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| Original Message |
| Author | Lui Chen |
| Topic | VOIP Equipment needed |
| Date Entered | 3/31/2006 10:30:02 PM |
| Message | Hi,
I m looking for PAP2 Link sys Phone adapter, which is not locked by vonage, I need 50 of them if some one have any idea please email me on the following address.
liuchen_33@yahoo.com
Thank you,
Have a nice day!!! |
| |
| Author | VoxMaster |
| Topic | Wifi Cell Phones |
| Date Entered | 3/31/2006 10:32:16 PM |
| Message | Wifi Cellphone, where to get the cheapest wifi cellular phone, do thye really work, anyone ever used one, how good are they, how long the battery last. please provide details as much as possible.
TY |
| |
| Author | ShakirKD |
| Topic | DID's for Pakistan |
| Date Entered | 4/1/2006 12:46:20 AM |
| Message | We sell Pakistan DID's in USA and Canada, complete solution for resellers, include instant ready to go package just $1500, include 1005 functional website, customer service, DID numbers for more than 20 major cities in Pakistan.
Contact : pakistandid@gmail.com
|
| |
| Author | EbayGuru |
| Topic | Wifi Cell Phones |
| Date Entered | 4/1/2006 12:52:47 AM |
| Message | Hi,
try Ebay, Zyxel is offering Wifi cellular phone , motorolla's wifi cellphone is due this year, so this thing is the future its going to be a cracker for cellphone service providers, you can read more about it on following links.
http://news.com.com/Wi-Fi+phones+make+a+splash/2100-7351_3-5296745.html
http://www.pcworld.com/news/article/0,aid,116334,00.asp
http://www.engadget.com/2004/11/01/bts-wifi-cellphone/
bye,
EG |
| |
| Author | tarek |
| Topic | Asterisk : Welcome to Asterisk |
| Date Entered | 4/1/2006 2:26:47 AM |
| Message | Starting a forum section for Asterisk only, need to know anyhting about asterisk , how to's or information about buying and selling custom asterisk please feel free to message here.
Thank you,Tarek |
| |
| Author | Aleem Dar |
| Topic | Portaone |
| Date Entered | 4/1/2006 2:30:19 AM |
| Message | Hi,
everyone, we just started selling services using Portaone soft switch, as a beginers in this feild like to know about portaone more any place where we can get more information about this soft switch SIP solution.
please email links or related info to : aleem_d@yahoo.com |
| |
| Author | tarek |
| Topic | Call center solutions in Pakistan |
| Date Entered | 4/1/2006 2:34:36 AM |
| Message | Pakistan Call Center Setup
I see the post above have different topics related to Call Center, I would like to assist you with the right information and if interested can help you setting up one call center I am such a spammer idiot with Government involvement and support and this might also answer the above RISK and LACK of Knowledge
I have Government approved 1200 seats Call Center establishment using VOIP Technology
If you are interested to setup from 5-100 seats then please find below the information
New Call Center:
1-Government approved 100% Legal
2-Outsource your Own BPO Service or start to market one
3-If you are from Pakistan and can arrange your own location then I have no problem with that else for outsource people , I can help to arrange that
4-THERE IS NO INCOME TAX FOR 5 YEARS ON YOUR CALL CENTER BUSINESS in PAKISTAN, WORK WITH US
5-PTCL/Telecomunication approved and audited agreement to setup Call Center on VOIP and can setup with in short time
Fixed Cost:
6-Average Cost per seat: $500-$1000 one Time Setup
Monthly Cost:
7-Unlimited Incoming from US: $3-$5 Per seat
8-Outgoing US $0.02 per minutes
9-Internet Speed/Bandwidth: Monthly for 10 Seats : $700-$800 Per Month
(We will assist in finding and train your HR)
Existing BPO Looking Call Centers to outsource their work:
1-15-30 Days RISK FREE Trail, means you outsource your work and get the seat, we will bear all of your cost and wont bill you till you satisfied
Now read all above and compare any country you want!
