title: Why asterisk not properly working with android sip client?
body: <p>Asterisk= 1.8.11.0</p>
  
  <p>Android= 2.3/4.0.3</p>
  
  <p>Android Sip client=Native Android sip client/sipdemo</p>
  
  <p>When i call from my pc using zoiper/xlite to android (native android sip client) now i can hear audio from both sides but when i make call from android to pc (zoiper/xlite) i cannot hear anything on android.
  On the other hand i have tested this scenario on elastix (which also uses asterisk 1.8.11.0) with no problem in audio.
  pc(zoiper) ip 192.168.15.27
  android ip 192.168.15.71
  asterisk server ip 192.168.15.118</p>
  
  <p><strong>Sip debug when calling from android to zoiper .</strong></p>
  
  <pre><code>&lt;--- SIP read from UDP:192.168.15.71:45616 ---&gt;
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK5996b0d9;rport=5060;received=192.168.15.118
  From: "asterisk" &lt;sip:asterisk@192.168.15.118&gt;;tag=as05233e7d
  To: &lt;sip:211@192.168.15.71:45616;transport=udp&gt;
  Call-ID: 14151c2d29c039983aa449b56ce419e0@192.168.15.118:5060
  CSeq: 102 OPTIONS
  Content-Length: 0
  
  &lt;-------------&gt;
  --- (7 headers 0 lines) ---
  Really destroying SIP dialog '14151c2d29c039983aa449b56ce419e0@192.168.15.118:5060' Method: OPTIONS
  
  &lt;--- SIP read from UDP:192.168.15.71:45616 ---&gt;
  OPTIONS sip:192.168.15.118 SIP/2.0
  Call-ID: 5f7229b44e43a9e0eeaccabafd9a1e4f@192.168.15.71
  CSeq: 7757 OPTIONS
  From: "211" &lt;sip:211@192.168.15.118&gt;;tag=1758376458
  To: "211" &lt;sip:211@192.168.15.118&gt;
  Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK6c57f04e92e9e34153367ca0e6e29675353134;rport
  Max-Forwards: 70
  User-Agent: SIPAUA/0.1.001
  Content-Length: 0
  
  &lt;-------------&gt;
  --- (9 headers 0 lines) ---
  Looking for s in default (domain 192.168.15.118)
  
  &lt;--- Transmitting (NAT) to 192.168.15.71:45616 ---&gt;
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK6c57f04e92e9e34153367ca0e6e29675353134;received=192.168.15.71;rport=45616
  From: "211" &lt;sip:211@192.168.15.118&gt;;tag=1758376458
  To: "211" &lt;sip:211@192.168.15.118&gt;;tag=as6a8e1b47
  Call-ID: 5f7229b44e43a9e0eeaccabafd9a1e4f@192.168.15.71
  CSeq: 7757 OPTIONS
  Server: Asterisk PBX 1.8.11.0
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  Supported: replaces, timer
  Contact: &lt;sip:192.168.15.118:5060&gt;
  Accept: application/sdp
  Content-Length: 0
  
  
  &lt;--- SIP read from UDP:192.168.15.71:45616 ---&gt;
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1ecea84c;rport=5060;received=192.168.15.118
  From: "asterisk" &lt;sip:asterisk@192.168.15.118&gt;;tag=as167765df
  To: &lt;sip:211@192.168.15.71:45616;transport=udp&gt;
  Call-ID: 5f4d210800aa07065fd35fba215e9783@192.168.15.118:5060
  CSeq: 102 OPTIONS
  Content-Length: 0
  
