User s Manual VoIP Communication Server. Model No.: SP5210 Series

Transcription

1 User s Manual VoIP Communication Server Model No.: SP5210 Series

2 CONTENTS 1 INTRODUCTION ADVANTAGES FEATURES EXCHANGE 2007 INTEGRATION APPEARANCE DESCRIPTION QUICK START NETWORK SYSTEM TIME APPLY CHANGE CONFIGURATION REFERENCE SP5210 CALLING PROCESSING FLOW SYSTEM DEBUG SYSTEM LICENSE: AAA CUSTOM TIMEZONE HTTPS CERTIFICATE EVENT NOTICE BACKUP/ RESTORE SERVICE REFERENCE GROUP SUBSCRIBER UAC CALL ROUTING DIGIT MANIPULATION CALL INTERCEPTION RTP RESOURCE GROUP NAT GROUP DNIS SCREENING GROUP EMERGENCY CALL VOICE CODEC GROUP SYSTEM CONTROL REFERENCE SYSTEM SYSTEM TIME NETWORK SNMP ACCOUNT MANAGER PROVISIONAL IP SERVICE UPGRADE RELOGIN SYSTEM MONITOR REFERENCE SUBSCRIBER STATUS CALL STATISTICS

3 6.3 RTP STATUS RTP STATISTICS SERVER STATUS EVENT DEBUG INFO PING TELNET & RS-232 CONFIGURATION LCD DISPLAY CONFIGURATION APPENDIX 1 RETRIEVE CDR INFORMATION APPENDIX 2 SP5210 STATUS CODE APPENDIX 3 TIME ZONE TO COUNTRY MAPPING LIST

4 1 Introduction The Micronet SP5210 VoIP Communication Server provides a comprehensive, powerful platform for delivering IP telephony applications based on the Session Initiation Protocol (SIP). It offers call-control features to enable service providers to quickly and reliably deploy next generation packet-voice networks. 1.1 Advantages The Micronet SP5210 series SIP Telephony Server is the best choice to your convergence VOIP network which covert the requirements from enterprise to service provider. With built-in rich telephony services, SP5210 enables traditional PABX features to your VOIP convergence platform. Also you can easily upgrade the license or provide the high available service according to the growth of your business without any hardware changes. Intelligent Call Routing SP5210 provides multiple service routing polices to meet different service providers requirements (e.g. load balancing, priority, most idle etc.) It enables service provider to tell how to route the call depending on the call results or predefined rules. The incoming prefix match and outgoing prefix insert provides a very easy way to manage your VOIP exchange service. Easy to Configure and Management Full web management interface make you to manage your SP5210 anywhere of the world. You don t need remember the command lines or operate it on the specified console. Also the system event notice features keep you the system status updated remotely. NAT On-Demanded Traversal Due to the lack of IPV4 address, a lot of customer is using NAT for their network. SP5210 provides the NAT on-demanded traversal which will only route the voice when needed. It saves the bandwidth and provides better voice quality compare to route each call voice back to server. No CPE modification is required. Voice NAT/Firewall Router With built-in SIP and voice routing features, SP5210 provides a secure and easy way to migrate your Voice IP PBX solution. It acts as a NAT router and firewall role which voice RTP port is only opened when SIP signaling is established successfully. Rich Telephony Service The SP5210 provides build-in rich set of telephony service which enables the service provider quick time to market to delivery their service to their customers. Multiple Access to Receive Calls Anywhere With provided SIP TCP and UDP protocol, SP5210 can accept both type of signals and do the conversion when needed. For each protocol, SP5210 can support up-to 3 service ports which enabling to receive SIP service anywhere of world. Also SP5210 provides the SIP and RTP packets encryption/decryption features to breakthrough the ISR blocking of your SIP service

5 High Availability Redundant SP5210 provides high availability VOIP service by using active and stand-by redundant technologic which provides hot standby and hitless fail-over for stable call to reach mission-critical service requirement. It keeps your service continues running. Microsoft Unified Communication Server Integration SP5210 can work with Microsoft Live Communication Server as a total solution to meet the enterprise communication requirement. Without any extra settings in Live Communication Server, SP5210 connects your Office Communicator to PSTN and VOIP world. Also the SP5210 can become a perfect connecting between Exchange 2007 and PSTN/VOIP. 1.2 Features Selected Features: Call Transfer Call Forward Call Forwarded Notice Call Screening Caller ID Privacy Call Waiting Call Hold Call Pickup (Global, Group) Specified Call Pickup Find Me Short Code Do Not Disturb Miss Call Notify by ANI Replacement Call Return Hide ANI/Show ANI Selection Call Park/Retrieve Call Camp on Display Name Replacement PSTN Number Ring PSTN & IP Device Simultaneously Support Bandwidth Control based on Codec Allowance List Support Quality of Service (DiffServ and TOS) Support Automatic Subscribe for MWI and Presence Service Support Event Notice for Register or no Response Failure Support Reject Anonymous Call Service Support ENUM Support Busy Lamp Field (BLF) based on RFC 4235 Support Register to SoftSwitch (UAC) DID Support SIP TAPI Support Low Balance Announcement Support Call Routing Overflow Announcement Support Diversion Header for VMS/IVR Support Customer HTTPS Certificate - 5 -

