows all the current available voice prompt languages on the UCM6510 To add more languages in the list please download voice prompt package by selecting Check Prompt List under web GUI gt PBX gt Internal Options gt Language Select the event for each key pressing for 0 9 Timeout and Invalid The event options are e Extension e Voicemail e Conference Rooms e Voicemai
30. The default setting is disabled The recording files are saved in Auto Record De l external storage device if plugged in and can be accessed under web GUI gt CDR gt Recording Files 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 Fax Detection Note If enabled Fax Pass through cannot be used DIRECT OUTWARD DIALING DOD VIA DIGITAL TRUNKS Please refer to section DIRECT OUTWARD DIALING DOD VIA VOIP TRUNKS DIGITAL TRUNK TROUBLESHOOTING After configuring the digital trunk on the UCM6510 as described above if it doesn t work as expected users can go to capture signaling trace on the UCM6510 web UI for troubleshooting purpose Depending on the signaling selected for the digital trunk users can go to the following pages to capture trace PRI Signaling Trace web GUI gt Maintenance gt Troubleshooting gt PRI Signaling Trace SS7 Signaling Trace web GUI gt Maintenance gt Troubleshooting gt SS7 Signaling Trace MFC R2 Signaling Trace web GUI gt Maintenance gt Troubleshooting gt MFC R2 Signaling Trace Here is the step to capture trace 1 Click on Start to start capturing trace The output result shows Capturing 2 Once the test is done click on Stop to stop the trace 3 Click on Download to download the trace Firmware Version 1 0 0 25 UCM6510 IP PBX User M
31. Whitelist Blacklist strong password based access control Advanced Defense Physical Universal Power Supply Input 100 240VAC 50 60Hz Output DC 12VDC 1 5A Physical Unit Weight 2 165 KG Package weight 3 012 KG Dimensions 440mm L x 185mm W x 44mm H Operating 32 113 F 0 45 C Humidity 10 90 non condensing Storage 14 140 F 10 60 C Humidity 10 90 non condensing Mounting Rack mount and Desktop Environmental Additional Features English Simplified Chinese Traditional Chinese Spanish French Portuguese German Russian Italian Polish Czech for web GUI Multi language Support Customizable IVR voice prompts for English Chinese British English German Spanish Greek French Italian Dutch Polish Portuguese Russian Swedish Turkish Hebrew and Arabic Bellcore Telcordia ETSI FSK ETSI DTMF SIN 227 BT NTT Japan Caller ID pending Polarity Reversal Wink Yes with enable disable option upon call establishment and termination Multiple configurable call queues automatic call distribution ACD Call Center based on agent skills availability busy level in queue announcement Customizable Auto Attendant Up to 5 layers of IVR Interactive Voice Response Maximum Call Capacity Up to 2000 registered SIP endpoints up to 200 concurrent calls Conference Bridges Up to 8 bridges up to 64 simultaneous conference attendees Call park call forward call transfer DND DISA ring group picku
32. amp Video modify the remote contacts by adding this prefix Register SIP Trunk Configuration Parameters Provider Name Host Name Transport Keep Trunk CID Username Password Auth ID Codec Preference From Domain From User Firmware Version 1 0 0 25 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 VolP provider server of the trunk Configure the SIP transport protocol to be used in this trunk The default setting is All UDP Primary e UDP Only e TCP Only e TLS Only e All UDP Primary UDP is the primary transport protocol when all the other SIP transport methods are available too e All TCP Primary TCP is the primary transport protocol when all the other SIP transport methods are available too e All TLS Primary TLS is the primary transport protocol when all the other SIP transport methods are available too When enabled it can avoid overridden by extension s CID if the extension has CID configured The default setting is enabled Enter the username to register to the trunk from the provider Enter the password to register to the trunk from the provider This is the authentication ID for the UCM6510 to register to the trunk if required by the provider If not specified the CallerID name will be used for authentication Select audio and video codec fo
33. and void the manufacturer warranty Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 13 of 233 ndstream Innovative IP Voice amp Video This document is subject to change without notice The latest electronic version of this user manual is available for download here http www grandstream com support Reproduction or transmittal of the entire or any part in any form or by any means electronic or print for any purpose without the express written permission of Grandstream Networks Inc is not permitted Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 14 of 233 E ten Innovative IP Voice amp Video PRODUCT OVERVIEW FEATURE HIGHTLIGHTS e 1 GHz quad core Cortex A9 application processor large memory 1GB DDR3 RAM 32GB Flash and dedicated high performance multi core DSP array for advanced voice processing e 1 Integrated 1 T1 E1 J1 interface 2 PSTN trunk FXO ports 2 analog telephone Fax FXS ports with lifeline capability in case of power outage and up to 50 SIP trunk accounts e Hardware DSP based 128ms tail length carrier grade line echo cancellation LEC hardware based caller ID call progress tone and smart automated impedance matching for various countries e Gigabit network port s with integrated PoE USB SD card integrated NAT router with advanced QoS support e Strong defense against malicious attacks Fail2ban Whitelist Blacklist alerts etc e Data communication via T1 E1 J1 and data voice
34. answer or hang up the call to finish the detecting process 6 Once done the detected result will show Users could save the detecting result as the current UCM6510 settings Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 89 of 233 ndstream Innovative IP Voice amp Video Table 24 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 UCM6510 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 UCM6510 will control the call to be established and hang up between caller and callee to finish the detection Detect Model e Semi auto Detect Semi auto detection requires answering or hanging up the call manually Please make sure one channel is connected to the UCM6510 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 Destination Number Configure the number to be called to help the detection A Note e The PSTN
35. be listed as static agent and can log in log out at any time e Call queue feature code Agent Pause and Agent Unpause can be configured under web GUI gt PBX gt Internal Options gt Feature Codes The default feature code is 83 for Agent Pause and 84 for Agent Unpause Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 151 of 233 Andstream Innovative IP Voice amp Video This page intentionally left blank Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 152 of 233 Andstream Innovative IP Voice amp Video EXTENSION GROUPS The UCM6510 extension group feature allows users to assign extensions to different groups to better manage the configurations on the PBX For example when configuring Enable Filter on Source Caller ID users could select a group instead of each person s extension to assign This feature simplifies the configuration process and helps manage and categorize the extensions for business environment CONFIGURE EXTENSION GROUPS Extension group can be configured via web GUI gt PBX gt Call Features gt Extension Groups e Click on Create New Extension Group to create a new extension group e Click on to edit the extension group Select extensions from the list on the left side to the right side Edit Extension Group AccountingDept AccountingDept Available Extensions Selected Extensions Figure 67 Edit Extension Group e Click on to delete the extension gro
36. can contain characters letters digits and Fill in the Email address for the user Voicemail will be sent to this Email address Select the voice prompt language to be used for this extension The default setting is Default which is the selected voice prompt language under web GUI gt PBX gt Internal Options gt Language The dropdown list shows all the current available voice prompt languages on the UCM6510 To add more languages in the list please download voice prompt package by selecting Check Prompt List under web GUI gt PBX gt Internal Options gt Language Analog Settings Call Waiting User as SEND RX Gain TX Gain Firmware Version 1 0 0 25 Configure to enable disable call waiting feature for the FXS extension When enabled the FXS extension currently in an active call allows a new call to come in and can hear call waiting tone on the new incoming call The default setting is No If configured the key can be used as SEND key after dialing the number on the analog phone The default setting is Yes Configure the RX gain for the receiving channel of analog FXS port The valid range is 30dB to 6dB The default setting is 0 Configure the TX gain for the transmitting channel of analog FXS port UCM6510 IP PBX User Manual Page 73 of 233 MIN RX Flash MAX RX Flash Enable Polarity Reversal Echo Cancellation 3 Way Calling Send CallerID After dstream Innovative IP Voice amp Vid
37. combined communication via T1 E1 J1 with SS7 or PRI e Supports up to 2000 SIP endpoint registrations up to 200 concurrent calls up to 100 SRIP encrypted concurrent calls and up to 64 conference attendees e Flexible dial plan call routing site peering call recording manual and automatic per SIP call and SIP trunk central control panel for endpoints integrated NTP server and integrated LDAP contact directory e Automated detection and provisioning of IP phones video phones ATAs gateways SIP cameras and other endpoints for easy deployment e Strongest possible security protection using SRIP TLS and HTTPS with hardware encryption accelerator e Redundant power supply advanced support for Hot Standby Clustering and High Availability to minimize system down time pending e Automatic export of previous day s data periodically cleans up user data Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 15 of 233 tream Innovative IP Voice amp Video TECHNICAL SPECIFICATIONS Table 1 Technical Specifications Analog Telephone FXS Ports 2 RJ11 ports both with lifetime capability in case of power outage PSTN Line FXO Ports 2 RJ11 ports both with lifeline capability in case of power outage T1 E1 J1 Interface 1 RJ45 port Dual Gigabit ports switched or routed with PoE Network Interfaces A 3 Gigabit port for Hot Standby Clustering NAT Router Yes user configurable Peripheral Ports USB SD Power 1 2 PoE US
38. connection or experience unstable connection click on to reconnect to help resolve the problem 6 Users can always click on ON OFF switch wal ON in the web page to enable disable the configured data trunk PBX gt gt Basic Call Routes gt gt Data Trunk Data Trunk Configure digital channels for data communication Sometimes the line will have the problem of synchronization Please try to reconnect Jorr 1 Figure 46 Data Trunk Web Page Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 105 of 233 Data Trunk am I fi Data Enable D Channel Group G Encapsulation Local IP i Subnet Mask Remote IF i DNS Server 1 fi DNS Server 2 fi Default Interface Data Enable Channel Group Encapsulation Local IP Subnet Mask Remote IP DNS Server 1 DNS Server 2 Default Interface Firmware Version 1 0 0 25 ndstream Innovative IP Voice amp Video all test HDLC 10 10 10 10 255 255 255 0 10 10 10 11 4222 42 21 Figure 47 Data Trunk Configuration Table 32 Data Trunk Configuration Parameters Select the checkbox to enable disable data trunk Users can also click on the ON OFF switch in data trunk web page to enable disable this Select the digital channel group from the dropdown list to be used for data trunk Users will need create a new group under web UI gt PBX gt Ports Config gt Digital Hardware page for this purpose Select the pro
39. e Click the Record button A request will be sent to the UCM6510 The UCM6510 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 UCM6510 The following are required for the IVR prompt file to be successfully uploaded and used by the UCM6510 e PCM encoded e 16 bits e 8000Hz mono e In mp3 or wav format or raw ulaw alaw gsm file with ulaw or alaw suffix e File size under 5M Upload IVR Prompt Choose voice prompt to upload Sound file must be PCM encoded 16 bits at 000HzZ mono with mp3 wav format or raw ulaw alaw gsm file with ulaw _alaw suffix The file size must be under 5M Choose file to upload as O Upload Figure 55 Upload IVR Prompt Click on __ to select audio file from local PC and click on t to start uploading Once uploaded the file will appear in the IVR Prompt web page Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 134 of 233 E ten Innovative IP Voice amp Video LANGUAGE SETTINGS FOR VOICE PROMPT The UCM6510 supports multiple languages in web GUI as well as sy
40. enabled the UCM6510 will regularly send SIP OPTIONS to the device to check if the device is still online The default setting is No When Enable Qualify option is set to Yes configure the timeout in ms for the Qualify SIP message If no response is received within the timeout the device is considered offline The default setting is 1000ms When Enable Qualify option is set to Yes configure the interval in seconds of the SIP OPTIONS message sent to the device to check if the device is still online The default setting is 60 seconds 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 Table 34 IAX Trunk Configuration Parameters Create New IAX Trunk Type Provider Name Host Name Keep Trunk CID Username Firmware Version 1 0 0 25 Select the VolP trunk type e Peer IAX Trunk e Register IAX Trunk Configure a unique label to identify this trunk when listed in outbound rules inbound rules and etc Configure the IP address or URL for the VolP provider s 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 No Enter the username to register to the trunk from the
41. file for the call Click on to play the recording file click on to download the recording file in wav format click on to delete the recording file the call record entry will not be deleted 2013 07 03 18 27 47 2013 07 03 17 55 04 Figure 101 Call Report Entry with Audio Recording File DOWNLOADED CDR FILE Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 209 of 233 ndstream Innovative IP Voice amp Video The downloaded CDR csv file has different format from the web Ul CDR Here are some descriptions e Call From Call To Call From the caller ID Call To the callee ID If Call From shows empty Call To shows s see highlight part in the picture below and the Source Channel contains DAHDI this means the call is from FXO PSTN line For FXO PSTN line we only know there is an incoming request when there is incoming call but we don t know the number being called So we are using s to match it where s means start call from call to context start time answer time end time call time talk time source channel dest channel status 610 19097622990 from internal 1 29 2014 14 28 1 29 2014 14 28 1 29 2014 14 31 153 150 SIP 610 00000074 DAHDI 1 1 ANSWERED default 1 29 2014 14 33 1 29 2014 14 33 0 DAHDI pseudo 149089967 default 1 29 2014 14 33 1 29 2014 14 33 0 DAHDI pseudo 1067045536 688 from internal 1 29 2014 14 33 1 29 2014 14 33 1 29 2014 14 33 9 SIP 601 00000077 ANSWERED default 1
42. provider when Register IAX Trunk type is selected UCM6510 IP PBX User Manual Page 111 of 233 Password E ten Innovative IP Voice amp Video Enter the password to register to the trunk from the provider when Register IAX Trunk type is selected Peer IAX Trunk Configuration Parameters Provider Name Host Name Keep Trunk CID Caller ID CalleriID Name Codec Preference Enable Qualify Qualify Timeout Qualify Frequency Fax Detection Firmware Version 1 0 0 25 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 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 CallerID with this option and this option will be ignored When making outgoing calls the following rules are used to determine which CallerID will be used if they exist e The CallerID configured for the extension will be looked up first e f Keep Trunk CID is enabled no CallerlD configured for the extension the CallerID configured for the trunk will be used e If the above two are missing the Global Outbound CID defined in
43. ten Innovative IP Voice amp Video CDR API CONFIGURATION FILES The UCM6510 supports third party billing interface API for external billing software to access CDR on the PBX The API uses HTTPS to request the CDR data matching given parameters as configured on the third party application Before accessing the API the administrators need enable API and configure the access authentication information on the UCM6510 first Table 74 CDR API Configuration Files Enable Enable Disable CDR API The default setting is disabled TLS Bind Address Configure the IP address for TLS server to bind to 0 0 0 0 means binding to all interfaces The port number is optional and the default port number is 8443 The IP address must match the common name host name in the certificate so that the TLS socket won t bind to multiple IP addresses The default setting is 0 0 0 0 8443 TLS Private Key Upload TLS private key The size of the key file must be under 2MB This file will be renamed as private pem automatically TLS Cert Upload TLS cert The size of the certificate must be under 2MB This is the certificate file pem format only for TLS connection This file will be renamed as certificate pem automatically It contains private key for the client and signed certificate for the server TLS Authentication Configure the user name for TLS authentication If not configured authentication will Name be skipped TLS Authentication Le we Configure t
44. the RX gain for the receiving channel of digital port The valid range is from 24dB to 12dB TX Gain Configure the TX Gain for the transmitting channel of digital port The valid range is 24dB to 12dB Codec Select alaw or ulaw If set to default ulaw will be used for T1 J1 Advanced Settings WE Coding Select B8ZS or AMI Indicates the type of the called number The receiving switch may use this Called Nature of Address indicator during translations to apply the number s proper dial plan Users Indicator can select Unknown Subscriber National International or Dynamic Indicates the type of the calling number The receiving switch may use Calling Nature of Address this indicator during translations to apply the number s proper dial plan Indicator Users can select Unknown Subscriber National International or Dynamic International Prefix Configure the prefix in PRI Local Dial Plan and PRI Dial Plan for each Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 101 of 233 E ten Innovative IP Voice amp Video National Prefix type Local Prefix Private Prefix Unknown Prefix DIGITAL TRUNK CONFIGURATION After configuring digital hardware go to web GUI gt PBX gt Basic Call Routes gt Digital Trunks e Click on Create New Digital Trunk to add a new digital trunk e Click on to configure detailed parameters for the digital trunk e Click on to configure Direct Outward Dialing DOD for the digi
45. the SMTP server ASMTP server is required and users need login with correct credentials Domain Specify the domain name to be used in the Email when using type MTA R Specify the SMTP server when using type Client For example if using Gmail as the SMTP server you can configure it as smtp gmail com 465 Username is required when using type Client Normally it s the Email Username address Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 53 of 233 Andstream Innovative IP Voice amp Video Password to log in for the above Username Email address is required Password when using type Client Display Name Specify the display name in the FROM header in the Email Specify the sender s Email address Sender For example pobx example mycompany com The following figure shows a sample Email settings on the UCM6510 assuming the Email is using smto gmail com as the SMTP server and the port number is 465 Email settings TLS Enable Yes Type Client Server smtp gmail com 465 Username pbx company gr Password Display Name Company PBX sender pbx rcompany gmail com Figure 24 UCM6510 Email Settings Once the configuration is finished click on Save first Then click on Test button to make sure the Email setting is working The following figure shows the new dialog prompted to test the Email setting Fill in a valid Email address to send a test Email to verify the Email settings on the UCM6510 Fi
46. the digits the IVR will announce 8 matching results The caller can press number 1 to 8 to select and call or press 9 for results in next page Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 171 of 233 Andstream Innovative IP Voice amp Video This page intentionally left blank Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 172 of 233 san Innovative IP Voice amp Video CALL FEATURES The UCM6510 supports call recording transfer call forward call park and other call features via feature code This section lists all the feature codes in the UCM6510 and describes how to use the call features FEATURE CODES Table 50 UCM6510 Feature Codes Feature Maps Blind Transfer Attended Transfer Disconnect Call Park Firmware Version 1 0 0 25 Default code 1 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 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 Default code 2 Enter the 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 Options Disable A
47. the password to access the ring group extension s voicemail Note The password has to be at least 4 characters 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 Edit Ring Group 6400 Ring Group Name Extension 656 657 658 659 660 bb Ring Group Options Ring Strategy shipping 6400 Available Extensions Ring Group Members Ring simultaneously e G Ring Timeout on Each 30 Member s Enable Destination GD Default Destination Firmware Version 1 0 0 25 L Hanvoiceenu T Figure 62 Ring Group Configuration UCM6510 IP PBX User Manual Page 146 of 233 PAGING AND INTERCOM GROUP Andstream Innovative IP Voice amp Video The UCM6510 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 Name Extension Type Page Intercom Group Members Create New Paging ntercom Group Name Shipping Extension 6770 Type 2 Way Intercom Available Extensions Paging Intercom Group Members 670 871 872 Figur
48. transfers initiated by the endpoint in the UCM6510 will be disabled unless enabled in peers or users The default setting is Yes Enable DNS SRV Lookups on Select to enables DNS SRV lookups on outbound calls from the Outbound Calls UCM6510 The default setting is Yes When sending MWI NOTIFY requests this value will be used in the Allow Transfer MWI From 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 SIP Domain Support Configure the domain for the UCM6510 Incoming INVITE and REFER Domain 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 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 187 of 233 andstream Innovative IP Voice amp Video 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 From Domain l DS be required by some providers for authentication If enabled the UCM6510 will add local host name and local IP to domain list The default setting is No Auto Domain If enabled requests for external domains that are not served by the UCM6510 will b
49. will be enabled Some switches AT amp T especially require network specific facility NSF Currently the supported values are none sdn megacom tollfreemegacom accunet Table 27 Digital Hardware Configuration Parameters E1 SS7 Basic Settings All E1 T1 J1 spans generate a clock signal on their transmit side The parameter determines whether the clock signal from the far end of the E1 T1 J1 is used as the master source of clock timing If the far end is used as the master the PBX system clock will synchronize to it Clock l 4 Ko e Master The port will never be used as a source of timing This is appropriate when you know the far end should always be a slave to you e Slave The equipment at the far end of the E1 T1 link is the preferred source of the master clock SS7 Variant Select ITU ANSI or CHINA Originating point code is used to identify the node originating the De H l message always provided by the operator ISP Originating Point Code l e ITU Format decimal number e ANSI amp CHINA Format decimal number or XXX XXX XXX Destination point code is the address to send the message to always be Ee l provided by the operator ISP Destination Point Code l e ITU Format decimal number e ANSI amp CHINA Format decimal number or XXX XXX XXX Network Indicator NI should match in nodes otherwise it might cause Network Indicator issues Users can select National National Spare Intern
50. 0 Please see details in the following table Table 70 System Status gt General Status gt System Status gt General Model Part Number System Time Up Time Idle Time Boot Core Base Program Recovery NETWORK Product model Product part number Current system time The current system time is also available on the upper right of each web page System up time since the last reboot System idle time since the last reboot Boot version Core version Base version Program version This is the main software release version Recovery version Under web GUI gt Status gt System Status gt Network users could check the network information for the UCM6510 Please see details in the following table Table 71 System Status gt Network Status gt System Status gt Network MAG Address IP Address Gateway Subnet Mask DNS Server Firmware Version 1 0 0 25 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 Default gateway address Subnet mask address DNS Server address UCM6510 IP PBX User Manual Page 202 of 233 STORAGE USAGE ndstream Innovative IP Voice amp Video 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
51. 0 0 25 UCM6510 IP PBX User Manual Page 11 of 233 Andstream Innovative IP Voice amp Video CHANGE LOG This section documents significant changes from previous versions of the UCM6510 user manual Only major new features or major document updates are listed here Minor updates for corrections or editing are not documented here FIRMWARE VERSION 1 0 0 25 e This is the initial version Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 12 of 233 san Innovative IP Voice amp Video WELCOME Thank you for purchasing Grandstream UCM6510 IP PBX appliance The UCM6510 is an innovative IP PBX appliance for E1 T1 J1 networks that brings enterprise grade unified communications and security protection to enterprises small to medium businesses SMBs retail environments and residential settings in an easy to manage fashion Powered by an advanced hardware platform and revolutionary software functionalities the UCM6510 offers a breakthrough turnkey solution for converged voice video data fax security surveillance and mobility applications out of the box without any extra license fees or recurring costs 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 adapter with the UCM6510 as it may cause damage to the products
52. 03 11 46 49 9 6 ANSWERED DOCUMENTATION 1386092800 0 EXT 5300 5301 s 63 5300 5301 from internal pn01 lt 5300 gt SIP 5300 00000000 SIP 5301 00000001 Dal SIP 5301 60 2013 12 03 14 01 41 2013 12 03 14 01 43 2013 12 03 14 01 46 5 3 ANSWERED DOCUMENTATION 1386100901 0 EXT 5300 5301 s 64 5300 5301 from internal pn01 lt 5300 gt SIP 5300 00000002 SIP 5301 00000003 Dial SIP 5301 60 2013 12 03 14 02 23 2013 12 03 14 02 27 2013 12 03 14 02 31 8 4 ANSWERED DOCUMENTATION 1386100943 2 EXT 5300 5301 s Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 217 of 233 ndstream Innovative IP Voice amp Video lt root gt lt cdr gt lt Acctld gt 62 lt AcctId gt lt accountcode gt lt accountcode gt lt src gt 5300 lt src gt lt dst gt 5301 lt dst gt lt dcontext gt from internal lt dcontext gt lt clid gt amp quot pn01 amp quot amp lt 5300 amp gt lt clid gt lt channel gt SIP 5300 00000000 lt channel gt lt dstchannel gt SIP 5301 00000001 lt dstchannel gt lt lastapp gt Dial lt lastapp gt lt lastdata gt SIP 5301 60 lt lastdata gt lt start gt 2013 12 03 11 46 40 lt start gt lt answer gt 2013 12 03 11 46 43 lt answer gt lt end gt 2013 12 03 11 46 49 lt end gt lt duration gt 9 lt duration gt lt billsec gt 6 lt billsec gt lt disposition gt ANSWERED lt disposition gt lt a maflags gt DOCUMENTATION lt amaflags gt lt uniqueid gt 1386092800 0 lt uniqueid gt lt userfield gt EXT lt userfie ld gt lt chann
53. 25 UCM6510 IP PBX User Manual Page 60 of 233 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 Auto provision automaticaly provides an extension to the device There are three methods of auto provision SIP SUBSCRIBE DHCP Option 66 and mDNS For example when the device boots up it will send SIP SUBSCRIBE multicast in the LAN The PBX will find it create an account and return a URL of the config file for the device to download Enable ero Config Automaticaly Assign Extension Zero Config Extension Segment Enable Pick Extension Pick Extension Segment Pick Extension Period hour Enable Zero Config Automatically Assign Extension Zero Config Extension Segment Enable Pick Extension Firmware Version 1 0 0 25 5000 6299 Zero Config Extension Segment Pick Extension Segment Figure 28 Auto Provision Settings Table 17 Auto Provision Settings Enable or disable the zero config feature on the PBX The default setting is enabled If enabled when the device is discovered the PBX will automatically assign an extension within the range defined in Zero Config Extension Segment to the device The default setting is disabled Click on the link Zero Config Extension Segment to specify the extensi
54. 29 2014 14 34 1 29 2014 14 34 0 DAHDI pseudo 1124093033 NO ANSWER default 1 29 2014 14 34 1 29 2014 14 34 0 DAHDI pseudo 1719498666 NO ANSWER Figure 102 Downloaded CDR File Sample Call To Shows s e Context There are different context values that might show up in the downloaded CDR file The actual value can vary case by case Here are some sample values and their descriptions from internal internal extension makes outbound calls ext did XXXXX inbound calls It starts with ext did and XXXXX content varies case by case which also relate to the order when the trunk is created ext local internal calls between local extensions e Source Channel Dest Channel Sample 1 DAHDI means it is an analog call FXO or FXS For UCM6510 DAHDI 1 2 are FXO ports and DAHDI 3 4 are FXS ports For UCM6510 DAHDI 1 4 are FXO ports and DAHDI 5 6 are FXS ports For UCM6510 DAHDI 1 8 are FXO ports and DAHDI 9 10 are FXS ports For UCM6510 DAHDI 1 16 are FXO ports and DAHDI 17 18 are FXS ports Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 210 of 233 ndstream Innovative IP Voice amp Video Sample 2 context start time answer time endtime call time talk time source channel dest channel status 619 from internal 1 30 2014 14 31 1 30 2014 14 32 1 30 2014 14 32 9 3 SIP 609 00000150 SIP 619 00000151 ANSWERED Figure 104 Downloaded CDR File Sample Source Channel and Dest Channel 2 SIP