Note:
In all over the world, Asia is one of the cheap call center location and only a few countries, Professionals speaks English, remove China, Japan and many others and PAKISTAN is one of the good source to outsource, and for security concern you can use hosted Call Center and store all of your Data and information in Outsourced or your own managed US Data Center....what to loose...contact or at least I tried to cover most of the concern for right direction....
Only Serious People, I would give available approved seats to those who are serious to setup Call Center and outsource their own Business Process and control their own!
email:tareksol@yahoo.com |
| |
| Author | netman |
| Topic | DID's for Pakistan |
| Date Entered | 4/1/2006 2:51:45 AM |
| Message | I have available legal Pakistan DID's from the cities below. Each DID number cost $25.00 with unlimited calling to any destination number you point numbers to. Numbers are legal and permanent as long as paid for and cannot be cancelled. The numbers are for residential and regular business use only. These numbers are NOT for use by calling card companies or to call Pakistan city to city.
Terms are prepay per DID. You can test by prepaying for 1 DID.
Pak Cities we have numbers for -
City Prefix
1 Bahawalpur 62
2 Dinga 53
3 Faisalabad 41
4 Gujranwala 55
5 Gujrat 53
6 Kasur 49
7 Kharian 53
8 Lalamusa 53
9 Multan 61
10 Peshawar 91
11 Sargodha 48
12 Sheikhupura 56
13 Sialkot 52
14 Sukkur 71
15 Wazirabad 55
16 Karachi 21
17 Lahore 42
Contact: netman@carolina.rr.com |
| |
| Author | P1_passion |
| Topic | About Portaone info |
| Date Entered | 4/1/2006 2:52:39 AM |
| Message | Hey, Aleem your email doesn't work, i m receiving bounce emails. please correct.
TC, P1P |
| |
| Author | P1_passion |
| Topic | About Portaone info |
| Date Entered | 4/1/2006 2:52:39 AM |
| Message | Hey, Aleem your email doesn't work, i m receiving bounce emails. please correct.
TC, P1P |
| |
| Author | InfoMiami |
| Topic | Free wifi high speed internet, for everyone |
| Date Entered | 4/1/2006 3:21:52 PM |
| Message | hello,
Now for the first time, at miami south beach you can browse internet absolutely free. Now at south beach wifi internet are now transmitted through light poles for all pedesterians to browse internet 100% free.
Good luck and welcome to all miami tourist.
InfoMiami |
| |
| Author | voipbug |
| Topic | What is VOIP |
| Date Entered | 4/3/2006 2:20:02 AM |
| Message | What is VoIP/Internet Voice?
VoIP allows you to make telephone calls using a computer network, over a data network like the Internet. VoIP converts the voice signal from your telephone into a digital signal that travels over the internet then converts it back at the other end so you can speak to anyone with a regular phone number. When placing a VoIP call using a phone with an adapter, you'll hear a dial tone and dial just as you always have. VoIP may also allow you to make a call directly from a computer using a conventional telephone or a microphone.
How Can I Place a VoIP Call?
Depending on the service, one way to place a VoIP call is to pick up your phone and dial the number, using an adaptor that connects to your existing high-speed Internet connection. The call goes through your local telephone company to a VoIP provider. The phone call goes over the Internet to the called party's local telephone company for the completion of the call. Another way is to utilize a microphone headset plugged into your computer. The number is placed using the keyboard and is routed through your cable modem.
What Kind of Equipment Do I Need?
A broadband (high speed Internet) connection is required. This can be through a cable modem, or high speed services such as DSL or a local area network. You can hook up an inexpensive microphone to your computer and send your voice through a cable modem or connect a phone directly to a telephone adaptor.
Is there a difference between making a Local Call and a Long Distance Call?
