Asterisk 1.4

     The AOL Voice team is testing a number of SIP devices and clients with the AIM Call Out service.
We are publishing the steps we took to set up each device to help AIM Call Out customers get up and 
running with the product they use. Here are the instructions for Asterisk.

Setup

-- Software Release: Download the software from  http://www.asterisk.org/downloads
-- Build : Follow the standard Asterisk instructions from http://asterisk.org/support/install
-- Visit the following sites for detailed instructions of how to set up Asterisk in your home or business.
     http://www.voip-info.org/wiki-Asterisk+Step-by-step+Installation
     http://www.jeremy-mcnamara.com/index.php/2007/02/26/how-to-configure-ast...
-- If you are using Free PBX refer to the article from their site, http://nerdvittles.com/index.php?p=215

Note : Please be aware that we limit the number of simultaneous calls that can be made with each AIM Call Out account to three. 

Here is basic config information to terminate calls to our AIM Call Open SIP gateway.

SIP Network Configuration

In order for the SIP devices to place calls using the Asterisk server, two files "sip.conf" and "extensions.conf" have to be edited.

Note: These configuations were tested with the complete source code for Asterisk release 1.4.19.1.

sip.conf:

[general]
context=default ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
pedantic=no
disallow=all
allow=ulaw
dtmfmode=inband
t38udptlsupport=yes
canreinvite=no
;=======================;

; Sample SIP device Definition (Example: X-Lite Softphone)

[100]
;Softphone Phone #1
type=friend
regexten=100
context=default
user=100
secret=100
authuser=100:100
callerid="John Smith" <100>
host=dynamic ; This device needs to register
;nat=yes ; If X-Lite is behind a NAT router
canreinvite=no ; Typically set to NO if behind NAT
disallow=all
qualify=yes
allow=ulaw
allow=alaw


; Sample IP Phone Definition (Example: Linksys/Cisco/Polycom/Grandstream)
[101]
;IP Phone #1
type=friend
regexten=101
context=default
user=101
secret=101
authuser=101:101
callerid= "Phone 1" <101>
host=dynamic ; This device needs to register
qualify=yes
;nat=yes ; IP Phone is behind a NAT router
canreinvite=no ; Typically set to NO if behind NAT
disallow=all
allow=ulaw
allow=alaw

;=======================;

;AOL SIP Peer Termination Definition
[AIMCallOut]
disallow=all
allow=ulaw
allow=alaw
host=sip.aol.com
type=peer
username=johndoe@aim.com ; Replace with your AIM Call Out screen name
;Include the @aim.com or @aol.com after your screen name
secret=sippassword ; Replace with your own SIP password.
externip=sip.aol.com
fromuser=johndoe ; Replace with your AIM Call Out screen name
fromdomain=aol.com
canreinvite=no
qualify=yes
nat=yes
context=inbound_calls ; Not used for AIM Call Out service
=============================================================================

*sip.conf: 
- This file details the SIP configuration for Asterisk. 
- We are showing a sample configuration that would enable an X-lite Softphone and an IP Phone to register with an Asterisk
  server and place calls to the AIM CallOut SIP Gateway (sip.aol.com).
- The general context instructs Asterisk to use the common SIP port of 5060 and to bind to any available network port. 
- The disallow and allow directives define the specific codec(s) that are to be used. 
- The 100 and 101 contexts define the X-lite Softphone and any IP Phone. 
- "host=dynamic" instructs Asterisk that your SIP devices will be registering to your Asterisk server, because they use a dynamically assigned IP address. 
- Secret is the shared secret that you will use to authenticate each SIP device with the Asterisk server.
- "username" and "fromuser" are setup specific variables within the SIP protocol that allows AIM Call Out to authenticate.
- "canreinvite=no" is a specific option informing Asterisk to not attempt to re-invite or change any settings of the call. 
  Re-invites are typically one cause of one-way audio or no audio at all, when behind NAT or firewalls.


extensions.conf:

; Dial local SIP device extensions
[default]
exten => _1XX,1,dial(SIP/${EXTEN},30)

;Sample Global AIMCallOut Dial Plan
exten => _., 1, dial(SIP/+${EXTEN}@AIMCallOut,30)
exten => _., 2, congestion()
exten => _., 102, busy()

=============================================================================

*extensions.conf:
- Extensions are groups of actions that instruct Asterisk exactly how to deal with your telephone calls.
- Extensions typically consist of a number of priorities containing multiple actions, which are grouped inside of contexts. Each context
  contains a logical grouping of actions (extensions) that make your dialing plan. The extensions concept of Asterisk is very flexible
  and powerful, which can be used to create complex telephony applications. In this configuration, we establish a global extensions
  for all the SIP devices interacting with the Asterisk server to make calls out via the AIM Call Out SIP gateway.

Notes

    1. AIM Call Out SIP Clients settings page URL = https://dashboard.voice.aol.com/settings/sip
    2. If you require assistance in setting up your asterisk server , please visit one on the support sites below.
         Asterisk PBX & Telephony Platform Support : www.asteriskpbx.org/support
         Asterisk Forums / Support : www.forums.digium.com/viewforum.php
         Asterisk debugging : http://voip-info.org/wiki/view/Asterisk+debugging

The AIM services are provided 'as is' and are subject to the AIM Call Out Terms of Service.