  &lt;-------------&gt;
  --- (7 headers 0 lines) ---
  Really destroying SIP dialog '5f4d210800aa07065fd35fba215e9783@192.168.15.118:5060' Method: OPTIONS
  Really destroying SIP dialog '5e5f98ad4818911a86d4b438d054e39f@192.168.15.71' Method: OPTIONS
  Reliably Transmitting (NAT) to 192.168.15.71:45616:
  OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
  Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK15f7593b;rport
  Max-Forwards: 70
  From: "asterisk" &lt;sip:asterisk@192.168.15.118&gt;;tag=as53340ecf
  To: &lt;sip:211@192.168.15.71:45616;transport=udp&gt;
  Contact: &lt;sip:asterisk@192.168.15.118:5060&gt;
  Call-ID: 3c9da88024211f4b2ad1f3af2b2acd73@192.168.15.118:5060
  CSeq: 102 OPTIONS
  User-Agent: Asterisk PBX 1.8.11.0
  Date: Sat, 12 Jan 2013 19:44:34 GMT
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  Supported: replaces, timer
  Content-Length: 0
  
  
  ---
  
  &lt;--- SIP read from UDP:192.168.15.71:45616 ---&gt;
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK15f7593b;rport=5060;received=192.168.15.118
  From: "asterisk" &lt;sip:asterisk@192.168.15.118&gt;;tag=as53340ecf
  To: &lt;sip:211@192.168.15.71:45616;transport=udp&gt;
  Call-ID: 3c9da88024211f4b2ad1f3af2b2acd73@192.168.15.118:5060
  CSeq: 102 OPTIONS
  Content-Length: 0
  
  &lt;-------------&gt;
  --- (7 headers 0 lines) ---
  Really destroying SIP dialog '3c9da88024211f4b2ad1f3af2b2acd73@192.168.15.118:5060' Method: OPTIONS
  
  &lt;--- SIP read from UDP:192.168.15.71:45616 ---&gt;
  BYE sip:215@192.168.15.118:5060 SIP/2.0
  Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bKad00f5add2a1f2e940a91e9a3f827b26353134
  CSeq: 5511 BYE
  From: "211" &lt;sip:211@192.168.15.118&gt;;tag=2465683119
  To: &lt;sip:215@192.168.15.118&gt;;tag=as573c52b3
  Call-ID: d188757cd6c044783fd413d11f8a982f@192.168.15.71
  Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
  Supported: replaces,timer
  Max-Forwards: 70
  Content-Length: 0
  
  &lt;-------------&gt;
  --- (10 headers 0 lines) ---
  Sending to 192.168.15.71:45616 (NAT)
  Scheduling destruction of SIP dialog 'd188757cd6c044783fd413d11f8a982f@192.168.15.71' in 6400 ms (Method: BYE)
  
  &lt;--- Transmitting (NAT) to 192.168.15.71:45616 ---&gt;
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bKad00f5add2a1f2e940a91e9a3f827b26353134;received=192.168.15.71;rport=45616
  From: "211" &lt;sip:211@192.168.15.118&gt;;tag=2465683119
  To: &lt;sip:215@192.168.15.118&gt;;tag=as573c52b3
  Call-ID: d188757cd6c044783fd413d11f8a982f@192.168.15.71
  CSeq: 5511 BYE
  Server: Asterisk PBX 1.8.11.0
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  Supported: replaces, timer
  Content-Length: 0
  &lt;------------&gt;
  Scheduling destruction of SIP dialog '0d06690561757f98637241394cc8dbba@192.168.15.118:5060' in 6400 ms (Method: INVITE)
  set_destination: Parsing &lt;sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP&gt; for address/port to send to
  set_destination: set destination to 115.167.21.82:5060
  Reliably Transmitting (NAT) to 192.168.15.27:5060:
  BYE sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP SIP/2.0
  Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1500d2f6;rport
  Max-Forwards: 70
  From: "device" &lt;sip:211@192.168.15.118&gt;;tag=as404f0eb0
  To: &lt;sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP&gt;;tag=96055240
  Call-ID: 0d06690561757f98637241394cc8dbba@192.168.15.118:5060
  CSeq: 103 BYE
  User-Agent: Asterisk PBX 1.8.11.0
  X-Asterisk-HangupCause: Normal Clearing
  X-Asterisk-HangupCauseCode: 16
  Content-Length: 0
  ---
  == Spawn extension (incoming-calls-wildcard, 215, 1) exited non-zero on 'SIP/211-00000008'
  Retransmitting #1 (NAT) to 192.168.15.27:5060:
  BYE sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP SIP/2.0
  Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1500d2f6;rport
  Max-Forwards: 70
  From: "device" &lt;sip:211@192.168.15.118&gt;;tag=as404f0eb0
  To: &lt;sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP&gt;;tag=96055240
  Call-ID: 0d06690561757f98637241394cc8dbba@192.168.15.118:5060
  CSeq: 103 BYE
  User-Agent: Asterisk PBX 1.8.11.0
  X-Asterisk-HangupCause: Normal Clearing
  X-Asterisk-HangupCauseCode: 16
  Content-Length: 0
  ---
  &lt;--- SIP read from UDP:192.168.15.27:5060 ---&gt;
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1500d2f6;rport=5060
  Contact: &lt;sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP&gt;
  To: &lt;sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP&gt;;tag=96055240
  From: "device"&lt;sip:211@192.168.15.118&gt;;tag=as404f0eb0
  Call-ID: 0d06690561757f98637241394cc8dbba@192.168.15.118:5060
  CSeq: 103 BYE
  User-Agent: Zoiper for Windows 2.39 r16838
  Content-Length: 0
  &lt;-------------&gt;
  -- (9 headers 0 lines) ---
  Really destroying SIP dialog '0d06690561757f98637241394cc8dbba@192.168.15.118:5060' Method: INVITE
  