55. 6 Ethernet Capture The output result is in ocap 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 IP 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 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 228 of 233 TRACEROUTE G Target Host Output Result b4 bytes from 4 125 224 7 9 64 bytes from 74 125 224 179 64 bytes from 74 125 224 179 64 bytes from 74 125 224 179 64 bytes from 74 125 224 179 64 bytes from 74 125 224 179 64 bytes from 74 125 224 179 64 bytes from 74 125 224 179 64 bytes from 74 125 224 179 64 bytes from 74 125 224 179 64 bytes from 74 125 224 179 64 bytes from 74 125 224 179 64 bytes from 74 125 224 179 64 bytes from 74 125 224 179 www google com Ser DEA tme 13 600 ms seg 2 ttl 53 time 19 300 ms seg 3 ttl 53 time 13 500 ms seg 4 ttl 53 time 13 625 ms seg 5 ttl 53 time 13 950 ms seg 6 ttl 53 time 14 125 ms seg ttl 53 time 17 425 ms seg s ttl 53 time 13 500 ms seg 9 ttl 53 time 13 975 ms seg 10 ttl 53 time 14 100 ms seg 11 ttl 53 time 14 175 ms seg 12 ttl 53 time 14 025 ms seg 13 ttl 53 time 14 150 ms seg 14 ttl 53 time 13 900 ms www google com ping statistics 15 packets transmitted 15 packets received 0 pac
56. 89 For example if the LCD shows 192 168 40 167 please enter 192 168 40 167 in your web browser and the web page will be redirected to https 192 168 40 167 8089 The option Redirect From Port 80 can be configured under the UCM6510 web GUl gt Settings gt HTTP Server 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 System Events and CDR e PBX To configure extensions trunks call routes zero config for auto provisioning call features internal options IAX settings SIP settings as well as ports configuration for digital trunks e Settings To configure network settings firewall settings change password LDAP Server HTTP Server Email Settings Time Settings and NTP server e Maintenance To perform firmware upgrade backup configurations cleaner setup reset reboot syslog setup and troubleshooting WEB GUI LANGUAGES Currently the UCM6510 web GUI supports the following languages English Simplified Chinese Traditional Chinese Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 27 of 233 andstream Spanish French Portuguese Russian Italian Polish German Czech Users can select the displayed language in web GUI login page or at the upper right of the web GUI after logging in Username English Password English BPH English IEA English Sc Espafio
57. B SD T1 E1 J1 FXS 1 2 FXO 1 2 LAN WAN Cluster Heartbeat LCD Display 128x32 dot matrix graphic LCD with DOWN and OK buttons Reset Switch Yes long press for factory reset and short press for reboot LED Indicators Voice Video Capabilities LEC with NLP Packetized Voice Protocol Unit 128ms tail length carrier grade Line Echo Cancellation Dynamic Jitter Buffer Modem detection and auto switch to G 711 G 711 A law U law G 722 G 723 1 5 3K 6 3K G 726 G729A B iLBC GSM AAL2 G 726 32 ADPCM T 38 Video Codecs H 264 H 263 H 263 QoS Layer 3 QoS Layer 2 QoS Voice over Packet Capabilities Voice and Fax Codecs Signaling and Control DTMF Methods In Audio RFC2833 and SIP INFO Digital Signaling PRI SS7 MFC R2 TFTP HTTP HTTPS auto discovery amp auto provisioning of Grandstream IP endpoints via ZeroConfig DHCP Option 66 multicast SIP SUBSCRIBE mDNS eventlist between local and remote trunks TCP UDP IP RTP RTCP ICMP ARP DNS DDNS DHCP NTP TFTP Network Protocols SSH HTTP HTTPS PPPoE SIP RFC3261 STUN SRTP TLS LDAP HDLC HDLC ETH PPP Frame Relay pending Call Progress Tone Polarity Reversal Hook Flash Timing Loop Current Provisioning Protocol and Plug and Play Disconnect Methods Disconnect Busy Tone Security Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 16 of 233 andstream Innovative IP Voice amp Video Media SRTP TLS HTTPS SSH Fail2ban alert events
58. Configure the name of the IVR Letters digits and are allowed Enter the extension number for users to access the IVR If enabled all callers to the IVR can dial other extensions 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 if Dial Trunk is enabled The available permissions are Internal Local National and International from the lowest level to the highest level The default setting is Internal If the user tries to dial outbound calls after dialing into the IVR the UCM6510 will compared the IVR s permission level with the outbound route s privilege level If the IVR s permission level is higher than or equal to the outbound route s privilege level the call will be allowed to go through Select an audio file to play as the welcome prompt for the IVR Click on Prompt to add additional audio file under web GUI gt Internal Options gt IVR Prompt Configure the timeout between digit entries After the user enters a digit the user needs to enter the next digit within the timeout If no digit is detected within the timeout the UCM6510 will consider the entries complete The default timeout is 3 seconds After playing the prompts in the IVR the UCM6510 will wait for the DTMF UCM6510 IP PBX User Manual Page 131 of 233 Response Timeout
59. 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 A random secure password will be automatically generated It is SIP IAX Password recommended to use this password for security purpose 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 Voicemail Password A random password in digits will be automatically generated It is recommended to use this password for security purpose 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 forwarded to voicemail voicemail is enabled or hang up voicemail is disabled If not specified the default ring timeout is 60 seconds on the UCM6510 which can be configured in the global ring timeout setting l under web GUI gt Internal Options General Preference The valid range Ring Timeout l is between 5 seconds and 600 seconds Note If the end point also has a ring timeout configured the actual ring timeout used is the shortest time set by either device Enable automatic recording for the calls using this extension The default Rete 4 setting is disabled The recording files will be saved in external storage if uto Recor plugged in and can be acce
60. For example directly dial 5000 will have to call go into the extension 5000 s voicemail If the user would like to transfer the call to the extension 5000 s voicemail enter 5000 as the transfer target number CALL RECORDING The UCM6510 allows users to record audio during the call If Auto Record is turned on for extension or trunk the call will be automatically recorded when there is established call with the extension or trunk Otherwise please follow the instructions below to manually record the call 1 Make sure the feature code for Audio Mix Record is configured and enabled 2 After establishing the call enter the Audio Mix Record feature code by default 3 followed by or SEND to start recording 3 To stop the recording enter the Audio Mix Record feature code by default 3 followed by or SEND again Or the recording will be stopped once the call hangs up 4 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 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 176 of 233 andstream Innovative IP Voice amp Video Click on the title of the column to sort by column Click on the row to display full record View 10 EN Start Time 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 82 Download Re
61. For example patter 2XXX 1234 means the only extension with the caller ID 1234 is allowed to use this rule Select privilege level for the inbound rule when a VoIP trunk is selected in Trunks field Internal The lowest level required All users can use this rule Local Users with Local National or International level are allowed to use this rule National Users with National or International level are allowed to use this rule International The highest level required Only users with international level can use this rule This setting is used to compare with the outbound trunk s permission level when the inbound call dials out via a trunk on the UCM6510 Therefore it s usually used only when the Default Destination is set to By DID Select the default destination for the inbound call Extension Voicemail Conference Room Call Queue Ring Group Paging Intercom Voicemail Group Fax DISA UCM6510 IP PBX User Manual Page 120 of 233 Strip Dial Trunk DID Destination Andstream Innovative IP Voice amp Video IVR By DID When By DID is used the UCM6510 will look for the destination based on the number dialed which could be local extensions conference call queue ring group paging intercom group IVR voicemail groups and Fax extension as configured in DID destination If the dialed number matches the DID pattern the call will be allowed to go through Dial By Name Specify the number of digi
62. INTERCOM GROUP 147 JE e e IEN 149 CONFIGURE Evne 149 FETE NR NNN apsee eee ree tee enone eer eer eee ne ee nee ee etre errr er 153 CONFIGURE EXTENSION GROUPS kissssseseiiireesesetttirrsssesetttitrrsssbtttrrrrrsseabtttrrrrnssaaetrrrrrnsa nant 153 USE EXTENSION GROUPS 153 Pe GROUP G LNG 155 CONFIGURE PICKUP GROUPS sssssseiiiiissssetttttrisettkttt trises bbt ttr 1 rras sabt t errr n asebete rnrn nsa ae nern r raaa nant 155 MUSIC ON FIO UD ON 157 RN E 159 CNN 159 SAMPLE CONFIGURATION TO RECEIVE FAX FROM PSTN UNE 160 SAMPLE CONFIGURATION FOR FAX TO EMAIL rrrsvsrrnssrvrsserssserseserneseressersssersesernestensstersssennesnn 162 DEAL 163 BLE AND PENT LIST en 165 FE 165 EE ncininicdyeetwnetn hiner E EnA SEEE Errr rnan 165 EISE ECK NANE NN 169 DIAL BY NAME CONEIGURATION 169 CALL FEATURE ee 173 FEATURE CODES 173 MENE 176 SEG 177 iE O E 177 RETRIEVE THE PARKED CAL 177 INTERNAL OPNE eee eee eee eee eee eee 179 INTERNAL OPTIONSIGENERAL iissssseiiiiisessettttttteskkkttt trr s sesk b ktt n nnns Seabee rnnr nsaan errr r rasanne errr rnane 179 INTERNAL OPTIONS JITTER BUFFER sssessssessssesssseesssseesssessssesssseeesstessssessssessssenesseeessneersneesaee 181 INTERNAL OPTIONS RTP SETTINGS scccsssessssessssesssseessssessssessssesssseesssteesssesssiestsseetsseessseeessneetaee 181 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 4 of 233 E ten Innovative IP Voice amp Video INTERNAL OPTIONS STUN MONITOR
63. 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 Inbound routes can be configured via web GUI gt PBX gt Basic Call Routes gt Inbound Routes e Click on Create New Inbound Rule button to add a new inbound route e Click on Blacklist button to configure blacklist for all inbound routes e Click on to edit the inbound route e Click on M to delete the inbound route INBOUND RULE CONFIGURATIONS Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 119 of 233 Trunks DID Pattern Privilege Level Default Destination Firmware Version 1 0 0 25 ndstream Innovative IP Voice amp Video Table 36 Inbound Rule Configuration Parameters Select the trunk to configure the inbound rule All patterns are prefixed with the _ Special characters X Any Digit from 0 9 Z Any Digit from 1 9 N Any Digit from 2 9 Wildcard Match one or more characters I Wildcard Match zero or more characters immediately Example 12345 9 Any digit from 1 to 9 The pattern can be composed of two parts divided by a f character The first part is used to specify the dialed number the second part is used to specify the caller ID and it is optional if set it means only the extension with the specific caller ID is allowed to call in or call out
64. Manual Page 83 of 233 Andstream Innovative IP Voice amp Video This page intentionally left blank Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 84 of 233 andstream Innovative IP Voice amp Video ANALOG TRUNKS To set up analog trunk on the UCM6510 e Go to web GUI gt PBX gt Basic Call Routes gt Analog Trunks to add and edit analog trunks e Go to web GUI gt PBX gt Ports Config gt Analog Hardware to configure analog hardware settings ANALOG TRUNKS CONFIGURATION 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 Click on to edit the analog trunk e Click on DI to delete the analog trunk The analog trunk options are listed in the table below Table 23 Analog Trunk Configuration Parameters Channels Select the channel for the analog trunk Specify a unique label to identify the trunk when listed in outbound routes Trunk Name l inbound routes and etc Advanced Options If enabled a polarity reversal will be marked as received when an Outgoing call is answered by the remote party For some countries a Enable Polarity Reversal 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 Se RTA AIS this
65. N Shinninn Figure 60 Voicemail Group Table 43 Voicemail Group Settings l Enter the Voicemail Group Extension The voicemail messages left to this Voicemail Group Extension l l l l extension will be forwarded to all the voicemail group members Configure the Name to identify the voicemail group Letters digits _ and are allowed Name Select available mailboxes from the left list and add them to the right list Voicemail Group Mailboxes The extensions need to have voicemail enabled to be listed in available mailboxes list Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 143 of 233 Andstream Innovative IP Voice amp Video This page intentionally left blank Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 144 of 233 E ten Innovative IP Voice amp Video RING GROUP The UCM6510 supports ring group feature with different ring strategies applied to the ring group members This section describes the ring group configuration on the UCM6510 CONFIGURE RING GROUP Ring group settings can be accessed via web GUI gt PBX gt Call Features gt Ring Group Create New Ring Group Extension Ring Group Name Members 6400 techsupport 6005 6006 6007 Figure 61 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 e Click on to delete the ring group Table 44 Ring Group Param
66. P server as configured in Figure 18 LDAP Server Configurations Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 51 of 233 Server Address Port Base DN User Name Password LDAP Name Affributes 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 e Aandstrean 192 168 40 134 389 dc pbx dc com CallerDName AccountNumber CallerIDName AccountNumber YAccountNumber CallerDName 50 0 O0 Enable CO Enable Cancel Figure 23 GXP2200 LDAP Phonebook Configuration HTTP SERVER The UCM6510 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 IE Mozilla Firefox and Google Chrome By default the PBX can be accessed directly by typing IP address in the PC s web browser e g 192 168 40 50 It will then be automatically redirected to HTTPS using Port 8089 eg 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 UCM6510 IP PBX User Manual Firmware Version 1 0 0 25 Page 52 of 233 E ten Innovative IP Voice amp Video Table 14 HTTP Server Settings Enable or disable redirect from port 80 On the PBX the def
67. Strings without a date have a default value of 2000 01 01 Strings without a time of day have a default value of of 00 00 UTC while strings with a time of day specified may also optionally specify a time zone offset replace A in time zone offset with 2B see Page 214 of 233 ndstream Innovative IP Voice amp Video HH MM http www w3 org TR NOTE datetime HH MM SS HH MM SS SSS NOW DDDDDDDDDD minDur Filters based on the billsec value the duration between Number duration in seconds call answer and call end maxDur Example Queries The following illustrates the format of queries to accomplish certain requests In most cases multiple different queries will accomplish the same goal and these examples are not intended to be exhaustive but rather to bring attention to particular features of the CDR API connector Query 1 Request all records of calls placed on extension 5300 which last between 8 and 60 seconds inclusive with results in CSV format https 192 168 254 200 8088 cdrapi format CSV amp caller 5300 amp minDur 8 amp maxDur 60 OR https 192 168 254 200 8088 cdrapi caller 5300 amp minDur 8 amp maxDur 60 Query 2 Request all records of calls placed on extension 5300 or in the range 6300 6399 to extensions Starting with 5 with results in XML format https 192 168 254 200 8088 cdrapi format XML amp caller 5300 6300 6399 amp callee 5 OR https 192 168 254 200 8088 cdrapi cdrapi format XML a
68. TORS winisecidcievcciacccncdedsencdatececcdededsinceacavercatednexedicdccrcnbadedexsadinenendhdaexctdiaeeedetedaaades 25 USE THE WEB GU 26 ACCESS WEB GU 26 WEB GUI CONFIGURATIHONG ccccccececececsccecscecececscsensceneuenscusnananeneuscscnansnenenscsnauauenenscsenansnenens 27 WEB GUI LANGUAGES Lo ccccceccccccccecececscecececcececsusnsceueaueusnsnsusueatuausnsusueaentausnsususaenususnsusnssetenensnens 27 SAVE AND APPLY CHANGES 1 cccccccccececececececcccceusnsueneatuteusususueaecueusnsustenenunusnsusnetnuensnsnsuseenenes 28 MAKE YOUR FIRST CAL 29 SYSTEM SETTINGS uannnannununununananananannunununanananananannnnnnunnnnnanananavnvnnnnnnnnn 31 NETWORK GEIIINGS A LAADA LARLA A LEARDE LEARD A APARLE APAA Anan nanana 31 BA REENEN 31 Lg ler EE 36 STATIC ROUTE 38 PORT EOHIWOHDINGG rannan Pannaan ran ranra nnana aonana nnna 40 ERE NN 42 SIL NI OLD m 42 DYNAMIC DEFENSE wrecaxicsesctesanccdbadevcusoddenexdbasecesadediuesdnidcusiansdeuedsniadssediciacdadbeiiaseeupiaenedneicdesebeieres 44 FAIL BAN EE 46 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 1 of 233 E ten Innovative IP Voice amp Video NER NN DN ee 46 PENNEN hhv 47 LDAP SERVER GONFIGURA ENEE ege 47 KIN FRON GOOR STEEN EEE EE 49 LDAP CLIENT EE TR ET 50 ATIE ERVE E 52 EMAIL SETTINGS Ladens 53 TME SETEN G ceee E E E E 55 NENNE 55 SETENE 57 NIP ERVE MA scacasnsesctecinretesaansadetasntasatacseadttonpceactasnuatstanssasatesstqbetosntebatasaieiateastaietache
69. X gt Internal Options gt General will be used Configure the name of the caller to be displayed when the extension has no CallerID Name configured Select audio and video codec for the VoIP trunk The available codecs are PCMU PCMA GSM AAL2 G 726 32 G 726 G 722 G 729 G 723 ILBC ADPCM H 264 H 263 and H 263p In the selected codec list UCM6510 IP PBX User Manual Page 108 of 233 Auto Record DID Mode Enable Qualify Qualify Timeout Qualify Frequency Fax Detection SRTP Sync LDAP Enable Sync LDAP Password Sync LDAP Port LDAP Outbound Rule LDAP Dialed Prefix Firmware Version 1 0 0 25 san Innovative IP Voice amp Video users can click on UP or DOWN arrow to adjust the order for the codec priority using this trunk Enable automatic recording for the calls using this trunk The default setting is disabled The recording files are saved in external storage if plugged in and can be accessed under web GUI gt CDR gt Recording Files Configure where to get the destination ID of an incoming SIP call from SIP Request line or To header The default is set to Request line If enabled the UCM6510 will regularly send SIP OPTIONS to the device to check if the device is still online The default setting is No When Enable Qualify option is set to Yes configure the timeout in ms for the Qualify SIP message If no response is received within the timeout the device is considered offline The d
70. able 34 IAX Trunk Configuration Parameterg 111 Table 35 Outbound Route Configuration Parameters 001 nn00nnn00annnnannnonnnnnnnnenonnrrennnrnrnnnrennnrrrsnrrennnnne 117 Table 36 Inbound Rule Configuration Parameters ccccccccsseccccceesceeceeeceeseeeeceeseeceeseeeeesseeeeessaeeeeeas 120 Table 37 Conference Bridge Configuration Parameters cccccccsesccecsseceeceeseceeceeeceeseeeeeseeeeeeesaaeeeeeas 125 Table 38 Conference Caller IVR Menu secessione ienie EENE 129 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 7 of 233 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 Table 64 Table 65 Table 66 Table 67 Table 68 Table 69 Table 70 Table 71 Table 72 Table 73 Table 74 Table 75 Table 76 Table 77 Table 78 Firmware Version 1 0 0 25 E ten Innovative IP Voice amp Video IVR Configuration Parameters ccccccccccceceeeeeeeeaaeeeseeseeeeeeeeeeeeeeeeeeeaeaaaaaaaedeeeeeeeeeeeeeeeeeeeaaaaaas 131 Meles nate dl Le EEE ee eee ee eee ee 139 Voicemail IVR Men 140 Von Email SS WANG sanere dene 142 erte EU RE el ie Ett Te CN 143 Ang Group g Ge Er 145 Paging Intercom Group Configuration Parameters 000nnn0nnnnaannnnannnonnnnnnenrnonnrrnnnnrrrnnreenen
71. able 78 Cleaner Configuration Enable CDR Cleaner Enable the CDR Cleaner function CDR Clean Time Enter 0 23 to specify the hour of the day to clean up CDR Clean Interval Enter 1 30 to specify the day of the month to clean up CDR Enable VR Cleaner Enter the Voice Records Cleaner function VR Clean Threshold Specify the Voice Records threshold from 0 to 99 by using local storage status in percentage VR Clean Time Enter 0 23 to specify the hour of the day to clean up Voice Records Clean Interval Enter 1 30 to specify the day of the month to clean up Voice Records All the cleaner logs will be listed on the bottom of the page RESET AND REBOOT Users could perform reset and reboot under web GUI gt Maintenance gt Reset and Reboot To factory reset the device select the mode type first There are two different types for reset e User Data All the data including voicemail recordings IVR Prompt Music on Hold CDR and backup files will be cleared e All All the configurations and data will be reset to factory default Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 226 of 233 ndstream Innovative IP Voice amp Video Reset amp Reboot Factory Reset User Data Figure 115 Reset and Reboot SYSLOG On the UCM6510 users could dump the 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 i
72. ace 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 UCM6510 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 within LAN area This allows users to finish the installation with ease and start using the SIP devices in a managed way To provision a phone three steps are involved i e discovery assignment and provisioning The UCM6510 creates XML config file to the detected assigned Grandstream device and accomplishes the following configurations on the device after the provisioning e AUCM6510 extension will be assigned and registered on the phone e SIP related network settings such as NAT traversal and Use Random Port are configured on the phone e Call feature settings such as Public Mode Voicemail User ID Dial Plan and Auto Answer e LDAP client configurations will be set up automatically on the phone to use the default LDAP directory generated in the UCM6510 LDAP server This section explains how zero config works on the UCM6510 The settings for this feature can be accessed via web GUI gt PBX gt Basic Call Routes gt Zero Config AUTO PROVISIONING By default the Zero Config feature is enabled on the UCM6510 for auto provisioning Three methods of auto p
73. age 189 of 233 TLS Client Protocol TLS Do Not Verify TLS Self Signed CA TLS Cert TLS CA Cert TLS CA List SIP SETTINGS NAT External IP Address External Host External Refresh External TCP Port Firmware Version 1 0 0 25 san Innovative IP Voice amp Video Select the TLS protocol for outbound client connections The default setting is TLSv1 If enabled the TLS server s certificate won t be verified when acting as a client The default setting is Yes 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 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 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 the accessed servers Note The size of the uploaded CA certificate file must be under 2MB Display a list of files under the CA Cert directory Table 62 SIP Settings NAT Configure a static address and port optional that will be used in outbound SIP messages if the UCM6510 is behind NAT If it s a hostna
74. al ring timeout setting under web GUI gt Internal Options General Preference The valid range Ring Timeout l is between 5 seconds and 600 seconds Note If the end point also has a ring timeout configured the actual ring timeout used is the shortest time set by either device Enable automatic recording for the calls using this extension The default setting is disabled The recording files will be saved in external storage if plugged in and can be accessed under web GUI gt CDR gt Recording Files Auto Record l l i When user dials voicemail code the password verification IVR is skipped Skip Voicemail Password S l If enabled this would allow one button voicemail access By default this option is disabled Verification User Settings l Configure the first name of the user The first name can contain First Name characters letters digits and Configure the last name of the user The last name can contain Last Name characters letters digits and _ Fill in the Email address for the user Voicemail will be sent to this Email address Email Address Select the voice prompt language to be used for this extension The default setting is Default which is the selected voice prompt language under web GUI gt PBX gt Internal Options gt Language The dropdown list Language shows all the current available voice prompt languages on the UCM6510 To add more languages in the list please download voice prompt pac
75. ameterg 65 Table 19 IAX Extension Configuration Harameters 69 Table 20 FXS Extension Configuration Parameters ccccccssecccceseceeceeceeceueceeseeseeeeceeeceeseaseeessaeeessaaes 71 Table 21 Batch Add SIP Extension Parametere 75 Table 22 Batch Add IAX Extension Parametere 77 Table 23 Analog Trunk Configuration Parameters cccccccccccseecceceeeeeeeceeeceeceeceeseeeecessaeseesseeeeessaneeessaaes 85 Table 24 PSTN Detection for Analog Trunk cccccecccccceescecceeeceeceeeeceeceeeceeceueeeessesecesseueceessaeeeesaeeeessaaes 90 Table 25 Analog Hardware Configuration ParaMete rs cccccsesccccessccecceeeceeceeeceeseeeecesseeceesseeeessaeeeseaees 92 Table 26 Digital Hardware Configuration Parameters E1 PRI NET PRI CRE 94 Table 27 Digital Hardware Configuration Parameters E1 Di 96 Table 28 Digital Hardware Configuration Parameters E1 MECH 97 Table 29 Digital Hardware Configuration Parameters T1 J1 PRI NET PRI CORE 99 Table 30 Digital Hardware Configuration Parameters T1 J1 G 100 Table 31 Digital Trunk Configuration Parameters rrannrrnnnnnnronnnnnnorrnnnnronnnnnnennnnnnennnnnnrennnnnnennnnnnennnnnneene 102 Table 32 Data Trunk Configuration Parameters ccccccccccceceseeessseceeeeeeseeeeeeeceeeeesesuaeseeseeeeeesssaaseeeeeeees 106 Table 33 SIP Trunk Configuration Parameters ccccccccccsssecceeccesseceececeeeeceeeeeaseeceeesseeseeeeseeaaaeceeeteaenes 107 T
76. an UCM6510 extension already registered To successfully reboot the user Zero Config needs to be enabled on the UCM6510 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 checkbox for the extension s Then click on Modify Selected Extensions to edit the extensions in a batch e Delete selected extensions Select the checkbox for the extension s Then click on Delete Selected Extensions to delete the extension s Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 80 of 233 Andstream Innovative IP Voice amp Video EXPORT EXTENSIONS The extensions configured on the UCM6510 can be exported to csv format file with selected technology SIP IAX or FXS Click on Export Extensions button and select technology in the prompt Export Extensions Export Options Technology Figure 33 Export Extensions The exported csv file can also serve as a template for users to fill in desired extension information to be imported to the UCM6510 IMPORT EXTENSIONS The capability to import extensions to the UCM6510 provides users flexibility to batch add extensions with similar or different configurations quickly 1 Export extension csv file from the UCM6510 by clicking on Export Extension
77. anual Page 103 of 233 andstream Innovative IP Voice amp Video ge PRI Signaling Trace Start a Stor a Download Delete Output Result Capturing Done Click on Download to download the captured packets Troubleshooting Figure 45 Troubleshooting Digital Trunks After capturing the trace users can download it for basic analysis Or you can contact Grandstream Technical support in the following link for further assistance if the issue is not resolved http www grandstream com index php support Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 104 of 233 andstream Innovative IP Voice amp Video DATA TRUNK The UCM6510 E1 T1 J1 interface also supports data trunk function that allows users to access Internet Users can select HDLC HDLC ETH Cisco and PPP protocol for the data trunk To use data trunk 1 Go to web UI gt PBX gt Ports Config gt Digital Hardware page and click do to create a new group Designate a channel for data trunk usage in the group setting 2 Goto web Ul gt PBX gt Basic Call Routes gt Data Trunks page click on to edit the data trunk 3 Save the configuration and click on Apply Changes for the change to take effect 4 Once connected the data trunk will periodically ping and check the status with status indicator shown for the data trunk on the web page The status indicator shows if connected successfully Se i S 5 Ifthe user happens to lost
78. arding Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 41 of 233 E ten Innovative IP Voice amp Video FIREWALL The UCM6510 provides users firewall configurations to prevent certain malicious attack to the UCM6510 system Users could configure to allow restrict or reject specific traffic through the device for security and bandwidth purpose The UCM6510 also provides Fail2ban feature for authentication errors in SIP REGISTER INVITE and SUBSCRIBE To configure firewall settings in UCM6510 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 firewall settings The following table shows a sample current service status running on the UCM6510 Table 9 UCM6510 Firewall gt Static Defense gt Current Service 22 Ls LILI Asterisk tcp IPv4 389 Slapd tcp IPv4 22 Dropbear tcp IPv4 80 Lighthttpd tcp IPv4 8089 Lighthttpd tcp IPv4 69 Opentttpd udp IPv4 9090 Asterisk udp IPv4 6060 zero config udp IPv4 5060 Asterisk udp IPv4 4569 Asterisk udp IPv4 5353 zero config udp IPv4 37435 Syslogd udp IPv4 For typical firewall settings users could configure the following options on the UCM6510 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 42 of 233 dstream Table 10 Typical Firewall Settings If
79. ational or International Spare Usually National or International is used The line build out LBO is the distance between the operators and the ae PBX Please use the default value 0dB unless the distance is long RX Gain Configure the RX gain for the receiving channel of digital port The valid range is from 24dB to 12dB TX Gain Configure the TX Gain for the transmitting channel of digital port The Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 96 of 233 andstream Innovative IP Voice amp Video valid range is 24dB to 12dB Codec Select alaw or ulaw If set to default alaw will be used for E1 Advanced Settings WE Coding Select HDB3 or AMI CRC Select whether to use CRC4 or not Indicates the type of the called number The receiving switch may use this Called Nature of Address indicator during translations to apply the number s proper dial plan Users Indicator can select Unknown Subscriber National International or Dynamic Indicates the type of the calling number The receiving switch may use Calling Nature of Address this indicator during translations to apply the number s proper dial plan Indicator Users can select Unknown Subscriber National International or Dynamic International Prefix National Prefix l Configure the prefix in PRI Local Dial Plan and PRI Dial Plan for each type Local Prefix Private Prefix Unknown Prefix Table 28 Digital Hardware Co