Some VoIP providers offer their services for free, normally only for calls to other subscribers to the service. Your VoIP provider may permit you to select an area code different from the area in which you live. This means you may not incur long distance charges if you call a number in your area code regardless of geography. It also means that people who call you may incur long distance charges depending on their area code and service.
Some VoIP providers charge for a long distance call to a number outside your calling area, similar to existing, traditional wireline telephone service. Other VoIP providers permit you to call anywhere at a flat rate for a fixed number of minutes.
If I have VoIP service, who can I call?
Depending upon your service, you might be limited only to other subscribers to the service, or you may be able to call any phone number, anywhere in the world. The call can be made to a local number, a mobile phone, to a long distance number, or an international number. You may even utilize the service to speak with more than one person at a time. The person you are calling does not need any special equipment, just a phone.
What Are Some Advantages of Internet Voice?
Because VoIP is digital, it may offer features and services that are not available with a traditional phone. If you have a broadband internet connection, you need not maintain and pay the additional cost for a line just to make telephone calls.
With many VoIP plans you can talk for as long as you want with any person in the world (the requirement is that the other person has an Internet connection). You can also talk with many people at the same time without any additional cost.
What Are Some disadvantages of Internet Voice?
If you're considering replacing your traditional telephone service with VoIP, there are some possible differences:
Some VoIP services don't work during power outages and the service provider may not offer backup power.
Not all VoIP services connect directly to emergency services through 9-1-1.
VoIP providers may or may not offer directory assistance/white page listings.
Can I use my Computer While I talk on the Phone?
Yes
Can I Take My Phone Adapter with me When I Travel?
You may be able to use your VoIP service wherever you travel as long as you have a high speed Internet connection available. In that case it would work the same as from your home or business.
Does my Computer Have to be Turned on?
Not if you are making calls with a phone and adaptor or special VoIP phone, but your broadband Internet connection needs to be active. You can also use your computer while talking on the phone.
How Do I Know If I have a VoIP phone Call?
It will ring like any other call.
Does the FCC Regulate VoIP?
The Federal Communications Commission (FCC) has worked to create an environment promoting competition and innovation to benefit consumers. Historically, the FCC has not regulated the Internet or the services provided over it. On February 12, 2004, the FCC found that an entirely Internet-based VoIP service was an unregulated information service. On the same day, the FCC began a broader proceeding to examine what its' role should be in this new environment of increased consumer choice and what it can best do to meet its role of safeguarding the public interest.
Aspects of these considerations may change with new developments in internet technology. You should always check with the VoIP service provider you choose to confirm any advantages and limitations to their service. |
| |
| Author | voipbug |
| Topic | How VOIP works |
| Date Entered | 4/3/2006 2:21:35 AM |
| Message | If you've never heard of VoIP, get ready to change the way you think about long-distance phone calls. VoIP, or Voice over Internet Protocol, is a method for taking analog audio signals, like the kind you hear when you talk on the phone, and turning them into digital data that can be transmitted over the Internet.
How is this useful? VoIP can turn a standard Internet connection into a way to place free phone calls. The practical upshot of this is that by using some of the free VoIP software that is available to make Internet phone calls, you are bypassing the phone company (and its charges) entirely.
This person is using a computer to talk to a friend in another state.
VoIP is a revolutionary technology that has the potential to completely rework the world's phone systems. VoIP providers like Vonage have already been around for a little while and are growing steadily. Major carriers like AT&T are already setting up VoIP calling plans in several markets around the United States, and the FCC is looking seriously at the potential ramifications of VoIP service.
