title: Asterisk call drop after 30 seconds
body: <p>I've installed Asterisk and made a call using Android Zoiper app. It successfully connects two users and hear sound, but call drops after 30 seconds.  </p>
  
  <p><strong>asterisk logs</strong></p>
  
  <pre><code>[Apr 14 18:40:34] WARNING[27959]: chan_sip.c:4176 retrans_pkt: 
  Retransmission timeout reached on transmission lPsW4atWG- for seqno 20 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
  Packet timed out after 31999ms with no response
  [Apr 14 18:40:34] WARNING[27959]: chan_sip.c:4205 retrans_pkt: 
  Hanging up call lPsW4atWG- - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
    == Spawn extension (from-sip, 1000, 1) exited non-zero on 'SIP/2000-0000000a'
  </code></pre>
  
  <p><strong>Sip.conf</strong></p>
  
  <pre><code>[general]
   context=default                       ; Default context for incoming calls
   ;
   bindport=5060                   ; bindport is the local UDP port that Asterisk will listen on
   bindaddr=0.0.0.0           ; IP address to bind to (0.0.0.0 binds to all)
   ;
   disallow=all                    ; First disallow all codecs
   allow=gsm
   allow=ulaw                      ; Allow codecs in order of preference
   ;
   register =&gt; 12121111111:1234:11111111@sipauth.deltathree.com/1000
  
  
  allow=g729
  allow=alaw
  srvlookup=no
  canreinvite=no
  directrtpsetup=no
  trustpid=yes
  sendrpid=yes
  qualify=yes
  callevents=yes
  insecure=invite
  pedantic=no
  useragent=Glastender PBX
  videosupport=no
  t38pt_udptl=no
  t38pt_rtp=no
  t38pt_tcp=no
  
  nat=yes
  media_address = XXX.52.91.XXX ; server ip address
  </code></pre>
  
  <p>It looks like I need to change something on sip.conf, and tried different configs. It is not working yet.. Do you see any problems? </p>
  
  <p><strong>SIP logs</strong> </p>
  
  <pre><code>interface: eth0 (10.7.21.0/255.255.255.0)
  filter: ( port 5060 ) and (ip or ip6)
  #
  U 2014/04/15 00:22:15.941072 XX.53.122.134:5060 -&gt; 10.8.21.XX:5060
  INVITE sip:1000@sipdomain.com SIP/2.0.
  Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;rport.
  From: &lt;sip:2000@sipdomain.com&gt;;tag=dGlp5o0FS.
  To: sip:1000@sipdomain.com.
  CSeq: 20 INVITE.
  Call-ID: wh8Ai1e~0c.
  Max-Forwards: 70.
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO.
  Content-Type: application/sdp.
  Content-Length: 280.
  User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4).
  Contact: &lt;sip:2000@XX.53.122.134&gt;;+sip.instance="&lt;urn:uuid:0b49a090-f01c-41a2-b771-bdc956e9b516&gt;".
  .
  v=0.
  o=2000 274 59 IN IP4 192.168.0.38.
  s=Talk.
  c=IN IP4 192.168.0.38.
  b=AS:380.
  t=0 0.
  m=audio 7076 RTP/AVP 0 8 101.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-11.
  m=video 9078 RTP/AVP 103 99.
  a=rtpmap:103 VP8/90000.
  a=rtpmap:99 MP4V-ES/90000.
  a=fmtp:99 profile-level-id=3.
  
  #
  U 2014/04/15 00:22:15.945220 10.8.21.XX:5060 -&gt; XX.53.122.134:5060
  SIP/2.0 100 Trying.
  Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
  From: &lt;sip:2000@sipdomain.com&gt;;tag=dGlp5o0FS.
  To: sip:1000@sipdomain.com.
  Call-ID: wh8Ai1e~0c.
  CSeq: 20 INVITE.
  Server: Asterisk PBX 11.8.1.
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  Supported: replaces, timer.
  Contact: &lt;sip:1000@10.8.21.XX:5060&gt;.
  Content-Length: 0.
  .
  
