Page 2
Jing Jie Co., Ltd. concentrates to provide the IPV6+IPV4 SIP server farm solution including SIP proxy server, IP-PBX, SIP surveillance server and QoS Monitor to our partner, system integrator and value added reseller. All Jing Jie solutions are provided to support both IPV4 and IPV6 dual stack simultaneously.
Page 3
Jing Jie make no claim to these trademarks. Jing Jie Co., Ltd. (Jing Jie) makes no representations or warranties with respect to the contents hereof. In addition, information contained herein are subject to change without notice. Every precaution has been taken in the preparation of this manual.
Table of Contents Part I Getting Start 1 Logon the system ........................... 8 2 Change Default Password ........................... 8 3 Setting SIP Service ........................... 9 4 Create Office ........................... 11 5 Create Extensions ........................... 12 6 Create AA Flow from Template ...........................
Page 5
Office ................................. 64 Access Code ..............................69 Pickup Group ..............................72 Black List ..............................73 Holiday ..............................74 Prompt File ..............................75 VMS Routing ..............................76 Menu Designer ..............................79 Extension ................................. 83 Call Feature ..............................92 Follow Me ............................96 Incoming Call Blocking List ............................
Page 6
7 Diagnostic ........................... 152 System Status ................................. 153 Extension Status ................................. 155 Call Status ................................. 156 High Available Status ................................. 157 Blocked IP ................................. 159 SIP Trunk Status ................................. 160 AA/VMS Status ................................. 161 Ping ................................. 162 Call Capture ................................. 163 System Inform ation .................................
Getting Start After successfully installed the system , first of all is to login to the web management interface. You can either using IPv4 or IPv6 address to access GUI management interface by using popular browser such as Internet Explorer or Firefox. If you are installing a HA version, make sure that MYSQL database replication service is working correctly and both server is running to make the setting simplified.
Getting Start Click ADMINISTRATION -> Account -> admin and the following screen will appear. Input the new password at the Password and Confirm Password fields and click the Apply button to take effective. Click logout to quit the system UI and relogin by new password for confirmation.
Page 10
Click Default button to get the default setting of SIP service and change the following settings: Parameter Name Value Attached WAN interface eth0 Name Attached LAN interface eth1 (for 2 Ethernet Leg Mode), none (for 1 Ethernet Name Leg Mode) UDP Service Port 1 5060 UDP Service Port 2...
Getting Start Parameter Name Value IPV6 Service disable IPV6 UDP Service Port Contact Update Method overwrite Default Register TTL NAT Register TTL Click button and COMMIT to take effect. Create Office Before we can create the required SIP extensions, the administrator need create an office which includes the SIP extensions.
Click the default button to set those access to a default value and enter the following values: Parameter Value Name Office ID Office Name office 1 Auto Attendant set operator number (e.g. 999) for working hour operator, Operator after work operator and holiday operator Description office 1 Click Apply to save it.
Page 13
Getting Start Create extension 1001 based on the following values: P a r a m e te r Value Na m e Extension Mode enable Extension Number 1001 SIP User ID 1001 SIP Password 1001 Belonged User Select group 1 "Office 1" Group Name 1001...
Page 14
Enable the Voice Mail for 1001: P a r a m e te r Value Na m e Voice Mail enable Voice Mail 1001 Password Voice Mail English Language Click Apply to save it. Create extension 1002 based on the following values: P a r a m e te r Value Na m e...
Getting Start P a r a m e te r Value Na m e Voice Mail enable Voice Mail 1001 Password Voice Mail English Language Click Apply to save it. Create operator as following: P a r a m e te r Value Na m e Extension Mode...
Page 16
Click to prepare the copy from a template and the following popup screen will appear. Select 'Copy Menu From Template' and select a suitable template to apply. After apply it, the system will duplicate the call flow and voice prompt into this office. It will become like as below.
Getting Start Create AA/VMS Access Key In order to have Auto Attendant and Voice Mail services enabled for your company. You need create the corresponding routing to be associated to it. Click Office -> Office 1 -> VMS Routing and the following will appear. Click Create Default Route to have the system to create the default routing for the created office.
Page 18
The next is to create the auto attendant incoming routing number. Click New to add the routing number for AA as follows. Enter the following example for the auto attendant service: Parameter Value Name Pilot Number 9000 Max Calls Time to Answer (sec)
Getting Start Parameter Value Name Service Type Auto Attendant Service Language English Click Apply to save it. Click COMMIT to take effective. Verify the Device Register After create extension 1001,1002 and operator 999, you need to configure the SIP phone, gateway or soft-phone to register to the system. To confirm whether those two extensions are registered correctly or not, click DIAGNOSTIC ->...
Make a Extension Call Use extension 1001 calls extension 1002. 1002 should ring and you should able to answer it and talk. To confirm the calls status from the system, click DIAGNOSTIC -> Call Status and the following screen will appear. Input 1001 and 1002 as above for the search criteria, click Search button.
Page 21
Getting Start Create a extension 1003 for FXO gateway. Please use the following values for the gateway. P a r a m e te r Value Na m e Extension Mode enable Extension Number 1003 SIP User ID 1003 SIP Password 1003 Extension Type FXO/Trunk/Proxy...
P a r a m e te r Value Na m e setup this field. Belonged Office Select group 1 "Office 1" Name 1003 Click Apply to save it. Click COMMIT to take effective. Then you need to configure the FXO gateway to register to the system. Verify the FXO is registered or not by using the extension status as the description in "Verify the Device Register".
Page 23
Getting Start Input the following values for the SIP trunk as an example. You should get those value from your VOIP carrier. Parameter Name Value SIP Trunk ID SIP Domain sip.carrier.com Register TEL 9900 Registrar Server 112.3.3.1 Registrar Port 5060 Outbound Proxy 112.3.3.1 Server...
Page 24
Input the following values for extension 1004. P a r a m e te r Value Na m e Extension Mode enable Extension Number 1004 SIP User ID 1004 SIP Password 1004 Extension Type SIP trunk SIP Trunk ID Select ID "1 -My SIP Carrier" Belonged User Select group 1 "My first calling group"...
Getting Start P a r a m e te r Value Na m e Name 1004 Description My VOIP Carrier Extension Click Apply to save it. Click COMMIT to take effective and verify the FXO is registered or not by using the extension status as the description in "Verify the Device Register".
Page 26
Input the following values to create the routing plan. Parameter Name Value Routing Plan Mode enable Pilot Number Length ignore Belonged Office Route Period All the time Hunt Type Round Robin Hunt Remove Pilot Number Click Apply to save it. Click Back to return the Routing Plan page. Select the created route plan and click the routing list button below.
Page 27
Getting Start Select New to add a routing list as below. Input the following value to add the routing list. Parameter Name Value Extension Number 1003 (PSTN gateway), 1004 (VOIP Carrier)
Parameter Name Value Preference Click Apply to save it. Click COMMIT to take effective. 1.14 Make a PSTN Call Use extension 1001 or extension 1002 to dial the 002123456 (this number should be replaced to your own telephone number). If everything is getting smoothly, your phone should ring and you should able to answer it and talk.
