Telephony Sip Proxy Server - Justec International Technology Inc.

Transcript

Telephony SIP Proxy Server User Guide Release2.1 V2.1 -1- CONTENTS Chapter 1 Telephony SIP Proxy server Introduction....................................................3 Multiple Advantages........................................................................................................3 Telephony SIP Proxy server V2.1 Features ...................................................................5 Telephony SIP Proxy server Appearance Description..................................................10 Chapter 2 Telephony SIP Proxy Quick Start................................................................. 11 Network.........................................................................................................................13 System Time .................................................................................................................14 Apply Change ...............................................................................................................15 Chapter 3 Configuration Reference ..............................................................................16 Telephony SIP Proxy Server Calling Processing Flow .................................................16 System..........................................................................................................................17 Debug ...........................................................................................................................23 Group............................................................................................................................24 Subscriber.....................................................................................................................28 UAC ..............................................................................................................................34 NAT ...............................................................................................................................35 RTP Resource ..............................................................................................................36 Call Interception............................................................................................................37 Prefix Route ..................................................................................................................37 Digit Manipulation .........................................................................................................40 DNIS Screening ............................................................................................................43 Emergency Call ............................................................................................................44 AAA...............................................................................................................................45 Configuration Manager .................................................................................................46 Chapter 4 System Control Reference ...........................................................................48 System..........................................................................................................................48 System Time .................................................................................................................48 Network.........................................................................................................................50 SNMP ...........................................................................................................................52 Account Manager .........................................................................................................53 Provisional IP................................................................................................................53 Upgrade ........................................................................................................................54 Relogin..........................................................................................................................54 Chapter 5 System Monitor Reference...........................................................................56 Subscriber Status..........................................................................................................56 Call Statistics ................................................................................................................58 RTP Status....................................................................................................................59 RTP Statistics................................................................................................................59 Server Status ................................................................................................................60 Event Log......................................................................................................................61 Debug Info ....................................................................................................................62 Ping...............................................................................................................................63 Chapter 6 Telnet & RS-232 Configuration ....................................................................64 Chapter 7 LCD Display Configuration ..........................................................................71 Appendix 1 Retrieve CDR Information..........................................................................74 Appendix 2 THE SIP TELEPHONY PROXY SERVER Status Code .............................76 Appendix 3 Time zone to Country Mapping List .........................................................77 V2.1 -2- Chapter 1 Telephony SIP Proxy Server Introduction The Telephony SIP Proxy 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 packetvoice networks. Multiple Advantages Intelligent Call Routing The Telephony SIP Proxy Server 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 Telephony SIP Proxy 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. The Telephony SIP Proxy Server provides the NAT ondemanded 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, The Telephony SIP Proxy Server 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 Telephony SIP Proxy Server provides build-in rich set of telephony service which enables the service provider quick time to market to delivery their service to their customers. By cooperating to IPCentrex, the service provider can provide Announcement, Auto Attendant, VMS, CRBT etc. immediately. Multiple Access to Receive Calls Anywhere With provided SIP TCP and UDP protocol, The Telephony SIP Proxy Server can accept both type of signals and do the conversion when needed. For each protocol, The Telephony SIP Proxy Server can support up-to 3 service ports which enabling V2.1 -3- to receive SIP service anywhere of world. Also a proprietary voice and SIP encryption can break through all ISP blockings. High Availability Redundant The Telephony SIP Proxy Server 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 missioncritical service requirement. It keeps your service continues running. Microsoft Unified Communication Server Integration The Telephony SIP Proxy Server 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, The Telephony SIP Proxy Server can connect your Office Communicator to PSTN and VOIP world. Also the The Telephony SIP Proxy Server can become a perfect connecting in between Exchange 2007 and PSTN/VOIP. V2.1 -4- The Telephony SIP Proxy Server V2.1 Features No. A. 1 2 3 4 5 6 B. 1 C. 1 2 3 D. 1 2 3 4 5 6 E. 1 2 3 4 F. 1 2 3 4 5 6 7 G. 1 2 3 4 5 6 7 8 9 Features Physical Dimensions Width Height Depth Industrial rack mount Color Weight Certification CE Power / Environmental Power Operating temperature Relative humidity Processors & Storage DSP vendor Operation System RAM Program/Data Storage OS Upgradeable Program Upgradeable Front Panel Display Power LED Dom Access LED System Ready LED LCD Status Network Interface 10/100/1000 Base Ethernet IP Address Required Fixed IP DNS Dynamic DNS Voice Router Mode (Voice NAT) Static Route for Private IP Leg Standard Protocol Support RFC 3261 RFC 2976 - SIP INFO RFC 3262 - PRACK draft-ietf-sip-cc-transfer-05 RFC 2327 - SDP RFC 2833 - DTMF RFC 3264 Offer Answer Model with SDP RFC 3265 Specific Event Notification RFC 3325 - Private Extensions to SIP V2.1 The Telephony SIP Proxy Server V2.1 483mm 44mm 450mm Yes black 6Kg Yes 90-240V auto switch 0~60 C 5%~95% Intel Pentium 4 XP Embedded 1024 MB 512 MB DOM Yes Yes Yes Yes Yes Yes 2 SIP Service & Management Yes Yes Yes Yes (NAT/Public Legs) Yes Yes Yes Yes Yes Yes Pass Through for NAT Yes Yes Yes -5- 10 11 12 13 14 15 16 17 18 H. 