  #
  U 2014/04/15 00:22:15.951499 10.8.21.XX:5060 -&gt; 223.XX.130.50:40764
  INVITE sip:1000@223.XX.130.50:40764 SIP/2.0.
  Via: SIP/2.0/UDP 10.8.21.XX:5060;branch=z9hG4bK70816646;rport.
  Max-Forwards: 70.
  From: &lt;sip:2000@10.8.21.XX&gt;;tag=as679b5fe7.
  To: &lt;sip:1000@223.XX.130.50:40764&gt;.
  Contact: &lt;sip:2000@10.8.21.XX:5060&gt;.
  Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
  CSeq: 102 INVITE.
  User-Agent: Asterisk PBX 11.8.1.
  Date: Mon, 14 Apr 2014 15:22:15 GMT.
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  Supported: replaces, timer.
  Content-Type: application/sdp.
  Content-Length: 258.
  .
  v=0.
  o=root 1811076761 1811076761 IN IP4 192.168.0.38.
  s=Asterisk PBX 11.8.1.
  c=IN IP4 192.168.0.38.
  t=0 0.
  m=audio 7076 RTP/AVP 0 8 101.
  a=rtpmap:0 PCMU/8000.
  a=rtpmap:8 PCMA/8000.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-16.
  a=ptime:20.
  a=sendrecv.
  
  #
  U 2014/04/15 00:22:16.045285 223.XX.130.50:40764 -&gt; 10.8.21.XX:5060
  SIP/2.0 100 Trying.
  Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK70816646;rport.
  From: &lt;sip:2000@10.8.21.XX&gt;;tag=as679b5fe7.
  To: sip:1000@223.XX.130.50:40764.
  Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
  CSeq: 102 INVITE.
  .
  
  #
  U 2014/04/15 00:22:16.445425 223.XX.130.50:40764 -&gt; 10.8.21.XX:5060
  SIP/2.0 180 Ringing.
  Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK70816646;rport.
  From: &lt;sip:2000@10.8.21.XX&gt;;tag=as679b5fe7.
  To: &lt;sip:1000@223.XX.130.50:40764&gt;;tag=coOV3rP.
  Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
  CSeq: 102 INVITE.
  User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4).
  .
  
  #
  U 2014/04/15 00:22:16.447116 10.8.21.XX:5060 -&gt; XX.53.122.134:5060
  SIP/2.0 180 Ringing.
  Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
  From: &lt;sip:2000@sipdomain.com&gt;;tag=dGlp5o0FS.
  To: sip:1000@sipdomain.com;tag=as1ba98ffc.
  Call-ID: wh8Ai1e~0c.
  CSeq: 20 INVITE.
  Server: Asterisk PBX 11.8.1.
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  Supported: replaces, timer.
  Contact: &lt;sip:1000@10.8.21.XX:5060&gt;.
  Content-Length: 0.
  .
  
  #
  U 2014/04/15 00:22:16.838201 XX.53.122.134:5060 -&gt; 10.8.21.XX:5060
  .
  .
  
  #
  U 2014/04/15 00:22:19.275720 223.XX.130.50:40764 -&gt; 10.8.21.XX:5060
  SIP/2.0 200 Ok.
  Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK70816646;rport.
  From: &lt;sip:2000@10.8.21.XX&gt;;tag=as679b5fe7.
  To: &lt;sip:1000@223.XX.130.50:40764&gt;;tag=coOV3rP.
  Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
  CSeq: 102 INVITE.
  User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4).
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO.
  Contact: &lt;sip:1000@223.XX.130.50:40764&gt;;+sip.instance="&lt;urn:uuid:307db642-fa79-44d8-835f-15152558c31a&gt;".
  Content-Type: application/sdp.
  Content-Length: 176.
  .
  v=0.
  o=1000 3792 2294 IN IP4 223.XX.130.50.
  s=Talk.
  c=IN IP4 223.XX.130.50.
  b=AS:380.
  t=0 0.
  m=audio 45068 RTP/AVP 0 8 101.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-11.
  
