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Disclaimer: Telcom SA. reserves the right to change any information found in this document without any written notice to the user.

License Agreement

You must accept the license agreement (LicenseAgreement.doc) before you use any Tresto hardware or software component!

Trademark Acknowledgement

LINUX is a registered trademark of Linus Torvalds in the United States and other countries.

Windows and Microsoft SQL Server is a registered trademark of Microsoft Corporation in the United States and other countries.

Oracle is a registered trademark of Oracle Corporation.

OpenH323 (used in test tools) are licensed under MPL: http://www.mozilla.org/MPL/MPL-1.0.html. Source code is included on the install CD.

Other logos and product and service names contained in this document are the property of their respective owners.

1. Introduction

1.1. Short description

This document describes the administration of Tresto Gateways, SoftSwitches and SimBanks. A unique set of proprietary software and hardware based capabilities and processes in VoIP network planning and network management.

These components are designed to cover the telecommunication needs for small to very large companies. The main power of the system is the sophisticated GSM and VOIP components, which are strongly used in today’s telecommunication infrastructures.

The Tresto components can be used as standalone or as centralized intelligent VOIP/ISDN/GSM platform, capable to handle millions of minutes/months.

1.2. Features

1.2.1. Hardware Components

VoIP-GSM gateway

-8 channel gateway, best fit to any cheap DSL connection

-up to 64 simcard/gateway

-SIM server interworking capability

-Integrated antenna splitter

SIM Server

-up to 750 simcard

VOIP-GSM Server

-industrial PC

-fault tolerant

-server failovering capability

-distributed architecture

Built in watchdogs to monitor the operation of the system components

1.2.2. H323

H.323 Standard Features (v.1,2,3,4)

Full H.323 proxy

H.225.0 Call Signaling

Fast Connect/Fast Start

H.245

H245 tunneling

H245 in setup

DTMF send/receive

Watchdog

Direct endpoint call signaling.

Gatekeeper routed: call signaling (H.225.0).

Gatekeeper routed: call signaling (H.225.0) and control channel (H.245)

Gatekeeper routed: call signaling (H.225.0), control channel (H.245) and voice

RTP Port Range (For firewalls)

Child Gatekeeper capability

Backup Gatekeeper capability

Gatekeeper clustering support (neighbors, parent/child, alternates)

1.2.3. GSM

Dual Band (900 / 1800 MHz or 850 / 1900 MHz)

Half rate, full rate, enhanced full rate, SMS, USSD

SIM server support

Integrated antenna splitter

8 channels/box

Up to 8 SIM cards per engine

Multiple ways to handle incoming calls

Call Forwarding

Sending and receiving SMS messages

Email To SMS Feature

Inter gateway SIM routing

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

SIM server interworking

Virtual Channels

1.2.4. SIP

Fully compliant with SIP rfc's

SIP proxy

SIP register

Routed and Direct voice

Automatic NAT detection

Voice Recording and Playback

Class 5 features (see details below)

RFC 2543 compatibility

RFC 3261 compatibility

RFC 2976 The SIP INFO Method

RFC 3262 Reliability of Provisional Responses in Session Initiation

RFC 2617 HTTP Authentication

RFC 3263 Locating SIP Servers

RFC 3265 Specific Event Notification

RFC 3420 Internet Media Type message/sipfrag

RFC 3515 Refer Method

RFC 3311 UPDATE Method

RFC 3581 Symmetric Response Routing

RFC 3842 Message Summary and Message Waiting Indication Event Package

RFC 3891 "Replaces" Header

RFC 3325 Private Extensions to the Session Initiation

RFC 2778 A Model for Presence and Instant Messaging

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.263 Video Streams -only routing

RFC 2327 SDP: Session Description Protocol

RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals

RFC 3264 An Offer/Answer Model with Session Description Protocol

RFC 3550 RTP: A Transport Protocol for Real-Time Applications -replaces RFC 1889

RFC 3555 MIME Type Registration of RTP Payload Formats

draft-ietf-mmusic-ice-02 A Methodology for NAT Traversal for Multimedia Session Establishment Protocols

draft-ietf-avt-rtp-ilbc-04

draft-ietf-sipping-cc-transfer Call Control - Transfer

draft-ietf-sip-referredby-05

Custom protocol extensions are possible

1.2.5. SIP-H.323 protocol conversion

Signaling and media when needed

1.2.6. Codecs

G.723.1

G.729

G.711 A-law

G.711 u-law

GSM 06.10

G.726 (16,24,32,40 KHz)

T.38

DTMF

Voice:

Adaptive de-jitter buffer

Voice Activity Detection/Silence Suppression

Recording conversations

QoS

Packet saver technology

1.2.7. IP

Ethernet 10/100 Base-T

Static IP

PPPoE (DSL or cable modem)

DialUpISDN

VPN

1.2.8. VOIP-GSM Server

First centralized architecture for GSM termination

Multiple signaling protocol support

Load distribution between the operational channels

No hard limit on the number of simultaneous calls

High availability

High throughput (more than 50 million minutes/month)

No additional Tresto hardware required

Equipment management

Channel management

Simcard management

Automatic recharge

Access Control Lists

Routing (see below)

Billing (see below)

Exploits almost any SIM tariff model

Number translation

Protocol encryption

Media proxy

Automatic time synchronizations

H.323/SIP Gateway Topology Hiding

Embedded firewall

Enhanced Security (automatic detection of flood attacks)

Web GUI for end-users

Encrypted communications

Distributed absolute fault tolerant system

External system supervisor service (email and sms alerts, watchdog can restart failed subsystems)

Class 5 Features:

Call Forward All/Busy/No Answer

Caller ID

RingGrouops

Call Return

Call Waiting, Call Hold

Caller ID Block

Selective Caller ID Blocking/Unblocking

Speed Dial

Three-Way Calling, Conference calls

Call Transfer (conditional/unconditional)

Message Waiting Indicator

Hotline

Call transfer, Attended transfer, Unattended transfer

Voicemail

DTMF transcoding on server side

Interactive Voice Response (IVR) supporting applications such as credit card and prepaid services

Video

T.38 fax relay

1.2.9. Call Center

Automatic Call Distribution: like simple automatic dialing, power dialing, predictive dialing, predictive intelligent dialing

Call Recoding: All calls can be recorded and stored

Real time call check out: Supervisors can listen to the ongoing calls real time

PBX Features: Call hold, call wait, call transfer, call forward (conditional and unconditional), call conference, CLIP, CLIR

Customizable Scripts: script tree, with any number of branches, answers, and reason codes.

Customizable IVR: Any number of language, any number of branches, voice and faxmail, call transfer to the operators

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

1.2.10. Routing

ACL

Sophisticated configurations

Load Balancing on available GSM channels and any other devices

Rerouting

Number rewriting (calling and called)

Failovering (multiple levels)

Least Cost Routing

Call routing based on PLMN tariff packages

Blacklist/White list filtering

RADIUS

Support for NAT traversal

Automatic capacity rebalancing

Automatic channel management

Number portability support

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

-Desired

-desired minute on packet

-packet multiplier

-Rotate

-“minrotateival”, “desired”, “maxrotateival”

-Price

-min/max pricediff on obj, maxpricepermin for system/partner

-Timetable

-BRS

-LCR

-and many other options

1.2.11. Billing

Flexible pricing

Automatic and Real Time billing (CDR records already includes the prices)

Prepaid and Postpaid platforms

Directions (traffic sender,prefix,gateway,sim packet) and time based billing. Lots of configuration settings.

Reporting and price comparisons (LCR)

Invoice generation in different formats, PDF generation, email scheduler and invoice printing

Complete call rating & accounting services for complex rating schemes

RADIUS

Currency and VAT can be set for every packet. Time zone can be changed.

1.2.12. Management

Centralized configuration and management for all software and hardware components

TManage:

-easy to use, mdi style

-almost every data query is parameterized with traffic direction and time

-all data in one place

-lots of data can be obtained from sl,asr,acl forms

-global system analysis

Create and edit network elements

Remote maintenance of Tresto gateways

Display of system information

Service restart functions

Display of the current status of each gateway and channel

Real time call supervision (with many grouping options)

Real time channel supervision (with many grouping options)

Statistics (Text based and graphical ASR,ACD,SL, etc) on any traffic direction and time scale

Disconnect Reasons (with many grouping options)

CDR monitoring, retrieval, direct CDR access

Global system analysis!

Routing pattern selection

Routing time selection

Failovering (in case of channel, gateway, direction etc errors)

Best Route Selection

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

Capacity and system load reports

And many more features!

1.2.13. Limitations

-The ammount of the traffic that can be handled depends on the routing speed mainly. If you have the database on a separate server, make sure that the network connection is fast.

-The media routing will consume havy CPU resources too. You can speed up the media routing if you use more than one processor.

-The maximum database size for basic gateways and servers is 4GB. If you need to work with more than 5 mil calls for more than 3 month, you should upgrade your license to the advanced version.

1.2.14. Known issues

Some features will work only with SIP protocoll

H323 GK doesn’t support username/password authentication

RADIUS is compatibile only with some servers

Conference, VoiceMail, Number Portability and SIM Bank business logic will come soon

1.3. Contact and tech support

Full remote administration supported.

24/7 technical support.

Visit http://www.gsmtermination.com for more details.

2. Modules

Depending on licensing, some modules may not be available in your release!

2.1. Soft switch

The Tresto Soft switch (Server) is the “brain” of the system. Depending on your needs, you can connect as many gateways as you want. Small companies can use “all in one” solutions, where the gateway and the server are placed in the same box. Large organizations will divide the server in multiple units, adding more power and fault tolerance. Up to 6 gateways the server can be used with the built-in database engine. With more gateways or users it is strongly recommended to use one or more separate database servers (MS-SQL or ORACLE). The soft switch is built from several modules: sip stack, h323 stack, sip – h323 conversion module, media server, ACL, routing, billing, alerting.

2.1.1. SIP Stack

The Tresto SIP stack was written in C++. It’s very fast and robust, currently used by voip service providers handling thousands of users.

2.1.2. H323 Stack

Capable to work as a simple Gateway or as a fully featured Gatekeeper.

2.1.3. SIP-H323 converter

Thank to this module, the protocol conversion is very transparent. You don’t even need to know if your partners use SIP or H323.

2.1.4. Media Server

If your server needs to route the media channels for many concurrent calls, you may need to use a separate media server, thus offloading the server traffic, and maximizing media throughput.

2.1.5. Routing

With the Tresto softswitch you can build very sophisticated routing scenarios. The routing is usually based on traffic direction and time. LCR and BRS routing are available.

2.1.6. Billing

The server will generate the detailed CDR records after each call. Thus the billing can work nearly real-time. (very important for prepaid systems). You can generate various reports and invoices based on a set of predefined rules.

2.1.7. Alerting and Daily report

The server can send various reports and alerts based on predefined rules. The reports are sent by email or SMS.

2.1.8. Call Center

Manage operators, automatic call distribution, IVR and other callcenter specific tasks.

2.2. VOIP-GSM Gateway

Tresto VOIP-GSM Gateways support 8 concurrent calls and up to 64 simcards.

See the features section for more details.

2.2.1. GSM

All standard GSM capabilities are supported.

2.2.2. VOIP

Tresto VOIP-GSM Gateways can accept SIP and H323 registrations, can act as a SIP proxy or a H323 Gatekeeper or Gateway. These functions can be run simultaneously.

2.2.3. SIM Bank

The built-in simbank will allow to virtually route the simcards in other Tresto gateways.

Tresto VOIP-GSM Gateways can take advantage of an external simbank, so you can have all your simcards in one place, easing the maintenance and administration tasks.

2.3. Other components

2.3.1. VPC

Simple monitoring software fot business purposes. Each partners (gateway or simcard owners, traffic senders, etc) can have their own VPC to monitorize their own traffic and create reports.

VPC Setup

You can give the VPC for any of your partners. The partners can login to the VPC with they username and password configured in the “Users and Devices” form in TManage. 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 TManage has the capability to login as a superuser. To do so, you have to enter your partner username, but use the admin password (from the “ad” account). Then you have access to the “Add Query” button in the VPC. Here you can add,delete or modify the existing queries and their access rights.