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 I. 1 2 3 4 5 6 7 8 9 10 11 12 J. 1 2 3 4 5 6 RFC 4028 - Session Timers SIP UDP Support SIP TCP Support SIP URI Support SIP Service Port Multiple SIP UDP/TCP Service Port SIP UDP/TCP Transversal MWI/Presence Support Conference URI SIP Registrar Dynamic Register Predefine User Predefine NAT User Register Authentication Nonce Live Time Authentication Check Period General Max Register Time Max NAT Register Time Register Type Selection Register to External Soft switch or Proxy Separate NAT/Normal Register TTL Contact Replacement Policy Unregister all Contacts Effective/Expired Date Global or Subscriber based Register TTL Device ACL Subscriber based Device Control SIP Outbound Proxy Server Stateful Proxy Server Call-based Authentication Multiple Contacts Sequential Call Forking Parallel Call Forking Proxy Peering Support Hierarchical Proxy Support Subscriber/System based Response Timer NAT User Detect & Transversal Max Concurrent Calls per Subscriber Incoming Call Prefix Match Outgoing Call Prefix Add Telephony Features Call Transfer Unconditional Call Forward No Answer Call Forward Busy Call Forward Unavailable Forward Calling Number Screening (Allowance List) V2.1 Yes Yes Yes Yes Programmable Yes Yes Pass Through Pass Through Yes Yes (up to 2 URI) Yes MD5 Programmable Programmable Programmable Programmable Yes Yes (multiple) Yes Yes Yes Yes Yes Yes Yes Yes MD5/RADIUS Up to 5 Yes Programmable ring or answer Yes Yes Yes (no answer, first response) Auto Yes Yes & removable Yes Yes (Client-based) Yes Yes Yes Yes Yes (subscriber/group) -6- 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 K. 1 2 3 4 5 6 7 8 9 L. 1 2 3 4 5 6 7 8 9 10 M. 2 Calling Number Screening (Black List) Outgoing Number Screening (Allowance Prefix) Outgoing Number Screening (Black List) Digit Manipulation for ANI Digit Manipulation for DNIS Flexible Digit Manipulation per User Group Caller ID Privacy Call Waiting Call Hold Emergency Call Global Call Pickup Group Call Pickup Specified Call Pickup Find Me Short Code Do Not Disturb Miss Call Notify by Email ANI Replacement Telephone Keypad Setup for Service Call Return Hide ANI/Show ANI Prefix Call Park/Retrieve Subscriber based Display Name Replacement Call is been Forwarded Notice (181) Call Hunting Round Robin Group Hunting Priority Group Hunting Max Idle Time Group Hunting Ring All with First Ring Back Group Hunting Ring All with First Answer Group Hunting Load Balance Hunting for Gateway Load Balance Hunting based on Return Code Group Hunting with Dedicated Length User Group Based Hunting Special Enhanced Service Coloring Ring Back Tone Service Announcement Service Voice Mail Outbound Call/DTMF Integration Number Change Notice Call Forward Notice Call Forward Notice and Forward Call Interception (DNIS or ANI Matched) Call Interception (Subscriber Based) Recording on Demand Computer Telephony Interface CTI Attached Data V2.1 Yes (subscriber/group) Yes (subscriber/group) Yes (subscriber/group) Yes Yes Yes Yes (Server Base) Client based Client based ANI Base Yes Yes Yes Yes (up to 5 contacts) Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes External resource is required Yes (IP Centrex ) Yes (IP Centrex ) Yes (IP Centrex ) Yes (SIPIVR ) Yes (IP Centrex ) Yes (IP Centrex ) Yes (IP Centrex ) Yes (WellRec 5600) Yes (WellRec 5600) Special CPE and WellRec 5600 is required Contact Justec for available Yes -7- N. 1 2 3 4 5 6 7 8 9 10 11 O. 1 2 3 4 5 6 7 8 P. 1 2 3 4 5 6 7 8 9 Q. 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 NAT Traversal NAT Traversal for Outbound and Inbound Automatically NAT detection and RTP Proxy NAT Partition Support Intelligent RTP Proxy Resource Management External RTP Proxy Resource Support Preferred NAT Proxy Group None NAT User only RTP Resource Group Hunting Subscriber based Video NAT Enable Behind NAT Predefined Session Boarder WellBG 5800 Support AAA Flat CDR File in Local Storage Real Time RADIUS Billing Message Support Support Redundant RADIUS Server Authorization Message for Prepaid Service Real Time Call Disconnect for Prepaid User On-Net Call Billing Disable RAIDUS Accounting Sending Selectable DM before Authorization or not Subscriber Management Subscriber Access Control Separate Web Password Subscriber only Parameter change User Group Provisional Interface Pickup Group Synchronize Web Password and User Password Unregister from Web GUI Free Calls from Web GUI Maintenance RS232 Console Port Telnet Real Time Log System Event Log HTTP server HTTPS server Password Security FTP Server Browser-based GUI Real Time Subscriber Status Monitor Front Panel LCD User Account Manager SNTP time synchronization Auto Daylight Saving Customizable Time Zone V2.1 Yes Yes Yes Yes Yes (WellRTP 5100) Yes Yes Yes Yes Yes Yes Yes Yes, Start/Stop Yes, Active/Standby Yes Yes Programmable Yes Yes Yes (global setting) Yes Yes Yes Yes (TCP, data encrypted) Yes Yes Yes Yes Yes (limited feature only) Yes (limited feature only) Programmable (Module/Level) Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes Yes -8- 16 17 19 20 21 22 23 24 25 26 R. 1 2 3 4 5 6 7 S. 1 2 3 4 5 6 7 S. 1 2 3 T. 1 Provisional Interface System Event Email Notice Call/RTP Traffic Report Provisional Gateway (Windows AP) Ping SNMP V2 MIB I & II SNMP get command SNMP set command SNMP Trap Server SyslogD Server MS Unified Communication Integration Multiple LCS Server Support Subscriber to LCS Mapping PSTN/VOIP to LCS User Mapping Parallel Call Ring for SIP and LCS Auto LCS User RTP proxy Exchange 2007 Integration Exchange 2007 fax Max System Capacity Max Subscribers Support Max Concurrent Call Max Call Attempt per Seconds Max Concurrent RTP Support for NAT user Max DM List Max Concurrent Contacts WellRTP 5100 (RTP resource server) High Available Active/Standby Redundant Hitless for Stable Call Auto Sync Subscriber Related Information Manual English User Guide Yes (TCP, data encrypted) Yes Yes Provided Yes Yes Yes Yes Yes Yes Optional license is required Yes* Yes* Yes* Yes* Yes (no LCS to LCS RTP proxy) Yes Yes depended on license 20000 2000 100 384 4096 5 384 Optional license is required Yes* Yes* Yes* Yes * Required Additional License V2.1 -9- The Telephony SIP Proxy Server Appearance Description The Telephony SIP Proxy Server Front Panel: 2 7 5 1 3 4 8 6 Functions: 1: Power LED 2: Network1 Interface LED (not used) 3: Network2 Interface LED (not used) 4: H/D LCD 5: Power Switch 6: System Status LED 7: LCD Panel 8: LCD Touch Panel The Telephony SIP Proxy Server Rear Panel: 1 2 3 4 5 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) V2.1 - 10 - 6 7 8 9 Chapter 2 The Telephony SIP Proxy Server Quick Start After connecting Ethernet cables into the Telephony SIP Proxy Server 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 The Telephony SIP Proxy Server, 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 1.4.1 (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. V2.1 - 11 - Logon The Telephony SIP Proxy Server Setp1: Start IE6.0 (or later version) to navigate The Telephony SIP Proxy Server web management system by typing the default URL is http://192.168.67.1:10087 or https://192.168.67.1:10087 the screen will display User ID and Password as figure 2.1-1. Figure 2.1-1 ☺Note: The default network IP address is: SIP Service Interface: 192.168.67.1 255.255.0.0 192.168.67.254 Management Interface: 192.168.67.2 255.255.0.0 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 2.1-2 Step 3: The screen shows the Home Page of The Telephony SIP Proxy Server as figure 2.1-3. Figure 2.1-3 V2.1 - 12 - Network The Telephony SIP Proxy Server has 2 network interfaces: - WAN interface: Public net interface - LAN interface: Private net interface 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 2.2-1. Figure 2.2-1 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 2.2-2. Figure 2.2-2 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 2.2-3. Figure 2.2-3 V2.1 - 13 - Step 4: When screen shows “After configuration changed, please relogin system with new IP address and execute Soft-Reset!” click OK button as figure 2.2-4. Figure 2.2-4 Step 5: Follow Step 1 to 4 to change management interface network configuration as figure 2.2-5. Figure 2.2-5 ☺ Note: “Network Control” takes around 5-second to apply the new network configuration. Please logon again with new IP address after 5 seconds. System Time Step 1: When relogon to the new IP address, the next is to setup the system time zone. Click Control > System Time Zone to setup the system. Enter current date and time. Apply the change by clicking Apply button as figure 2.3-1. Figure 2.3-1 V2.1 - 14 - 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 2.3-2. Figure 2.3-2 Step 3: After successfully base setup, restart The Telephony SIP Proxy Server to take effect as figure 2.3-3. Figure 2.3-3 Apply Change When you loaded a new working or configuration or changed any configurations, you