• BYPASS THE PSTN
  • NO PER MINUTE CHARGES (IN OR OUT)
  • US NUMBERS OR INTERNATIONAL NUMBERS
  • INCLUDES TOLL FREE NUMBERS
  • ROUTE ALL CALLS DIRECTLY TO THE RECEIVING SIP SERVER
  • SIGN UP NOW AS A CHARTER MEMBER AND NO COST FOR LIFE

Introducing VoipConnect. A working voip interconnect service. Now accepting Charter Memberships. You can become a founding member of voipConnect.tel. FREE Access for Life

To enroll as a Charter Member, it is necessary that you register 100 or more live phone numbers.
When you enroll, your credit card will be charged for $10. After your identity has been verified, your $10 subscription fee will be refunded, and no annual fee will be assessed ever.

Do you remember ENUM?

Here is what NetworkWorld had to say about it in 2004:

Once ENUM is widely deployed, the idea is that you should be able to dial any "phone number," and have your SIP infrastructure look it up using DNS. If a SIP URI exists for that phone number, the call could be routed directly via IP. If no URI exists, the call would have to be routed over the normal PSTN.

The only problem was that ENUM NEVER got widely deployed. There were several attempts: freenum.org, e164.org. Both have vanished.

We think the problem might have been these two words: "using DNS"

As soon as you put your sip server address into DNS, you make it available for all the world to see, and you make it clear that it is a sip server.

NOW THERE IS VoipConnect.
A working voip interconnect service BASED ON AGI, NOT DNS.

  • Bypass the PSTN. No per call charges (IN AND OUT)

  • Get the direct dial number with a simple AGI query in less than a second.

  • All system users must pay a small fee (Helps defray some of our costs and establishes the true identity of the user)

  • Your IP addresses are never published worldwide, only made available to paid users on specific phone number request.

  • There are no per call or per minute charges (either to the call receiver or the originator). Running your calls through the PSTN typically costs the receiver $0.015 per minute and the originator $0.005 per minute.

How it Works

  1. Pay your subscriber fee ($10 per year if you list more than 5 real telephone numbers on your system, $50 if you don't)
  2. Send us a list of your numbers in E164 format (11 digits for US; start with County Code for others) Include your toll free numbers
  3. Set up a 6 digit PIN that must accompany any call attempt.
  4. In your sip.conf or pjsip.conf, allow calls from guests.
  5. Route guest calls to a context where you check that the caller knows your PIN number; reject all others.
  6. Calls that include your pin number: accept and process the call.

How to make an outbound call

  1. On every attempt to dial a phone number, do this first:
    • exten=>_1NXXNXXXXXX,1,AGI(agi://<voipConnect's IP>:<port>,voipConnect,${EXTEN},25,TtWw,<your account id>)
      exten=>_1NXXNXXXXXX,n,GotoIf($[ ${LEN(${dialString})} < 6 ]?telco)
    • exten=>_1NXXNXXXXXX,n,dial(${dialString})
  2. That's it. If you get a valid dial string back, you dial it. If you get back an empty one, then just go on into your regular PSTN dial out routine

If you are running an Asterisk installation with more then 2 phones, you are probably running more than one Asterisk box.

And you may have found Asterisk deficient in the following areas:

  1. Keeping track of where phones are registered and sending calls to the right box.

  2. Maintaining BLF light status across multiple Asterisk boxes.

  3. Keeping intruders from DOSing your boxes.

We've spent the last 15 years wrestling with and learning about Asterisk.

And we are willing to share that experience with you.

Give us a call and ask a question. If we can answer it, we will.

If it looks like it is going to require some work, we'll let you know.

AMI, AGI, and ARI

These 3 acronyms provide you with a multitude of ways to enhance and improve on Asterisk.

We have a lot of experience in developing these types of add-ons. If you need to work with these, give us a shout.