p 2014 Mizu Softswitch Administrator’s Guide VoIP Server documentation Mizu Server is a software soft switch solution running on Microsoft Windows platforms that can replace traditional hardware based PBX and ISDN solutions.
1. Introduction ......................................................................................................................................................................................................... 12 1.2. Features ........................................................................................................................................................................................................ 13 1.2.1. Hosting ..........................................................
2.1.14. Self-check and reporting ..................................................................................................................................................................... 26 2.1.15. VoIP-GSM Gateway ........................................................................................................................................................................... 26 2.1.16. Other components .....................................................................................
.2.7. Capacity Check ..................................................................................................................................................................................... 47 4.2.8. System Load.......................................................................................................................................................................................... 47 4.2.9. Server Console ...........................................................................
4.5.2. Price List ............................................................................................................................................................................................... 93 4.5.3. Billing ................................................................................................................................................................................................... 93 4.5.4. Currency Conversion ..........................................................
4.7.10. To-do ................................................................................................................................................................................................. 130 4.7.11. Notes ................................................................................................................................................................................................. 130 4.7.12. Holidays .......................................................................
4.10.5. Automatic software upgrades ........................................................................................................................................................... 161 4.11. PBX / IPCentrex functionality ................................................................................................................................................................. 161 4.11.1. Call Rerouting ...........................................................................................
4.12.12. Encryption and tunneling ................................................................................................................................................................ 183 5. FAQ................................................................................................................................................................................................................... 186 5.1. How to make a H323 call directly to a GW (without the gatekeeper) ....................
5.33. How to disable PIN request on GSM gateways ................................................................................................................................... 191 5.34. What are the minimal global settings that must be correct on servers? ............................................................................................... 192 5.35. How to add a new traffic sender? ................................................................................................................
5.67. How to setup a new virtual server instance ......................................................................................................................................... 200 5.68. Fax detection ........................................................................................................................................................................................ 201 5.68. Handling dynamic & private ip/port .......................................................................
6. 5.100. How to rewrite caller/called numbers ................................................................................................................................................ 219 5.101. Reseller checklist ............................................................................................................................................................................... 219 5.102. Register timeout .....................................................................................
1. Introduction About This document provides an overall technical description of Mizu VoIP platform. For non technical user guides please check out our website. Your installation may not include all modules described in this document and you will notice small changes if you program version doesn’t conform to the documented version! Version MizuServer v5.0 Administrator’s Guide Revisited January 14, 2014 Copyright Mizu server and other MizuVoIP software are copyrighted by MizuTech SRL.
1.2.
1.2.1. Hosting MS-SQL backend (Express or Full versions) Ethernet 10/100/1000 Base-T Static IP, PPPoE (DSL or cable modem), DialUpISDN,VPN Encrypted communications Virtual servers STUN/ICE Support NAT Support Near-End and Far-End NAT traversal Multi-homed and multi-domain support 1.2.2.
CLIP/CLIR DTMF generation Call-Forward on out-of-service Codec transcoding Advanced statistics support NAT traversal of signaling NAT traversal of media SIP Session timers RTP Timers and media timeout Blind SIP Registration Late Codec Negotiation Multiple SIP registrations per user account Can act as an SBC Max Session Setting Manage Presence Detailed call logs SIP/SIMPLE SIP Reinvites SIP-H.
RFC 3428 Session Initiation Protocol (SIP) Extension for Instant Messaging RFC 1889 RTP: A Transport for Real-Time Applications RFC 2190 RTP Payload Format for H.
• Packet saver technology 1.2.4.
Statistic generation: customer statistics, operator statistics, call related statistics, work time statistics, campaign statistics Campaign creation: supervisors can create a campaigns Invitation letter: customization, and automatic printing Report generation: Specific hourly, daily and weekly reports 1.2.6. Accounting ACD Features Unlimited accounts / Unlimited Extensions Automatic pincode generation Flexibile authentication (digest,IP,port,user,etc) 1.2.7.
Automatic capacity rebalancing Remote Linked Servers Automatic channel management Number portability support User authentication by username/password, IP address, techprefix, caller number GeoIP database Automatic SIM allocation: Sim allocation rules: Rules can be defined on multiple levels: global, partner, gateway, engine, simpacket, simcard, time -Static -will not modify gw settings -Limits -sl (day/month) -packet allowed intervals -min/max lines for partner -Priorities -sim partnerm, sim, gw
Invoice generation in different formats, PDF generation, email scheduler and invoice printing Complete call rating & accounting services for complex rating schemes Currency and VAT can be set for every packet. Time zone can be changed. Automatic online currency conversion Paypal and lot’s of other payment gateways are supported 1.2.9.
Backup Gatekeeper capability Gatekeeper clustering support (neighbors, parent/child, alternates) 1.2.11. VoIP-GSM GSM components are not included with the default standard install and the required hardware is for extra costs. Please contact us about the possibilities. Hardware Components No hardware components are required for H323 and SIP networks.
SIM server interworking GSM cell selection and locking DTMF send/receive CLI restriction SIM Rerouting Locking to a given gsm cell Automatic SIM credit request and charge Voice Recording and Playback (In Stereo caller/callee left/right) SIM server interworking Virtual Channels VoIP-GSM First centralized architecture for GSM termination Unlimited Gateway and TrafficSender support Multiple signaling protocol support Load distribution between the operational channels No hard lim
License management Distributed absolute fault tolerant system External system supervisor service (email and sms alerts, watchdog can restart failed subsystems) Can be used as virtual servers 1.2.12. Tunneling and encryption Tunneling and encryption are built-in into the Mizutech VoIP server and you can enable from the global configuration or from the Config Wizard. Details: http://www.mizu-voip.com/Software/VoIPTunnel.aspx 1.2.13.
Billing module Balance module Real Time Capacity check Ability to insert queries directly into the database Blacklist filtering Self-analysis tools Detailed logging (multiple levels). Detailed call tracing capability Call simulations Reseller/Agent Registration and Management Capacity and system load reports Import/Export from/to any format (SQL, text, excel, etc) CRM Integration Restore and Backup 1.2.14.
The Mizu Soft switch (Server) is the “brain” of the system. Depending on your needs, you can connect as many gateways as you want. The soft switch is built from several modules: sip stack, h323 stack, sip – h323 conversion module, media server, ACL, routing, billing, alerting. Almost all modules are installed by default and they can be enabled/disabled as needed. 2.1.1. SIP Stack The Mizu SIP stack was written in C++.
2.1.10. Watchdog service This NT service can automatically detect critical service problems and restart the VoIP server, the SQL database or the OS. 2.1.11. HTTP service The built-in HTTP server will listen for service requests. 2.1.12. Direct command interface Administrative tasks and class5 service requests can be handled by the command line TCP interface. 2.1.13. Enduser web portal Template portal is available with source code.
2.1.16.
You can give the VPC for any of your partners. The partners can login to the VPC with the username and password configured in the “Users and Devices” form in MManage. Usually only “Owner” users will receive VPC access. You can define what users can see in their VPC by setting the “Can watch sim packets”, “Can watch users/devices” and “Access Rights” in the user configuration form (billing tab). See section 4.3.1 for more details. The VPC included with MManage has the capability to login as a super user.
3.2. Gateway quick setup Skip this chapter if you are not using VoIP-GSM gateways. In order to get a working system, here is a checklist which may help you: 1. Connect the gateway(s) and/or the server to the network. 2. Install the MManage programs in a separate PC used for monitoring your Mizu devices. network (you can find it on the Mizu install CD which is shipped with every product) 3. Set up the gateway(s) and/or the server network parameters with the VnetCfg utility 4.
8. Set up the simcards. You can add simcards manually, but it’s easier to wait for them to register. Then you only have to modify its packets, owners and the recharging settings (“Simcards” form) 9. Add some traffic sender in the “Users and Devices” form. Be careful with the authorization settings 10. Set up the routing (“Routing” form) Add at least one routing pattern (name it as you wish) Add at least one entry to its priority list (your newly created packet or some other direction) 11.
-and many others Since all data is stored in the SQL database, you have to protect only your database (the program directory can be recreated anytime by simple file copy or reinstall. Only the database path is stored here in the mizuserver.ini file which need to be reentered correctly after a restore). Although there is no special maintenance needs with the mizu server, to have a working backup is a very important task to be set and tested properly.
dbmaint_backuplevel: 0=no backups,1=daily,2=daily,monthly,weekly,3=hourly,daily,monthly,weekly,4=keep lots of files dbmaint_backupdbdir: local path (accessible for the server) dbmaint_backupdbnetworkdir: path accessible for the database engine (it is necessary when the database is located on a separate box. Otherwise can be left empty –default to dbmaint_backupdbdir) 3.5.4. Using backup database tables To keep your active size smaller, you can move the content of big database tables to another database.
3.5.6. Saving recorded voice Serverftpvoice: where to store recorded audio Serverftpdailydir: set to true to create separate directory for recorded voices daily Keeprecorded: days to keep voice records Voicebackupdir: backup your recorded voices to this secondary location Keepbackuprecorded: days to keep the voice backups The voice recording option can be set for any user by checking the “Record” checkbox on the user configuration form in MManage.
if the backup server fails (the server behind the main server, with private ip) -connect to the main server with the remote desktop "root" account -On the main server, do the followings: 1. launch the stop batch file (from the gk directory) 2. Enable and Start the SQLSERVER service 3. Restore latest database 4. launch the start batch file 5. check the vservdebuglog and MManage (you must have current calls) 6. you are ready 3.5.9.
At Source/ Database select the database that you want to back up. At Backup type select either full for full type of backup or differential, for differential backup. At Destination click on the Add button, select the disk, and path to the file where you want to save the backup file. If this is on another server type \\ double-slash, followed by the IP address and path on the another machine. 3.5.11.
Once the time and frequency of a given schedule is set the tasks from the left side of the window can be dragged and dropped in the field under the subplans list. For every subplan a ‘Backup database task’ and a ‘Maintenance cleanup task’ can be dragged/dropped and defined on this field. For example you can create 3 folders in the shared folder on the application server, one for hourly differential backup which is backed up once every hour, and the backup is deleted once every day.
Fee backup & FTP tools (for one single database): http://sqlbackupandftp.com/download/ http://www.sqlbackupmaster.com/download Script for SQL backup & FTP: http://www.morrenth.com/script-to-backup-zip-and-ftp-sql-server-database.aspx 4. Administration 4.1. MManage 4.1.1. Overview All administration tasks can be done using the MManage (MizuManage) client application.
