Asterisk Configurations for connecting with VOIP providers

Business SIP Provider
Provider Plan Details Monthly Rate *
Nextiva SIP Trunking PBX SIP Trunking
  • Compatible with nearly all PBX systems
  • US-based support
  • No contracts at all
$14.95
 Details

150 Reviews
Vocalocity Unlimited Extension
  • Unlimited calling and long distance
  • Company Call Record, Voicemail to email transcription
  • No contracts, not setup or installation fees
$24.99
 Details

35 Reviews
RingCentral SIP RingCentral Office
  • Unlimited calling and faxing
  • No Commitment, No Setup Fees, No Installation
$19.99
 Details

64 Reviews
8x8 8x8 IP Trunking
  • No contract required.
  • Unlimited Calls (9 free countries)
  • Save 50% or more
$29.99
 Details

88 Reviews
Jive Communications Smart PBX
  • Unlimited Use of All Features
  • Free Long-Distance
  • No Contracts or Hidden Fees
$21.95
 Details

147 Reviews
Layer Four VoIP Hosted SIP PBX
  • Free IP phones with most plans.
  • Simple flat rate billing
  • Your choice of usage based, extension based, or unlimited
  • Free 30 day trial, no credit card required
$19.00
 Details

1 Reviews
*Taxes and other fees may apply.
VoIP Hardware Solutions
Provider Solution Details
VoIP Supply VoIP Phones and Hardware
  • Over 90 manufacturers and 6,500 products
  • Live experts ready to answer your questions
  • Volume pricing and financing for resellers
 Details
Sangoma VoIP Hardware Hardware for Asterisk
  • TDM Interface Boards & Gateways
  • Guaranteed for life
  • Often Copied, Never Duplicated
 Details
Business PBX Solution
Provider Solution Details
Bicom VoIP Become an ITSP Now!
  • Become a serious competitor in VoIP Immediately
  • FULL Consultancy, Installation, Training & Support
  • Sell Hosted IP PBXs, Biz Lines, Call Centre
  • Turnkey Provisioning at your data center
 Details

Asterisk provider specific settings

This page is intended to provide a place for people to place excerpts from configuration files for specific providers.

In General

It is considered bad style to use a dialstring with a password as in "user:pass@voipserver.com" since username and password will then show up on the console. Instead, for each provider enter a peer entry in sip.conf or iax.conf and use that peer entry's name in your dialstring as in "user@peername". Username and password pairs in dialstrings should only be used for testing!

Provider Specific Settings




Settings for softswitches


More sample scripts are to be found on the Asterisk tips and tricks page.

Created by: jjhall, Last modification: Tue 07 of Jun, 2011 (15:31 UTC) by o.hammami


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