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Hi FRIENDS,

Looking for best Billing Solution for your VoIP Company? OR Looking to start your own calling card business under your own company brand name?

We offers the latest VOipswitch Versions with all its latest modules, one year free support,

ten hours remote training and full system and modules support (24/7) with installation and configuration in just 1300 USD.
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INTRODUCTIoN OF SOFTSWITCH:
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Softswitch is the main element of the platform, which merges the functionality of the following VOIP architecture’s elements.

H323 switch

H323 gatekeeper

SIP Proxy

SIP registrar

Each of the described elements can operate simultaneously with the others. Moreover, the clients, regardless of the protocol, or the way they transfer connections, can connect between one another.

This option allows connecting the networks, which because of the differences in implemented protocols or dialects inside the particular protocol, cannot directly transfer connection between one another.

Implementing APNAVOIP as a central traffic controller also introduces a number of additional management, supervision and network security facilitations.

The main characteristics of the softswitch include:

· Simultaneous and transparent support of SIP and H323 protocols (sip?h323 and h323?sip translator.

· Possibility of implementing various types of proxy (e.g. RTP-proxy or signaling proxy), possibility of choosing proxy for each prefix defined in dialing plan.

· Advanced routing and rating system

· Full internetworking with most commercially available switches, softswitches, session border controllers and VOIP gateways.

· VOIP equipment support.

· NAT support both for SIP and h323 equipment.

· Calling to sip devices behind NAT (without the necessity of configuring NAT).

· Calling among users registered to softswitch, support for dynamic IP addresses.

1· Authentication of VOIP equipment:
===================================

o. by IP address

o by ANI

o by h323id

o by the pair of login/password (according to the SIP standard)

0· Flexible routing

o· Individual, integrated billing system

o· Managing pre-paid and post-paid accounts

o· Setting up users in the VSConfig program

o· Managing users, blocking, setting limits

o· Generating the groups of users and managing lots

o· Creating and managing tariffs, the possibility of attributing a tariff to an individual user

o· Data stored in the MSSQL or MySQL database

o· Graphic management interface (presentation of the statistical data, billing information, managing clients’ accounts, generating PIN, managing the tariffs, dialing plan and others)

o· Graphic interface presenting the current traffic in the real time, number of the logged in clients, with the division into different types of services, presentation of logs and others

o· Web interface for clients – presentation of the connections history, possibility of exporting to the file, presentation of the current account status, possibility of making payments online and others

o· Easy to set up architecture

o· Automatic software re-start facilities in case of system failure

o· Scalability for new telecommunication services by enabling additional modules.

STANDARD APPLICATIONS

Central point of your VOIP network

2.Main benefits:
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Management of authorization rules of VoIP-gateways

Setting up call routing rules

Provisioning of compatibility for H323 and SIP- equipment of various vendors

Security and load planning of VoIP-traffic by using optional RTP-proxying

Access to the statistical data (ASR, PDD and others)

Transparent interface of the billing system

3.Network security:
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When using RTP-proxying SoftSwitch provides a single entry point for VoIP traffic.Both for clients and carriers there is only one IP address available.

Integration of equipment with support of different protocols

One of the most important features of RSF1000 is its ability to support widely accepted signaling IP-protocols - SIP and H323.

The system provides transparent converging of one protocol into another, thus allowing performing calls from one type of equipment to another.

4.SCALABILITY:
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Through launching subsequent modules, it is very convenient for a provider to extend the range of services offered. Available modules:

IVR for calling cards

Web/SMS/ANI callback (with IVR)

Reseller’s module

Online shop

CallShop

5.SPECIFICATIONS:
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Supported protocols

1 H.323 v.2 (H.245 v7, H225 v4) with/without FAST START

2 SIP (RFC 3261)

3 proxying of RTP/RTCP streams

4 Signalling proxy

5 Support of T38 (SIP, H323)

6 Transparent conversion of SIP to H323 and vice versa

Support of the Devices Behind the NAT

1 SIP-devices

2 H323-devices

6.Authentication:
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1 by IP address – SIP and H323

2 by H323ID – h323 terminals/gateways

3 by ANI (calling party number) – SIP and H323

4 by login and password- SIP equipment

5 by login and password – HearLink pc to phone/web to phone dialer (included in the package)