6 Ready-to-Run Value Added Service System Announcement Service Multi-Company Auto Attendant Voice Mail Service Coloring Ring Back Tone Service Number Change Notice Call Forward Notice Call Forward Notice and Forward Call Interception Call Recording Service IP Centrex External RTP Resource Up-to 384 RTP channels extension for SP5210 Subscriber Selectable Automatically load balancing by SP5210 Application Examples: - VOIP Core Service - VOIP Class 5 Subscriber Service - Internet Exchange Service Center - Hosted IP-PBX - Medium to Large Enterprise IP-PBX - VOIP Call Center - Microsoft LCS 2005 Integration - 6 -

7 1.3 Exchange 2007 Integration Release Note of Version 3.0: - Support Bandwidth Control based on Codec Allowance List - Support Quality of Service (DiffServ and TOS) - Display Subscriber Status in Subscriber list - Support External DB (MS-SQL only) - Support Log Filter Based on ANI, DNIS or IP - Support Local Time Zone for Missed Call - Support Local Time RADIUS Format - Support Automatic Subscribe for MWI and Presence Service - Support Event Notice for Register or no Response Failure - Support Reject Anonymous Call Service - Support ENUM - Support Anonymous Incoming Call to Subscriber - Support Busy Lamp Field (BLF) based on RFC Support Server-based Support Register to SoftSwitch (UAC) DID - Support SIP TAPI from I(Outlook Integration) - Support Broadcasting Real Time Subscriber Status via UDP - Support Call Routing Overflow Announcement - Support Diversion Header for VMS/IVR - Support Customer HTTPS Certificate - Support Subscriber Contact Information - Support Enable/Disable SNMP & ftp - 7 -

8 1.4 Appearance Description SP5210 Front Panel: Functions: 1: Power LED 2: H/D LCD 3: System Status LED 4: LCD Panel 5: LCD Touch Panel 6: Power Switch SP5210 Rear Panel: Functions: 1: Electric Fan 2: AC Power outlet 3: AC Power switch 4: Keyboard/Mouse 5: Console port 6: SIP Service Ethernet port (WAN) 7: Management (Voice Gateway) Ethernet port (LAN) 8: VGA 9: USB (not used) - 8 -

9 2 Quick Start After connecting Ethernet cables into the SP5210 Management Interface & SIP interface, turn on the power. The first step is to logon the system and set up the IP address. Before you can use the browser to config SP5210, you need to install Java Plug-in before using subscriber status, call detail, debug info, remote terminal and upgrade. Please confirm your JRE version is (preferred & tested) if your PC has already installed Java. You also need to set newer versions of stored pages. Click Tool > Internet Option > General > Setting. After success, restart your browser to take effect. Logon SP5210 Setp1: Start IE6.0 (or later version) to navigate SP5210 web management system by typing the default URL is or the screen will display User ID and Password as figure

10 Figure Note: The default network IP address is: SIP Service Interface: Management Interface: Step 2: Enter login user name and password (the default user id is root and user password is root). You can manage your user account via web (refer to section Account Manager ) later. Figure Step 3: The screen shows the Home page of SP5210 as figure Network Figure SP5210 has 2 network interfaces: - SIP Service interface: This is the main service interface or WAN interface when voice gateway is ON. - Management interface: It is the LAN interface and should only be used when voice gateway mode is ON. It is recommended to not connect Ethernet cable when voice gateway mode is OFF. Step 1: After successfully logon to the system, we need to change the network configuration. Click Control > Network to setup the SIP Service Interface parameters as figure

11 Figure Step 2: Enter the deserved IP address, Submask and default gateway or selected to Use DHCP. Apply the change by clicking Apply button as figure Figure Step 3: When screen shows Change network configuration may cause server disconnected, are you sure? click on OK button to changes IP address as figure Figure Step 4: When screen shows After configuration changed, please relogin system with new IP address and execute Soft-Reset! click OK button as figure Figure Step 5: Follow Step 1 to 4 to change management interface network configuration as figure Figure

12 Note: Network Control takes around 5-second to apply the new network configuration. Please logon again with new IP address after 5 seconds. 2.2 System Time Step 1: When re-logon to the new IP address, the next is to setup the system time zone. Click Control > System Time Zone to setup the system. Enter the current date and time. Apply the change by clicking Apply button as figure Figure Step 2: If you would like to use SNTP to sync time with a SNTP V4 Server, click Time Sync button to setup it as figure Figure Step 3: After successfully base setup, restart SP5210 to take effect as figure Figure

13 2.3 Apply Change When you load a new working configuration or chang any configuration, you need click Apply Change to take effect as figure Configuration quick step Figure Please refer to the Configuration Detail Reference to do the system configuration setup as follows: - Setup SIP Domain if use DNS - Create subscriber or gateway - Setup Prefix hunting for gateway - Create required Digit Manipulation

14 3 Configuration Reference 3.1 SP5210 Calling Processing Flow Caller Validation Valid Start > Incoming call Invalid Call Reject Remove matched Prefix for G/W Emergency Call Yes Make E.C. call No Short Code Yes No Caller DM Invalid PSTN Number Valid Valid \ Invalid Called DM (ANI Only) Valid Caller DM Valid DNIS Screening Invalid Invalid Remove Add Prefix for G/W Call Reject Valid Subscriber Search Not Found Target Found ANI Screening Valid Invalid Target Found No Call Making RADIUS Call Permission Valid Prefix Hunting Yes RADIUS Validation Not Found Call Reject Invalid