  &lt;--- SIP read from UDP:192.168.15.27:5060 ---&gt;
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1500d2f6;rport=5060
  Contact: &lt;sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP&gt;
  To: &lt;sip:215@115.167.21.82:5060;rinstance=49cb21467969bef8;transport=UDP&gt;;tag=96055240
  From: "device"&lt;sip:211@192.168.15.118&gt;;tag=as404f0eb0
  Call-ID: 0d06690561757f98637241394cc8dbba@192.168.15.118:5060
  CSeq: 103 BYE
  User-Agent: Zoiper for Windows 2.39 r16838
  Content-Length: 0
  
  &lt;-------------&gt;
  -- (9 headers 0 lines) ---
  Reliably Transmitting (NAT) to 192.168.15.71:45616:
  OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
  Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK30d9d70f;rport
  Max-Forwards: 70
  From: "asterisk" &lt;sip:asterisk@192.168.15.118&gt;;tag=as4f0724aa
  To: &lt;sip:211@192.168.15.71:45616;transport=udp&gt;
  Contact: &lt;sip:asterisk@192.168.15.118:5060&gt;
  Call-ID: 05f27bf10f0a0759637b90252dc4366a@192.168.15.118:5060
  CSeq: 102 OPTIONS
  User-Agent: Asterisk PBX 1.8.11.0
  Date: Sat, 12 Jan 2013 19:44:37 GMT
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  Supported: replaces, timer
  Content-Length: 0
  
  
  ---
  
  &lt;--- SIP read from UDP:192.168.15.71:45616 ---&gt;
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK30d9d70f;rport=5060;received=192.168.15.118
  From: "asterisk" &lt;sip:asterisk@192.168.15.118&gt;;tag=as4f0724aa
  To: &lt;sip:211@192.168.15.71:45616;transport=udp&gt;
  Call-ID: 05f27bf10f0a0759637b90252dc4366a@192.168.15.118:5060
  CSeq: 102 OPTIONS
  Content-Length: 0
  
  &lt;-------------&gt;
  --- (7 headers 0 lines) ---
  Really destroying SIP dialog '05f27bf10f0a0759637b90252dc4366a@192.168.15.118:5060' Method: OPTIONS
  
