title: Asterisk outgoing call fails with 'Busy Here' error
body: <p>I have Asterisk 11.7.0 configured on CentOS 6.4 x64 with following sip.conf configuration:</p>
  
  <pre><code>[general]
  register =&gt;myuser:pass@voipproviderip
  registertimeout=20
  context=incoming
  allowoverlap=no
  bindport=5060
  bindaddr=192.168.0.3
  srvlookup=no
  subscribecontext=from-sip
  
  
  [VoIPProvider]
  canreinvite=yes
  username=myuser
  fromuser=myuser
  secret=mypass
  context=incoming
  type=friend
  ;fromdomain=voipproviderip
  host=voipproviderip
  dtmfmode=rfc2833
  disallow=all
  allow=alaw
  allow=ulaw
  nat=yes
  insecure=very
  
  
  
  
  ; ext 100
  [100]
  type=friend
  host=dynamic
  secret=MyPass123
  context=default
  mailbox=100@default
  callgroup=1
  pickupgroup=1
  dtmfmode=rfc2833
  canreinvite=no
  
  ; ext 101
  [101]
  type=friend
  host=dynamic
  secret=MyPass123
  context=default
  mailbox=101@default
  callgroup=1
  pickupgroup=1
  dtmfmode=rfc2833
  canreinvite=no
  
  ; ext 102
  [102]
  type=friend
  host=dynamic
  secret=MyPass123
  context=default
  mailbox=102@default
  callgroup=1
  pickupgroup=1
  dtmfmode=rfc2833
  canreinvite=no
  
  ; ext 103
  [103]
  type=friend
  host=dynamic
  secret=MyPass123
  context=default
  mailbox=103@default
  callgroup=1
  pickupgroup=1
  dtmfmode=rfc2833
  canreinvite=no
  
  ; ext 104
  [104]
  type=friend
  host=dynamic
  secret=MyPass123
  context=default
  mailbox=104@default
  callgroup=1
  pickupgroup=1
  dtmfmode=rfc2833
  canreinvite=no
  
  ; ext 105
  [105]
  type=friend
  host=dynamic
  secret=MyPass123
  context=default
  mailbox=105@default
  callgroup=1
  pickupgroup=1
  dtmfmode=rfc2833
  canreinvite=no
  
  ; ext 106
  [106]
  type=friend
  host=dynamic
  secret=MyPass123
  context=default
  mailbox=106@default
  callgroup=1
  pickupgroup=1
  dtmfmode=rfc2833
  canreinvite=no
  
  ; ext 100
  [107]
  type=friend
  host=dynamic
  secret=MyPass123
  context=default
  mailbox=107@default
  callgroup=1
  pickupgroup=1
  dtmfmode=rfc2833
  canreinvite=no
  
  ; ext 108
  [108]
  type=friend
  host=dynamic
  secret=MyPass123
  context=default
  mailbox=108@default
  callgroup=1
  pickupgroup=1
  dtmfmode=rfc2833
  canreinvite=no
  </code></pre>
  
  <p>and following extensions.conf:</p>
  
  <pre><code>[default]
  include =&gt; internal
  include =&gt; incoming
  include =&gt; outgoing
  
  
  [incoming]
  ; Ring on extension 100, 200 and the mobile phone.
  exten =&gt; s,1,Answer()
  exten =&gt; s,n,Dial(SIP/101&amp;SIP/103,30,r,t,)
  
  
  ; Pass unanswered call to a mobile phone
  exten =&gt; s,n,Dial(SIP/101&amp;SIP/103/&amp;SIP/100,30,r)
      same =&gt; n,GotoIf($["${DIALSTATUS}" = "BUSY"]?busy:unavail)
      same =&gt; n(unavail),VoiceMail(100@default,u)
      same =&gt; n,Hangup()
      same =&gt; n(busy),VoiceMail(100@default,b)
      same =&gt; n,Hangup()
  
  
  [outgoing]
  ; Outbound calls can be routed based on the number of digits dialled (or the value of the first few digits)
  ;exten=&gt; _XXXXXXXXXXXX.,1,Dial(SIP/VoIPProvider/${EXTEN})
  ;exten=&gt; _XXXXXXXXXXXX.,2,Hangup
  
  exten =&gt; _XXXXXXXXXXXX,1,Dial(SIP/VoIPProvider/${EXTEN})
  exten =&gt; _XXXXXXXXX,1,Dial(SIP/VoIPProvider/${EXTEN})
  
  exten =&gt; _XXXXXXXXX,1,Dial(SIP/VoIPProvider/${EXTEN})
  exten =&gt; _XXXXXXX,1,Dial(SIP/VoIPProvider/${EXTEN})
  
  [internal]
  ; Calls between employees (between extensions)
  exten =&gt; _XXX,1,Dial(SIP/${EXTEN},60)
  include =&gt; outgoing
  
  
  ; Calls to ext 100
  exten =&gt; 100,1,Dial(SIP/100,20)
  exten =&gt; 100,n,VoiceMail(100,u)
  exten =&gt; 100,n,Hangup
  
  ; Calls to ext 101
  exten =&gt; 101,1,Dial(SIP/101,20)
  exten =&gt; 101,n,VoiceMail(101,u)
  exten =&gt; 101,n,Hangup
  
  ; Calls to ext 102
  exten =&gt; 102,1,Dial(SIP/102,20)
  exten =&gt; 102,n,VoiceMail(102,u)
  exten =&gt; 102,n,Hangup
  
  ; Calls to ext 103
  exten =&gt; 103,1,Dial(SIP/103,20)
  exten =&gt; 103,n,VoiceMail(103,u)
  exten =&gt; 103,n,Hangup
  
  ; Calls to ext 104
  exten =&gt; 104,1,Dial(SIP/104,20)
  exten =&gt; 104,n,VoiceMail(104,u)
  exten =&gt; 104,n,Hangup
  
  ; Calls to ext 105
  exten =&gt; 105,1,Dial(SIP/105,20)
  exten =&gt; 105,n,VoiceMail(105,u)
  exten =&gt; 105,n,Hangup
  
  ; Calls to ext 106
  exten =&gt; 106,1,Dial(SIP/106,20)
  exten =&gt; 106,n,VoiceMail(106,u)
  exten =&gt; 106,n,Hangup
  
  ; Calls to ext 107
  exten =&gt; 107,1,Dial(SIP/107,20)
  exten =&gt; 107,n,VoiceMail(107,u)
  exten =&gt; 107,n,Hangup
  
  ; Calls to ext 108
  exten =&gt; 108,1,Dial(SIP/108,20)
  exten =&gt; 108,n,VoiceMail(108,u)
  exten =&gt; 108,n,Hangup
  </code></pre>
  
  <p>Whenever I dial a number from <code>CLI&gt;console dial mynumber</code>, I see the folowing output:</p>
  
  <pre><code>    -- Executing [mynumber@default:1] Dial("Console/dsp", "SIP/VoIPProvider/mynumber") in new stack
    == Using SIP RTP CoS mark 5
      -- Called SIP/VoIPProvider/mynumber
      -- Got SIP response 486 "Busy Here" back from voipproviderip:5060
      -- SIP/VoIPProvider-00000000 is busy
    == Everyone is busy/congested at this time (1:1/0/0)
      -- Auto fallthrough, channel 'Console/dsp' status is 'BUSY'
  </code></pre>
  
  <p>I have tried all the possible ways to dial a number, (including / excluding country code, etc.) but it always replays with <code>486 "Busy Here"</code> could it be that I have to pay the bill of my voip service? </p>