15 3.2 System Start Path: System Core > System Figure Parameter Description: SIP Domain: Micronet Telephony SIP Proxy domain name. It s normally used when you have a DNS record setup for SP5210. Listen UDP Port, 2 and 3: The local UDP port on which the SIP service listened - Encrypt: The SIP signal and voice RTP will be encrypted while passing through the UDP port. TCP Enable: Enable the local TCP port or not Local TCP Prot, 1 and 2: The local TCP port on which the SIP service listened No Answer Timeout: The default maximum time (in second) to wait the remote party Answer (pick up phone). Max Forward Times: The maximum times to forward the calls Default Max Register Time: The default maximum register for public network user when a subscriber user is crated Default NAT Max Register Time: The default maximum register time for a inside user when a default subscriber user is crated Enable Device ACL: Authenticate specified device type or not. First Response Timeout: The default maximum time to wait for response. It s depended on the network speed. Subscriber Login: Enable Subscriber login to SP5210 or not Over Max Contact Rule: Over Max Contact Rule, reject or update. - Reject: The system will reject the new contact REGISTER request when the subscriber s used contacts reached the max contact - Update: The system will replace the oldest contact by new received contact. Support Video: Support video RTP proxy or not. Enable video will great reduce the number of concurrent RTP channel and bandwidth

16 Voice Gateway: Enabling voice gateway feature, SP5210 will be able to play the role as a NAT server to pass through SIP and voice call. Please refer to Voice Gateway Example for a configuration example. Password Number Only: When it is enabled, the web password only allows using dial pad (0-9). It is necessary to turn on this function when you are using privilege access code. Forward Caller ID: - Caller: use original caller ID when call is forwarded - Forwarder: use forwarder caller ID when call is forwarded to another user CPE Billing Enquiry: Enable the CPE billing enquiry or not. Accept Anonymous Call to Subscriber: Whether to allow the anonymous caller to dial to the subscriber. It is mainly used when ENUM support is enabled. Enable ENUM: Enable the ENUM or not. It is required to setup the DNS server in SIP service interface and input the correct ENUM Domain Suffix. ENUM Domain suffix: The suffix of the ENUM domain to be used for query ENUM address. QOS Type: Quality of Service Type - None: Not using QOS Tag - DiffServ: Differentiated Services Value - TOS: Type of Service. IP Precedence: Voice package priority setting. - Routine Precedence - Priority Precedence - Immediate Precedence - Flash Precedence - Flash Override Precedence - Critical Precedence - Internetwork Precedence - Network Precedence IP TOS: Type of Service with the following priority selection. - Normal Service - Maximize Reliability - Maximize Throughput - Minimize Delay Advance System Configuration: Start Path: System Core > System > Advance

17 Figure Advance Parameter Description: Call Validation Time: The timer to check periodically call is valid or not. SP5210 will send REINVITE periodically for call validation. NAT Compare Method: How to detect a NAT user - IP Only: Compare IP only - IP / Port: Compare IP and UDP port RetransmissionT1 (msec.): T1 determines several timers as defined in RFC3261. For example, when an unreliable transport protocol is used, a Client Invite transaction retransmits requests at an interval that start at T1 seconds and doubles after every retransmission. A Client General transaction retransmits requests at an interval that starts at T1 and doubles until it reaches T2. (Default Value: 500ms) ** RetransmissionT2 (msec.): Determines the maximum retransmission interval as defined in RFC3261. For example, when an unreliable transport protocol is used, general requests are retransmitted at an interval which starts at T1 and doubles until reaches T2. If a provisional

18 response is received, retransmission continue but at an interval of T2. (Default Value: 4000ms) ** RetransmissionT4 (msec.): T4 represents the amount of time the network takes to clear message between client and server transactions as defined in RFC3261. For example, when working with an unreliable transport protocol, T4 determines the time that UAS waits after receiving an ACK message and before terminating the transaction. (Default Value: 5000) ** Cancel General No Response Timer (msec.): When sending a CANCEL request on a General transaction, the User Agent waits cancel General No Response Timer milliseconds before timeout termination if there is no response for the cancelled transaction(default Value: 10000ms).** General Request Timer (msec.): After sending a General request, the User Agent waits for a final response general Request Timeout Timer milliseconds before timeout termination (in this time the User Agent retransmits the request every T1, 2*T1, T2, milliseconds)** Proxy 2xx Rcvd Timer (msec.): A successful client INVITE transaction of a Proxy server includes only the INVITE request and the 2xx response. (The ACK is not part of the transaction.)after receiving the 2xx response, the Proxy will wait proxy2xxrcvdtimer before the transaction terminates. (Default: ms)** Proxy 2xx Sent Timer (msec.): A successful server INVITE transaction of a Proxy server includes only the INVITE request and the 2xx response. (The ACK is not part of the transaction).after sending the 2xx response the Proxy will wait proxy2xxsenttimer before the transaction will terminate. (Default: 8000 ms)** Use Domain for Auth: Send Domain in 401 or 407 for authentication or not General Guard Time: The general guard time for internal purpose only Nonce Valid Period: The max valid time for a nonce. Once time out, SP5210 will issue a new nonce for authentication. Set it to 0 will cause SP5210 to generate new nonce for each call or register Valid Period Auth Mode: During the nonce valid period, need a subscriber send response on the register / invite message or not - None: User agent doesn t need send MD response in register invite message - MD: User agent should send MD response over the current nonce. Or a new nonce will be send by SP5210 Message Pool Page Size: Used to hold and process all incoming and outgoing in the form of encoded message or message objects. It is recommended that you configuration the page size to the average message size your system is expected to manage. General Pool Page Size: Used by SIP Stack objects, such as call-legs and transactions, to store the internal fields. For example, the call-leg object will store the To, Form and Call-ID headers and the local and remote contact addresses on the general pool pages. The general pool is also used for other activates that demand memory allocation. Send Receive Buffer Size: The buffer size used by the SIP Stack for receiving and sending SIP messages