Above all else, VoIP is basically a clever "reinvention of the wheel." In this article, we'll explore the principles behind VoIP, its applications and the potential of this emerging technology, which will more than likely one day replace the traditional phone system entirely. |
| |
| Author | VOIP_Helper |
| Topic | How to Setup your VOIP Phone Adapter |
| Date Entered | 4/3/2006 2:23:02 AM |
| Message | Step 1: Network Installation Instructions A. Please Check to Make Sure That You Have the Following Package Contents:
1. Voip Adapter
2. Ethernet Cable
3. A 5 Volt Power Adapter
You Will Also Need:
1. One or Two Analog Touch Tone Telephones (or Fax Machine) Connect the Networked Devices with Ethernet Cables:
1. Power down all the devices you will be networking: Cable/DSL Modem, PC, Voip Adapter. Connect an Ethernet cable from the Cable or DSL Modem to the ETHERNET port of the VOIP Adapter. Note: If you are connecting the VOIP Adapter to a local area or home network, connect an Ethernet cable from the WAN port of the VOIP Adapter to the LAN port on your network switch/router. Do not connect the VOIP ADAPTER WAN Port to an Ethernet hub.
Step 2: Telephone / Fax Installation and Power-UP From the Rear of the VOIP Adapter: A. Insert a standard RJ-11 telephone cable into the VOIP Adapter PHONE 1 port. B. Connect the other end of the cable to an analog telephone or fax machine. C. Insert a standard RJ-11 telephone cable into the VOIP Adapter PHONE 2 port . D. Connect the other end of the cable to an analog telephone or fax machine. Note: Do not connect RJ-11 telephone cable from the VOIP Adapter to the wall jack to prevent any chance of connection to the circuit switched telco network. You may now insert the plug end of the power adapter into a live power outlet which will power up the VOIP Adapter. At this time, you may now power on the Cable / DSL modem.
http://www.callpakistandirect.com/devicesetup.aspx
For more information please call 1-888-250-6484. |
| |
| Author | Linksys Seller |
| Topic | Linksys PAP2 dual port phone adapter. |
| Date Entered | 4/3/2006 2:28:47 AM |
| Message | Great Product. No problems with it with MyFone. Was making calls 5 minutes after it came out of the box.
Dial-tones are US, which takes a little getting used to.
Documentation on features is extremely limited, despite having many functions. Interface does not make it easy to find out how to use these features either. But basic functionality is easy to use.
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| |
| Author | VOIPHelper |
| Topic | What is a VoIP codec? |
| Date Entered | 4/3/2006 2:30:45 AM |
| Message | What is a VoIP codec?
A codec (Coder/Decoder) converts analog signals to a digital bitstream, and another identical codec at the far end of the communication converts the digital bitstream back into an analog signal.
In the VoIP world, codec's are used to encode voice for transmission across IP networks.
Codec's for VoIP use are also referred to as vocoders, for "voice encoders".
Codecs generally provide a compression capability to save network bandwidth. Some codecs also support silence suppression, where silence is not encoded or transmitted. |
| |
| Author | LinkSys |
| Topic | Link Sys Special for Small bussiness |
| Date Entered | 4/7/2006 3:20:39 AM |
| Message | Linksys SPA-9000 SIP PBX Appliance w/16 Extensions
The SPA9000 marries the rich feature set of high-end PBX telephone systems with the convenience and cost advantages of Voice over IP. It has common voice system features such as an auto-attendant, shared line appearances, three way call conferencing, intercom, music on hold, call-forwarding and much more. The SPA9000 opens up access to the benefits of VoIP, including low cost long distance service, telephone number portability, and one network for both voice and data.
Linksys SPA-9000 SIP PBX Appliance w/16 Extensions
Please Note: VOIPSupply is currently accepting pre-orders for the SPA-9000. Call 1-800-398-VOIP for shipping status.
Features:
IP PBX system with high-end features comparable to traditional large business voice services
Sixteen (16) SIP compatible IP Phones per SPA9000 system
Powerful self-configuration capabilities enabled with Linksys IP Phones
Works with most Internet Telephone Service Providers
The SPA9000 does not have onboard voicemail capabilities. Voicemail is provided by the Internet Telephony Service Provider, though voicemail may also be provisioned via a specially modified version of the Asterisk Open-Source PBX.