Using the System Using the System The administrator can logon the web GUI interface to manage the system service. It provides the sip service provisioning, real time system status monitor and system statistic report. The default login URL for administrator login is http:// xxx.xxx.xxx.xxx:9200 https://xxx.xxx.xxx.xxx:9201)
Page 30
The detail of each parameter are described as below: Parameter Name Description Product Name The product name System Release The current running system release Web Release The current running web release Sip Domain Accepted SIP domain or FQDN of the system IPV4 IPV4 SIP service status IPV6...
Using the System Parameter Name Description RADIUS RADIUS Settings RADIUS Server RADIUS Server Licensed Feature Licensed Feature Expired License Expires Date Extension Groups Current created Extension Group Created Extension Created Current created Extension Sip Trunk Created Current created SIP Trunk Routing Plan Created Current created Routing Plan System...
2.2.1 SIP Service The SIP Service page is the main configuration for SIP core. Click SYSTEM -> SIP Service to view and change the settings.
Page 33
Using the System The detail of each parameter is described as below: Parameter Description Name Domain Name 1-6 Accepted SIP domain or FQDN of the system Attached WAN If system acts as a SIP router, WAN interface indicates the interface Name Ethernet leg connected to public IP network.
Page 34
Parameter Description Name UDP Service Port IPV4 UDP port used for SIP service. The default value is 5060. UDP Service Port IPV4 UDP port used for SIP service. The default value is 8080. UDP Service Port IPV4 UDP port used for SIP service TCP Service Port TCP port used for SIP service.
Page 35
Using the System Parameter Description Name Max Forward The max forward counts for a call to be forwarded. When it Count reach the count, the forward setting will be ignored. The default value is 5. Max Forward/ The system wide max allowed forward or transferred calls. It Transferred Call is recommended to set it to 2 or 5 instead of unlimited.
2.2.1.1 TLS Certficate Upload This is used to upload the SIP required TLS certificate. Click TLS Certificate Upload button to upload the certificate for SIP TLS service. 2.2.1.2 SIP Reject Code This is a mapping table for SIP Proxy reject reason code. The detail of each parameter is described as below: Parameter Name Description...
Page 37
Using the System The detail of each parameter is described as below: Parameter Name Description No Answer Time Out The default time to wait for the called party to answer. The recommended value is 300. Session Validation The time to check whether the call is still connected or Period not.
Page 38
Parameter Name Description SIP QoS Diff-Serv Tag The DiffServ tag used for SIP signaling. The default value is 0 which means no QOS tag is used. RTP QoS Diff-Serv Tag The DiffServ tag used for RTP packets. The default value is 0 which means no QOS tag is used. 302 Moved Handling When system receive a 302 moved response, send it back to original caller if it is set to "No".
Page 39
Using the System Parameter Name Description the ENUM user and its DNIS screening in order to protect your network will not be attacked by any SIP caller. ENUM User This is the user will be used for ENUM incoming call in order to protect the system against the sip attack by taking the advantage of ENUM.
Page 40
Parameter Name Description Send New Replaced Whether to send a new replaced Invite to transferee or Invite to Transferee not? Use Global No Answer When it enabled and no answer forward was not Timeout When No enabled, the global no answer forward will be used, Answer Services were otherwise user's no answer forward will be used.
Using the System Parameter Name Description Talking RTP Timeout The RTP Timeout for voice calls. If system doesn't receive any RTP for any side, the call will be dropped. Video Talk RTP Time The RTP timeout for video calls. If system doesn't receive any RTP for any side, the call will be dropped.
Parameter Description Name SIP T1 The T1 timer, which is defined in milliseconds, specifies the amount of round trip time (RTT), that the client will attempt to send a SIP Request and expect a response. By default, the T1 timer is set to 500ms. SIP T2 Maximum retransmission interval for non-INVITE requests and INVITE responses.
Page 43
Using the System Parameter Name Description RADIUS Enable RADIUS authorization and accounting or not. Administrator need also turn on RADIUS call authorization from extension in order to let system send the RADIUS authorization to RADIUS server when the extension calling. The default value is disabled.
Parameter Name Description Database CDR If it is enabled, the system will do the local billing calculation. It will store call detail into local database and use the tariff plan to calculate the charge amount. And you will able to see the top usage or division report from the Billing.
Using the System 2.2.6 Call detail record (CDR) can be turn on or off here. Click SYSTEM -> CDR to view and change the settings. The detail of each parameter is described as below: Parameter Name Description CDR Logging Whether to enable the CDR recording or not. The default value is Yes.
Page 46
The detail of each parameter is described as below: Parameter Name Description HTTP Service Port The TCP service port for web GUI management. The default value for administrator and supervisor login is 9000. The default value for extension login is 80. HTTPS Service Port The TCP service port for HTTPS (SSL) web GUI management.
Using the System Parameter Name Description Use Validation Code Use CAPTCHA to against the response is not generated on Login by a computer or not for logon. It is recommended to enable it for security reason. Extension Web The default web language will be used when a extension Language is login.
Parameter Name Description Login ID The ID was tried for the failed login Login Time The time was blocked Time to Unblock The time will be unblocked 2.2.8 Database This is for system database settings. Click SYSTEM -> Database to view and change the settings.
Using the System Parameter Name Description MYSQL DB Server MYSQL database server IP address. The default value is 127.0.0.1 MYSQL Port MYSQL database connection port. The default port is 3306. MYSQL User ID MYSQL access user ID MYSQL Password MYSQL access password MYSQL Database MYSQL Database Name Name...
Parameter Name Description License Key The license key generated Expired Date The expired date for the license Click Import to upload a granted license, Export to download a existing license. Click Activate to active a granted serial number license. Click Deactivate to deactivate a serial number in order to move to other server.
Using the System Parameter Name Description Syslog Debug Enable syslog debug or not Syslog Debug Server IP The syslogd server to receive the debug information. The port to receive the syslog is 514. Trace Target The target to be used for debugging. It could be the telephony number or IP address.
Page 52
The detail of each parameter is described as below: Parameter Description Name System Alert The filter level to send the alert out. The default is level of Threshold "Notice". Alert to Syslog Whether to send the system alert to syslogd server or not. Syslog Receiver IP The syslogd server to receive the system alert.
Using the System Parameter Description Name Email Subject The email subject for the system alert notice. The variable "$HOSTIP$", Host IP address, could be used in the subject to make the subject easy to be read (e.g. System Alert Notice from $HOSTIP$). Email User ID The email sending account ID Email User...
The voice logging service requires additional license for running it. Please contact "Jing Jie" when you need it. The system support MP3 compression for voice logged files including VBR and CBR coding. Click SYSTEM -> Voice Logging to view and...