  #
  U 2014/04/15 00:22:19.276630 10.8.21.XX:5060 -&gt; 223.XX.130.50:40764
  ACK sip:1000@223.XX.130.50:40764 SIP/2.0.
  Via: SIP/2.0/UDP 10.8.21.XX:5060;branch=z9hG4bK730c16dd;rport.
  Max-Forwards: 70.
  From: &lt;sip:2000@10.8.21.XX&gt;;tag=as679b5fe7.
  To: &lt;sip:1000@223.XX.130.50:40764&gt;;tag=coOV3rP.
  Contact: &lt;sip:2000@10.8.21.XX:5060&gt;.
  Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
  CSeq: 102 ACK.
  User-Agent: Asterisk PBX 11.8.1.
  Content-Length: 0.
  .
  
  #
  U 2014/04/15 00:22:19.276978 10.8.21.XX:5060 -&gt; XX.53.122.134:5060
  SIP/2.0 200 OK.
  Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
  From: &lt;sip:2000@sipdomain.com&gt;;tag=dGlp5o0FS.
  To: sip:1000@sipdomain.com;tag=as1ba98ffc.
  Call-ID: wh8Ai1e~0c.
  CSeq: 20 INVITE.
  Server: Asterisk PBX 11.8.1.
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  Supported: replaces, timer.
  Contact: &lt;sip:1000@10.8.21.XX:5060&gt;.
  Content-Type: application/sdp.
  Content-Length: 287.
  .
  v=0.
  o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
  s=Asterisk PBX 11.8.1.
  c=IN IP4 223.XX.130.50.
  t=0 0.
  m=audio 45068 RTP/AVP 0 8 101.
  a=rtpmap:0 PCMU/8000.
  a=rtpmap:8 PCMA/8000.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-16.
  a=ptime:20.
  a=sendrecv.
  m=video 0 RTP/AVP 103 99.
  
  #
  U 2014/04/15 00:22:19.776861 10.8.21.XX:5060 -&gt; XX.53.122.134:5060
  SIP/2.0 200 OK.
  Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
  From: &lt;sip:2000@sipdomain.com&gt;;tag=dGlp5o0FS.
  To: sip:1000@sipdomain.com;tag=as1ba98ffc.
  Call-ID: wh8Ai1e~0c.
  CSeq: 20 INVITE.
  Server: Asterisk PBX 11.8.1.
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  Supported: replaces, timer.
  Contact: &lt;sip:1000@10.8.21.XX:5060&gt;.
  Content-Type: application/sdp.
  Content-Length: 287.
  .
  v=0.
  o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
  s=Asterisk PBX 11.8.1.
  c=IN IP4 223.XX.130.50.
  t=0 0.
  m=audio 45068 RTP/AVP 0 8 101.
  a=rtpmap:0 PCMU/8000.
  a=rtpmap:8 PCMA/8000.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-16.
  a=ptime:20.
  a=sendrecv.
  m=video 0 RTP/AVP 103 99.
  
  #
  U 2014/04/15 00:22:20.778018 10.8.21.XX:5060 -&gt; XX.53.122.134:5060
  SIP/2.0 200 OK.
  Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
  From: &lt;sip:2000@sipdomain.com&gt;;tag=dGlp5o0FS.
  To: sip:1000@sipdomain.com;tag=as1ba98ffc.
  Call-ID: wh8Ai1e~0c.
  CSeq: 20 INVITE.
  Server: Asterisk PBX 11.8.1.
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  Supported: replaces, timer.
  Contact: &lt;sip:1000@10.8.21.XX:5060&gt;.
  Content-Type: application/sdp.
  Content-Length: 287.
  .
  v=0.
  o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
  s=Asterisk PBX 11.8.1.
  c=IN IP4 223.XX.130.50.
  t=0 0.
  m=audio 45068 RTP/AVP 0 8 101.
  a=rtpmap:0 PCMU/8000.
  a=rtpmap:8 PCMA/8000.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-16.
  a=ptime:20.
  a=sendrecv.
  m=video 0 RTP/AVP 103 99.
  