19 Memory Pages: Number of memory page allocated. RTP Resource Timeout: The maximum time to wait for RTP server response. It s depended on the network speed. (only available when working with Micronet external RTP resource server) System Announcement: Used when personal announcement cannot be located (e.g. user not found). System Announcement URI: URI for system announcement server Announcement Prefix: Extra prefix to be added when personal or system announcement service is enabled. Invalid TTL Process: Response policy when register expired is too small. - Use Proxy TTL: Response proxy expires time to UAC and expect it will use it as default TTL. - Reject: Send 423 Interval Too Brief to UAC AAA Sending Stage: Send AAA message before or after DM Global Call Validation: Call Validation through both, none, caller or called (default is caller) Use Local Time: Use Local Time for RADIUS Accounting Message or not. It is recommended to use UTC time instead of Local time. Please make sure your RADIUS server can support local time (YYYY/MM/DD HH:MM:SS) format. Missed Call from Domain: The From host name for missed call - None: no host name or IP after from user name - IP: From host name is IP, xxx@xxx.xxx.xxx.xxx - Domain: From host name is SIP domain, Missed Call Tel No: The From User part for missed call - Tel no: The From user will be original called number - Replaced ANI: The From user will be the replaced number in the called subscriber service. Camp On Timeout: The max time to wait the camped on user to free the call. If it the max camp on is over and the called user is still busy, SP5210 will cancel the camp on silently. Web Call: Enable to use demo web call for subscriber or not. It requires the web call license from Micronet. Web Call ID: The demo web caller ID, you can get a number by click. Camp On Codec: The codec will be used for a camp on calls. It is required to use a most of common codec within your PABX. Change Call-ID: Whether change the call-id in the SIP message when forking a call out. Please contact Micronet Engineer when you want to change it. Dedicate UAC Failure: It is used when you are using dedicated UAC feature. If a subscriber is using the dedicate UAC ID and it is not registered, this option will decide which actions will be taken. Select Local Prefix Routing when you would like to use backup route. Select Disconnect when you are not allowed using backup route. BLF (RFC 4235) State Code: Busy Lamp Field State Code when report the subscriber status to Subscriber. - Unavailable: The default is void. - Idle: The default is terminated. - Ringing: The default is early

20 - Connect: The default is confirmed. Subscriber server-based 302: Enable Server based 302 Move or not. When it is enabling, the 302 will be processed by SP5210 instead of sending it to caller. ** SIP and network knowledge is required to change these parameters. Click the Database button to configure the system database: The system will allow using external DB (MSSQL only). Please contact Micronet for the MSSQL DB script. Start Path: System Core > System >Database Figure Parameter Description: System Database DataBase Type: The type of database which SP5210 used - Internal: This is the default settings to use the internal embedded database. - External MSSQL: To use external MSSQL database. Please contact Micronet if you would like to use it. DataBase Server: The database server IP address DataBase Name: The database name DataBase Port: The access port of database (default: 1433). It is not required to define the port when use OLEDB. DataBase User ID: The user ID to connect to DB. DataBase Password: Database user ID password Broadcasting You can set to broadcast subscriber status by using Micronet proprietary UDP format. Please use it under Micronet instruction and it might hit the overall system performance. Click the Broadcasting button: Figure

21 Parameter Description: Broadcasting: Enable the broadcasting function or not Broadcasting Interval (sec): The broadcasting interval to update the all registered subscriber. Broadcasting Port: The local port used for broadcasting Broadcasting IP: The broadcasted IP address. It could be or other broadcasting address. 3.3 Debug Debug can be turn on or off based on each system module and level to minimum the debug information. Please only turn on the debug information for debug purpose under Micronet FAE's instruction and turn off when complete. Or the system performance will be greatly hit. Start Path: System Core > Debug

22 Figure System License: Start Path: System Core > License Figure License Parameter Description: Feature: System parameter Serial No: System parameter License Key: System parameter Note: Please don t change it unless under Micronet s instruction 3.5 AAA When the subscriber users do the AAA (Authorization, Authentication and Accounting), enter the correct parameter the Radius setting. Start Path: System Core > AAA Figure Parameter Description: Authorization IP: Radius Authentication/Authorization Server IP address Authorization Port: Radius Authentication/Authorization Server Port Accounting IP: Radius Account Server IP address Accounting Port: Radius Account Server Port Backup Authorization IP: Backup Radius Authentication/Authorization Server IP address

23 Backup Authorization Port: Back Radius Authentication/Authorization Server Port Backup Accounting IP: Back Radius Account Server IP address Backup Accounting Port: Back Radius Account Server Port Max Retry: The maximum retry times Response Timeout (msec): The maximum wait for response time from RADIUS Server Switch Threshold: Switch to alternate RADIUS Server when failures are occurred more than switch threshold. Local Port: The local port used for RADIUS Client Secret Key: The shared secret key with RADIUS Server CISCO Mode: - Yes: Use Cisco RADIUS mode (have redundant string in vender attribute) - No: no CDR Mode: - Enable: Log CDR into the file - Disable: no CDR Keeper Days: CDR system keeping days Vendor ID: RADIUS vender attribute s vender ID.(Default is 9) Send Zero Session Time: - Yes: Send 0-balance session time for RADIUS when the call failed - No: no Inter-Subscriber RADIUS Authentication: - Yes: When a subscriber is calling another subscriber, SP5210 will send RADIUS for call permission - No: When a subscriber calling another subscriber, SP5210 will not send RADIUS for call permission Inter-Subscriber RADIUS Billing: - Yes: Send RADIUS billing message for Inter-Subscriber calls - No: Do not send RADIUS billing message for Inter-Subscriber calls Billing Message: Send RADIUS billing message out Micronet extend RADIUS attrib: Enable the Micronet extended RADIUS attributes or not. 3.6 Custom Timezone This is the place to define the customized time zone for subscriber. It is mainly used for missed call notify purpose. Start Path: System Core > Custom Timezone > New