need click “Apply Change” to take effect as figure 2.4-1. Figure 2.4-1 Configuration quick step Please refer the “Configuration Detail Reference” to do the system configuration setup as follows: V2.1 Setup “SIP Domain” if use DNS Create subscriber or gateway Setup “Prefix hunting” for gateway Create required “Digit Manipulation” - 15 - Chapter 3 Configuration Reference The Telephony SIP Proxy Server Calling Processing Flow Caller Validation Valid Remove matched Prefix for G/W Invalid Not Found Emergency Call Target Found Valid No Start > Prefix Hunting Subscriber Search Caller/ Both Digits Manipulation Target Found Not Found Incoming call Short Code Yes Yes No Call Reject Invalid PSTN Number Make E.C. call ANI Screening DNIS Screening Valid Invalid Invalid Call Reject Valid RADIUS Call Permission No Call Reject Yes Invalid Called DM Valid Remove Add Prefix for G/W V2.1 - 16 - Valid Valid Call Making RADIUS Validation Invalid System Start Path: Configuration > System Figure 3.1-1 Parameter Description: • SIP Domain: the Telephony SIP Proxy domain name. It’s normally used when you have a DNS record setup for The Telephony SIP Proxy Server. • 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. - Non-Encrypt: not encrypt the SIP signal and voice RTP. • 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. • Enable System Log: Enable to send system information to syslogD Server or not • SyslogD Server IP 1, 2: syslogd server IP address • Over Max Contact Rule: Over Max Contact Rule, reject or update. V2.1 - 17 - - 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. 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 SMTP Server: SMTP server host for email notice Email From: Email sender account Email To: Email receiver (semicolon is used for multiple receiver) Subject: Email 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 - • • • • • • V2.1 - 18 - Advance System Configuration: Start Path: Configuration > System > Advance Figure 3.1-2 Advance Parameter Description: V2.1 - 19 - • • • • • • • • • • • V2.1 NAT Compare Method: How to detect a NAT user - IP Only: Compare IP only - IP / Port: Compare IP and UDP port RetransmissionT1: 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: 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 response is received, retransmission continue but at an interval of T2. (Default Value: 4000ms) ** RetransmissionT4: 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: 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: 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: 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: 10000 ms)** Proxy 2xx Sent Timer: 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, The Telephony SIP Proxy Server will issue a new nonce for authentication. Set it to 0 will cause The Telephony SIP Proxy Server to generate new nonce for each call or register - 20 - • • • • • • • • • • • • • • • • • • V2.1 Valid Period Auth Mode: During the nonce valid period, does a subscriber need 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 The Telephony SIP Proxy Server Subscriber Login: Enable Subscriber log or not 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 Pool Size: The buffer size used by the SIP Stack for receiving and sending SIP messages. 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 Justec external RTP resource server) Forward Caller ID: - Caller: use original caller id when call is forwarded - Forwarder: use forwarder caller id when call is forward to anther user Redundant Forward: contact Justec for detail implementation Forward Host: contact Justec for detail implementation 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) Voice Gateway: Enabling voice gateway feature, The Telephony SIP Proxy Server will able to play the role as a NAT server to pass through SIP and voice call. Please refer Voice Gateway Example for a configuration example. Support Video: Support video RTP proxying or not. Enable video will great reduce the number of concurrent RTP channel and bandwidth. - 21 - • • Missed Call from Domain: The Email From host name for missed call None: no host name or IP after email from user name IP: Email From host name is IP, [email protected] Domain: Email From host name is SIP domain, xxx @domain Missed Call Tel No: The Email From User part for missed call Tel no: The Email From user will be original called number Replaced ANI: The Email 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, 6500 will cancel the camp on silently. ** SIP and network knowledge is required to change these parameters. Device ACL: The Telephony SIP Proxy Server can use SIP User Agent to validate whether it is a tested device or not. This is system wide parameter and only listed devices are allowed to access The Telephony SIP Proxy Server. It is a good way to protect your system. Start Path: Configuration > System > Device ACL Figure 3.1-3 Parameter Description: • Device: User type to validate • User Agent: SIP User agent • Desc: Description System License: Start Path: Configuration > System > License V2.1 - 22 - Figure 3.1-4 License Parameter Description: • Feature: System parameter • Serial No: System parameter • License Key: System parameter • Version: Server version ☺ Note: Please don’t change it unless under Justec’s instruction 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 Justec FAE's instruction and turn off when complete. Or the system performance will be greatly hit. Start Path: Configuration > Debug Figure 3.2-1 V2.1 - 23 - Group Each user group can have its owned access code and related settings to minimum the management effort. Start Path: Configuration > Subscriber > Group Figure 3.3-1 Click on the user group you want to modify: Figure 3.3-2 Parameter Description: • Close Group: Enabled for in-group subscriber to subscriber call only. To call to another group need to get thought prefix hunting. • User Group: User Group ID • DM Group ID: Group-winded digit manipulation applied • User Group Description: user group description • SMTP Host: SMTP server host (i.e. mail.Justec.com.tw) for delivery missed call message • Miss Call Subject: Missed call notify subject You can have the following variables for notify subject. $FROM$: caller party number V2.1 - 24 - • • • • • • $TO$: 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. Click on the Detail button: It is Service code definition for the selected user group. Figure 3.3-3 Parameter Description: • Service Code: Telephony Keypad used for the service code • Service Type: Applied service type V2.1 - 25 - The others please refer to the examples below: Forward Service: Access Code Parameter (optional) 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 Service Enable don’t disturb *205 Don’t disturb time 1 (hhmmhhmm) Don’t disturb time 1 &2 (hhmmhhmmhhmm hhmm) Example *2010933111666 *201 (use existing setting) *2020282265699 *202 (use existing setting) *2030936111222 *203(use existing setting) *204302 *204(use existing setting) *20518002359 *2051800235901000900 *205(use existing setting) Enable Notify Enable Fine Me Enable CRBT Enable Announcement Enable VMS Disable unconditional forward Disable no answer forward Disable busy forward Disable unavailable forward Disable don’t disturb Disable Notify Disable Fine Me Disable CRBT Disable Announcement Disable VMS *206 *207 *208 *209 *210 *301 *302 *303 *304 *305 *306 *307 *308 *309 *310 n/a n/a n/a n/a n/a n/a n/a n/a n/a n/a n/a n/a n/a n/a n/a *206 (need pre-config by web) *207 (need pre-config by web) *208 (need pre-config by web) *209 (need pre-config by web) *210 (need pre-config by web) *301 *302 *303 *304 *305 *306 *307 *308 *309 *310 Hide ANI Service: Access Code Parameter Hide ANI *314 Dialed number Show ANI *214 Dialed number Service Example *5070100001 (Hide caller ID) *607010062 (Show caller ID) Pickup Call Service: Service Global Pickup Group Pickup Access Code *0 *1 Parameter n/a n/a Call Pickup *0 (global pickup) *1 (group pickup) VAD Service: Service Camp On Cancel Camp On Call Return V2.1 Access Code *211 *311 *212 Parameter 070700001 n/a n/a - 26 - Call Pickup *211070700001 *311 (cancel camp on) *212 Call Pickup *213 301 Call Park *215 Enable Privilege Access *216 070700001 (user’s password) Disable Privilege Access *316 n/a Disable Call Waiting Enable Call Waiting *217 *317 n/a n/a (No answer call return) *213301(assigned call pickup) *216 (enable privilege access) *316 (disable privilege access) *217 *317 Click on the Pickup button: Grouping the subscribers for group pickup service, you can set a subscriber to belong to a pickup group. Figure 3.3-4 Parameter Description: • Pickup Group ID: Pickup group ID • Description: Description V2.1 - 27 - Subscriber Start Path: Configuration > Subscriber Figure 3.4-1 Modify: Click on the subscriber you want to modify: Figure 3.4-2 Parameter Description: • 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 V2.1 - 28 - • • • • • • • V2.1 User Group: Belonged User group Authentication Mode: Authenticate subscriber by MD or not - None: No - None: None - Radius: Send to RADIUS for call permission - 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) - Gateway/RTP - Proxy/RTP - SIP Proxy: SIP proxy server - IVR/VMS: IP IVR or VMS server - IVR/VMS/RTP: IVR or VMS server - Recorder - Outbound Caller: Outbound Caller (SIPIVR 6800 is required) - 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. Welltech will provide OCX and sample code for integrating into the customer’s Web server. - Exchange Fax: Allow to provide the fax feature for Exchange 2007. Regular T.38 device can work with Exchange 2007 UMS when using this exchange fax user. - Hunting Method: Call forking method. - Sequential: Call hunting each contact in sequence - Parallel (ringing): Send multiple call invites to multiple contacts simultaneously. When receive the first 180 (ringing), use it and disconnect the others. - - 29 - • • • • • • • • • • • • • V2.1 - Parallel (answer): Send multiple call invites to multiple contacts simultaneously. When a user pickup the phone, disconnect others contacts. Preferred RTP Host: Preferred RTP resource server 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:[email protected]) Predefine URI2: Predefine subscriber URI2 (i.e. sip:[email protected]) - 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:[email protected]: 7777) Public TA: mapped NAT Server IP address and port (i.e. 61.218.42.217:5060) 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 Proxying or not (recommended) - Recorder: Use Recorder - Recording on Demand: Use Recorder on demand service - NAT Group: NAT group can be used for enterprise user. When two subscribers have same NAT group defined, The Telephony SIP Proxy Server will not use NAT Proxying 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 Fast Response Time: The maximum time to wait for response. It’s depended 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 The Telephony SIP Proxy Server. Max Concurrent Call: The maximum of concurrent call Call Validation: The call validation type: none, update or invite o None: disable call validation features o Update: Use SIP UPDATE instead of INVITE o Invite: Use SIP INVITE message for call validation Over Max Contact Rule: Over Max Contact Rule: reject, update or use Global Setting (Configuration > System) - 30 - • • • • • • o Reject: The system will reject the new contact REGISTER request when the subscriber’s used contacts reached the max contact o Update: The system will replace the oldest contact by new received contact. o Global Setting: Use system defined policy. AAA Sending Stage: Send AAA message before or after DM. Or use Global Setting (System > Advance) 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) 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. Description: Description Sync Web Password: Sync web password per subscriber Service: Click on the subscriber > service you want to modify telephony service for the selected user. V2.1 - 31 - Figure 3.4-3 Parameter Description: Forward Service: • 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. • No Answer Forward: Forward to the URI when the subscriber has no answer. • 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 • 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 • 181 for Call Forward: Send SIP181 for Call Forwarded service • Announcement Before Forward: Play Announcement before call forward • Forward Subscriber Only: Forward to proxy subscriber only. • 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. • Number Change: Change the original number to a new number. Welltech IP Centrex 6850 is required for announcement service. • Auto Call Forward: Auto Call forward or not after number changed Call Pickup: • Group Pickup (Picker): Allowed to be picked-up within group or not • Global Pickup (Picker): 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 on-list ANI can get through. For all TEL are set to disallow, only onlist ANI will be screened. Otherwise, “disallow” has higher priority than “allow”. V2.1 - 32 - • • 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: Coloring Ring Back Tone service • Announcement Service: Announcement Service • VMS: Voice Mail Service Subscriber Service • Short Code: short code to be used within same group • 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 6500. 6500 will look at the PSTN number first. • Disable call forward display name: Add display name as subscriber id in SIP from header when call forward caller mode is "forwarder". • Display Name: Assigned the display caller name for the subscriber. It will be showed on SIP IP phone. • Missed Call URI: Missed call notify service • Disable Conference Call: Disallow to call a conference call • Support Video: Enable Support Video or not • Disable Call Waiting: Disable call waiting feature. When disable call waiting features, the second incoming call to the user will be rejected by the SIP Telephony Server. • 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 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) V2.1 - 33 - • Disable Qop: Disable sending qop tag in SIP 401 and 407 authentication header. 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. UAC The SIP Telephony Server 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 on the subscriber > UAC you want to modify: Figure 3.5-1 Parameter Description: • User Agent ID: Identifier used for subscriber setting (type UAC) • Register ID: SIP registrar user ID • Register Password: SIP registrar user password • Register Realm: SIP registrar realm (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. V2.1 - 34 - 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. NAT NAT group can be used for enterprise user. When two subscribers have same NAT group defined, The Telephony SIP Proxy Server will not use NAT Proxying when both subscriber have same NAT group. You have to define NAT group definition by detail, or same IP address detect policy will be used. Start Path: Configuration > NAT Figure 3.6-1 Modify: Click on the group ID > Modify you want to modify: Figure 3.6-2 Basic Parameter Description: • IP Address: public IP address on NAT • Submask: network mask For example: IP: 61.218.42.224 Submask: 255.255.255.248 NAT Group Definition: 61.218.42.224 to 61.218.42.238 V2.1 - 35 - IP: 61.218.42.0 Submask: 255.255.255.0 NAT Group Definition: 61.218.42.0 to 61.218.42.225 RTP Resource This feature can be used only for external RTP resource server. By using this feature, the SIP Telephony Server can have more concurrent RTP proxying channels. You can define different RTP group for different purposes (e.g. by different ISR providers). Start Path: Configuration > RTP Resource Figure 3.7-1 Basic Parameter Description: • Group ID: RTP Server Group ID • Description: Description Click on the Detail button: Figure 3.7-2 Parameter Description: • User ID: RTP Server User ID • Priority: The RTP Server priority V2.1 - 36 - After define the installed RTP resource server, you can set the preferred RTP proxy server group, in subscriber menu, to be used. The SIP telephony Proxy Server will use RTP resource severs according to priority assigned within same group. Call Interception Call Interception can provide interception service through target call number, be sure that you have an external Recorder Server installed Start Path: Configuration > Call Interception Figure 3.8-1 Basic Parameter Description: • Target Number: The call number to be intercepted • Description: Description Prefix Route The Telephony SIP Proxy Server Prefix Route can provide prefix hunting base on priority, max idle time or round robin method. The SIP telephony Proxy Server will use prefix 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. V2.1 - 37 - Start Path: Configuration > Prefix Route Figure 3.9-1 Modify Prefix Route List: Click on the Modify button: Figure 3.9-2 Parameter: • 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. V2.1 - 38 - • • • - 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 base most idle policy. Stop hunting when a call is failed except receiving the defined reason code in load balance reason. - No Answer Timeout: The maximum time (in second) to wait the remote party answer (pick up phone) First Response Timeout: The maximum time to wait for device response. It’s depended on the network speed. Remove Prefix: Remove prefix matched or not RADIUS Authorization Resend: Send RADIUS authorization for each prefix hunting. Click Detail to define member of prefix routing group. Click on the Detail button: Figure 3.9-3 Parameter: • TEL NO: Subscriber TEL no for route • Priority: Used only for priority hunting. V2.1 - 39 - Click on the L.B Reason button: When enable load balance hunting, here is the cause reason to enable The Telephony SIP Proxy Server to continue the hunting. Figure 3.9-4 Parameter: • State Code: SIP State code to continue the hunting Digit Manipulation The Telephony SIP Proxy Server Digit Manipulation can provide operator target called number and calling number to “insert, replace or drop”. Start Path: Configuration > Digit Manipulation Figure 3.10-1 V2.1 - 40 - Click on the Detail button: Figure 3.10-2 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 on the Modify button: Figure 3.10-3 Parameter: • Matched Prefix: Calling/Called number party matched • Matched Target: Matched target is ANI(calling number) or DNIS(called number) • OP Target: Operator target is ANI(calling number) or DNIS(called number) • Matched Length: matched length of the target number • Apply Target: The target which the operation is executed. V2.1 - 41 - • • • • o Caller: When the subscriber is making a call, the DM will be applied o Called: When the subscriber is received a call, the DM will be applied. o Both: When the subscriber is making a call or receive a call, the DM will be applied. Active Mode: The DM group is active or inactive Start Position: Start position to be replaced Stop Position: Stop position to be replaced Replace Value: Replaced value Example: Group ID 88 35701 811360601 070700023 070700024 Digit Manipulation Setting 881 314 8226 070 DNIS ANI DNIS ANI 8 0 Caller Both 0 0 0 3 886 557 Subscriber ID DM Group ID Matched Prefix Matched Target OP Target Matched Length DM Apply Target Start Position Stop Position Replace Value Call Test Example: Call Test Number Display DNIS: 88682265555 ANI: 070700023 DNIS: 070100007 ANI: 99170700023 DNIS: 811360601 ANI: 557700023 DNIS: 070700023 ANI: 99111360601 DNIS: 070700023 ANI: 811360601 DNIS: 070700023 ANI: 811360601 DNIS: 070700023 ANI: 070100007 DNIS: 811360601 ANI: 557100007 DNIS: 070100007 ANI: 557700024 DNIS: 070700023 ANI: 557700024 070700023 > 82265555 070700023 > 070100007 070700023 > 811360601 811360601 > 070100007 811360601 > 82265555 811360601 > 070700023 070100007 > 070700023 070100007 > 811360601 070700024 > 070100007 070700024 > 070700023 V2.1 - 42 - 99 070100007 991 070 DNIS ANI 0 Called 0 1 991 DNIS Screening DNIS screen group can be used to limit the called prefix. Start Path: Configuration > DNIS Screening Figure 3.11-1 Click on the Detail button: Figure 3.11-2 Parameter: • Screening Prefix: Called number prefix • Screening Type: Allow or disallow When all TEL are set to allow (prefix match), only on-list DNIS can get through. When all TEL are set to disallow, only on-list DNIS will be screened. Otherwise, “disallow” has higher priority than “allow” V2.1 - 43 - Emergency Call To have subscriber-based emergency call setting, please define the required emergency call here and select it on subscriber basis. Start Path: Configuration > Emergency Call Figure 3.12-1 Click on the Detail button: Figure 3.12-2 Parameter: • Emergency Number: Emergency called number (e.g. 911) • Routed Number: Actually called number to be dial out (e.g. 0222211111) V2.1 - 44 - AAA When the subscriber users do the AAA (Authorization, Authentication and Accounting), enter the correct parameter the Radius setting. Start Path: Configuration > AAA Figure 3.13-1 Parameters: • Auth IP: Radius Authentication Server IP address • Auth Port: Radius Authentication Server Port • Acct IP: Radius Account Server IP address • Acct Port: Radius Account Server Port • Backup Auth IP: Backup Radius Authentication Server IP address • Backup Auth Port: Back Radius Authentication Server Port • Backup Acct IP: Back Radius Account Server IP address • Backup Acct Port: Back Radius Account Server Port • Secret Key: The shared secret key with RADIUS Server • Max Retry: The maximum retry times • Response Time (sec): The maximum wait for response time from RADIUS Server • Auth Retry Interval (sec): The internal to resend the Authentication packet to RADIUS Server. • Acc Retry Interval (sec): The internal to resend the Account packet to RADIUS Server. • Switch Threshold: Switch to alternate RADIUS Server when failures are occurred more than switch threshold. • 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) • Cisco Mode: - Yes: Use Cisco RADIUS mode (have redundant string in vender attribute) - No: no V2.1 - 45 - • Send Nero 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, The Telephony SIP Proxy Server will send RADIUS for call permission - No: When a subscriber calling another subscriber, The Telephony SIP Proxy Server 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 Configuration Manager Configuration Management provides a way to save and backup and restore the working configuration here. Backup the working configurations: Step 1: To backup the running configuration, click on Backup Configuration, to back up local hard disk as figure 3.11-1. Figure 3.14-1 Step 2: The whole running configuration will be compress into a zip file (file name: export.zip) and transfer back to local as figure 3.11-2. Figure 3.14-2 Restore Configuration: Step 3: To restore the backup configuration file, click Restore Configuration as figure 3.11-3. Figure 3.14-3 V2.1 - 46 - Step 4: Select backup file (i.e. c:\export.zip) click on Import button to restore the configuration to the working configuration as figure 3.11-4. Figure 3.14-4 ☺Note: It is need to restart the system to take effect of the new-restored working configuration. Compact the database file: Step 5: In order to optimize the system performance, you can optional compact the database by click Compact button as figure 3.11-5. Figure 3.14-5 ☺ Note: Please make sure that there is no others person to use database concurrently. V2.1 - 47 - Chapter 4 System Control Reference System Start path: Click Control > System Figure 4.1-1 Parameter: • Soft Reset: Soft Reset at the Telephony SIP Proxy Server • Upgrade Reset: Soft reset after application upgrade. The new application image will be extracted and executed without reboot. • Restart: Restart the the Telephony SIP Proxy Server • Shutdown: Shutdown the the Telephony SIP Proxy Server System Time Time Zone Setting Step 1: If you would like to use time zone, click Time Zone button to setup the system time zone as figure 4.2-1. Figure 4.2-1 Standard: Step 2: Select the Standard option to setup the system predefined time zone as figure 4.2-2 Figure 4.2-2 V2.1 - 48 - Parameter: • Time Zone: o Standard: Use a predefined standard time zone (Refer to section “Timezone to Country Mapping List” ) o Customize: Use a user defined time zone • Auto Daylight Saving: Auto adjust daylight saving time or not User defined Time Zone: Step 3: Select the Customized option and enter the time zone bias to set a user defined time zone as figure 4.2-3 Figure 4.2-3 Parameter: • Daylight Bias: The offset added to the Bias when the time zone is in daylight saving time • Daylight Start: The date that a time zone enters daylight time o Month: 01 to 12 o Week Day: Sunday to Saturday o 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) o Hour: 00 to 23 • Standard Start: The date that a time zone enters daylight time o Month: 01 to 12 o Week Day: Sunday to Saturday o 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) o Hour: 00 to 23 V2.1 - 49 - Network Management interface it’s used for management purpose. If you have DNS record, also you must be setup DNS server to effect. ☺Note: SIP Service and Manager Interface Setting: Please refer to section “Network Configuration” DNS Server Setting: Step 1: Enter correct DNS server IP address, host name, domain name and dynamic DNS registration to “Yes”. Apply change by click Apply button as figure 4.3-1. Figure 4.3-1 Parameter: • Primary DNS Server: Primary DNS Server IP network • Secondary DNS Server: Secondary DNS Server IP network • Host Name: Host name used to register to DNS Server • Domain Name: Domain name used to • Dynamic DNS Registration: Enable Dynamic DNS registration or not Voice Gateway Setting: Voice gateway mode needs 2 network legs. SIP service Ethernet leg need to be on WAN side and management interface Ethernet leg will be used for private IP leg. This feature private IP leg. This feature provides NAT server and voice only firewall functions. Step 1: Select Configuration > System > Advance > Voice Gateway to “Yes”. Apply change by click Apply button as figure 4.3-2. V2.1 - 50 - Figure 4.3-2 Step 2: Select Control > Network > LAN, click Route button as figure 4.3-2. Figure 4.3-3 New the routing table: Figure 4.3-4 In Private IP Ethernet leg, The Telephony SIP Proxy Server can also provides routing command for LAN to route their IP traffics. It is useful for those companies had different LAN or VPN network. Parameter: • Destination: Target IP address or network • Netmask: Network mask • Gateway: Destination gateway • Metric: Routing priority V2.1 - 51 - Voice gateway service example: SNMP Start path: Click Control > SNMP > Community Figure 4.4-1 Parameter: • Community Name: Community name for network manager system accessing • Access Rights: Giving access right to the community Start path: Click Control > SNMP > Trap Figure 4.4-2 V2.1 - 52 - Parameter: • Trap Community: Trap community name for NMS • Trap Host: Trap host IP address ☺Note: It takes around 1-minute to update SNMP configuration and display successful message. Account Manager You can manage (Modify, Add and Delete) the login user account as follows: Step 1: Click Control > Account Manager as figure 4.5-1 Figure 4.5-1 Field Description: • User ID: Login User ID • Password: Login Password • Confirm Password: Confirm new password again • Ownership: The ownership of the web management - Admin: super user - Monitor: view only ☺Note: The system provides 2 USER ID by default: User 1: “root” Password: “root” User 2: “admin” Password: “admin” Provisional IP The Telephony SIP Proxy Server can be integrated with other system, such billing system, web server etc, by using provisional interface. To implement the provisional interface, high security communication protect is required. However, to minimize the developing effort, the trusted provisional host can be defined in here. For those host/IP defined here, it will communicate without any security protect for provisional. V2.1 - 53 - Step 1: Click “Control > Provisional IP” to upgrade the software as figure 4.6-1. Figure 4.6-1 Field Description: • Trust IP: Trust provisional host IP • Enable: -Yes: enabled - No: disabled Upgrade The Telephony SIP Proxy Server provide upgrade new version at remote side. You can upgrade it from Welltech technical support web page by yourself. Step 1: Click “Control > Upgrade” to upgrade the software as figure 4.7-1. Figure 4.7-1 Field Description: • File Name: Upload the software file name • Upload: Remote Upload the software at The Telephony SIP Proxy Server Relogin Step 1: Click Control > Relogin to relogon by another user account as figure 4.8-1. V2.1 - 54 - Figure 4.8-1 V2.1 - 55 - Chapter 5 System Monitor Reference It provides a way to monitor the system status. Subscriber Status Show subscriber users status. Start Path: Monitor > Subscriber Status > Monitor button key-in the TEL No. and click Apply button to be controlled. Figure 5.1-1 Figure 5.1-2 V2.1 - 56 - See the Subscriber Detail: Select a subscriber and double-click to “Subscriber Detail” as figure 5.1-2. Figure 5.1-3 Field Description: • TEL NO: registered TEL number • Register: Registered or not • Call Count: number of concurrent calls for the user • Call Status: Detail is showed for subscriber. Summary only is used for gateway user. • Contact: Registered contact URI • NAT: NAT IP address • Register Time: Register time • TTL: Register time to live • Registrar: Registered Proxy IP and port • Unreg: Un-register the subscriber • Disconn: Disconnect the connected call V2.1 - 57 - Call Statistics Show total call statistics records . Start Path: Monitor > Call Statistics Figure 5.2-1 Field Description: • Time: statistic period in 24 hours format • Call: For the current period, this field showed real time concurrent call. For the past period, this field showed the concurrent call for last seconds. For example, if current time is 10:30, the 10-11 time period show the real time concurrent call and the 9-10 time period show the concurrent call right on 10:00. • Peak Call: In this period, the max call reached. • Total Call: The total call processed in the period • Connected Call: For the current period, this field showed real time connected call. For the past period, this field showed the connected call for last seconds. For example, if current time is 10:30, the 10-11 time period show the real time connected call and the 9-10 time period show the connected call right on 10:00. • Peak Connected Call: The max connected call in the period • Total connect call: The total connected call in this period • Register: For the current period, this field showed real time registered count. For the past period, this field showed the register count for last seconds. For example, if current time is 10:30, the 10-11 time period show the real time registered call and the 9-10 time period show registered count right on 10:00. • Peak Register: The max registered count V2.1 - 58 - RTP Status Show RTP resource server status. At least the sip telephony proxy server itself will be showed here as an internal RTP resource server when have no extra RTP resource server is defined. Start Path: Monitor > RTP Status Figure 5.3-1 RTP Statistics Show RTP count statistics Start Path: Monitor > RTP Statistics Figure 5.4-1 Field Description: • Time: statistic period in 24 hours format • Max NAT: The NAT resource capacity • NAT Call: For the current period, this field showed real time NAT resource is used. For the past period, this field showed the NAT count for last seconds. For example, if current time is 10:30, the 10-11 time period show the real time NAT call in this server and the 9-10 time period show NAT calls right on 10:00. V2.1 - 59 - • • • NAT Peak Call: The max NAT call in this period NAT Total Call: Total NAT calls in this period NAT Fail Call: NAT failed call in this period. It might be indicating the resource is exhausted. Server Status Show current server status Start Path: Monitor > Server Status Figure 5.5-1 Field Description: • Application : System application • Version: Application version • Status: Server status • Call Attempt: The number of concurrent call attempts • NAT Call: The number of concurrent NAT Calls • Register: The number concurrent Register clients • Max Call: Max current call • Current Call: Used call • Max Transaction: Max transaction • Used Transaction: Used transaction • Memory Pool: Max memory pool • Used Memory: Used memory pool V2.1 - 60 - Event Log Show system event log status. Start Path: Configuration > Event Log Figure 5.6-1 Field Description: • Type: Event Log type - Information - Warring - Error • Date: Event created date • Time: Event created time • Source: Executable program • Category: Event type (none, Welltech Sys…) • Event ID: Event Log ☺ Note: You can click Clear button to clear all event log. See the detail event log: Click the event log or select the log and click detail to see the log detail. Figure 5.6-2 V2.1 - 61 - Event Description: Event ID Event Description Description 8000 Get initial data fail Failed to start application 8001 Create DB Connection fail Failed to start application 8002 Create thread fail Failed to start application 8003 Unable to start socket Failed to start application 8004 Failed to start application 9501 Initial SIP callback function fail Get new application transaction fail Over subscriber license limit AP Logger Miss Call Notify Telnet Svr AP Logger Over subscriber license limit SIPPD program started Miss Call Notify Started Telnet Svr Started On the fly change (system change) 9510 AP Logger NAT Proxy Started 9530 AP Logger AAA Mgr start 9540 AP Logger Status Teller Started 9550 AP Logger Provisional Gateway Started 9600 SNTP client application started SNTP is started 8700 8701 9500 Failed to start application ☺ Note: You should see the Event ID 9500~ 9600 in normal time. Debug Info Shows detail trace level messages. Start Path: Click “Monitor > Debug Info” Figure 5.7-1 Filed Description: • Get Log: Get debug logs (-1~999) • Clear: Clear logs V2.1 - 62 - Ping You can use the “Ping” to check an IP is active or not. Start Path: Configuration > Ping Figure 5.8-1 Field Description: • Host IP Address: The IP address to ping V2.1 - 63 - Chapter 6 Telnet & RS-232 Configuration The Telephony SIP Proxy Server also can support to be managed by Telnet or Console port (RS-232) for basic operations. Interface: Network: TCP/IP Telnet (i.e. telnet 192.168.67.1 10086) RS232: - Connect using: COM1 - Baud Rate: 9600 - Data bits: 8 - Parity: None - Stop bits: 1 - Flow Control: None - Wire: Null modem line (crossed) Logon The Telephony SIP Proxy Server by Telnet Use Windows build-in Hyper Terminal or other telnet terminal emulator to login (e.g. telnet 192.168.67.1:10086). User ID & password will be required for login (default login user id: admin, password: admin & user id: root, password: root). Command List: Command echo Event log exit IP config ping reboot reset shutdown time Time zone User admin help & ? Description Auto echo on or off Clean or show system log message Quit the current session Configure or show network1,2 information Check an IP address is available or not Reboot Soft-reset Shutdown Reset or show system time. Setup or show system time zone Manage user account. View command list Echo: auto echo on or not Command Purpose [root#]echo ? Usage: echo on/off Example: echo on [root#]echo on [root#]echo off Echo is on Echo is off (default value) V2.1 - 64 - Event log: show system log message Command Purpose [root#]event log ? Usage: event log [-clear] Example: event log Event log -clear [root#]event log Show system event log: Event log example: [root#] event log Time: 2005-04-11 14:07:34 Event ID: 9501 Type: Information Source : SIPPD Description: [14:07:34-461][Information]: SIPPD on the fly change Time: 2005-04-11 14:01:12 Event ID: 9500 Type: Information Source : SIPPD Description: [14:01:12-141][Notice]: SIPPD Started(ver 2.02) Time: 2005-04-11 13:57:32 Event ID: 9500 Type: Information Source : SIPPD Description: [13:57:32-054][Notice]: SIPPD Started(ver 2.02) Press any key to continue or press 'Q' to quit Press any key to continue or press 'Q' to quit [root#]event log clear Clear all event log Exit: Quit the current session Command [root#]exit Purpose Quit the current session ipconfig: Configuration or show network information Command Purpose [root#] ipconfig ? Usage: ip config [-network boardno][-delete dns] [-dhcp] [dns IPAddress1 IP Address2 ] [-ip IP Address -mask Mask -gateway Gateway] Example: ipconfig -network 1 -ip 192.168.67. 