Page 55
Using the System change the settings. The detail of each parameter is described as below: Parameter Name Description MP3 Encoding MP3 encoding method, it could be CBR (Constant Bit Rate) or VBR(Variable Bit Rate) depending on the compression ration and quality. Bit Rate The selected bit rate will be used when CBR is selected.
Parameter Name Description Mixed Mono Channel Whether enable mix calling and called parties' voice into a mono channels for MP3 or not. Recording RTP The maximum keeping days for recording voice logging Keeping Days files. 2.2.14 VMS Settings The VMS Settings includes the settings of voice mail system, auto attendant and conference.
Page 57
Using the System Parameter Name Description Local SIP V6 UDP This is local IPV6 SIP UDP port will be used for AA, VMS Port and conference service. Local Media UDP The is the media UPD starting port will be used for AA, Start Port VMS and conference service.
Parameter Name Description Min MWI Subscribe The minimum time for MWI SUBSCROBE request. The Time (mins) recommended value is 30 minutes. Codec 1-5 The codec will be used for AA, VMS and conference. 2.2.15 High Available The system supports active/standby redundant mode. It relies on MYSQL database replication and high available software to build the system redundant as follows: To make the redundant working smoothly, you need to the following to be prepared: 1.
Using the System Active/Standby Redundant for 2 Ethernet Legs Above case but IPV4 only Some system architecture examples will be showing in the following topic. For high available, if the fail-over is happened, the call will be continue for 2-5 second silence and voice recording will became 2 separate recorded file.
2.2.15.2 Active/Standby (2 Ethernet) In this mode, the system is serving for both WAN and LAN interface and another node is a standby node. In the normal case, the SIP CPE is register to their server through either WAN's VIP or LAN's VIP. If one of the machine is down, the another peering node will taking over the VIP and continue the service.
Using the System 2.2.15.3 IPV4 Only Redundant IPV4 only redundant is a simplified architecture for described above. In this mode, IP V6 is disabled. 2.2.15.4 High Available Settings Click SYSTEM -> High Available to change the HA settings. Some other parameters might also affect the HA settings such as IPV6 enabled and Attached LAN Interface Enable/Disable.
Page 62
Parameter Description Name Cluster ID Cluster ID is used to identify the cluster. Different Cluster ID will not able to working together. For different Cluster ID, it is required to use different Cluster Service Port. The maximum length of ID is 6 bytes. Cluster Service The UDP port will be used for intra-cluster communication to Port...
Page 63
Using the System Parameter Description Name Heartbeat Keep The interval to send the heartbeat message. This value will Alive Interval decide how long the failure can be detected. The minimum value is 300ms and maximum is 3000ms. The default value is 700ms.
Parameter Description Name IPV4 VIP for LAN Virtual IPV4 address for LAN interface. IPV6 VIP for WAN Virtual IP V6 address for WAN interface. The IPV6 address must be a global unicast addressed, not a link-local or site- local address. IPV6 VIP for WAN is only available when 1 Ethernet leg mode is used (Attached LAN Interface is disabled).
Page 65
Using the System Select New, Modify, Delete to change the office settings. The following web page will appear:...
Page 66
The detail of each parameter is described as below: Parameter Description Name Office ID Office ID Office Name The name of office Digit Manipulation The Digit Manipulation Group ID will be used for this group. Group Time Zone The time zone will be used for this office's extension. It need to be set if using CPE auto provisioning feature.
Page 67
Using the System Parameter Description Name Voice Mail Subject The email subject for sending a new voice mail notice. The administrator can input the system variable to make the subject easy to be read. Voice Mail Body The email body for sending a new voice mail notice. . The administrator can input the system variable to make the subject easy to be read.
Page 68
Parameter Description Name *505: Toggle working hour and after call work mode. (This will be also the BLF to subscribe in order to get the working hour (LED is OFF or Green) or not (LED is ON or Red). Hold Tone Music The music will be used for music on hold and transferring Prompt music in AA.
Using the System 2.3.1.1 Access Code The Access Code parameters are used to define those service activation or deactivation from telephone set. Once the feature access code is accepted by the system, the system will send SIP "180 ring" and user will hear ring back tone. If it is rejected by the system, the will send "406 Not Acceptable"...
Page 70
Parameter Name Description ACCESS_CDOE (using the existing setting). Disable Call Forward The access code to disable "Call Forward for no No-Answer answer". The default value is "*06". Dialing Rule: ACCESS_CODE. Enable Call Forward The access code to enable "Call Forward" for busy. Busy The default value is "*03".
Page 71
Using the System Parameter Name Description Calling with Caller ID The access code to enable calling ID for this call. The default value is "*15". Dialing Rule: ACCESS_CODE+DIAL_TEL. Calling without Caller ID The access code to disable calling ID for this call. The default value is "*16".
Parameter Name Description which means disble this feature. Set Call Forward Always The access code to set Call Forward Always for an for Extension unregistered extension. Dialing Rule: ACCESS_CODE+Extension+* +web_password+*+forwarded number. 2.3.1.2 Pickup Group The pickup group is used for pickup the ring call within the same office. Each pickup group cannot cross the office.
Using the System The detail of each parameter is described as below: Parameter Description Name Pickup Group ID Pickup group ID for call pickup Description Description for this group 2.3.1.3 Black List Black List can be used for auto attendant service to filter those unwanted calls. When auto attendant receive a call, it will try to map the incoming caller ID against the black list.
Click New to add a new black list ID as follows: The detail of each parameter is described as below: Parameter Description Name Office ID Office ID for this black list Blocking Number The number will be in black list. 2.3.1.4 Holiday The holiday definition for the office.
Using the System The detail of each parameter is described as below: Parameter Description Name Office ID Office ID Holiday The holiday date in formation of MM/DD. Description The holiday name or description for this holiday. Holiday Prompt The holiday prompt could be used for this specified holiday in menu editor.
2.3.1.6 VMS Routing The AA/VMS Routing is used to define those AA and VMS related service call routing number. For each service, you need create a service routing number in order to use it. Each service type was defined as follows: Service Type Description Auto Attendant...
Page 77
Using the System Service Type Description Meeting Me The dial in conference service. You need create each Conference conference room here. Call Park The call park room will be created. If you create it, total 10 room will be created. For example, the pilot number is 812, thus you will have park room from 8120 to 8129.
Page 78
Service Type Description will be played. The interface to manage AA/VMS routing is showed as below: Click New to add a new routing plan as follows: The detail of each parameter is described as below:...
Using the System Parameter Description Name Office ID Office ID Pilot Number The AA/VMS service routing number Max Calls Maximum allowed calls for this service Time to Answer The time to wait before answer this service call. Service Type The AA/VMS service type described above Service Language The language will be used for those AA service such as voice mail main menu, meet me conference or outgoing...
Page 80
In the top left, you will able to see the menu icon as For each menu, click right key and your will see the Modify action for reviewing and modify the menu parameters. To create a new menu, click add icon and you will see the following:...