24 Figure Parameter Description: Timezone ID: The customized time zone ID Time Bias: The offset from GMT time zone Auto Daylight Saving: Auto adjust daylight saving time or not Daylight Bias: The offset added to the standard Bias when the time zone is in daylight saving time Daylight Start: The date that a time zone enters daylight time - Month: 01 to 12 - Week Day: Sunday to Saturday - Apply Week(Day:01 to 05,Specifies the occurrence of day in the month;01=first occurrence of day,02=second occurrence of day, and 05 = Last occurrence of day) - Hour:00 to 23 Standard Start: The date that a time zone enters daylight time - Month: 01 to 12 - Week Day: Sunday to Saturday - Apply Week(Day:01 to 05,Specifies the occurrence of day in the month; 01 = First occurrence of day,02 = Second occurrence of day, and 05 = Last occurrence of day) - Hour:00 to 23 Description: The description Click the Apply button to take effect as figure Figure

25 3.7 HTTPS Certificate Secure Hypertext Transfer Protocol Certificate, it can enhance the web page security. The user can put their owned HTTPS SSL certificate here. Start Path: System Core > HTTPS Certificate Figure Parameter Description: SSL Certificate file name: The encrypted certificate file name in SSL layer SSL Certificate Password: The password for the SSL certificate. It required to be gotten from the SSL certificate company. 3.8 Event Notice Server event notice method setup Start Path: System Core > Event Notice Figure Parameter Description: Enable System Log: Enable to send system information to syslogd Server or not SyslogD Server IP 1, 2: syslogd server IP address SNMP Sending Interface: The SNMP sending interface Event Notice: Enable the event notice or not SMTP Event Filter Level: The level of filter SMTP Server: SMTP server host for notice

26 From: sender account To: receiver (semicolon is used for multiple receiver) Subject: subject to be send to receiver. The following variable parameters can be used to create dynamic subject for system notice: - $LOGLEVEL$: Information Level - $HOSTNAME$: Host name - $HOSTIP$: Host IP address Auth-type: The authentication type for the SMTP server 3.9 Backup/ Restore Backup/ Restore provides a way to backup and restore the working configuration here. Backup the working configurations: Step 1: To backup the running configuration, click the Backup Configuration to backup to local hard disk as figure Figure Step 2: The whole running configuration will be compress into a zip file (file name: export.zip) and transfer back to local as figure Restore Configuration: Figure Step 3: To restore the backup configuration file, click Restore Configuration as figure

27 Figure Step 4: Select backup file (i.e. c:\export.zip) and click Apply button to restore the configuration to the working configuration as figure Figure Note: It is need to restart the system to take effect of the new-restored working configuration

28 4 Service Reference 4.1 Group Each user group can have its owned access code and related settings to minimum the management effort. Start Path: Service > Subscriber > Group Figure Click the user group you want to modify: Figure Parameter Description: Closed Group: Enabled for in-group subscriber to subscriber call only. To call to another group need to get thought prefix hunting. User Group ID: User Group ID DM Group ID: Group-winded digit manipulation applied Description: user group description SMTP Host: SMTP server host (i.e. mail.micronet.com.tw) for delivery missed call message Miss Call Subject: Missed call notify subject

29 You can have the following variables for notify subject. $FROM$: caller party number $TO$: Called party number $UTCTIME$: UTC Time $LTIME$: Local Time $DOMAIN$: SIP Domain $HOSTIP$: Host IP address For example: You have a missed call from $FROM$ at $LTIME$ CRBT Prefix: Extra prefix to be added when Coloring Ring Back Tone service is enabled. Announcement Prefix: Extra prefix to be added when Announcement service is enabled. VMS Prefix: Extra prefix to be added when VMS service is enabled. Enable MWI: Enable MWI Service or not. MWI server subscriber ID is required. Enable Presence: Enable Presence Service or not. Presence Server subscriber ID is required. Call Park: Enable Call Park or not - Park Source: The announcement server to play the music after call part for first party. - Call Park Location: The Call Park Location starting code (e.g. 800, and the system will automatically add to 809, 10 locations in all.) It cannot be conflict with subscriber or prefix. Billing Announcement URI: Click the Detail button: It is Service code definition for the selected user group. Figure Parameter Description: Service Code: Telephony Keypad used for the service code Service Type: Applied service type The others please refer to the examples below: Forward Service: Service Access Code Parameter (optional) Example