The SPA-9000 is shipped unlocked and may be used with most Internet Telephony Service Provider offer SIP credentials or ITSP's who will provision a user-provided device. Note that service providers that provision devices with a "Profile Rule" may partially or completely restrict your ability to configure certain settings in the device. Please check with your VoIP service provider for details.
The SPA9000 is so easy to configure that a fully working system can be set up in minutes. New telephones are automatically detected and registered when they are connected to the SPA9000. The SPA9000 has an integrated web server that allow features to be configured using a web browser. The web server has multiple levels of password protected access to user and service level features. Service level settings may be locked by the Internet Telephone Service Provider to ensure they are not inadvertently corrupted. The Internet Telephone Service Provider also can remotely update the software and settings through a secure encrypted connection.
With its integrated router, the SPA9000 can be either connected directly to the internet connection or to another router on your network. The SPA9000 has separate WAN and LAN Ethernet ports. The WAN connection can be connect through DHCP or a fixed IP address. The LAN port can assign IP addresses to IP Phones and computers using NAT and DHCP.
While the SPA9000 will work with any SIP compatible IP Phone, it is the ideal host for Linksys IP Phones, such as the SPA901, SPA921, SPA922, SPA941, and SPA942. Powerful configuration capabilities enable the SPA9000 to support a greater set of advanced features with these IP Phones, such as shared line appearances, hunt groups, call transfer, call parking lot, and group paging. With its two FXS ports, the SPA9000 can support traditional analog devices such as telephones, answering machines, FAX machines, and media adapters.
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| |
| Author | Vault1 |
| Topic | Dedicated Hosting Solution |
| Date Entered | 4/8/2006 12:26:53 AM |
| Message | Hi,
this is to let you know now vault 1, is offering for the first time 100% disaster recovery facility, loaded with specialized equipment in South Florida offering special suites and seating for your employees to take care of bussiness with 0 down time.
Vault1 |
| |
| Author | Call Pakistan Direct |
| Topic | Unlimited incoming calls from Pakistan to USA |
| Date Entered | 4/15/2006 3:44:38 PM |
| Message | Offering unlimited calls from Pakistan to USA, Canada, Middle east, Europe and many other countries, just sign up for a Calling Plan at http://www.callpakistandirect.com and receive unlimited incoming calls from Pakistan .
Please check this website http://www.callpakistandirect.com
contact@callpakistandirect.com
1-888-250-6484. |
| |
| Author | Sub |
| Topic | Cancellation of dids. |
| Date Entered | 4/27/2006 7:31:14 PM |
| Message | Hi netman,
This message is for you. What will happen, if I want to cancel the did. Another thing is that for $25 per month, the number will ring on a phone in Canada or US or computer. Are there any other charges? Do you have dids from Canada to ring on a phone in Pakistan. What are the charges? |
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| Author | |
| Topic | VOIP Traffic |
| Date Entered | 5/18/2006 9:31:06 AM |
| Message | |
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| Author | anony |
| Topic | VOIP Traffic from UAE |
| Date Entered | 5/18/2006 9:31:50 AM |
| Message | All VOIP traffic from UAE has been blocked by the national career.
Etisalat the national career is to expensive. and no competition
Please suggest ways to go thru. thanks |
| |
| Author | Suresol |
| Topic | VOIP Traffic from UAE |
| Date Entered | 5/19/2006 7:25:58 PM |
| Message | Hi,
http://www.callpakistan.com offers Device based solutions for such VOIP restrictions usually voip providers block certain ports to stop VOIP Traffic, Call Pakistan Direct, one of the VOIP providers from Pakistan to anywhere in the world. Call Pakistan Direct provides unlimited calls from Pakistan to USA, Canada, UAE, UK , South Africa and many other countries.