6 gatekeeper registration based on aliases

7.Intelligent routing:
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1 based on prefixes (the possibility of defining prefixes differentiating individual users)

2 based on accessibility of the VOIP gateway

3 based on priorities when choosing a gateway

4 depending on available voice codecs

5 depending on prefixes specified in the tariff of an individual client

Phone Numbers Translation

1 Deletion of the set number of digits from the called party number

2 Addition of the set number of digits to the called party number

3 Deletion of the set number of digits from the caller number

4 Addition of the set number of digits to the caller number

5 Virtual prefixes (for differentiation of the dialing plans)

8.Information for the Billing System:
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1 Real-time, built in billing system

2 Storage in SQL database (MSSQL or MYSQL)

3 pre-paid and post-paid accounts

4 Payments history

5 CDR – examining the logs of the calls carried out from the VSCConfig level, possibility of filtering data according to the set parameters, possibility of exporting data to the file (html, excel, txt, or csv type), presenting the CDR on the WWW pages available for clients

9.System Management and Control Features:
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1 Graphic User Interface for managing the overall functionality of the system

2 Visual presentation of current connections along with the information on their status

3 The number of statistical data presenting the information on the traffic intensity with its various parameters e.g. ASR, PDD. Possibility of limiting the number of data presented by using available filters e.g. only incoming traffic from the particular client, traffic directed to the particular gateway, or prefix etc.

4 Visual presentation of logged in clients and their current status, with the division into types of services e.g. gatekeeper users, SIP users, pc2phone, callback.

10.Operating Systems

1 Windows 2000, 2003, XP
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Please contact us if you are interested.

MSN;  sales(AT)apnavoip(DOT)com
      support(AT)apnavoip(DOT)com

OR Please Email us on 
   apnavoip(AT)hotmail(DOT)com
   SOLUTION PROVIDER  

BEST REGARDS.

Dear Friends:

Want to setup VOIP company, a business under your own brand name? We have complete solution to launche VOIP (Voice Over Internet Protocol) company. All support comes included.

Features: PC2Phone, Device2Phone, Calling Card, Callback,sms callback solution, Ani Callback solution, DID callback solution, Cli Callback, Pin Callback, Wholesale Termination, Online Billing, Unlimited Resellers Creating, online shop, invoice generator, paypal integrated online shop, pin recharge modules, H323 and SIP.

No ties to us, deal with the wholesalers you want and buy VoIP minutes from any one in the market without any expensive equipment.

Basically all kind of services that are available in the market. We always observe and follow the current trends in technology trying to react timely to changing customers demand by enhancing our system and adding new features.

The services can be divided on retail and wholesale:

[b]Retail : [/b]

•  [b]PC to phone and PC to PC – [/b]using our own sip based softphone with embedded voip tunnel client which allows for communication (both receiving and making calls) from behind voip blockades/firewalls, for example from UAE or other countries where voip ports are closed;

•  [b]device to phone[/b] i.e. making and receiving calls from SIP or h323 terminals (bidirectional NAT support), full compatibility with all popular brands of soho gateways and IP phones, support for 5 class services, voicemail, waiting message indicator feature, actual balance information on phone's display and more.

• [b] DIDs/Virtual phone numbers services – [/b]assigning phone numbers from various geographical regions to SIP or h323 devices or softphones. Customers can pick up a phone number directly from the web interface with immediate delivery. The system can work with leading DIDs provider thru their API or/and using own DIDs numbers stored in the local database.

•  [b]Class 5 services –[/b] call waiting, call forwarding, follow me – possibility to set forwarding rules conditioned on various events like busy, no answer, not logged with different forwarding destinations/phone numbers (possibility to set multiple numbers with different priorities), hold, hold with conversation

•  [b]Voicemail –[/b] checking voice messages by calling predefined voicemail number from softphone or SIP/h323 device or from PSTN thru the IVR system (calling cards) or using the callback service. Detailed information about left messages on the web interface. Possibility to record own welcome prompt. Email notification about new voice messages

•  [b]Callback services – [/b]triggered by SMS, Web or a missed call. Works with DIDs as an access numbers. Authentication by callerID (ANI) or PIN. Also supported DID callback with unique DID associated with each customer and his/her phone number. Manageable thru the web interface, possibility to add multiple ANIs per user. Multilingual Interactive Voice Response system (IVR). More on the callback and its variants here