30 Enable unconditional forward *201 Forward number Enable no answer forward *202 Forward number Enable busy forward *203 Forward number Enable unavailable forward *204 Forward number Enable don t disturb *205 Don t disturb time 1 (hhmmhhmm) Don t disturb time 1 & 2 (hhmmhhmmhhmmhhmm) * *201 (use existing setting) * *202 (use existing setting) * *203(use existing setting) * *204(use existing setting) * * *205(use existing setting) Enable Notify *206 n/a *206 (need pre-config by web) Enable Fine Me *207 n/a *207 (need pre-config by web) Enable CRBT *208 n/a *208 (need pre-config by web) Enable Announcement *209 n/a *209 (need pre-config by web) Enable VMS *210 n/a *210 (need pre-config by web) Disable unconditional *301 n/a *301 forward Disable no answer forward *302 n/a *302 Disable busy forward *303 n/a *303 Disable unavailable forward *304 n/a *304 Disable don t disturb *305 n/a *305 Disable Notify *306 n/a *306 Disable Fine Me *307 n/a *307 Disable CRBT *308 n/a *308 Disable Announcement *309 n/a *309 Disable VMS *310 n/a *310 Hide ANI Service: Service Access Code Parameter Example Hide ANI *314 Dialed number * (Hide caller ID) Show ANI *214 Dialed number * (Show caller ID) Pickup Call Service: Service Access Code Parameter Call Pickup Global Pickup *0 n/a *0 (global pickup) Group Pickup *1 n/a *1 (group pickup) VAD Service: Service Access Code Parameter Call Pickup Camp On * * Cancel Camp On *311 n/a *311 (cancel camp on) Call Return *212 n/a *212(No answer call return) Call Pickup * *213301(assigned call pickup) Call Park *215 Enable Privilege Access * (user s password) *216 (enable privilege access) Disable Privilege Access *316 n/a Disable Call Waiting *217 n/a *217 Enable Call Waiting *317 n/a *317 *316 (disable privilege access)

31 Disable Caller ID *218 n/a *218 Enable Call ID *318 n/a *318 Set CTI Status* *219 n/a *219 * Set CTI Status can be only used on ICCS model. Click the Pickup button: Grouping the subscribers for group pickup service, you can set a subscriber to belong to a pickup group. Figure Parameter Description: Pickup Group ID: Pickup group ID Description: Description 4.2 Subscriber Start Path: Service > Subscriber Figure Modify: Click the subscriber you want to modify:

32 Figure Parameter Description: Configuration Active Mode: The subscriber user is active or inactive TEL No: Register TEL no User Account: Register used ID User Password: Register user password (device password only) Web Password: Password used only for web access only User Group: Belonged User group Authentication Mode: Authenticate subscriber by MD or not - None: None - Register Only: Authenticate subscriber only for register - Register Invite: Authenticate subscriber for register and each call DNIS Screening Group: DNIS screening group Call Authorization Mode: Send authorization to Radius server or not Emergency Group: Emergency call group Call ID Mode: Displace caller ID or not - Inhibit: Hide the called party number - Transparent: Pass through the caller ID Device Type: Subscriber device type - Subscriber: Subscriber user - Gateway: Gateway (e.g. trunk gateway or FXO gateway) - Pay Phone: The public coin telephone (reserved type). Please contact Micronet for your application. - Gateway/RTP: SP5050/SP5064,SP5002/SP5004/SP5052A/SP5054A/SP5012/SP

33 - Proxy/RTP: Micronet SP SIP Proxy: SIP proxy server - IVR/VMS: IP IVR or VMS server - IVR/VMS/RTP: Micronet IVR or VMS server - Recorder: Micronet Recorder - Outbound Caller: Outbound Caller - Register UAC: Register user agent client User Agent ID: User agnet ID in UAC Caller Info: Display calling parting information Caller TEL No: Display original caller ID Registered TEL No: Registered UAC user ID Caller Display Name: SIP display name for original caller - LCS Server: Microsoft Live Communication Server - MWI Server: MWI Server - Presence Server: Presence Server - RTP Server: RTP Server - VMS (Diversion): The voice mail server which support diversion header as draft-levy-sip-diversion-08.txt. - Web Caller: Allow unregistered subscriber to make call. Web Caller license is required. Micronet will provide OCX and sample code for integrating into the customer s Web server. - Exchange Fax: Allow to provide the fax feature for Exchange Regular T.38 device can work with Exchange 2007 UMS when using this exchange fax user. - Softphone: It is Micronet soft-phone device type. - Diversion/VMS/RTP: - Gateway/RTP/Fax: Hunting Method: Call forking method. - Sequential: Call hunting each contact in sequence - Parallel (answer): Send multiple call invites to multiple contacts simultaneously. When a user pickup the phone, disconnect others contacts. Preferred RTP Group: Preferred RTP resource server group to be used. Register Type: Subscriber register type - Dynamic: Subscriber need send register message for availability - Predefine: Subscriber will be handle as a permanent user Predefine URI1: Predefine subscriber URI1 (i.e. sip:9001@ ) Predefine URI2: Predefine subscriber URI2 (i.e. sip:9001@ ) - Predefine/NAT: Subscriber will be handled as a permanent NAT user (manual IP/Port mapping is required) Predefine URI 1: Predefine NAT subscriber URI (i.e. sip:8001@ : 7777) Public TA: mapped NAT Server IP address and port (i.e :5060) - LCS Dynamic: This is used only when the subscriber type is a regular subscriber and also a LCS user. You can use IP phone or CPE to register this account and also it will ring the LCS communicator client software simultaneously