80. ault 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 Enable Select HTTP or HTTPS as the protocol to access the HTTP server The default setting is HTTPS This also defines whether to use HTTP or Protocol Type HTTPS to download the config file in zero config as the UCM6510 is served as HTTP HTTPS server that has the device config files for zero Redirect From Port 80 config Specify port number to access the HTTP server The default port number is 8089 Port 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 UCM6510 can be used to send out alert event Emails Fax Fax To Email Voicemail Voicemail To Email and etc The configuration parameters can be accessed via web GUI gt Settings gt Email Settings Table 15 Email Settings Enable or disable TLS during transferring submitting your Email to other TLS Enable Ee SMTP server The default setting is Yes e MITA Mail Transfer Agent The Email will be sent from the configured domain When MTA is selected there is no need to set up SMTP Pee 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
81. ault setting is Yes If set to Yes the UCM6510 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 UCM6510 can display correct time accordingly 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 MT Z 6MDT 5 M4 1 0 M11 1 0 MTZ 6MDT 5 This indicates a time zone 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 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 UCM6510
82. block collect calls If Double Answer users have problem blocking collect calls using Group B signals please try enabling this option Accept On Offer By default it s enabled In most of cases this option should be enabled If enabled the callee side will request the caller to send caller category Skip Category before sending caller number Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 98 of 233 Charge Calls andstream Innovative IP Voice amp Video Note Get ANI First and Skip Category cannot be enabled at the same time Whether or not report to the other end accept call with charge This setting has no effect with most telecos The default setting is enabled recommended Table 29 Digital Hardware Configuration Parameters T1 J1 PRI_LNET PRI_CPE Basic Settings Clock LBO RX Gain TX Gain Codec Play Local RBT All E1 T1 J1 spans generate a clock signal on their transmit side The parameter determines whether the clock signal from the far end of the E1 T1 J1 is used as the master source of clock timing If the far end is used as the master the PBX system clock will synchronize to tt e Master The port will never be used as a source of timing This is appropriate when you know the far end should always be a slave to you e Slave The equipment at the far end of the E1 T1 J1 link is the preferred source of the master clock The line build out LBO is the distance between the operato
83. bnet Only Only the user in specific subnet can register this extension Up to three subnet addresses can be specified e A 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 outbound rule to make outbound calls The default setting is No Select audio and video codec for the extension The available codecs are PCMU PCMA GSM AAL2 G 726 32 G 722 G 729 G 723 ILBC ADPCM LPC10 H 264 H 263 and H 263p In the selected codec list users can click on UP or DOWN arrow to adjust the order for the codec priority UCM6510 IP PBX User Manual Page 79 of 233 san Innovative IP Voice amp Video EDIT EXTENSION All the UCM6510 extensions are listed under web GUI gt PBX gt Basic Call Routes gt Extensions with status Extension CallerID Name Technology SIP IAX and FXS IP and Port Each extension has a checkbox for users to Modify Selected Extensions or Delete Selected Extensions Also options Edit Reboot and Delete are available per extension e Status Users can see the following icon for each extension to indicate the SIP status G Green Free G Blue Ringing Gi Yellow In Use Grey Unavailable e Edit single extension Click on to start editing the extension parameters e Reboot the user Click on to send NOTIFY reboot event to the device which has
84. 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 Display trunk name Display trunk Type e Analog e E1 T1 e SIP e AX Display username for this trunk 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 via web GUI gt PBX gt Basic Call Routes gt Analog Trunks e Click on O to refresh the trunk status e Click on to expand the status detail table e Click on to hide the status detail table EXTENSIONS Users could see all the configured extension status in this section Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 194 of 233 ndstream Innovative IP Voice amp Video Extensions gt AN Analog 1AX SIP Status Extension a 1000 GAP2140 1001 HT701 1005 1002 1003 1004 Total 6 Show 1 1 coto ey Figure 85 Extension Status Table 65 Extension Status Display extension number including feature code The color indicator has the following definitions e G Green Free d Blue Ringing e Yellow In Use Grey Unavailable Status Extension Display the extension number Name Label Display name callerID name or label
85. cation for the invited users The default web 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 UCM6510 conference bridge there are two ways to invite other parties to join the conference e Invite from web GUI For each conference bridge in UCM6510 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 Extension Figure 51 Conference Invitation From web GUI e Invite by dialing 0 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 0 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 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 127 of 233 ndstream 0 If O is entered to invite other pa
86. ccccccsseseceecceeeeeeeeeeeeseeeeeeaeeeceeessaaeeeeeessaaeeeeeesaaeeeeessaaess 220 Data Syne Contig rall N ss scesccchecessccenacnevonecensexect vos lt tocatacdyscavesneeaenedecneuwedelenatacesnesvecuiegenssecaesmetene 224 Cleaner Configuration BE 226 UCM6510 IP PBX User Manual Page 8 of 233 E ten Table of Figures UCM6510 IP PBX User Manual Figure 1 UCMGS1T0 FOR TT 19 Figure 2 UNS TN BACK E 19 Figure 3 UCM6510 T1 E1 J1 Crossover Cable Pm out 20 Figure 4 UCM6510 web GUI Login Page 26 Figure 5 UCM6510 web GUI Language 28 Figure 6 UCM6510 web GUI Apply Changes ccccccccceeesseeeeeeeeeeeeeeeeeeeeeeeeseeaeseeeeeeeesssaeseeeeeeeessaaaaeses 29 Figure 7 UCM6510 Network Interface Method Houte 34 Figure 8 UCM6510 Network Interface Method Switch nnnannnnnneannnnnnennnnnnnennnnnnnnnerensnnrrrnnennnresnnnrreennnne 35 Figure 9 UCM6510 Network Interface Method Dua 36 Figure 10 UCM6510 Using 802 1X as Cent 36 Figure 11 UCM6510 Using 802 1X EA MIR 37 Figure 12 UCM6510 Static Route Sample 2 0 0 ceeccccccccccceseseeeeeeeeeeeaeeeeeceeeeeseeseeeeseeeeeessseeaeeeeeeeessaaeasees 39 Figure 13 UCM6510 Static Route Contfguratton 39 Figure 14 UCM6510 Port Forwarding Configuration ccccccccccccecccseeseseceeeeseeeaeeeeeeeeeessaeeeeeseeeeeessaeaaeees 41 Figure 15 GXP2160 Web Access Using UCM6510 Port Forwarding nnnnnnnn0nannnnnnnnnennnnnnnenssnnnnenneennne 41 Figure 16 Create New Firewall Rule
87. ck on t to start uploading The music file uploaded has to be 16 bit 8 KHz Mono in wav format with size smaller than 5M e e Clickon next to the sound file to delete it from the selected Music On Hold Class Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 157 of 233 ndstream Innovative IP Voice amp Video A Note In case the users have deleted the system MOH files there are two ways to recover 1 Users could download the MOH file from this link htto downloads asterisk org pub teleohony sounds releases asterisk moh opsound wav 2 03 tar qz After downloading unzip the pack and upload the music files to the UCM6510 2 Factory reset could also recover the MOH file on the UCM Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 158 of 233 E ten Innovative IP Voice amp Video FAX T 38 The UCM6510 supports T 30 T 38 Fax and Fax Pass through It can also convert the received Fax to PDF format and send it to the configured Email address Fax T 38 settings can be accessed via web GUI gt PBX gt Internal Options gt FAX T 38 CONFIGURE FAX T 38 e Click on Create New Fax Extension In the popped up window fill the extension name and Email address to send the received Fax to e Click on Fax Settings to configure the Fax parameters Enable Error Correction Mode Maximum Transfer Rate Minimum Transfer Rate Default Email Address Template Variables Firmware Version 1 0 0 25 Tab
88. conds if a host exceeds the max times of retry as Max Retry Duration l l l SE defined in MaxRetry the host will be banned The default setting is 5 Configure the number of authentication failures during Max Retry Duration MaxRetry LE before the host is banned The default setting is 10 Configure IP address CIDR mask or DNS host in the whiltelist Fail2Ban will not Fail2Ban Whitelist ban the host with matching address in this list Up to 5 addresses can be added into the list Local Settings Enable Asterisk service for Fail2Ban The default setting is disabled Please make Asterisk Service sure both Enable Fail2Ban and Asterisk Service are turned on in order to use Fail2Ban for SIP authentication on the UCM6510 Configure the listening port number for the service Currently only 5060 for UDP Port is Supported Configure the number of authentication failures during Max Retry Duration Verse before the host is banned The default setting is 10 Please make sure this option is properly configured as it will override the MaxRetry value under Global Settings CHANGE PASSWORD After logging in the web GUI for the first time it is highly recommended for users to change the default password admin to a more complicated password for security purpose Follow the steps below to change the web GUI access password Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 46 of 233 E ten 1 Goto web GUI gt Settings g
89. cording File from CDR Page The above recorded call s recording files are also listed under the UCM6510 web GUI gt CDR gt Recording Files CALL PARK The UCM6510 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 call is not retrieved after the timeout the original extension who parks the call will be called back Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 177 of 233 Andstream Innovative IP Voice amp Video This page intentionally left blank Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 178 of 233 andstream 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 UCM6510 including general configura
90. d or time of day in any of the following formats YYYY MM DDTHH MM YYYY MM DDTHH MM SS YYYY MM DDTHH MM SS SSS literal T character separator in above three formats UCM6510 IP PBX User Manual Details Define the format for output of matching CDR rows Default is csv comma separated values Number of records to return Default is 1000 which is also the maximum allowed value Number of matching records to skip This will be combined with numRecords to receive all matches over multiple responses Default is 0 Filters based on src caller or dst callee value matching any extension contained in the parameter input string Patterns containing one or more wildcards en or _ will match as a regular expression and treat as a literal hyphen rather than a range signifier The Q wildcard matches any number of characters including zero while matches any single character Otherwise patterns containing a single hyphen will be matching a range of numerical extensions with non numerical characters ignored while patterns containing multiple hyphens will be ignored The pattern 0 0 will match all non numerical and empty strings Filters based on the start call start time value Calls which start within this period inclusive of boundaries will match regardless of the call answer or end time An empty value for either field will be interpreted as range with no minimum or maximum respectively
91. d to the UCM6510 e f users have other backup files on PC to restore on the UCM6510 click on Upload Backup File first and select it from local PC to upload on the UCM6510 Once the uploading is done this backup file Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 224 of 233 ndstream Innovative IP Voice amp Video will be displayed in the list of previous configuration backups for restore purpose Click on po to restore from the backup file Create New Backup Upload Backup File Config File Voice File Voicemail File Voice Records Oooo D 8 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 113 Restore UCM6510 from Backup File A Note e The uploaded backup file must be a tar file with no special characters like 4 amp space in the file name e The 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 0 25 UCM6510 IP PBX User Manual Page 225 of 233 Gran Innovative IP Voice amp Video CDR Cleaner Gi Enable CDR Cleaner CO CDR Clean Time GJ Clean Interval Voice Records Cleaner GJ Enable VR Cleaner CO VR Clean Threshold CO VR Clean Time CO VR Clean Interval Figure 114 Cleaner T
92. de The default setting is 6000 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 Note This option requires system reboot to take effect Configure whether normal ringing voltage 40V or maximum ringing voltage 89V for analog phones attached to the FXS port is required The default setting is Normal Configure to increase the ringing speed to 25HZ This option can be used with Low Power option The default setting is Normal Configure the peak voltage 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 default setting is Standard Configure the type of Message Waiting Indicator on FXS lines The default setting is FSK e FSK Frequency Shift Key Indicator e NEON Light Neon Bulb Indicator UCM6510 IP PBX User Manual Page 92 of 233 san Innovative IP Voice amp Video DIGITAL TRUNKS The UCM6510 supports E1 T1 J1 which are physical connection technology used in digital network T1 is the North American standard J1 is used in Japan whereas E1 is the European standard UCM6510 supports three signaling protocols PRI MFC R2 and SS7 PRI provides a va
93. detection process will keep the call up for about 1 minute e 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 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 90 of 233 Andstream Innovative IP Voice amp Video ANALOG HARDWARE CONFIGURATION The analog hardware FXS port and FXO port on the UCM6510 can be configured under web GUI gt PBX gt Ports Config gt Analog Hardware 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 Port1 Loop Star Port2 Kewl Start Figure 42 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 UCM6510 to automatically detect the ACIM value The detecting value will be automatically filled into the settings ACIM Setting G ACIM Detection Figure 43 FXO Ports ACIM Settings A Note ACIM setting is very important for the FXO PSTN line to work properly on the UCM6510 If the users experience ech
94. ditional Deactivate Firmware Version 1 0 0 25 Default code 77 Default code 78 Default Code 90 Enter the code and follow the voice prompt Or enter the code followed by the extension to forward the call Default Code 91 Default Code 92 Enter the code and follow the voice prompt Or enter the code followed by the extension to forward the call Default Code 93 Default Code 72 Enter the code and follow the voice prompt Or enter the code followed by the extension to forward the call Default Code 73 UCM6510 IP PBX User Manual Page 174 of 233 andstream Innovative IP Voice amp Video Feature Code Digits Timeout Call Park Parked Lots Parking Timeout s Default Setting 1000 Configure the maximum interval in milliseconds between the digits input to activate the feature code Default Extension 700 During an active call initiate blind transfer and then enter this code to park the call Default Extension 701 720 These are the extensions where the calls will be parked i e parking lots that the parked calls can be retrieved Default setting 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 Intercom Prefix Blacklist Add Blacklist Remove Firmware Ver
95. dstream Innovative IP Voice amp Video This page intentionally left blank Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 186 of 233 Andstream Innovative IP Voice amp Video SIP SETTINGS The UCM6510 SIP global settings can be accessed via web GUI gt PBX gt SIP Settings SIP SETTINGS GENERAL Table 58 SIP Settings General Configure the host name or domain name for the UCM6510 Realms MUST be globally unique according to RFC3261 The default setting is grandstream Bind UDP Port 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 means binding to all addresses If enabled the UCM6510 allows unauthorized INVITE coming into the PBX and the call can be made The default setting is No Realm For Digest Authentication Bind IP Address Allow Guest Calls Warning Please be aware of the potential security risk when enabling Allow Guest Calls as this will allow any user with the UCM6510 address to dial into the UCM6510 Select to enable overlap dialing support Overlap dial provides for a longer time out period between digits also called the inter digit timer If ee Diane set to Yes the PBX expects to receive the digits one right after the other coming in to this line with very little delay between digits If set to No the PBX waits up to about 2 seconds between digits The default setting is No If set to No all
96. e 147 Call Queue Configuration Parameters ccccccceceeeeeeeeeeeeeaaeaeeseeeeeeeeeeeeeeeeeeeeeeeeaaaaaaaseseeeeees 149 FPA dl Le EEE NE VEE NN 159 DES NE rbk 163 Event dee E 166 UCM6S10 Te 173 Internal eidele 179 Internal Options Jitter Butter 181 Internal Options RTP Settings ccccccccceeeeeeeeeeeeeeeeeseeeeeeeeeeeeeeeeeeeeessaaeeaeesdeeeeeeeeeeeeeeseessaaaaaas 181 Internal Options STUN Monitor rronnnrrnrrnnnrrnnnnnnnnvnnnrrrnnnnnnnssnnnnrnnnnnnnnssnnnnnnnnnnnnsssnnnnnnnnnsnnsesennn 182 IAX Settings GEN EE 183 PX SEG RS AN OMY een 183 IAX Settings Static Defense rrrnrnnnnnnnrnvvvvnnrnnnnnrrrrrrnrnrnnnnnrrsrennnnnnnnnsnrsrennnnnnnnnersssennrnnnnnssesssennn 184 PLEN ee 187 elle ON 188 SIP Selings SESSION IMET sereesssstiocttcncedecenieseasnaausvncdnsawvncihsndededesiiuensmnauiyaadaautnnthsndededeqibuensiaausebaxk 188 SIP Settings TOP and A UCNE 189 VPT 190 SG NOS e EE EE NN 191 TOR ENDRE EE EEE NE ENN 193 BUE 162 UD EEE Eed 195 PCIE E 196 Interface e e e te 197 Digital Channel Status Indicators 0 00 0 ceeeceeeceeeeecaeeeeeeceeeeeesaeeeeceeeeeesseeeaseeeeeeeesssaeseeeeeeeeeseaas 200 Parking Bee TE 201 System Status gt General ENEE 202 System Status gt Network EE 202 COR FIRE OE EEE EE 208 CDR Statistics Filter Crtera cc cccccccccccceeeeeeeeeeeceeeeeeeseeeeeeeceeeeeseeeeeeeeeeeesssaaaaeeeeeeeeessaaaaeses 212 COR API Conigu lon E 213 CDRAPLURI AMEN 214 Network Upgrade Configuration ccc
97. e 2 UCM6510 Back View Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 19 of 233 dstream Innovative IP Voice amp Video Follow the steps below to connect the UCM6510 for initial setup 1 Connect one end of an RJ 45 Ethernet cable cable type straight through into the WAN port of the UCM6510 connect the other end into the uplink port of an Ethernet switch hub 2 Connect the 12V DC power adapter into the DC 12V power jack 1 on the back of the UCM6510 Insert the main plug of the power adapter into a surge protected power outlet Connect the second power adapter into the DC 12V power jack 2 for failover purpose in case the first one is down 3 Wait for the UCM6510 to boot up The LCD in the front will show its hardware information when the bootup process Is done 4 Once the UCM6510 is successfully connected to the network the LED indicator for the WAN port in the front will be in solid green and the LCD shows up the IP address Depending on how the UCM6510 is used users can follow the steps below for optional setup 1 PSTN Line Connection connect PSTN lines from the wall jack to the UCM6510 LINE ports FXO ports Analog Line Connection connect analog lines phone and fax to the PHONE ports FXS ports 3 T1 E1 J1 Line Connection connect one end of the T1 E1 J1 cable provided from the service provider into the T1 E1 J1 port of the UCM6510 connect the other end into the T1 E1 J1 wall jack T1 E1 J1 crossover cabl
98. e 63 Paging Intercom Group Table 45 Paging Intercom Group Configuration Parameters Configure paging intercom group name Configure the paging intercom group extension Select 2 way Intercom or 1 way Page member list on the right e Click on i to edit the paging intercom group Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Select available users from the left side to the paging intercom group Page 147 of 233 Andstream Innovative IP Voice amp Video e Click on I to delete the paging intercom group e 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 amp Intercom Settings GJ AlertInfo Header Intercom Paging Intercom Feature Code Settings Please goto Feature Codes Configure the paging intercom feature code Figure 64 Page Intercom Group Settings e The UCM6510 has pre configured paging intercom feature code By default the Paging Prefix is 81 and the Intercom Prefix is 80 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 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 148 of 233 ndstream Innovative IP Voice amp Video CALL QUEUE The UCM6510 supports call queue by using static agents or dynamic agents This
99. e UCM6510 system will send the alert 4 System Reboot Alert Settings System Reboot LO Detect Cycle i minute s e Cancel Save Figure 95 System Events gt Alert Events Lists System Reboot Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 205 of 233 Andstream Innovative IP Voice amp Video e Detect Cycle The UCM6510 will check the system reboot based on this cycle Users can enter the number and then select second s minute s hour s day s to configure the cycle 5 System Update Alert Settings System Update G Detect Cycle minute s e Figure 96 System Events gt Alert Events Lists System Update e Detect Cycle The UCM6510 will check the system update based on this cycle Users can enter the number and then select second s minute s hour s day s to configure the cycle 6 System Crash Alert Settings System Crash G Detect Cycle minute s e Figure 97 System Events gt Alert Events Lists System Crash e Detect Cycle The UCM will detect the event at each cycle based on the specified time Users can enter the number and then select second s minute s hour s day s to configure the cycle Click on the switch OFF ONE to turn on off the alert and Email notification for the event Users could also select the checkbox for each event and then click on button Alert On Alert Off Email Notification On Email Notification Off to control the alert and Email notif
100. e allowed The default setting is Yes Allow External Domains SIP SETTINGS MISC Table 59 SIP Settings Misc Outbound SIP Registrations Register Timeout Configure the register retry timeout in seconds The default setting is 20 Configure the number of registration attempts before the UCM6510 gives Register Attempts up The default setting is 0 which means the UCM6510 will keep trying until the server side accepts the registration request Video Configure the maximum bit rate in kb s for video calls The default Max Bit Rate kb s setting is 384 Support SIP Video Select to enable video support in SIP calls The default setting is Yes If enabled the UCM6510 will generate manager events when SIP UA Generate Manager Events performs events e g Hold The default setting is No If enabled when rejecting an incoming INVITE or REGISTER request the UCM6510 will always reject with 401 Unauthorized instead of Reject Non Matching INVITE notifying the requester whether there is a matching user or peer for the request This reduces the ability of an attacker to scan for valid SIP usernames The default setting is No SIP SETTINGS SESSION TIMER Table 60 SIP Settings Session Timer Select the session timer mode The default setting is Accept The options are Session Timers Wi e Originate Always request and run session timer Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 188 of 233 Session Exp
101. e is disabled this option will not take effect Configure the password for the user A random secure password will be automatically generated when the extension is created It is SIP IAX Password recommended to use this password or other strong password for security purpose Enable voicemail for the user so that the call will be forwarded to the Enable Voicemail user s voicemail if there is no answer or the call is rejected The default setting is Yes Configure voicemail password digits only for the user to access the Voicemail Password voicemail box A random numeric password is automatically generated when the extension is created It is recommended to use the random Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 65 of 233 Call Forward Unconditional Call Forward No Answer Call Forward Busy Ring Timeout Auto Record Skip Voicemail Password Verification Support Hot Desking Mode san Innovative IP Voice amp Video generated password for security purpose Configure the Call Forward Unconditional target number so that the incoming call to this extension will be always forwarded to the target number If not configured the Call Forward Unconditional feature is deactivated The default setting is deactivated Configure the Call Forward No Answer target number so that the incoming call to this extension will be forwarded to the target number if the call is not answered until the ringing times out
102. e left side to the right side Create New Pickup Group TechSupport Available Extensions Pickup Group Members Figure 69 Edit Pickup Group e Click on I to delete the pickup group Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 155 of 233 Andstream Innovative IP Voice amp Video This page intentionally left blank Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 156 of 233 andstream 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 KI Music On Hold Classes defaut e iil Upload an amp KHz Mono Music file file size under 5M Choose file to upload as Upload List of Sound Files macroform cold_day wav macroform robot_dity wav macroform the_simplicity wav manolo camp morning coffee wav reno project system wav Figure 70 Music On Hold Default Class e Click on Create New MOH Class to add a new Music On Hold class e Clickon to configure the MOH class sort method to be Alpha or Random for the sound files le e Clickon U next to the selected Music On Hold class to delete this Music On Hold class e Click on 1 to select music file from local PC and cli
103. e port goes into BLUE alarm when it receives all unframed 1s on all timeslots from the remote switch This is a special signal to indicate that the remote switch is having problem with its upstream connection Cannot start up Other operations are also available in interface status section e Click on O to refresh the interface status e Click on to expand the interface details e Click on to hide the interface details DIGITAL CHANNELS STATUS This section displays the status of the digital trunks on the UCM6510 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 199 of 233 ndstream Innovative IP Voice amp Video Digital Channels Status 10 Ed 18 Figure 88 Digital Channels Status Table 68 Digital Channel Status Indicators Voice Channel Not available O 3 C Connected and in talking status or incorrect configuration Connected and in idle status Connected and in ringing status Data Channel Always shows blue square D Channel Always shows grey with channel number in blue Other operations are also available in interface status section e Click on G to refresh the Digital Channels status e Click on to expand the Digital Channels Status details e Click on to hide the Digital Channels Status details PARKING LOT Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 200 of 233 andstream The UCM6510 supports call
104. e should be used and it s not provided in the UCM6510 package Please see T1 E1 J1 crossover cable pin out in the figure below 2 i 1 EE CS Ce h A a a EE El b h u Figure 3 UCM6510 T1 E1 J1 Crossover Cable Pin out Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 20 of 233 E ten Innovative IP Voice amp Video SAFETY COMPLIANCES The UCM6510 series IP PBX complies with FCC CE and various safety standards The UCM6510 power adapter is compliant with the UL standard Use the universal power adapter provided with the UCM6510 package only The manufacturer s warranty does not cover damages to the device caused by unsupported power adapters WARRANTY If the UCM6510 series IP PBX 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 Networks contact our Technical Support Team for a RMA Return Materials Authorization number before the product is returned Grandstream Networks reserves the right to remedy warranty policy without prior notification A Warning Use the power adapter provided with the UCM6510 series IP PBX Do not use a different power adapter as this may damage the device This type of damage is not covered under warranty Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 21 of 233 Andstream Innovative IP Voice amp Video This page intentionally
105. each the company employee if he she doesn t want to receive them directly Edit Extension 1000 Setting General Extension CallerlD Number G Permission Internal CO SIPNAX Password P95Cih Create New User Modify Select v Enable Voicemail Voicemail Password 47545707 Call Forward Unconditional Call Forward No Answer Skip Voicemail Password Auto Record Verification Qi Call Forward Busy G Ring Timeout Mi Suppor Hot Desking Mode User Settings First Name LastName Doe Email Address Language Default w Figure 81 Configure Extension First Name And Last Name 4 Query Type Specify the query type This defines how the caller will need to enter to search the directory By First Name enter the first 3 digits of the first name to search the directory By Last Name enter the first 3 digits of the last name to search the directory By Full Name enter the first 3 digits of the first name or last name to search the directory 5 Select Type Specify the select type on the searching result The IVR will confirm the name number for the party the caller would like to reach before dialing out By Order After the caller enters the digits the IVR will announce the first matching party s name and number The caller can confirm and dial out if it s the destination party or press to listen to the next matching result if it s not the desired party to call By Menu After the caller enters