  #
  U 2014/04/15 00:22:22.777522 10.8.21.XX:5060 -&gt; XX.53.122.134:5060
  SIP/2.0 200 OK.
  Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
  From: &lt;sip:2000@sipdomain.com&gt;;tag=dGlp5o0FS.
  To: sip:1000@sipdomain.com;tag=as1ba98ffc.
  Call-ID: wh8Ai1e~0c.
  CSeq: 20 INVITE.
  Server: Asterisk PBX 11.8.1.
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  Supported: replaces, timer.
  Contact: &lt;sip:1000@10.8.21.XX:5060&gt;.
  Content-Type: application/sdp.
  Content-Length: 287.
  .
  v=0.
  o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
  s=Asterisk PBX 11.8.1.
  c=IN IP4 223.XX.130.50.
  t=0 0.
  m=audio 45068 RTP/AVP 0 8 101.
  a=rtpmap:0 PCMU/8000.
  a=rtpmap:8 PCMA/8000.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-16.
  a=ptime:20.
  a=sendrecv.
  m=video 0 RTP/AVP 103 99.
  
  #
  U 2014/04/15 00:22:25.139894 XX.53.122.134:32840 -&gt; 10.8.21.XX:5060
  .
  .
  
  #
  U 2014/04/15 00:22:26.777002 10.8.21.XX:5060 -&gt; XX.53.122.134:5060
  SIP/2.0 200 OK.
  Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
  From: &lt;sip:2000@sipdomain.com&gt;;tag=dGlp5o0FS.
  To: sip:1000@sipdomain.com;tag=as1ba98ffc.
  Call-ID: wh8Ai1e~0c.
  CSeq: 20 INVITE.
  Server: Asterisk PBX 11.8.1.
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  Supported: replaces, timer.
  Contact: &lt;sip:1000@10.8.21.XX:5060&gt;.
  Content-Type: application/sdp.
  Content-Length: 287.
  .
  v=0.
  o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
  s=Asterisk PBX 11.8.1.
  c=IN IP4 223.XX.130.50.
  t=0 0.
  m=audio 45068 RTP/AVP 0 8 101.
  a=rtpmap:0 PCMU/8000.
  a=rtpmap:8 PCMA/8000.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-16.
  a=ptime:20.
  a=sendrecv.
  m=video 0 RTP/AVP 103 99.
  
  #
  U 2014/04/15 00:22:30.179568 XX.53.122.134:55180 -&gt; 10.8.21.XX:5060
  .
  .
  
  #
  U 2014/04/15 00:22:30.777462 10.8.21.XX:5060 -&gt; XX.53.122.134:5060
  SIP/2.0 200 OK.
  Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
  From: &lt;sip:2000@sipdomain.com&gt;;tag=dGlp5o0FS.
  To: sip:1000@sipdomain.com;tag=as1ba98ffc.
  Call-ID: wh8Ai1e~0c.
  CSeq: 20 INVITE.
  Server: Asterisk PBX 11.8.1.
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  Supported: replaces, timer.
  Contact: &lt;sip:1000@10.8.21.XX:5060&gt;.
  Content-Type: application/sdp.
  Content-Length: 287.
  .
  v=0.
  o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
  s=Asterisk PBX 11.8.1.
  c=IN IP4 223.XX.130.50.
  t=0 0.
  m=audio 45068 RTP/AVP 0 8 101.
  a=rtpmap:0 PCMU/8000.
  a=rtpmap:8 PCMA/8000.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-16.
  a=ptime:20.
  a=sendrecv.
  m=video 0 RTP/AVP 103 99.
  
