just like Perkild co. be very careful But ?? Nothing at all, and all money lost with L Perkild he offer different kinds of VOIp also offer 24x7 technical supports. Encrypted VoIP gateway
complete pack. which include: VM software, SM, dialer....great lost in my company thanks to L Perkild No service at all

Hi, I tried so many time...and it's not working...><
i did port forwarding.....and still its not working....i dont know what to do anymore...
please help......thank you

Hmmmm, spammer that uses spamblocks?

Spiders, please feel free to add sales@apnavoip.com, support@apnavoip.com and service4voip@hotmail.com to your lists. That's sales@apnavoip.com , support@apnavoip.com and service4voip@hotmail.com.

Thanks,


for adding sales@apnavoip.com, support@apnavoip.com and service4voip@hotmail.com to your lists.

Any company that cannot understand simple rules, and continues to spam multiple websites should certainly not be trusted with your business!

People and companies like you ruin the internet.

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 free support,

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.

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· 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.

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· Advanced routing and rating system

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

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· VOIP equipment support.

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· NAT support both for SIP and h323 equipment.

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· 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:

---

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:

---

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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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.

I have recently purchased a linksys voip wip330 phone with the latest firmware and i just wanted to clear up what i have read on the site re changing the firmware to get the use of skype and the IE as i need the IE to access hotspots where i live acording to what i have read on this site i can flash the firmware back to 1.00.00 if i understand it correctly. if that is true then i can then flash it to upgrade it to the 1.01.00 and have the use of skype msn and IE for getting on to hot spots in my area.
I would be very thankfull if some one could confirm this for me. And if they have any other advice for me in doing this would be great. never done this before
thank-you much

Please service4voip@hotmail.com don't spam this site. Thanks.

Dear friends.

we are offering DIDs and toll free numbers for 40 countries including Pakistan from around the globe.

No Setup Fee. No Hidden Charges. Unlimited incoming Calls. Map to anywhere and no minimum purchases.

---

countries Available for dids :

---

THE AMERICAS:
=========> ARGENTINA.BRAZIL.CANADA.CHILE.GUATEMALA.MEXICO.UNITED STATES

EMEA :
======> AUSTRIA.BELGIUM.BULGARIA.CYPRUS.CZECH REPUBLIC.DENMARK.ESTONIA.FINLAND
FRANCE.GERMANY.HUNGARY.IRELAND.ISRAEL.ITALY.LATVIA.LITHUANIA.LUXEMBOURG
NETHERLANDS.NORWAY.POLAND.PORTUGAL.ROMANIA.SPAIN.SWEDEN.SWITZERLAND.UNITED KINGDOM

ASIA PACIFIC:
=========> AUSTRALIA.JAPAN.NEW ZEALAND.PAKISTAN.SLOVAKIA.SLOVENIA


we are also offering toll free numbers for the following countries.

---

COUNTRIES AVAILABLE:

---

AUSTRALIA, AUSTRIA, CYPRUS, LUXEMBOURG, NETHERLANDS, SLOVAKIA, SWEDEN, UK.

---

only for interested parties please contact:

MSN: service4voip (@) hotmail (.)com

contact: +923218940265

I am trying to get call forwarding working the equivilant to 72# on a PSTN line .. Basically I want anyone who calls my sip number to be forwarded to another number. which I want to do using my sip phone by pressing *21*1NxxNxxxxxx and then all calls coming in go to the number I provided.. here is my config in extensions but it does not work can anyone help..
; create call forward
exten => _*21*X.,1,GotoIf($${EXTEN:-1} = #?2:3)
exten => _*21*X.,2,DBput(CFIM/+1${CALLERIDNUM}=${EXTEN:4})
exten => _*21*X.,3,Hangup

; delete call forward
exten => **21,1,DBdel(CFIM/+1${CALLERIDNUM})
exten => **21,2,Hangup

Any help would be appreciated..

• sigh*