Page 81
Using the System The detail of each parameter is described as below: Parameter Description Name Office ID Belonged office ID Menu ID The call flow menu ID Menu Type The menu type to indicate this menu is the entry point for difficult call flow.
Page 82
Parameter Description Name Max DTMF Maximum DTMF digits to be received. Retry Count The max retry count within this call flow menu Main Prompt The prompt will be played when execute the call flow menu. Retry Prompt The prompt will be played for retrying this call flow menu. It could be played such as no DMTF received, not a extension, invalid input etc.
Using the System Parameter Description Name Description The description of this menu. Key Action The action will be executed when this key was matched. The following actions can be selected: Transfer to operator: will try to transfer this call to operator. Repeat Prompt: Will repeat the Main prompt.
Page 84
simultaneously. Click EXTENSION -> Extension to view or modify the extension settings. Or you can click EXTENSION -> Office -> Extension to see the office owned extension only. Click Rebuild can rebuilt this mac's device configuration if auto provisioning is enabled.
Page 85
Using the System The detail of each parameter is described as below: Parameter Description Name Extension Mode Whether to activate this extension or not. Extension Number The extension telephone number for SIP registration (from/ to header). SIP User ID The SIP user ID for authentication SIP Password The SIP user password for authentication SIP Display Name...
Page 86
Parameter Description Name need to be set to enable. The web password can only allow digits (0-9), since it will be used for "outgoing call privilege access" as a password. Belonged Office An extension should be only belong to a office. Please select the office here.
Page 87
Using the System Parameter Description Name SIP Trunk: It is used for connecting to another VOIP carrier. In this case, VOIP carrier will give you a SIP account for calling and you need to set it on SIP trunk then associate it here.
Page 88
Parameter Description Name Permanent Contact When the contact policy is permanent contact, this is the defined contact URI and the target interface. The SIP URI is used for the contact address. For example: sip:1001@112.25.26.3:5060 or sip: 1030@113.111.222.333:9099. Permanent Contact When the contact policy is permanent contact, this is the defined contact URI and the target interface.
Page 89
Using the System Parameter Description Name No Answer Time Out The time to wait in seconds for the called party to answer. The default value is to use the global settings in SYSTEM- >Service Parameter->No Answer Time Out. Dedicate Device 1 The allowed device to be used for this extension.
Page 90
Parameter Description Name Not Matched Policy The way to handle when IP network or country is not matched the defined network. Send Alert Only: enable this if administrator need receive an alert only Send Alert and Unregister: This option will send alert message and unregister this unmatched contact.
Page 91
Using the System The detail of AA/VMS Setting parameters are described as below: Parameter Description Name Voice Mail Whether enable or disable the voice mail. Voice Mail The password to access the voice mail. Password Outgoing Call within Whether allow to dial out to PSTN (not extension) within Personal Greeting The personal greeting when get into the extension's voice mail.
Parameter Description Name number. 2.3.2.1 Call Feature Each extension can enable or disable the call feature individually by click Call Feature button. The following screen will appear. The detail of each parameter is described as below: Parameter Description Name Call Forward Always Enable call forward always.
Page 93
Using the System Parameter Description Name Call Forward No Enable call forward for no answer call. Answer No Answer Forward The telephone number to be forwarded Number Call Forward Busy Enable call forward for a busy call. Busy Forward The telephone number to be forwarded Number Call Forward Enable call forward when SIP client is not registered.
Page 94
Parameter Description Name Incoming Call When it is checked, the incoming call will be filtered by Blocking matching the "calling party number" with "Incoming call blocking list". If it is matched, the call will be rejected. Outgoing Call When it is checked, the dialed number will be filtered by Blocking "Outgoing Call Blocking List".
Page 95
Using the System Parameter Description Name Disable RADIUS Whether to disable the RADIUS Billing send or not. If it is Billing set to yes, the system will not send any RADIUS billing out and this number will not be billing. Normally, it should be set to No.
Parameter Description Name Disable Video Call Whether to allow video call to be madden or not? Enable Distinctive Whether enable or disable distinctive ringing feature? This Ringing feature need a compatible SIP phone to support this feature. 2.3.2.1.1 Follow Me The follow me time should be defined here when Follow Me service was enabled in Call Feature.
Using the System 2.3.2.1.2 Incoming Call Blocking List When enabled the incoming call block feature in call feature screen. The calling party number defined here will be filtered based on the blocking type. Select New, Modify, Delete to change the screening setting. The following web page will appear: The detail of each parameter is described as below: Parameter...
Parameter Description Name Pilot Number The calling number used to be matched. If incoming calling number (SIP user part) is matched, the call might be rejected or accepted based on the "Blocking Type". Blocking Time The system allow to have time restricted screening feature.
Using the System The detail of each parameter is described as below: Parameter Description Name Blocking Target Incoming call or outgoing call to be screened. Pilot Number The called number prefix used to be matched. If the outgoing number prefix is matched the pilot number, the call might be rejected or accepted based on the "Blocking Type".
You can double click the item to hear the voice mail. The detail of each parameter is described as below: Parameter Description Name Calling Time The time to start the call Calling From The calling party number Status Whether the voice mail was read or not? 2.3.2.3 Batch Create Batch create is mainly used for creating testing data.
Page 101
Using the System The detail of each parameter is described as below: Parameter Description Name Batch Mode The way to batch create extension: Batch Create: Create extension numbers in between "From Extension Number" and "To Extension Number". The existing number will be ignored. Batch Modify: Replace the existing record by using the current settings in between "From Extension Number"...
Parameter Description Name Web Password The prefix for the web password. The system will set the Prefix default web password as "Web Password Prefix"+"Extension Number". For security reason, it is recommended to set a prefix instead of keeping blank. SIP Password Prefix The prefix for SIP password.
Page 103
Using the System Click New to add a new phone book entry as follows. Input the name and telephone number to create an entry of phone book. If you want to create a group to be used later, click Phone Book Group -> New and the following will appear.
Click >> to assign the extension to the group or << to un-assign it. Feature The system provides the flexible SIP trunking, digit manipulation, routing plan, DNIS screening group and others can be defined here. Those features is core for providing required services for customers.
Page 105
Using the System Select New, Modify, Delete to change the SIP Trunk setting. The following web page will appear: The detail of each parameter is described as below:...
Parameter Name Description SIP Trunk ID SIP trunk ID SIP Domain The SIP register domain for SIP trunk user Register TEL The SIP User (normally, it is TEL number) for register Registrar Server The SIP registrar proxy server IP address or DNS name. Registrar Port The SIP service port to register (default value is 5060) Outbound Proxy...
Page 107
Using the System Select New, Modify, Delete to change the routing plan. The following web page will appear: The detail of each parameter is described as below: Parameter Name Description Routing Plan Mode Activate this routing plan or not Pilot Number The leading number (prefix) used to be matched with the called number.
Page 108
Parameter Name Description Length The length to be matched for the called number length. If "ignore" is checked, the length matching is ignored. Belonged Office The selected office will be applied to this routing. Select "All" if don't need group filter. Route Period The time of day and weekday to execute this route.