  &lt;--- SIP read from UDP:192.168.15.71:45616 ---&gt;
  OPTIONS sip:192.168.15.118 SIP/2.0
  Call-ID: a5a311df861221d42844a8c485d4fee8@192.168.15.71
  CSeq: 5815 OPTIONS
  From: "211" &lt;sip:211@192.168.15.118&gt;;tag=3109248316
  To: "211" &lt;sip:211@192.168.15.118&gt;
  Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK1e09e6f6c30b4b74ddc93c590abd3383353134;rport
  Max-Forwards: 70
  User-Agent: SIPAUA/0.1.001
  Content-Length: 0
  
  &lt;-------------&gt;
  --- (9 headers 0 lines) ---
  Looking for s in default (domain 192.168.15.118)
  
  &lt;--- Transmitting (NAT) to 192.168.15.71:45616 ---&gt;
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK1e09e6f6c30b4b74ddc93c590abd3383353134;received=192.168.15.71;rport=45616
  From: "211" &lt;sip:211@192.168.15.118&gt;;tag=3109248316
  To: "211" &lt;sip:211@192.168.15.118&gt;;tag=as51223faf
  Call-ID: a5a311df861221d42844a8c485d4fee8@192.168.15.71
  CSeq: 5815 OPTIONS
  Server: Asterisk PBX 1.8.11.0
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  Supported: replaces, timer
  Contact: &lt;sip:192.168.15.118:5060&gt;
  Accept: application/sdp
  Content-Length: 0
  
  
  &lt;------------&gt;
  Scheduling destruction of SIP dialog 'a5a311df861221d42844a8c485d4fee8@192.168.15.71' in 32000 ms (Method: OPTIONS)
  Reliably Transmitting (NAT) to 192.168.15.71:45616:
  OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
  Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1f8e44dc;rport
  Max-Forwards: 70
  From: "asterisk" &lt;sip:asterisk@192.168.15.118&gt;;tag=as7a9a1ea3
  To: &lt;sip:211@192.168.15.71:45616;transport=udp&gt;
  Contact: &lt;sip:asterisk@192.168.15.118:5060&gt;
  Call-ID: 7ebcafc7159379fd047075a85c424588@192.168.15.118:5060
  CSeq: 102 OPTIONS
  User-Agent: Asterisk PBX 1.8.11.0
  Date: Sat, 12 Jan 2013 19:44:41 GMT
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  Supported: replaces, timer
  Content-Length: 0
  
  
  ---
  
  &lt;--- SIP read from UDP:192.168.15.71:45616 ---&gt;
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK1f8e44dc;rport=5060;received=192.168.15.118
  From: "asterisk" &lt;sip:asterisk@192.168.15.118&gt;;tag=as7a9a1ea3
  To: &lt;sip:211@192.168.15.71:45616;transport=udp&gt;
  Call-ID: 7ebcafc7159379fd047075a85c424588@192.168.15.118:5060
  CSeq: 102 OPTIONS
  Content-Length: 0
  
  &lt;-------------&gt;
  --- (7 headers 0 lines) ---
  Really destroying SIP dialog '7ebcafc7159379fd047075a85c424588@192.168.15.118:5060' Method: OPTIONS
  Really destroying SIP dialog 'd188757cd6c044783fd413d11f8a982f@192.168.15.71' Method: BYE
  Really destroying SIP dialog 'a81e6a5f591141abd73f9dad478a6b56@192.168.15.71' Method: OPTIONS
  Reliably Transmitting (NAT) to 192.168.15.71:45616:
  OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
  Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK017f2b0a;rport
  Max-Forwards: 70
  From: "asterisk" &lt;sip:asterisk@192.168.15.118&gt;;tag=as5367b37c
  To: &lt;sip:211@192.168.15.71:45616;transport=udp&gt;
  Contact: &lt;sip:asterisk@192.168.15.118:5060&gt;
  Call-ID: 1f5a66a10c6dc1e4126a681d2001b173@192.168.15.118:5060
  CSeq: 102 OPTIONS
  User-Agent: Asterisk PBX 1.8.11.0
  Date: Sat, 12 Jan 2013 19:44:44 GMT
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  Supported: replaces, timer
  Content-Length: 0
  