In the “rights_allow” field you can put a list of user id, “all” or “nobody” fields. The same for the “rights_deny”. Thus you can configure which partner can see and execute which queries.

2.3.2. Helper modules

server service: the brain of the system

H323 GK: standard H323 gatekeeper

SIP Server: sip stack

Media Server: rtp routing

VGW: voip-gsm gateway, the most essential part of the client

client service: this service supervises the gsm gateway and gives a clear interface to the server

TManage: smart client software, capable to manage the whole system

supervisor service: this service supervises the vserver

alerter service: collects statistical information and reports it

recplayer: can play g729, g723, encrypted, raw PCM and wave files

loganalizer: log file parser

gwtest: handle gsm terminal (no h323)

ipmux: packet saver client and server

serveremulator: server interface to gk

simalloctest: test the automatic sim allocation

smtp_test: test smtp functionality

tariffcalc: estimate sim packet real price

tcperver: tcp server for test

udptest: udp through test

valerterclient: alerter sw which can be installed on client computers

vchargecards: manage chargecards

vclientinterface: platform specific functions for the gw

partnerclient: admin sw. for our partners

pricesettings: for packet price configuration

routingandprices: for config. routes, prices and sim packet priorities

servertest: brute force test for the server

supervisor: supervises the server

updater: automatically updates client software from the server ftp

mediasrv: media server for routing rtp packets

businesslgc: controls the routing, registration, endpoint list, endpoint creation, udp initialization

3. Maintenance Tasks

When properly set up, Tresto software doesn’t need too many administration tasks. The routing will adjust automatically to the external conditions. Every software module has auto repair features. However if you have millions of minutes/month, you may watch the system parameters every day.

3.1. Overview

3.2. Quick Setup

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 TManage programs in a separate PC used for monitoring your Tresto devices. network (you can find it on the Tresto 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. Put your simcards into the gateway (see the image below)

5. Connect to the gateway or server with the TManage (by typing its ip address and username/password in the login form)

The default username/password is admin/tpwdadmin

6. Set up the basic parameters from the “Configurations” form

Be careful.

7. Set up one ore more packets for the simcards in the “SIM Packets”

Be careful with the following settings: prepaid/postpaid, allowedpartners

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. Set up advanced routing –Optional

Firewall, prefix rules, BRS, etc

12. Set up the billing module –Optional

13. You are ready to accept traffic now.

3.3. Daily Maintenance

You should check at least the followings every day:

-Current Calls –to quickly check if you have the required amount of traffic

-GSM Channels (channels with problems are marked with red)

-Quality Statistics by traffic senders and terminating gateways

-Run a global system analysis (“Analyze” form)

3.4. Monthly Maintenance

-Check your cash flow (“billing” form) to check if your routing is still profitable

-Logs (errors and critical levels)

-Analyze your traffic by using the “Advanced Statistics” form

-Remove blacklisted but good numbers

4. Administration

4.1. TManage

4.1.1. Overview

Although the server and the gateway are PC based, you will newer have to login to the PC. All administration tasks are done from TManage.

4.1.2. TManage Installation

The TManage program group is shipped with all Tresto Hardware components. Occasionally you may visit our website to download the newer versions. The software is shipped as a standard windows install package. Requirements are:

-Windows 2000/XP/2003

-At least 1024x768 screen resolution for better operation

-You may need a headset for tescalls

-Network connection

Double click on the install exe and follow the instructions.

During the install procedure the following modules and files will be copied:

-TManage.exe –tha main executable

-VPC.exe –admin version

-VSQLRouter service –for compressing and encrypting sql requests and answers

-VOIP client programs (SIP and H323)

-Adobe Acrobat Reader –optional (can be canceled during the install process)

-Cute PDF Writer – pdf printer driver used for reports and invoice pdf creation -optional (can be canceled during the install process)

-Utilty programs: tariffcalc.exe, smtptest.exe, rptest.exe etc

-Required dll files

-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.

The following values are required on login:

App Server: server ip address

DB Server: databse ip address (“default” can be used if the same as “App Server” address )

DB: Application and database instance (because a single server can hold several virtual server)

Username: login name

Password: login password

Use encryption: encrypt and compress the server comunication (requires the “vsqlrouter” service to run)

4.1.3. TManage Framework

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 TManage.Here you can define your preference regarding the traffic direction including Source and Destination. You can filter on Item Type, Item, Group, Number Prefix, Packet and SIM Card. For example you may select one SIM ID, and when loading logs, you will see only the messages related to the selected simcard.

In the left-bottom side of the form, you can find an edit box used for quick search. You can use the ‘*’ character in the begin and the end of expressions. (For example when searching for CDR records).

Most of the report will be filtered after the selected Date Interval also.

In the Thresholds you can set some thresholds used for TManage. This setting doesn’t have any effects on the server or gateway. Server and gateway thresholds may be set up from the Configurations Form explained below. In the Options Windows (still from the Tools Menu), you can set up several important TManage parameters.

In the Help Menu you have access to documentation.

In the Licensing box you can see your server parameters (there is no effect if you change these values, because these ar used only for informing you). Depending on licensing, some modules may not be available in your release! Occasionally you may need to know the software version you use, which you can find in the About box.

Example: How the check your ASR for the traffic sender “A” in the last week.

1. In the date-time drop-down list, select the “Last Week” field

2. In the “Select Direction” form set the “Source” (left side) “Type” to traffic sender, and select “A” in the “Name” drop-down list (or type “A” manually)

3. Launch the “Basic Statisitcs” form under Monitoring.

4. Clear the “Group by” option (select the first “-“ line)

5. Make sure the ASR checkbox is checked

6. Click on (Re)Load

7. Depending on current server config and current load this query may take some time (on a usual configuration this will take 2 second)

4.2. Monitoring –TManage

4.2.1. Current Calls

Currently running calls are listed here. Calls terminated on Tresto 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. Can have the following values: Gateway Disabled, Off (no info), Not Active, Gateway Disconnected, Closed, Not Ready, Ready, Dialing, Ringing, Speaking, Call Ending, DTMF, Simulating Outgoing, Simulated Incoming, Routing to SIMID, Routing to Alias, Routing

Duration: seconds elapsed from Setup (not from Connect!)

Caller: source name (user name or traffic sender name)

Called: destination name (user name or traffic sender name)

CallerNumber: the phone number of the caller party

CalledNumber: the phone number of the called party

Dialed: number routed to called user or gateway (with techprefix)

Line: the number of the gsm channel (usually from 0 to 7)

SimPos: the position of the active simslot in the current engine (usually from 0 to 7)

SIM Owner: the owner of the SIM Card

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

SIM ID: sim identification number

4.2.2. GSM Channels

Usually this is the most frequently used form by the technical support. You can see the status of each gsm channel on your gateway(s).

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. (ok status and with enough credit)

Credit problem: will list the channels with low credit and when the credit request/recharge functionality doesn’t work properly

Wrong lines: list all “bad” channels

Last week detected: all active simcards in the last week

All: all channels including disabled ones

Sim distribution: all existing simcards

Not used postpaid: Some simcards may not receive calls for many days due to some misconfigurations. You may check this list occasionally to be sure that all of your postpaid simcards are working.

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. The following values are defined:

-unknown: you may have to reload

-missing: no simcard detected

-sim disabled: the “enabled” property of the simcard is set to false. No calls are routed to that simcard.

-gw disabled: the “enabled” property of the gateway is set to false. No calls are routed to that gateway.

-gw missing: no status from this gateway for more than 8 minutes

-sim missing: no status from this simcard for more than 8 minutes

-sim temp. disabled: the simcard “temporarily disabled” property is set to true. You must reenable the simcard to receive calls.

-gw temp. disabled: the gateway “temporarily disabled” property is set to true. You must reenable the gateway to receive calls.

-packet disabled: : the “enabled” property of the simpacket is set to false. No calls are routed to the members of that packet.

-closed: the channel is in the “closed” status. Can be for simchange or maybe is in restart.

-failovered: call quality has dropped below the predefined values, so the sim priority is lowered

-cannot get credit: credit automation malfunction. There are simcards from which the operator may restrict the credit request if they have no credit. Also you may need to check the packet settings related to the credit request. Check the logs too.

-wrong statistics: wrong ASR or ACD in that channel in the current day

-wrong ASR: the ASR is low in that channel in the current day

-wrong ACL: the ACD is low in that channel in the current day

-expired: the simcard has reached the predefined limits (you can configure this limits in the SIM Packets form)

-low credit: not enough credit on this simcard. Check if you have enough chargecards and the credit automation is working correctly.

-autodisabled: same as failovered

-ready (in black): no calls have been routed in the last 10 minutes on that channel (but the simcard is working without problems)

Status: channel status as reported by the gateway. Can have the following values: Gateway Disabled, Off (no info), Not Active, Gateway Disconnected, Closed, Not Ready, Ready, Dialing, Ringing, Speaking, Call Ending, DTMF, Simulating Outgoing, Simulated Incoming, Routing to SIMID, Routing to Alias, Routing

Line: the number of the gsm channel (usually from 0 to 7)

SimPos: the position of the active simslot in the current engine (usually from 0 to 7)

SIM Owner: the owner of the SIM Card

PartnerID: The database ID of the owner user

~~CanWatchPartnerID~~~~: database id of the partner who can see this simcard in there VPC~~

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 day of the current month in weekends

Username: Gateway Alias

Credit: current credit on the simcards. Refreshed after all calls, and corrected after credit requests (VAT included!)

InitialCredit: you may save the initial credit of the simcard here

Tpercek: special field for TMobile Tminutes

AllowedPartners: comma separated list of allowed partners and traffic senders. ‘*’ will allow all. You may restrict the access on gateway or simpacket level instead of setting it for all simcards separately. Try to use the packet “allowedpartners” setting and leave it as ‘*’ for the simcards!

Prepaid: loaded from the packet settings (1 if prepaid, 0 if postpaid)

Datum: the date when the simcard was inserted in the database (first use)

Comment: you may place any comment here

LastError: last error message received from the gateway related to the actual simcard

LastLog: last log message received from the gateway related to the actual simcard

LastFailedCalls: the number of subsequent failed calls (not connected calls)

LastWrongCalls: the number of subsequent wrong calls (below the predefined speech length)

LastGoodCalls: : the number of subsequent good calls (above the predefined speech length)

FieldStrength: combination of last reported field strength value in percent (0-100%) and the rx quality (from 0 to 7. 9 is invalid).

Value = field strength*10+rxqual (divide with 10 to get the fieldstrength. The remaining is the rxqual)

Pin: the security code of the simcard

LastRecTime: : the date-time of the last message received from the simcards. Every channel will send status messages in every 2 minutes and on status changes

LastCallerid: the destination id of the last call attempt

LastDialedNum: the called party number of the last call on the simcard

LastCallBegin: the date-time of the last call attempt on the simcard

LastCallEnd: the date-time of the last call attempt on the simcard

Enabled: set to 0 to disable the simcards instead of deleting it

TemporarilyDisabled: you can disable the simcard temporarily for maintenance tasks by setting this value to 1

DisabledUntil: used for automatic failovering. If the value is above the current time, the simcard is in failovered state

DisabledCause: last disable cause explained

ReenabledCount: how many times have the simcard reenabled after a failover

LastReenabled: the date-time of the last reenabling operation

TodayCallCount: call attempts from 00:00

ThisMonthCallCount: call attempts from the first day of the current month

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 „sure” level: 0

4-block all blacklist

5-allow only known numbers (listed in tb_knowngoodnumbers)

6- allow only known numbers that are 100% ok (sure is 1 in tb_knowngoodnumbers)

Co_....: fields used by server for fake call and sms simulations

BestDirection: used for automatic simallocation

BestPrice: used for automatic simallocation

EngineID: the corresponding engine (tb_engines.id)

Credit automation related fields:

CheckCredit: credit calculation or request/charge operations needed

CrequestEnabled: automatic credit request enabled/disabled (1/0)

LastCreditRequestTry: the date-time of the last credit request command issued by the server

AllCreditRequestCount: the number of credit requests

LastCreditAnswer: the date-time of the last answer to the credit request command

CreditRequestFails: subsequent failed credit request. Check the credit automation logs if this goes above 3

LastCreditRequestFail: : the date-time of the last failed credit request

ManualCreditRequestNeed: when set to 1, the server will request the credit from the simcard in 5 minutes

ChargeEnabled: automatic recharge is enabled/disabled (1/0)

MustCharge: when set to 1, the server will charge the simcard in 5 minutes

LastCreditChargeTry: the date-time of the last credit charge command issued by the server

LastChargeCardID: the database identifier of the last charge card used for this simcard

LastChargecardPrice: the value of the last charge card used for this simcard

CreditWhenCharged: the credit value after the last recharge operation

AllChargeTryCount: number of charge operations until now

AllChargePrice: the sum of the total charge card value

FailedCharges: subsequent failed charge requests. Check the credit automation logs if this goes above 3

LastChargeSuccess: the date-time of the last successfully completed charge operation

LastChargeFail: the date-time of the last failed charge operation

CreditDiffErrors: too big difference detected on sim credit reports

4.2.3. Basic Statistics

Shows the main quality parameters of your system.