34 RTP Proxy: Use RTP Proxy or not - Yes: Always use the RTP Proxying - No: Always not use RTP Proxying - Auto: Automatic decide to use RTP Proxy or not (recommended) - Recorder: Use Recorder - Recording on Demand: Use Recorder on demand service is required) - Auto (n-nat): This is only used when you have a NAT box behind a NAT box. It is mainly for testing or a special environment. NAT Group: NAT group can be used for enterprise user. When two subscribers have same NAT group defined, SP5210 will not use NAT Proxy when both subscriber have same NAT group. Max Register Time: The maximum register time when a user is coming from public network Max Register NAT Time: Time: The maximum register time when a user is sited behind NAT First Response Time: The maximum time to wait for response. It depends on the network speed. No Answer Timer: The maximum time (in second) to wait the remote party answer (pick up phone). Max Contact Allowed: The maximum contact allowed for a subscriber. The new contact will not able to register when old one doesn't free up. Pickup Group: Pickup group for subscriber Device_1, 2: Allowed device to be connected to SP5210. Max Concurrent Call: The maximum of concurrent call Call Validation: The call validation type: none, update or invite - None: disable call validation features - Update: Use SIP UPDATE instead of INVITE - Invite: Use SIP INVITE message for call validation Over Max Contact Rule: Over Max Contact Rule: reject, update or use Global Setting (Configuration > System) - Reject: The system will reject the new contact REGISTER request when the subscriber s used contacts reached the max contact - Update: The system will replace the oldest contact by new received contact. - Global Setting: Use system defined policy. AAA Sending Stage: Send AAA message before or after DM. Or use Global Setting (System > Advance) Codec Group: Belonged codec group. When you select a codec group, only selected voice codec can be get through. The others will be filtered. It is very useful when you want to limit your users codec. Dedicate Outgoing UAC: When enable it, the subscriber will use the selected UAC for making outgoing call. You can use PSTN number together to provide a DID and DOD feature. Assign a special UAC device for outgoing calls Effective Period: The subscriber effective period (Format: yyyymmdd - yyyymmdd) Remove Tag for Cancel: When cancel the call, remove the to tag (for CISCO device only)

35 Disallow Register From NAT: Enable this option will not allow a subscriber to register behind NAT. In other words, this subscriber will never consume the RTP resource. Sync Web Password: Sync web password per subscriber Timezone: Set the time zone (for missed call notice). You can select the custom time zone by clicking. - Auto Daylight Saving: Auto adjust daylight saving timer or not Information User Name: The user contact name Contact TEL: The telephone number to contact to the subscriber Address: The address of the subscriber Address: The subscriber live address Description: Description Service: Click the Subscriber > Service you want to modify telephony service for the selected user. Parameter Description: Figure Forward Service: Forward Subscriber Only: Forward to proxy subscriber only

36 No Answer Forward: Forward to the URI when the subscriber has no answer. Unconditional: When enabled, any calls to this subscriber will be forward to this URI unconditionally. You can use SIP URI or subscriber ID here. - Disable Call Originate: When enable unconditional forward, the user will not able to make call out if it is checked. Unavailable Forward: Forward to the URI when the subscriber is unavailable (not registered). You can use SIP URI or subscriber ID here. You can use SIP URI or subscriber ID here. Busy Forward: Forward to the URI when the subscriber is busy. You can use SIP URI or subscriber ID here. You can use SIP URI or subscriber ID here Announcement Before Forward: Play Announcement before call forward 181 for Call Forward: Send SIP181 for Call Forwarded service Number Change: Change the original number to a new number. Find Me: Locate subscriber based on different time segment when the original (registered) contact cannot be reached. - Find Me Hunting First: Hunting find me contact first. - Hunting Subscriber: Applicable only for Find me hunting checked will hunt subscriber registered contact when find me can t be reached. Auto Call Forward: Auto Call forward or not after number changed Call Pickup: Group Pickup (Picked): Allowed to be picked-up within group or not Global Pickup (Picked): Allowed to be picked-up globally or not Screening Service: Personal ANI Screening: Personal ANI screening can be used to filter the caller based on caller ID. For all TEL are set to allow (full match), only onlist ANI can get through. For all TEL are set to disallow, only on-list ANI will be screened. Otherwise, disallow has higher priority than allow. Personal DNIS Screening: Personal DNIS screening can be used to limit the called prefix. For all TEL are set to allow (prefix match), only on-list DNIS can get through. For all TEL are set to disallow, only on-list DNIS will be screened. Otherwise, disallow has higher priority than allow Do not disturb: Up-to 2 time segments can be set to reject all incoming calls. VAD Service Coloring Ring Back Tone: Coloring Ring Back Tone service VMS: Voice Mail Service Server Hold Tone: Enable Server Hold Tone or not. The server hold tone requires an external announcement service. You also need to define the CRBT prefix in the user group when the URI subscriber is a regular announcement. For a diversion supported device, you can ignore the CRBT prefix definition. Announcement Service: The announcement service requires an external announcement service. You also need to define the Announcement prefix in the user Group when the URI subscriber is a regular announcement. For a diversion supported device, you can ignore the Announcement prefix definition