-Help files -Uninstall.exe -Other files (depending of your install package configuration, OS version, etc) When properly installed, you are ready to login on your server(s) and/or your gateways. If you have a central server, all administration tasks can be done connecting only to the server. If your gateways are running without a server, you must connect to each gateway separately for doing administration tasks.
Almost all tasks are done by selecting an item from the left side of the main form. For detailed descriptions please read below. In the Menu you can find common tasks such as “Settings”, “Save As”, etc. The selected action usually has effect only on the current active form.
From the File Menu you can save, print or export the selected form. Usual database operations are performed from the Edit Menu. In the Favorites Menu you can see the most frequently used items. In the Tools Menu you can find a set of helper applications explained later in this document. In the Settings Menu the most important form is the “Select Direction” which will filter almost all listing used in MManage.
Other datasources wich can be accessed by ADO or ODBC. 4.2. Monitoring 4.2.1.
Currently running calls are listed here. Calls terminated on Mizu Gateways are displayed in separate list from other directions. You can filter the listing by selecting your preferences in the “Set Direction” box (as you can do in many other parts of the program). The following grouping is available: by caller, by called, by called prefix, by simowner and by sim packet. Field Explanations: Status: engine or simcard status.
CDRC: call attempt count SL: speech length (duration in minutes) ASR: average success ratio (percent) ACL, ACD: average call length, average call duration (in second) You can select any direction in the “Select Directions” Box, to check only that specific traffic. Also there are some simple groupings available: -No grouping: will display the total sum.
You can make reports based on the following fields: -CDRC: number of calls -SL: speech length (all call duration) -ASR: the number of successfully answered calls divided by the total number of calls attempted (seizures) Since busy signals and other rejections by the called number count as call failures, the calculated ASR value can vary depending on user behavior -ACD: average call duration
-ASRB: average success ration, but here the “success” means a minimum amount of duration. Configurable in Settings Menu -> Thresholds Box -ACT: average connect time.
Partner/Day: group by partner and day Partner/Hour: group by partner and hour Partner/Minute: : group by partner and minute Called Country: : group by called user country Called Direction: : group by callednumber zone Provider direction (prefix):: group by callednumber prefix Provider direction (name): group by callednumber direction Direction and packet: group by prefix and simpacket Provider direction and packet: group by callednumber zone and simpacket By caller root endusers: group by billed or company
The most common reason codes are the followings: -SIP, Bye: normal SIP close code -SIP, CANCEL: the call was canceled by the caller (not connected call) -H323, Remote endpoint application cleared call: normal H323 disconnect -H323, Remote endpoint stopped calling: the call was canceled by the caller (not connected call) -GSM, Normal call clearing: normal GSM close code -GSM, Normal unspecified: normal GSM close code -Server, Blacklisted: dropped due to ACL (blacklist) -Wrong Media: no voice activity detecte
gwrestart restart the gateway process pcrestart restart the gateway (hardware) 4.2.10. Server Monitor Will connect to the server logport. The trace level depends on configuration (Open the Configuration form, type “log” in the filter box, and hit the enter button. Then you can see all options regarding to log levels) 4.2.11. Logs Here you can see the log records for the server and every connected Mizu Gateways in the selected time period.
4.1.12. Analyze You will get detailed system analysis in this module. Thus you can see through the system by only one mouse click. Malfunctions are colored in red. 4.1.13. CDR Records After every call (and SMS, etc), a new CDR is stored in the database tb_cdrs table (and in tb_cdrresellers when the reseller option is used). CDR records can be filtered, analyzed, exported and lots of vital statistics are based on this records. CDR records will contain the following fields: Id: database identifier.
Callenddate: first disconnect code or CANCEL/BYE received or sent Connectdate: first 200 OK received or ACK for 200 OK sent Connecttime: time elapsed until call fail or call pickup (routing+ringing time) Workenddate: used for callcenters and represents the time when the operator have finished to work with the current client (CRM updates, etc) Realduration: speech length Discparty: disconnect origination. 1=called or gsm, 2=caller or h323, 3=router (server) Discreason: disconnect reason code.
Mark (marker): for special CDR records: EMAIL (e-mail), SMS (sms), FAX (fax calls), FAIL (failowered), RER (rerouted), FWD (forwarded), TRANS (transferred), CONF (conference), PRED (predictive) and to signal other important call types Opworktime: used in callcenters to store the actual operator worktime Opwaittime: used in callcenters to specify how much time the operator have been waiting for the current cal Billingstep: loaded from price settings (endusercost packet) Unitprice: loaded from price settings
Rtpsent and rtprec is 0 when media routing has failed (if we don’t route the media, or the terminating endpoint don’t send media info to us, the system will set there values to 1, so this condition will be true) All prices in the cdr records are calculated with VAT included! 4.3.14. Balance Duration lists of several traffic types. 4.3.15. Callcenter Statistics Statistics related to callcenter operations: Campaign and operator statistics.
If you need both to accept and sent traffic to another server (carrier) then you have to add it both as traffic sender and SIP server. Open the “Users and devices” form to see your users. There are a few template user created during setup. You can add new user by just cloning one existing user (select user with the appropriate type, click on the “new user” button, the select “yes” when asked for cloning).
groups For more flexibility you might use direct SQL queries against the “tb_users” table.
11: isdngw, (parentid is the gatewayowner) 12: sms, fax,email gateway 14: support users (can operate with MManage, has ftp account) 15: admin users (can see and modify everything, can receive email/sms reports) 17: other users (for various reasons) ParentID: This field is very important for resellers/subresellers relationships. Logical parent of the user. Checked for routing pattern, max lines, etc.
ShortTelnumber: sip short telnumber (for example if several users has the same BelongsToCompany field) DisplayName: how the user will be displayed. Can be null Username: the most important field. Used for authentification and also as a DID number. This field is unique and cannot be empty. Password: password applicable everywhere (sip, web, VPC, etc) Ip: sipphone, sipproxy or gsmgateway ip address.
Postpaid: if the user will prepaid or postpaid PaymentMode: Check (0), Bank Tranfer (1), Cash (2), Else (3) ContractNumber: contract for end-users Allowedpartners: Allowed traffic senders for the gateway, or allowed gateways for traffic senders. A list of user id separated by comma or ‘*’ Note that parent users will be checked too Enabledprefixes: can be one prefix (with any length) or a list of prefixes with 3,4 or 5 digit separated by comma. Can be used for trafic senders and gateways too.
3= route rtp if both behind nat 4= route rtp if caller is behind nat 5= route rtp if called is behind nat 6= route rtp if any endpoint is behind nat 7=always route rtp Rtp settings will be checked first for the called and then the caller (so if the caller RouteRtpCaller settings is not 0, then it will overwrite the called RouteRtpCalled settings) RtpIp: last rtp ip RtpPort: last rtp port ServerRtpPort: last bind (we try to use the same for every user) NatDetected: 0= no and don’t change, 1=no but can be cha
CurrCallCount: current running calls (usable for traffic senders) enablefakegw: if we don’t have capacity, we can route h323 calls to a fake gateway to prevent congestions candisablesim: if the router will check the disableduntil field from tb_sims alarmat: we can ring the sipuser if it is set forwardonbusy: telnumber where we have to forward the calls when busy forwardonnoanswer: telnumber where we have to forward the calls when we have no answer forwardalways: rerouting voicemail: if we can send messages
autopriority: set by server.
Denyaddr: because the server will try to send the sip messages to all possibile addresses, sometime it will missroute. With this setting you can restrict the address posibilities. Check the FAQ for more details. sendfakealert: used for gsm gateways. Specifies the timeout in sec after that the gsm gateway will send an alert to voip even if no ringing have been received from gsm network. Set to 0 or -1 to disable.
selecting only one code. If set, than only one codec will be left in the sdp (plus the dtmf codecs). This will help, when the server answer to invitation with more than one codec in the 200. The client should answer with the final codec in the ACK, but many endpoint fail to do so. SessionTimer: use session keep alive.
Default users: Owner mycompany: template for owner
GsmGw LOCAL_GW: used for advanced gateways. IP=127.0.0.1 GsmGw NOGW: used when no route found. IP=1.2.3.4, isfake=1,realgw=0 GsmGw FBACKUPGW: used to handle traffic exceed. IP=127.0.0.1, callsigaddr=1725,isfake=1,realgw=0 SipProxy: sip2h323: convert signaling form h323 to sip and from sip to h323. TechPrefix=-1,IP: 127.0.0.1?, Port? TrafficSender: virtserver X: for routing traffic from virtual servers. AuthIPList: 192.168.0.1, TechPrefix=XXX Enduser: PREDECTIVE_DIALER: IP: 4.3.2.
Password: sip password used in authentification CallerNumber: usually the same as username. If left as blank, then the server use the actual caller username. Credit: account balance.
banned. For example if there are too much meaningless or not authenticated request from an IP address (probably the attacker), than that IP address will be banned for a time period and the incoming messages from that IP will be silently dropped. Users and devices will be allowed to access the system (and create new dialogs) only if they pass basic authorization which can be set from MManage -> Users and Devices -> Edit tab.
The server can authenticate the user based on the following methods: ANI/CLI authentication: if the CLI is known and this method is allowed. A number authentication can be used to try user authentication for a call coming from a traffic sender. If user is found with the actual A number then the caller will be authenticated as the enduser, otherwise will be authenticated as traffic sender. In this case you can require a PIN number from the user.
When using the pin field based authentication, make sure that user has valid pin codes set in this field (when the users are automatically generated, the pin is set to be username+password). User rights User rights can be further restricted by several configuration option. The most useful tool for this is the routing table. You can define were a certain user or a user group can initiate calls.
callmaxwait: max waittime allowed for operators between calls (for administrative purposes) enforcestrongauth: enforce authorization and strong passwords Try to avoid prioritizations by users, gateways, simpackets or channels (absolutepriority, priority, allowedpartners, prioritypartners, etc) Almost all kind of configuration can be set up by using only the “routing” form. 4.4. Routing 4.4.1. Dial patterns The “dial patterns” terminology is not used with the mizu voip server.