CDRC: call attempt count

SL: speech length (duration in minutes)

ASR: average success ration (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. Chart views are supported only with this option

-Group by Called Gateway: list of destination gateway statistics

-Group by Traffic Sender: list of statistics by source

-Group by SIM Packet: statistics by SIMCard type

-Group by Provider Direction: statistics by called number prefix (first 4 digits)

4.2.4. Advanced Statistics

This is an extended version of Basic Statistics. You can find more grouping options here.

Additional reports:

-ASRB: average success ration, but here the “success” means a minimum amount of duration. Configurable in Settings Menu -> Thresholds Box

-ACT: average connect time. The time elapsed from setup until the connect in seconds

-PF: profit. This require your billing module to be properly configured

-SUCC: successful call count (same as ASR but not in percent)

-CCC: concurrent (simultaneous) call count

-RTP: media channel statistics

You can make the grouping by minute in this form by checking the “on minute” box.

The following “grouopby” options are available:

-: display summary data (no groupby)

Caller and Called: group by caller and called users

Caller: group by caller (source) user

Called: group by called (destination) user

Traffic Sender: group by caller (source) user, but show only traffic senders

Called Gateway: group by called (destination) user, but show only gsm gateways

GSM Engine: group by called gsm channels

Gateway, Packet and SIM Card: group by called simcard (and show the actual gateway and packet)

SIM Card: group by called simcard

Caller IP: group by caller ip address

Week –absolute: group by week, but with sum (don’t groupby to months)

Day –absolute : group by day, but with sum (don’t groupby to weeks)

Hour –absolute: group by hour, but with sum (don’t groupby to day)

Week: group by weeks

Day: group by days

Hour: group by hours

Minute: group by minutes

Day Compare: comapare current weekday with last week the same day

Called SIM Packet: group by called simcards group

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 callerusers

4.2.5. Disc. Reasons

Disconnect codes in graphical form by any traffic direction.

The server will collect the reason in the most appropriate format depending on the protocols used. For example for a call from voip to gsm if the disconnect was caused by the gsm party, then you will se the GSM network reason code here. Otherwise, if the disconnect source was the caller party, and then you will see H323 or SIP reason codes here.

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 detected.

4.2.6. Line Monitor

This is a simple listing of your channels. You can discover all simcard problems by scrolling down this list. (Missing channels are highlighted)

4.2.7. Capacity Check

This module tests the capacity for the predefined direction in priority order.

4.2.8. System Load

Shows system utilization statistics.

4.2.9. Server Console

Direct interface to the server command port. Type help to see the available commands. You can connect directly to any gateway interface.

Command defined on gateway interface:

help show this command list

info show status and important parameters

cmd launch the predefined process

exec launch the predefined process

file will send the requested file

showlog will send the last lines from the requested file

timeset will sent the current time

setini write to config file

getini read from config file

dtmf send dtmf

ftpget load from ftp

ftpput put file to ftp

selfupgrade do a selfupgrade

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 Tresto Gateways in the selected time preiod. You can restrict the listing by defining the source, severity or filtering.

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, a new CDR is stored in the database.

Id: database identifier. Auto increment

Datum: the date-time when the CDR were inserted into the database (call end time)

Connecttime: time elapsed until call fail or call pickup (routing+ringing time)

Realduration: speech length

Discparty: disconnect origination. 1=called or gsm, 2=caller or h323, 3=router (server)

Discreason: disconnect reason code. Explanations in tb_reasoncodes

Callerid: caller database id from tb_users

Callerip: the origination ip

Callernumber: caller phone number (or sip username)

Calledid: called database id from tb_users

Simid: called simid (if any)

Calledline: Engine (phone line) or the called proxy authorization id (from tb_proxyauth)

Calledip: the ip address of the called party

OrigCalledNumber: received called party number (not modified)

Callednumber: techprefix and the normalized called number. If the server will block the call too early, than you may have the “origcallednumber” here (no techprefix and normalization)

DialedNumber: the forwarded called number (sometimes only the “callednumber” will be insterted here)

Rtpsent: rtp packets from caller to called. 0 if no rtp routing. At least 1 if routed. If remains 1, then routing has failed

Rtprec: rtp packets from called to caller. 0 if no rtp routing. At least 1 if routed. If remains 1, then routing has failed

Rtplost: lost rtp packets

Rtpcodec: voice codec name

Rtpframes: rtp payload framed in one udp packet

Signalin: audio signal strength into the playback device

Signalout: audio signal strength received from the audio recorder device

Costprovider: call cost to the provider (ex. Tmobile)

Costenduser: cost for the caller (ex: a sipuser or traffic sender)

Costsales: sales commission if any

Costcompany: price for the reseller company

Costadditional: can be used for anything

Recfileid: if we have recorded the voice, then after this field we can found the recorded file

Comment: with details about the call setup and disconnect

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. Agent Statistics

Statistics related to callcenter operations.

4.3. Access -TManage

4.3.1. Users

This form will allow to manage the users of the system (Endusers, SÍP users, Administrators, Tech. Support users)

You can list the users with the following filters:

-ActiveNow: gateways with received status in the last 5 minute or endusers active (register or invite received) in the last 3 hour

-Active:

-gateways with received status in the last 24 hour or when “mustbeactive” is set to 1

-endusers active (register or invite received) in the last 24 hour

-All Enabled: where the “Enabled” field is not 0

-All: all users

-New Users: users added in the last month

-New Web Registrations: users added in the last month by the web registration form

-Low Credits: will list users with credit lower than 3000

ID: database id. Auto increment

Type: user type

0=enduser (usually a sip user). Operator if isoperator set to 1

1=reseller company (usually a sip reseller)

4=sim,gw or traffic owner (sim partnerid or gateway parentid show this id)

5=traffic_sender (parentid can be a simowner or a gatewayowner)

6=sales (parentid is the reseller id)

8=gsmgw, (parentid is the gatewayowner)

9=sipproxy, (parentid is the gatewayowner)

10=h323gw, (parentid is the gatewayowner)

11=isdngw, (parentid is the gatewayowner)

14=support (can operate with tmanage, has ftp account)

15=admin (can see and modify everything)

ParentID:

if a sipenduser then reseller company

if traffic sender, then traffic owner

if gateway, then gateway owner

BillingUserID:

If the current type is an end-user, then can have a BillingUserID where we send the invoices. If not set or the same ID as the current, than the bills will be generated to itself. For example in a company, all bills will be sent to the boss (company address), nit the employee

IsBilledUser: set to 1 if this user is not a real service user, but a user who pays for other user. Usually this is a company who pays for its employee.

UserGroup: users can call each other only if the user group is the same (default: 0)

usually users with the same parentid (reseller) has common parentid

Ringroup: a list of userid separated by comma (all number will ring when the actual user will be called)

Name: user first an last –name

Country: sip phone country (important for prefix rules)

ContactName: additional name

UseCallingCard: if has calling card (usable with pin codes)

CanDial: example: for sipuser is 1. for simowners is 0

IsCompany: if the current user actually is a company

BelongsToCompany: when a company has more then one subscriber. Used for example for short sip numbers.

Phone: user phone number (but not the sip phone)

Email: where the user can be contacted

Address: where the user can be contacted

Billaddress: where to send the invoices

TelNumber: sip telnumber.users can be contacted if we call there username or telnumber

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 in authentification.

Password: password applicable everywhere (sip, web, VPC, etc)

Ip: sipphone, sipproxy or gsmgateway ip address. The server will overwrite with the last known ip address

AuthIp: if we want to authenticate after ip, not after username/password

NeedAuth:

-If NeedAuth is 0, then the system is an open voip relay !!!!

-If NeedAuth is 1, then AuthIP must match (usually from SIP traffic senders)

-If NeedAuth is 2, then TechPrefix must match (usually from H323)

-If NeedAuth is 3, then TechPrefix and IP must match (usually from H323)

-If NeedAuth is 4, then user/pwd must match (usually from SIP end-users)

-If NeedAuth is 5, then username must match

AddDate: when the user has been inserted in the database

Rights: rights on user interfaces

0: no access

10: cannot login (disabled)

20: can login but no rights

30: a normal user

40: sales

50: admin

60: general admin

AddedBy: the user id who have added this user (sales, web registration, etc)

Commission: used for sales to define their comission percent from the enduser price

Reduction: sales user can give to enduser some percent (substracted from their comission)

LateFee: applicable when the user is late paying the invoice cost

PacketID: billing for users, traffic senders

BillingDay: usually 1 (the first day in every month)

Qualification: the importance for the user. From 0 to 10. for example if the user has big priority, then we route its calls to better routes

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 searated 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 4 or 5 digit separated by comma.

Can be used for trafic senders and gateways too. No need to setup a separate routing pattern if you use this restriction.

BlockPrefixes: list of called prefixes that will be blocked for the user (techprefix will not be considered here). Numbers listed here must have 7 digit length and separated with comma.

ContractState: the status of the contract

0- Unknown

1- Not applicable

2 -In Progress

3 -Active

4 -Terminated

ContractComment: additional comment for sales

Credit: when postpaid, then we also can set a max amount (which will reset in every month)

Enabled: if disabled, it behaves as if it were deleted

DomainName: sipproxy domain name

Port: signaling port

TransIp: secondary signaling ip

TransPort: secondary signaling port

RouteRtpCaller: routing mode if this endpoint is the caller

0=check called settings –this is the preferred settings

1=don't touch the sdp and the rtp

2=sdp correction if necessary

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

RouteRtpCalled: routing mode if this endpoint is the called

0=check caller settings

1=don't touch the sdp and the rtp

2=sdp correction if necessary –this is the preferred settings

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 changed, 2=yes but can be changed, 3 yes, and don’t change it

NatDetectDisabled: deprecated

Status: 0=inactive,1=registered, 2=speaking (if statusdate is too old, then treat as 0)

StatusDate: last status change

CalledNumber: last called number

CalledID: last called id

Discount1: discount percent. users can have discounts in for max 3 directions

Direction1: prefix. users can have discounts in for max 3 directions

Discount2: discount percent. users can have discounts in for max 3 directions

Direction2: prefix. users can have discounts in for max 3 directions

Discount3: discount percent. users can have discounts in for max 3 directions

Direction3: prefix. users can have discounts in for max 3 directions

TechPrefix:

The server can authorize and/or route the traffic after the incoming techprefix.

Sip users can have techprefixes too. this is usually common for reseller company users.

If no techprefix is specified, then it will be loaded from tb_pxrules if any.

Sim owners and vpc users can have a list of prefixes separated by comma.

If no techprefix is specified, 111 will be inserted for incoming called numbers.

If the techprefix is „-1”, then the original techprefix will be forwarded.

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.