37 Subscriber Service ANI Replacement: Replace calling number for Gateway or all subscriber - Replace ANI: Replace calling number - Replace Type: Gateway only or all subscribers - PSTN Number: This number will be handled as a PSTN number. It will work like you have a second number in SP5210. SP5210 will look at the PSTN number first. Short Code: short code to be used within same group Display Name: Assigned the display caller name for the subscriber. It will be showed on SIP IP phone. Disable call forward display name: Add display name as subscriber id in SIP from header when call forward caller mode is "forwarder". Support Video: Enable Support Video or not Missed Call: Missed call notify service Disable Call Waiting: Disable call waiting feature. When disable call waiting features, the second incoming call to the user will be rejected by SP5210. Disable Conference Call: Disallow to call a conference call Server Transfer: The server will do the transfer instead of send to CPE. It is recommended to use it only when CPE doesn t support call transfer features. It is only happened when the user is transferred party. Security Disable Un-Register All: Disable Un-Register all (use * to un-register all contacts) Disable RADIUS Billing Send: Disable RADIUS Billing Send Reject Anonymous Call: Reject the anonymous incoming call or not Misc Sync to Address: Make SIP TO head to be same as Request URI Response to Sending Port: Response to CPE sending port instead of register port. Response to Top Via: Response to Top Via instead of register port CTI: Computer Telephony Integration (reserved item) Auto Subscriber for MWI: Automatically subscriber MWI service to MWI server when the user is registered. Disable Qop: Disable sending qop tag in SIP 401 and 407 authentication header. Auto Subscriber for Presence: Automatically subscriber Presence service to Presence Server when the user is registered. Register Event Log: The system will trigger a system event when the subscriber is registered and unregistered. For a predefine user, the system will also fire a system event when the predefined user is no response and responding. Parameter Button Copy: Copy service setting from a subscriber Mask: Set the subscriber visual view of the service. If you uncheck the mask of a service, the subscriber login will not able to see it

38 Search: Click the search button and you can search the item you want: Figure Parameter Description: User ID: The subscriber user ID to be searched User Name: The subscriber user name Contact TEL: The subscriber contact telephone number User Group ID: The group ID that the subscriber user belonged to Device Type: The device type of the searched subscriber 4.3 UAC SP5210 can register to another proxy server as a standard SIP UAC (User Agent Client). You can have hieratical SIP proxy architecture by using UAC settings. Modify: Click subscriber > UAC you want to modify: Figure Parameter Description: User Agent ID: Identifier used for subscriber setting (type UAC) Register ID: SIP registrar user ID Register TEL No: SIP registrar telephone number

39 Register Password: SIP registrar user password SIP Domain: SIP registrar domain Register IP: SIP registrar IP address Register Port: SIP registrar UDP port number Register TTL: The registration maximum time to live setting when registered to the SIP registrar Outbound Proxy User ID: SIP outbound proxy server user ID Outbound Proxy Password: SIP outbound proxy server user password Outbound Proxy IP: SIP outbound proxy server IP address Outbound Proxy Port: SIP outbound proxy server port number Description: Description Encrypt: The device for UAC is encrypted or not. To set a subscriber as a register client, choose register type to "register UAC". Then you can use this subscriber Tel number for prefix hunting. 4.4 Call Routing SP5210 call Routing can provide prefix hunting base on priority, max idle time or round robin method. SP5210 will use call routing plan to do the corresponding routing. The routing target can be a UAC (register client), another proxy, gateway or subscribers...etc. Routing policy is defined here. Start Path: Service > Call Routing Modify Call Routing List: Click the Modify button: Figure

40 Figure Parameter Description: Active Mode: The prefix group is active or inactive Prefix Matched: Called number prefix to be matched Description: Description Matched Length: Applied only when specified length of DINS is matched. Zero (0) indicate ignore length option. Matched User Group: Applied only for specified user group. Others group will not be applied. Hunting Method: Hunting method used for this group - Round Robin: Call is hunting rotationally until user answer - Priority: Call is hunting base on priority set until user answer - Max Idle Time: Max idle one will be hunt first until user answer - Ring All (First Ring): Send request to all members. When a user response ringing, cancel the others request. - Ring All (First Answer): Send request to all members. When a user pickup the phone, cancel the others request. - Round Robin (Ring Only): Send request based on round robin member selection. Stop hunting when a user response ringing. - Priority (Ring Only): Send request based on member's priority. Stop hunting when a user response ringing. - Max Idle Time (Ring Only): Send request base most idle policy. Stop hunting when a user response ringing. - Round Robin (Load Balance): Send request based on round robin member selection. Stop hunting when a call is failed except receiving the defined reason code in load balance reason. - Priority (Load Balance): Send request based on member's priority. Stop hunting when a call is failed except receiving the defined reason code in load balance reason. - Max Idle Time (Load Balance): Send request based on max idle policy. Stop hunting when a call is failed except receiving the defined reason code in load balance reason. - ACD Route: It is used only for ICCS model. - RADIUS Route: Let the RADIUS to decide the routing. It is mainly used for least cost routing or QOS routing

41 - ENUM: This routing is use ENUM DMS server to decide the routing. You have to set the corresponding ENUM DNS in the Networking settings. No Answer Time Out: The maximum time (in second) to wait the remote party answer (pick up phone) First Response Time Out: The maximum time to wait for device response. It s depended on the network speed. Announcement URI: This is the place to define an announcement service when the routing is not able to get through. It could be an IVR or IVR which support Diversion header. URI for announcement server Remove Prefix: Remove prefix matched or not RADIUS Authorization Resend: Send RADIUS authorization for each prefix hunting. Click Detail to define member of call routing group. Click Detail button: Figure Parameter Description: TEL NO: Subscriber TEL no for route Priority: Used only for priority hunting. Click the Reason button: When enable load balance hunting, here is the cause reason to enable SP5210 to continue the hunting. Figure Parameter: State Code: SIP State code to continue the hunting

42 4.5 Digit Manipulation SP5210 Digit Manipulation can provide operator target called number and calling number to insert, replace or drop. Start Path: Service > Digit Manipulation Click the Detail button: Figure Figure Modify Digit Manipulation: System is able to execute 1 ANI DM and 1 DNIS DM separately for calling and called party. The default will be the caller which is same as the old version. Matched ANI DM will be executed first and use the result for DNIS DM. Click the Modify button: Figure