4.4.3. Normalize numbers By default the server will "normalize" all numbers. This means that it will clean the numbers from garbage characters, sql injection attacks and might remove/add IEC/CC/NEC prefix based on the circumstances and global configuration (international escape code, country code).
normalizenumbers: how to normalize 0=no,1=basic,2=normal,3=more (hu/ro),4=extra default is 2 normalizecallednumbersnec: NEC normalisation (National Destination Code) -1=depends on normalizenumbers and normalize_localpx, 0=no,1=yes,2=extra default is -1 rewriteinternalcalled: rewrite the called numbers for calls to endusers 0=don't set,1=don't rewrite,2=client to as received,3=client to username,4=client to telnumber,5=client to best if match,6=all to as received,7=all to username,8=all to telnumber,9=all to
country: country (2 chars) iec1,iec2,iec3: international escape codes countryprefix: default country prefix (country code) prefix1,prefix2,prefix3: national destination codes cfg_mobileprefixes: list of common mobile prefixes (useful for callenters to differentiate landline and mobile calls) Deprecated settings (don't use these anymore): normalize_localpx: replaced by normalizecallednumbersnec setcalledtosipusername: replaced by rewriteinternalcalled normalize_clean: replaced by validateinput normalize_che
More settings: forcedcallernidentity: user and global setting //0=no,1=default checks (best num; default),2=dbusername,3=dbsipnumber,4=bestnumber,5=first good num,6=check minassertedidentitylen,7=with phonenumber,8=anonymous affects only P-Asserted-Identity will owerwrite any cli settings default is -1 replacecalleroncalls: user and global setting //0=no,1=default checks (best num; default),2=dbusername,3=dbsipnumber,4=bestnumber,5=first good num,6=check minassertedidentitylen,7=with phonenumber,8=anonymou
user setting replace client->callernumber (controlled by config.identityrwmode) //config.identityrwmode = 2; //0=no rewrite, 1=basic, 2=conform sip specification (identity) //config.
global config option used only for callforward //config.replacecalleronforward = 9; //0=no (default),1=yes,2=yes with username as callername, 3=yes with phonenumber as callername, 4=yes with username as authname too, 5=yes with phonenumber as authname too 9=auto parseidentity global config option 0=no at all, 1=normal,2=rewrite caller number,3=rewrite callername also default is 1 parsesipdisplayname global config option 0=no,1=callername,2=displayname,3=can be used for auth default is 1 4.4.5.
4.4.6. Prefix rules You should instead use the “Rules” form mentioned above. You can rewrite prefixes before they arrive to the routing by entering your preferences here. The Mizu routing engine will accept only 3 digit length techprefixes or no thechprefix, so you must convert them here if your traffic sender will send the traffic with techprefix that are not three digit length. Example 1: To set up a rule which defines that every incoming number from ip 111.111.111.
Input parameters: @calledat TINYINT, /*1=first check, 2=after authentication, 3=before routing out, 4=after routing out*/ @protocoll TINYINT, /*0=SIP, 1=H323, 2=GSM, 3=Other*/ @fromip varchar(22), /*caller ip address*/ @fromport SMALLINT, /*caller port*/ @callerid int, /*caller device database id from tb_users*/ @callernumber varchar(35), /*caller number or sip username*/ @callername varchar(35), /*caller name or displayname*/ @calledid INT, /*called device database id from tb_users*/ @origcallednumber varc
sql help: sql tutorial: http://www.w3schools.com/SQL/sql_intro.asp stored procedures: http://msdn.microsoft.com/en-us/library/aa174792(SQL.80).aspx string functions: http://msdn.microsoft.com/en-us/library/aa258891(SQL.80).aspx like operator: http://msdn.microsoft.com/en-us/library/aa933232(SQL.80).
4.4.8. Blacklisting List the blacklisted numbers on the selected time interval and direction. This query will generate high server load. Use it only in off-peak time if possible 4.4.9. Access Lists You can define the “Blacklist” and the “Whitelist” here. The listing will be appreciated in the routing depending of the actual packet “filtering” setting. Check section 4 for details regarding filtering. Blacklist fields: -telnumber: country code + operator + telnum -sure: levels tb_blacklist.
In Caller Prefix, you can place only one prefix. Tech prefix can be empty string, asterisk (*) or 3 digit length number. Called prefix can be one prefix (with any length) or a list of prefixes with 3 or 4 digit separated by comma. *Tip: you don’t need to enforce traffic sender rights by routing. The routing can be done as generic as possible for example by specifying only Called Prefixes (Leave the other direction option blank or ‘*’).
Routing Configurations Try to set up all routing rules and prioritizations using this form.
Caller “CanDial” setting is set to false Caller tb_users.
Called gateway and simcard absolutepriority Positive routing priority (deprioritze simpackets with negative routing priority -these are “emergency packets”) SimPacket absolute priority partner (absprioritypartner -if set and if match the caller) Simcard caller priority (absprioritypartner -if set and if match the caller) Gateway absolute priority Gateway called priority (if set and if match the caller) Simcard absolute priority Routing list priority/100 (differences more than 100 in priority list) Called ga
Technical description: Call arrive from traffic sender or enduser via SIP or H323 Check if MAXCCALS restriction reached (licensing option). Drop if yes. Check if maxslperdayreached reached. Drop if yes. Check if maxroutereqpermin reached. Drop if yes. Check if the current call is a routing retry (forked calls). Drop if too much retry Normalize caller ip addres Check if the call was from the local SIP2H323 module. Return with the already prepared target if yes Chech if the call was arrived from GSM gateway.
To activate BRS based routing, the “brs_lcr” global configuration option have to be changed to 4 or 5. Then the routing will check both the pricing and the routing, which you can finetune using the “BRS” form (only after you have some traffic, so the table is populated with the statics. If you would like to change the default values, you can do this only directly in the database). Finetuning means the change of the following fields: accuracy,minprice,maxprice,minasr,maxasr,minacl,maxacl.
Fields have the following meanings: -Id: database identifier. Auto increment -Gateway: gateway id (called) -Direction: called prefix -QualityPercent: how much the quality will contribute to the final result. If price is very important for us, set this value lower. Default is 50% -Accuracy: how accurate the final result will be. If we set it too high, then we probably will have only one route as the best. If we set it too low then too little discrimination will be made between routes.
-MinPrice, MaxPrice: min-max prices/minute. set it to a very wrong price to that direction and the max value to a very good one. Calculate it with consideration to billing step and min minutes (so you must fill in as 1/1 price) -PriceCalcSec: we estimate the price values with this value to get a gross value -TryedCount: how much time we have tried this alternative route until now. Helps us the decide how to increment NextTry.
GatewayID: called gateway or sipproxy Direction: called direction (prefix) MaxSubsFail: if we get more wrong calls than MaxSubsFail we failover to the next route if any MinASR: if we get more lower ASR than MinASR we failover to the next route if any MinACL: if we get more lower ACL than MinACL we failover to the next route if any MinCallCount: we calculate ASR and ACL statistics only if we have MinCallCount cdr SubsFails: current subsequent wrong calls detected NoPriority: We have done a failover until thi
Note: failowering will occur with increased thresholds (more slowly) if the priority between the routing directions (SIP servers) is more than 100. 4.4.15. Channel reservation Skip this chapter if you are not using VoIP-GSM gateways. Best quality (ASR and ACD) SIM channels can be reserved for sip or h323 originated calls.
tb_portednumbers: id: autoincrement database primary key number: original (normalized) called (B) number providerpx: new prefix (for example instead of 3630 changed to 3620) newnumber: the changed number newdomain: the new service provider ip or domain newport: service port (defaults to 5060) priority: checked for duplicate numbers datum: record insertion date You must have the providerpx OR newnumber OR newdomain: newport specified. If the newnumber is set, than the providerpx has no effect. 4.5.
Field descriptions: Title: the name of the invoice group
Schedule: how often the report will be generated DueIn: allowed time for payment in day (used only if the report is an invoice) Status: billing status Invoice Type: specifies the format of the invoice Group By: specifies the format of the invoice Separate by caller: every caller will receive a separate invoice (used for billing to end-users) MailTo: list of email address where to send the generated report Last Invoice Sent: date-time when the invoice was emailed to the recipient Last Payment Received: date-
These directions will be checked in priority order on routing and billing ValidSince, ValidUntill: the pricelist may be applied only after a specified date-time Prefix: called number prefix (this will be loaded after “best fit”).
The following prices are calculated: -enduser cost: used for invoicing for costumers -provider cost: cost that needs to be paid for service operators -sales cost: sales commission. If not defined in price setup, then will be loaded from user’s settings (“commission”) if any -company cost: usually used for profit calculations -other cost: for any other purpose Billing can be done from 1.
Money Precision: how many floating point digit would you like in money fields. Completion date: defaults to the end of filling period if not modified Method of payment: can be specified here, or loaded from user setting. By clicking on the “&Generate report for the selected directions” button, you can generate the actual invoice(s) 4. Invoices and Payments The invoice records for the selected user(s) are in this form. You can watch the debt for every user by checking the topmost record debt value.
Lastbilled: last time when the it was invoiced Description: any comment here 4.5.4. Currency Conversion You can set currency in 3 places: 1. -native currency in the global configuration “currency” This will be the default currency for all internal operations 2. -currency for pricelist packet When you receive pricelist in other currency, with this setting you can easily convert it to your native currency 3.
You can generate random prepaid codes here. Prepaid account can be charged over the website or by ivr: Website operation: -user authentication (tb_users.username and password) -check if pincard is valid (tb_prepaidcodes) -increase credit for the user (tb_users.credit) IVR operation: -automatic user authentication based on sip registration or require entering the phone number to be charged -require pincode -check if pincard is valid (tb_prepaidcodes) -increase credit for the user (tb_users.
2: Peak 3: Offpeak 4: Weekday 5: Weekend 6: Offpeak and weekend 7: Evening 8: Night 9: Holiday 10: All times 11: Other Times (Rest) example: v_getprice '1','36301234567',6555,'003615555555','150',666,-1,4,2,99,99,4,11,41,500 The v_getprice stored procedure will return the following fields: tb_billentries.currency,tb_billingtimes.isdiff,tb_billingtimes.origprice,tb_billingtimes.price, tb_billentries.billingstep, tb_billentries.minammount,tb_billentries.freeammount,tb_billentries.freeafter,tb_billentries.
The following fields are defined: ID: autoincrement database primary key Type: 0=All or Recreate (technical) 1=Report 2=Proform 3=Advance 4=Invoice 5= CreditNote 6=Storno 7=Correction 8=Payment (technical: payment received) Copynum: printed copies CompanyID: emmitent company ID (tb_billsources) UserID: costumer ID from tb_users (if any) User_name: costumer name or company name User_address: costumer address User_regnum: costumer registration number User_euregnum: costumer eu registration number User_bank:
tb_invoice_entries: Id: autoincrement database primary key Invoiceid: foreign key to tb_invoices datum: record insert date description: product description with code fromdate: billing period if applied todate: billing period if applied Ammount: ammount AmmountName: name of the amount (minute) AmmountPrice: unit price (net) The emmitent (company) settings are stored in the tb_providers table. Only one company entry can be stored (conform to Hungarian laws).