Addtechprefix: we insert this number before the callednumber if the caller don’t send its calls with tech prefix

MaxLines: max concurrent calls allowed

maxlinetouse: deprecated

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 as email

mincreditonroute: if user has less credit, then we don’t even route the call

regtimeout: reregistration interval for sip proxies

maxsubsfail: we set the „nopriority” field when we reach „maxsubsfail” failed calls

subsfails: successive calls with duration smaller than 20 sec

nopriority: this gateway has lowered priority in the routing until this date

noprioritycount: successive lowered priority countminasr:

minasr: minimum asr before failovering

minacl: minimum acl before failovering

mincallcount: min. Cdr records to calculate minasr and minacl

lastrouted: last call time

active: applicable for gsm gateways.

display: text to display instead of username

description: important comment

comment: any comment

lastrectime: last status receive from this gsm gateway

realgw: we can have fake voip-gsm gateways

temporarilydisabled: gsmgw is temporarily disabled

onlytestcalls: we allow only calls with techprefix 999

testprefix: we allow only this techprefix

datum: when the user has bee inserted into the database

mustbeactive: if the gsm gateway must be active. Will do actions if this field is 1 and the gateway is not active

notactivecount: how many time we found that the gw is not active

channelcount: gsm channel count

minline: minimum active lines. If we found less line active, then we do actions

nominlinecount: : how many time we found that the gw has not enough line

prioritypartner: this partner will have priority on this gateway

callerpriority: this caller prefix will have priority on this gateway

calledpriority: this called prefix will have priority on this gateway

autopriority: set by server. If the gateway is wrong, then we lower the priority until this time

absolutepriority: if we set it greater then for other gateways, all calls will be routed here, until it is filled, regardless to other routing settings

priority: gateway priority

swversion: gateway sw version

lastrestart: gateway last restart

pingtime: deprecated

avgkbitssec: deprecated

maxkbitssec: deprecated

bandwidth: deprecated

restartcount: gsm sw restart count

pcrestartcount: gsm gw (pc) restart count

lasterror: last error message from this gw

lastlog: last log message from this gw

callsigaddr: h323 port

isfake: we can have fake voip-gsm gateways

forwardearlystart: if we can send media parameters before callstart (OK for INVITE). 2 if check called

changesptoring: if we have to change the session in progress message to ring. . 2 if check called

identityforward: we can toward these kinds of usernames and the other we rewrite to „identityrewrite”

identityrewrite: if the caller username don’t match the identityforward prefix, then we rewrite it

PlayAdv: if we can play advertisements for this user

Maxmonthlycredit: max allowed credit/month even if the user is postpaid (in ft not in filler)

Maxmonthlycreditend: max Maxmonthlycredit (because we increase Maxmonthlycredit by maxmonthlycreditinc every month if the user was active)

Maxmonthlycreditinc: determines how much money we add to Maxmonthlycredit every month

ContractNumber:

Contact Status:

0-Unknown

1-Not applicable

2-In Progress

3-Active

4-Terminated

Contract comment: any usefully comment for sales here

Noanswertimeout: will redirect if no answer received

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. Gsm gateway settings will overwrite the traffic sender settings if is not set to -1

sendsmsalert: use for support and admin acounts. Will send sms notification to the configured “phone” when a critical error occurs

sendemailalert: use for support and admin acounts. Will send email notification to the configured “email” when a critical error occurs

sendsmsreport: daily sms report for support and admins

senddailyemail: daily email report for support and admins

sendmonthlyemail: monthly email report for support and admins

Missed by SMS: notify about missed calls by sms. Usually used for endusers

Missed by Email: notify about missed calls by email. Usually used for endusers

Can watch sim packets: list of packetid separated by comma, used for VPC access. The actual partner can see this simpackets with his VPC account

Can watch users/devices: list of users and gateways id separated by comma, used for VPC access. The actual partner can see this devices with his VPC account

Access Rights: specify wich fielss are allowed for the user in the VPC application

0: simcard and traffic sender fields are not shown

1: simcard related fields are not shown (simid, packetname)

2: traffic sender related fields are not shown (name, username)

3: all fields are shown

CLI: CLIR and CLIP settings

0: forward always

1: normal handling

2: forward as asserted identity always

3: forward as asserted identity only to trusted domains

4: hide

5: force hide

IsOperator: specif if the user is a callcenter operator, or a normal enduser

Choosecodecs: list of supported rtp payload formats in priority order separated by comma. Only one will be selected. Don’t set this field 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.

4.3.2. Devices

Administarton of Tresto Gateways, Other GSM Gateways, H323 Endpoints, SIP Proxies, ISDN Gateways and other compatible devices. The fields are the same as for the Users (see above)

If the actual sipproxy require authentification, then we store the accounts in this table

Id: database identifier. Auto increment

Priority: Account priority (accounts will be used in priority order or in round-robin if they have equal priority)

Username: sip username used in authentification

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. When it reach 0, then we switch to the next account if any

DateEntered: record insertion date

LastUsed: the date-time when the server was routed some calls with this account

ProxyID: to which proxy the account belongs

Enabled: set to 0 to disable the usage of this account

SubsFails: the number of subsequent wrong calls with this account. If subsfails will reach a predefined value (30 as default), it means that there is some problem with this account or the money/time limit have been expired, and the server will switch to the next account if any

4.3.3. Groups

Grouping of several items will ease the administrations tasks. The following type of items can be grouped:

SIM Packets

Users

Gateways

Traffic Senders

4.4. Routing -TManage

4.4.1. Firewall

The firewall rules are checked first when a call are initiated (SETUP or INVITE received), so this is the most effective way to block some unwanted traffic sender.

All ip address are allowed except those are listed if the ip with ‘*’ is 1.

Otherwise (if the ip ‘*’ is set to 0) all address are blocked except those that are listed here.

4.4.2. Prefix Rules

You can rewrite prefixes before they arrive to the routing by entering your preferences here.

The Tresto 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.

For example you can set up a rule which defines that every incoming number from ip 111.111.111.111 on H323 if begins with 1234 must be rewritten to begin with 56. Number 123499999 will be rewritten to 5699999.

4.4.3. Blacklisted

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

4.4.4. 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 4.5.1 for details regarding filtering.

Blacklist fields:

-telnumber: country code + operator + telnum

-sure: levels

tb_blacklist.sure:

0 –probably good numbers (reput)

1 - not sure (holes)

2 - probably wrong number (monthly autodisabled)

3 - very sure (roaming numbers)

6 – always block (not only to gsm)

filtering: determines how we check the blacklist and the known numbers

0 -no filtering

1 - filter if very sure blacklisted (tb_blacklist.sure >=3)

2 - filter if probably blacklisted (tb_blacklist.sure > =2)

3 –filter if suspicious (tb_blacklist.sure > =1)

4 - filter if present in blacklist (any tb_blacklist.sure)

5 - filter if not a known number

6 - filter out if no sure known number

4.4.5. Routing

For every time period and direction a “Routing Pattern” needs to be defined. Every Routing pattern has a list of routing directories which may be Tresto GSM Gateteways, other H323 gateways or gatekeepers, ISDN gateways or SIP proxies in priority order. Set up as much directories with the same priority order as possible so the routing engine can prioritize itself after other settings (device priority, LCR, quality,BRS)

Generic rules can be defined by setting the pattern priority lower. For example for every call that doesn’t have a specific route can be routed to a specific direction (otherwise is dropped)

There is a list of typical time definition. If none of them mach your needs, the “Start-End Time” entry can be selected to specify proprietary intervals.

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.

*Tipp: 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 ‘*’). Rights can be enforced by setting “Enabled Prefixes” for all Traffic Senders

Routing Configurations

Try to set up all routing rules and prioritizations using this form.

Try to avoid prioritizations by gateways, simpackets or channels (absolutepriority, priority, allowedpartners, prioritypartners, etc)

Almost all kind of configuration can be set up by using only the “routing” form

4.4.6. Routing workflow

Introduction

The routing in the tresto sofswitch means deciding on which active gateway or gsm channel should we route the incoming call from traffic senders and andusers.

The routing is influenced mainly by the following:

-device ownership and access rights (allowedpartner settings)

-routing time, direction and the selected pattern (device/packet priority list)

-various priority settings

The routing is blocked if the following conditions are met:

General conditions

syntax error in incoming number (or not known)

max call/day, max speachlength reached (licensing option)

Caller user check

the traffic sender reached their maximum channels

aller gateway, simcard or simpacket is not enabled or temporarily disabled

failed authorization (wrong originating ip, bad username or password or wrong techprefix)

Caller “CanDial” setting is set to false

Caller tb_users.enabledprefixes not match (‘*’ allow all numbers)

Check if other traffic sender has the same ip/techprefix (caller mismatch)

Routing

direction and time don’t match a routing pattern

no active device or simpacket from the selected pattern priority list

Called device/gateway checks

Called gateway(s) is not enabled, not active, temporaydisabled, allowedpartners don’t match or any other problem

Called gateway onlytestcalls not match

Called gateway enabledprefixes not match (‘*’ allow all numbers)

Called gateway blockprefixes match

Called device filtering option doesn’t allow blacklisted number level (if the incoming number is blacklisted)

gateway has testprefix but does not match

Called simpacket check (only for gsm directions)

Packet is not enabled

Caller is not listed in allowedpartners

packets.waitaftercall second not elapsed since last call

packets.filtering. blacklist/whitelist restriction (filtering option doesn’t allow blacklisted number level (if the incoming number is blacklisted))

Simcard check (only for gsm directions)

Called simcard(s) is not enabled, temporaydisabled, not ready, allowedpartners don’t match or any other problem

partner is not allowed on the simcard (allowedpartners)

the simcard is prepaid, but it doesn’t have enough credit

two subsequent calls cannot be routed to the same simcards (configurable)

there was a credit request or recharge in the simcard in the last minute

cannot request credit from prepaid card more than 5 times

maxmonthlyminutes,maxdailyminutes,maxallminutes, maxmonthlyminutespeak are reached

no report from the channel for more than 5 minutes (the gateway may have lost its network connection or power)

Routing priority order

If emergency number, than the defined emergency route has the biggest priority

Routing pattern priority (if two or more pattern overlap)

Routing pattern direction/time best match (if two or more pattern overlap)

Called gateway Globalabsolutepriority

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 gateway is not failowered -value lower or higher than the current date-time (for automatic failovering)

Called gateway is not failowered for the currenct called prefix (direction)

Simcard is not failowered - value lower or higher than the current date-time (for automatic failovering)

SimPacket is not failowered

Tpercek priority (hungarian tmobile specific)

Routing list priority

Elapsed time from last call disconnect is more than 10 sec

Gateway callerpriority match the caller number

Gateway prioritypartner match

Simcard priority partner match

SimPacket nopriority partner not match

SimPacket priority partner match

Gateway priority (simple)

Simcard priority (simple)

Simcard minimum monthly speechlength not reached

Simcard minimum daily speechlength not reached

Simcard desired monthly speechlength not reached

Simcard desired daily speechlength not reached

Simcard todayspeachlength desc order (simcards with more callduration has lower priority)

Simcard thismonthcallcount desc order (simcards with more callcount has lower priority)

Simcard thismonthspeachlength desc order (simcards with more callduration has lower priority)

Simcard a.creditrequestfails desc order (simcards with more failed credit requests with lower priority)

Gateway ready channels (balance calls across gateways)

Last call begin on the simcard (balance calls across simcards -simcards with the most recent calls has lower priority)

Simcard currspeachlen desc

Simcard GSM Fieldstrength

Simcard lastrectime (for randomizations)

The routing process. Short 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. (callin option). Replace caller and called after the config.

Correct the called number string if it is corrupt.

Check min/max length of the called number

Check if the incoming call is a testcall. Set the testcall flag is yes

Check and apply prefix roules (tb_pxrules – rewriting the called number)

Authenticate the caller (after username/password, ip pr techprefix). Drop the call with “no such user” reason on fail

Add techprefix if needed

Setup sip parameters if the call was arrived from sip

Normalize called number (check prefix, area code, etc). Drop if wrong number

Check if subsecvent wrong call

Check if the caller exceed its max line restriction

Check blacklist and whitelist

Check the embedded firewall

Check if caller called itself

check if called if a sipuser (Username, telnumber, short telnumber)

check the forwardalways option

check the ringgroup option

setup called endpoint if found

get time variables (peak, holiday, etc)

Get the correct routing pattern

Check routing list in priority order

If simpacket found, than Check simrouting

Drop the call if no route found

4.4.7. RADIUS

Define the radius servers, protocol and login information here. Used for authorization and billing.

4.4.8. BRS

Short name for “Best Route Selection”.

In addition to LCR (Least Cost Routing), the Tresto routing engine can take in consideration the quality of the route.

If you would like only LCR, simply set the “Quality Percent” field to 0.