106. efault setting is 1000ms When Enable Qualify option is set to Yes configure the interval in seconds of the SIP OPTIONS message sent to the device to check if the device is still online The default setting is 60 seconds 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 VolP trunk The default setting is No If enabled the local UCM6510 will automatically provide and update the local LDAP contacts to the remote UCM6510 SIP peer trunk In order to ensure successful synchronization the remote UCM6510 peer also needs to enable this option on the SIP peer trunk The default setting is No This is the password used for LDAP contact file encryption and decryption during the LDAP sync process The password must be the same on both UCM6510 peers to ensure successful synchronization Configure the TCP port used LDAP sync feature between two peer UCM6510 It could be any open port available not used by other services on the UCM6510 Specify an outbound rule for LDAP sync feature The UCM6510 will automatically modify the remote contacts by adding prefix parsed from this rule Specify the prefix for LDAP sync feature The UCM6510 will automatically UCM6510 IP PBX User Manual Page 109 of 233 san Innovative IP Voice
107. el are allowed to use this rule e National Users with National or International level are allowed to use _ this rule Privilege Level l EE l e International The highest level required Only users with international level can use this rule The default setting is Disable Please be aware of the potential security risks when using Internal level which means all users can use this outbound rule to dial out from the trunk When enabled users could specify extensions allowed to use this outbound route Privilege Level is automatically disabled if using Enable Filter on Source Caller ID The following two methods can be used at the same time to define the extensions as the source caller ID 1 Select available extensions extension groups from the left to the right This allows users to specify arbitrary single extensions available in the PBX Custom Dynamic Route define the pattern for the source caller ID Enable Filter on Source Caller ID This allows users to define extension range instead of selecting them one by one e 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 one or more characters Wildcard Match zero or more characters immediately Example 12345 9 Any digit from 1 to 9 Send This Call Through Trunk Use Trunk Select the trunk for this outbound rule Allows the user to specify the numb
108. el ext gt 5300 lt channel ext gt lt dstchannel ext gt 5301 lt dstchannel ext gt lt service gt s lt service gt lt cdr gt lt cdr gt lt Acctld gt 63 lt Acctld gt lt accountcode gt lt accountcode gt lt src gt 5300 lt sre gt lt dst gt 5301 lt dst gt lt dcontext gt from internal lt dcontext gt lt clid gt amp quot pn01 amp quot amp lt 5300 amp gt lt clid gt lt channel gt SIP 5300 00000000 lt channel gt lt dstchannel gt SIP 5301 00000001 lt dstchannel gt lt lastapp gt Dial lt lastapp gt lt lastdata gt SIP 5301 60 lt lastdata gt lt start gt 2013 12 03 14 01 41 lt start gt lt answer gt 2013 12 03 14 01 43 lt answer gt lt end gt 2013 12 03 14 01 46 lt end gt lt duration gt 5 lt duration gt lt billsec gt 3 lt billsec gt lt disposition gt ANSWERED lt disposition gt lt a maflags gt DOCUMENTATION lt amaflags gt lt uniqueid gt 1386100901 0 lt uniqueid gt lt userfield gt EXT lt userfie ld gt lt channel ext gt 5300 lt channel ext gt lt dstchannel ext gt 5301 lt dstchannel ext gt lt service gt s lt service gt lt cdr gt lt cdr gt lt Acctld gt 64 lt Acctld gt lt accountcode gt lt accountcode gt lt src gt 5300 lt src gt lt dst gt 5301 lt dst gt lt dcontext gt from internal lt dcontext gt lt clid gt amp quot pn01 amp quot amp lt 5300 amp gt lt clid gt lt channel gt SIP 5300 00000002 lt channel gt lt dstchannel gt SIP 5301 00000003 lt dstchannel gt lt lastapp gt Dial lt lastapp gt lt
109. emote extension status from this event list Once successfully configured the event list page will show the status of total extension and subscribers for each event list Users can also select the event URI to check the monitored extension s status and the subscribers details A Note e To configure LDAP sync please go to UCM6510 web GUI gt PBX gt Basic Call Routes gt VolP Trunk You will see Sync LDAP Enable option Once enabled please configure password information for the remote peer UCM6510 to connect to the local UCM6510 Additional information such as port number LDAP outbound rule LDAP Dialed Prefix will also be required Both the local UCM6510 and remote UCM6510 need enable LDAP sync option with the same password for successful connection and synchronization e Currently LDAP sync feature only works between two UCM6510s e Theoretically Remote BLF monitoring will work when the remote PBX being monitored is non UCM6510 PBX However it might not work the other way around depending on whether the non UCM6510 PBX supports event list BLF or remote monitoring feature Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 167 of 233 Andstream Innovative IP Voice amp Video This page intentionally left blank Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 168 of 233 Andstream Innovative IP Voice amp Video DIAL BY NAME Dial By Name is a feature on the PBX that allows caller to search a person by
110. 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 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 interface Enable to prevent Ping of Death attack to the device The default setting is disabled To enable or disable it click on the check box for the LAN or WAN interface Ping Defense Enable SYN Flood Defense Enable Ping of Death Defense Enable Under Custom Firewall Settings users could create new rules to accept reject or drop certain traffic going through the UCM6510 To create new rule click on Create New Rule button and a new window will pop up for users to specify rule options The following figure shows a firewall rule example that will deny SSH access for the UCM6510 from WAN side Create new firewall rule Gi Rule Name RejectSSH Action REJECT Gi Type Gi Interface G Service Figure 16 Create New Firewall Rule Table 11 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 Type o e Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 43 of 233 E ten Innovative IP Voice amp Video If selected users wil
111. eo The valid range is 30dB to 6dB The default setting is 0 Configure the minimum period of time in milliseconds that the hook flash must remain unpressed for the PBX to consider the event as a valid flash event The valid range is 30ms to 1000ms The default setting is 200ms Configure the maximum period of time in milliseconds that the hook flash must remain unpressed for the PBX to consider the event as a valid flash event The minimum period of time is 256ms and it can t be modified The default setting is 1250ms 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 Yes Specify ON OFF or a value the power of 2 from 32 to 1024 as the number of taps of cancellation Note When configuring the number of taps the number 256 is not translated into 256ms of echo cancellation Instead 256 taps means 256 8 32 ms The default setting is ON which is 128 taps Configure to enable disable 3 way calling feature on the user The default setting is enabled For example when enabled if the FXS extension has established a call with User A gt Press Flash to open a new line gt FXS extension calls User B gt Press flash again it will establish 3 way conference call with U
112. er if the call is rejected or the extension is in talking busy status If not configured the Call Forward Busy feature is deactivated The default setting 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 If not specified the default ring timeout is 60 seconds on the UCM6510 IP PBX User Manual Page 72 of 233 Auto Record Skip Voicemail Password Verification Andstream Innovative IP Voice amp Video UCM6510 which can be configured in the global ring timeout setting under web GUI gt Internal Options General Preference The valid range is between 5 seconds and 600 seconds Note If the end point also has a ring timeout configured the actual ring timeout used is the shortest time set by either device Enable automatic recording for the calls using this extension The default setting is disabled The recording files will be saved in external storage if plugged in and can be accessed under web GUI gt CDR gt Recording Files When user dials voicemail code the password verification IVR is skipped If enabled this would allow one button voicemail access By default this option is disabled User Settings First Name Last Name Email Address Language Configure the first name of the user The first name can contain characters letters digits and Configure the last name of the user The last name
113. er of digits that will be stripped from the beginning of the dialed string before the call is placed via the selected trunk Strip 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 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 118 of 233 Prepend Andstream 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 Use Failover Trunk Failover Trunk Strip Prepend INBOUND ROUTES 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 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
114. er web GUI gt Maintenance gt Backup gt Data Sync Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 223 of 233 Enable Backup Account Password Server Address Backup Time Before saving Andstream Innovative IP Voice amp Video Manage Configuration Network Backups Backup your voice records voicemails CDR Fax every day via SFTP protocol automatically Backup Configuration Enable Backup Ve Account root Password Oo Server Address ucmpbx backup server com Backup Time Cancel J Test Connection Save Figure 112 Data Sync Table 77 Data Sync Configuration Enable the auto backup function The default setting is No Enter the Account name on the SFTP backup server Enter the Password associate with the Account on the SFIP backup server Enter the SFTP server address Enter 0 23 to specify the backup hour of the day the configuration users could click on Test Connection The UCM6510 will then try connecting the server to make sure the server is up and accessible for the UCM6510 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 UCM6510 from a backup file users could go to web GUI gt Maintenance gt Backup gt Local Backup e A list of previous configuration backups is displayed on the web page Users could click on SS of the desired backup file and it will be restore
115. eters l Configure ring group name to identify the ring group Letters digits _ and Ring Group Name are allowed Extension Configure the ring group extension Select available users from the left side to the ring group member list on the right side Click on ASAE to arrange the order Select the ring strategy The default setting is Ring in order Ring Group Members e Ring simultaneously Ring all the members at the same time when there is incoming call to the ring group extension If any of the member answers the call it will Ring Strategy 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 Configure the number of seconds to ring each member If set to 0 it will l l keep ringing The default setting is 30 seconds Ring Timeout on Each Member Note Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 145 of 233 Enable Destination Secret Email Address ndstream Innovative IP Voice amp Video The actual ring timeout might be overridden by users if the phone has ring timeout settings as well If enabled users could select extension voicemail ring group IVR call queue voicemail group as the destination if the call to the ring group has no answer Secret and Email address are required if voicemail is selected as the destination Configure
116. evices ASSIGNMENT EI In the discovered list click on to open the edit dialog to assign an extension or multiple extensions to this device Hot Desking can also be enabled from this edit page Edit Device 000B822B2D94 Enable Hot Desking MAC Address 000B822B62094 IP Address 192 168 40 143 Version Model CXP2110 Account Select Firmware Version 1 0 0 25 Account 1 Account 2 Account 3 Account 4 500 el 602 l 614 Le 615 l Cancel Save Figure 31 Assign Extension To Device UCM6510 IP PBX User Manual Page 63 of 233 Andstream Innovative IP Voice amp Video After saving the edit dialog the XML config file will be generated in the UCM6510 Reboot the phone or trigger the phone to download the config file by clicking on icon for the entry in the zero config device list CREATE NEW DEVICE Users could also directly create a new device and assign the extension before the device is discovered by the UCM6510 Once the device is plugged in it can then be discovered and provisioned by the UCM6510 Click on Create New Device and the following dialog will show Enabled Hot Desking optional fill in the MAG address required IP address optional Version optional Model optional and the extension required to assign to the device Click on Save to add the device to the provision list Create New Device Enable Hot Desking MAC Address IP Address Version Model
117. ferred 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 IAX Trunk Options Trunk Frequency Trunk Time Stamps Configure the frequency of trunk frames in milliseconds The default 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 Unvalidated Call Numbers Call Number Limits IP or IP Range Firmware Version 1 0 0 25 Table 57 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 UCM6510 IP PBX User Manual Page 184 of 233 fndstream Innovative IP Voice amp Video Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 185 of 233 An
118. first or last name via his her phone s keypad The administrator can define the Dial By Name directory including the desired extensions in the directory and the searching type by first name or last name After dialing in the PBX IVR Auto Attendant will guide the caller to spell the digits to find the person in the Dial By Name directory This feature allows customers clients to use the guided automatic system to get in touch with the enterprise employees without having to know the extension number which brings convenience and improves business image for the enterprise DIAL BY NAME CONFIGURATION The administrators can create the dial by name group under web GUI gt PBX gt Call Features gt Dial By Name Create New Dial By Name Group Name DialByNameGP1 Extension 7101 Available Extensions Selected Extensions Group Options Query Type By Last Name By First Name By Full Name Select Type By Order By Menu Figure 78 Create Dial By Name Group 1 Group Name Enter the Group Name This is to identify the Dial By Name group The Dial By Name group can be used as the destination for inbound route and key pressing event for IVR The group name defined here will show up in the destination list when configuring IVR and inbound route Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 169 of 233 Create New IVR Name Extension Dial Other Extensions Dial Trunk Permission Welcome Prompt Digit T
119. for the extension Display message status for the extension Example 2 4 1 Message ee Description There are 2 urgent messages 4 messages in total and 1 message that has been already read Displays extension type e SIP User Type e AX User e Analog User FXS 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 e Click on O to refresh the extension status Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 195 of 233 andstream Innovative IP Voice amp Video e Click on one of the tabs Anse to display the corresponding extensions accordingly e Click on to expand the status detail table e Click on to hide the status detail table QUEUES Users could see all the configured call queue status in this section The following figure shows the call queue 6500 being in used Queues No Calls 4 1000 2 1001 Service Level SL 0 0 within Us Calls Completed 0 Calls Abandoned 1 Figure 86 Queue Status The current call status caller ID duration agent status service level calls summary completed abandoned are shown for the call queue The agent status is defined as below Table 66 Agent Status The agent is available idle The agent is ringing The agent is talking busy ja je je je
120. g Record Trace Here is the step to capture trace 1 Select FXO or FXS for Record Ports If the issue happens on FXO 1 select FXO port 1 to record the trace Select Record Direction Select Record File Mode to separate the record per direction or mix Click on Start Make a call via the analog port that has the issue Once done click on Stop Click on Download to download the analog record trace AE oe oS Pe Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 230 of 233 andstream Innovative IP Voice amp Video LH Status PBX Settings K 1 Maintenance gt gt Troubleshoo Analog Record Trace Record Ports FXO Ports wv 1 FXS Ports 1 Record Direction Both 3 Record File Mode separate Troubleshooting Start a Download Delete Output Result Capturing Analog record tracing has been stoped Compressing file now please wait Done Click on Download to download the captured packets SS Signaling Trace Figure 119 Troubleshooting Analog Trunks After capturing the trace users can download it for basic analysis Or you can contact Grandstream Technical support in the following link for further assistance if the issue is not resolved http www grandstream com index php support Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 231 of 233 Andstream Innovative IP Voice amp Video This page intentionally left blank Firmware Version 1 0 0 25 UCM6510 IP
121. g on the UCM6510 as a failover backup and etc e Click on in the table below B 55 IO create a new static route The configuration parameters are listed e Once added users can select to edit the static route e Select to delete the static route e Static routes configuration can be reset from LCD menu gt Network Menu Destination Netmask Gateway Interface Firmware Version 1 0 0 25 Table 7 UCM6510 Network Settings gt Static Routes Configure the destination IP address or the destination IP subnet for the UCM6510 to reach using the static route Example IP address 192 168 66 4 IP subnet 192 168 66 0 Configure the subnet mask for the above destination address If left blank the default value is 255 255 255 255 Example 255 255 255 0 Configure the gateway address so that the UCM6510 can reach the destination via this gateway Gateway address is optional Example 192 168 40 5 Specify the network interface LAN or WAN on the UCM6150 to reach the destination using the static route UCM6510 IP PBX User Manual Page 38 of 233 andstream Innovative IP Voice amp Video The following diagram shows a sample application of static route usage on UCM6510 Network 192 168 66 0 192 168 66 238 l e gt 8 A o J lg bk _ VPN connection is established between be ap network 192 168 40 0 and network 192 168 66 0 H ng 8 Ey zateway 192 168 40 3 Zi 4 Network
122. gh cannot be used Select the Caller ID scheme for this trunk If you are not sure which Caller ID Scheme scheme to choose please select Auto Detect The default setting is Bellcore Telcordia Enable automatic recording for the calls using this trunk The default setting is disabled The recording files are saved in external storage Auto Record gee device if plugged in and can be accessed under web GUI gt CDR gt Recording Files Tone Settings l Busy Detection is used to detect far end hangup or for detecting busy Busy Detection a signal The default setting is Yes 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 even 8 Please note that the higher the number is 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 a default setting is Yes l If Congestion Detection is enabled users can specify the number of Congestion Count l l SE congestion tones to wait for The default setting is 2 Select the country for tone settings If Custom is selected users could Tone Country manually configure the values for Busy Tone and Congestion Tone The default setting is United States of America USA Firmware Versi
123. grade G Upgrade Via HTTP Le GJ Firmware Server Path fw ipvideotalk com gs GJ Firmware File Prefix Firmware File Suffix CO HTTP HTTPS User Name 9 HTTP HTTPS Password Figure 107 Network Upgrade Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 219 of 233 E ten Innovative IP Voice amp Video Table 76 Network Upgrade Configuration Upgrade Via Allow users to choose the firmware upgrade method TFTP HTTP or HTTPS Firmware Server Path Define the server path for the firmware server Firmware File Prefix If configured only the firmware with the matching encrypted prefix will be downloaded and flashed into the UCM6510 Firmware File Suffix If configured only the firmware with the matching encrypted postfix will be downloaded and flashed into the UCM6510 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 process e Please be patient during the upgrading process Once done a reboot message will be displayed in the LCD e Manually reboot the UCM6510 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
124. haracters 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 well as invite other parties to join the conference by dialing 0 permission required from the invited party or 1 permission not required from the invited party during the conference call Note e If Public Mode is enabled the password is not required to join the conference bridge thus this field is invalid e The password has to be at least 4 characters UCM6510 IP PBX User Manual Page 125 of 233 Enable Caller Menu Record Conference Quiet Mode Wait For Admin Enable User Invite Announce Callers Public Mode Play Hold Music For First Caller Music On Hold Firmware Version 1 0 0 25 ndstream Innovative IP Voice amp Video 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 participan
125. he LED test select Back in the menu and the device will show the LED actual status again RTC Test Patterns Select 2022 02 22 22 22 or 2011 01 11 11 11 to start the RTC Real Time Clock test pattern Check the system time from LCD idle screen by pressing DOWN button or from web GUI gt System Status gt General page After the test reboot the device manually and the device will display UCM6510 IP PBX User Manual Page 24 of 233 E ten Innovative IP Voice amp Video the correct time e Hardware Testing Select Test SVIP to perform SVIP test on the device This is mainly for factory testing purpose which verifies the hardware connection inside the device The diagnostic result displays on 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 UCM6510 has LED indicators in the front to display connection status The following table shows the status definitions Table 4 UCM6510 LED INDICATORS LED Indicator LED Status Power 1 Power 2 PoE LAN SS Solid Connected WAN SS Fast Blinking Data Transferring USB SS Slow Blinking Trying to connect SD OFF Not Connected Phone 1 Phone 2 FXS Line 1 Line 2 FXO SS Solid Connected and working Fast Blinking 0 5s 0n 0 5s off No cable is connected or connected but the link is not T1 E1 J1 working at all SS Slow Blink
126. he UCM6510 Table 5 UCM6510 Network Settings gt Basic Settings Select Route Switch or Dual mode on the network interface of UCM6510 The default setting is Route e Route Method WAN port interface will be used for uplink connection LAN port interface will be used to serve as router e Switch WAN port interface will be used for uplink connection LAN port interface will Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 31 of 233 Preferred DNS Server andstream Innovative IP Voice amp Video be used as bridge for PC connection e Dual Both ports can be used for uplink connection Users will need assign LAN 1 or LAN 2 as the default interface in option Default Interface and configure Gateway IP for this interface if static IP is used for the interface Enter the preferred DNS server address WAN when Method is set to Route IP Method IP Address Subnet Mask Gateway IP DNS Server 1 DNS Server 2 User Name Password Layer 2 QoS 802 1Q VLAN Tag Layer 2 QoS 802 1p Priority Value 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 subnet mask address for static IP settings The default setting is 255 255 0 0 Enter the gateway IP address for static IP settings The default setting is 0 0 0 0 Enter the DNS server 1 address for static IP settings The default setting is 0 0 0 0
127. he UCM6510 is set to Route under web GUI gt Settings gt Network Settings gt Basic Settings Method port forwarding is available for configuration The port forwarding configuration is under web GUI gt Settings gt Network Settings gt Port Forwarding page Please see related settings in the table below Table 8 UCM6510 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 Protocol Type Select protocol type UDP Only TCP Only or TCP UDP for the forwarding in the selected port The default setting is UDP Only The following figures demonstrate a port forwarding example to provide phone s web Ul access to public side The UCM6510 network mode is set to Route The UCM6510 WAN port is connected to uplink switch with a public IP address configured e g 1 1 1 1 The UCM6510 LAN port provides DHCP pool that connects to multiple phone devices in the LAN network 192 168 2 x The UCM6510 is used as a router with gateway address 192 168 2 1 There is a GXP2160 connected under the LAN interface network of the UCM6510 It obtains IP address 192 168 2 100 from UCM6510 DHCP pool On the UCM6510 web Ul gt Settings gt Network Settings gt Port Forwarding configure a port forwarding entry as the figure shows below WAN Port This is the port opened up on the WAN side for access purpose LAN IP This is
128. he available codecs are PCMU PCMA GSM AAL2 G 726 32 G 726 G 722 G 729 G 723 ILBC ADPCM H 264 H 263 and H 263p In the selected codec list users can click on UP or DOWN arrow to adjust the order for the codec priority If enabled the UCM6510 will regularly send SIP OPTIONS to the device to check if the device is still online The default setting is No When Enable Qualify option is set to Yes configure the timeout in ms for the Qualify SIP message If no response is received within the timeout the device is considered offline The default setting is 1000ms When Enable Qualify option is set to Yes configure the interval in seconds of the SIP OPTIONS message sent to the device to check if the device is still online The default setting is 60 seconds UCM6510 IP PBX User Manual Page 113 of 233 E ten Innovative IP Voice amp Video 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 Fax Detection Note If enabled Fax Pass through cannot be used DIRECT OUTWARD DIALING DOD VIA VOIP TRUNKS The UCM6510 provides Direct Outward Dialing DOD which is a service of a local phone company or local exchange carrier that allows subscribers within a company s PBX system to connect to outside lines directly Example of how DOD is used Company ABC has a SIP trunk This SIP tr
129. he outbound route e Click on to delete the outbound route e On the UCM6510 the outbound route priority is based on Best matching pattern For example the UCM6510 has outbound route A with pattern 1xxx and outbound route B with pattern 10xx configured When dialing 1000 for outbound call outbound route B will always be used first This is because pattern 10xx is a better match than pattern 1xxx Only when there are multiple outbound routes with the same pattern configured users can click on to move the outbound route up down to arrange the priority among those outbound routes Table 35 Outbound Route Configuration Parameters Configure the name of the calling rule e g local long_distance and Calling Rule Name Ea etc Letters digits _ and are allowed e All patterns are prefixed with the _ e Special characters X Any Digit from 0 9 Z Any Digit from 1 9 Pattern N N Any Digit from 2 9 Wildcard Match one or more characters Wildcard Match zero or more characters immediately Example 12345 9 Any digit from 1 to 9 Configure the password for users to use this rule when making outbound Password calls Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 117 of 233 Andstream Innovative IP Voice amp Video Select privilege level for the outbound rule e Internal The lowest level required All users can use this rule e Local Users with Local National or International lev
130. he password for TLS authentication This is optional Password Permitted Specify a list of IP addresses permitted by CDR API This creates an AIP specific access control list Multiple entries are allowed For example 192 168 40 3 255 255 255 255 denies access from all IP addresses except 192 168 40 3 The format of the HTTPS request for the CDR API is as below https UCM IP Port cdrapi option1 J value amp option2 value amp By default the port number for the API is 8443 The options included in the request URI control the record matching and output format For CDR matching parameters all non empty parameters must have a match to return a record Parameters can appear in the URI in any order Multiple values given for caller or callee will be concatenated The following table shows the parameter list used in the CDR API Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 213 of 233 Field format numRecords offset caller callee startTime endTime Firmware Version 1 0 0 25 sn Innovative IP Voice amp Video Table 75 CDR API URI Parameters Value csv xml json Number 0 1000 Number Comma separated extensions ranges of extensions or regular expressions Example caller 5300 5302 5304 4 OR caller 5300 amp caller 5302 5304 amp caller 4Q Matches extensions 5300 5302 5303 5304 and any extension containing 4 as the second digit character Date an
131. how for users to select the exact date and time To Date Specify To 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 The call report will display as the following figure shows Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 208 of 233 andstream Innovative IP Voice amp Video Start Time 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 2013 07 03 17 53 11 Delete All Download Records Prev Next Figure 100 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 e Play Download Delete Recording File per entry If the entry has audio recording file for the call the three icons on the most right column will be activated for users to select In the following picture the second entry has audio recording
132. ial Extensions G Special Extensions BOGE Figure 77 Create New Event List Table 49 Event List Settings Configure the name of this event list for example office event list Please note the URI name cannot be the same as the extension name on the UCM6510 The valid characters are letters digits and Select the available extensions listed on the local UCM6510 to be monitored in the event list If LDAP sync is enabled between the UCM6510 and the peer UCM6510 the remote extensions will be listed under Available Extensions If not manually enter the remote extensions under Special Extensions field Manually enter the remote extensions in the peer register trunk to be monitored in the event list Valid format 5000 5001 9000 Remote extension monitoring works on the UCM6510 via event list BLF among Peer SIP trunks or Register SIP trunks register to each other Therefore please properly configure SIP trunks on the Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 166 of 233 san UCM6510 first before using remote BLF feature Please note the SIP end points need support event list BLF in order to monitor remote extensions When an event list is created on the UCM6510 and remote extensions are added to the list the UCM6510 will send out SIP SUBSCIRBE to the remote UCM6510 to obtain the remote extension status When the SIP end points registers and subscribes to the local UCM6510 event list it can obtain the r