Using the System Parameter Name Description Hunting No-Answer No answer time out in seconds for this route. The Timer default value is " Use Global Setting" which means use the global setting in SYSTEM-> Service Parameter. SIP Request Response SIP Request response time out for this route. The Timer default value is "...
Page 110
Select New, Modify, Delete to change the Hunting Stop Code. The following web page will appear: The detail of each parameter is described as below: Parameter Name Description Stop Code The reason code to be used for stopping the hunting. If the reason code are not listed, you can enter the SIP response code here to stop the hunting.
Using the System 2.4.2.2 Routing List Each routing plan contains multiple routing devices, such as gateway, VOIP carrier or extension. Here is the place to define where to be routed. Click Routing List button after select a routing plan. The following screen will appear: Select New, Modify, Delete to change the Routing List setting.
Parameter Name Description Extension Number The extension to be added to this routing plan. Preference The preference priority number: 0 is lowest and 9 is highest. The higher value indicate higher preference for preference route. 2.4.3 Digit Manipulation Digit Manipulation is used to manipulate the calling or called number. The administrator can insert, delete or change some digits from original number.
Using the System The detail of each parameter is described as below: Parameter Description Name Group ID Digit Manipulation Group ID Description The description for this Digit Manipulation Group 2.4.3.1 DM Group List The detail operation list for the digit manipulation group. The process policy of digit manipulation list within the group is showing as below: Step 1.
Page 114
Select New, Modify, Delete to change the Digit Manipulation List. The following web page will appear: The detail of each parameter is described as below:...
Page 115
Using the System Parameter Description Name Mode Activate this digit manipulation group or not Pilot Number The leading number (prefix) to be matched Incoming The incoming number type to be matched. It could be calling Number Type number (ANI) or called number (DNIS). For most of case, the DM incoming type will be DNIS.
2.4.4 Abbreviated Dialing The abbreviated dialing group is used to replace the dialed abbreviated number to the real telephone number. Click FEATURE -> Abbreviated Dialing Group to view the current groups of emergency call as follows: Select New, Modify, Delete to change the Abbreviated Dialing Group setting. The following web page will appear: The detail of each parameter is described as below: Parameter...
Using the System 2.4.4.1 Abbreviated Dialing Group List Here is the place to define the replacement of abbreviated number call. Click the Group List after select a created Abbreviated Dialing Group as follows: The detail of each parameter is described as below: Parameter Name Description Abbreviated Dialing...
Page 118
Select New, Modify, Delete to change the Emergency Group setting. The following web page will appear: The detail of each parameter is described as below: Parameter Description Name Emergency The Emergency Call Group ID Group ID Description The description for this Emergency Call Group...
Using the System 2.4.5.1 Emergency Group List Here is the place to define the replacement of emergency call. Click the Group List after select a created Emergency Group as follows: The detail of each parameter is described as below: Parameter Name Description Emergency Telephone The called emergency number, such as 911, 119, 110...
Page 120
Select New, Modify, Delete to change the Screening Group setting. The following web page will appear: The detail of each parameter is described as below: Parameter Description Name Screening Group DNIS (called number) screening group ID Description The description for this DNIS screening group...
Using the System 2.4.6.1 Screening List The detail of telephone number to be blocked or un-blcoked for the selected DNIS screening group should be defined here. Click Screening List button after select a created DNIS Screening Group to view the screening list as below: Select New, Modify, Delete to change the Screening List setting.
Pilot Number The called number prefix used to be matched. If the outgoing number prefix is matched the pilot number, the call might be rejected or accepted based on the "Blocking Type". Screening Time The system allow to have time restricted screening feature.
Page 123
Using the System Click Import Supported Provisioning Devices if you don't see any provisioned device was here. You can view the supported provisioning device as follows by click an item and modify. The following will appear: Click Rebuild Configuration will rebuild all device configuration files for this device model.
Page 124
The detail of each parameter is described as below: Parameter Name Description Device Name The name to be selected in the dedicate device of extension. User Agent The SIP "User Agent" header used to be filtered for dedicate device. Auto Provisioning Whether enable auto provisioning feature for this device or not.
Using the System 2.4.8 Block Device Block Device is used to filter the incoming request. If the incoming SIP request's "SIP User Agent" header match the defined block device by using prefix matching, the incoming SIP request will be ignored silently. Click FEATURE ->...
Parameter Name Description Device Name The name to be blocked by the system. The request from this type of agent will be ignored. User Agent The SIP "User Agent" header used to be filtered for blocking. 2.4.9 DID Routing This service is used to have a central management for your DID number routing. You can use to route your DID number to any extension, gateway or proxy.
Using the System The detail of each parameter is described as below: Parameter Name Description DID Number DID number to be routed Routed Extension When system receive this DID number, which extension is going to be routed. If the routed extension type is phone/ ATA, the called will be replaced to the extension tel number in order to reach the phone or ATA.
Page 128
Select New, Modify, Delete to change the Voice Logging Target. The following web page will appear: The detail of each parameter is described as below: Parameter Name Description Logging Target The target to be recorded. If caller number or called number matched the logging target, this call will be recorded.
Using the System 2.4.11 Queue Prompt The system support call queue feature. The max call queue can be supported is 1000, starting from 000 to 999. You need config call queuing feature in Routing Plan to enable call queuing feature. Here is the place to put the required queue prompts. There are 2 prompts for each call queue.
Parameter Description Name Prompt ID There are 2 prompts will be played for each queue. The first prompt is xxx_0.wav which will be played once and repeat play the xxx_1.wav after it. xxx is the queuing music ID. Play Once Prompt This file will be play continues after play once prompt.
Using the System The detail of each parameter is described as below: Parameter Name Description BLF Number BLF group representative number Group Type Only Proxy BLF group can be created here. ACD BLF group need to be created in ACD module. Description The description for this BLF group 2.4.13 MAC List...
Page 132
The detail of each parameter is described as below: Parameter Description Name Batch ID The ID for this batch Device Name The device model will be applied for this provisioning batch Description The description for this batch Click Detail to view and edit the detail of this batch provisioning list and the following will appear: It will be easier to export to Excel, edit and Import to system.
Using the System Provision to provision it. Report The system provides system statistic and status reports for management purpose. 2.5.1 Call Statistic Report Daily call statistic report provides the administrator to understand the call attempts, connected call and access success ration for each hour. Click REPORT -> Call Statistic and select the day to view the daily report as follows.
Field Name Description Access Success The average ASR (access success ratio) for this period Ratio 2.5.2 Extension Statistic Report The extension statistic report provides the current register user per hour. The administrator can use this report to know whether all user are registered or not. Click REPORT ->...
Using the System The detail of each report field is described as follows: Field Name Description Time The time for the event Extension The extension number for the event User Name The SIP user name for the event State The extension status changed which could be registered or unregistered Private IP The private IP address from SIP contact address...