1 -mask 255.255.0.0 -gateway 1 92.168.5.254 example : ipconfig -network 1 -dhcp example : ipconfig -network 1 -dns 192.168.1.1 example : ipconfig -network 1 -delete dns [root#]ipconfig Show current network configuration [Network 1] Local Area Connection USE FIXED IP (or DHCP) IP Address : 192.168.5.7 Subnet Mask : 255.255.0.0 Default Gateway : 192.168.5.254 DNS Servers : [Network 2] Local Area Connection 2 USE FIXED IP IP Address : 192.168.5.8 Subnet Mask : 255.255.0.0 Default Gateway : DNS Servers : V2.1 - 65 - [root#]ipconfig network 1 –delete dns Delete the DNS servers setting [Network 1] Local Area Connection USE FIXED IP IP Address : 192.168.5.113 Subnet Mask : 255.255.0.0 Default Gateway : 192.168.1.254 DNS Servers : [root#]ipconfig network 1– Enable DHCP [Network 1] Local Area Connection dhcp USE DHCP IP Address : 192.168.5.10 Subnet Mask : 255.255.0.0 Default Gateway : 192.168.1.254 DNS Servers : 192.168.5.1 168.95.1.1 [root#]ipconfig network 1– Use fixed network configuration [Network 1] Local Area Connection ip 61.220.126 28 –mask USE FIXED IP 255.255.0.224 –gateway IP Address : 61.220.126.28 61.220.126.1 Subnet Mask : 255.255.255.1 Default Gateway DNS Servers : 61.220.126.254 : [root#]ipconfig network 1– Changes IP address only. [Network 1] Local Area Connection ip 61.220.126.115 USE FIXED IP IP Address Subnet Mask Default Gateway DNS Servers : 61.220.126.115 : 255.255.255.1 : 61.220.126.254 : [root#]ipconfig network 1– Changes DNS configuration only. [Network 1] Local Area Connection dns 210.59.126.53 USE FIXED IP IP Address Subnet Mask Default Gateway DNS Servers : 61.220.126.115 : 255.255.255.1 : 61.220.126.254 : 210.59.126.53 Ping: Check an IP address is available or not Command [root#] ping ? [root#]ping 61.220.126.1 Purpose Usage: ping IP. Example: ping 127.0.0.1 Ping result Reply from 61.220.126.1 bytes=64 time=1ms TTL=29 Reply from 61.220.126.1 bytes=64 time=1ms TTL=29 Reply from 61.220.126.1 bytes=64 time=1ms TTL=29 Reply from 61.220.126.1 bytes=64 time=1ms TTL=29 Reboot: Command [root#] reboot ? Purpose Reboot System Are You Sure? (Y/N) [root#]reboot Are You Sure?(Y/N)y V2.1 The Telephony SIP Proxy Server are rebooting - 66 - Shutdown: Command [root#] shutdown ? Purpose Shutdown System Are You Sure? (Y/N) [root#]shutdown Are You Sure?(Y/N)y The Telephony SIP Proxy Server are shutting down Reset: Command [root#] reset ? Purpose Soft reset System Are You Sure? (Y/N) [root#]reset Are You Sure?(Y/N)y Time: Reset or show system time Command [root#] time ? [root#]time Purpose Usage : time YYYY-MM-DD HH:NN:SS Example : Time 2002-01-01 12:00:00 Show current time The current time is 2003-06-20 15:17:30 [root#]time 2003-07-29 23:14:53 Change system bios time Time Zone: Setup or show system time zone Command V2.1 Purpose - 67 - [root#] timezone ? V2.1 Fixed Zone List: 01. Afghanistan Standard Time 03. Arab Standard Time 05. Arabic Standard Time 07. AUS Central Standard Time 09. Azores Standard Time 11. Cape Verde Standard Time 13. Cen. Australia Standard Time 15. Central Asia Standard Time 17. Central European Standard Time 19. Central Standard Time 21. Dateline Standard Time 23. E. Australia Standard Time 25. E. South America Standard Time 27. Egypt Standard Time 29. Fiji Standard Time 31. GMT Standard Time 33. Greenwich Standard Time 35. Hawaiian Standard Time 37. Iran Standard Time 39. Korea Standard Time 41. Mexico Standard Time 2 43. Mountain Standard Time 45. N. Central Asia Standard Time 47. New Zealand Standard Time 49. North Asia East Standard Time 51. Pacific SA Standard Time 53. Romance Standard Time 55. SA Eastern Standard Time 57. SA Western Standard Time 59. SE Asia Standard Time 61. South Africa Standard Time 63. Taipei Standard Time 65. Tokyo Standard Time 67. US Eastern Standard Time 69. Vladivostok Standard Time 71. W. Central Africa Standard Time 73. West Asia Standard Time 75. Yakutsk Standard Time - 68 - 02. Alaskan Standard Time 04. Arabian Standard Time 06. Atlantic Standard Time 08. AUS Eastern Standard Time 10. Canada Central Standard Time 12. Caucasus Standard Time 14. Central America Standard Time 16. Central Europe Standard Time 18. Central Pacific Standard Time 20. China Standard Time 22. E. Africa Standard Time 24. E. Europe Standard Time 26. Eastern Standard Time 28. Ekaterinburg Standard Time 30. FLE Standard Time 32. Greenland Standard Time 34. GTB Standard Time 36. India Standard Time 38. Israel Standard Time 40. Mexico Standard Time 42. Mid-Atlantic Standard Time 44. Myanmar Standard Time 46. Nepal Standard Time 48. Newfoundland Standard Time 50. North Asia Standard Time 52. Pacific Standard Time 54. Russian Standard Time 56. SA Pacific Standard Time 58. Samoa Standard Time 60. Singapore Standard Time 62. Sri Lanka Standard Time 64. Tasmania Standard Time 66. Tonga Standard Time 68. US Mountain Standard Time 70. W. Australia Standard Time 72. W. Europe Standard Time 74. West Pacific Standard Time Usage1 : timezone zone (1 to 75) Auto Daylight (Y or N) Example1 : time zone 1 Y Usage2 : time zone -custom Bias Daylight Bias Daylight Start Standard Start Bias : -12:00 to +13:00 Daylight Bias : -12:00 to +13:00 Daylight Start : MM (Month: 01 to 12) ; WD (Day of week: 00 to 06) DD (Day:01 to 05 ;Specifies the occurrence of day in the month; 01 = First occurrence of day, 02 = Second occurrence of day, ..., 05 = Last occurrence of day HH (Hour:00 to 23) Standard Start : MM (Month: 01 to 12) ; WD (Day of week: 00 to 06) DD (Day:01 to 05 ;Specifies the occurrence of day in the month; 01 = First occurrence of day, 02 = Second occurrence of day, ..., 05 = Last occurrence of day HH (Hour:00 to 23) Example2 : time zone -custom +08:00 -01:00 04-00-01-02 10-0005-02 [root#]time zone Show current time zone info Time Zone : (40) Mexico Standard Time (GMT -06:00) Daylight Bias : -01:00 Daylight Start : 05-00-01 02:00 Standard Start : 09-00-05 02:00 Auto Daylight : Y [root#]time zone 40 n Use pre-defined time zone [root#]timezone custom +08:00 01:00 05-00-01-03 09-00-05-03 Use customized time zone Time Zone : (40) Mexico Standard Time (GMT -06:00) Daylight Bias : -01:00 Daylight Start : 05-00-01 02:00 Standard Start : 09-00-05 02:00 Auto Daylight : n Time Zone : (99) Customized (GMT 08:00) Daylight Bias : -01:00 Daylight Start : 05-00-01 03:00 Standard Start : 09-00-05 03:00 Auto Daylight : Y Useradmin: Manager User account Command Purpose [root#] useradmin ? Usage: useradmin [-add User] [-delete User] [password User] Example: useradmin -add irene [root#]useradmin Show the current login user account root [root#]useradmin -list V2.1 Show the current user account list admin root irene - 69 - [root#] useradmin -add irene Password : irene Confirm : irene Add user Success. [root#] useradmin -delete 1111 Are You Sure?(Y/N)y [root#] useradmin -password root New Password : 1234 Confirm : 1234 V2.1 Add the new user account: irene Delete the user: 1111 Change the user: root’s password. - 70 - Chapter 7 LCD Display Configuration The Telephony SIP Proxy Server provides a front panel LCD for basic operations. Button List: Description Button List When the The Telephony SIP Proxy Server is ready, the LCD screen shows as blow Ready | 04-03-03 16:40 Press Enter to select command Event Log IP Config Enter ESC ▲ ▼ Quit the current command Up or previous edit mode Next or previous edit mode Command Tree: Main Menu Event Log Show system log message Show IP Info IP Config Network 1 Network 2 Use DHCP Use Fixed IP Yes Reboot No Reset PWD Yes No Yes Soft Reset No Yes Shut V2.1 - 71 - Down No Event Log: Configure ▲ ▼ Enter ▲ ▼ ESC ESC LCD Display Previous event log Next event log Show detail event log Previous line Next line Quit detail event log viewing Quit to main menu IP Config: Configure ▲ ▼ Enter ▲ ▼ Enter ▲ ▼ Enter ESC ESC LCD Display Select Network1 or Network2 configuration Select Network1 or Network2 configuration Configure Network1 or Network2 Select Network configuration Select Network configuration Configure Network Increase the digit apply to network setting Decrease the digit apply to network setting Apply change to network information Quit network setting Quit to main menu Reboot: Configure ▲ ▼ Enter ESC ESC LCD Display Select Reboot or not Select Reboot or not Reset user: root’s (or admin) user password Quit Reboot configure Quit to main menu Reset: Configure ▲ ▼ Enter ▲ ▼ ESC ESC V2.1 LCD Display Select user to change password Select user to change password Change user password Increase the alphabet apply to user password setting Decrease the alphabet apply to user password setting Quit Reset configure Quit to main menu - 72 - Soft Reset: LCD Display Configure ▲ ▼ Enter ESC ESC Select Reset or not Select Reset or not Reset or not Quit Reset configure Quit to main menu Shutdown: LCD Display Configure ▲ ▼ Enter ESC ESC V2.1 Select Shutdown or not Select Shutdown or not Shutdown or not Quit Shutdown configure Quit to main menu - 73 - Appendix 1 Retrieve CDR Information ☺ Retrieve method example (stop20040305.log) by ftp: C:\>ftp 192.168.5.8 Connected to 192.168.5.8. 