Chargecards can be generated from Billing -> Pincodes CallingCards: There is a special user type called “callingcards” but any usual user can act as a calling card. Users can access the system via Web or IVR typing a pincode. The pincode will represent the “pincode” column from the user table or the username+password combination for enduser or only the username field for calling cards. PayPal: direct or indirect handling of PayPal payments are supported.
Iongate iTransact RediCharge HTML LinkPoint Merchant Anywhere Merchant Partners Moneris MPCS Weblink NetBilling Network Merchants NexCommerce NOVA's My Virtual Merchant NOVA's Viaklix OGONE Optimal Payments PayFlow Link PayFlow Pro PayFuse - First National MS Paygea PayJunction Trinity Paymentech - Orbital Payment Express Payments Gateway Payready Link PayStream Planet Payment Plug 'n Pay PRIGate Protx PSIGate RTWare WebLink SECPay SecurePay SkipJack www.iongate.com www.itransact.com www.linkpoint.com www.
Sterling SurePay / YourPay TransFirst eLink TrustCommerce USA ePay uSight Verifi Verisign PayFlow Pro WorldPay Select Junior Invisible YourPay and more ... www.sterlingpayment.com www.surepay.com www.transfirst.com www.trustcommerce.com www.usaepay.com gateway.usight.com www.verifi.com www.verisign.com www.worldpay.com www.yourpay.com Search for epayment in the global settings for the configuration details. To enable the E-Payment module follow these steps: -1.
4.5.12. Promotions • • • • First X seconds are free You might create packets when the first few seconds are not billied. Just use the “Free amount” option on the Price Setup form for this Call X direction for only … cent Just setup a separate packet with lowered prices for this. Use only the directions (prefixes) you wish to promote. 10 USD free to direction X Set the “packetcredit” field in tb_users to 10 Create a special packet with the desired directions (prefixes).
4.5.13. Notes Call forward billing: 2 cdr record will be generated. A->B and B->C Call forward from IVR: one cdr will be generated. Whether we charge the call to the IVR or only bill the forwarded call can be controlled by “resetdurationonfwd” Call transfer by SIP signaling: the second call is completely different from the first call.
0-no filter: allow all numbers 1-allow blacklist „sure” level: 0,1 and 2 (tb_blacklist) 2- allow blacklist „sure” level: 0 and 1 3-allow only blacklist „sure” level: 0 4-block all blacklist 5-allow only knownnumbers (listed in tb_knowngoodnumbers) 6- allow only knownnumbers that are 100% ok (sure is 1 in tb_knowngoodnumbers) Dialplan: 0: international number format with 00... (e.g.: 003630xxxxxxx) 1: international number format with +... (e.g.
The “chargecode” string in the message will be replaced with a valid code if found. You can introduce delays by inserting ‘#’ characters in the message. The action parameter can be -0: used to send USSD messages The message parameter must have the following format “AT+CUSD=command” where command is the ussd string. Example: DTMF,0,simid,"AT+CUSD=1,*121*chargecode#" -1: will send the specified message to the engine.
Status Filter: Existing lines: List only current running channels. (this doesn’t mean that the channel is workable. We list all channels who have reported there status in the last 5 minutes) Good lines: only workable lines are listed.
Active and not used: Working simcard without calls on it Monitor: simcards grouped on gsm channels. You may detect missing “holes” very easily by scrolling down this list. This listing is almost the same as in the “Line Monitor” form. Field Explanations: ID: database unique identifier SIM ID: sim identification number (you can find this number written on the simcard) IMEI: unique gsm engine identifier Monitor: the status of the channel.
Packet: the type of the SIM Card TodaySpeachLength: the number of active minutes on the current simcard since 00:00 ThisMonthSpeachLength: the number of active minutes on the current simcard since the first day of the current month ThisMonthSpeachLengthPeak: the number of minutes since the first day of the current month in peaktime ThisMonthSpeachLengthOffPeak: the number of minutes since the first day of the current month in offpeak times ThisMonthSpeachLengthWeekend: the number of minutes since the first
AllCallCount: all call attempts on the simcard until now AllWrongCalls: all wrong calls on the simcard until now (speech length below the predefined value) AbsolutePriority: if you set it higher then on other sims, all calls will be routed here primary Priority: routing priority boost Filtering: determines how we check the blacklist and the known numbers 0-no filter: allow all numbers 1-allow blacklist „sure” level: 0,1 and 2 (tb_blacklist) 2- allow blacklist „sure” level: 0 and 1 3-allow only blacklist „su
4.6.5. SIMCards Same as “GSM Channels”. See section 4.2.2 The first field will show the status of the simcard (Monitor). The most frequently used values are the followings: Unknown: the last list refresh is too old. Status cannot be determined. Click on the reload button to refresh Missing: simid not found.
Note: dialing, ringing and call ending messages may not be shown in the monitor depending from the gsm gateway configuration. If the “sendallstatus” setting is set to false, than instead of “dialing” and “ringing” only the “speaking” message will be shown. 4.6.6.
4.6.11. SIM Bank In the SIM Bank form you can monitor the sim flying activity. 4.7. Other -MManage 4.7.1. Configurations Global system configurations. Basic configuration are vital for the system to run correctly.
The following config settings are defined: Category Key Description CallCenter allowdbcalls Allow calls from database in MAgent CallCenter allowmanualcalls Allow manual calls from MAgent CallCenter allowopcampchange Allow operators to change its campaign from MAgent CallCenter autoformname Type of AutoCall GUI to load CallCenter callbackautorecall 1= schedule for recall if number is in campaign CallCenter callbackhandling handling incoming calls: 0=dropp all CallCenter callbackivr pl
CallCenter maxrecalltrycount max attemt of REcalls for a client CallCenter mobileratio_08_12 percent of mobile calls in the specified period [1-100] callcenter mobileratio_12_16 percent of mobile calls in the specified period [1-100] CallCenter mobileratio_16_20 percent of mobile calls in the specified period [1-100] CallCenter mobileratio_20_08 percent of mobile calls in the specified period [1-100] CallCenter mobileratio_weekend percent of mobile calls in the specified period [1-100] Cal
license CAN_hash323 default license (will have no effect if you change it here) license CAN_hassimbank default license (will have no effect if you change it here) license CAN_hassimplatform default license (will have no effect if you change it here) license CAN_hassip default license (will have no effect if you change it here) license CAN_recharge default license (will have no effect if you change it here) license CAN_runsipproxy default license (will have no effect if you change it here)
licensecfg hassip modules to load (has effect only if not prohibited by builtin license restriction) licensecfg maxallusers limitations (has effect only if not prohibited by builtin license restriction) licensecfg maxcallspermin limitations (has effect only if not prohibited by builtin license restriction) licensecfg maxccalls limitations (has effect only if not prohibited by builtin license restriction) licensecfg maxccallsblock limitations (has effect only if not prohibited by builtin licens
settings checkcallerids wich calls are to be checked on the selfcheck thread (useful if you have wrong traffic senders) settings checkcputime if cputime is constantly high settings checkcredirights if to check the owner of the sim and chargecards settings checkgkcdrs if to check cdr records from the gk statusport settings checkgsmnumlen if to allow incoming number only with this size settings checkknownnumbersex 0 = no settings checklocalnumberpx local endusers prefix to check (to not ro
settings dbmaint_backupdbdir database backup directory settings dbmaint_backupdbnetworkdir database backup directory path (if the database engine is located on a remote server) settings dbmaint_backuplevel 0=no backups settings dbmaint_backuptables backup cdr records and other tables to xxx_backup: 0=no settings dbmaint_do do database maintanance settings dbmaint_removecdrs remove cdr records after x days settings dbmaint_removelogs remove logfiles after x days settings dbmaint_removeo
settings faxsuffix fax sender configuration (email to fax server) settings faxuser fax sender configuration (email to fax server) settings fileloglevel file server loglevel (0=only errors to monitor settings filetransferbufflen fileserver buffer length settings filetransfertick fileserver speed settings freenumlen number length that can be called free of charge settings gkcommand how to start the h323 gatekeeper settings gkstoptick gk setting settings globalcdrid server generated.
settings maxpriceperminute will alarm if providerprice is bigger settings maxrecdiff recorded voice stereo sync in msec settings maxrecdiff2 recorded voice stereo sync in msec settings maxroutereqpermin allow only "maxroutereqpermin" routing request/minute settings maxudpselect max socket on select (set to -2 to autoconfigure.
settings ppriority 0=low settings removetrailinghash remove # when routing settings restartatnight set to true if you want a reset every night settings restartpcatfirst restart the pc immediately on error (don't restart the sw) settings rotatelogfile create separate logfiles for every day settings rrr_black blacklist q931 disconnect code. defaults to ResourceUnavailable settings rrr_denied access denied q931 disconnect code.
settings voicebackupdir where to store a backup of recorded conversations settings weekendispeak peaktime settings for billing and routing settings weekendispeaktr peaktime settings for various operation settings wrongnumcache remembered callednumbers. used if usewrongnumfilter is not 0 settings wrongnumdropcount drop call if this number of failed calls found in cache. .
SimPlatform gkrstifnotconnected restart the gk if not connected SimPlatform gsmgwport port for the gsm gateways SimPlatform gsminccalled forward incoming call from gsm gateway to this number SimPlatform gsminccaller convert incoming caller number from gsm gateways when forwarding to a support phone (incall action in gsm gatew SimPlatform gsminccallerip gsminccallerip SimPlatform gsminfromip fromip SimPlatform gwstatistics to build the table or not SimPlatform incomigcalls simulate inc
SIPSettings CanAcceptLocalIp Can call from 127.0.0.
SipSettings maxstatchangepermin max allowed enduser status changes/60 sec (slower if exceed) SIPSettings MAXSUBSMSGCOUNT max subsequent messages before block SIPSettings MAXSUBSMSGPERIOD max subsequent messages before block are checked for this interval (sec) SIPSettings MAXWRONGMSGALLOWED dos attack protection SIPSettings MEDIATIMEOUT will disconnect if the media disappears SIPSettings MEDIATIMEOUTSTART will disconnect if no media detected SIPSettings MINRESENDIVAL sip udp resend timer
SipSettings supportlist privacy SIPSettings traceep1 user id SIPSettings traceep2 user id SipSettings udpkeepalive send keepalive messages SIPSettings udppriority rtp thread priority: 1=normal SipSettings upperexpire register expire SIPSettings usedateheader send date to user agents SIPSettings userofflinemin enduser will be considered offline if no register or invite for this period supervisor canrestartformalfunctions if supervisor can do restarts.