Page 136
The detail of each report field is described as follows: Field Name Description Period The time period for this statistic NAT Resource The licensed NAT resource Peak NAT Req The peak number of NAT resource request during this period NAT Utilization (%) The utilization for NAT resource NAT Serviced The NAT request serviced during this period...
Using the System 2.5.5 System Alert Report This report provides system alert notice report. The administrator can use it to understand when and which service had problem. Click REPORT -> System Alert to view the report. The detail of each report field is described as follows: Field Name Description Time...
2.5.6 Web Provisioning Report The system will record down all the access to the system from web. The administrator can use it to audit the system and tracking the changes. Click REPORT -> Web Provisioning to view the report as follows: The detail of each report field is described as follows: Field Name Description...
Using the System 2.5.7 Voice Logging Report The voice logging service requires additional license to run. Please contact "Jing Jie" for detail. If a target was recorded, you can query and listen the recorded prompt here. Click REPORT -> Voice Logging to view the report as follows: Click Detail, can see the detail information for this logged call.
2.5.8 Voice Logging Statistic This report provides the utilization of Voice Logging resource. The administrator can verify how many Voice Logging resource are used. Click REPORT -> Voice Logging Statistic to view the report as follows: Field Name Description Period The time period for this statistic Logging Resource The licensed logging resource...
Using the System Field Name Description Logging Serviced The logging request serviced during this period Logging Failure The count of failed logging request during this period Logging Failure Rate (%) The logging request failure rate for this period 2.5.9 AA/VMS Statistic Daily AA/VMS statistic report provides the administrator to understand the AA/VMS and conference resource usage for each hour.
Page 142
Field Name Description Total Conference Total service count for conference service Resource Peak Conference Peak service count for conference service Resource Auto Attendant Service count of auto attendant Peak Auto Attendant Peak service count for auto attendant Total Auto Attendant The total service count for auto attendant within this period Voice Mail service count...
Using the System Field Name Description Peak Adhoc Conference The peak service count for adhoc conference service. Total Adhoc Conference The total service count for adhoc conference service within this period. Voice Message The service count for extension to extension voice mail or direct to voice mail service.
Page 144
Select New, Modify, Delete to change the division. The following web page will appear: The detail of each parameter is described as below: Parameter Name Description Division ID The charge division ID. Division Name The name of the division Admin Account The administration account for this division.
Using the System You can select extension from left window (no charged extension) and click >> to be assigned to this charge division. 2.6.2 Tariff Plan The tariff plan is used to calculate the charge amount based on the charge unit. It is recommended to assign a default rate for those undefined prefix.
Page 146
Select New, Modify, Delete to change the Tariff Plan. The following web page will appear: The detail of each parameter is described as below: Parameter Name Description Plan ID The tariff plan ID Plan Name The tariff plan name Click Detail to view and modify the tariff rate plan. The following screen will appear. Select New, Modify, Delete to add the Tariff Detail.
Using the System The detail of each parameter is described as below: Parameter Name Description Plan ID The tariff plan ID Pilot Number The prefix to be matched the called number. Check default tariff to set a default rate for this plan. The prefix name The name of this prefix Charge Unit...
Page 148
The detail of each report field is described as follows: Field Name Description Ext. Number Extension Number Division belonged division Caller calling party number Called called party number Duration call duration Amount charged amount Call Type Call type could be the following: Extension: extension to extension calls Outgoing: Extension outgoing call Incoming: Incoming call to extension...
Using the System Field Name Description Destination IP The IP address for the called party SIP Call ID SIP Call ID for this call which could be used for tracking. Universal Call ID Universal Call ID for tracking purpose 2.6.4 Division Billing Report Division Billing Report shows the charge amount and percentage for each division.
2.6.5 Top Usage User Report Top Usage User Report show the top usage user for whole company or division for administrator. Click BILLING -> Top Usage User Report and select the queried period to see the following report. The detail of each report field is described as follows: Field Name Description Ranking...
Using the System 2.6.6 Top Prefix Usage Report Top Prefix Usage Report show the top usage user for whole company or division for administrator. Click BILLING -> Top Prefix Usage Report and select the queried period to see the following report. The detail of each report field is described as follows: Field Name Description...
2.6.7 Prefix Summaries Report Prefix Summaries Report show the status of each defined prefix for selected period based on the selected divsion. Click BILLING -> Prefix Summaries Report and select the queried period to see the following report. The detail of each report field is described as follows: Field Name Description Prefix...
Using the System 2.7.1 System Status The System Status provides the current status of system status. You can see whether the system is up and the resource usage. Click DIAGNOSTIC -> System Status to view the current system status. The following screen will appear. The detail of each filed is described as below: Field Name Description...
Page 154
Field Name Description Peak User The peak of user registered to the system within this hour Peak Call Attempt The peak call attempt to the system within this hour Peak Call The peak connected call within this hour Peak NAT Used The Peak NAT resource used within this hour Failed NAT Request The count of NAT resource request failure...
Using the System 2.7.2 Extension Status The administrator can query the current registered extension by clicking DIAGNOSTIC -> Extension Status. The following screen will appear. The detail of each filed is described as below: De s c r ip tio n Field Name Extension The extension is currently registered...
The contact list will show the current registered devices. Click it and it will allow you to unregister if need. The call list show the current calls for this extension number. Click it and it will allow you to disconnect the call. 2.7.3 Call Status The real time call status can be checked here.
Using the System Field Name Description Calling The calling party number Called The called party number State The current call state Connect Time The connected time for the call Call ID The SIP call ID 2.7.4 High Available Status The HA status can be checked by click DIAGNOSTIC -> High Available Status. If both server are working correctly, you could see that the status of each HA member is "online"...
Page 158
Field Name Description Active/Active Cluster: Both servers are acting as an independent server and backup for each other. The register information will be forwarded to backup node to speed up the fail over timing. Once the active one is failed to service, it will switch over to backup node.
Using the System Field Name Description IPV4 VIP for WAN Virtual IP V4 address for WAN interface for HA Group 2. IPV4 VIP for LAN Virtual IP V4 address for LAN interface for HA Group 2. IPV6 VIP for WAN Virtual IP V6 address for WAN interface for HA Group 2. The IPV6 address must be a global unicast addressed, not a link- local or site-local address.
Field Name Description Block time The time to block this IP address. 2.7.6 SIP Trunk Status The administrator can query the current SIP trunk status by clicking DIAGNOSTIC -> SIP Trunk Status. The following screen will appear. The detail of each filed is described as below: Field Name Description SIP Trunk ID...
Using the System 2.7.7 AA/VMS Status AA/VMS Status show the current status of service. Click DIAGNOSTIC -> AA/VMS Status to check the resource status as follows. The detail of each filed is described as below: Field Name Description Office ID Office ID for this statistic Auto Attendant Service count for auto attendant...
Field Name Description Peak Meet Me The peak service count for meet me conference Conference service. Total Meet Me The total service count for meet me conference within Conference this period. Call Park The service count for call park service Peak Call Park The peak service count for call park service Total Call Park...