If we put some directions with equal priority in the pattern, then the system will choose the routing automatically depending on price and quality, when other settings don't modify the routing (min/max minutes, gw/sim priorities, failovered directions, etc)

The server automatically calculates am „autopriority” on every route. This priority is the combination of the quality and the price. The quality is calculated as an average of daily asr/acl and monthly asr/acl. The price is calculated to pricecalcsec seconds with the given minammount and billingstep (from packet prices). The server route the traffic on the higher priority direction, BUT it will try the other routes periodically (to check if quality have changed). You can change this „next time try” setting by changing the values of TryedCount,NextTry, NextTryCount. If the best quality gateway for a route will change, then we will reset TryedCount,NextTry and NextTryCount values to there defaults. (so we can recheck quicker)

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. So, the final result (AutoPriority) will be lowered only if we have too wrong acl, asr or price.

Default is 30%. This default means that the AutoPriority will change only if price will change with 2-3 ft or asr will change with at least 15% (considering asr between 10 and 80, price between 0 and 40 and QualityPercent as 50%)

-ARSDay: last day ARS (automatically calculated every day)

-ACLDay: last day ACL (automatically calculated every day)

-ARSMonth: last month ARS (automatically calculated every month in the last day)

-ACLMonth: last month ACL (automatically calculated every month in the last day)

-MinASR, MaxASR, MinACL,MaxACL: when asr or acl reach the min value, then the line is considered very wrong. When it reaches the max values, then the line is considered very good. Must be configured manually for every direction, because the statistics will change dramatically by country

-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. It will grow only until 7

-NextTry: we will route calls to this route beginning with this date. Will grow exponentially until 1 month.

-NextTryCount: we will route NextTryCount calls on this route next time. ( > CurrTryCount)

-CurrTryCount: counter to know how many times we have routed in this direction

-AutoPriority: the current priority calculated from these values and from the price settings (the result)

To see how much a parameter change will modify the final AutoPriority value, you can find a demo named AutoPriorityDemo in Tmanage, Tools menu. Before changing any value in the BSR table, please play a little with this demo.

4.4.9. Failovering

Tresto server and gateway will make automating failovering between sim channels, sim packets and gateways. The rules can be defined using this form.

You can check the route status also from here.

ID: database id. Auto increment

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 this date. When the time elapses, we try this route again. This will grow exponentially.

NoPriorityCount: we have failovered NoPriorityCount until now because of SubsFails. The bigger is NoPriorityCount, the longer we do the deprioritization (NoPriority)

NoPriorityCountD: : we have failovered NoPriorityCount until now because of statistics

Manual: all routes will be added automatically to failover table with a minimum of quality requirements

Enabled: failovering enabled

Datum: record insertion or last modification date

Comment: why was the record modified last time (reason)

4.4.10. SIM Channel reservation by caller protocol

Best quality (ASR and ACD) SIM channels can be reserved for sip or h323 originated calls.

In the sim table the reserverfor field can have the following values:

0=cannotreserve: this channel will not be reserved

1=sip: always reserver only for SIP (manually assigned)

2=h323: always reserver only for H323 (manually assigned)

3=dynamic: can be allocated by the server dinamically (hourly check) -this is the default value

4=sipdynamic: allocated automatically for SIP

5=h323dynamic: allocated automatically for H323

For every simpacket you can restrict the maximum allowed reservations by the maxalloc field. (usefull to not reserver all channel from the same simpacket)

To setup the channel reservation use the following configuration values (vserver, simplatform config):

-reserveforh323: reserve capacity for h323. reservations will be disabled if less than 1

-reserveforsip: reserve capacity for sip . reservations will be disabled if less than 1

By exmaple if you set the reserveforsip field to 5, you can be sure that 5 channels always remain free to be used by calls received with SIP protocoll (H323 originated calls didn’t consume all your channels)

4.5. Billing –TManage

Tresto Servers and Gateways are ready for prepaid and postpaid billing.

The pricing must be set from TManage –Prices Settings form

You can list and compare the prices for different directions in the Price List

The Billing are done from the “Billing” form

4.5.1. Price Settings

Pricing of the CDR records are done after the prices defined on this form.

You can define “Price Groups”. All price settings that belong together (accounting with a partner for example). This is located in the left side of the Prices form.

~~Invoices can be generated automatically by the server and send by email, or can be loaded manually by using the “Billing” form.~~

~~You can schedule when to send the invoice or report to the partner or for you (defined by the mailto entry)~~

~~The report format can be defined by “Invoice Type” and “Group by” fields.~~

Below a “Price Group” you can have several Price Setups named “Directions” (the middle column in the form)

For example “Traffic from Telcom SA” or “Traffic to T-Mobile direction”

Here you have to set up the actual prices. The price setup is further divided into different prefixes (the right side of the form -Pricelist), because it is very common that you have lots of directions in a provider pricelist.

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-time when the payment was received~~

Direction name: name of the billing entry

Type: specify the type of the price. For exampe the prices used when billing to endusers, or our minute costs to service provicers.

Price culations will be saved directly in the CDRs, thus can be used in prepaid billing. In the CDR records, the following fields are used for price calculation:

-costprovider: used to calculating the minute price thay needs to paid to service provider (Tmobile for example)

-costenduser: used for billing to our endusers (sip endusers, traffic senders)

-costcompany: can be used for profit calculations

-costsales: sales comission. If not set, than will be calculated by the comission value in users settings

-costother: can be used for any custom price calculation

Action: How to handle the calculated price in reporting. For example in “profit” calculations whe have expenses (prices payd for service providers) and incomings (from our endusers). Thus we can simply substarct the expenses from the incomings to get the “profit”

Billing Steps: provider specific billing interval in sec

Min. Amount: the minimum payable duration in sec

Free Amount: you may have packets when the first X second is free

Free After: you may have packets when after X seconds the conversation is free

Currency: different providers may have different currency. Used for billing.

VAT Included: if the pricelist applied for this user is with VAT included. Set to 0 if VAT is not included. Used for billing.

VAT Value: the ammount of VAT applied for the pricelist. Will have effect only if “VAT Included” is checked

Convert to NET value: if you have defined the pricelist with included VAT, you should check this option, othervise you overcomplicate the billing process. Thus the VAT value will be substracted from the price, and you will have NET values in CDR records (try to use net values whenever possible)

Convert to HUF: if you have defined the pricelist in other currency than the native (configurations->currency), than your prices will be automatically converted to native currency in CDR records.

Traffic Direction: here you have to define the rules when the current pricing will be applied

Usually only one field needs to be specified here (for example all traffic from Telcom SA -caller)

The caller field will check the caller parent also, but the called field will not check the parent.

ValidSince, ValidUntill: the pricelist may be applied only after a specified date-time

Prefix: called number prefix (this will be loaded after “best fit”). Set to ‘*’ to be applied to all directions

Price: the actual price

CPrice: the price converted in your currency (“currency” entry in the Configuration form and converted after the values specified in the “Currency Converter” form)

Time Definitions: the time period when this rule is applied

Diff between enduserprice and providerprice means that price will be calculated by extracting the provider cost from the enduser cost for an already existing cdr record. Cannot be used for realtime (prepaid) price calculations. Usually used when calculating “profit” values.

By clicking on the “Clone” button, you can easily duplicate a price list (it is very usefull when you have to add only a few modification to a long pricelist)

The Billing button is a shortcut to the billing form (does not make the billing automatically)

Importing price definitions from file are done by clicking on the “Import from file” button.If you use the “default peak time definition”, the peak settings will be loaded from the global configuration (peaktimebegin and peaktimeend values). If this is not suitable (different service providers may calculate with different peak-offpeak definitions), you can set up the peak time definition manually (start – end hour).

The imported file must have four comma separated field: prefix (direction definition), flat price, peak price and offpeak price. If you use flat price, than leave the peak and offpeak price fields emty and vice-versa.

The easiest way to generate such files is to use Excel, fill the first four columns with these values and save as CVS file. (Don’t leave emty coloumns before the columns with data)

Importing price files may take some time, depending from your network connection speed.

4.5.2. Price List

On the List tab you can list all prices for a packet (by using the “Packet” list box) or to a direction (by entering a direction name or a prefix to the filter box)

On the Least Cost tab you can compare the prices from different service providers.

The Reference Packet usually is the price for your end-users.

Only peak (max) prices are compared for every direction.

On the Directory Check tab, lookups from the directory table are possible ( directory name – prefix match).

4.5.3. Billing

The server automatically calculates the price field for every incoming CDR record, based on price settings ( Section 4.5.1)

The following prices are calculated:

-enduser cost: used for invoicing for costumers

-provider cost: cost that needs to be payed for service operators

-sales cost: sales comission. If not defined in price setup, than will be loaded from users settings (“comission”) if any

-company cost: usually used for profit calculations

-other cost: for any other purpose

Billing can be done from

1. the “Users and Devices” form, Billing tab, by clicking on the “Generate &Invoice or Report” button (billing for the actual user)

2. set up required directions and click on the “Billing” form (in this manner, billing reports can be generated for more users)

The billing process will always take in consideration the selected date interval.

Billing form:

1. On the Customized Billing tab after selecting the required date-time interval and direction, the prices are calculated after predefined parameters (price/minute, billingstep). So you can do simple calculations using this form.

2. The CDR Prices tab will load the “enduser cost” and “provider cost” directly from cdr records (already calculated after realtime price settings)

3. Generating Reports and Invoices tab

Used for billing and reporting.

Fields explanations:

Provider: you must select the invoice emmitent here. By clicking on the “…” button, you can customize the company invoicing details.

Delete old invoices: if checked, than will clear the invoice files directory before saving the new ones.

Include inactive users: uncheck this checkbox, if you don’t want to generate reports (invoices) for inactive users (inactive for the selected period)

Include child users: for example you can select a Reseller as direction source, and all “child” users will be included in billing (where the parent id will point to that reseller)

Include CDR records: include call detail records in appendix

Language: language of the invoice

Grouping: you can select any grouping options to be generated as appendix for the report

Price values: select the price field from the CDR record after wich the billing are done.

Reporting: you can automatically save the generated reports or invoices to file, or open it one by one (you can decide what to do for every report -save, print or just preview)

Format: file format (text, pdf) or printig

Real Invoice: if you would like only a quick report for the selected user(s), you can do it by setting this option to “Don't generate real invoices”. If you choose to generate real invoices, than it will take special processing for it (required for bookeeping)

If you have selected a reseller, you should choose the “For Resellers” option. In this manner a real invoice only for the reseller company is generated. (A report will be generated for all child endusers, but those report are skipped from the bookeeping)

Invoice Comment: any comment here. This will not be shown on the report

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.

5. You can change the price settings whenever you want, but don’t forgot to Rebill your CDR records after the new settings. All CDR prices will be recalculated for the selected time interval and direction. Users and simcards credits will NOT be modified by rebilling!

Note: prior to generate pdf report you should configure the installed print to pdf driver to save automatically in the specified directory. The defult pdf printer can be configuted in the TManage menu on the Settings-> Options from. The “cutepdf” driver is included in the TManage install package.

For printig jobs, the default configured printer will be used.

If you would like to save more pdf to file at once, you should install a pdf printer driver wich support to set a default directory for files. (The Cute PDF driver found in install package don’t support this feature)

4.5.4. Currency Converter

Defines the conversion between your native currency and other currencies used in price settings. You should update this conversion prices as many times as possible.

4.5.5. Finances

You can use this form for your cash flow administration regarding your voip business. (Other simple alternative is Excel :)

4.5.6. Pin codes

Recharge codes used if you have prepaid cards printed.

You can generate random prepaid codes here.