  ---
  &lt;--- SIP read from UDP:192.168.15.71:45616 ---&gt;
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK017f2b0a;rport=5060;received=192.168.15.118
  From: "asterisk" &lt;sip:asterisk@192.168.15.118&gt;;tag=as5367b37c
  To: &lt;sip:211@192.168.15.71:45616;transport=udp&gt;
  Call-ID: 1f5a66a10c6dc1e4126a681d2001b173@192.168.15.118:5060
  CSeq: 102 OPTIONS
  Content-Length: 0
  
  &lt;-------------&gt;
  --- (7 headers 0 lines) ---
  Really destroying SIP dialog '1f5a66a10c6dc1e4126a681d2001b173@192.168.15.118:5060' Method: OPTIONS
  </code></pre>
  
  <p><strong>When calling from pc (zoiper) to android</strong></p>
  
  <pre><code>&lt;--- SIP read from UDP:192.168.15.71:45616 ---&gt;
  BYE sip:215@192.168.15.118:5060 SIP/2.0
  Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bKe2f448b90c482342407298e309be5aed353134
  CSeq: 1 BYE
  From: &lt;sip:211@192.168.15.71:45616;transport=udp&gt;;tag=4162167884
  To: "device" &lt;sip:215@192.168.15.118&gt;;tag=as5805dc66
  Call-ID: 2732e4564ce8534c5765a456045a9960@192.168.15.118:5060
  Max-Forwards: 70
  Content-Length: 0
  
  &lt;-------------&gt;
  --- (8 headers 0 lines) ---
  Sending to 192.168.15.71:45616 (NAT)
  Scheduling destruction of SIP dialog '2732e4564ce8534c5765a456045a9960@192.168.15.118:5060' in 8576 ms (Method: BYE)
  
  &lt;--- Transmitting (NAT) to 192.168.15.71:45616 ---&gt;
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bKe2f448b90c482342407298e309be5aed353134;received=192.168.15.71;rport=45616
  From: &lt;sip:211@192.168.15.71:45616;transport=udp&gt;;tag=4162167884
  To: "device" &lt;sip:215@192.168.15.118&gt;;tag=as5805dc66
  Call-ID: 2732e4564ce8534c5765a456045a9960@192.168.15.118:5060
  CSeq: 1 BYE
  Server: Asterisk PBX 1.8.11.0
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  Supported: replaces, timer
  Content-Length: 0
  
  
  &lt;------------&gt;
  == Spawn extension (incoming-calls-wildcard, 211, 1) exited non-zero on 'SIP/215-0000000a'
  Scheduling destruction of SIP dialog 'MDhhMTg1YjllYTUwZThlMmNiZTQwYTMzODYyZmEzNjk.' in 6400 ms (Method: ACK)
  set_destination: Parsing &lt;sip:215@115.167.21.82:5060;transport=UDP&gt; for address/port to send to
  set_destination: set destination to 115.167.21.82:5060
  Reliably Transmitting (NAT) to 192.168.15.27:5060:
  BYE sip:215@115.167.21.82:5060;transport=UDP SIP/2.0
  Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK498da08b;rport
  Max-Forwards: 70
  From: &lt;sip:211@192.168.15.118;transport=UDP&gt;;tag=as10377813
  To: &lt;sip:215@192.168.15.118;transport=UDP&gt;;tag=50312112
  Call-ID: MDhhMTg1YjllYTUwZThlMmNiZTQwYTMzODYyZmEzNjk.
  CSeq: 102 BYE
  User-Agent: Asterisk PBX 1.8.11.0
  Proxy-Authorization: Digest username="215", realm="asterisk", algorithm=MD5, uri="sip:192.168.15.118", nonce="", response="c897390cc8e4f674d7e9cd1efa7319a6"
  X-Asterisk-HangupCause: Normal Clearing
  X-Asterisk-HangupCauseCode: 16
  Content-Length: 0
  