4.7.4. Test Call H323 test calls can be done here. 4.7.5. Rfile system Upload/download files from gateways. 4.7.6. Rdesktop Use this form to login directly in gateways and on the server. 4.7.7. DB Admin Database administration tool. Only for database experts! 4.7.8. Web Admin Direct link to the Costumers website if you have any.
4.7.9. Phone Numbers Numbers allocated by authorities. You may add new endusers with telnumbers set to a free number from this database. Don’t forgot to set the “free” field to 0 if the number is allocated to an enduser. The web interface will get free numbers for newly registered users from this database too. 4.7.10. To-do You can define tasks for technical support with the ease of this form. 4.7.11. Notes Any quick note here to be shared between the support and admin users. 4.7.12.
o o o o Ring a phone number Restart the VoIP service Restart the SQL service Reboot the server 4.8. Gateway Configuration Skip this section if you are not using VoIP-GSM gateways. All configurations can be done from the MManage Client Utility GUI and the VnetCfg utility. For better understandings we present the gateway configuration settings here: 4.8.1. Phone Settings [PhoneX] //serial port PortNumber=1 //control port (not used in 1.
simcard3= Simchange settings explanation: format: simchange1= 2004.03.05/13:00:00 - 2004.03.07/13:00:00 - 8936302403070132426 (from date - to date) or simchange2= 10:20:00 - 10:26:00 - 8936302403070132426 (every day from time to time) or simchange3= 2/10:20:00 - 7/10:26:00 - 8936302403070132426 (from Tuesday 10:00 to Sunday 10:00) or simchange4= 6/00:00:00 - 7/24:00:00 - 8936302403070132426 (Saturday and Sunday) there is a priority order from top to bottom (simchange1, simchange2, etc.
onlyg72x=1 //useserver if false, then don't connect to the simserver. will save cdr records to file. may be limited due to licensing options useserver=true //load configuration from the server (at startup, at regular intervals and when specified) loadcfgfromdb=true //gatekkep ip address (leave it empty if you don't want arq registration) gkip= //gatekeeper H.235 security gkpassword= 4.8.3.
record=0 //what kind of logs to send to server (1-5) tracetoserver=1 //process priority priority=1 //ModemControllPort used only with hw 1.0 controlportnumber=1 //if we use prefXXX settings useseparatesettings=0 //signaling endpoint port. Defaults to 20001 mintcpport=20001 //max h323 signaling endpoint port. Defaults to 29999 maxtcpport=29999 //min h323 udp endpoint port. Defaults to 36000 minudpport=36000 //max h323 udp endpoint port. Defaults to 37999 maxudpport=63999 //min media port.
//wrong call criteria wrongcallmaxduration=30 //call duration limit in sec (defaults to 3 hour -10800 sec) callimit=10800 //max time to wait for ring signal from gsm network in msec maxringewait=36000 //ring limit in msec (defaults to 52 sec) maxringtime=52000 //deprecated statusintervall=600 //do Q931 progress indication doprogreessindicator=0 //reset the gw if we have fewer lines minactivelines=2 //delay of initialization of the lines (msec) initdelay=2200 //delay of registration of the lines (msec) destr
faststart=1 //used for debug purposes ringtime=6000 //desktop access desktoppwd= //if set, then will try to autologin loginpwd= //if we want to play a background sound backgroundsound=0 //4 or 8.
nosmsread=0 //socket read/write timeout and system checks operations modifier. default=4 timeoutmultiplier=4 //backup server address serverip2= //route incoming calls here (defaults to serverip if not specified) outserverip= //keep connected to the internet (redial, reconnect, repair, enable/disable network interface, restart) keepinternet=1 //ethernet interface name. configure from the vnetcfg tool net_interfacename= //network connection type (STATICIP/DHCPIP/ISDNIP/ADSLIP,CARDNAME).
inccalls=1 //file to play when requesting number to call on dtmf (when incalls is 4). "please enter phone number to forward call" playdtmfreqfile= //file to play when requesting number to call on dtmf failed (when incalls is 4) "forwarding failed" playdtmffail= //file to play when requesting number to call on dtmf succeed, and forwarding begins (when incalls is 4) "your call has been forwarded.
MAXPCRESTARTIVAL= //max time to wait for watchdog reset. defaults to 1000*60*20 msec MUSTRECEIVEOKIVAL= 4.8.5.
3. When incalls is set to 2 -call will be forwarded to the number specified by the “forwardnum“ option in the GSM network. -the simcards must support the forwarding options, otherwise this operation will fail 4. When incalls is set to 3 -the call will be forwarded to the Mizu server specified by the “outserverip” setting in the gateway configuration. -on the server, the call will be forwarded to the “gsminccalled” number (SimPlatform configuration).
Each simcards can have it’s own GSM engine (in other gsm gateway the engines are used by more simcards) GSM Cell Lock Because Mizu GSM Gateways use only 8 channels, they don’t overuse the gsm network. However, you can setup individual GSM channels to use separate cells Virtual Simcards With the ease of Mizu simbank, your simcards can be stored in a central location, and used in gsm gateways installed at different locations.
For advanced gateways and servers the following port ports must be forwarded also: -TCP: 1720 (default h323 GK signaling) -UDP: 5060 (default SIP port) -TCP: 9885, 9886, 9889 (gsm server, admin port, log port) -TCP: 1433 or 2223 (for SQL server) -TCP: 80 (optional for HTTP) *you may use other ports than the defaults listed above 4.9. Call Center This section will describe the outgoing callcenter module but can also be useful to read when using the incoming callcenter module (e.g.
4.9.3. Scripts Every campaign can have different operator instructions. These instructions can be defined in this form. For every step (question) the operator can select from different actions (answers). The call will follow these selected instructions. Pay attention to cover all possibilities. tb_ccscripts: (used for questions) id: autoincrement PK campaignid: the campaign where this script belongs to ordernum: display order (used when jumping to next question) title: script alias.
id: autoincrement PK questionid: answer belongs to this question ordernum: display order answer: the answer text alias: will be stored as answer (usually the same as the ‘answer’) actioncontext: can be used for storing reject reasons or for data imput field specification fields: used grids to determine its columns. Values must be separated by comma ‘,’ or semicolon ‘;’ jumpto: next presented question.
set the datainputtype to “Run” specify application name and parameters in the “Command” field. keywords can be used as part of command. the following Parameters are defined (separe them with comma) -hide: will launch the application hided -wait: will wait for the application to terminate Text Mask: Use EditMask to restrict the characters a user can enter into the masked edit control to valid characters and formats.
Any character that does not appear in the preceding table can appear in the first part of the mask as a literal character. Literal characters must be matched exactly in the edit control. They are inserted automatically, and the cursor skips over them during editing. The special mask characters can also appear as literal characters if preceded by a backslash character (\).
all script answers (including data input) will be saved to tb_ccscript_processing list answers will be separated by semicolon ; grid answer will be saved in separate columns where the column title will be the tb_script.title + the grid column title 4.9.4. GUI Designer The MAgent Automatic Call form user interface can be customized in the MManage -> GUI Designer. Basic MSWindows controls and specific controls are supported.
-Next Client: force jump to next client (will hang-up the current call if any) -Recall: set client for later recall -Reject: will set the rejected flag -Hold: call hold (mute all directions) -Display call status: display called party name and call duration -Call Statistics: operator statistics 4.9.5. Quotas You can define campaign target quotas by launching the “CC Quotas” form from MManage. By defining quotas, you can restrict your calls to well defined target groups (called clients).
Note: call record must reach a completition question to be valid in quota calculations 4.9.6. Presentations Used to store the different presentation locations. When a client is invited, the operator will select a presentation for them. Presentation can be selected dynamically in MAgent by setting the sql condition in the script form. It can be done by tree way: 1.
-City (string) -Age (number) -Passport (0 if unknown, 1 if no or 2 if yes) -Married (0 if unknown, 1 if no or 2 if yes) -Sex (0 if unknown, 1 if no or 2 if yes) -Robinson (0 if unknown, 1 if no or 2 if yes) -Address (string) -Comment (string) DBF files must contain the following columns (can contain other columns too): -UNEV: user name -TEL: phone number -VAROS: city -UTCA: address -ROBIN: robin -IRSZ: zipcode At least the landline or mobile phone must contaion a valid entry. 4.9.9.
comment: any comment (can be filled by operators) randorder: deprecated numstatus: wich number is present. 0=both are missing,1=only landline,2=only mobile,3=both (denormalized here for speedup) guidrand: used when clients need to be called randomly (not in database or A-Z order) rejected: 1 if the call was rejected (rejected by user not by the phone) rejectedreason: optional comment when rejected 4.9.10.
callbackivr: play special messgae (callback.
4.9.11. Predictive dialer To restrict the operator wait times, the calls can be prepared on the server side and dropped to operators when they are waiting for it. There are separate predictive threads for all campaigns.
- Waituntillconnected: if set to false, than will dropp started call when no more operator waiting (no more "No free operator on predective" disconnect reason) 4.9.12. Outgoing callback There are two types of callbacks: 1. operator set Recall (1) manually 2. server receive an incoming call and set to Callback (2) automatically Client records will be marked when callbacks or recalls are needed.
2. By specifying a callbacknumber and setting the callbackhandling option. 3. Handling by the IVR 4. Make incoming campaigns Incoming callers (clients) identity can be loaded from database when the call arrives. If the client data is not found, then can be added to database automatically. In the MAgent the call can be handled in the following ways: 1. Dropp all 2. Show in Manual Call 3. Show with client data 4.
other=add to the specified campaign (campaign id) Handlingincoming: 0=not handled 1=allow to search the current campaingn for the incomig number 2=allow to search and edit the current campaingn for the incomig number 3=allow to search the whole database for the incomig number 4=allow to search and edit the whole database for the incomig number 5=show scripts form Handlingincoming: 0=not handled 1=allowed 2=show scripts form IncSearch 0=no automatic search 1=allow to search and edit the current campaingn for
scriptquestions => tb_ccscripts scriptanswers => tb_ccscript_answers quota(s) => tb_ccquotas The following keywords are defined: -all database fields from tb_cclient, ccampain, tb_ccampain_clients, tb_ccscripts, tb_ccscript_answers and tb_ccquotas -[currentnumber] current called number -[callduration] call duration -[ringduration] ring duration -[callstarttime] begin time -[callcount] number of call attemts -[currdatetime], [currdate], [currname] current time/date/date-time -[currweekday], [curryear] weekda
storedani dtmfstoredtrimmed opname currdatetime currdate currdatetimesql currtime currweekday curryear currmonth currday currhour currmin currsec calleruserid calleduserid proxyid origcallernumber origcallernumber_nq orignumber_nq realcallernumber callernumber callernumber_nq number_nq callername origcallednumber callednumber calledname techprefix called_norm auth_username auth_password transportip fromip rtprecip transportport fromport rtprecport callertype usertype callstate cc_clientid cc_ccid credit max
currency centname disccausetextex otherpartyname otherpartydisplayname otherpartyfullname Embedding controls in texts: Basic: Keyword embedded in square brackets [xxx] will be replaced with dynamically loaded data. (Label) Keyword embedded in brackets {xxx} will be editable (loaded and saved to/from edit control) (Edit) Advanced: You can define other controls by the following text rule: {ControlName|Width|BindValue|Item1|Item2,...