Using the System Input the Host IP address and start the ping test. 2.7.9 Call Capture Call capture is a debug tool for tracking a call and suitable for low traffic mode. If you need large traffic capture and analyse, you need have a qos monitor product to do it. Click DIAGNOSTIC ->...
Click each button to see the different status. For detail, please refer to Linux administration guide. 2.7.11 Search Number Search number can be used to search matched number in DID routing, Extension number, PSTN number, short code or routing plan and display the result for your reference.
Using the System Select a line and click Assign Extension Number, the following will appear: Choice the extension and register interface and click Apply. This MAC's CPE will automatically use it without touching to it. Administration The Administration setting includes the user account management, restart or reboot the service.
Click restart button to restart the whole service. 2.8.2 Reboot System Click ADMINISTRATION -> Reboot System and the following screen will appear. Click Reboot button to reboot the whole machine. 2.8.3 Account The system provides 3 different level of user to login the web, Administrator, Supervisor and Extension.
Page 167
Using the System Click New to add a new user and the following screen will appear. The detail of each parameter is described as below: Field Name Description User Mode Activate or de-activate the user...
Field Name Description User ID The user ID to login Password The user password Authorization The authorized role for the user. As an administrator, it could do anything while supervisor can be customized to have different access right. Language The web GUI language when the user login. 2.8.3.1 Supervisor Access Right For supervisor, the administrator can define the access list to limit the access of web...
Using the System The administrator can set access deny, read only or full access right for each module. Click Apply to save. 2.8.4 Clear Hitory Data It is recommended to clean the unnecessary historical data periodically. Here is the place to clean those historical data. Click Administration -> Clear History Data to clean those historical data.
After upgrade, reboot the machine to take effective. 2.8.7 Logout To quit the management web for the current user, click ADMINISTRATION -> Logout and the following pop screen will appear. Click OK to logout. Commit After you change the system settings, you need to apply it by clicking the COMMIT and the following popup screen will appear: Select OK to commit the changes.
Using the System 2.10 Help The system provides pop up help hint when you move the cursor to the filed as follows. Also you can click HELP to see on line help which provides the same information as this guide.
Division Manger Login The division manager use the same login URL as administrator. After login, it can only access to those division owned extension. Each division can only have 1 division manager. After login the following screen will appear. Please refer to Billing and Extension settings for detail. Please don't forget to click COMMIT to apply the configuration to running system.
Extension Login Extension Login In order to make the system more secure, the system provides a separate port for extension login. The default login for http port is 80 and the default SSL login port is 443. Both are de-fact port for web access and make the customer easily to remember.
Page 174
Parameter Description Name Extension Number The extension telephone number for SIP registration (from/ to header). Web Password The password for extension owner to login the extension web for service settings. In order to allow extension login, the "SYSTEM->WEB Service->Allow Extension Logon" need to be set to enable.
Extension Login Parameter Description Name Anonymous Call When the incoming call doesn't include the caller ID, Blocking whether to reject it or not. Email Missed Call If the extension is unable to take the call, whether to send a email to extension owner or not. Do Not Disturb Enable Do Not Disturb or not.
Page 176
Select New, Modify, Delete to change the screening setting. The following web page will appear: The detail of each parameter is described as below: Parameter Description Name Blocking Target Incoming call or outgoing call to be screened. Pilot Number The calling number used to be matched. If incoming calling number (SIP user part) is matched, the call might be rejected or accepted based on the "Blocking Type".
Extension Login Parameter Description Name Blocking Time The system allow to have time restricted screening feature. When you enter the blocking time for a screening list, this screen will only affected by this certain period. Blocking Type Whether to block or unblock it. When all entries in the same group are set to "block", it means all call can be passed unless those listed pilot number.
The detail of each parameter is described as below: Parameter Description Name Blocking Target Incoming call or outgoing call to be screened. Pilot Number The called number prefix used to be matched. If the outgoing number prefix is matched the pilot number, the call might be rejected or accepted based on the "Blocking Type".
Extension Login The detail of each parameter is described as below: Parameter Description Name Voice Mail Whether enable or disable the voice mail. Voice Mail The password to access the voice mail. Password Personal Greeting The personal greeting when get into the extension's voice mail.
You can double click the item to hear the voice mail. The detail of each parameter is described as below: Parameter Description Name Calling Time The time to start the call Calling From The calling party number Status Whether the voice mail was read or not? Call History Report Extension can query hist own call history list by clicking Call History Report.
Extension Login The detail of each report field is described as follows: Field Name Description Caller calling party number Called called party number Duration call duration Call Type Call type could be the following: Extension: extension to extension calls Outgoing: Extension outgoing call Incoming: Incoming call to extension Misc: Others call type Connect Time...
Page 182
2. Allow to call your customers using office extension 3. Allow to create a 16-parties conference 4. Allow to monitor the meeting me conference room To enable you to use smart calling feature, the extension need have "Outgoing Call within AA" enabled. And then you can use your smart phone or smart pad to login the extension office (default URL is http://xxx.xxx.xxx.xxx:81/ ).
Extension Login 4.7.1 Settings Click Settings and the following will appear. The detail of each parameter is described as below: Parameter Description Name Forward to My Phone Whether to forward my extension to "My Phone" or not. My Phone The telephone number will be used as my phone number for forward, calling out and conference.
4.7.3 Meet Me Conference By using your smart phone, you can manage the meeting me conference from anywhere. Click Meet Met Conference and the following will appear. Enter the monitored meet me conference room number and host PIN code, press Enter and the following will appear.
Extension Login 4.7.4 Create Conference You can create a conference on demand from your smart phone or pad anywhere. Click Create Conference and the following will appear. Click to add a phone book group into conference room. Click to add a contact from phone book into conference room. Click to add telephone numbers into conference room.
Page 186
Un-mute the participant or whole conference room. Disconnect the user from the conference room. The participant was disconnect, click to redial to invite him to join again. Quit the meet me conference control and back to menu.
RADIUS Attribute List This appendix including the system provides RADIUS attribute list for connecting to a RADIUS server. 5.2.1 Authorization Request Message The authorization message will be send if RADIUS service is turn on and RADIUS Call Authorization is check in Extension. If the RADIUS return failed, the call will be rejected.
Appendix Attribu Attribute Description Form Example Name SIP: sip:user@ip:port User- 16 octets user password String Password 5.2.2 Authorization Response Message The RADIUS server could response the following attributes for authorization request. Attri Attribute Description Example bute Name A ID h323- The reason for failing Stin 0: Authenticated...
5.2.3 Start Accounting Message When a call is connected, the RADIUS billing start could be set to send to RADIUS server. The following is the Start Accounting Message which will be sent out. Attri Attribute NameVSA ID Description Example bute NAS-IP-Address IP Address of the In- 4 bytes unsigned long...