4.6. SIM Platform -TManage

4.6.1. SIM Packets

Id: database primary key. Autoincrement

Provider, type, subtype: the name of the packet

Owner: simowner in case of simpackets

Allowedpartners: applied when it is a simpacket

AbsPriorityPartner: this partner will have big priority on sims that belong to this packet

PriorityPartner: this partner will have increased priority on sims that belong to this packet

NopriorityPartner: this partner will have lowered priority on sims that belong to this packet

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 „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.: +3630xxxxxxx)

2: area code + number (0630xxxxxx, 061xxxxxxx)

3: shortest possible number (xxxxxxx in the same simpacket or 0630xxxxxxx in other simpacket)

4: correct it to the most appropriate format if original is not correct

WaitAfterCall: how much time must be elapsed between calls to simcard belonging to this packet

MaxMonthlyMinutes: we don’t route more than MaxMonthlyMinutes to simcards belonging to this packet

MaxMonthlyMinutesPeak: maximum allowed minutes in peak time / month

MaxMonthlyMinutesOffPeak: maximum allowed minutes in offpeak time / month

MaxMonthlyMinutesWeekend: maximum allowed minutes in weekends / month

MinMonthlyMinutes: this packet will run on higher priority until the min minutes is reached

Price: default minute price if not set in tb_packetprices

BillingStep: second increments

MinAmmount: min billing seconds

FreeAmmount: free speech seconds

MinCreditOnRoute: if the sim has less credit, then we don’t route call to it

MinCreditOnCharge: if the sim has less credit, then we begin trying to charge it

Prepaid: 0=postpaid, 1=prepaid

SendFakeSMS: we send dummy sms on this sim

CanCallEachOther: the simcard in this packet will call each other periodically to generate incoming traffic

IncludeVAT: used when credit message information are received from simcards (typically via SMS) and the simcards credits are calculated without the VAT value

Currency: used when the credit messages received needs to be converted in native currency (“currency” global setting) format. If the currency is not the same as the native currency and the “convertsimcreditcurrency” global setting is set to true, than the received credit value is converted to the native currency, based on “Currency Converter” settings, found in TManage under the “Billing” section

MaxAlloc: helper settings when automatically alocating channels for a direction. (Depending on reserverfor simcard setting).

Here you can define the maximum count of simcards that can be reserved for the actual packet. Set to 0 to disable rezerving from that packet.

Credit Request Command: the command used by the server for sim credit request (used for recharge automation)

Credit Charge Command: the command used by the server for sim credit charge (used for recharge automation)

The request and the charge command must have the following syntax: <DTMF,action,simid,”message”,telnum>

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. The message can be any valid AT command

-2: will dial the specified telnum, and than send the message as DTMF.

If the message string if emty, tha only will dial the requested telnum, hold a little and than drop.

Example: DTMF,2,simid,"",172

-3: reserved for future ussage

-4: will send the specified message as SMS to telnum

4.6.2. Gateways

Used to configure your Tresto VOIP-GSM gateways.

The fields are the same as listed in section 4.3.1

4.6.3. Engines

Listing of gsm channels. The fields are self explanatory.

4.6.4. 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. Corrupt entry

Sim Disabled: simcard “Enabled” is set to false
GW Disabled: gateway “Enabled” field is set to false

GW Missing: last message received from gateway is more than 8 minute old

SIM Missing: last message received from simcard is more than 8 minute old

SIM Temp. Disabled: simcard “Temporary disabled” field is set to true

GW Temp. Disabled: gateway “Temporary disabled” field is set to true

No Packet Set: no packet settings are present for this sim. You always need to set the correct packet settings for all simcards

Packet Disabled: simpacket “Enabled” field is set to false

Closed: simcard channel status is set to closed. A simchannel can be closed for different reason. Cannot register to gsm network, Sim Change, Just restarted, etc. If this status persist, check the logs for that simcard

Failovered: server has detected wrong quality on the simcard. Traffic will be forwarded to other simcards if possible

AutoDisabled: same as “Failowered”

Cannot Get Credit: automatic credit request failed. Check the credit automation log for errors

Wrong Statistics: wrong statistics for the current day

Wrong ASR: wrong ASR detected on the channel. Treshold values can be set up from the TManage -> Menu -> Settings

Wrong ACD: too small average speechlenth detected on that simcard

Expired: maximum monthly or daily speechlength limit reached (SimPacket option)

Low Credit: prepaid simcard expired

Gateway Disc.: gateway is offline or just restarting.

Not Ready: simcard is not ready for some reason. Maybe just starting. Checj the logs if this status persist

Ready: simcard is ready to accept incoming call

Dialing: outgoing call setup in progress

Ringing: ringing signal received from gsm network

Speaking: gsm engine is ringing or call in progress

Call ending: dropping the current call

DTMF: dtmf or credit request/recharge message in progress

Simulating incoming/outgoing: calls between simcards generated by the server

Routing: the call have been routed from the server, but still not arrived to the gsm gateway. If this persist, check the log for errors. Usually means firewall/NAT problems

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.

4.6.5. Credits

For Identification of sms and dtmf messages received from simcards that are useful for credit request and charge

Type: 0=other, 1=succ charge without credit info,2=credit start/end, 3=failed charge, 4=need charge

Msgbgn: begins with

Msgeng: ends with -used if type is 0 (replace) or 2 (end of credit), 4 (new credit. usually 0)

Priority: check order (longer messages usually first, to not include shorter) –higher values first

4.6.6. SIM Distribution

All simslots are listed here.

Probability values:

not sure: the simcards was seen more than one month

probably: the simcards in the last month

sure: the simcards in the last week

The other fields are the same as described in section 4.2.2.

4.6.7. SIM Utilization

List of simcards in call duration order.

4.6.8. New Simcard

You can add new simcards by using this form.

However, the simcards are usually added automatically. If they are active in the gateway they will register automatically. Usually only the owner and the packet must be set manually.

4.6.9. New Charge Card

Add new chargecards with this form.

The charge card will be charged only on the simpackets selected (“packets for”) and if the owner will match.

4.7. Other -TManage

4.7.1. Configurations

Global system configurations.

Basic configuration are vital for the system to run correctly.

Check the “Comment” field for each setting for more help.

4.7.2. Direct Query

From this form, direct SQL queries can be done against the Tresto backend. Use it carefully!

4.7.3. Voice Here

With this utility, the conversations on Tresto gateways can be listened in realtime.

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 (instead of notepad :)

4.8. Gateway Configuration

All configurations can be done from the TManage 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 1.6 hardware)

ModemControllPort=X

//if there are no "In" and "Out" device, we use this settings both for in and out

##AudioDevice="Xaaaaaa"

//from engine

AudioDeviceIn="1Audio Codec 1000"

//to engine

AudioDeviceOut="2Audio Codec 1000"

//simchange settings

simchange1= 00:00:00 - 00:00:00 - 01234567890123456789

//if 1 then the conversations (voice) will be saved to files on encrypted, compressed format

record=0

//init commands only for this engine: atinit1,atinit2 ... atinit19

##atinit1=XXXX

##atinit2=XXXX

##etc

//simcard id's in the slots

simcard0=01234567890123456789

simcard1=

simcard2=

simcard3=

Simchange settings explanation:

format:

simchange1= 2004.03.05/13:00:00 - 2004.03.07/13:00:00 - 8936302403070132426 (from date - to date)

simchange2= 10:20:00 - 10:26:00 - 8936302403070132426 (every day from time to time)

simchange3= 2/10:20:00 - 7/10:26:00 - 8936302403070132426 (from Tuesday 10:00 to Sunday 10:00)

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

special characters are: - , / . :

4.8.2. Gateway Basic Settings

//the name of the gateway. uppercase with "GW" suffix. must be descriptive

alias=NEWGW

//hardware version: 10,16,18 or 19

hwversion=18

//mode of operation. virtual available from hw 1.9

virtualmode=0

//server ip address

serverip=195.70.36.43

//number of hardware audio buffers (the jitter base is soundbuffcount*10)

sndbuffnum=8

//min jittertime in milisec (the minimum of the dynamic maximum jitter time. must be larger than soundbuffcount*5)

minjitter=130

//maximum jittertime in milisec (the maximum of the dynamic maximum jitter time. must be larger than maxjitter. if equal, then static jitter will be applied)

maxjitter=350

//0=off,1=dynamic,2=fixed,3=dynamic+off

silencedetection=3

//codecs to use: onlyg723, onlyg729, onlyg72X, onlyg711

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. Gateway Advanced Settings

//search for gatekeeper

gkdiscover=0

//gatekeeper supported prefixes (from 1 to 100)

gkprefixes1=

gkprefixes2=

gkprefixesX=

//volume in (sound device recorder from the gsm engine). defaults to 40 in hw. 1.8, 100 in hw 1.6

volumein=

//volume out (sound device player to the gsm engine) defaults to 75 in hw. 1.8, 100 in hw 1.6

volumeout=

//gsm engine receive gain. defaults to 0 in hw. 1.8, 64 in hw 1.6

vgr=

//gsm engine transmit gain. defaults to 0 in hw. 1.8, 64 in hw 1.6

vgt=

//ethernet interface to use. leave it empty to listen on all

netinterface=

//don't touch it usually

launchcmd=voipgsmgw

//install status: 0=idle, 1=wait, 2=normal

opmode=1

//will be set to false after first init

firstinit=true

//tracelevel 1-6 't'

trace=t

//record voice

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. Defaults to 38000

minrtpport=36000

//max media endpoint port. Defaults to 63999

maxrtpport=63999

//call with immediately pick up

fakecalls=0

//set to 1 if you want error report

errreport=0

//codec frames in one packet: g723frames, g729frames, g72xframes, g72xframes

g72xframes=1

g723frames=1

g729frames=2

//minimum frame count in 1 packet (apply even if the other end says another settings)

g72xminframes=0

//if set to 0, then we send connect when the call arrives

waitforring=1

//reset the engine/gw if we reach this limit

maxnotconnectedcalls=25

//reset the engine/gw if we reach this limit

maxwrongcalls=40

//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)

destroydelay=100

//max simchange wait in sec (if sim in call, we will wait until disconnect). default is 5 min

simchangewait=300

//max simcard/channel (will auto detect. don't overwrite)

maxsimcount=

//additional hang-up on the call end (to increase the real duration)

delayonhangup=0

//if we can retry the call

allowreroute=1

//deprecated, as we use only self reroute now

onlyselfreroute=1

//all calls will be routed on the onlyphone if enabled (no simcard requested from the server). deprecated

##onlyphone=3

//automatically increased on every gw (re)start

restartcounter=0

//usually set to 1

enableh245tuneling=1

//usually set to 0

connectwithmedia=1

//usually set to 1

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. no problem if we use 8 on a 4 channel gateway

chanellnum=8

//pincode applied globally to all channels (if not specified in phonex section)

pincode=

//will set the simcards to don't request for pincode (pincode must be set in gateway or phonex sections)

autoremovepincode=true

//volume in/out (will be overwritten with volumein and volumeout)

volume=

//auto gain enable/disable

doautogain=0

//listening tcp port (may be changed on NAT configurations)

signalport=1721

//0=no watchdog, 1=yes, 2=unknown

paralellwatchdog=2

//set to 1 if you want to remap usb audio lines

mustremapaudio=0

//set to 1 if you want to reread all simcards

readallsims=0

//set to 0 if you don't want an usb remap on every pc restart

canremaponstart=1

//if we have usb audio and don’t have other usb device then allow to remap if needed

canremapusbaudio=1

canrenewusbaudio=1

//set to 0 if you don’t want panel reset

canpanelreset=1

//set to 0 if you want an usb remap when the service will start

mustremapaudio=0

//disable reading sms messages

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). configure from the vnetcfg tool

net_conntype=

//network interface ip address. configure from the vnetcfg tool

net_ip=

//network netmask. configure from the vnetcfg tool

net_netmask=

//network default gateway. configure from the vnetcfg tool

net_defgw=

//network primary dns server. configure from the vnetcfg tool

net_dns=

//dialup phone number

net_phonenum=

//network ppp username. configure from the vnetcfg tool

net_username=

//network ppp password. configure from the vnetcfg tool

net_pwd=

//maximum speech length allowed in sec. defaults to 10800 (3 hour). set to 0 to disable

maxcallduration=

//maximum ringtime allowed in msec. defaults to 52000 (52 sec)

maxringtime=

//password on local command line. default is cmdpwd1234

cmdpwd=cmdpwd1234

//towarding dtmf from voip to gsm

forwarddtmf=1

//what to do with incoming calls (0=drop,1=hold a little then drop,2=auto forward,3=forward to server as forwardnum,4=forward to number requested by dtmf)

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. please wait for connect"

playdtmfforward=

//auto forward number (used if inccalls is 2)

forwardnum=

//used to require the number to forward to (when inccalls is 4)

promtfile=

//allow towarding dtmf messages to gsm network

allowdtmf=true

//how we send the ring signal. 0=send immediately and always, 1=send when received from gsm, (on the server you can set a timeout)

exactring=1

//used by the ipconfig tool. don't edit manually

ethcfg=

//local ip stored here. don't modify

localip=

//date-time of the last config download from the server

lastinisave=

//date-time of the last config upload to the server

lastiniupload=

4.8.4. Watchdog settings

[watchdog]

//set to 0 if you don’t want pc restarts

canrestartpc=1

//set to 0 if you don’t want service restart (then the watchdog will have no effect)

canrestartservice=1

//set to 1 if you want a reset on every night

canrestartdaily=0

//how often can the watchdog restart the service. defaults to 1000*60*25 msec (will change dynamically)

MAXSERVICERESTARTIVAL=

//how often can the watchdog restart the pc. defaults to 1000*60*45 msec (will change dynamically)

MAXPCRESTARTIVAL=

//max time to wait for watchdog reset. defaults to 1000*60*20 msec

MUSTRECEIVEOKIVAL=

4.8.5. Other settings

//at commands sent only once for all engines

[atonce]

#hardware version

cmd0=AT+WHWV

#sw version

cmd1=AT+WSSW

//at commands sent for all engine at every init

[atinit]

##cmd0=XXXX

##cmd1=XXXX

##etc

//prefix depending settings

[prefXXX]

connectwithmedia=0

g723frames=3

g729frames=6

[ipmux]

ipmuxenabled=0/1

[sounddevices]

//will be filled when reading all sims, so you can copy device names from here

4.8.6. Handling incoming calls from GSM network

Depending on the “incalls” (gateway configuration) settings, incoming calls from gsm network can be handled in several ways.