4.10.2.
4.10.3. Calls from database Call to any client presented in the central database. 4.10.4. Automatic calls Will handle calls automatically if the operator is part of a campaign. 4.10.5. Automatic software upgrades Mizu client applications can check for newer version and selfupgrade. The following modules are responsible for handling software updates: -TUpdater.exe: to download the new software from a central directory (usually ftp) -TUpdLaunch.
4.11.1. Call Rerouting Failed calls (error code, no answer) will be rerouted automatically to the next device according to routing preferences. 4.11.2. Ring Groups User devices can be configured to ring in different locations (by filling tb_users.ringgroup with phonenumbers/usernames separated by comma). Single line extensions are supported. 4.11.3. Caller ID You can setup the caller id display/hide (CLIP/CLIR) function by changing the CLI setting for the actual user.
Usually In-Band DTMF are supported by any vendors in endpoint devices but will not be parsed on IVR servers because of high computation requirements to decode the encoded RTP channels.
4.11.7. Call Transfer -The standard sip calltransfer protocol is supported (REFER, replaces methods) using the transfer button on sip phones, but because many sip devices have problems with call-transfer, the DTMF mode transfer is supported. If the transfer fails, than the Cisco “Also” method is also supported for transfer failover.
4.11.9. Call Waiting and queuing Call waiting scenarios are supported. Need to be implemented in UA to work. Group hunting and queuing are implemented only in callcenters. 4.11.10. Call Take-Over You can pick up or transfer an incoming or existing call to any other device, by dialing *7*origcalled on the target device. You must be in the same user group or billed user for this feature to work. The SIP replace method is also supported. 4.11.11.
The exact location will be: serverftpvoice\databasename\currentday\voice.xxx A separate backup can be created in the directory specified by the “voicebackupdir” global config option. Out of date recorded files can be deleted by setting the “keeprecorded” option accordingly (days to keep). Recorded files are compressed and encrypted by default. On new server installation, make sure that the voice directory is accessible via ftp for the MManage (for listening on the “CDR Record” form).
For skill based callcenters you can use the “Forward to Group” or the “Forward to SQL” actions. Voicemail can be implemented by the “record” actions.
fromport rtprecport callertype callstate cc_clientid cc_ccid Technical description: On routing if calleduser.ivrid > 0 than set ep ivr_id and calltype to eIvr No other ep will be created ivr_id is the campaignid On invite (after routed) if calltype = eIvr () call IvrAction(actIvrStart); IvrAction(EIvrActionType actiontype) 1. the actiontype will tell what was happened: actDefault, actIvrStart, actReady, actUserInput, actTimeout, actFail 2. load the scripts if already not loaded 3.
iscallback 1=act based on anumberhandling (default - connect to the ivr if not authenticated as enduser, otherwise callback immediately) 2=connect to the ivr if not authenticated as enduser, otherwise callback immediately 3=connect to the ivr if authenticated as enduser, otherwise callback immediately 4=drop the call if not authenticated as enduser, otherwise callback immediately 5=drop the call if authenticated as enduser, otherwise callback immediately 6= callback immediately always 7=alw
-CallingCardAuthentication: (Will check the pincode in the database). On success go to next item, on fail play a prompt to try again -Play File: Please enter the destination number -Forward to phone number [dtmftrimmed] (call forwarding) -Finish (release ivr) If you want to bill the user for calling the ivr, then set the “ivrbilling” global config to “0”. Otherwise set to “1” and only the forwarded call will be billed. If you need 2 leg billing, then set the “ivrbilling” value to 2.
The billed user will be the logged in user. The p2p command has 3 parameters. A number, B number and IVR id (optional). 2 CDR records are generated for these calls, both of them billed to the initiator (based on your usual pricing). The call to the A number will be initiated from “callbackcallernumber” configurable number or from the user who initiated the call.
SMS messages billing can be specified by selecting SMS as the service type on the Price setup form. The “smstime” is set to 60 sec by default. If no pricing found, then the sms messages will be billed by “smsprice” value. If you don’t have an SMS provider yet then please consider this list: clickatell.com infobip.com messagingbay.com hqsms.com tm4b.com smsbts.com bulksms.co.uk smsxchange.com textmagic.com truesenses.com txtlocal.co.
general username password number1, number2 pincode number number pincode b (pincode based authentication. the A number will be called back) general requests will result in a p2p call when at least 2 number is known. If only one number is know, than it will result in generating a callback to that number. Two numbers can be supplied, one by sms and the another number can be known from the sender CLI.
Pragma: no-cache Accept: Referer: 99.99.99.99 Content-Length: 157 Content-Type: application/x-www-form-urlencoded api_id=3144985&from=3611111111&to=3622222222×tamp=2009-07-04 21:39:40&text=Test+1+2+3&charset=ISO-8859-1&udh=&moMsgId=5jv45t5tb42bavu34drftr For callbacks you must define the campaign which will have the proper IVR content by the defcallbackivr global configuration setting. 4.12.4.
-initiate payments -endusers: recharge -endusers: setup call forwarding -endusers: initiate phone to phone calls and callbacks -enduser: send SMS -endusers: launch webphone -resellers: create sub reseller and enduser accounts -resellers: create own tariffs -resellers: generate pins and users -callshop: watch cabin’s status -admin: manipulate users -admin: global statistics To enable the SMS,phone to phone and the callback functionality the “checkdbconf” must be set to 1 or 2.
Then this top resellers can login on the web interface and create their own billing and add sub resellers and/or child endusers. Reseller can create a “base tariff” and other tariffs assigned to individual users. Multiple packets are allowed and packets can be assigned to other users or resellers directly from web (in the same way like on the “Users and Devices” form -> “Billing” tab -> Billing packet setting.
Credit from the reseller account is deducted when he add credit for the users (create new users with some default credit, increase user credit, generate recharge PIN’s).
1. The reseller can generate PIN codes regardless of the balance in the account. When a call is made, both the end user and the reseller are charged. When the reseller's credit ends, the end user will no longer be allowed to make calls. 2 The reseller can generate PIN codes only if he has enough credit in the account. If you opt for this method, you have to enable the early payments for the reseller. You can find this option on the reseller's billing tab.
4.12.9. Mobile dialers With the “all in one” platform the mizutech support will send your own mobile client build (hardcoded with your server address) For more details about this application please follow this link: http://www.mizu-voip.com/Software/MobileClient.aspx 4.12.10. SIM-Bank Skip this chapter if you are not using VoIP-GSM gateways. Introduction: Mizu servers can automatically switch to the best simcards based on routing settings and traffic estimations.
minstayonce: minimum time to stay (not fly) at once maxstayonce: maximum time to stay (not fly) at once minflydeep: minimum gateway count to fly trough at once maxflydeep: maximum gateway count to fly trough at once Simcard fields: origgatewayid: original gateway for the simcard (gatewayid can be the virtual gateway) origline: original line for the simcard (line can point to the virtual engine) origsimpos: original position for the simcard (simpos can be the virtual position) lastpresent: last seen (date-ti
flyto=ip/engine/simpos/simid //template flyto=local/local/2/999999 //local simpos flyto=local/2/2/99999999 //other engine flyto=1.1.1.
To enable h323-SIP conversion the runsipproxy global config values must be set to true. Local and LAN IP’s should be also enabled. These are can be enabled from the Configuration Wizard. The atarongk application can be controlled from the console port if you type the “gk” command Two important user entry is created automatically during setup used internally for sip to h323 conversion (and inverse): "sip2h323caller" traffic sender "sip2h323" sip proxy Make sue that these are present and enabled.
1. 2. 3. 4. 5. 6. 7. 8. 9. replace the atarongk file with the _dbg version change the "sipcommand" global config option to "vsip -vvvvvvvv -l siplog.dat" change the "gkcommand" global config option to "atarongk -ttttt -o gklog.txt" set loglevel to 5 check the callsigaddress field for h323 gateways (usually 1720) start the service make sure that atarongk and vsip processes are running make a test call and check the logs for h323 test calls you can use the openphone.exe or the ohphone.
-strong but slower encryption mode (when usequickencryption is false) Encryption can be enabled/disabled with the “useencryption” global configuration value: 0=disabled, 1=only when rec encrypted (default) 2=use 3=always The encryption method is stored in the “candisablesim” field in tb_users (refreshed automatically by the server on client connect) Also, there is a possibility to define a list of ip addresses with the “encryptedpeerlist”. All communications with these peers will be done encrypted.
Checkmaxlines = 0 ? Checkmaxlinetb=0 ? Maxsessionspeechlen= 1000UL * 60UL * 60UL * 6UL Ringtimeout=120 MAXEPCOUNTTRESHOLD=30000 MAXSUBSMSGCOUNT=99999 MAXWRONGMSGALLOWED=9999 Running the server as a VPN access point (forced encryption) Fwdregistrations=1 //only from encrypted clients (bug: set to 2) Alternatelocalport=5088 //or any other port; same like the port that was set in the softphone and webphone Alternatelocalportencrypt=3 //0=default,1=never,2=auto,3=always normalize_clean=1 normalizenumbers=0 all
Set the udptunnelsamesock congig option to at least 1. (udptunnelsamesock = 0; //0=no,1=if received so, 2=yes). This will tunnel only encrypted sessions so it is safe to use. Increase the “udpbuffsize” configuration option Tunneling between two servers Setup the sip servers and traffic senders with forced encryption and desired protocol Set the “usehttpclient“ global config option to “true”. Set tb_users.
5.4. How can I make test calls? 1. simply right click on a channel (“Simcards” form) and select the “Test call” option 2. or use one of the voip clients from the “Tools” menu 5.5. How to check the call quality on a specific channel? 1. In the “Set Directions” box set the preffered simid. Then go to the “Statistics” form and check the ASR/ACD values. 2. Start some tescalls (right click on the preferred channel and then hit the “Test Call” menu) 3. Listen to conversation. (“Voice Here” form) 5.6.
a) Restart the server. b) Call the administrator. 5.9. Calls in „routing” status 1. If all calls are in routing status, then restart the gateway. 2. If this behavior is specific only for some of the gateways, then check if you have enabled the voipgsmgw.exe and the vclientsrv.exe on the windows firewall. 3. If enabling this programs on the firewall and restarting the service (stop.bat, start.bat) will not help, then do a software upgrade and restart the PC. 4.
