title: Asterisk outgoing call fails
body: <p>I have Asterisk 11.7.0 configured on CentOS 6.4 x64 with following <code>sip.conf</code> configuration:</p>
  
  <pre><code>[general]
  register =&gt;mynumber:mypass@xxx.xxx.xxx.xxx
  registertimeout=20
  context=incoming
  allowoverlap=no
  bindport=5060
  bindaddr=192.168.0.3
  srvlookup=no
  subscribecontext=from-sip
  
  ; The SIP provider
  [VoIPProvider]
  canreinvite=no
  username=mynumber
  fromuser=mynumber
  secret=mypass
  context=incoming
  type=friend
  fromdomain=xxx.xxx.xxx.xxx
  ;host=xxx.xxx.xxx.xxx
  dtmfmode=rfc2833
  disallow=all
  allow=alaw
  allow=ulaw
  nat=yes
  insecure=very
  
  ; ext 100
  [100]
  type=friend
  host=dynamic
  secret=MyPass123
  context=internal
  mailbox=100@default
  callgroup=1
  pickupgroup=1
  dtmfmode=rfc2833
  canreinvite=no
  
  ; ext 200
  [200]
  type=friend
  host=dynamic
  secret=MyPass123
  context=internal
  callgroup=1
  pickupgroup=1
  dtmfmode=rfc2833
  canreinvite=no
  </code></pre>
  
  <p>And following <code>extensions.conf:</code></p>
  
  <pre><code>[incoming]
  ; Ring on extension 100, 200 and the mobile phone.
  exten =&gt; s,1,Answer()
  exten =&gt; s,n,Dial(SIP/100&amp;SIP/200&amp;SIP/VoIPProvider/*320423456789,150,r,t,)
  
  ; Pass unanswered call to a mobile phone
  exten =&gt; s,n,Dial(SIP/VoIPProvider/*320423456789,150,r)
  
  ; Still not answered? Pass unanswered calls to voicemail
  exten =&gt; s,n,Voicemail(100,u)
  exten =&gt; s,n,Hangup
  
  [outgoing]
  exten =&gt; _XXXXXXXXXXXXXXX,1,Dial(SIP/VoIPProvider/${EXTEN})
  exten =&gt; _XXXXXXXXXX,1,Dial(SIP/VoIPProvider/${EXTEN})
  exten =&gt; _XXXXXX,1,Dial(SIP/VoIPProvider/${EXTEN})
  
  [internal]
  exten =&gt; _XXX,1,Dial(SIP/${EXTEN})
  
  ; 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 200
  exten =&gt; 200,1,Dial(SIP/100,20)
  exten =&gt; 200,n,Hangup
  </code></pre>
  
  <p>When I try to dial my mobile number from an IP phone I see following output in Asterisk CLI:</p>
  
  <pre><code>-- Executing [XXXXXXXXX@outgoing:1] Dial("SIP/XXX-00000002", "SIP/VoIPProvider/XXXXXXXXX") in new stack
  
  WARNING[19884][C-00000003]: app_dial.c:2437 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
  
  == Everyone is busy/congested at this time (1:0/0/1)
  -- Auto fallthrough, channel 'SIP/XXX-00000002' status is 'CHANUNAVAIL'
  </code></pre>
  
  <p>Now, I know what <code>cause 20 - Subscriber absent</code> means, but I know for sure that my mobile number exists and is reachable, because when I dial the IP Phone's number from my mobile (incoming call) it works.</p>
  
  <p>Any suggestions?</p>