VoipJet Inc. delivers guaranteed quality Voice-over-IP termination.
Instant online test account setup, with 25 cents free to test.

You can fund your account with PayPal, and payments are instantly credited to your account. No minimums.
Activation is right away, and you'll be calling in no time.
All deposits less then $500 are subject to a 3% + 30 cents processing and handling fee (it's the fees that they are charged by paypal)

Setting up is easy. On their Accounts screen they 'll have personalized settings for you (with your username and passwords right in the page with your asterisk settings)

But here they are just for those who need a quick reference:
  • Peer1 East Coast Server (recommended):
  • NAC East Coast Server:
  • NAC East Coast Server II:
  • InterNap West Coast Server:
(Choose depending on your location)

For Asterisk@Home AMP see this screenshot. For the regular Asterisk PBX setup, see below:

Asterisk PBX Step 1: Add the following lines to the end of iax.conf (found in /etc/asterisk)

secret= {{ this is your password, see your website for setup }}

Step 2: Add the following to extensions.conf (found in /etc/asterisk)

; NANPA: North American Numbers dialed as 1 + area code
; For example, the New York Public Library is dialed as 12123400849
; 1 (North American call) 212 (New York area code) 3400849 (libary's phone number)
; WORLD: International Numbers dialed as 011 + country code + number
; For example, the Tate Modern Museum in London, U.K. is dialed as 011442078878000
; 011 (International call) 44 (U.K. country code) 2078878000 (museum's number)
; Finally, the number just before @voipjet in the Dial string is your VoipJet userid #

exten => _1NXXNXXXXXX,1,SetCallerID(4153574000); Set your CallerID as a ten digit number like this. See our FAQ
exten => _1NXXNXXXXXX,2,Dial,IAX2/2135@voipjet/${EXTEN} ; VoipJet.com NANPA
exten => _011.,1,SetCallerID(4153574000); Set your CallerID as a ten digit number like this. See our FAQ.
exten => _011.,2,Dial,IAX2/2135@voipjet/${EXTEN} ; VoipJet.com WORLD
;Do not change IAX2/2135 in the above two lines!

Step 3A (recommended):

; Set your codec to G.711 ulaw for optimal sound clarity and minimal transmission delay.
;They accept iLBC, and GSM Codecs
;In iax.conf (found in /etc/asterisk) locate the codec section and include the following only.
disallow=all ; Prevent all codecs...
allow = ulaw ; ...except G.711 ulaw
allow = ilbc ; ...except G.711 ulaw
allow = gsm ; ...except G.711 ulaw

Step 3B (recommended):

;Also in iax.conf, enable the jitter buffer. This section is usually immediately below the codecs section.
jitterbuffer=yes ; Jitter buffer enabled...
dropcount=1 ; ...to drop at most 0.5% of VoIP packets 

Terms of Service

What protocols do you support?
IAX/IAX2 Inter-Asterisk Exchange for all customers. For larger carriers only, we support SIP and H323 interconnection.

Unresolved problems you can find:
Using their service you will probably see an error message that makes calls impossible to be routed:
WARNING[xxxx]: chan_iax2.c:10190 socket_process: Call rejected by xx.xx.xx.xx: No authority found
If you see that, just forget this provider. It is probably due to the IP restrictions they use for connections and you will never get any support. if it works, it is OK, do not except anything more.

VOIP Service Providers
Created by: jht2, Last modification: Mon 11 of Jun, 2012 (03:39 UTC) by admin
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