5.15. Too many wrong calls on a simpacket (low ASR/ACL) 1. Check disconnect reasons 2. Check if gateway is working ok (another type of simcards on that gateway are working) 3. Check if simcards are not blocked by service provider (make a test call and listen) 5.16. Not enough or too many calls on a sim or simpacket 1. Check absolutepriority, min/max daily/monthly minutes on sim and simpacket 2. Check the routing for that packet 5.17. Calls are routed to wrong simcards 1.
3. No calls Check gw,sim and packet priority Check the routing table 4. Wrong statistics (ASR/ACL) Check disconnect reasons Check if simcard is not blocked by service provider 5. Other problems Check statistics Check disconnect reasons Check gw, sim and packet priorities Check the routing table Check the log files Restart the gateway 5.23. Wrong disconnect reasons 1. Check firewalls 2. Check the log file for that directions 5.24. MManage cannot connect to the server 1. Ping the server box.
5.28. How to restart the server service MManage->Administration->Server Console->Connect and send the „servicerst” command 5.29. How to restart the server box -MManage->Administration->Server Console->Connect and send the „pcrst” command -If you cannot connect with MManage, you can find a small program in the vclients directory named „serverrst” (usually at C:/Program Files/VCLIENTS/ serverrst.exe -If these does’nt work, then the server has a serious problem.
1. Start GWTest and switch to the preffered channel/simpos 2. „login” with: AT+CPIN=xxxx (where xxxx is the original pin code) 3. Disable pin code request with: AT+CLCK=”SC”,0,”xxxx” (where xxxx is the original pin code) 4. in the next switch on, the sim will login to the gsm network automatically 5.34.
5.41. How to set up basic billing? On the “Price Setup” form add a new “Invoice and statistics” entry. Then you can add packets to it, which will define the traffic direction when the actual packet will be active and the price. 5.42. Where can I check the logs and traces? 1. “Logs” form 2. “Server Monitor” form 3. Set up your trace level in the “Configurations” form (filer after the “log” expression) 5.43. The conversation volume is too loud.
5.47. H323 signaling problems Check your firewalls. Check Gateway Configuration: onlyg7x, connectwithmedia, enableh245tuneling, faststart. 5.48. How to set up the automatic credit recharge? First you have to set up the “Message Rules”. The packet must be set to prepaid. Proper Credit Request/Charge command must be defined. See at 4.6.1. SIM Packets SIMcard “Credit and Recharge” setting must be set accordingly.
3. If the caller is blocked (e.g. DOS attack protection), then call will be silently dropped 4. Caller authorization (by source IP address, username/password, techprefix, etc) 5. Check the call parameters. If doesn’t fit into the predefined limits, the call will be dropped (example: too long called number) 6. Rewriting the called number if any Prefix Rule Match 7. Normalizing the called number (validating call prefix) 8. Searching for the best routing pattern 9.
simchange4= 6/00:00:00 - 7/24:00:00 - 8936302403070132426 (Saturday and Sunday) there is a priority order from top to bottom (simchange1, simchange2, etc.) numbering begins from 1 without holes tip: you can set date-hour prioritization tip: 24:60 is a wrong time (minutes ends with 59) tip: on day and exact date settings the roundrobin trick is not working 5.57.
For simcards you can setup the VAT value in the Packet options (“VAT” editbox). If you set the “convertsimcredittonovat” global configuration options to true, than sim credits will be automatically converted to net values. For examlpe after an automatic credit request, the credit value in the received messages (SMS) will be automatically conveted to net values. You should set up the appropiate VAT values for users too, wich will be taken in consideration during the billing process. 5.60.
5.63. Abbreviations ASR: average success ratio (percent of the connected calls) ACD: average call duration. The same as ACL (ACD: Automatic Call Distributor) ACL: average call length. The same as ACD SIMID: sim identifier. 13-17 digit number stored in the simcard (and written on the simcard) IMEI: gsm engine identifier (should be globally unique) ACT: average connect time. The time elapsed from setup until the connect in seconds PF: profit.
-local DID number: normal access numbers. Usually you will have separate DID numbers for different regions to minimize enduser costs -callback: DID or toll free number configured as enduser with iscallback set to the required IVR -ANI callback: same as callback with User-ID based authorisation (A number) -Virtual Numbers (DID): "real" phone numbers allocated for users. You have to buy DID numbers from CLEC or any other service provider like didx.net -SMS callback: callback triggered by received SMS message.
2. Open the printer “Options” Form This can be achieved -when you save invoices as PDF first time or -from Start Menu -> Programs -> PDF Creator -> PDF Creator -> Printer Menu -> Options 3.
16. allocate numbers in the mainserver for the new virtserver instance (callback numbers) 17. start the new virtual server 18. make a testcall 19. check the logfile for errors. test MAgent. check MManage 5.68. Fax detection Usually you can filter out fax calls by setting the “block711” global setting. 5.68. Handling dynamic & private ip/port For sip proxyes, endusers and gsm gateways the server will handle ip and port changes automatically.
MAXALLUSERS: to restrict the total number of users and devices (including endusers,gateways, etc) Devices behind this limit will be disabled (calls blocked) MAXSIPUSERS: to restrict the total number of sip endusers. Users behind this limit will be disabled (calls blocked) MAXGATEWAYS: to restrict the total number of sip, h323 and gsm gateways, gatekeepers and proxies.
5.71. Redirect or forward sessions to other domains Routing to other domain can be restricted by the “fwdtootherdomains” global config setting.
The defined commission percent will be calculated for the profit or for the enduserprice. This behavior can be set by the “salescomissionfromprofit” global config. Sales can give up for the users some reduction wich will be substracted from their cost. This can be configured with the “reduction” value for the individual endusers. 5.74. Multihomed setup UDP ports must be binded to different ip interface (at least the default sip port 5060), or set to different values.
SU1=unknown SS1=trying (scan success 1: the last status was: trying) -2 SS2=progress (scan success 2: the last status was: progress) -3 SS3=ringing (scan success 3: the last status was: ringing) -4 SS4=connect (scan success 4: the last status was: connect) -5 SF1=disc code (scan failed 1: the last status was: disc code received) -6 SF2=bye (scan failed 2: the last status was: bye received) -7 For example the following query will list all live numbers when the scan_stop_at is set to 1: select callednumber fr
nameX=MyPlugin cmdX=MyPlugin.exe X is a number starting from 0 to MAXPLUGINCOUNT The MManage will pass the database access parameters in the command line. Any other parameter must be readed from the inifile. Command Line: dbserverip,dbport dbname username password connectionstring *Please note that appserverip may differ from dbserverip! Direction parameters will be located under the “parameters” section. Read them as soon as possibile.
specificip, specificaddr //any ip or ip:port Examples: to restrict the target address for a sipuser you don’t allow local and lan ip: privateip to restrict the target address for a virtserver you don’t allow: localaddr to restrict a misconfigured proxy wich is always telling you 11.12.13.14: privateip, 11.12.13.14 on a virtserver you must deny only the localaddr for the mainserver (wich is configured usually as a sipproxy) and for the operators you can deny privateip 5.80.
prefixrewritestr: the original prefix prefixrewritefrom: keep from prefixrewriteto: inserted string for example to handle the hungarian roaming prefix: 08 + SK + BK + NSN +SN you have to set the following values: prefixrewritestr: 08X… prefixrewritefrom: 9 prefixrewriteto: 36 5.83. Short number and internal billing You can set the short number billing mode by the “shortnumbill_type” config value.
-don’t install any virus scanner (it is meaningless) -choose a strong password for your OS accounts (especially for the Administrator account) -choose a strong password for your database access (especially for the sa account) A good password means at least 8 characters and you should use both lower and upper case letters and numbers.
-passwords are stored in clear text in the user table by default to ease the support (md5 checksum option is also available). Make sure that your database server cannot be stolen, hacked, etc.
Full Cone NAT A full cone NAT is a solution, where all requests from the same internal IP address and port are mapped to the same external IP address and port. Furthermore, any external host can send a packet to the internal host, by sending a packet to the mapped external address. Restricted Cone: A restricted cone NAT is a solution, where all requests from the same internal IP address and port are mapped to the same external IP address and port.
5.91. How to change the database username/password The auto installer will use the “sa” username with “srEgtknj34f” password by default. Install the SQL Server Management Studio if not already installed. Login with the old credentials. Open “Security” -> “Logins”. From here you can easily add/remove any users or change passwords. Before changing the default user settings, stop the mserver service (By launching the stop.bat file or from Services). Change the database settings in the mizuserver.
Increase REGISTERIVAL,upperexpire, userofflinemin, upperexpiremin Set allowforkforsignaling to 0 Set usertpmutex to false Set allowlogloss to 2 Increase cfg_sipmsgresendival Turn off creditcheckforpostp Turn off allowforkforsignaling Turn off creditcheckforts Turn off lookupcdrcalldirection Turn off brs if not needed. (“lcr_brs”) Turn off the dynamic firewall.
SetCfgStr("settings","priorityboost","false"); SetCfgStr("settings","wrongnumcache","100"); SetCfgStr("SIPSettings","MAXSUBSMSGCOUNT","35000"); SetCfgStr("SIPSettings","MAXWRONGMSGALLOWED","4000"); SetCfgStr("SIPSettings","MAXWRONGAUTHFROMIP","3000"); SetCfgStr("SIPSettings","MAXFAILEDAUTHENABLEDIPPORT","170"); SetCfgStr("supervisor","maxnocdrmin","6000"); SetCfgStr("supervisor","maxnologival","20"); SetCfgStr("SIPSettings","MAXH323GKCDRCACHE","400"); SetCfgStr("CallCenter","maxcallatonce","1000"); SetCfgSt
5.94. How to remove the 3 digit techprefix system wide setting This might be required for old server versions (before 2009) to convert to the latest database format: Run the following queries and reload the server config.
For unified pricing you can use the “smsprice” and the “smstime”. (in this case all sms messages will be billed with the same price, regardless of the destination) Otherwise the pricing can be set on the “Price Setup” form the same way as for voice calls. The only difference is that all SMS messages appear with 60 sec duration. Unicode messages are supported.