  #
  U 2014/04/15 00:22:34.777660 10.8.21.XX:5060 -&gt; XX.53.122.134:5060
  SIP/2.0 200 OK.
  Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
  From: &lt;sip:2000@sipdomain.com&gt;;tag=dGlp5o0FS.
  To: sip:1000@sipdomain.com;tag=as1ba98ffc.
  Call-ID: wh8Ai1e~0c.
  CSeq: 20 INVITE.
  Server: Asterisk PBX 11.8.1.
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  Supported: replaces, timer.
  Contact: &lt;sip:1000@10.8.21.XX:5060&gt;.
  Content-Type: application/sdp.
  Content-Length: 287.
  .
  v=0.
  o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
  s=Asterisk PBX 11.8.1.
  c=IN IP4 223.XX.130.50.
  t=0 0.
  m=audio 45068 RTP/AVP 0 8 101.
  a=rtpmap:0 PCMU/8000.
  a=rtpmap:8 PCMA/8000.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-16.
  a=ptime:20.
  a=sendrecv.
  m=video 0 RTP/AVP 103 99.
  
  #
  U 2014/04/15 00:22:38.777721 10.8.21.XX:5060 -&gt; XX.53.122.134:5060
  SIP/2.0 200 OK.
  Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
  From: &lt;sip:2000@sipdomain.com&gt;;tag=dGlp5o0FS.
  To: sip:1000@sipdomain.com;tag=as1ba98ffc.
  Call-ID: wh8Ai1e~0c.
  CSeq: 20 INVITE.
  Server: Asterisk PBX 11.8.1.
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  Supported: replaces, timer.
  Contact: &lt;sip:1000@10.8.21.XX:5060&gt;.
  Content-Type: application/sdp.
  Content-Length: 287.
  .
  v=0.
  o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
  s=Asterisk PBX 11.8.1.
  c=IN IP4 223.XX.130.50.
  t=0 0.
  m=audio 45068 RTP/AVP 0 8 101.
  a=rtpmap:0 PCMU/8000.
  a=rtpmap:8 PCMA/8000.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-16.
  a=ptime:20.
  a=sendrecv.
  m=video 0 RTP/AVP 103 99.
  
  #
  U 2014/04/15 00:22:42.777667 10.8.21.XX:5060 -&gt; XX.53.122.134:5060
  SIP/2.0 200 OK.
  Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
  From: &lt;sip:2000@sipdomain.com&gt;;tag=dGlp5o0FS.
  To: sip:1000@sipdomain.com;tag=as1ba98ffc.
  Call-ID: wh8Ai1e~0c.
  CSeq: 20 INVITE.
  Server: Asterisk PBX 11.8.1.
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  Supported: replaces, timer.
  Contact: &lt;sip:1000@10.8.21.XX:5060&gt;.
  Content-Type: application/sdp.
  Content-Length: 287.
  .
  v=0.
  o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
  s=Asterisk PBX 11.8.1.
  c=IN IP4 223.XX.130.50.
  t=0 0.
  m=audio 45068 RTP/AVP 0 8 101.
  a=rtpmap:0 PCMU/8000.
  a=rtpmap:8 PCMA/8000.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-16.
  a=ptime:20.
  a=sendrecv.
  m=video 0 RTP/AVP 103 99.
  
  #
  U 2014/04/15 00:22:46.776449 10.8.21.XX:5060 -&gt; XX.53.122.134:5060
  SIP/2.0 200 OK.
  Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
  From: &lt;sip:2000@sipdomain.com&gt;;tag=dGlp5o0FS.
  To: sip:1000@sipdomain.com;tag=as1ba98ffc.
  Call-ID: wh8Ai1e~0c.
  CSeq: 20 INVITE.
  Server: Asterisk PBX 11.8.1.
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  Supported: replaces, timer.
  Contact: &lt;sip:1000@10.8.21.XX:5060&gt;.
  Content-Type: application/sdp.
  Content-Length: 287.
  .
  v=0.
  o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
  s=Asterisk PBX 11.8.1.
  c=IN IP4 223.XX.130.50.
  t=0 0.
  m=audio 45068 RTP/AVP 0 8 101.
  a=rtpmap:0 PCMU/8000.
  a=rtpmap:8 PCMA/8000.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-16.
  a=ptime:20.
  a=sendrecv.
  m=video 0 RTP/AVP 103 99.
  