133. ic IP settings The default setting is 0 0 0 0 UCM6510 IP PBX User Manual Page 32 of 233 Andstream Innovative IP Voice amp Video Enter the DNS server 1 address for static IP settings The default setting is DNS Server 1 0 0 0 0 DNS Server 2 Enter the DNS server 2 address for static IP settings User Name Enter the user name to connect via PPPoE Password Enter the password to connect via PPPoE Layer 2 QoS Assign the VLAN tag of the layer 2 QoS packets for LAN port The default value is 802 1Q VLAN Tag 0 Layer 2 QoS 802 1p Assign the priority value of the layer 2 QoS packets for LAN port The default Priority Value value is 0 LAN 1 LAN 2 when Method is set to Dual If Dual is selected as Method users will need assign the default interface to be LAN 1 mapped to UCM6510 WAN port or LAN 2 mapped to UCM6510 LAN port and then configure network settings for LAN 1 and LAN 2 The default interface is LAN 2 IP Method Select DHCP Static IP or PPPoE The default setting is DHCP IP Address Enter the IP address for static IP settings The default setting is 192 168 0 160 Default Interface Enter the subnet mask address for static IP settings The default setting is 255 255 0 0 Enter the gateway IP address for static IP settings when the port is assigned as default interface The default setting is 0 0 0 0 Subnet Mask Gateway IP Enter the DNS server 1 address for static IP settings The default setting is
134. ic password is automatically generated It Voicemail Password l is recommended to use the random generated password for security purpose Configure the Call Forward Unconditional target number so that the 7 incoming call to this extension will be always forwarded to the target Call Forward Unconditional l l number If not configured the Call Forward Unconditional feature is deactivated The default setting is deactivated Configure the Call Forward No Answer target number so that the incoming call to this extension will be forwarded to the target number if Call Forward No Answer the call is not answered until the ringing times out If not configured the Call Forward No Answer feature is deactivated The default setting is deactivated Configure the Call Forward Busy target number so that the incoming call Call F JB to this extension will be forwarded to the target number if the call is all Forward Bus d rejected or the extension is in talking busy status If not configured the Call Forward Busy feature is deactivated The default setting is Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 69 of 233 Andstream Innovative IP Voice amp Video 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 If not specified the default ring timeout is 60 seconds on the UCM6510 which can be configured in the glob
135. ication configuration ALERT LOG Under web GUI gt Status gt System Events gt Alert Log system messages are listed when the alert is triggered for the configured system events The following picture shows disk usage alert log We can tell the detect cycle for the disk usage is 10 minutes and the disk usage is restored to normal after the administrator cleans up the disk storage below the threshold Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 206 of 233 ndstream Innovative IP Voice amp Video 2013 10 09 21 32 00 Disk Usage Generate Alert Disk usage exceeds the threshold 2013 10 09 21 42 00 Disk Usage Generate Alert Disk usage exceeds the threshold 2013 10 09 21 52 00 Disk Usage Generate Alert Disk usage exceeds the threshold 2013 10 09 22 02 00 Disk Usage Restore to normal Disk usage has been restored to normal Figure 98 System Events gt Alert Log ALERT CONTACT Users could add administrator s Email address under web GUI gt Status gt System Events gt Alert Contact to send the alert notification to Up to 10 Email addresses can be added CDR 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 UCM6510 e Start Time Format 2013 03 27 16 47 03 e Call From Format John Doe lt 6012 gt e Call To Format
136. ice Info Network Info and Web Info which do not have Back option simply press the OK button to go back to the previous menu Additionally 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 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 23 of 233 san Innovative IP Voice amp Video The following table shows the LCD menu options View Events Device Info Network Info Network Menu Factory Menu Firmware Version 1 0 0 25 Table 3 LCD Menu Options Critical Events Other Events Hardware Hardware version number Software Software version number P N Part number WAN MAC WAN side MAC address LAN MAG LAN side MAC address Uptime System up time since the last reboot WAN Mode DHCP Static IP or PPPoE WAN IP IP address WAN Subnet Mask LAN IP IP address LAN Subnet Mask WAN Mode Select WAN mode as DHCP Static IP or PPPoE Static Routes Reset Click to reset the static route setting Reboot Factory Reset LCD Test Patterns Press OK to start Then press Down button to test different LCD patterns When done press OK button to exit Fan Mode Select Auto or On LED Test Patterns Select All On All Off or Blinking and check LED status for USB SD T1 E1 J1 Phone 1 Phone 2 Line 1 Line 2 ports After t
137. imeout Response Timeout Response Timeout Prompt Invalid Prompt WRA 7000 Internal welcome e Prompt 3 10 ivr createtimeout e invalid e andstream Innovative IP Voice amp Video Response Timeout Repeat Loops Invalid Repeat Loops 3 o Language Default Key Fressing Events Extension Conference Rooms Figure 79 Dial By Name Group In IVR Key Pressing Events Create New Inbound Rule Trunks SIP Trunks PeerSIFTrunk e DID Pattern 1000 Default Destination Dial By Name DialByNameGP1 e CU Privilege Level Internal D Dial Trunk Time Condition Click to add Time Condition Save Figure 80 Dial By Name Group In Inbound Route Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 170 of 233 clean 2 Extension Configure the direct dial extension for the Dial By Name group 3 Available Extensions Selected Extensions Select available extensions from the left side to the right side as the directory for the Dial By Name group Only the selected extensions here can be reached by the Dial By Name IVR when dialing into this group The extensions here must have valid first name and last name configured under web GUI gt PBX gt Basic Call Routes gt Extensions in order to be searchable in Dial By Name directory through IVR By specifying the extensions here the administrators can make sure unscreened calls will not r
138. imum value is reached the caller will be treated with busy tone followed by the next calling rule after attempting to enter the queue UCM6510 IP PBX User Manual Page 150 of 233 ndstream Innovative IP Voice amp Video If enabled the UCM6510 will report to the agent the duration of time of Report Hold Time the call before the caller is connected to the agent The default setting is No 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 users to be the static agents in the call queue Agents Choose from the available users on the left to the static agents list on the right Click on AMY Ato arrange the order e Click on 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 Figure 66 Agent Login Settings 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
139. in menu UCM6510 IP PBX User Manual Page 140 of 233 3 Advanced options 0 Mailbox options san Innovative IP Voice amp Video Cancel 1 Send a reply 2 Call the person who sent this message 3 Hear the message envelop 4 Leave a message Return to the main menu 1 Accept this recording 1 Record your unavailable message 2 Listen to it 3 Re record your message 1 Accept this recording 2 Record your busy message 2 Listen to it 3 Re record your message 1 Accept this recording 3 Record your name 2 Listen to it 3 Re record your message 1 Accept this recording 4 Record temporary greeting 2 Listen to it 3 Re record your message 5 Change your password Return to the main menu VOICEMAIL EMAIL SETTINGS The UCM6510 can be configured to send the voicemail as attachment to Email Click on Voicemail Email Settings button to configure the Email attributes and content Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 141 of 233 Andstream Innovative IP Voice amp Video Voicemail Email Settings Gi Atach Recordings to E mail Template for Voicemail Emails Template Variables 7 TAB Subject New voicemail from VM_CALLERID for HVM_MAILBOX Message Hello SCHM NAME you received a message lasting DVM DUR at VM DATE from 34V M_CALLERID This is message BVM_MSGNUM in your voicemail Inbox fe Load Default Settings Figure 59 Voicemail Email Settings
140. ination based on the number dialed which could be local extensions conference call queue ring group paging intercom group IVR voicemail groups and Fax extension as configured in DID destination If the dialed number matches the DID pattern the call will be allowed to go through Configure the number of digits to be stripped in Strip option e Dial By Name BLACKLIST CONFIGURATIONS In the UCM6510 Blacklist is supported for all inbound routes Users could enable the Blacklist feature manage the Blacklist by clicking on Blacklist Blacklist Blacklist Enable Blacklist Manage Blacklist list 1234567 E 12345678 r Total 2 Figure 50 Blacklist Configuration Parameters e G Add Blacklist Number 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 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 122 of 233 ndstream Innovative IP Voice amp Video e To remove a number from the Blacklist select the number in Blacklist list and click on A Note Users could also add a number to the Blacklist or remove a number from the Blacklist by dialing the feature code for Blacklist Add default 40 and Blacklist Remove default 41 from an extension The feature code can be configured under web GUI gt PBX gt Internal O
141. ing 1s on 1s off Connected but the link is only working one way Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 25 of 233 ream Innovative IP Voice amp Video USE THE WEB GUI ACCESS WEB GUI The UCM6510 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 IE version 8 Mozilla Firefox Google Chrome and etc O e F ndstean UCM6510 IPPBX Appliance Passwor Figure 4 UCM6510 web GUI Login Page To access the web GUI 1 Connect the computer to the same network as the UCM6510 2 Ensure the device is properly powered up and shows its IP address on the LCD 3 Open a web browser on the computer and enter the IP address in the address bar The web login page will display as shown in Figure 4 UCM6510 web GUI Login Page 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 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 26 of 233 E ten Innovative IP Voice amp Video A Note By default the UCM6510 has Redirect From Port 80 enabled Therefore if users type in the UCM6510 IP address in the web browser the web page will be automatically redirected to the page using HTTPS and port 80
142. ing is Allow All If enabled users will not need enter the PIN Set required by the Skip Trunk Auth WER outbound rule to make outbound calls The default setting is No Select audio and video codec for the extension The available codecs are PCMU PCMA GSM AAL2 G 726 32 G 726 G 722 G 729 G 723 ILBC ADPCM H 264 H 263 and H 263p In the selected codec list users can click on UP or DOWN arrow to adjust the order for the codec priority Codec Preference CREATE NEW FXS EXTENSION To manually create new FXS user go to web GUI gt PBX gt Basic Call Routes gt Extensions Click on Create New User gt Create New FXS Extension and a new dialog window will show for users to fill in the extension information The configuration parameters are as follows Table 20 FXS Extension Configuration Parameters Extension The extension number associated with the user Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 71 of 233 Analog Station CallerID Number Permission Enable Voicemail Voicemail Password Call Forward Unconditional Call Forward No Answer Call Forward Busy Ring Timeout Firmware Version 1 0 0 25 san Select the FXS port to be assigned for this extension Configure the CallerID Number that would be applied for outgoing calls from this user Note The ability to manipulate your outbound Caller ID may be limited by your VolP provider Assign permission level to the user The avai
143. ing is No Enable Jitter Buffer Configure the time in ms to buffer This is the jitter buffer size used in Jitter Buffer Size Fixed jitter buffer or used as the initial time for adaptive jitter buffer The default setting is 100 Configure the maximum time in ms to buffer for Adaptive jitter buffer Max Jitter Buffer implementation or used as the jitter buffer size for Fixed jitter buffer implementation The default setting is 200 Configure the jitter buffer implementation on the sending side of a SIP 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 INTERNAL OPTIONS RTP SETTINGS Table 53 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 RIP stream will be dropped The default setting is Disable Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 181 of 233 E ten Innovative IP Voice amp Video Configure to enable or disable RTP Checksums on RIP traffic The RTP Checksums GE l default setting is Disable INTERNAL OPTIONS STUN MONITOR Table 54 Inte
144. interval is shorter than the value of Polarity on Answer Delay the d Polarity Reversal will be ignored Otherwise the FXO will onhook to disconnect the call The default setting is 600ms l This is the periodic time in ms that the UCM6510 will use to check on a Current Disconnect Threshold vs voltage drop in the line The default setting is 200 The valid range is 50 to 3000 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 85 of 233 Gran Innovative IP Voice amp Video 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 Ring Timeout This value can be used to configure how long it takes before the UCM6510 considers a non ringing line with hangup activity The default setting is 8000 Configure the RX gain for the receiving channel of analog FXO port The E valid range is from 13 5 dB to 12 0 dB The default setting is 0 TX Gain 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 Use CallerID Configure to enable CallerID 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 The default setting is No Fax Detection Note If enabled Fax Pass throu
145. ire Min SE Session Refresher san 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 default setting is 1800 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 SIP SETTINGS TCP and TLS A Note The configuration in this section require system reboot to take effect TCP Enable TCP Bind Address TLS Enable TLS Bind Address Firmware Version 1 0 0 25 Table 61 SIP Settings TCP and TLS Configure to allow incoming TCP connections with the UCM6510 The default setting is No Configure the IP address for TCP server to bind to 0 0 0 0 means binding to all interfaces The port number is optional If not specified 5060 will be used Configure to allow incoming TLS connections with the UCM6510 The default setting is No Configure the IP address for TLS server to bind to 0 0 0 0 means binding to all interfaces The port number is optional If not specified 5061 will be used 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 UCM6510 IP PBX User Manual P
146. irmware Version 1 0 0 25 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 VolP provider server of the trunk Configure the SIP transport protocol to be used in this trunk The default setting is All UDP Primary e UDP Only e TCP Only e TLS Only e All UDP Primary UDP is the primary transport protocol when all the other SIP transport methods are available too e All TCP Primary TCP is the primary transport protocol when all the other SIP transport methods are available too e 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 extension 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 CallerID with this option and this option will be ignored When making outgoing calls the following rules are used to determine which CallerID will be used if they exist e The CallerlD configured for the extension will be looked up first e If Keep Trunk CID is enabled the CallerID configured for the trunk will be used e If the above two are missing the Global Outbound CID defined in web GUI gt PB
147. irmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 222 of 233 andstream Innovative IP Voice amp Video LOCAL BACKUP Users could backup the UCM6510 configurations for restore purpose under web GUI gt Maintenance gt Backup gt Local Backup Before creating new backup file select the backup option first e lf the Config File is selected only the backup file will be saved in the flash of the UCM6510 e If Voice File Voicemail File Voice Records CDR or VFAX is selected external storage devices USB Flash drive or SD Card will be required because the backup file might be too large Click on Create New Backup button to start backup Once the backup is done the list of the backups e will be displayed with date and time in the web page Users can download restore a or delete it from the UCM6510 internal storage or the external device Maintenance gt gt Backup gt gt Local Backup Manage Configuration Backups Backup Configuration Create New Backup Upload Backup File Config File Voice File Voicemail File Voice Records CDR VFAX List of Previous Configuration Backups The files saved in the local disk Date backup_2014may19_123223 tar 2014 05 19 12 32 29 UTC 07 00 Total 4 Show AN Go to Figure 111 Local Backup DATA SYNC 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 und
148. is Internal If the user tries to dial outbound calls after dialing into the DISA the UCM6510 will compared the DISA s permission level with the outbound route s privilege level If the DISA s permission level is higher than or equal to the outbound UCM6510 IP PBX User Manual Page 163 of 233 Response Timeout Digit Timeout Allow Hangup ndstream route s privilege level the call will be allowed to go through Configure the maximum amount of time the UCM6510 will wait before hanging up if the user dials an incomplete or invalid number The default setting is 10 seconds Configure the maximum amount of time permitted between digits when the user is typing the extension The default setting is 5 seconds If enabled during an active call users can enter the UCM6510 hangup feature code 0 by default to disconnect the call or hang up directly A new dial tone will be heard shortly for the user to make a new call The default setting is No Once successfully created users can configure the inbound route destination as DISA or IVR key event as DISA When dialing into DISA users will be prompted with password first After entering the correct password a second dial tone will be heard for the users to dial out Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 164 of 233 E ten Innovative IP Voice amp Video BLF AND EVENT LIST BLF The UCM6510 supports BLF monitoring for extensions ring group call queue
149. isted in the table below Table 33 SIP Trunk Configuration Parameters Create New SIP Trunk Select the VoIP trunk type Type e Register SIP Trunk e Peer SIP Trunk Configure a unique label to identify this trunk when listed in outbound rules inbound rules and etc Provider Name Configure the IP address or URL for the VoIP provider s server of the Host Name ee unk If enabled the trunk CID will not be overridden by extension s CID when Keep Trunk CID l l ao the extension has CID configured The default setting is No Enter the username to register to the trunk from the provider when Username Register SIP Trunk type is selected Enter the password to register to the trunk from the provider when Password l Register SIP Trunk is selected Auth ID Enter the Authentication ID for Register Trunk type Outbound Proxy Enter the IP address or URL of the outbound proxy for Register SIP Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 107 of 233 Auto Record dstream Innovative IP Voice amp Video Trunk type Enable automatic recording for the calls using this trunk for SIP trunk only The default setting is disabled The recording files are saved in external storage if plugged in and can be accessed under web GUI gt CDR gt Recording Files Peer SIP Trunk Configuration Parameters Provider Name Host Name Transport Keep Trunk CID Caller ID CalleriID Name Codec Preference F
150. ive if this channel Music On Hold Interpret l l l l has no music class configured and the bridged channel putting the call on hold has no Music On Hold Suggest setting Specify which Music On Hold class to suggest to the bridged channel Music On Hold Suggest when putting the call on hold Bandwidth Configure the bandwidth for IAX settings The default setting is Low IAX SETTINGS REGISTRATION Table 56 IAX Settings Registration IAX Registration Options Configure the minimum period in seconds of registration The default Min Reg Expire au setting is 60 Configure the maximum period in seconds of registration The default Max Reg Expire RS setting is 3600 IAX Thread Count Configure the number of IAX helper threads The default setting is 10 IAX Max Thread Count Configure the maximum number of IAX threads allowed The default Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 183 of 233 Auto Kill Authentication Debugging Codec Priority Type of Service Andstream 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 pre
151. joined user from the conference Decrease the volume of the conference call Increase the volume of the conference call Decrease your volume Oo N O A OO PD Increase your volume More options e 1 List all users currently in the conference call e 2 Kick all non Administrator participants from the conference call e 3 Mute Unmute all non Administrator participants from the conference call e 4 Enable disable conference call recording e 8 Exit the caller menu and return to the conference Conference User IVR Menu 1 Mute unmute yourself 4 6 7 9 8 Decrease the volume of the conference call Increase the volume of the conference call Decrease your volume Increase your volume Exit the caller menu and return to the conference A Note When there is participant in the conference the conference bridge configuration cannot be modified RECORD CONFERENCE The UCM6510 allows users to record the conference call and retrieve the recording from web GUI gt PBX gt Call Features gt Conference Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 129 of 233 ndstream Innovative IP Voice amp Video 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 The recording files will be li
152. kage by selecting Check Prompt List under web GUI gt PBX gt Internal Options gt Language Configure the maximum number of calls allowed for each remote IP address Max Number of Calls Configure to enable disable requiring call token If set to Auto it might l lock out users who depend on backward compatibility when peer Require Call Token d authentication credentials are shared between physical endpoints The default setting is Yes Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 70 of 233 andstream 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 send the received Fax to the default Email address in Fax setting page under web GUI gt PBX Fax Detection 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 e Allow All Device in any network can register this extension e Local Subnet Only Strategy Only the user in specific subnet can register this extension Up to three subnet addresses can be specified e A Specific IP Address Only the device on the specific IP address can register this extension The default sett
153. ket loss round trip min avg max 13 5600 14 588 19 300 ms Done Figure 117 PING andstream Innovative IP Voice amp Video 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 G Target Host Output Result traceroute Dignostic run www google com Start Or traceroute to www google com 74 125 224 179 nr T ae 81 81 csw3 LosAngeles1 Level3 net 4 69 137 10 14 700 ms 33 675 ms 14 675 ms ae 160 edgel LosAngeles9 Level3 net 4 69 144 10 14 000 ms ae 4 90 edge1 LosAngeles9_Level3_net 4 69 144 202 17 900 ms 11 725 ms 9 GOOGLE NGC edget LosAngeles9 Level3 net 4 53 276 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 lax02501 inf19 te100 net 74 125 224 179 13 775 ms 11 925 ms Done PRI SS7 MFC R2 SIGNALING TRACE Figure 118 Traceroute Please see section DIGITAL TRUNK TROUBLESHOOTING Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 229 of 233 ndstream Innovative IP Voice amp Video ANALOG RECORD TRACE Analog record trace can be used to troubleshoot analog trunk issue for example the UCM6510 user has caller ID issue for incoming call from Analog trunk Users can access analog record trance under web GUI gt Maintenance gt Troubleshooting gt Analo
154. l Table 40 Voicemail Settings er Configure the maximum number of seconds for the voicemail greeting ax Greetin 9 The default setting is 60 seconds If enabled the caller can press 0 to exit the voicemail application and Dial 0 For Operator 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 Max Messages Per Folder l l EEN 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 e 1 minute Max Message Time e 2minutes e 5 minutes e 15 minutes e 30 minutes e Unlimited Configure the minimum effective duration in seconds of a voicemail message Messages will be automatically deleted if the duration is shorter than the Min Effective Message Time The default setting is 3 seconds The available options are e No minimum e 1 second Min Effective Message Time e 2 seconds e 3 seconds e 4 seconds e 5 seconds Note Silence and noise duration are not counted in message time Announce Message Caller ID If enabled the caller ID of the user who has left the message will be Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 139 of 233 Announce Message Duration Play E
155. l Efe PX Francais Espafiol Francais Portugues Portugu s FYCCKANA Pycckui Figure 5 UCM6510 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 Apply Changes English Logout Status settings Maintenance 2014 10 01 21 16 UTC 04 00 Create New User Modify Selected Extensions Delete Selected Extensions Batch Add SIP Extensi Import Extensi Email To User Auto Refresh v Extension Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 28 of 233 E ten Innovative IP Voice amp Video Figure 6 UCM6510 web GUI Apply Changes MAKE YOUR FIRST CALL Power up the UCM6510 and your SIP end point phone Connect both devices to the network Then follow the steps below to make your first call 1 Log in the UCM6510 web GUI go to PBX gt Basic Call Routes gt Extensions 2 Click on Create New SIP Extension 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 with the SIP User ID SIP server and SIP Password information The SIP server address is the UCM6510 IP address 4 When your phone i
156. l Group e IVR e Ring Group e Queues e Page Group e Fax e IVR Prompt e Hangup e DISA e Dial By Name UCM6510 IP PBX User Manual Page 132 of 233 Andstream Innovative IP Voice amp Video 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 VR Name Main Menu Extension 7011 Dial Other Extensions Dial Trunk Permission Internal Welcome Prompt None Figure 53 Click On Prompt To Create IVR Prompt Once the IVR prompt file is successfully added to the UCM6510 it will be added into the prompt list options for users to select in different IVR scenarios RECORD NEW IVR PROMPT In the UCM6510 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 Le Dial This User Extension to 6000 Le Record a New Voice Prompt Figure 54 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 UCM6510 to record the IVR prompt Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 133 of 233 Andstream
157. l need specify the network interface LAN WAN or Both for the incoming traffic e OUT Select the service type e FTP e SSH e Telnet e TFTP Service ee e 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 Please note if the source or the destination field is left blank it will be used as Anywhere The new rule will be listed at the bottom of the page with sequence number rule name action protocol type source destination and operation Users can click on to edit the rule or click on to delete the rule Save the change and reboot the device for the configuration to take effect DYNAMIC DEFENSE Dynamic defense can blacklist hosts dynamically when the UCM6510 is set to Route under web GUI gt Settings gt Network Settings gt Basic Settings Method If enabled the traffic via TCP connection coming into the UCM6510 can be monitored which helps prevent massive connection attempts or brute force attacks to the device The blacklist can be created and updated by the UCM6510 firewall which will then be displayed in the web page Please refer to the following table for dynamic defense options on the UCM6510 Table 12 UCM6510 Firewall Dynamic Defense Dynamic Defense Enable dynamic defense The default setting is disabled Enable Configure the dynamic defense periodic time interval in minutes If the Periodical Ti nu
158. l on their transmit side The parameter determines whether the clock signal from the far end of the UCM6510 IP PBX User Manual Page 100 of 233 dstream Innovative IP Voice amp Video E1 T1 J1 is used as the master source of clock timing If the far end is used as the master the PBX system clock will synchronize to tt e Master The port will never be used as a source of timing This is appropriate when you know the far end should always be a slave to yOu e Slave The equipment at the far end of the E1 T1 link is the preferred source of the master clock SS7 Variant Select ITU ANSI or CHINA Originating point code is used to identify the node originating the WA l message always provided by the operator ISP Originating Point Code l e ITU Format decimal number e ANSI amp CHINA Format decimal number or XXX XXX XXX Destination point code is the address to send the message to always be er l provided by the operator ISP Destination Point Code l e ITU Format decimal number e ANSI amp CHINA Format decimal number or XXX XXX XXX Network Indicator NI should match in nodes otherwise it might cause Network Indicator issues Users can select National National Spare International or International Spare Usually National or International is used The line build out LBO is the distance between the operators and the Sa PBX Please use the default value 0dB unless the distance is long RX Gain Configure
