MP Asterisk hook up

Revision as of 12:22, 23 July 2009 by Alan Levin (talk | contribs) (→‎Configure an IAX2 connection)
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One only needs to add the trunk to the asterisk configuration files. In order to do that you will need the hostname of the upstream trunk as well as the username and password and codec. Note: the MP does not include codec G729 so you will need an Asterisk that has IAX or GSM codec support.

Each of the following tasks will be scripted and at some point added to the management gui interface. The scripts are required to configure multiple Mesh Potatoes (MPs) as a batch. Without the scripts each MP must be configured independently.

Configure the routing on the MP to use an Internet gateway

Elektra to help here please... the following will instruct the MP to get it's routes from a dhcp server

ip r (to see routing table)
udhcpc -i eth0

Note: it is important to know if you will have a fully routed global IP of if you are behind a NAT. If behind a NAT make sure the correct port forwarding has been setup.

Configure an IAX2 connection

  • Note: IAX2 is currently not working on the MP but we expect this will work best when behind a NAT and/or firewall

1. Setup registration in /etc/asterisk/iax.conf (this will enable inbound calls)

register => username:secretpassword@server.dabba.net

To enable inbound calls to ring on the MP phone edit /etc/asterisk/extensions.conf so

exten => s,1,Dial(MP/1)  
  • this will be set to default

2. Setup outbound calls via dialplan in /etc/asterisk/extensions.conf (in the following example when you dial 4006, the MP will connect to the upstream server ct1.dabba.net with user: user and password:password and then dial ext 111 on that server)

exten => 4006,1,Dial(IAX2/user:password@ct1.dabba.net/111)

When the above is completed don't forget to:

iax2 reload

and:

dialplan reload

to test that the IAX connection is registered you can use the following checks

show iax2 registry
show dialplan

Configure a SIP connection

this may not work well behind a firewall or NAT connection

Create/edit the 'ast' route:

Edit the /etc/asterisk/extensions.conf file (using vi editor)

Find [default] configurations (/[default in vi)

Add/edit the line with ast in the brackets so you have one:

exten => 4003,1,Dial(SIP/ast/10)


Add the SIP login details:

Edit the /etc/asterisk/sip.conf file (using vi editor)

Find ast (/ast in vi) and setup the server host, username and password and codec (in the following example username=mp1 pw=mp server host: 196.35.197.196 and codec=gsm)

[ast]
type=peer                                                                                    
username=mp1                                                                                  
fromuser=ast                                                                                 
secret=mp                                                                                    
context=default                                                                              
disallow=all                                                                                 
qualify=500                                                                                  
dtmfmode=rfc2833                                                                             
callerid=server                                                                              
canreinvite=no                                                                               
host=196.35.197.196                                                                          
allow=gsm 

Once that is done go to asterisk console and type:

dialplan reload

Notes:

  • Ensure the Asterisk gateway has required codecs
  • show dialplan default
  • dialplan reload
  • sip debug (to turn off sip set debug off)
  • sip show peers
  • show dialplan default