•  [b]Calling cards –[/b] phone to phone services in which a customer dials an access number first, then he or she is asked about PIN (or authenticated by callerID/ANI) and then prompted for a destination phone number. The system comes with own IP based IVR system that supports several languages (possibility to add new), balance announcement, max duration for the call announcement, also it allows to recharge account by PIN and register actual ANI during the call and more. There are several scenarios which can be associated with particular access numbers. Moreover new scenarios can be added or the existing ones be modified by softswitch's administrator as they are programmed in xml format. More on the IP IVR can be found here

•  [b]CallShops –[/b] by callshop is understood a place like internet café or a shop where are cabins/booths with IP phones from which customers can make calls and then pay for them at a cash desk. The callshop is a windows based application which shows all the cabins and the calls that are taking place. When a customer has finished, the cashier can see the made calls with their details and costs and can charge the customer. The application also allows to print bills. The advantage of this solution is that the callshop application does not require any special hardware as the calls are not sent thru it, instead they go directly to SoftSwitch (thru the internet connection) and the callshop program exchanges with SoftSwitch only little data needed for billing purposes. Also as the clients can be used softphones installed on PCs or FXS gateways with analogue phones connected, not necessarily IP phones.

[b]Wholesale services:[/b]

• [b] Wholesale termination –[/b] bulk voice traffic termination, in this scenario Sofswitch acts as the traffic controller which on one side collects calls from clients gateways or other equipment, authenticate them, and then depending on the routing tables send the traffic further to termination gateways/gatekeepers. The whole traffic is billed and controlled in real time, the billing supports both prepaid and post-paid accounts types. Also the media (voice) flow can be set per destination and be sent in full proxy mode where all packets are going through the switch or in the modes where only signalization is sent through the system while the media goes directly between endpoints. The system allows to hide the source and the destination what is often important especially if you work as a broker buying and selling traffic from various parties. Additionally the system offers high flexibility in modifying calls setup, called numbers, enables you to manipulate prefixes, caller IDs and all other significant data before sending calls to the destination. There is also a web interface both for clients and terminators with CDRs (which can be exported to file) and other reports. More on the softswitch's functionality here.

The Rental Packages are designed a/c to concurrent calls, Please Select your Rental Package a/c to concurrent calls.

------------------------ [b]Rental Packages Detail[/b] ----------------------

[b]50 Concurent calls 250 USD /month [/b]
( Pentium 3, 1.6 GHz, 256 MB RAM, 2 Mbps Dedicated Bandwidth )

[b]120 Concurrent calls 350 USD /month[/b]
( Pentium 4, 2.6 Ghz 1 GB Ram, 4 Mbps Dedicated Bandwidth

[b]300 Concurrent Calls 450 USD /Month[/b]
(Pentium 4, 2.8 GHz, 2 GB RAM, 8 Mbps Dedicated Bandwidth )

[b]500 Concurrent Calls 600 USD /month[/b]
(Xeon Pentium 4, 2.4 GHz, 1 GB RAM, 13 Mbps Dedicated Bandwidth )

[b]1000 Concurrent Calls 950 USD /month [/b]
(Dual Xeon Pentium 4, 2.8 GHz, 2 GB RAM, 25 Mbps Dedicated Bandwidth)

------------------------ [b] rental Packages Details [/b] ------------------------------------

-------------------------[b]Detail of Pc2Phone Gallery [/b]--------------------

For SIPLINK Tunnel and without Tunnel Gallery:

[code]http://www.solution4voip.com/Pc2Phone_gallery.html[/code]
-------------------------Detail of Pc2Phone Gallery----------------------------

Contact us if you are interested.

Thank you,
[code]Solution4VOIP (.) com[/code]
VOIP Solution
Solution 4 VOICE OVER INTERNET PROTOCOL (VOIP)

Us Toll Free : +18888217060
Support Office : +92217019182

Sales / MSN : [code]Sales@Solution4VOIP.com [/code]    ,[code]sales@solution4voip.net[/code]
Support / MSN : [code]Support@Solution4VOIP.com [/code],[code]support@solution4voip.net[/code]
Billing : [code]Billing@Solution4VOIP.com[/code]
WebSite : [code]www.solution4voip.com[/code]           ,[code]www.solution4voip.net[/code],[code]www.solution4dids.com[/code]