159. lable 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 an outbound rule s privilege in order to make outbound calls using this rule If the outbound rule privilege is disabled this option will not take effect Enable voicemail for the user so that the call will be forwarded to the user s voicemail if there is no answer or the call is rejected The default setting is Yes Configure voicemail password digits only for the user to access the voicemail box A random numeric password is automatically generated when the extension is created It is recommended to use the random generated password for security purpose Configure the Call Forward Unconditional target number so that the incoming call to this extension will be always forwarded to the target number If not configured the Call Forward Unconditional feature is deactivated The default setting is deactivated Configure the Call Forward No Answer target number so that the incoming call to this extension will be forwarded to the target number if the call is not answered until the ringing times out If not configured the Call Forward No Answer feature is deactivated The default setting is deactivated Configure the Call Forward Busy target number so that the incoming call to this extension will be forwarded to the target numb
160. lastdata gt SIP 5301 60 lt lastdata gt lt start gt 2013 12 03 14 02 23 lt start gt lt answer gt 2013 12 03 14 02 27 lt answer gt lt end gt 2013 12 03 14 02 31 lt end gt lt duration gt 8 lt duration gt lt billsec gt 4 lt billsec gt lt disposition gt ANSWERED lt disposition gt lt a maflags gt DOCUMENTATION lt amaflags gt lt uniqueid gt 1386100943 2 lt uniqueid gt lt userfield gt EXT lt userfie ld gt lt channel ext gt 5300 lt channel ext gt lt dstchannel_ ext gt 5301 lt dstchannel ext gt lt service gt s lt service gt lt cdr gt lt root gt cdr Acctld 62 accountcode src 5300 dst 5301 dcontext from internal clid on01 lt 5300 gt channel SIP 5300 00000000 dstchannel SIP 5301 00000001 lastapp Dial lastdata SIP 5301 60 start 2013 12 03 11 46 40 answer 2013 12 03 11 46 43 end 2013 12 03 11 46 49 duration 9 billsec 6 disposition ANSWERED amaflags DOCUMENTATION uniqueid 1386092800 0 userfield EXT channel evt 5300 dstchannel ext 5301 service s Acctld 63 accountcode src 5300 dst 5301 dcontext from internal clid on01 lt 5300 gt channel SIP 5300 00000000 dstchannel SIP 5301 00000001 lastapp Dial lastdata SIP 5301 60 start 2013 12 03 14 01 41 answer 2013 12 03 14 01 43 end 2013 12 03 14 01 46 duration 5
161. le 47 FAX T 38 Settings Configure to enable Error Correction Mode ECM for the Fax The default setting is Yes Configure the maximum transfer rate during the Fax rate negotiation The possible values are 2400 4800 7200 9600 12000 and 14400 The default setting is 14400 Configure the minimum transfer rate during the Fax rate negotiation The possible values are 2400 4800 7200 9600 12000 and 14000 The default setting is 2400 Configure the Email address to send the received Fax to if user s Email address cannot be found 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 Fill in the Subject and Message content to be used in the Email when sending the Fax to the users The template variables are e CALLERIDNUM Caller ID Number e CALLERIDNAME Caller ID Name e RECEIVEEXTEN The extension to receive the Fax e FAXPAGES Number of pages in the Fax e VM DATE The date and time when the Fax is received UCM6510 IP PBX User Manual Page 159 of 233 Andstream Innovative IP Voice amp Video e Click on e to edit the Fax extension e Click on to delete the Fax extension SAMPLE CONFIGURATION TO RECEIVE FAX FROM PSTN LINE The following instructions describes how to use the UCM6510 to receive Fax from PSTN line on the Fax machine connected
162. left blank Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 22 of 233 E ten Innovative IP Voice amp Video GETTING STARTED The UCM6510 provides LCD interface LED indication and web GUI configuration interface e The LCD displays hardware software and network information Users could also navigate in the LCD menu for device information and basic network configuration e The LED indication at the front of the device provides interface connection and activity status e The web GUI gives users access to all the configurations and options for UCM6510 setup This section provides step by step instructions on how to use the LCD menu LED indicators and web GUI of the UCM6510 Once the basic settings are done users could start making calls from UCM6510 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 UCM6510 hardware version e g V1 4A and IP address Press Down button and the system time will be displayed e g 2014 10 21 14 20 e Menu Access Press OK button to start browsing menu options Please see menu options in Table 3 LCD Menu Options e Menu Navigation Press the Down arrow key to browser different menu options Press the OK button to select an entry e Exit If Back option is available in the menu select it to go back to the previous menu For Dev
163. llow 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 Default code 0 Enter the code during active call It will disconnect the call 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 Default code 72 Enter the code during active call to park the call UCM6510 IP PBX User Manual Page 173 of 233 Audio Mix Record clean 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 Default code 3 Enter the code followed by or SEND to start recording the audio call and the UCM6510 will mix the streams natively on the fly as the call is in progress 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 DND Call Forward Do Not Disturb DND Activate Do Not Disturb DND Deactivate Call Forward Busy Activate Call Forward Busy Deactivate Call Forward No Answer Activate Call Forward No Answer Deactivate Call Forward Unconditional Activate Call Forward Uncon
164. lue 0dB unless the distance is long RX Gain Configure the RX gain for the receiving channel of digital port The valid range is from 24dB to 12dB TX Gain Configure the TX Gain for the transmitting channel of digital port The valid range is 24dB to 12dB This configured whether to play the ringback tone from local UCM6510 or not If enabled the local UCM6510 will play ringback tone to the caller Play Local RBT ra Otherwise the caller will listen to the tone from peer device The default setting is disabled Advanced Settings Bee Coding Select HDB3 or AMI CRC Select whether to use CRC4 or not l MFC R2 value in milliseconds for MF timeout Values smaller than 500ms MF Back Timeout ms are not recommended 1 represents default value MFC R2 value in milliseconds for the metering pulse timeout Metering pulse is sent by some telcos for some R2 variants during a call Metering Pulse Timeout ms presumably for billing purposes to indicate costs Should not last more than 500ms 1 represents default value and for Argentina the default value is 400ms for others is Oms Brazil has a special calling party category for collect calls llamadas por cobrar instead of using the operator as in Mexico The R2 spec in Brazil Alllow Collect Calls Says a special GB tone should be used to reject collect calls By default this is disabled which means collect calls will be blocked Some PBXs require a double answer process to
165. mber of TCP connections from a host exceeds the Connection Threshold eriodical Time int within this period this host will be added into Blacklist The valid value is nterva between 1 and 59 when dynamic defense is turned on The default setting is 59 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 44 of 233 Blacklist Update Interval Connection Threshold Dynamic Defense Whitelist san Innovative IP Voice amp Video Configure the blacklist update time interval in seconds The default setting is 120 This defines how long the IP will be blocked once added into the UCM6510 blacklist For example if it s set to 300 seconds the blocked IP address will only be able to establish TCP connection with the UCM6510 again after 300 seconds Configure the connection threshold Once the number of connections from the same host reaches the threshold during Periodical Time Interval it will be added into the blacklist The default setting is 100 Configure the dynamic defense whitelist This is a list of IPs that will not be blocked by the UCM6510 For example 192 168 1 3 192 168 1 4 The following figure shows a configuration example like this e Ifa host at IP address 192 168 40 7 initiates more than 20 TCP connections to the UCM6510 within 1 minute it will be added into UCM6510 blacklist e This host 192 168 40 7 will be blocked by the UCM6510 for 300 seconds e Since IP address 1 92 168 40 5 is i
166. me Email FirstName Department MobileNumber HomeNumber D D LastName T T T Fax Figure 22 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 Assuming the server base dn is dc pbx dc com configure the LDAP clients as follows case insensitive Base DN dc pbx dc com Login DN Please leave this field empty Password Please leave this field empty Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 50 of 233 Gran Anonymous Please enable this option Filter Caller lDName AccountNumber Port 389 To configure Grandstream IP phones as the LDAP client please refer to the following example Server Address The IP address or domain name of the UCM6510 Base DN dc pbx dc com User Name Please leave this field empty Password Please leave this field empty LDAP Name Attribute CallerIDName Email Department FirstName LastName LDAP Number Attribute AccountNumber MobileNumber HomeNumber Fax LDAP Number Filter AccountNumber LDAP Name Filter CallerIDName LDAP Display Name AccountNumber CallerIDName LDAP Version If existed please select LDAP Version 3 Port 389 The following figure shows the configuration information on a Grandstream GXP2200 to successfully use the LDA
167. me it will only be looked up once Specify an external host name which is similar to External Address except the host name will be looked up periodically based on the 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 UCM6510 is behind a Static NAT or PAT UCM6510 IP PBX User Manual Page 190 of 233 External TLS Port Local Network Address 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 DI MF DTMF Mode RTP Timeout Firmware Version 1 0 0 25 E ten Innovative IP Voice amp Video Configures the externally mapped TLS port when UCM6510 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 Table 63 SIP Settings ToS Configure the Type of Service for SIP packets The default setting is None Configure the Type of Service for RTP audio packets The default setting is None Configure the Type of Service for RTP video packets The default setting is None Configure the defaul
168. means it s a SIP call There are three possible format a SIP NUM XXXXXX where NUM is the local SIP extension number The last XXXXX is a random string and can be ignored c SIP trunk_X NUM where trunk_X is the internal trunk name and NUM is the number to dial out through the trunk c SIP trunk_X XXXXXX where trunk_X is the internal trunk name and it is an inbound call from this trunk The last XXXXX is a random string and can be ignored Sample 3 call from callto context start time answer time end time call time talk time source channel dest channel status s default 1 30 2014 14 30 1 30 2014 14 37 386 0 DAHDI pseudo 1665832080 NO ANSWER s default 1 30 2014 14 30 1 30 2014 14 37 390 0 DAHDI pseudo 1946772436 NO ANSWER Figure 105 Downloaded CDR File Sample Source Channel and Dest Channel 3 This is a very special channel name If it shows up most likely it means a conference call There are some other possible values but these values are almost the application name which are used by the dialplan IAX2 NUM XXXXXXX it means this is an IAX call Local from internal XXXXX it is used internally to do some special feature procedure We can simply ignore it Hangup the call is hung up from the dialplan This indicates there are some errors or it has run into abnormal cases Playback play some prompts to you such as 183 response or run into an IVR ReadExten collect numbers from user It may occur when you input PIN c
169. mp caller 5300 amp caller 6300 6399 amp callee 5Q Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 215 of 233 ndstream Innovative IP Voice amp Video Query 3 Request all records of calls placed on extensions containing substring 53 prior to January 23 2013 00 00 00 UTC to extensions 5300 5309 with results in CSV format https 192 168 254 200 8088 cdrapi caller 53 amp callee 5300 5309 amp endTime 2013 01 23 OR https 192 168 254 200 8088 cdrapi caller 53 amp callee 530_ amp endTime 2013 01 23T00 00 00 Query 4 Request all records of calls placed by an Anonymous caller during July 2013 Central Standard Time to extensions starting with 2 or 34 or ending with 5 with results in CSV format https 192 168 254 200 8088 cdrapi caller Anonymous amp callee 2 34 5 amp startTime 2013 07 01T00 00 00 06 00 amp endTime 2013 07 317T23 59 59 06 00 Query 5 Request all records during July 2013 Central Standard Time 200 at a time with results in CSV format https 192 168 254 200 8088 cdrapi startTime 2013 07 01T00 00 00 06 00 amp end Time 2013 07 31 1T23 59 59 06 00 amp numRecords 200 amp offset 0 THEN https 192 168 254 200 8088 cdrapi sstartTime 2013 07 01T00 00 00 06 00 amp endTime 2013 07 31 T23 59 59 06 00 amp numRecords 200 amp offset 200 THEN https 192 168 254 200 8088 cdrapi startTime 2013 07 01T00 00 00 06 00 amp end Time 2013 07 31123 59 59 06 00 amp numRecords 200 amp offset 400
170. n whitelist if the host at IP address 192 168 40 5 initiates more than 20 TCP connections to the UCM6510 within 1 minute it will not be added into UCM6510 blacklist It can still establish TCP connection with the UCM6510 Firmware Version 1 0 0 25 Settings gt gt Firewall gt gt Dynamic Defense 3 Dynamic Defense Dynamic Defense Dynamic Defense Enable w Periodic Time Interval min 1 Blacklist Update Interval s 300 Connection Threshold 20 Dynamic Defense Whitelist 192 168 40 5 Figure 17 Configure Dynamic Defense UCM6510 IP PBX User Manual Page 45 of 233 andstream Innovative IP Voice amp Video FAILZBAN Fail2Ban feature on the UCM6510 provides intrusion detection and prevention for authentication errors in SIP REGISTER INVITE and SUBSCRIBE Once the entry is detected within Max Retry Duration the UCM6510 will take action to forbid the host for certain period as defined in Banned Duration This feature helps prevent SIP brute force attacks to the PBX system Table 13 Fail2Ban Settings Global Settings Enable Fail2Ban The default setting is disabled Please make sure both Enable Enable Fail2Ban Fail2Ban and Asterisk Service are turned on in order to use Fail2Ban for SIP authentication on the UCM6510 Configure the duration in seconds for the detected host to be banned The Banned Duration SR default setting is 300 If set to 1 the host will be always banned l Within this duration in se
171. n you know the far end should always be a slave to you e Slave The equipment at the far end of the E1 T1 J1 link is the preferred source of the master clock The line build out LBO is the distance between the operators and the PBX Please use the default value 0dB unless the distance is long Configure the RX gain for the receiving channel of digital port The valid range is from 24dB to 12dB Configure the TX Gain for the transmitting channel of digital port The valid range is 24dB to 12dB Select alaw or ulaw If set to default alaw will be used for E1 This configured whether to play the ringback tone from local UCM6510 or UCM6510 IP PBX User Manual Page 94 of 233 Andstream Innovative IP Voice amp Video not If enabled the local UCM6510 will play ringback tone to the caller Otherwise the caller will listen to the tone from peer device The default setting is disabled Advanced Settings EE Switch Type Coding CRC PRI Dial Plan PRI Local Dial Plan International Prefix National Prefix Local Prefix Private Prefix Unknown Prefix PRI Indication Reset Interval PRI Exclusive Firmware Version 1 0 0 25 Select switch type e EurolSDN EurolSDN common in Europe e Nl2 National ISDN type 2 common in the US e DMS100 Nortel DMS100 e 4ESS AT amp T 4ESS e 5ESS Lucent 5ESS e NI old national ISDN type 1 e QSIG Select HDB3 or AMI Select whether to use CRC4 or not This setti
172. nal Options gt Music On Hold Configure whether the callers will be disconnected from the queue or not 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 e 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 e 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 The default setting is 15 seconds 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 max
173. nd rule s privilege in order to make outbound calls using this rule If the outbound rule privilege is disabled this option will not take effect 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 A random secure password will be automatically generated It is SIP IAX Password recommended to use this password for security purpose 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 Voicemail Password A random password in digits will be automatically generated It is recommended to use this password for security purpose 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 forwarded to voicemail voicemail is enabled or hang up voicemail is disabled If not specified the default ring timeout is 60 seconds on the Ring Timeout UCM6510 which can be configured in the global ring timeout setting under web GUI gt Internal Options General Preference The valid range is between 5 seconds and 600 seconds Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 75 of 233 Andstream Innovative IP Voice amp Video Note If the end point also has a ring timeout configured the actual ring timeou
174. nebook with Phonebook DN ou people dc pbx dc com only 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 UCM6510 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 47 of 233 andstream Innovative IP Voice amp Video 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 Confirm Root Password Cancel Save Figure 18 LDAP Server Configurations The UCM6510 LDAP server supports anonymous access read only by default Therefore the LDAP client doesn t have to configure username and password to access the phonebook directory The Root DN and Root Password here are for LDAP management and configuration where users will need provide for authentication purpose before modifying the LDAP information The default phonebook list in this LDAP server can be viewed and edited by clicking on for the first phonebook under LDAP Phonebook Figure 19 Default LDAP Phonebook DN Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 48 of 233 andstream Innovative IP Voice amp Video Edit Phonebook ou pbx dc pbx dc com LDAP Attributes Contact List AccountNumber 5000 CallerlbName John Doe Stacy Green Email TEEN FirstName Ricky Chan LasiName Front Desk Warehouse Department Sales MobileN
175. nfiguration Parameters E1 MFC R2 Basic Settings All E1 T1 J1 spans generate a clock signal on their transmit side The parameter determines whether the clock signal from the far end of the E1 T1 J1 is used as the master source of clock timing If the far end is used as the master the PBX system clock will synchronize to it Clock l E ea e Master The port will never be used as a source of timing This is appropriate when you know the far end should always be a slave to you e Slave The equipment at the far end of the E1 T1 link is the preferred source of the master clock MFC R2 multinational adaption UCM6510 supports MFC R2 standards varian by ITU and MFC R2 standards in different countries or regions including Argentina Brazil China Czech Republic Colombia Ecuador Indonesia Mexico the Philippines and Venezuela If enabled the callee side will request the caller to send caller number Get ANI First first and then called number Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 97 of 233 dstream Innovative IP Voice amp Video Note Options Get ANI First and Skip Category cannot be enabled at the same time Select the category of the caller UCM6510 supports four categories Category National Subscriber National Priority Subscriber International Subscriber and International Priority Subscriber The line build out LBO is the distance between the operators and the SEN PBX Please use the default va
176. nformation The default syslog level for all modules is error which is recommended in your UCM6510 settings because it can be helpful to locate the issues when errors happen Some typical modules for UCM6510 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 FXS app meetme This module is related to Conference Bridge A Note Syslog is usually for debugging and troubleshooting purpose Turning on all levels for all syslog modules is not recommended for daily usage Too many syslog print might cause traffic and affect system performance Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 227 of 233 Andstream Innovative IP Voice amp Video TROUBLESHOOTING On the UCM6510 users could capture traces ping remote host and traceroute remote host for troubleshooting purpose under web GUI gt Maintenance gt Troubleshooting ETHERNET CAPTURE The captured trace can be downloaded for analysis Also the instructions or result will be displayed in the web GUI output result Ethernet Capture G Interface Type LAN Capture Filter host 192 168 40 178 k Start e Download Output Result capture Dignostic run Package capturing Done Click on Download to download the captured packages Figure 11
177. ng is used to specify the type of the callee number The service provider will usually verify this The default setting is unknown In some very unusual circumstances you may need set to Dynamic or Redundant Note When one type is selected you might not be able to dial another class of numbers For example if National is configured you won t be able to dial local or international numbers This setting is used to specify the type of the caller number The service provider will usually verify this Configure the prefix in PRI Local Dial Plan and PRI Dial Plan for each type Select the PRI Indication e outofoband Use RELEASE DISCONNECT or other messages with CAUSE to indicate call progress e g cause unassigned number or user busy e inband use in band tones to play busy or congestion signal to the other side This is the default setting The interval that restarts idle channels This setting is used to set up the ChannellD in SETUP message If enabled only the specified B channel can be used Otherwise select one UCM6510 IP PBX User Manual Page 95 of 233 Andstream Innovative IP Voice amp Video of the channels in B channel If you need override the existing channels selection routine and force all PRI channels to be marked as exclusively selected please enable it If selected transmission of facility based ISDN supplementary services Facility Enable ae l such as caller name from CPE over facility
178. nnrtreornnrrrrrnnrrennanrrernnne 64 glede gt DOME e Le 81 FOU TETEN nd 81 Figure 35 Email To User Prompt Information n00nnnnnnnannnnnnnnnnnnnsnnnnnnnrnnnnnonrnrnnnsrnneressnnrrrrnenrerennnnrrenennne 82 Figure 36 Email To User Account Registration Information and QR Code eennrrrnnnnrrnnvvnnrrnnnnnrnnnnnnnrnnnnnn 83 Figure 37 Email To User LDAP Client Information and QR Code 83 Figure 38 UC M6510 FXO TONE SENGS EE 88 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 9 of 233 Figure 39 Figure 40 Figure 41 Figure 42 Figure 43 Figure 44 Figure 45 Figure 46 Figure 47 Figure 48 Figure 49 Figure 50 Figure 51 Figure 52 Figure 53 Figure 54 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 Figure 70 Figure 71 Figure 72 Figure 73 Figure 74 Figure 75 Figure 76 Figure 77 Figure 78 Figure 79 Firmware Version 1 0 0 25 E ten Innovative IP Voice amp Video UCM6510 PSTN Detection cccccccceeeececcceeeeeeeeseeeeceeeeesaeeseeeeeeseseseeeseeeeeeeesseeeaeeeeeeseesaaeaeses 88 UCM6510 PSTN Detection Auto Deier 89 UCM6510 PSTN Detection Semi Auto Detect rrrrrrnrnnnnvvvnnnnvvrrrrnnnnnnnnnnnrrnrrrerennnnnrrrrnnnnnnnnnn 89 FXS Ports Signaling GE e 91 ele Ode EE 91 Digital Hardware Confouraton 93 Troubleshooting Digital Trunks
179. no HTTP TFTP server users could also upload the firmware to the UCM6510 directly via web GUI Please follow the steps below to upload firmware locally 1 Download the latest UCM6510 firmware file from the following link and save it in your PC http www grandstream com support firmware 2 Log in the web GUI as administrator in the PC c 3 Go to web GUI gt Maintenance gt Upgrade upload the firmware file by clicking on and select the firmware file from your PC The default firmware file name is ucm6510fw bin Local Upgrade LO Firmware File Path uUcm5500fw bin ons Upgrade Figure 108 Local Upgrade Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 220 of 233 fream Innovative IP Voice amp Video 4 Click on O to start upgrading Loading 3 Upgrading Firmware files Figure 109 Upgrading Firmware Files 5 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 Cancel Figure 110 Reboot UCM6510 6 Click on OK to reboot the UCM6510 and check the firmware version after it boots up Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 221 of 233 E ten Innovative IP Voice amp Video A Note Please do not interrupt or power cycle the UCM6510 during upgrading process NO LOCAL FIRMWARE SERVERS F
180. nsions Click on Create New User gt Create New IAX Extension and a new dialog window will show for users to fill in the extension information The configuration parameters are as follows Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 68 of 233 ndstream Innovative IP Voice amp Video Table 19 IAX Extension Configuration Parameters Extension The extension number associated with the user Configure the CallerID Number that would be applied for outgoing calls from this user CallerID 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 an outbound rule s privilege in order to make outbound calls using this rule If the outbound rule privilege is disabled this option will not take effect Configure the password for the user A random secure password will be automatically generated when the extension is created It is SIP IAX Password recommended to use this password or other strong password for security purpose Enable Voicemail Enable voicemail for the user The default setting is Yes Configure voicemail password digits only for the user to access the l l voicemail box A random numer
181. nts Lists Disk Usage 204 System Events gt Alert Events Lists Modify Admin Password 205 System Events gt Alert Events Lists Memory Usage ssnsssnnssennnenseesernnrrrreesrrrnrtrrressnrnnrennee 205 System Events gt Alert Events Lists System Reboot rrrrrrrrrrrrrnrrnnnnnnnovrrnnrrrnnnnnnnrrnnnrrnnnnnnnnee 205 System Events gt Alert Events Lists System Update nnnnnennneennneneeennnnnnnrreenrrrnrrrrrensnrnnreneee 206 System Events gt Alert Events Lists System Crash 206 System Events gt Alert Lon 207 CDER Ir 208 BAS A COU E 209 Call Report Entry with Audio Recording File 209 Downloaded CDR File Sample Call To Shows ei 210 Downloaded CDR File Sample Source Channel and Dest Channel 210 Downloaded CDR File Sample Source Channel and Dest Channel 2 211 Downloaded CDR File Sample Source Channel and Dest Channel 3 211 REG 212 Ne NN 219 Mole Edo elg TEE 220 Upgrading Firmware Eiles 221 Reboot UCM6510 eens 221 Foue TITO PN EE 223 Figure 112 Figure 113 Figure 114 Figure 115 Figure 116 Figure 117 Figure 118 Figure 119 DE Ge EEE EE EE EEE 224 Restore UCM6510 from Backup File rrrrrnrrnrnnnnnrorrrnnnnnrrvnrnnnnnrnnnnnnnnnrnnnnnnnnnsnnnnnnnnssnnnnnnsnsenn 225 SACI AMC EEE EE 226 ue Fie 00 EE 227 ETEN 228 PN 229 GL EEE A E A E ES 229 Troubleshooting Analog Trunks ccccccecccseeeeeeeeeeeeseeeeeeeeesceeeeseeeeceeessaaeeeeeesseaeeeeeessaneeeeseeas 231 Firmware Version 1
182. nvelope Allow User Review Gran Innovative IP Voice amp Video announced at the beginning of the voicemail 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 ACCESS VOICEMAIL If the voicemail is enabled for UCM6510 extensions the users can dial the voicemail access feature code oy default 98 or 97 to access the extension s voicemail The users will be prompt to enter the voicemail password and then can enter digits from the phone keypad to navigate in the IVR menu for different options Main Menu 1 New messages 2 Change folders Firmware Version 1 0 0 25 Table 41 Voicemail IVR Menu Sub Menu 1 3 Advanced options 5 Repeat the current message 7 Delete this message 8 Forward the message to another user 9 Save Help Exit 0 New messages 1 Old messages 2 Work messages 3 Family messages 4 Friend messages Sub Menu 2 1 Send a reply 2 Call the person who sent this message 3 Hear the message envelop 4 Leave a message Return to the ma
183. o caller ID or disconnecting issue please make sure to run the ACIM detection to find out the correct value for impedance setting Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 91 of 233 Tone Region dstream Innovative IP Voice amp Video Table 25 Analog Hardware Configuration Parameters Select country to set the default tones for dial tone busy tone ring tone and etc to be sent from the FXS port The default setting is United States of America USA Advanced Settings FXO Opermode FXS Opermode FXS TISS Override PCMA Override Boost Ringer Fast Ringer Low Power Ring Detect FXS MWI Mode Firmware Version 1 0 0 25 Select country to set the On Hook Speed Ringer Impedance Ringer Threshold Current Limiting TIP RING voltage adjustment Minimum Operational Loop Current and AC Impedance as predefined for your country s analog line characteristics The default setting is United States of America USA Select country to set the On Hook Speed Ringer Impedance Ringer Threshold Current Limiting TIP RING voltage adjustment Minimum Operational Loop Current and AC Impedance as predefined for your country s analog line characteristics The default setting is United States of America USA Configure to enable or disable override Two Wire Impedance Synthesis TISS The default setting is No If enabled users can select the impedance value for Two Wire Impedance Synthesis TISS overri
184. o send the account registration and configuration information to the user Please make sure Email setting under web Ul gt Settings gt Email Settings is properly configured and tested on the UCM6510 before using Email To User When click on Email To User button the following message will be prompted in the web page Click on OK to confirm sending the account information to all users Email addresses Prompt information Are you sure you want to send the account infomations to the user s email Figure 35 Email To User Prompt Information The user will receive Email including account registration information and LDAP configuration A QR code is also generated for Mobile applications to scan it and get automatically provisioned QR code provisioning is supported on Grandstream Softphone GS Wave Android application Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 82 of 233 E ten Innovative IP Voice amp Video Account Name 1001 SIP Server 192 168 2 1 SIP User ID 1001 Authenticate ID 1001 Authenticate Password t 297eoS1h Name This is the QR code of this account Figure 36 Email To User Account Registration Information and QR Code Server Address 192 168 2 1 Port 389 Base dc pbx dc com Tis is the QR code of this LDAP config EN A AA JE i a J Figure 37 Email To User LDAP Client Information and QR Code Firmware Version 1 0 0 25 UCM6510 IP PBX User