Visit
http://www.callpakistandirect.com
1-888-250-6484
|
| |
| Author | anony |
| Topic | VOIP Traffic from UAE |
| Date Entered | 5/20/2006 2:53:25 PM |
| Message | UAE has blocked gizmo, delta3, mywebcalls, vbuzzer, yahoo messenger and MSN as of today
r u sure your service is still working |
| |
| Author | Sure Sol |
| Topic | VOIP Traffic from UAE |
| Date Entered | 5/21/2006 4:18:15 PM |
| Message | Hi,
We can have it work for you but still need you to buy a Voip Phone adapter for it. please contact http://www.callpakistandirect.com , email us at support@callpakistandirect.com and refer the you are contacting us after the discussion at Voxzine the VOIP Forum.
Thank You.
AH |
| |
| Author | batman |
| Topic | VOIP Traffic from UAE |
| Date Entered | 6/21/2006 9:19:36 PM |
| Message | Hi
Etisalat has blocked access to websites of known VOIP providers (Mediaring, Harbibi, Vonage, Net2Call, hotfoon, etc.) however you can still enjoy making cheap international calls by using adaptors and IP phones that is H323 and SIP protocol compliant.
If anyone is interested in the device please drop me an email at batman13@gmail.com
Best regards |
| |
| Author | parvez |
| Topic | solution for resellers |
| Date Entered | 7/24/2006 8:13:00 PM |
| Message | will you please elaborate and send me an e-mail in detail.
thank you.
texcomm1@sbcglobal.net |
| |
| Author | CallPak |
| Topic | VOIP Traffic from UAE |
| Date Entered | 9/14/2006 3:17:16 AM |
| Message | Blocking port and sites does cause problem, still contact us at support@callpakistandirect.com and we can provide easier way to receive calls from Pakistan,USA,Canada and other country. |
| |
| Author | Route Searcher |
| Topic | VOIP Route for India. |
| Date Entered | 9/14/2006 3:21:40 AM |
| Message | Hi,
We are looking for a Route for India from USA, Canada and UK over SIP Protocol. Anyone with white route please provide us a contact. |
| |
| Author | Newbug |
| Topic | Welcome to Asterisk |
| Date Entered | 9/16/2006 6:59:54 PM |
| Message | Hi,
Tarek, we are setting up a four seater call center using Asterisk in Karachi, how can we contact you for, technical help we need to have things setup, what are your charges. Please email us at : imran@callcentersworld.biz
Thank you |
| |
| Author | Qadir |
| Topic | Islamabad DID's |
| Date Entered | 12/3/2006 11:05:07 PM |
| Message | Hi,
We are looking for Islamabad, Hyderabad, and Abbotabad DID's. Pakistan DID's are avaialable but we need Islamabad DIDs. If anayone has any information we are looking for 50 DIDs.
Contact Us,
apnakeisha@gmail.com.
Thank you,
Qadir. |
| |
| Author | M.Saimullah |
| Topic | Call Pak Net |
| Date Entered | 12/20/2006 2:20:38 AM |
| Message | I used your calling service its good, it helped me in my Spouse's immigration process too, thank you for your help and providing Call Detail report on each of my request.
CallPak.Net has one way voice problem at late night time. I think they are selling more and have less capacity.
I m using CallPak.net from Dubai, there service is good and dirt cheap its just $24.00 it is saving me a lot of money, if any one finds a cheaper deal post a message.
|
| |
| Author | Ismail Lakhani |
| Topic | Call Pak Net's Service |
| Date Entered | 12/21/2006 1:56:43 AM |
| Message | CallPak.net's service is excellent, Unlimited calling from Pakistan just for $15.99 with average customer service is not bad.
CallPak.net doesn't have 100% Connection some time there are connecting problem and it also goes busy.