  #
  U 2014/04/15 00:22:46.927655 XX.53.122.134:5060 -&gt; 10.8.21.XX:5060
  .
  .
  
  #
  U 2014/04/15 00:22:50.776948 10.8.21.XX:5060 -&gt; XX.53.122.134:5060
  SIP/2.0 200 OK.
  Via: SIP/2.0/UDP 192.168.0.38:5060;branch=z9hG4bK.5GT~xTFUf;received=XX.53.122.134;rport=5060.
  From: &lt;sip:2000@sipdomain.com&gt;;tag=dGlp5o0FS.
  To: sip:1000@sipdomain.com;tag=as1ba98ffc.
  Call-ID: wh8Ai1e~0c.
  CSeq: 20 INVITE.
  Server: Asterisk PBX 11.8.1.
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  Supported: replaces, timer.
  Contact: &lt;sip:1000@10.8.21.XX:5060&gt;.
  Content-Type: application/sdp.
  Content-Length: 287.
  .
  v=0.
  o=root 1836373944 1836373944 IN IP4 223.XX.130.50.
  s=Asterisk PBX 11.8.1.
  c=IN IP4 223.XX.130.50.
  t=0 0.
  m=audio 45068 RTP/AVP 0 8 101.
  a=rtpmap:0 PCMU/8000.
  a=rtpmap:8 PCMA/8000.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-16.
  a=ptime:20.
  a=sendrecv.
  m=video 0 RTP/AVP 103 99.
  
  #
  U 2014/04/15 00:22:51.278124 10.8.21.XX:5060 -&gt; XX.53.122.134:5060
  INVITE sip:2000@XX.53.122.134 SIP/2.0.
  Via: SIP/2.0/UDP 10.8.21.XX:5060;branch=z9hG4bK348a4dc2;rport.
  Max-Forwards: 70.
  From: sip:1000@sipdomain.com;tag=as1ba98ffc.
  To: &lt;sip:2000@sipdomain.com&gt;;tag=dGlp5o0FS.
  Contact: &lt;sip:1000@10.8.21.XX:5060&gt;.
  Call-ID: wh8Ai1e~0c.
  CSeq: 102 INVITE.
  User-Agent: Asterisk PBX 11.8.1.
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  Supported: replaces, timer.
  Content-Type: application/sdp.
  Content-Length: 259.
  .
  v=0.
  o=root 1836373944 1836373945 IN IP4 117.52.91.12.
  s=Asterisk PBX 11.8.1.
  c=IN IP4 117.52.91.12.
  t=0 0.
  m=audio 19152 RTP/AVP 0 8 101.
  a=rtpmap:0 PCMU/8000.
  a=rtpmap:8 PCMA/8000.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-16.
  a=ptime:20.
  a=sendrecv.
  
  #
  U 2014/04/15 00:22:51.278285 10.8.21.XX:5060 -&gt; 223.XX.130.50:40764
  INVITE sip:1000@223.XX.130.50:40764 SIP/2.0.
  Via: SIP/2.0/UDP 10.8.21.XX:5060;branch=z9hG4bK59c0124b;rport.
  Max-Forwards: 70.
  From: &lt;sip:2000@10.8.21.XX&gt;;tag=as679b5fe7.
  To: &lt;sip:1000@223.XX.130.50:40764&gt;;tag=coOV3rP.
  Contact: &lt;sip:2000@10.8.21.XX:5060&gt;.
  Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
  CSeq: 103 INVITE.
  User-Agent: Asterisk PBX 11.8.1.
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH.
  Supported: replaces, timer.
  Content-Type: application/sdp.
  Content-Length: 259.
  .
  v=0.
  o=root 1811076761 1811076762 IN IP4 117.52.91.12.
  s=Asterisk PBX 11.8.1.
  c=IN IP4 117.52.91.12.
  t=0 0.
  m=audio 15858 RTP/AVP 0 8 101.
  a=rtpmap:0 PCMU/8000.
  a=rtpmap:8 PCMA/8000.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-16.
  a=ptime:20.
  a=sendrecv.
  