185. odes or run into DISA STATISTICS CDR Statistics is an additional feature on the UCM6510 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 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 211 of 233 E ten Innovative IP Voice amp Video Figure 106 CDR Statistics Table 73 CDR Statistics Filter Criteria Trunk Type Select one of the following trunk type e All e SIP Calls e PSTN Calls e ISDN Calls Call Type Select one or more in the following checkboxes e Inbound calls e Outbound calls e Internal calls e External calls e Allcalls 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 e By range For example 2013 01 To 2013 03 RECORDING FILES The recording files recorded by Auto Record per extension per trunk or via feature code Audio Mix Record are listed here Users could click on to play the recording file click on to download the recording file in wav format or click on to delete the recording file To sort the recording file click on the title Caller Callee or Call Time for the corresponding column Click on the title again can switch the sorting mode between ascending order or descending order Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 212 of 233 E
186. on Create New Inbound Rule Trunks AnalogTrunks PSTN1 DID Pattern e G Default Destination Time Condition Click to add Time Condition Cancel Save Figure 75 Inbound Route to Fax Extension 5 Once successfully configured the incoming Fax from external Fax machine to the PSTN line number will be converted to PDF file and sent to the Email address Faxtest ucm6510mycompany com as attachment Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 162 of 233 The UCM6510 supports DISA to web GUI gt Call Features gt DISA ndstream Innovative IP Voice amp Video DISA be used in IVR or inbound route Before using it create new DISA under e Click on Create New DISA to add a new DISA e Click on to edit the DISA configuration e Click on l to delete the DISA Create New DISA Name i Password Permission i Response Timeout Digit Timeout fi Allow Hangup Name Password Permission Firmware Version 1 0 0 25 Internal 10 5 Cancel Figure 76 Create New DISA Table 48 DISA Settings Configure DISA name to identify the DISA Configure the password digit only required for the user to enter before using DISA to dial out Note The password has to be at least 4 digits Configure the permission level for DISA The available permissions are Internal Local National and International from the lowest level to the highest level The default setting
187. on 1 0 0 25 UCM6510 IP PBX User Manual Page 86 of 233 E ten Innovative IP Voice amp Video 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 Busy Tone 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 Congestion Tone 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 PSTN Detection l l l more than one channel configured and working properly If the detection has busy tone the Tone Country option will be set as Custom PSTN DETECTION The UCM6510 provides PSTN detection function to help users detect the busy tone Polarity Reversal and Current Disconnect by making a call from the PSTN line to another destination The detecting call will be answered and up for about 1 minute Once done the detecting result will show and can be used for the UCM6510 settings 1 G
188. on range to be assigned if Automatically Assign Extension is enabled The default range is 5000 6299 Zero Config Extension Segment range can be defined in web Ul gt PBX gt Internal Options gt General Auto Extensions page gt Extension Preference section Provision If enabled the extension list will be sent out to the device after receiving the device s request This feature is for the GXP series phones that support selecting extension to be provisioned via phone s LCD The default setting is disabled UCM6510 IP PBX User Manual Page 61 of 233 dstream Innovative IP Voice amp Video Click on the link Pick Extension Segment to specify the extension list to be sent to the device The default range is 4000 to 4999 Pick Extension Pick Extension Segment i Segment range can be defined in web Ul gt PBX gt Internal Options gt General page gt Extension Preference section Pick Extensions l Specify the number of minutes to allow the phones being provisioned to Pick Extension Period hour l l pick extensions Please make sure an extension is manually assigned to the phone or Automatically Assign Extension is enabled during provisioning After the configuration on the UCM6510 web GUI click on Save and Apply Changes Once the phone boots up and picks up the config file from the UCM6510 it will take the configuration right away MANUAL PROVISIONING DISCOVERY Users could manually discover the device by s
189. or users that would like to use remote upgrading without a local TFTP server Grandstream offers a NAT friendly HTTP server This enables users to download the latest software upgrades for their devices via this server Please refer to the webpage http www grandstream com support firmware Alternatively users can download a free TFTP or HI TP server and conduct a 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 tftod32 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 UCM6510 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 UCM6510 web configuration interface 5 Configure the Firmware Server Path to the IP address of the PC 6 Update the changes and reboot the UCM6510 End users can also choose to download a free HTTP server from http httod apache org or use Microsoft IIS web server BACKUP The UCM6510 configuration can be backed up locally or via network The backup file will be used to restore the configuration on UCM6510 when necessary F
190. oto UCM6510 web GUI gt PBX gt Basic Call Routes gt Analog Trunks page Click to edit the analog trunk created for the FXO port In the dialog window to edit the analog trunk go to Tone Settings section and click on EED for PSTN Detection Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 87 of 233 andstream Innovative IP Voice amp Video Tone Settings G Busy Detection Yes e CO Busy Tone Count Congestion Detection Yes e i Congestion Count Tone Country United States of America USA Busy Tone f1 480 50 f2 620 50 c 500 500 Congestion Tone M 480 50 f2 620 50 c 2 50 250 PSTN Detection Figure 38 UCM6510 FXO Tone Settings 4 Click on Detect to start PSTN detection Edit Analog Trunk trunk_ 1 Detect model Auto Detect Source Channel to be detected Destination Channel Destination Number Note Detection will keep the call up for about 1 minute If you have selected Semi auto Detect please pick up the phone only after you are informed Figure 39 UCM6510 PSTN Detection e lf there are two FXO ports connected to PSTN lines use the following settings for auto detection Detect Model Auto Detect Source Channel The source channel to be detected Destination Channel The channel to help detecting For example the second FXO port Destination Number The number to be dialed for detecting This number must be the actual PSTN number for the FXO port used as the des
191. p Call Features SE group blacklist paging intercom and etc e FCC Part 15 CFR 47 Class B Part 68 e CE EN55022 Class B EN55024 EN61000 3 2 EN61000 3 3 EN60950 1 TBR21 RoHS e A TICK AS NZS CISPR 22 Class B AS NZS CISPR 24 AS NZS 60950 AS ACIF S002 e ITU T K 21 Basic Level UL 60950 power adapter e 11 TIA 968 B Section 5 2 4 e Ei TBR12 TBR13 E1 AS ACIF Compliance Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 17 of 233 Andstream Innovative IP Voice amp Video This page intentionally left blank Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 18 of 233 clean Innovative IP Voice amp Video INSTALLATION Before deploying and configuring the UCM6510 series 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 UCM6510 series EQUIPMENT PACKAGING Table 2 UCM6510 Equipment Packaging Main Case Yes x 1 Power Adapter Yes x 2 Ethernet Cable Yes x 1 Wall Mount Yes x 2 Screws Yes x 6 Quick Installation Guide Yes x 1 CONNECT YOUR UCM6510 CONNECT THE UCM6510 USB Port T1 E1 J1 Port Navigation Keys um en ren Po DOC Ka gt SD Card Slot 2xFXS Port 2 x FXO Port LED Indicators LCD Figure 1 UCM6510 Front View Ground Gam eT Le KI ll ee 2x DC 12V Power Jack Reset LAN Port WAN Port Heartbeat Port Figur
192. park using feature code When there is call being parked this section will display the parking lot status i S ha nnel SIP 6010 00000050 SIP 6005 00000052 Figure 89 Parking Lot Status Table 69 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 Dr e Click on to refresh the parking lot status e Click on to expand the parking lot details e Click on to hide the parking details SYSTEM STATUS The UCM6510 system status can be accessed via web GUI gt Status gt System Status which displays the following system information e General e Network e Storage Usage e Resource Usage Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 201 of 233 GENERAL andstream Innovative IP Voice amp Video Under web GUI gt Status gt System Status gt General users could check the hardware and software information for the UCM651
193. 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 SD Card will display if connected Storage Usage Configuration Partition Total 96MB Available 54 MB E Used 40 MB USB Disk a Partition 1 Total 1 888MB Available 1 855 MB PS Used 31 MB RESOURCE USAGE Data Partition Total 3 168MB Available 3 071 MB E Used 95 MB SD Card 1 Partition 1 Total 3 808MB Available 3 455 MB MM Used 351 MB Figure 90 System Status gt Storage Usage When configuring and managing the UCM6510 users could access resource usage information to estimate the current usage and allocate the resources accordingly Under web GUI gt Status gt System Status gt Resource Usage the current CPU usage and Memory usage are shown in the pie chart Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 203 of 233 Andstream Innovative IP Voice amp Video Resource Usage CPU Usage Memory Usage Figure 91 System Status gt Resource Usage SYSTEM EVENTS The UCM6510 can monitor important system events log the alerts and send Email notifications to the system administrator ALERT EVENTS LIST The system alert events list can be found under web GUI gt Status gt System Events gt Alert Events List Click on f to configure
194. pecifying the IP address or scanning the entire LAN network Three methods are supported to scan the devices e PING e ARP e SIP Message NOTIFY Click on Auto Discover fill in the Scan Method and Scan IP The IP address segment will be automatically filled in based on the network mask detected on the UCM6510 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 To successfully discover the devices Zero Config needs to be enabled on the UCM6510 web GUI gt PBX gt Basic Call Routes gt Zero Config gt Auto Provisioning Settings Auto Discover i Scan Method Lea Scan IP Figure 29 Auto Discover Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 62 of 233 andstream The following figure shows a list of discovered phones The MAC address IP Address Extension if assigned Version Vendor Model Connection Status Create Config Options Edit Delete Update are displayed in the list Grandstream GXP2110 Connected 000B822B2D94 192 168 40 143 000B82273BE4 Grandstream GXP2110 Connected 000B822B1B29 Grandstream GXP2100 Connected 000B822A852C 192 168 40 3 Grandstream GXP2100 Connected 000B82382BE7 Grandstream GXP1450 Connected 000B82382FFB Grandstream GXP1450 Connected 000B8233A045 Grandstream GXP1450 Connected Figure 30 Discovered D
195. port number e Very Allow peers matching by IP address without matching port number Also authentication of incoming INVITE messages is not Insecure required e 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 Enable Keep alive ee keep the NAT port open The default setting is Yes K ive F Configure the number of seconds for the host to be up for Keep alive The eep alive Frequenc 5 j d default setting is 60 seconds Other Settings Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 76 of 233 clean Innovative IP Voice amp Video SRTP 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 Fax Detection l 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 e Allow All Device in any network can register this extension e Local Subnet Only Strategy Only the user in specific subnet can register this extension Up to three subnet addresses can be specified e A Specific IP Address Onl
196. ptions gt Feature Codes Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 123 of 233 Andstream Innovative IP Voice amp Video This page intentionally left blank Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 124 of 233 ndstream Innovative IP Voice amp Video CONFERENCE BRIDGE The UCM6510 supports Conference Bridge allowing 64 participants with up to 8 bridges at the same time 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 e 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 Extension Password Admin Password Firmware Version 1 0 0 25 Table 37 Conference Bridge Configuration Parameters Configure the conference number for the users to dial into the conference When configured the users who would like to join the conference call must enter this password before accessing the conference bridge Note e f Public Mode is enabled the password is not required to join the conference bridge thus this field is invalid e The password has to be at least 4 c
197. qt tesesatatduanetatecntenstes 58 PROVISIONING givanannguennnsaivenaxanspsicsungiunvansssbaiecaednnseananeenesmncanasexanauniientennnnins 59 OY ERVE N hj 59 AUTO RRONIGSIONING erect eee terre eee aaa eee etter rere tees eee e eee e aa aaa ee aceeeeeeeeeeeeeeeeseeaaaaaaaaneeeenees 59 MANUALE PROVISIONING ipscstccaseancenanseacconinncbenssaaucnesaanvesanctoteneinseseadiiaiesaacaaneansssonteentese eeii Enne 62 GE 62 SONEN 63 E ANNA 64 FV SN 64 EATEN STON LE 65 CREATE NEW UGER ee 65 CREATE NEW SIP ESTENGION 65 CREATE NEW NN eegen 68 CREATE NEW PXS EXTENSION vrsscdvcaxecinnsasasciuaasbaadecaBacadesiteegeusataeciusasbaadeeeBetadenteageumayeecieeatesaeds 71 BATCH ADD EXTENSIONS AAA 75 BATCH ADD SIP EXTENSIONS rrrrrrrnnnnnnnnnnorrnnrrrnvvvnrrrrrnrrnnnnnnnnrrrrrntnnnnnnennnrrnrrrnnnnnnnnrrrrnnnnnneneen 75 HE RTR E ER ET 77 EN EATEN ON 80 EXPORT EXTENSIONS rrrnnrrrrrrrorrvvvvenrnrrrrrrrnnnnnnnnrernrneveeeenennrrrrrrnnnnnnnnsennresenenenennnrrrrnnnnnnnnneennneenee 81 IMPORT eege 81 EAE VO USE P EE 82 POA ERIN vvs 85 ANALOG TRUNKS CONEIGUDATION 0 cccceeeeeeeeeeeee cece cece teen eee ee aaa aa ee aaaeeeeeeeeeeeeeeeeeeseaaaaaaaaneeeeeees 85 Fe Ne NE 87 ANALOG HARDWARE CONFIGURATION uvvvvvvvvvrrrrrnnnnnnnrrrrrrrrveveennerrrrrrnnnnnnnnnnrrnnnererereenrnrrrrnnnnnnnnn 91 DIGITALE TRINN sr rn 93 DIGITAL HARDWARE CONFIGURATION ccecesrecaesisanieioine ornar e EE ETEei 93 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 2 of 233 dstream
198. r the VoIP trunk The available codecs are PCMU PCMA GSM AAL2 G 726 32 G 726 G 722 G 729 G 723 ILBC ADPCM H 264 H 263 and H 263p In the selected codec list users can click on UP or DOWN arrow to adjust the order for the codec priority Configure the actual domain name where the extension comes from This can be used to override the From Header For example trunk UCM6510 provider com is the From Domain in From Header sip 1234567 Qtrunk UCM6510 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 trunk UCM6510 provider com UCM6510 IP PBX User Manual Page 110 of 233 Outbound Proxy Support Outbound Proxy Auto Record DID Mode Enable Qualify Qualify Timeout Qualify Frequency Fax Detection SRTP Andstream 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 Enable automatic recording for the calls using this trunk The default setting is disabled The recording files can be accessed under web GUI gt CDR gt Recording Files Configure where to get the destination ID of an incoming SIP call from SIP Request line or To header The default is set to Request line If
199. re Extension For Fax Machine snnnnnnnnnneseeenessssrrrrnnnrtrtrrrreeeesssssrnnnnnrttrtrrreeeeesssenen 161 Configure Inbound Rule for Fan 161 GC gl 0 EEE EEE E 162 Inbound Route to Fax Extension EE 162 Create New DEAL 163 Create New Event List EEE 166 Create Dial By Name Group 169 Dial By Name Group In IVR Key Pressing Events rrrrrrrrnnnrrvvrnnnnvrrvnnnnnnrrrrnnrnnnerrennnnnrrreennnnn 170 UCM6510 IP PBX User Manual Page 10 of 233 Figure 80 Figure 81 Figure 82 Figure 83 Figure 84 Figure 85 Figure 86 Figure 87 Figure 88 Figure 89 Figure 90 Figure 91 Figure 92 Figure 93 Figure 94 Figure 95 Figure 96 Figure 97 Figure 98 Figure 99 Figure 100 Figure 101 Figure 102 Figure 103 Figure 104 Figure 105 Figure 106 Figure 107 Figure 108 Figure 109 Figure 110 E ten Innovative IP Voice amp Video Dial By Name Group In Inbound Route rrrnrrrrrrrnnnnnnnovornrrrnnnnnnnsrnnnnrnnnnnnnnsrnnnnrnnnnnnnnssnnnnnnnnnn 170 Configure Extension First Name And Last Name nnaannnnnnannnnnnnnnnnnnnennnnnsnnrnrnnnnnrrensnnrreennnne 171 Download Recording File from CDR Page nnnnnnannnnonennnnnnennnnennnnrnnesenrrrnsrnnrrensnnrrresenrrrenne 177 JE EE 193 NS 193 FN 195 OTER eebe 196 Conference Room Statu Sissies anie E aiea Eine E E 197 Digital Channels EE 200 Pt 201 System Status gt Storage Usage 203 System Status gt Resource Usage 204 System Events gt Alert Eve
200. ritish English Deutsch English Espafiol EMnvk Francais Italiano Nederlands Polski Portugu s Pycckun Svenska T rk e nay Ay all Firmware Version 1 0 0 25 1 0 1 14 1 0 1 0 1 14 1 0 1 04 1 0 1 0 1 0 LI LI 1 04 1 00 1 04 1 14 Figure 57 Voice Prompt Package List UCM6510 IP PBX User Manual Page 136 of 233 andstream Innovative IP Voice amp Video Click on to download the language to the UCM6510 The installation will be automatically started once the downloading is finished Language Settings Upload Voice Prompt Package Choose Voice Promptto as Upload Upload Voice Prompt Package List Language English Delete Check Prompt List Figure 58 New Voice Prompt Language Added A new language option will be displayed after successfully installed Users then could select it to apply in the UCM6510 system voice prompt or delete it from the UCM6510 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 137 of 233 Andstream Innovative IP Voice amp Video This page intentionally left blank Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 138 of 233 ndstream Innovative IP Voice amp Video VOICEMAIL CONFIGURE VOICEMAIL If the voicemail is enabled for UCM6510 extensions the configurations of the voicemail can be globally set up and managed under web GUI gt PBX gt Call Features gt Voicemai
201. rmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 54 of 233 ndstream Innovative IP Voice amp Video Send atest Email to the Email address below Email Address testucm6510 mycompany pbx com Figure 25 UCM6510 Email Settings Send Test Email TIME SETTINGS AUTO TIME UPDATING The current system time on the UCM6510 is displayed on the upper right of the web page It can also be found under web GUI gt Status gt System Status gt General To configure the UCM6510 to update time automatically go to web GUI gt Settings gt Time Settings gt Auto Time Updating A Note The configurations under Web GUl gt Settings gt Time Settings gt Time Auto Updating page require reboot to take effect Please consider configuring auto time updating related changes when setting up the UCM6510 for the first time to avoid service interrupt after installation and deployment in production Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 55 of 233 Remote NTP Server Enable DHCP Option 2 Enable DHCP Option 42 Time Zone Self Defined Time Zone Firmware Version 1 0 0 25 Gan Innovative IP Voice amp Video Table 16 Auto Time Updating Specify the URL or IP address of the NTP server for the UCM6510 to synchronize the date and time The default NIP server is ntp ipvideotalk com If set to Yes the UCM6510 is allowed to get provisioned for Time Zone from DHCP Option 2 in the local server automatically The def
202. rnal 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 STUN Server l 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 STUN Refresh ak setting is 30 seconds Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 182 of 233 clean Innovative IP Voice amp Video AX SETTINGS The UCM6510 IAX global settings can be accessed via web GUI gt PBX gt IAX Settings IAX SETTINGS GENERAL Table 55 IAX Settings General Bind Port Configure the port number that the IAX2 will be allowed to listen to The default setting is 4569 GE SE the SE that the IAX2 will be forced to bind to The default setting is 0 0 0 0 which means all addresses IAX1 Compatibility Select to configure IAX1 compatibility The default setting is No If selected UDP checksums will be disabled and no checksums will be No Checksums 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 Delay Reject er default setting is No ADSI Select to enable ADSI phone compatibility The default setting is No Specify which Music On Hold class this channel would like to listen to l when being put on hold This music class is only effect
203. rovisioning are used Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 59 of 233 andstream Innovative IP Voice amp Video UCM6510 SIP End Device Boot up Discover Device T Assign Extension to Device Create XML Config File Download Config File Send Downloading i URL to Device Reboot Get Provisioned Figure 27 UCM6510 Zero Config e SIP SUBSCRIBE When the phone boots up it sends out SUBSCRIBE to a multicast IP address in the LAN The UCM6510 discovers it 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 UCM6510 and reboot again to take the new configuration e DHCP OPTION 66 This method should be used only when the UCM6510 is set to Route mode under web GUI gt Settings gt Network Settings gt Basic Settings Method When the phone restarts by default DHCP Option 66 is turned on it will send out a DHCP DISCOVER request The UCM6510 receives it and returns DHCP OFFER with the config server path URL in the Option 66 for example https 192 168 2 1 8089 zccgi The phone will then use the path to download the config file generated in the UCM6510 e mDNS When the phone boots up it sends out MDNS query to get the TFTP server address The UCM6510 will respond with its own address The phone will then send TFTP request to download the XML config file from the UCM6510 Firmware Version 1 0 0
204. rs and the PBX Please use the default value 0dB unless the distance is long Configure the RX gain for the receiving channel of digital port The valid range is from 24dB to 12dB Configure the TX Gain for the transmitting channel of digital port The valid range is 24dB to 12dB Select alaw or ulaw If set to default ulaw will be used for T1 J1 This configured whether to play the ringback tone from local UCM6510 or not If enabled the local UCM6510 will play ringback tone to the caller Otherwise the caller will listen to the tone from peer device The default setting is disabled Advanced Settings EE Switch Type Firmware Version 1 0 0 25 Select switch type e EurolSDN EurolSDN common in Europe e NI2 National ISDN type 2 common in the US e DMS100 Nortel DMS100 e 4ESS AT amp T 4ESS e 5ESS Lucent 5ESS e NI old national ISDN type 1 e QSIG UCM6510 IP PBX User Manual Page 99 of 233 Coding PRI Dial Plan PRI Local Dial Plan International Prefix National Prefix Local Prefix Private Prefix Unknown Prefix PRI Indication Reset Interval PRI Exclusive Facility Enable NSF Basic Settings Andstream Innovative IP Voice amp Video Select B8ZS or AMI This setting is used to specify the type of the callee number The service provider will usually verify this The default setting is unknown In some very unusual circumstances you may need set to Dynamic or Redundant
205. rty 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 If 1 is entered to invite other party no permission will be required from the invited party A Note Conference 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 UCM6510 web GUI during the conference call the participants in each conference bridge will be listed 1 Click on 9 to kick a participant from the conference 2 Click on to mute the participant 3 Click on M to lock this conference bridge so that other users cannot join it anymore 4 Click on 9 to invite other users into the conference bridge e Manage the conference call 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 0 25 UCM6510 IP PBX User Manual Page 128 of 233 andstream Innovative IP Voice amp Video Table 38 Conference Caller IVR Menu 1 Mute unmute yourself Lock unlock the conference bridge Kick the last
206. rying number of channels depending on the standards in the country of implementation E1 T1 or J1 MFC R2 is a signaling protocol heavily used over E1 trunks SS7 uses out of band signaling which travels on a separate dedicated channel rather than within the same channel as the telephone call providing more efficiency and higher security level when the telephone calls are set up To set up digital trunk on the UCM6510 1 Go to web Ul gt PBX gt Ports Config gt Digital Hardware to configure port type and channels 2 Goto web Ul gt PBX gt Basic Call Routes gt Digital Trunks to add and edit digit trunk 3 Go to web Ul gt PBX gt Basic Call Routes gt Outbound Routes and Inbound Routes to configure outbound and inbound rule for the digital trunk DIGITAL HARDWARE CONFIGURATION Go to web GUI gt PBX gt Ports Config gt Digital Hardware page and configure the following PBX gt gt Ports Config gt gt Digital Hardware 8 Digital Hardware Type T1 Group Name DefaultGroup1 Figure 44 Digital Hardware Configuration e Step 1 Click on to edit digital ports Please see configuration parameters in the tables below e Step 2 Click on to edit group This assigns channels to be used for the digital port For E1 30 B channels can be assigned to the default group for T1 J1 23 B channels can be assigned to the default group e Step 3 If fewer than 30 B channels for E1 or 23 B channels for T1 J1 are assigned in defaul
207. s SRTP Fax Detection Strategy Skip Trunk Auth Codec Preference CREATE NEW IAX EXTENSION 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 send the received Fax to the default Email address in Fax setting page under UCM6510 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 e 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 three subnet addresses can be specified e A 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 outbound rule to make outbound calls The default setting is No Select audio and video codec for the extension The available codecs are PCMU PCMA GSM AAL2 G 726 32 G 726 G 722 G 729 G 723 ILBC ADPCM H 264 H 263 and H 263p In the selected codec list users can click on UP or DOWN arrow to adjust the order for the codec priority To manually create new IAX user go to web GUI gt PBX gt Basic Call Routes gt Exte
208. s Conference Rooms Interfaces Digital Channels and Parking lot It presents administrators the real time status in different sections under web GUI gt Status gt PBX Status Trunks G Status Unmonitored PeerSiPTrunk Total 1 Show 1 4 Goto Extensions G Status e 1000 e 1001 1002 1003 1004 1005 1006 Total 7 Show 4 4 Goto ey Queues gt TRUNKS John Doe William Cheung Sandy Fang Steve Mitchell Nancy Lin Ted Smith Joseph White Messages 0 0 0 Messages 0 0 0 Messages 0 0 0 Messages 0 0 0 Messages 0 0 0 Messages 0 0 0 Messages 0 0 0 First P Conference Rooms ZC Notin Use Interfaces Status Heartbeat Power 2 Parking Lot G No Parking Lot defined Figure 83 Status gt PBX Status Users could see all the configured trunk status in this section Trunks Unmonitored Unavailable Firmware Version 1 0 0 25 Grandstream Trunk1 192 16840 140 Ports 1 Figure 84 Trunk Status Table 64 Trunk Status UCM6510 IP PBX User Manual Page 193 of 233 Status Trunks Type Username Port Hostname IP E ten Innovative IP Voice amp Video Display trunk status e Analog trunk Digital trunk status Available Busy Unavailable Unknown Error Error Configured Incorrect signaling configuration between the two devices For example both of the devices are configured as CPE or NET e SIP Peer trunk status Unreachable The hostname
209. s button 2 Fill up the extension information you would like in the exported csv template 3 Click on Import Extensions button The following dialog will be prompted Import Extensions Import Options On Duplicate Extension Skip sd F Delete and Recreate Extension File Update Information Figure 34 Export Extensions 4 Select the option in On Duplicate Extension to define how the duplicate extension s in the imported csv file should be treated by the PBX Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 81 of 233 ndstream e Skip Duplicate extensions in the csv file will be skipped The PBX will keep the current extension information as previously configured without change e Delete and Recreate The current extension previously configured will be deleted and the duplicate extension in the csv file will be loaded to the PBX e Update Information The current extension previously configured in the PBX will be kept However if the duplicate extension in the csv file has different configuration for any options it will override the configuration for those options in the extension 5 Click on to select csv file from local directory in the PC for uploading 6 Click on Save to import the csv file 7 Click on Apply Changes to apply the imported file on the UCM6510 EMAIL TO USER Once the extensions are created with Email address the PBX administrator can click on button Email To User t
210. s 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 to the voicemail you will be prompted with the Voice Mail Main menu 6 You are successfully connected to the PBX system now Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 29 of 233 Andstream Innovative IP Voice amp Video This page intentionally left blank Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 30 of 233 san Innovative IP Voice amp Video SYSTEM SETTINGS This section explains configurations for system wide parameters on the UCM6510 Those parameters include Network Settings Firewall Change Password LDAP server HTTP server Email settings Time Settings and NTP Server settings NETWORK SETTINGS After successfully connecting the UCM6510 to the network for the first time users could log in the web GUI and go to Settings gt Network Settings to configure the network parameters for the device Select each tab in web GUI gt Settings gt Network Settings page to configure LAN WAN settings 802 1X and Port Forwarding A Note To connect the UCM6510 to network T1 E1 J1 data trunk can also be used instead of using the WAN LAN port Please see section DATA TRUNK to use UCM6510 data trunk to connect the device to Internet BASIC SETTINGS Please refer to the following tables for basic network configuration parameters on t