  
  ---
  Retransmitting #1 (NAT) to 192.168.15.27:5060:
  BYE sip:215@115.167.21.82:5060;transport=UDP SIP/2.0
  Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK498da08b;rport
  Max-Forwards: 70
  From: &lt;sip:211@192.168.15.118;transport=UDP&gt;;tag=as10377813
  To: &lt;sip:215@192.168.15.118;transport=UDP&gt;;tag=50312112
  Call-ID: MDhhMTg1YjllYTUwZThlMmNiZTQwYTMzODYyZmEzNjk.
  CSeq: 102 BYE
  User-Agent: Asterisk PBX 1.8.11.0
  Proxy-Authorization: Digest username="215", realm="asterisk", algorithm=MD5, uri="sip:192.168.15.118", nonce="", response="c897390cc8e4f674d7e9cd1efa7319a6"
  X-Asterisk-HangupCause: Normal Clearing
  X-Asterisk-HangupCauseCode: 16
  Content-Length: 0
  
  
  ---
  
  &lt;--- SIP read from UDP:192.168.15.27:5060 ---&gt;
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK498da08b;rport=5060
  Contact: &lt;sip:215@115.167.21.82:5060;transport=UDP&gt;
  To: &lt;sip:215@192.168.15.118;transport=UDP&gt;;tag=50312112
  From: &lt;sip:211@192.168.15.118;transport=UDP&gt;;tag=as10377813
  Call-ID: MDhhMTg1YjllYTUwZThlMmNiZTQwYTMzODYyZmEzNjk.
  CSeq: 102 BYE
  User-Agent: Zoiper for Windows 2.39 r16838
  Content-Length: 0
  
  &lt;-------------&gt;
  --- (9 headers 0 lines) ---
  SIP Response message for INCOMING dialog BYE arrived
  Really destroying SIP dialog 'MDhhMTg1YjllYTUwZThlMmNiZTQwYTMzODYyZmEzNjk.' Method: ACK
  
  &lt;--- SIP read from UDP:192.168.15.27:5060 ---&gt;
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK498da08b;rport=5060
  Contact: &lt;sip:215@115.167.21.82:5060;transport=UDP&gt;
  To: &lt;sip:215@192.168.15.118;transport=UDP&gt;;tag=50312112
  From: &lt;sip:211@192.168.15.118;transport=UDP&gt;;tag=as10377813
  Call-ID: MDhhMTg1YjllYTUwZThlMmNiZTQwYTMzODYyZmEzNjk.
  CSeq: 102 BYE
  User-Agent: Zoiper for Windows 2.39 r16838
  Content-Length: 0
  
  &lt;-------------&gt;
  --- (9 headers 0 lines) ---
  Reliably Transmitting (NAT) to 192.168.15.71:45616:
  OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
  Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK01e7e47d;rport
  Max-Forwards: 70
  From: "asterisk" &lt;sip:asterisk@192.168.15.118&gt;;tag=as73902c1e
  To: &lt;sip:211@192.168.15.71:45616;transport=udp&gt;
  Contact: &lt;sip:asterisk@192.168.15.118:5060&gt;
  Call-ID: 7b77192d5d07c4bf5d0a82ab2c11c3c1@192.168.15.118:5060
  CSeq: 102 OPTIONS
  User-Agent: Asterisk PBX 1.8.11.0
  Date: Sat, 12 Jan 2013 19:54:09 GMT
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  Supported: replaces, timer
  Content-Length: 0
  
  
  ---
  
  &lt;--- SIP read from UDP:192.168.15.71:45616 ---&gt;
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK01e7e47d;rport=5060;received=192.168.15.118
  From: "asterisk" &lt;sip:asterisk@192.168.15.118&gt;;tag=as73902c1e
  To: &lt;sip:211@192.168.15.71:45616;transport=udp&gt;
  Call-ID: 7b77192d5d07c4bf5d0a82ab2c11c3c1@192.168.15.118:5060
  CSeq: 102 OPTIONS
  Content-Length: 0
  
  &lt;-------------&gt;
  --- (7 headers 0 lines) ---
  Really destroying SIP dialog '7b77192d5d07c4bf5d0a82ab2c11c3c1@192.168.15.118:5060' Method: OPTIONS
  