Appendix Attri Attribute NameVSA ID Description Example bute h323-connect- Connect time Strin yyyy/mm/dd hh:mm:ss time gw-rxd-cdn The called number as Strin 1002 received by the gateway in the incoming signalling message before any translation rules are applied. call-id SIP call ID kept for whole Strin call fdcnt...
Page 198
Attrib Attribute Description Form Example Name Acct-Status- Account Request Type Nume 2: Stop Accounting Type Service-Type Type of service requested Nume 5: Outbound h323-gw-id Name of gateway String SIP Proxy IP h323-conf-id GUID String xxxx h323-call-type Protocol type used on this String VOIP leg of the call - Telephony or...
Appendix Attrib Attribute Description Form Example Name fdcnt Forward Count String 0: normal call, 1: 1 st forward incoming-req- Incoming call leg request String sip:1001@192.168.1.1:506 SIP: sip:user@ip:port outgoing-req-uri outgoing call leg request URI String sip:1001@192.168.1.1:506 (after DM) SIP: sip:user@ip:port Acct-Session- A unique accounting String 8 bytes, like 12345678...
Page 200
Field Field Name Description Index Account A unique accounting identifier-match start & stop Session ID (Acct-Session-Id) Talk Time Call Duration (Acct-Session-Time) Disconnect The SIP caused code for a disconnected call (h323- Cause Code disconnect-cause) Incoming Leg incoming call leg INVITE received time (h323-setup- Setup Time time) Outgoing Leg...
Page 201
Appendix Field Field Name Description Index IP Type 0: IPv4 o IPV4 calls, 1: IPv4 to IPV6 calls 2: IPv6 to IPv4 calls, 3: IPv6 to IPV6 calls Additional reserved Parameters If you have turned on enhanced CDR, you will have more fields to descript the quality of calls as follows: Field Field Name...
Page 202
Field Field Name Description Index Called Audio Called Audio Octet Count Octet Count Called Audio Called Audio Payload size Payload size Called Audio Called Audio Lost Packets Lost Packets Called Audio Called Audio Mean Jitter Mean Jitter Called Audio Called Audio lost Rate lost Rate Called Peak Called Peak Audio Jitter...
Appendix Field Field Name Description Index Octet Count Called Video Called Video Payload size Payload size Called Video Called Video Payload size Lost Packets Called Video Called Video Payload size Mean Jitter Called Video Called Video Payload size lost Rate Called Peak Called Video Payload size Video Jitter...
Page 204
Modul Level Event NOTICE The IP x.x.x.x has been removed from blocking IP list. NOTICE SIP communication service NOTICE: (database connection resumed) NOTICE 01:35:47 (Registered) Extension: xxxx was registered from x.x.x.x:5060 (x.x.x.x:5060) NOTICE SIP communication service NOTICE: (service started) INFO User ID: XXX call attempt had over the max concurrent calls INFO...
Appendix Modul Level Event WARN HA Group 1 stopped at xxx NOTICE This node is set to ON-LINE by administrator. NOTICE HA Group 1 started at xxx (xxx is hostname) Digit Manipulation Example This appendix includes some digit manipulation examples for reference. Assumed that the following is the digit manipulation defined in the system.
Pilot Incomi Applie Applie Star Stop Repl Description d Ext. Posi Numbe Numbe Target Posi tion Valu r Type r Type tion DNIS caller Change 4th to 5th digit to 00 for DNIS when ANI's leading digit is 5 and length is 4.
Page 207
Appendix Pilot Number Screening Type 0204 block Extension 1001 had the following personal outgoing call screening group: Personal Pilot Number Personal Screening Type 00286 unblock The following is the calling example and result Called Number Result 002132342663 block the call 00286123456 allow to call 0091234567...
Page 208
Called Number Result 002132342663 allow to call 00286123456 allow to call 0091234567 allow to call 0204123456 allow to call 12345678 block the call Case 3: Assumed the following outgoing call screening group setting are assigned to extension 1003. Pilot Number Screening Type block 0204...
Appendix screening easy to be predicted. Call Processing Policy The system call processing policy helps administrator to understand the handling procedure on system point of view.
Appendix Extension Import Description The appendix is described the imported CSV file format. The tab is used to be used as a separator. The following is the field description for extension. [SIPPD_UserM ] Field DB Field Description Index Name UGroup_ID Extension Group ID User_ID Extension Number...
Page 212
Field DB Field Description Index Name 14. Hunting Extension 15. Send 181 before Start Forward 16. set SIP TO as request URI 17. Enable/Disable VMS 18. Enable/Disable "Disable Authentication qop tag" 19. Enable/Disable Anonymous Call Blocking 21. Enable/Disable Privilege Access 22.
Page 213
Appendix Field DB Field Description Index Name Locate_T1 Follow Me time period 1 (format: hhmm-hhmm) Locate_URI2 Follow Me's follow number for time period 2 Locate_T2 Follow Me time period 2 (format: hhmm-hhmm) Locate_URI3 Follow Me's follow number for time period 3 Locate_T3 Follow Me time period 3 (format: hhmm-hhmm) Locate_URI4...
Page 214
Field DB Field Description Index Name 2: WAN/UDP port 3 3: LAN/UDP port 1 4: LAN/UDP port 2 5. LAN/UDP port 3 6. IPV6 UDP port 7. TCP Port 8. TLS Port Transport_Typ Permanent contact address 2 transportation address, refer to Transport_type1 Device_1 Dedicate Device 1 Device_2...
Appendix Field DB Field Description Index Name Screen_Prefix Pilot Number Screen_Target Blocking Target (0: incoming call, 1:outgoing call) Screen_Type Blocking Type (0: Block, 1:Unblock, 2: Privilege Access) List of Used Network Ports The following is the list of used TCP/IP ports. The network administrator can use it to set the firewall when necessary.
SYSTEM -> VMS Settings -> Local Media UDP Start Port HA heartbeat SYSTEM -> High Available -> Cluster broadcasting port Service Port 5060 SIP TCP service SYSTEM -> SIP Service -> TCP Service port Port 5061 SIP TLS service SYSTEM -> SIP Service -> TLS Service port Port 9200...
Page 217
Appendix Step 3: Install it on your computer as a stand along application Step 4: Start up the Kiwi Syslog Server, you will see the following main screen: Step 5: Click File -> Setup and you should see the setup screen. Do the following settings: 1.
Page 218
Step 6: Right click in main windows and select Show/High columns, uncheck all items except 'message'. Step 7: Login to the system and click SYSTEM -> Debug. The following screen will display.
Page 219
Appendix Change the following: Syslog Debug: Enable Syslog Debug Serve IP: xxx.xxx.xxx.xxx (IP address you have installed syslog server) Change the required module's debug level to "Debug" Check the required module list for debug. Click Apply to save it. Step 9: Click COMMIT to start the syslog sending and start your testing. Step 10: You should able to see the debug log in the kiwi syslog screen.
Need help?
Do you have a question about the UniPBX-2000 and is the answer not in the manual?
Questions and answers