211. 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 accessed via web GUI gt PBX gt Call Features gt Call Queue e Click on Create New Queue to add call queue Name Strategy TechSupport1 Linear Warehouse Ringall Sales Ringall TechSupport2 Least Recent Figure 65 Call Queue e Click on to edit the call queue The call queue configuration parameters are listed in the table below Table 46 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 Sr 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 0 25 UCM6510 IP PBX User Manual Page 149 of 233 Music On Hold Leave When Empty Dial in Empty Queue Dynamic Login Password Ring Time Out Wrapup Time Max Queue Length Firmware Version 1 0 0 25 san 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 GUI gt PBX gt Inter
212. ser A and User B Configure the number of rings before sending CID The default setting is Other Settings Fax Detection Skip Trunk Auth Firmware Version 1 0 0 25 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 If enabled users will not need enter the PIN Set required by the outbound rule to make outbound calls The default setting is No UCM6510 IP PBX User Manual Page 74 of 233 Andstream Innovative IP Voice amp Video BATCH ADD EXTENSIONS BATCH ADD SIP EXTENSIONS Under web GUI gt PBX gt Basic Call Routes gt Extensions click on Batch Add Extensions gt Batch Add SIP Extensions Table 21 Batch Add SIP Extension Parameters Configure the starting extension number of the batch of extensions to be added Create Number Specify the number of extensions to be added The default setting is 5 Start Extension 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 an outbou
213. sion 1 0 0 25 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 UCM6510 IP PBX User Manual Page 175 of 233 san Innovative IP Voice amp Video Default Code Call Pickup on Ringing To pick up a call for any extension xxxx enter the code followed by the extension number xxxx Default Code 8 This code is for the pickup group which can be assigned for l l each extension on the extension configuration page Pickup Extension lf there is an incoming call to an extension the other extensions within the same pickup group can dial 8 directly to pick up the call Default Code This code is for the user to directly dial or transfer to an extension s voicemail Direct Dial Voicemail Prefix
214. ssed under web GUI gt CDR gt Recording Files l l l When user dials voicemail code the password verification IVR is skipped Skip Voicemail Password l l er If enabled this would allow one button voicemail access By default this Verification Osten option is disabled Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 78 of 233 Max Number of Calls Require Call Token Andstream Innovative IP Voice amp Video Configure the maximum number of calls allowed for each remote IP address Configure to enable disable requiring call token If set to Auto it might lock out users who depend on backward compatibility when peer authentication credentials are shared between physical endpoints The default setting is Yes Other Settings SRTP Fax Detection Strategy Skip Trunk Auth Codec Preference Firmware Version 1 0 0 25 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 e Allow All Device in any network can register this extension e Local Su
215. sted as below once available Users could click on to download the recording or click on Il to delete the recording meetme conf rec 6300 1372865271 25 wav 2013 07 03 12 39 38 UTC 03 00 10 67 MB meetme conf rec 6300 1372451238 6 wav 2013 06 28 17 27 46 UTC 03 00 120 04 KB meetme conf rec 6300 1372205127 347 wav 2013 06 25 21 05 56 UTC 03 00 62 86 KB meetme conf rec 6300 1372867161 40 wav 2013 07 03 13 10 29 UTC 03 00 10 17 MB meetme conf rec 6300 1372864546 12 wav 2013 07 03 12 16 01 UTC 03 00 35 67 KB meetme conf rec 6300 1372866438 36 wav 2013 07 03 12 47 47 UTC 03 00 322 86 KB meetme conf rec 6300 1372204987 337 wav 2013 06 25 21 03 30 UTC 03 00 315 98 KB meetme conf rec 6300 1372864583 17 wav 2013 07 03 12 16 36 UTC 03 00 65 67 KB meetme cont rec 6300 13 70385024 71 wav 2013 06 04 19 35 28 UTC 03 00 4 22 MB Figure 52 Conference Recording Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 130 of 233 CONFIGURE IVR san Innovative IP Voice amp Video IVR IVR configurations can be accessed under the UCM6510 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 Digit Timeout Response Timeout Firmware Version 1 0 0 25 Table 39 IVR Configuration Parameters
216. stem voice prompt The following languages are currently supported in system voice prompt English United States Arabic Chinese Dutch English United Kingdom French German Greek Hebrew Italian Polish Portuguese Russian Spanish Swedish Turkish English United States and Chinese voice prompis are built in with the UCM6510 already The other languages provided by Grandstream can be downloaded and installed from the UCM6510 web GUI directly Additionally users could customize their own voice prompts package them and upload to the UCM6510 Language settings for voice prompt can be accessed under web GUI gt PBX gt Internal Options gt Language DOWNLOAD AND INSTALL VOICE PROMPT PACKAGE To download and install voice prompt package in different languages from UCM6510 web GUI click on Check Prompt List button Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 135 of 233 Language Settings Upload Voice Prompt Package T Choose Voice Promptto Upload Voice Prompt Package List CO Language a English ORE andstream Innovative IP Voice amp Video Upload Check Prompt List Figure 56 Language Settings for Voice Prompt 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 B
217. t used is the shortest time set by either device Enable automatic recording for the calls using this extension The default ME S setting is disabled The recording files will be saved in external storage if uto Recor plugged in and can be accessed under web GUI gt CDR gt Recording Files l When user dials voicemail code the password verification IVR is skipped Skip Voicemail Password l l l l oS If enabled this would allow one button voicemail access By default this Verification option is disabled Use NAT when the PBX is on a public IP communicating with devices hidden behind NAT e g broadband router If there is one way audio issue usually it s related to NAT configuration or Firewall s support of SIP NAT and RTP ports The default setting is enabled 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 No Select DTMF mode for the user to send DTMF The default setting is RFC2833 If Info is selected SIP INFO message will be used If DTMF Mode 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
218. t Change Password page 2 Enter the old password first Enter the new password and retype the new password to confirm The new password has to be at least 4 characters The maximum length of the password is 16 characters 4 Click on Save and the user will be automatically logged out 5 Once the web page comes back to the login page again enter the username admin and the new password to login LDAP SERVER The UCM6510 has an embedded LDAP server for users to manage corporate phonebook in a centralized manner e By default the LDAP server has generated the first phonebook with PBX DN ou pbx dc pbx dc com based on the UCM6510 user extensions already e Users could add new phonebook with a different Phonebook DN for other external contacts For example ou people dc pbx dc com e All the phonebooks in the UCM6510 LDAP server have the same Base DN dc pbx dc com If users have the Grandstream phone provisioned by the UCM6510 the LDAP directory has been set up on the phone and can be used right away for users to access all phonebooks Additionally users could manually configure the LDAP client settings to manipulate the built in LDAP server on the UCM6510 If the UCM6510 has multiple LDAP phonebooks created in the LDAP client configuration users could use dc pbx dc com as Base DN to have access to all phonebooks on the UCM6510 LDAP server or use a specific phonebook DN for example ou people dc pbx dc com to access to pho
219. t 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 UCM6510 The default setting is 3600 Configure the minimum duration in seconds of incoming registration and subscription allowed by the UCM6510 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 DIMF 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 RIP activity within the timeout in seconds the call will be terminated The default setting is no timeout UCM6510 IP PBX User Manual Page 191 of 233 RTP Hold Timeout Trust Remote Party ID Send Remote Party ID Generate In Band Ringing Server User Agent Send Compact SIP Headers Add user phone to URI Firmware Version 1 0 0 25 E ten
220. t group users can click on to add more groups This is not necessary in most cases and only default Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 93 of 233 group is needed A Note clean Innovative IP Voice amp Video Currently the group configuration in digit trunks settings is to manage outbound routes only It doesn t control inbound routes Therefore if the users have configured multiple groups for the digital trunk please make sure the inbound routes for those groups have the same inbound rule configured Otherwise inbound call using the digital trunk might not work properly The UCM6510 currently supports E1 T1 and J1 digital hardware type When different signaling is selected for E1 T1 or J1 the settings in basic options and advanced options will be different The following tables list all the settings to configure digital ports when selecting each signaling Table 26 Digital Hardware Configuration Parameters E1 PRI NET PRI CPE Basic Settings Clock LBO RX Gain TX Gain Codec Play Local RBT Firmware Version 1 0 0 25 All E1 T1 J1 spans generate a clock signal on their transmit side The parameter determines whether the clock signal from the far end of the E1 T1 J1 is used as the master source of clock timing If the far end is used as the master the PBX system clock will synchronize to it e Master The port will never be used as a source of timing This is appropriate whe
221. tal Trunk e Click on to delete the digital trunk The digital trunk parameters are listed in the table below Table 31 Digital Trunk Configuration Parameters Trunk Name Configure trunk name to identify the digital trunk Channel Group Configure the digital channel group used by the trunk Hide CallerID Configure to hide outgoing caller ID The default setting is No If enabled the trunk CID will not be overridden by extension s CID when Keep Trunk CID oe 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 CallerID with this option and this option will be ignored When making outgoing calls the following rules are used to determine which CallerID will be used if they exist Caller ID e The CallerID configured for the extension will be looked up first e If Keep Trunk CID is enabled the CallerID configured for the trunk will be used e If the above two are missing the Global Outbound CID defined in web GUI gt PBX gt Internal Options gt General will be used Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 102 of 233 san Innovative IP Voice amp Video Configure the new name of the caller when the extension has no CallerID Name configured CallerID Name Enable automatic recording for the calls using this trunk for SIP trunk only
222. ters for 802 1X on UCM6510 Identity and MD5 password are required for authentication which should be provided by the network administrator obtained from the RADIUS server If EAP TLS or EAP PEAPvO MSCHAPVv2 is used as the 802 1X mode users will also need upload 802 1X CA Certificate and 802 1X Client Certificate which should be also generated from the RADIUS server Table 6 UCM6510 Network Settings gt 802 1X Select 802 1X mode The default setting is Disable The supported 802 1X mode are 802 1X Mode e EAP MD5 e EAP TLS e EAP PEAPv0 MSCHAPv2 Identity Enter 802 1X mode identity information MD5 Password Enter 802 1X mode MD5 password information 802 1X CA 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 Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 37 of 233 STATIC ROUTES E ten Innovative IP Voice amp Video A static route is a pre determined path that the network traffic travels to reach a specific host or network On the UCM6510 the static route function allows the device to use manually configured routes rather than dynamically assigned routes or default gateway configured in the UCM6510 web GUI gt Network Settings gt Basic Settings to forward traffic It can be used to define a route when no other routes are available or necessary or used in complementary with existing routin
223. the GXP2160 IP address under the LAN interface network of the UCM6510 Protocol Type We select TCP here for web UI access using HTTP Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 40 of 233 Port Forwarding Please configure the LAN mode as Router to enable this function WAN Port 8088 WAN Port WAN Port WAN Port WAN Port WAN Port WAN Port WAN Port 192 168 2 100 LAN Port 80 LAN Port LAN Port LAN Port LAN Port LAN Port LAN Port LAN Port Figure 14 UCM6510 Port Forwarding Configuration ndstream Innovative IP Voice amp Video Protocol Type TCP Only Protocol Type UDF Only Protocol Type UDP Only Protocol Type UDF Only Protocol Type UDP Only Protocol Type UDP Only Protocol Type UDP Only Protocol Type UDP Only This will allow users to access the GXP2160 web Ul from public side by typing in address 1 1 1 1 8088 8 page status_network Grandstream GXP2160 Ce d Co Network Status Account Status Status Accounts Settings Network Network Status MAC Address 00 0B 82 59 A9 8D System Info IP Setting IPv4 Address 192 168 2 100 0 0 0 0 0 0 0 0 DHCP IPv6 Address Subnet Mask Gateway 192 168 2 1 DNS Server 1 255 255 255 0 8 8 8 8 DNS Server 2 208 67 222 222 Disabled PPPoE Link Up eames English v Maintenance Phonebook Version 1 0 4 16 Figure 15 GXP2160 Web Access Using UCM6510 Port Forw
224. the parameters for each event 1 Disk Usage Alert Settings Disk Usage OU Detect Cycle minute s e GJ Alert Threshold Figure 92 System Events gt Alert Events Lists Disk Usage e Detect Cycle The UCM6510 will perform the internal disk usage detection based on this cycle Users can enter the number and then select second s minute s hour s day s to configure the cycle Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 204 of 233 Andstream Innovative IP Voice amp Video e Alert Threshold If the detected value exceeds the threshold in percentage the UCM6510 system will send the alert 2 Modify Admin Password Alert Settings Modify Admin Password Detect Cycle minute s e Cancel Save Figure 93 System Events gt Alert Events Lists Modify Admin Password e Detect Cycle The UCM6510 will initiate the admin password check based on this cycle Users can enter the number and then select second s minute s hour s day s to configure the cycle 3 Memory Usage Alert Settings Memory Usage Gi Detect Cycle second s e G Alert Threshold Cancel Save Figure 94 System Events gt Alert Events Lists Memory Usage e Detect Cycle The UCM6510 will perform the memory usage detection based on this cycle Users can enter the number and then select second s minute s hour s day s to configure the cycle e Alert Threshold If the detected value exceeds the threshold in percentage th
225. tination channel Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 88 of 233 Andstream Innovative IP Voice amp Video Edit Analog Trunk trunk_ 1 Detect model Auto Detect Source Channel to be 1 e detected Destination Channel 2 ER Destination Number 1234567 Note Detection will keep the call up for about 1 minute If you have selected Semi auto Detect please pick up the phone only after you are informed Figure 40 UCM6510 PSTN Detection Auto Detect e f there is only one FXO port connected to PSTN line use the following settings for auto detection Edit Analog Trunk trunk 1 CO Detect model Semi auto Detect e G Source Channel to be 1 e detected G Destination Number 1234567 Note Detection will keep the call up for about 1 minute If you have selected Semi auto Detect please pick up the phone only after you are informed Figure 41 UCM6510 PSTN Detection Semi Auto Detect Detect Model Semi auto Detect Source Channel The source channel to be detected Destination Number The number to be dialed for detecting This number could be a cell phone number or other PSTN number that can be reached from the source channel PSTN number 5 Click Detect to start detecting The source channel will initiate a call to the destination number For Auto Detect the call will be automatically answered For Semi auto Detect the UCM6510 web GUI will display prompt to notify the user to
226. tions jitter buffer RTP settings hardware config and STUN monitor The options can be accessed via web GUI gt PBX gt Internal Options INTERNAL OPTIONS GENERAL General Preferences Table 51 Internal Options General Global OutBound CID Global OutBound CID Name Operator Extension Ring Timeout Record Prompt Extension Preferences Configure the global CallerID used for all outbound calls when no other CallerID is defined with higher priority If no CallerlD is defined for extension or trunk the global outbound CID will be used as CallerID Configure the global CallerID Name used for all outbound calls If configured all outbound calls will have the CallerID Name set to this name If not the extension s CallerID 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 Note This is the global value used for each extension if Ring Timeout field is left empty on the extension configuration page If enabled users will hear voice prompt before recording is started or stopped For example before recording the UCM6510 will play voice prompt The call will be recorded The default setting is No Enforce Strong Passwords Firmware Version 1 0 0 25 If enabled
227. to LAN 2 interface Users will need assign LAN 1 or LAN 2 as the default interface in option Default Interface and configure Gateway IP if static IP is used for this interface Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 35 of 233 andstream Innovative IP Voice amp Video Method Dual x Router IP Address 192 168 40 1 Router IP Address 172 18 25 1 172 18 25 x 192 168 40 x LAN 2 LAN Port Je Je N yr A SR LAN 1 WAN Port IP address 172 18 25 10 WR eo ZE Figure 9 UCM6510 Network Interface Method Dual 802 1X IEEE 802 1X is an IEEE standard for port based network access control It provides an authentication mechanism to device before the device is allowed to access Internet or other LAN resources The UCM6510 supports 802 1X as a supplicant client to be authenticated The following diagram and figure show UCM6510 uses 802 1X mode EAP MD5 on WAN port as client in the network to access Internet Authenticator Authentication server Switch WAN port connected to Switch UCM6510 Figure 10 UCM6510 Using 802 1X as Client Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 36 of 233 E ten Innovative IP Voice amp Video Settings gt gt Network Settings gt gt 802 1X O 802 1X Settings Gi 802 1 Mode EAP MD5 Gi Identity 8021xUCM6510 CO MDS Password Figure 11 UCM6510 Using 802 1X EAP MD5 The following table shows the configuration parame
228. to the UCM6510 FXS port Connect Fax machine to the UCM6510 FXS port Connect PSTN line to the UCM6510 FXO port Go to web GUI gt PBX gt Analog Trunks page Create or edit the analog trunk for Fax as below SCHER n Fax Detection Make sure Fax Detection option is set to No Edit Analog Trunk FAX LINE Channels 1 wi J CO Trunk Name FAX_LINE Advanced Options CO Enable Polarity Reversal Current Disconnect ag CO Ring Timeout Thresholdims RX Gain W G TX Gain Use CallerlD wi i Fax Detection Caller ID Scheme Bellcore Telcordia Auto Record Figure 71 Configure Analog Trunk without Fax Detection 5 Goto UCM6510 web GUI gt PBX gt Basic Call Routes gt Extensions page 6 Create or edit the extension for FXS port e Analog Station Select FXS port to be assigned to the extension By default it s set to None e Once selected analog related settings for this extension will show up in Analog Settings section Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 160 of 233 andstream Innovative IP Voice amp Video Create New User sip G Analog Station SIP Settings NAT i Can Reinvite G DTMF Mode RFC2833 e i Insecure G Enable Keep alive Keep alive Frequency G AuthiD Analog Settings Call Waiting Use as SEND RX Gain i TXGain MIN RX Flash 200 i MAX RX Flash G Enable Polarity Reversal Yes e i Echo Cancellation G 3 Way Calling Figure
229. tocol used for the data trunk The UCM6510 supports HDLC HDLC ETH PPP and Cisco Configure the local IP address for the data port This IP address shouldn t conflict with the WAN or LAN side IP of the UCM6510 Configure the subnet mask for the data port Configure the remote IP address for the data port This IP will be the gateway IP address if Default Interface is enabled for the datat trunk Configure DNS server 1 Configure DNS server 2 If enabled this data port will be used as the default interface for Internet connection The Remote IP will be the gateway IP address This has higher priority than the Default Interface assignment LAN 1 or LAN 2 under web Ul gt Settings if Dual is selected as the network method UCM6510 IP PBX User Manual Page 106 of 233 andstream Innovative IP Voice amp Video VOIP TRUNKS VOIP TRUNK CONFIGURATION VoIP trunks can be configured in UCM6510 under web GUI gt PBX gt Basic Call Routes gt VolP Trunks Once created the VoIP trunks will be listed with Provider Name Type Hostname IP Username and Options to edit detect the trunk e Click on Create New SIP Trunk or Create New IAX Trunk to add a new VolP trunk e Clickon to configure detailed parameters for the VoIP trunk Ga e Click on to configure Direct Outward Dialing DOD for the SIP Trunk e Click on i to start LDAP Sync e Click on 1 to delete the VoIP trunk The VoIP trunk options are l
230. ts to strip from the beginning of the DID This is used when By DID is selected in Default Destination Configure to allow the inbound call to dial out from the PBX s trunk or not The default setting is disabled Please be aware of potential security risk if Dial Trunk is enabled The inbound call might be able to dial out international calls from the PBX s trunk if allowed by the privilege level Select the DID destination if By DID is selected in Default Destination Only the selected category can be reached by DID using this inbound route Extension Conference Call Queue Ring Group Paging Intercom Group IVR Voicemail Groups Fax Extension Dial By Name Time Condition Select the start time hour minute for the trunk to use the inbound rule Start Time End Time Date Week Destination Firmware Version 1 0 0 25 Select the end time hour minute for the trunk to use the inbound rule Select By Week or By Day and specify the date for the trunk to use the inbound rule Select the day in the week to use the inbound rule Select the destination for the inbound call under the defined time condition Extension Voicemail Conference Room Call Queue UCM6510 IP PBX User Manual Page 121 of 233 ndstream Innovative IP Voice amp Video e Ring Group e Paging Intercom e Voicemail Group e Fax e DISA e IVR e By DID When By DID is used the UCM6510 will look for the dest
231. ts 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 0 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 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 no authentication will be required when joining the conference call The default setting is Yes If enabled the UCM6510 will play Hold music to the first participant in the conference until another user joins in The default setting is No Select the music on hold class to be played in conference call This option shows up if Play Hold Music For First Caller is enabled Music On Hold class can be set up under web Ul gt PBX gt Internal Options gt Music On Hold UCM6510 IP PBX User Manual Page 126 of 233 Andstream Innovative IP Voice amp Video Skip Authentication When If enabled the invitation from web GUI for a conference bridge with Inviting User via Trunk from password will skip the authenti
232. umber Tech Support HomeNumber Customer Service Fax RMA shipping Test Cancel Figure 20 Default LDAP Phonebook Attributes 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 G Phonebook Prefix G Phonebook DN Figure 21 Add LDAP Phonebook Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 49 of 233 Andstream Innovative IP Voice amp Video Configure the Phonebook Prefix first The Phonebook DN will be automatically filled in For example if configuring Phonebook Prefix as people the Phonebook DN will be filled with ou people dc pbx dc com Once added users can select to edit the phonebook attributes and contact list see figure below or select H to delete the phonebook Edit Phonebook ou people dc pbx dc com LDAP Attributes Contact List AccountName AccountName CallerIDNa
233. unk has 4 DIDs associated to it The main number of the office is routed to an auto attendant The other three numbers are direct lines to specific users of the company At the moment when a user makes an outbound call their caller ID shows up as the main office number This poses a problem as the CEO would like their calls to come from their direct line This can be accomplished by configuring DOD for the CEO s extension Steps on how to configure DOD on the UCM 1 To setup DOD go to UCM6510 web GUI gt PBX gt Basic Call Routes gt VolP Trunks page 2 Click to access the DOD options for the selected SIP Trunk 3 Click Create anew DOD to begin your DOD setup 4 For DOD Number enter one of the numbers DIDs from your SIP trunk provider In the example above Company ABC received 4 DIDs from their provider ABC will enter in the number for the CEO s direct line 5 Select an extension from the Available Extensions list Users have the option of selecting more than one extension In this case Company ABC would select the CEO s extension After making the selection click on the button to move the extension s to the Selected Extensions list Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 114 of 233 andstream Innovative IP Voice amp Video Edit DOD Direct Outward Dialing DOD is a service of a local phone company or local exchange carrier that allows subscribers within a company s PBX system to connect to o
234. up USE EXTENSION GROUPS Here is an example where the extension group can be used Go to web GUI gt PBX gt Basic Call Routes gt Outbound Routes and select Enable Filter on Source Caller ID Both single extensions and extension groups will show up for users to select Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 153 of 233 andstream Innovative IP Voice amp Video Edit Outbound Rule usa Calling Rule Name Pattern Password Privilege Level National Enable Filter on Source Caller IB Available Extensions Extension Groups Selected Extensions Extension Groups Extension Group Accounting Dept Extension Group Marketing Dept Extension Group IT Dept Extension Group Sales Dept Extension Group TechSupport Dept Gi Custom Dynamic Route Figure 68 Select Extension Group in Outbound Route Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 154 of 233 Andstream Innovative IP Voice amp Video PICKUP GROUPS The UCM6510 supports pickup group feature which allows users to pick up incoming calls for other extensions if they are in the same pickup group by dialing Pickup Extension feature code by default CH CONFIGURE PICKUP GROUPS Pickup groups can be configured via web GUI gt PBX gt Call Features gt Pickup Groups e Click on Create New Pickup Group to create a new pickup group e Click on to edit the pickup group Select extensions from the list on th
235. urning on Public Mode On the device users can log in and log out using the SIP UserID and password If enabled on the UCM6510 the SIP Password for the extension will accept only alphabet characters and digits AuthID will also be changed to the same as Extension User Settings Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 66 of 233 dstream Innovative IP Voice amp Video Configure the first name of the user The first name can contain First Name Ze characters letters digits and Configure the last name of the user The last name can contain Last Name SE characters letters digits and Fill in the Email address for the user Voicemail will be sent to this Email address Email Address Select the voice prompt language to be used for this extension The default setting is Default which is the selected voice prompt language under web GUI gt PBX gt Internal Options gt Language The dropdown list Language shows all the current available voice prompt languages on the UCM6510 To add more languages in the list please download voice prompt package by selecting Check Prompt List under web GUI gt PBX gt Internal Options gt Language Use NAT when the UCM6510 is on a public IP communicating with NAT 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 The default setting is enabled
236. utside lines directly DOD Number 68861711234 Available Extensions Selected Extensions Figure 48 DOD extension selection 6 Click Save at the bottom Once completed the user will return to the Edit DOD page that shows all the extensions that are associated to a particular DOD Edit DOD Direct Outward Dialing DOD is a service of a local phone company or local exchange carrier that allows subscribers within a company s PBX system to connect to outside lines directly Create a new DOD Edit DOD 6176518241 3002 4451234567 2005 Total 2 Show 1 1 Go e ER Figure 49 Edit DOD Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 115 of 233 Andstream Innovative IP Voice amp Video This page intentionally left blank Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 116 of 233 E ten Innovative IP Voice amp Video CALL ROUTES OUTBOUND ROUTES In the UCM6510 an outgoing calling rule pairs an extension pattern with a trunk used to dial the pattern This allows different patterns to be dialed through different trunks e g Local 7 digit dials through a FXO while Long distance 10 digit dials through a low cost SIP trunk Users can also set up a failover trunk to be used when the primary trunk fails Go to web GUI gt PBX gt Basic Call Routes gt Outbound Routes to add and edit outbound rules e Click on Create New Outbound Rule to add a new outbound route e Click on to edit t
237. y 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 woe outbound rule to make outbound calls The default setting is No Select audio and video codec for the extension The available codecs are PCMU PCMA GSM AAL2 G 726 32 G 722 G 729 G 723 ILBC Codec Preference ADPCM LPC10 H 264 H 263 and H 263p In the selected codec list users can click on UP or DOWN arrow to adjust the order for the codec priority BATCH ADD IAX EXTENSIONS Under web GUI gt PBX gt Basic Call Routes gt Extensions click on Batch Add Extensions gt Batch Add IAX Extensions Table 22 Batch Add IAX Extension Parameters Configure the starting extension number of the batch of extensions to be Start Extension added Firmware Version 1 0 0 25 UCM6510 IP PBX User Manual Page 77 of 233 ndstream Create Number Specify the number of extensions to be added The default setting is 5 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 an outbound rule s privilege in order to make outbound calls using this rule If the outbound rule privilege is disabled this option will not take effect

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