SIP call is rejected with an asterisk

I am using Asterisk and Adhearsion. I created an application ahn

and followed these steps:

Added dialplan.rb

to the root directory of the project. Content:

adhearsion {
  play 'hello-world'
  hangup
}

      

        
        
        
      

    

Contents config/adhearsion.rb

:

  ...
  ##
  # Use with Asterisk
  #
  config.punchblock.platform = :asterisk # Use Asterisk
  config.punchblock.username = "myuser" # Your AMI username
  config.punchblock.password = "mypassword" # Your AMI password
  config.punchblock.host = "127.0.0.1" # Your AMI host
  ...

      

        
        
        
      

    

Asterisk config files:

extensions.conf

:

[adhearsion]
exten => test,1,AGI(agi://127.0.0.1)                                                                                                                                      

[adhearsion-redirect]

      

        
        
        
      

    

Content sip.conf

[general]
context=adhearsion
....

      

        
        
        
      

    

I am getting an error when calling (SIP) from SJPhone app:

== Manager 'myuser' logged on from 127.0.0.1
  == Manager 'myuser' logged on from 127.0.0.1
    -- Added extension '1' priority 1 to adhearsion-redirect
  == Using SIP RTP CoS mark 5
    -- Executing [test@adhearsion:1] AGI("SIP/127.0.0.1-00000000", "agi://127.0.0.1") in new stack
[Jan 30 12:02:26] ERROR[4211]: utils.c:1164 ast_carefulwrite: write() returned error: Connection refused
[Jan 30 12:02:26] WARNING[4211]: res_agi.c:1509 launch_netscript: Connect to 'agi://127.0.0.1' failed: Connection refused
    -- Auto fallthrough, channel 'SIP/127.0.0.1-00000000' status is 'UNKNOWN'