Please remove this problem besides this I m one satisfied customer.
|
| |
| Author | kashif |
| Topic | Call Pak Net's Service |
| Date Entered | 12/22/2006 6:28:30 AM |
| Message | I am using Call Pakistan Direct since I saw them on google they are providing good services, espcially Lahore,Islamabad numbers I recommend there service and they have good customer suport too. |
| |
| Author | kami |
| Topic | voip switch and its all latest modules in 1300usd |
| Date Entered | 7/18/2007 6:57:33 AM |
| Message | Hi FRIENDS,
Looking for best Billing Solution for your VoIP Company??
Apnavoip is a platform that allows to implement various types of Voice Over the Internet Protocol (VOIP) services,
with retaining shared, uniform management interface. The feature that distinguishes this platform is the implementation
of an integrated, embedded billing system that cooperates wfith SQL – MS SQL or MySQL databases’ servers.
This solution results in the simplicity of preparing the system, by the operator, to be fully functional for providing
services and also for administrating it in the future.
VoipSwitch’s software consists of the following modules:
1.)VoipSwitch manager – the main part of the application. It allows to monitor the whole incoming traffic.
Apart from the current connection status and the type of logged in clients, it also presents a number of additional
information on the processes that are taking place.
2.)VoipSwitch Config management interface. It is a tool for configuring the whole system. It has a number of features
that allow to manage clients’ billing and analyze traffic information, based on the statistical data.
Web CDR the module that allows clients to check their current account status and the history of the calls made.
There is a possibility of exporting data to the file from this level.
3.)PC to Phone Client softphone based on the g723.1 codec.
4.)Webphone softphone that can be initiated directly from the website.5.)
5.)Callback Client client’s software that allows to initiate calls between two telephones.
6.)Webcallback client’s callback version that is initiated directly from the website
7.)SMS callback module that cooperates with the SMS operators. It also allows to create access number for SMS callback
service using the mobile phone
8.)IVR module module responsible for playing back voice messages utilized by various services e.g. 2 stage dialing,
account balance or IP PBX
9.)Online Shop web based system that allows resellers to create users accounts, manage them, manage the tariffs,
and analyze the information on the traffic that is taking place etc.
10.)Reseller’s module web based system that allows resellers to create users accounts, manage them, manage the tariffs,
and analyze the information on the traffic that is taking place etc. .
11.)CallShop windows based application that is to be installed in internet callshops, it gives real time information on
connection status, the telephone number, time of the connection, its cost etc
LATEST VOIPSWITCH SUPPORTS THE FOLLOWING MODULES:
new Webportal , support 15/30/60 sec for CDR display format
OnlineShop for this is probably paypal scam, don't listen to me , online pay
Tunnel Server/Tunnel pc2phone Dialer/Tunnel Proxy **** for UAE,dubai good VPN dialer
Calling card Module , pin recharge
Callshop Module
DID Callback / SMS Callback / ANI Callback / WEB Callback / SMS recharge
Internal Billiing / prepaid / web reseller / web config
WEB CDR
IVR voice play UserBalance , IVR voice build , IVR recharge
SIP and H323 Switch
SIP to H323
Codec G711/G723/G729/GSM610
NAT private network pass through
Endpoint as caller or called(call to private network)
Virtual Private Voice Network
include h323 , sip softphone / webphone
Operating Systems
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.
|
| |
| Author | kami |
| Topic | VOIPSWITCH + FULL SUPPORT 24/7 + ALL LATEST MODULE |
| Date Entered | 7/18/2007 12:33:39 PM |
| Message | 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.
------------------------------------------
· 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.
|
| |
| Author | kami |
| Topic | VOIPSWITCH + FULL SUPPORT 24/7 + ALL LATEST MODULE |
| Date Entered | 7/19/2007 3:26:08 AM |
| Message | 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.
------------------------------------------
· 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.
|
| |
| Author | Reviewer |
| Topic | VOIP or NO VOIP |
| Date Entered | 10/8/2007 4:57:42 PM |
| Message | With VOIP as a cheap medium to communication companies are emerging fast and adopting VOIP to be the solution for there evreyday communication needs. |
| |
|
|
|