1. When incalls is set to 0

-all incoming calls to gsm simcards will be dropped immediately

2. When incalls is set to 1

-the engine will pickup the call, hold a little (random time, but maximum 1 minute), and than drop. Also used in call simulations.

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 tresto server specified by the “outserverip” setting in the gateway configuration.

-on the server, the call will be forwarded to the “gsminccalled” number (SimPlatform configuration). If the “gsminccaller” option is filled with a valid phone number, than the callernumber will change accordingly.Otherwise the caller number will be the original caller. The ip caller address can be changed with the “gsminccallerip” option. (thus you can simulate the routing from a predefined user)

5. When incalls is set to 4

-the caller will be asked to enter the target number (handled with dtmf), and the call will be forwarded to that number

-the prompt played to ask the target number can be set by the “playdtmfreqfile” setting. This will have to point to a PCM 8000kHz, 8 bit mono wave audio file.

-the prompt to be played if the forwarding has failed can be specified by the “playdtmffail” setting. When the forwarding is in progress, the “playdtmfforward” file will be played to the user.

-the call will arrive to the server with the ‘222’ techprefix, and you can setup a separate routing roule for this tecprefix

4.8.7. Operator friendly gsm termination

Not using industrial engines

On request, we can deploy our gateways equipped with normal gsm phones instead of industrial gsm engines. Ask the Tresto support for more details

Virtual Engines

Each simcards can have it’s own GSM engine (in other gsm gateway the engines are used by more simcards)

GSM Cell Lock

Because Tresto 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 tresto simbank, your simcards can be stored in a central location, and used in gsm gateways installed at different locations.

Delayed network registration

A delay time can be configured to elapse between succesive engine (re)registrations.

Intelligent routing

Ballancing the traffic across your simcard based on price and quality

Handling of incoming calls

In usual GSM gateways there are no simple mechanisms to handle incoming calls from the gsm network. In a tresto system all calls can be forwarded to your support team, so each call can be responded accordingly.

No GSM network owerload

Tresto GSM gateways occupy only 8 channels

Fast detection of dead channels

Failovering from simcards blocked by the operator or with wrong quality

Automatic blacklist calculation

Wrong numbers will be detected and blocked on the server (not forwarded to the gsm network)

Minute limits

Each simcard can have different daily, monthly and other limits

Time between subsequent calls

Calls will not be forwarded to gsm gateway without a delay between (configurable) them

Many other tricks

Ask the Tresto support for more details

4.9. Call Center –TManage

4.9.1. CC Users

Will load the callcenter operators (agents). Here you can add, delete and edit them.

The basic settings are placed on the Edit Operator tab. SIP enduser related settings can be edited on the Advanced tab.

With the Campaign drop-down list you can assign the selected operator to a campaign.

Operators must have entered and quit date set correctly. (If the quit date is elapsed, than the operator is not allowed to work with TAgent)

Technically operators are just sip endusers (tb_users.type =0) by the isoperator flag is set to 1.

4.9.2. CC Campaigns

You can setup the campaigns in this form.

Campaign will start to run when the StartDate is reached and will run untill no more clients (phone number) are assigned or the EndDate was reached. This means that Tagent -> Automatic Calls will run if this conditions are met.

The “Display” field will be displayed for the operators in TAgent “Automatic Call” window.

By clicking on the “Load Statistics”, a sort statistics window will be displayed regarding the selected campaign.

Handling invitations:

Load Invitation: will download the assigned invitation fron tha database. This can be any file, but Microsoft Word document are preffered.

Save Invitatio: will save the document back to database. Prior to hit this button, the document must be edited, saved and closed.

Print Invitations: will print a separate invitation for all invited clients in this campaign.

You can use special keywords in word documents and that will be replaced with the coresponding value. This keywords are the followings:

[client_name]

[client_address]

[presentation_name]

[presentation_price]

[presentation_display]

[operator_name]

You must include the brakets too.

4.9.3. CC 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.

4.9.4. CC Presentations

Used to store the different presentation locations. When a client is invited, the operator will select a presentation for them.

4.9.5. CC Checklist

Can be used in persentations to print the list of invited users.

4.9.6. CC Clients

The client (phone number) database.

Clients can be assigned and/or reassigned to campaigns with the ease of this form.

You can search across client by a lot of condition presented on this form.

By selecting the “Last Status” filer, the users can be searched by the reason code in the last campaign

By selecting the “Any Status” filer, the users can be searched by the reason code in any campaign

Importing client database can be done from external csv or dbf files. These files must have the following fields:

CSV file columns (must be in this order):

-Name (string)

-Landline phone number (string)

-Mobile phone nuber (string)

-Zipcode (short string)

-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):

-IRSZ: zipcode

-VAROS: city

-UTCA: address

-ROBIN: robin

-IRSZ: zipcode

At least the landline or mobile phone must contaion a valid entry.

4.9.7. Callcenter global settings

Allowdbcalls: allow calls from database in tagent

Allowmanualcalls: allow manual calls in tagent

Callmaxring: max ringtime when automatic calls in sec

Callnumbertype: 0=start with landline, and if fail, call mobile, 1=start with mobile, and if fail, call landline, 2 =call only landline, 3 = call only mobile

Maxcalltrycount: max number of calls to a client including recalls. the first recall increment the maximum allowed calls by one

Recallrestrictions: 0=try to recall with the same operator, but allow other if no recall, 1 = only with the same operator, 2=any operator can recall, 3 =disable recalls

4.10. TAgent

4.10.1. Login

Enter server settings and authentication info here to login.

The following values are required on login:

App Server: server ip address

Instance: Application and database instance (because a single server can hold several virtual server)

Data port: defaults to 2223

Database username: the same for all agents

Database password: the same for all agents

Username: agent username

Password: agent password

4.10.2. Manual Call

Simple VOIP client window where the operator are free to make calls to any number

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.

4.11. Virtual server settings

-Virtual server directory, database and exe name should be the same (virtserverX)

The service name will be the same as the exe file name

-Be sure to assign a different SIP port for every virtual server (Configurations->SipSettings->LocalPort)

-an ftp directory must be created for all virtual server under the “voice” direcory named after the service name

-be sure that the absolute ftp path is set properly (Configurations-> settings -> serverftpvoice defauls to C:\Inetpub\ftproot\)

-database and windws users can be the same must be set properly (windows user must have access to its ftp voice directory)

Suggested usernames has the form: callcenterX

-in the main server all virtual servers must be configured as traffic senders with proper authentication settings

-admin and monitor port numbers will be the default + X*100 where X represents the number after the service file name

5. FAQ

5.1. How to make a H323 call directly to a GW (without the gatekeeper)

set the signalport to 1720 in the gateway inifile

launch ohphone g729: 999simid#telnumber

5.2. The voice are cutting. How can I improve the voice quality?

Set up to active silencedetection (silencedetection=1)

Increase the jitter buffer (minjitter, maxjitter)

Use a low bandwidth codec (onlyg72x=1)

5.3. Using Netmeeting

MICROSOFT NETMEETING (H.323) tcp port 522, 389, 1503, 1720 and 1731 plus two secondary dynamically negotiated udp ports in the range 1024-65535 for the H.323 streaming protocol transmission of audio and video. For transmission of audio and video you only have to enable outgoing for these ports. Unfortunately to allow incoming audio and video you need to open up the entire 1024-65536 range as well as tcp 1503, 1720, 1731. Due to the complexity of the H.323 protocol which pre-dates the introduction of network address translation. Unless you have a firewall or proxy that specially supports the H.323 protocol at the application level, and thus supports the virtual opening of dynamic incoming udp ports, you have to open them all up. See Microsoft's Knowledge Base "How to Establish NetMeeting Connections Through a Firewall" Q158623.

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. Typical Cisco Config

! dial-peer voice 3630 pots incoming called-number 0036T direct-inward-dial port 2:D ! dial-peer voice 3631 voip destination-pattern 0036 voice-class codec 1 voice-class h323 1 session target ipv4:195.70.36.43

5.7. Server Recovery (in a separate app and db server configuration)

if the application server fails (the server directly connected to the internet, with your public ip

1. call your ISP support to change the internet cable to the backup server, and when it will be available connect to the "backupserver" with the remote desktop

"root" account

-on the backup server do the following:

2. enable the "vserver" service

3. launch the start batch file (from gk directory)

4. check the vservdebuglog and the tmanage

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 tmanage (you must have current calls)

6. you are ready

5.8. No incoming calls (no new calls in current call list in peak time)

1. Check the logs (filtered to „Server”)

2. If you cannot find the solution then.

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. If still in routing mode, then call the administrator.

5.10. SIP caller cannot call

1. Check disconnect reasons in cdr record for that caller

2. Check username/password

3. Check credit (if prepaid user)

4. Check caller techrefix, and the routing settings for that techprefix

5.11. SIP called cannot be called

1. Check disconnect reasons in cdr record for that called

3. Check if username exists

4. Check if usergroup matches the caller usergroup

5. Check user firewall settings

5.12. No call on Gateway

1. Check gateway absolutepriority, priority, enabled, temporarilydisabled and allowedpartners

2. Check if the gateway is online and sims are registered

3. Check sim packet settings (allowedpartners)

4. Check the routing on that simcards

5.13. No call on SIM

1. Check if simcard is active

2. Check allowedpartners, absoluteprioritypartners, absolutepriority

3. Check sim packet settings, including min/max speechlengths

5.14. No voice (caller and called cannot hear each-other)

1. Check routertp settings for the caller and the called

2. Check called firewall and nat settings

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. Check absolutepriority for gateway, sim and simpacket

2. Check routing patterns and timetable

5.18. Too low ASR

1. Check disconnect reasons for that direction

2. Check if gateway audio is ok

5.19. Too low ACL

1. Check disconnect reasons for that direction

5.20. SIM cards with low credit

1. Check if you have chargecards for that simpacket

2. Check charge fields in tb_sims (check if charging is enabled, lastchargetry date, etc)

5.21. GSM Gateway not working

1. Do the required settings for that box (pc config)

2. Check logs

3. Check if voipgsmgw and vclientsrv is enabled on the firewall

4. Check if vclientsrv service is running

5.22. GSM Gateway malfunctions

1. Cannot open sound device

Restart the pc. The usb sound devices will be remapped on pcrestart if allowed in inifile (check gw inifile and allow usbremap)

2. Lines in routing status

Enable voipgsmgw and vclientsrv on the gw pc firewall

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. TManage cannot connect to the server

1. Ping the server box. If ping is working, then check your username/password

2. Restart the server if you are sure that it is blocked

3. If still is not working, call the administrator immediately

5.25. Too slow TManage

1. Check your internet connection

2. Check server processor load. If too high, then check server logs, and if necessary, restart the server

3. If the problem persist, call the administrator

5.26. Server software problem (service unavailable)

1. Restore the last good configuration (Stop the service with stop.bat, copy all files from the lastconfig directory, near the current config and restart the service with start.bat )

5.27. Server OS, Database or Hardware problem (server unavailable)

1. Follow the failover plan.

2. Call the administrator

5.28. How to restart the server service

TManage->Administration->Server Console->Connect and send the „servicerst” command

5.29. How to restart the server box

-TManage->Administration->Server Console->Connect and send the „pcrst” command

-If you cannot connect with TManage, you can find a small program in the vclients directory named „serverrst” (usually at C:/Program Files/VCLIENTS/ serverrst.exe

-If this not work, then the server has serious problem. Follow the failovering plan and call the administrator

5.30. How to restart a GSM gateway

-TManage->Administration->Server Console->Connect and send the „client,XXX” command, where XXX is the gateway name or ip address. When connected to the gateway, send the „pcrestart” command

-if this does not work, then try to connect with remote desktop to the required gateway

-if the gateway is unreachable, then the pc or the internet is down.