  #
  U 2014/04/15 00:22:51.344965 223.XX.130.50:40764 -&gt; 10.8.21.XX:5060
  SIP/2.0 100 Trying.
  Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK59c0124b;rport.
  From: &lt;sip:2000@10.8.21.XX&gt;;tag=as679b5fe7.
  To: &lt;sip:1000@223.XX.130.50:40764&gt;;tag=coOV3rP.
  Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
  CSeq: 103 INVITE.
  .
  
  #
  U 2014/04/15 00:22:51.355122 223.XX.130.50:40764 -&gt; 10.8.21.XX:5060
  SIP/2.0 200 Ok.
  Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK59c0124b;rport.
  From: &lt;sip:2000@10.8.21.XX&gt;;tag=as679b5fe7.
  To: &lt;sip:1000@223.XX.130.50:40764&gt;;tag=coOV3rP.
  Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
  CSeq: 103 INVITE.
  User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4).
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO.
  Contact: &lt;sip:1000@223.XX.130.50:40764&gt;;+sip.instance="&lt;urn:uuid:307db642-fa79-44d8-835f-15152558c31a&gt;".
  Content-Type: application/sdp.
  Content-Length: 176.
  .
  v=0.
  o=1000 3792 2296 IN IP4 223.XX.130.50.
  s=Talk.
  c=IN IP4 223.XX.130.50.
  b=AS:380.
  t=0 0.
  m=audio 45068 RTP/AVP 0 8 101.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-11.
  
  #
  U 2014/04/15 00:22:51.355539 10.8.21.XX:5060 -&gt; 223.XX.130.50:40764
  ACK sip:1000@223.XX.130.50:40764 SIP/2.0.
  Via: SIP/2.0/UDP 10.8.21.XX:5060;branch=z9hG4bK144199ce;rport.
  Max-Forwards: 70.
  From: &lt;sip:2000@10.8.21.XX&gt;;tag=as679b5fe7.
  To: &lt;sip:1000@223.XX.130.50:40764&gt;;tag=coOV3rP.
  Contact: &lt;sip:2000@10.8.21.XX:5060&gt;.
  Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
  CSeq: 103 ACK.
  User-Agent: Asterisk PBX 11.8.1.
  Content-Length: 0.
  .
  
  #
  U 2014/04/15 00:22:51.355619 10.8.21.XX:5060 -&gt; 223.XX.130.50:40764
  BYE sip:1000@223.XX.130.50:40764 SIP/2.0.
  Via: SIP/2.0/UDP 10.8.21.XX:5060;branch=z9hG4bK0ac3adc4;rport.
  Max-Forwards: 70.
  From: &lt;sip:2000@10.8.21.XX&gt;;tag=as679b5fe7.
  To: &lt;sip:1000@223.XX.130.50:40764&gt;;tag=coOV3rP.
  Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
  CSeq: 104 BYE.
  User-Agent: Asterisk PBX 11.8.1.
  X-Asterisk-HangupCause: Normal Clearing.
  X-Asterisk-HangupCauseCode: 16.
  Content-Length: 0.
  .
  
  #
  U 2014/04/15 00:22:51.408414 XX.53.122.134:5060 -&gt; 10.8.21.XX:5060
  SIP/2.0 100 Trying.
  Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK348a4dc2;rport.
  From: &lt;sip:1000@sipdomain.com&gt;;tag=as1ba98ffc.
  To: &lt;sip:2000@sipdomain.com&gt;;tag=dGlp5o0FS.
  Call-ID: wh8Ai1e~0c.
  CSeq: 102 INVITE.
  .
  
