title: Asterisk returns 501 - Not Implemented to INFO packets
body: <p>I am trying to send DTMFs through SIPp. Since the play_pcap_audio action was not 100% reliable, I wanted to construct SIP INFO messages to make my tests more robust, however when I send INFO packets I get 501 - Not implemented response from Asterisk.</p>
  
  <p>If I set my softphone to use SIP INFO for sending DTMFs, that works fine, so I am assuming it has to do with the messages I am sending. However comparing the actual messages did not reveal any difference.</p>
  
  <p>The INVITE I send:</p>
  
  <pre><code>INVITE sip:*203@192.168.200.208:5060 SIP/2.0
  Via: SIP/2.0/UDP 172.17.0.2:5060;branch=z9hG4bK-234-1-7;rport
  From: sipp &lt;sip:2018005@192.168.200.208&gt;;tag=1
  To: &lt;sip:*203@192.168.200.208:5060&gt;
  Call-ID: 1-234@172.17.0.2
  CSeq: 4 INVITE
  Contact: sip:2018005@172.17.0.2:5060
  Authorization: //auth header omitted
  Max-Forwards: 70
  Allow: OPTIONS, SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO
  Content-Type: application/sdp
  Content-Length:   195
  
  v=0
  o=user1 53655765 2353687637 IN IP4 172.17.0.2
  s=-
  c=IN IP4 172.17.0.2
  t=0 0
  m=audio 8192 RTP/AVP 0
  a=rtpmap:8 PCMA/8000
  a=rtpmap:101 telephone-event/8000
  a=fmtp:101 0-15
  a=sendrecv
  </code></pre>
  
  <p>and the INFO message:</p>
  
  <pre><code>
  INFO sip:192.168.200.208:5060 SIP/2.0
  Via: SIP/2.0/UDP 172.17.0.2:5060;branch=z9hG4bK-234-1-16
  Max-Forwards: 70
  Contact: &lt;sip:2018005@172.17.0.2:5060;transport=UDP&gt;
  To: &lt;sip:*203@192.168.200.208:5060&gt;
  From: "sipp" &lt;sip:2018005@192.168.200.208&gt;;tag=1
  Call-ID: 1-234@172.17.0.2
  CSeq: 5 INFO
  User-Agent: SIPp docker
  Authorization: // auth header omitted
  Content-Length: 22
  
  Signal=1
  Duration=160
  </code></pre>
  
  <p>I have made sure it's not to do with the <code>dtmfmode</code> configuration in Asterisk.
  One thing I noticed is when Asterisk responds to INVITE, it contains the following header:
  <code>Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER</code>
  I would expect INFO to be here - but again, it's the same when using the softphone and it all works.
  What other areas could affect the processing of SIP INFOs?</p>
  
  <p>Any help would be much appreciated on further debugging.</p>