5.31. How are the incoming calls from the gsm network handled?

Depending from Gateway Configuration inccalls value.

(0=drop,1=hold a little then drop,2=auto forward,3=forward to server as forwardnum,4=forward to number requested by dtmf)

Check the Gateway Configuration for more details.

5.32. Routing test calls to a dedicated gateway

set the calledpriority to the techprefix of the traffic sender

calledpriority: all calls with the specified techprefix will prioritize this gateway (but other techprefixes can go to this gateway also)

testprefix: only the specified techprefix can go to that gateway (but the specified testprefix can go to other gateways also)

so if you want a dedicated gateway for a techprefix, then you have to set the calledpriority and a testprefix too

example:

update tb_users set calledpriority = '987', testprefix = '987' where username = 'TESTGW'

then all calls with techprefix 987 will go to TESTGW with high priority

in case when the TESTGW channels are not available, the calls can be routed to other gateway

5.33. How to disable PIN request

The easy way

Set up the pincode entry under the [gateway] or [phoneX] section with the valid pincode. The gateway service will remove the pincodes automatically.

The hard way

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. What is the minimal global settings that must be correct?

On the “Configuration” form select “Basic” settings and check at least the following values:

LocalIP, LocalInternalIP, LocalDomain, currency, Routing, emergencydir, creditunit

5.35. How to add a new traffic sender?

In the “Users and Devices” form select Traffic Sender. Load the list and then hit the “New” button. Then you have the option to clone an already existing traffic sender. Set up the authorization correctly!

5.36. How to add a new sip enduser?

In the “Users and Devices” form select Endusers. Load the list and then hit the “New” button. Then you have the option to clone an already existing traffic sender. Set up the authorization correctly! Check the credit and prepaid/postpaid option!

5.37. How to add a new Tresto VOIP-GSM gateway to the server?

Tresto gateways will register automatically on the server. You may adjust its properties when the gateway is present. After that, you have to set up its sim channels correctly.

5.38. How to add new simcards (sim packet)?

Create a new packet in the “SIM Packets” form. Set up a meaningful name, specify if is postpaid or prepaid and walk through the other options (ownership, access list, recharging options, etc)

5.39. How to add a new simcard?

GSM channels will register automatically on the server. Then you have to set up its properties (to which packet it belongs, recharge options, owner, etc)

5.40. How to set up basic routing?

On the routing form add routing patterns to cover all possibilities (directions and times). Then you have to add your sim packets or other direction in desired priority order. Specify as many simpacket with the same priority as you can (so the server can do the routing after other conditions too. For example the quality.)

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. How can I change the volume?

In the Gateway Configuration check the followings: volumein, volumeout, vgr, vgt

5.44. How to register your Tresto Gateway to a H323 gatekeeper?

In the Gateway Configuration check the followings: gkip, gkpassword, gkdiscover, gkprefixesX

5.45. What ports are used in the system?

Standard SIP signaling port: 5060 (TCP and UDP)

Default H323 signaling port: 1720 (TCP)

H323 signaling port used by Tresto gateways: 1721 (TCP)

Rdesktop port: 8836 TCP

SQL Server port: 2223 TCP

“Voice Here” port: 44444 UDP

Tresto server admin port: 9885 TCP

Tresto server comm. port: 9886 TCP

Tresto server log port: 9889 TCP

Virtual SIM port: 9886 UDP

H323 additional port: configurable dynamic TCP

Media ports: configurable dynamic UDP

WebServer: 80 TCP

FTP: 21,22 TCP

5.46. My gateway restarts too often

Check the watchdog settings. For example the gateway will restart if no traffic are routed on it for 3 hour by default. Also check the maxwrongcalls and maxnotconnectedcalls settings

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.

SIMcard “Credit and Recharge” setting must be set accordingly.

5.49. The automatic credit recharge is not working

Check mincreditonrequest, creditrequestival for the packet.

SIMcard “Credit and Recharge” setting must be set accordingly.

Check the CreditRequestFail and CreditChargeFail (the server will try only 5 times. Reset to 0 if the problem is eliminated)

Check the other fields in the simcards regarding to credit charge and request. (fieldnames that contains the “credit” word. You have to check the “All Fields” checkbox on the SIM Channels form to see those fields)

Check if you have charge card for the required simpacket.

Check Logfiles (filter for “credit”)

5.50. How to monitor the credit automation?

Method 1: Launch TManage -> Sim Platform -> Credits and check the “credit history” queries

Method 2: Launch TManage -> Monitoring -> Logs and filter for “credit related”

To monitor the credit automation for a selected simcard, you can filter after the simid both in logs and in the Credit form.

5.51. Gateway and channels are inactive

Check if the gateway has internet connection.

5.52. How calls are processed

1. The SETUP or INVITE signal arrives from the traffic sender

2. If the caller is not allowed by the firewall, the call will be silently dropped

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. Searching for best route direction (available channels, priority order, round-robin, LCR, BRS, failovers, rerouting. etc)

10. Calculating the maximum speech length based on caller credit

11. Checking class 5 features and other endpoint settings (media routing, early-start, etc)

12. Initiating protocol conversion if needed

13. Routing the call to destination

14. Checking for call status, dropping if time exceed and other call monitoring tasks

15. Collecting CDR records at the end of the call

16. Calculating the prices of the call (realtime billing)

5.53. How to set up holiday billing

In the price form in “Time Definitions” select the “Holiday” entry

Set the priority higher in the Directions settings

5.54. How to treat specific weekends as weekdays

Set up a new entry in the holidays form and don’t set as holiday (uncheck the checkbox)

5.55. How the different currencies are handled?

Realtime price calculation in cdr records and the credit calculations for prepaid users are always done in the global currency (can be set up in configuration->currency)

However, you are able to set up your pricesettings in any currency. Automatic conversion is done when the given currency is not the same as the “global currency”. The conversion is done by predefined rates. You can set these rates in the “Currency convert” form in the TManage.

5.56. SimChange settings from the command line

format:

simchange1= 2004.03.05/13:00:00 - 2004.03.07/13:00:00 - 8936302403070132426 (from date - to date)

simchange2= 10:20:00 - 10:26:00 - 8936302403070132426 (every day from time to time)

simchange3= 2/10:20:00 - 7/10:26:00 - 8936302403070132426 (from Tuesday 10:00 to Sunday 10:00)

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. How to reenable blacklisted but good numbers

- In tmanage -> direct query, under the misc section check the “reenable blocked but good numbers” section

- delete old number from the helper table (section 0)

- run the query from section 1. this will load blacklisted but good number. The query execution may take 15 minutes

- list found numbers (section 2) and check it agains the blacklist (section 3)

- now you may delete blacklist entryies or set the “sure” level lower. First check the requested blacklist entry agains the query in section 4 (found numbers may be only a subset from the blacklist entry and in this case you may not delete or modify the blacklist. But if the asr and acl values are good for the blacklist entry, you may delete or modify it). Before you delete or modify the blacklist entry, check the comment (why was that number blocked). Number with comment “jukak” or “autdisabled monthly/weekely/daily” should be deleted or changed without problems.

5.58. How are different currencies handled?

In the global configuration, a global currency can be defined by the “currency” setting. For example ‘EUR’. Than there is the possibility to conver other currencies (used for pricelists, simpackets, users) to this “native” currency. For prices defined in “Price List” form, there is a possibilty to convert all imput prices in “native” currency by checking the “Convert to XXX” checkbox. In this manner for example you can import pricelist in other currency and that will be converted automatically in native currency when calculating CDR prices.

The conversion are done based on the settings in the “Currency Converter” form. You shoul update the conversion rates here as frequently as possible.
If you wish, you can leave the original value intact, so you can make your billing in other currencies than the native.

For every simpacket you can also define the currency, wich will affect the simcard credit calculation (automatic simcredit requests and recharges for prepaid simcards). Simcredits can be converted in the native currency format if the “convertsimcreditcurrency” configuration option is set to true. So you can have simcards in different countries, but all simcredits will be shown in the native currency.

For endusers and traffic senders you can also define different currency format in the Users and Devices form, Billing tab. The currency format defined here will be taken in consideration by the billing process.

5.59. How is VAT handled?

You should try to use prices without VAT included all ower in the system (for pricelist and for simcards)

VAT included pricelists can be easily converted to net values by checking the “Convert to NET value” checkbox in the “Price List”. You should enter the VAT percent in the “VAT Value” editbox for proper calculations.

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. How the check your ASR (or ACD, SL, CDRC) for the traffic sender “A” in the last week.

1. In the date-time drop-down list, select the “Last Week” field

2. In the “Select Direction” form set the “Source” (left side) “Type” to traffic sender, and select “A” in the “Name” drop-down list (or type “A” manually)

3. Launch the “Basic Statisitcs” form under Monitoring.

4. Clear the “Group by” option (select the first “-“ line)

5. Make sure the ASR checkbox is checked

6. Click on (Re)Load

7. Depending on current server config and current load this query may take some time (on a usual configuration this will take 2 second)

5.61. How to add endusers (basic settings)

1. Go to TManage -> Users and device form, and select enduser type

1. Select an already existing user wich has the same caracteristics as the required new endusers

2. Hit “New User” and than accept the the copy from existing option (cloning)

3. Check at least the following fields: username, password, parent id, authorizaton type (usually username/password), prepaid/postpaid, billed user

4. Check other settings

5. Save

5.62. Basic callcenter tasks

1. Setup your server as for a normal sofswitch (routes)

2. Create campaigns

3. Add callcenter operators

4. Assign operators to campaigns

5. Add or import clients

6. Assign clients to campaigns

7. Add presentation locations

8. Setup global callcenter configurations

9. Operators now are ready to start there TAgent application

10. Check statistics

11. Print invitations

12. Use checklist when you are on presentations

5.63. Abbreviations

ASR: average success ration (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. (for correct values, requires your billing module to be properly configured)

SUCC: successful call count (same as ASR but not in percent)

CCC: concurrent (simultaneous) call count

RTP: media channel protocol

SIP: The Session Initiation Protocol (SIP) is a signaling protocol used for establishing sessions in an IP network. A session could be a simple two-way telephone call or it could be a collaborative multi-media conference session.

H323: H.323 is an ITU (International Telecommunications Union) recommended standard, which provides a foundation for audio, video and data communications on non-guaranteed Quality of Service networks

RAS: used in H323. Used between the endpoint and its Gatekeeper in order to

Allow the Gatekeeper to manage the endpoint (Registration, Admission, and Status)

GK Registration: Endpoint will send an RRQ and expect to receive either an RCF or RRJ

H225: Call Signaling is used to establish calls between two H.323 entities

H245: generally transmitted on a separate TCP connections by most older endpoints

REGISTRAR: serverside component that allow SIP REGISTER requests

IEC: international escape code

NEC: national escape code

AC: area code

NUM: phone number

ANI / CLI – Automatic Number Identification or Caller Line Identification

IVR – Interactive Voice Recognition