Hi FRIENDS,

Looking for best Billing Solution for your VoIP Company??

We offers the latest VOip switch Version (2.0.0.879) with all its latest modules with one year support free,

ten hours remote training and full system and modules support (24/7) with installation + configuration in just 1300 USD.

We offers the best.

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:

---

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:

---

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:

---

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:

---

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:

---

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:

---

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:

---

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:

---

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

---

Contact us if you are interested.


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

OR for more information please visit our website.
  www.apnavoip.com
  SOLUTION PROVIDER   

BEST REGARDS.

Have any company use Ser to provide Prepaid internet phone service?
my company is looking for Prepaid internet phone solution.
Thank you, phanvanduc@gmail.com

R there any detailed docs on Ser+asterisk integration .

If you're trying to implement SER on your VoIP network, feel free to contact us voipinfo@idv.net or through our website http://www.idv.net
We have extensive background in implementing SER with Cisco voice gateways, Asterisk, ATAs, etc.
We can also help with custom development.

we are looking for a linux based programmer for improving our SIP proxy Project
sip professional programmer please contact me at grkashani@yahoo.com

Hi,

Can anyone expand on this comment (SER supports SIP connections with more features and more scalability than Asterisk. Normally, SER would be used in conjunction with Asterisk when a SIP phone needed to connect to the PSTN) so that a first-time Asterisk novice can understand the logic of how SER and Asterisk work together to connect to the PSTN? What functions/processes is Asterisk responsible for and what functions/processes is SER responsible for? What handoffs are going on? Thank You, dave_rep@yahoo.com

Hi,
We are investigating into integrating the two. Do you know if Asterisk supports VXML? If not, do you have any pointers or info on how the integration can be done?

Thanks.

Anyone implement SER & Asterisk@Home as a single box solution to allow home gateway NAT'd phones to easily communicate with Asterisk?

I've noticed that the iptel site is hardly ever functioning.. Does anyone know of a SER mailing list outside of iptel? (mail.iptel.org is always down..)

Thanks.

Roel

need help.
howto
config example
AX2400 (24fxs ports)
documentation,links... any