(select id from tb_groups where name = 'testgroup'), 1, id from tb_users where username like '%111%' --select users from this group select tb_users.id, tb_users.Username, tb_users.name from tb_groupentries, tb_users where tb_groupentries.groupid = (select id from tb_groups where name = 'testgroup') and tb_users.id = tb_groupentries.entryid --delete group delete from tb_groupentries where groupid = (select id from tb_groups where name = 'testgroup') delete from tb_groups where name = 'testgroup' 5.98.
If the techprefix is „-2”, then the original techprefix will be inserted in cdr record (but not forwarded). If the techprefix is empty, then only the normalized callednumber will be forwarded. The following techprefixes are reserved for the server: 111,222,999. Only 3 digit techprefix is allowed. If your traffic sender needs another techprefix length, you must rewrite the incoming number in the “Prefix Rules” form.
For more control over the A number you can use the prefix rules form or manually edit the v_check_pxrules stored procedure. Prefixed can be also used for authentication, routing and billing. 5.100. How to rewrite caller/called numbers The server is configured by default to do some basic number normalization like removing junk characters from the numbers, removing + or 00 and a few others.
-this relationship can be analyzed using the “Ownerships” form -make sure that you have a public reseller price listing (enduser costs) -reseller will be able to create their own prices on the website -reseller can create a “base tariff” and other tariffs assigned to individual users -individual reseller prices are stored in tb_billsources with their “resellerid” -if reseller has not tariffs, than the billing will be done after usual enduserprices -top reseller id is stored in tb_cdrs.
- set the " allowanonymouscaller" global config option to "true" - set the " allowanonymoususers " global config option to "1" or "2" -an enduser named "sysanonymous" must be present -set the " enforcestrongauth" global config option must be set to "false" -set the "autocreatereguser" config option to 2 (for voip tunnels only) 5.100. How to play advertisement for the callers For this you just have to fill the "playadv" field for the user who have to hear the advertisement (or for all users).
Window Server - Binding Web Interface to port 80 when IIS is running By default IIS listens on all IP addresses on port 80, assigned to a PC. So what needs to be done is to configure IIS to listen only on one IP address.
1 channel bandwidth: 40 kbits = 13 GB / month 10 channel bandwidth: 400 kbits = 130 GB / month 100 channel: 4 mbits = 1300 GB / month (1.3 TB) 1000 channel: 40 mbits = 13000 GB / month (13 TB) 10000 channel: 400 mbits = 130000 GB / month (130 TB) If both the traffic sender and the termination devices are located on remote peers, then you have to multiply these values with 2X (especially if you are billed for both “in” and “out” traffic) Nothe: 1.
The mizuupdate.
Make sure that your paypal account is “verified” before switching to production. For this you might have to send some documents to PayPal like your Driver ID. If the account is not verified, then PayPal might hold all transactions after a certain limit. The amount applied for the user credit will be the real (net) amount received which is usually lower than the amount sent by the user due to paypal transaction fee and other additional costs (for example currency conversion).
Currency: paypal currency Transactionid: unique id for all transactions Stage: 0=unknown,1=started,2=notification received,4=verified and applied,5=failed, 6=need to be checked (no email match) Comment: contains the reason of the failures and all patameters as received from paypal Example query to list all transactions that need to be approved from the last week: select top 1000 tb_users.Username, tb_users.credit as 'usercredit', tb_payprocessing.
Set the “haslinkedaccounts” global config option to “true” for this to work. 5.107. Server clustering You can setup multiple servers with one (clustered) database backend. For this you will have to edit the .ini file near the server and set the node value. Each server must have its unique node id (1,2,3…) [database] node=1 For any global configuration option specific to a node, a #NODENUMBER have to be appended for the key. For example bindip#1.
avg(case when b.realduration > 0 then b.realduration else null end) as 'ACD (sec)' FROM tb_cdrs b WITH(NOLOCK) WHERE b.datum >= :from AND b.datum <= :to GROUP BY b.marker ORDER BY b.marker --stats by call quality SELECT b.marker, count(b.id) as 'CDRC (count)', sum(b.realduration)/60 as 'SL (min)', sum(SIGN(b.realduration))*100/count(b.id) as 'ASR (%)', avg(case when b.realduration > 0 then b.
else null end ) as 'QUALITY (0 = worst,10=best)' FROM tb_cdrs b WITH(NOLOCK) WHERE b.datum >= GETDATE()-1 and b.realduration > 0 GROUP BY b.marker ORDER BY b.
5.109. How to obtain geolocation and advanced call quality statistics 1. Import GeoInfo database with MManage -> Tools menu -> Server setup -> Setup GeoIP database (GeoIP.exe) At least the countries must be imported 2. If you need geoinfo data from previous days, then run the “createcdrinfo” command from the server console with a “number of days” parameter. This will take 5-10 minutes (don’t run on high server load) 3.
i.srvlosspercent < 13 and i.clientlosspercent < 13 and i.totallosspercent < 13 and (i.srvlost < i.srvsent/6) ) then 8 when realduration > 65 and i.srvsent > 2000 and i.srvrec > 2000 and ( (i.srvsent > i.srvrec/4) and (i.srvrec > i.srvsent/4) and i.srvlosspercent < 5 and i.clientlosspercent < 5 and i.totallosspercent < 5 and (i.srvlost < i.srvsent/6) ) then 10 when realduration > 180 and i.srvlosspercent < 15 and i.clientlosspercent < 15 and i.
( (i.srvsent >= 0 and i.srvsent < i.srvrec/10) or (i.srvrec >= 0 and i.srvrec < i.srvsent/10) or i.srvlosspercent > 50 or i.clientlosspercent > 50 or i.totallosspercent > 50 or (i.srvlost > i.srvsent/2) ) then 'NOVOICE' when realduration > 5 and realduration < 55 and (i.srvsent > 400 or i.srvrec > 400) and ( (i.srvsent >= 0 and i.srvsent < i.srvrec/4) or (i.srvrec >= 0 and i.srvrec < i.srvsent/4) or i.srvlosspercent > 20 or i.clientlosspercent > 20 or i.totallosspercent > 20 or (i.srvlost > i.
CAST(mstr.physical_name AS VARCHAR(72)) AS 'File', stats.num_of_reads, stats.num_of_bytes_read/1000000 as 'MB read', stats.num_of_writes, stats.num_of_bytes_written/1000000 as 'MB write', stats.size_on_disk_bytes/1000000 as 'MB size' FROM sys.dm_io_virtual_file_stats(null,null) AS stats join sys.master_files AS mstr ON mstr.database_id = stats.database_id and mstr.FILE_ID = stats.FILE_ID WHERE DB_NAME(mstr.database_id) = 'tempdb' or DB_NAME(mstr.database_id) = 'mserver'; go 5.111.
In the mizuserver.ini leave the [database] settings as the main mizuserver.
from trace12 group by applicationname order by sum(cpu) desc select sum(duration), applicationname from trace12 group by applicationname order by sum(duration) desc select sum(cpu), applicationname, substring(textdata2,1,29) from trace12 group by applicationname, substring(textdata2,1,29) order by sum(cpu) desc select sum(duration), applicationname, substring(textdata2,1,29) from trace12 group by applicationname, substring(textdata2,1,29) order by sum(duration) desc select sum(cpu), substring(textdata2,1,
5.115. Pricing speedup If you have (or will plan to add) many prefix entries for multiple billing packets than you should consider to change to the int based alghoritm especiall if you are using LCR. Follow these steps: 1. Create tb_billingtimes_int with the prefix field set to int type. Don’t forget to add the entries. 2. Migrate all entries to the new table: INSERT INTO [msrv_2].[dbo].
-00, 0 and + prefixes are not allowed in the billing (this should be avoided anyway) -instead of * you should use -1 In this mode the _int version of the stored procedures are used (v_getprice_int and fgetprice_int) 5.116. Shared DID’s You can provide shared DID numbers for your users in the following way: -Add a few DID number from the “Users and devices” form.
peaktimebegin or peaktimebegintr: peaktime start hour. Default is 9. peaktimeend or peaktimeendtr: peaktime end hour. Default is 19. weekendispeak: treat weekend as peaktime (same traffic ammount). Default is false. restartatnight: if to restart at every night. Default is false. restartpcatfirst: don't restrart the service. Restart the pc immediately.
,'RA MANAGER') order by batch_duration desc declare @spid int , @stmt_start int , @stmt_end int , @sql_handle binary(20) set @spid = 124 -- Fill this in select top 1 @sql_handle = sql_handle , @stmt_start = case stmt_start when 0 then 0 else stmt_start / 2 end , @stmt_end = case stmt_end when -1 then -1 else stmt_end / 2 end from master.dbo.
OPEN DBCursor; DECLARE @DBName VARCHAR(200) = ''; FETCH NEXT FROM DBCursor INTO @DBName; WHILE @@FETCH_STATUS = 0 BEGIN DECLARE @SQL NVARCHAR(MAX) = N'USE ' + QUOTENAME(@DBName) + ' IF EXISTS ( SELECT 1 FROM sys.tables WHERE [Object_ID] = OBJECT_ID(N''dbo.tb_settings'') inisection=''settings'' ) BEGIN INSERT #tmp_333 (DatabaseName, valstr, keystr, inisection) SELECT @DB, [valstr], [keystr], [inisection] FROM dbo.
http://blogs.msdn.com/b/ntdebugging/archive/2007/07/05/desktop-heap-part-2.aspx 5.121. How to rewrite SMS number prefix Create a stored procedure named v_dialplansms with the following inputs: - tonum varchar(256) - fromid int - fromnum varchar(64) - clientip varchar(64) This sp must return one field containing the rewritten target number (“tonum”) Set the “usesmsprefixsp” global config option to “true” and reload. 5.122.
5.124. Enable session-timer With session timers (RFC 4028) you can prevent orphaned sessions (hung calls). Session timers are not enforced by default to prevent any complications or incompatibilities on a new server. If all the media are routed trough your server, then there is no much reason to enable session timers because the media timeout can usually prevent orphaned calls.
12: can’t add resellers 30: normal rights (default) 40: increased rights 60: full rights 5.126. Server ini file The database connection settings are stored in a configuration file which should be named “mizuserver.ini” and located near the service executable (mserver.exe). This file is readed when the server is starting so it can find its database (all other data and settings are stored in database).
You just have to upload the prices. To charge only connected calls, you have to apply the billing packet for B-leg only. You can set this option on the "Price Setup" form in the "Applied for" section. 6. Links VoIP Server homepage: http://www.mizu-voip.com/Software/VoIPServer.aspx Mizutech homepage: http://www.mizu-voip.com/ Documentations: http://www.mizu-voip.com/Support/Documentations.aspx Email support: serversupport@mizu-voip.