  &lt;--- SIP read from UDP:192.168.15.71:45616 ---&gt;
  OPTIONS sip:192.168.15.118 SIP/2.0
  Call-ID: c51761a244f0dc9639f87a5cf9af1b3f@192.168.15.71
  CSeq: 9273 OPTIONS
  From: "211" &lt;sip:211@192.168.15.118&gt;;tag=740019322
  To: "211" &lt;sip:211@192.168.15.118&gt;
  Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK5ac51b5cca5ad84320fc342692fdbb2d353134;rport
  Max-Forwards: 70
  User-Agent: SIPAUA/0.1.001
  Content-Length: 0
  
  &lt;-------------&gt;
  --- (9 headers 0 lines) ---
  Looking for s in default (domain 192.168.15.118)
  
  &lt;--- Transmitting (NAT) to 192.168.15.71:45616 ---&gt;
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK5ac51b5cca5ad84320fc342692fdbb2d353134;received=192.168.15.71;rport=45616
  From: "211" &lt;sip:211@192.168.15.118&gt;;tag=740019322
  To: "211" &lt;sip:211@192.168.15.118&gt;;tag=as1bed6ef2
  Call-ID: c51761a244f0dc9639f87a5cf9af1b3f@192.168.15.71
  CSeq: 9273 OPTIONS
  Server: Asterisk PBX 1.8.11.0
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  Supported: replaces, timer
  Contact: &lt;sip:192.168.15.118:5060&gt;
  Accept: application/sdp
  Content-Length: 0
  
  &lt;--- SIP read from UDP:192.168.15.71:45616 ---&gt;
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK3c24b1cb;rport=5060;received=192.168.15.118
  From: "asterisk" &lt;sip:asterisk@192.168.15.118&gt;;tag=as54c6581a
  To: &lt;sip:211@192.168.15.71:45616;transport=udp&gt;
  Call-ID: 7742af6c24e3e9a33efa0ba73051a29e@192.168.15.118:5060
  CSeq: 102 OPTIONS
  Content-Length: 0
  
  &lt;-------------&gt;
  --- (7 headers 0 lines) ---
  Really destroying SIP dialog '7742af6c24e3e9a33efa0ba73051a29e@192.168.15.118:5060' Method: OPTIONS
  
  &lt;--- SIP read from UDP:192.168.15.71:45616 ---&gt;
  OPTIONS sip:192.168.15.118 SIP/2.0
  Call-ID: 18347a0db6841423591b08250847a1e0@192.168.15.71
  CSeq: 3824 OPTIONS
  From: "211" &lt;sip:211@192.168.15.118&gt;;tag=841349553
  To: "211" &lt;sip:211@192.168.15.118&gt;
  Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK79bb575a5d0f832291b861987849d9a2353134;rport
  Max-Forwards: 70
  User-Agent: SIPAUA/0.1.001
  Content-Length: 0
  
  &lt;-------------&gt;
  --- (9 headers 0 lines) ---
  Looking for s in default (domain 192.168.15.118)
  
  
  &lt;-------------&gt;
  --- (9 headers 0 lines) ---
  Looking for s in default (domain 192.168.15.118)
  
  &lt;--- Transmitting (NAT) to 192.168.15.71:45616 ---&gt;
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 192.168.15.71:45616;branch=z9hG4bK40d81d1686d23c9133e52f4180c2accb353134;received=192.168.15.71;rport=45616
  From: "211" &lt;sip:211@192.168.15.118&gt;;tag=4017391219
  To: "211" &lt;sip:211@192.168.15.118&gt;;tag=as52fe1845
  Call-ID: 09809b7ed9a68b7f270f33f1d8de13dc@192.168.15.71
  CSeq: 4619 OPTIONS
  Server: Asterisk PBX 1.8.11.0
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  Supported: replaces, timer
  Contact: &lt;sip:192.168.15.118:5060&gt;
  Accept: application/sdp
  Content-Length: 0
  
