title: Why does not 'channel originate' call?
body: <p>I would like invite somebody into a context.</p>
  
  <p>I test with:</p>
  
  <pre><code>asterisk*CLI&gt; channel originate SIP/trunk-test/PHONE extension s@my-context
  </code></pre>
  
  <p>PHONE is phone number of somebody.</p>
  
  <pre><code>my-context is a context into my dialplan which contains a MeetMe Room
  </code></pre>
  
  <p>The command prints:</p>
  
  <pre><code>== Using SIP RTP CoS mark 5
      -- Called trunk-test/PHONE
  </code></pre>
  
  <p>But the phone don't ring.</p>
  
  <p><strong>Note</strong>: Yesterday it worked. I could use this command to call external phones. But I couldn't reach from out. </p>
  
  <p>Today, magically, without changes in my configuration, this is inversed. This command does not work. But I can reach from out.</p>
  
  <p>Here is my debug dump (from sip set debug on)</p>
  
  <p>SIP_EXTENSION is the phone number I call to reach the my-context.</p>
  
  <pre><code>asterisk*CLI&gt; channel originate SIP/trunk-test/PHONE extension s@my-context
    == Using SIP RTP CoS mark 5
  Audio is at 15896
  Adding codec ulaw to SDP
  Adding codec alaw to SDP
  Adding codec gsm to SDP
  Reliably Transmitting (NAT) to 91.121.129.23:5060:
  INVITE sip:PHONE@siptrunk.ovh.net SIP/2.0
  Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5158c2d4;rport
  Max-Forwards: 70
  From: "Anonymous" &lt;sip:SIP_EXTENSION@siptrunk.ovh.net&gt;;tag=as6a21d2d9
  To: &lt;sip:PHONE@siptrunk.ovh.net&gt;
  Contact: &lt;sip:SIP_EXTENSION@192.168.1.10:5060&gt;
  Call-ID: 3dfff18c396337fb3c55a3fb7bc8c4c3@siptrunk.ovh.net
  CSeq: 102 INVITE
  User-Agent: Asterisk PBX certified/13.1-cert4
  Date: Mon, 04 Apr 2016 01:34:56 GMT
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
  Supported: replaces, timer
  Content-Type: application/sdp
  Content-Length: 246
  
  v=0
  o=root 1530515320 1530515320 IN IP4 37.187.205.63
  s=Asterisk PBX certified/13.1-cert4
  c=IN IP4 MY_PUBLIC_IP
  t=0 0
  m=audio 15896 RTP/AVP 0 8 3
  a=rtpmap:0 PCMU/8000
  a=rtpmap:8 PCMA/8000
  a=rtpmap:3 GSM/8000
  a=maxptime:150
  a=sendrecv
  
  ---
      -- Called trunk-test/PHONE
  
  &lt;--- SIP read from UDP:192.168.1.1:5060 ---&gt;
  SIP/2.0 408 Request Timeout
  Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5158c2d4;rport
  From: "Anonymous" &lt;sip:SIP_EXTENSION@siptrunk.ovh.net&gt;;tag=as6a21d2d9
  To: &lt;sip:PHONE@siptrunk.ovh.net&gt;
  Call-ID: 3dfff18c396337fb3c55a3fb7bc8c4c3@siptrunk.ovh.net
  CSeq: 102 INVITE
  Content-Length: 0
  
  &lt;-------------&gt;
  --- (7 headers 0 lines) ---
  Transmitting (NAT) to 192.168.1.1:5060:
  ACK sip:PHONE@siptrunk.ovh.net SIP/2.0
  Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5158c2d4;rport
  Max-Forwards: 70
  From: "Anonymous" &lt;sip:SIP_EXTENSION@siptrunk.ovh.net&gt;;tag=as6a21d2d9
  To: &lt;sip:PHONE@siptrunk.ovh.net&gt;
  Contact: &lt;sip:SIP_EXTENSION@192.168.1.10:5060&gt;
  Call-ID: 3dfff18c396337fb3c55a3fb7bc8c4c3@siptrunk.ovh.net
  CSeq: 102 ACK
  User-Agent: Asterisk PBX certified/13.1-cert4
  Content-Length: 0
  
  
  ---
  Scheduling destruction of SIP dialog '3dfff18c396337fb3c55a3fb7bc8c4c3@siptrunk.ovh.net' in 32000 ms (Method: INVITE)
  
  &lt;--- SIP read from UDP:192.168.1.1:5060 ---&gt;
  SIP/2.0 408 Request Timeout
  Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5158c2d4;rport
  From: "Anonymous" &lt;sip:SIP_EXTENSION@siptrunk.ovh.net&gt;;tag=as6a21d2d9
  To: &lt;sip:PHONE@siptrunk.ovh.net&gt;
  Call-ID: 3dfff18c396337fb3c55a3fb7bc8c4c3@siptrunk.ovh.net
  CSeq: 102 ACK
  Content-Length: 0
  
  &lt;-------------&gt;
  --- (7 headers 0 lines) ---
  </code></pre>
  
  <p>Do you have some ideas what the issue is?</p>