  #
  U 2014/04/15 00:22:51.408837 XX.53.122.134:5060 -&gt; 10.8.21.XX:5060
  SIP/2.0 200 Ok.
  Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK348a4dc2;rport.
  From: &lt;sip:1000@sipdomain.com&gt;;tag=as1ba98ffc.
  To: &lt;sip:2000@sipdomain.com&gt;;tag=dGlp5o0FS.
  Call-ID: wh8Ai1e~0c.
  CSeq: 102 INVITE.
  User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4).
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO.
  Contact: &lt;sip:2000@XX.53.122.134&gt;;+sip.instance="&lt;urn:uuid:0b49a090-f01c-41a2-b771-bdc956e9b516&gt;".
  Content-Type: application/sdp.
  Content-Length: 170.
  .
  v=0.
  o=2000 274 61 IN IP4 192.168.0.38.
  s=Talk.
  c=IN IP4 192.168.0.38.
  b=AS:380.
  t=0 0.
  m=audio 7076 RTP/AVP 0 8 101.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-11.
  
  #
  U 2014/04/15 00:22:51.409343 10.8.21.XX:5060 -&gt; XX.53.122.134:5060
  ACK sip:2000@XX.53.122.134 SIP/2.0.
  Via: SIP/2.0/UDP 10.8.21.XX:5060;branch=z9hG4bK04d7bdd5;rport.
  Max-Forwards: 70.
  From: sip:1000@sipdomain.com;tag=as1ba98ffc.
  To: &lt;sip:2000@sipdomain.com&gt;;tag=dGlp5o0FS.
  Contact: &lt;sip:1000@10.8.21.XX:5060&gt;.
  Call-ID: wh8Ai1e~0c.
  CSeq: 102 ACK.
  User-Agent: Asterisk PBX 11.8.1.
  Content-Length: 0.
  .
  
  #
  U 2014/04/15 00:22:51.409471 10.8.21.XX:5060 -&gt; XX.53.122.134:5060
  BYE sip:2000@XX.53.122.134 SIP/2.0.
  Via: SIP/2.0/UDP 10.8.21.XX:5060;branch=z9hG4bK1b9de0d9;rport.
  Max-Forwards: 70.
  From: sip:1000@sipdomain.com;tag=as1ba98ffc.
  To: &lt;sip:2000@sipdomain.com&gt;;tag=dGlp5o0FS.
  Call-ID: wh8Ai1e~0c.
  CSeq: 103 BYE.
  User-Agent: Asterisk PBX 11.8.1.
  X-Asterisk-HangupCause: No user responding.
  X-Asterisk-HangupCauseCode: 18.
  Content-Length: 0.
  .
  
  #
  U 2014/04/15 00:22:51.453121 223.XX.130.50:40764 -&gt; 10.8.21.XX:5060
  SIP/2.0 200 Ok.
  Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK0ac3adc4;rport.
  From: &lt;sip:2000@10.8.21.XX&gt;;tag=as679b5fe7.
  To: &lt;sip:1000@223.XX.130.50:40764&gt;;tag=coOV3rP.
  Call-ID: 2dd1cc443fb8529e284d2b2412ef8f89@10.8.21.XX:5060.
  CSeq: 104 BYE.
  User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4).
  .
  
  #
  U 2014/04/15 00:22:51.495263 XX.53.122.134:5060 -&gt; 10.8.21.XX:5060
  SIP/2.0 200 Ok.
  Via: SIP/2.0/UDP 10.8.21.XX:5060;received=117.52.91.12;branch=z9hG4bK1b9de0d9;rport.
  From: &lt;sip:1000@sipdomain.com&gt;;tag=as1ba98ffc.
  To: &lt;sip:2000@sipdomain.com&gt;;tag=dGlp5o0FS.
  Call-ID: wh8Ai1e~0c.
  CSeq: 103 BYE.
  User-Agent: LinphoneAndroid/2.2.1.1 (belle-sip/1.2.4).
  .
  
  exit
  37 received, 0 dropped
  </code></pre>
  
  <p>Thank you.</p>