  
  &lt;------------&gt;
  Scheduling destruction of SIP dialog '09809b7ed9a68b7f270f33f1d8de13dc@192.168.15.71' in 32000 ms (Method: OPTIONS)
  Reliably Transmitting (NAT) to 192.168.15.71:45616:
  OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
  Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK13b8cca8;rport
  Max-Forwards: 70
  From: "asterisk" &lt;sip:asterisk@192.168.15.118&gt;;tag=as6e6638f8
  To: &lt;sip:211@192.168.15.71:45616;transport=udp&gt;
  Contact: &lt;sip:asterisk@192.168.15.118:5060&gt;
  Call-ID: 76f6f373769041c3462ebb676641f1b7@192.168.15.118:5060
  CSeq: 102 OPTIONS
  User-Agent: Asterisk PBX 1.8.11.0
  Date: Sat, 12 Jan 2013 19:54:31 GMT
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  Supported: replaces, timer
  Content-Length: 0
  
  
  ---
  
  &lt;--- SIP read from UDP:192.168.15.71:45616 ---&gt;
  SIP/2.0 200 OK
  Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK13b8cca8;rport=5060;received=192.168.15.118
  From: "asterisk" &lt;sip:asterisk@192.168.15.118&gt;;tag=as6e6638f8
  To: &lt;sip:211@192.168.15.71:45616;transport=udp&gt;
  Call-ID: 76f6f373769041c3462ebb676641f1b7@192.168.15.118:5060
  CSeq: 102 OPTIONS
  Content-Length: 0
  
  &lt;-------------&gt;
  --- (7 headers 0 lines) ---
  Really destroying SIP dialog '76f6f373769041c3462ebb676641f1b7@192.168.15.118:5060' Method: OPTIONS
  Really destroying SIP dialog '9eeee094f46eec920ac462e291314bde@192.168.15.71' Method: OPTIONS
  Reliably Transmitting (NAT) to 192.168.15.71:45616:
  OPTIONS sip:211@192.168.15.71:45616;transport=udp SIP/2.0
  Via: SIP/2.0/UDP 192.168.15.118:5060;branch=z9hG4bK47a8a134;rport
  Max-Forwards: 70
  From: "asterisk" &lt;sip:asterisk@192.168.15.118&gt;;tag=as76426de6
  To: &lt;sip:211@192.168.15.71:45616;transport=udp&gt;
  Contact: &lt;sip:asterisk@192.168.15.118:5060&gt;
  Call-ID: 3a98a25b41dc3b3e699ee4383669e984@192.168.15.118:5060
  CSeq: 102 OPTIONS
  User-Agent: Asterisk PBX 1.8.11.0
  Date: Sat, 12 Jan 2013 19:54:34 GMT
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
  Supported: replaces, timer
  Content-Length: 0
  </code></pre>
  
  <p>I am using asterisk on local network (LAN)....</p>
  
  <p><strong>My dial plan in extensions.conf is:</strong></p>
  
  <pre><code>[incoming-calls-wildcard]
  
  exten =&gt; _2XX,hint,(SIP/${EXTEN},,120)
  
  exten =&gt; _2XX,1,Dial(SIP/${EXTEN},,120)
  
  exten =&gt; _2XX,n,Hangup
  </code></pre>
  
  <p><strong>My sip account is:</strong></p>
  
  <pre><code>[215]
  
  deny=0.0.0.0/0.0.0.0
  
  secret=very123
  
  dtmfmode=rfc2833
  
  canreinvite=no
  
  context=incoming-calls-wildcard
  
  host=dynamic
  
  type=friend
  
  nat=yes
  
  port=5060
  
  qualify=yes
  
  callgroup=
  
  pickupgroup=
  
  dial=SIP/215
  
  mailbox=215@device
  
  permit=0.0.0.0/0.0.0.0
  
  callerid=device &lt;215&gt;
  
  callcounter=yes
  
  faxdetect=no
  </code></pre>