220 Server ready User (192.168.5.8 :( none)): root 331 Password required for root. Password: 230 User root logged in. ftp> cd cdr 250 CWD command successful. "d:/ap/cdr/" is current directory. ftp> dir 200 Port command successful. 150 Opening data connection for directory list. drw-rw-rw- 1 ftp ftp 0 Mar 06 00:02 . drw-rw-rw- 1 ftp ftp 0 Mar 06 00:02 .. -rw-rw-rw- 1 ftp ftp 53998192 Mar 05 23:57 STOP20040305.log -rw-rw-rw- 1 ftp ftp 20222855 Mar 05 23:50 STRT20040305.log 226 File sent ok ftp: 403 bytes received in 0.25Seconds 1.61Kbytes/sec ftp> bin 200 Type set to I. ftp> lcd Local directory now C:\. ftp> get stop20040305.log 200 Port command successful. 150 Opening data connection for stop20040305.log. 226 File sent ok ftp: 20222855 bytes received in 4.43Seconds 4569.10Kbytes/sec. ftp>bye 221 Goodbye Billing Start CDR: • File name: STRTyyyymmdd.log • Field delimit: , • Field description: NAS-IP-Address: The Telephony SIP Proxy Server IP address NAS-Port-Type : (Network Access Server Port Type) Asynchronous User-Name : User ID Called-Station-Id : Called station number Calling-Station-Id : Calling station number Acct-Status-Type : Message type (1: start) Service-Type : 1: login V2.1 - 74 - Calling-IP-Address Conf-ID Call-Type Call-Originate Setup-Time Acct-Session-Id : Calling IP Address : GUID : VOIP : originate : Call initiate time (UTC time) : Start from 00000001 Billing Stop CDR: • File name: STOPyyyymmdd.log • Field delimit: , • Field description: NAS-IP-Address : The Telephony SIP Proxy Server IP address NAS-Port-Type : (Network Access Server Port Type) Asynchronous User-Name : User ID Called-Station-Id : Called station number Calling-Station-Id : Calling station number Acct-Status-Type : Message type (2: Stop) Service-Type : 1: login Calling-IP-Address : Calling IP Address Conf-ID : Call ID Call-Type : VOIP Setup-Time : Setup Time (UTC time) Connect-Time : Connect Time (UTC time) Disconnect-Time : Disconnect Time (UTC time) Disconnect-Cause : Disconnect cause code Call-Originate : Call originate Remote ID : Called IP address Acct-Session-Id : start ID from 00000001 Acct-Session-Time : Talk time V2.1 - 75 - Appendix 2 The SIP Telephony Proxy Server Status Code SIP Code 200 400 OK Bad Request 403 Forbidden 404 Not Found 423 Interval too Brief 480 Temporarily Unavailable Reason 486 487 Call Leg/ Transaction Does not Existed Busy here Request Terminate 402 Payment Required 403 Forbidden 404 406 410 Not Found Not Acceptable Gone Service Unavailable 481 503 V2.1 Description Success Calling ID is not Registered ANI Screened DNIS Screened Close Group Group DNIS Screened From header is not a Trust host or gateway User ID Expired Not SIP service IP Caller unconditional forward Privilege Access code error Calling ID is not Existed Called ID is not Exist DNIS is empty User ID or Tel is empty Host Address is empty Empty Host Address Fail (VIA) Register expire < Proxy default register TTL No RTP Resource Remote Party no Response No Response Time out –Trying Called ID is not Registered Over max call count Call Leg/ Transaction Does not Existed User Busy Cancel Response AAA Failed – Account has no balance (4) AAA Failed – Insufficient balance (11) AAA Failed – Invalid pin number (2) AAA Failed – Denied User (7) AAA Failed – Account disabled (12) AAA Failed – Invalid account (1) AAA Failed – Called Number Block (9) AAA Failed – Account is expired (5) AAA Server Connect timeout - 76 - Appendix 3 Time zone to Country Mapping List Greenwich Mean Time & Country List (GMT-12:00) International Date Line West (GMT-11:00) Midway Island, Samoa (GMT-10:00) Hawaii (GMT-09:00) Alaska (GMT-08:00) Pacific Time (US & Canada); Tijuana (GMT-07:00) Mountain Time (US & Canada) (GMT-07:00) Chihuahua, La Paz, Mazatlan (GMT-07:00) Arizona (GMT-06:00) Saskatchewan (GMT-06:00) Guadalajara, Mexico City, Monterrey (GMT-06:00) Central Time (US & Canada) (GMT-06:00) Central America (GMT-05:00) Indiana (East) (GMT-05:00) Eastern Time (US & Canada) (GMT-05:00) Bogota, Lima, Quito (GMT-04:00) Santiago (GMT-04:00) Caracas, La Paz (GMT-04:00) Atlantic Time (Canada) (GMT-03:30) Newfoundland (GMT-03:00) Greenland (GMT-03:00) Buenos Aires, Georgetown (GMT-03:00) Brasilia (GMT-02:00) Mid-Atlantic (GMT-01:00) Cape Verde Is. (GMT-01:00) Azores (GMT) Greenwich Mean Time: Dublin, Edinburgh, Lisbon, London (GMT) Casablanca, Monrovia (GMT+01:00) West Central Africa (GMT+01:00) Sarajevo, Skopje, Warsaw, Zagreb (GMT+01:00) Brussels, Copenhagen, Madrid, Paris (GMT+01:00) Belgrade, Bratislava, Budapest, Ljubljana, Prague (GMT+01:00) Amsterdam, Berlin, Bern, Rome, Stockholm, Vienna (GMT+02:00) Jerusalem (GMT+02:00) Helsinki, Kyiv, Riga, Sofia, Tallinn, Vilnius (GMT+02:00) Harare, Pretoria (GMT+02:00) Cairo (GMT+02:00) Bucharest (GMT+02:00) Athens, Istanbul, Minsk V2.1 Time Zone 21. Dateline Standard Time 58. Samoa Standard Time 35. Hawaiian Standard Time 02. Alaskan Standard Time 52. Pacific Standard Time 43. Mountain Standard Time 41. Mexico Standard Time 2 68. US Mountain Standard Time 10. Canada Central Standard Time 40. Mexico Standard Time 19. Central Standard Time 14. Central America Standard Time 67. US Eastern Standard Time 26. Eastern Standard Time 56. SA Pacific Standard Time 51. Pacific SA Standard Time 57. SA Western Standard Time 06. Atlantic Standard Time 48. Newfoundland Standard Time 32. Greenland Standard Time 55. SA Eastern Standard Time 25. E. South America Standard Time 42. Mid-Atlantic Standard Time 11. Cape Verde Standard Time 09. Azores Standard Time 31. GMT Standard Time 33. Greenwich Standard Time 71. W. Central Africa Standard Time 17. Central European Standard Time 53. Romance Standard Time 16. Central Europe Standard Time 72. W. Europe Standard Time 38. Israel Standard Time 30. FLE Standard Time 61. South Africa Standard Time 27. Egypt Standard Time 24. E. Europe Standard Time 34. GTB Standard Time - 77 - (GMT+03:00) Nairobi (GMT+03:00) Moscow, St. Petersburg, Volgograd (GMT+03:00) Kuwait, Riyadh (GMT+03:00) Baghdad (GMT+03:30) Tehran (GMT+04:00) Baku, Tbilisi, Yerevan (GMT+04:00) Abu Dhabi, Muscat (GMT+04:30) Kabul (GMT+05:00) Islamabad, Karachi, Tashkent (GMT+05:00) Ekaterinburg (GMT+05:30) Chennai, Kolkata, Mumbai, New Delhi (GMT+05:45) Kathmandu (GMT+06:00) Sri Jayawardenepura (GMT+06:00) Astana, Dhaka (GMT+06:00) Almaty, Novosibirsk (GMT+06:30) Rangoon (GMT+07:00) Krasnoyarsk (GMT+07:00) Bangkok, Hanoi, Jakarta (GMT+08:00) Taipei (GMT+08:00) Perth (GMT+08:00) Kuala Lumpur, Singapore (GMT+08:00) Irkutsk, Ulaan Bataar (GMT+08:00) Beijing, Chongqing, Hong Kong, Urumqi (GMT+09:00) Yakutsk (GMT+09:00) Seoul (GMT+09:00) Osaka, Sapporo, Tokyo (GMT+09:30) Darwin (GMT+09:30) Adelaide (GMT+10:00) Vladivostok (GMT+10:00) Hobart (GMT+10:00) Guam, Port Moresby (GMT+10:00) Canberra, Melbourne, Sydney (GMT+10:00) Brisbane (GMT+11:00) Magadan, Solomon Is., New Caledonia (GMT+12:00) Fiji, KamChapterka, Marshall Is. (GMT+12:00) Auckland, Wellington (GMT+13:00) Nuku'alofa V2.1 22. E. Africa Standard Time 54. Russian Standard Time 03. Arab Standard Time 05. Arabic Standard Time 37. Iran Standard Time 12. Caucasus Standard Time 04. Arabian Standard Time 01. Afghanistan Standard Time 73. West Asia Standard Time 28. Ekaterin burg Standard Time 36. India Standard Time 46. Nepal Standard Time 62. Sri Lanka Standard Time 15. Central Asia Standard Time 45. N. Central Asia Standard Time 44. Myanmar Standard Time 50. North Asia Standard Time 59. SE Asia Standard Time 63. Taipei Standard Time 70. W. Australia Standard Time 60. Singapore Standard Time 49. North Asia East Standard Time 20. China Standard Time 75. Yakutsk Standard Time 39. Korea Standard Time 65. Tokyo Standard Time 07. AUS Central Standard Time 13. Cen. Australia Standard Time 69. Vladivostok Standard Time 64. Tasmania Standard Time 74. West Pacific Standard Time 08. AUS Eastern Standard Time 23. E. Australia Standard Tim 18. Central Pacific Standard Time 29. Fiji Standard Time 47. New Zealand Standard Time 66. Tonga Standard Time - 78 -