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  1. POSIP Callee does not get notification that call ended
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    <p>I have deleted my previous question and post this updated:</p> <p>I have a an issue with my SIP UAC, once I received a ringing from the B2BUA on both the caller and callee, and the caller hangs up the call while the call is ringing (I send cancel request and receive "request terminated" on the caller side), the callee does not get any notification that the call has been terminated by the caller.</p> <p>But when the callee declines the call, the caller gets a busy here.</p> <p>Here is the <strong>Callee</strong> side:</p> <pre><code>/----------------------- MEDIA SESSION ------------------------/ --- Multimedia-Session: Composed Audio --- 1. Media Session: "Audio" enabled=true States: [Disconnected] Capturers: (1 in total) Stream 1: audio device - DirectSoundCapture Formats: [PCMU/8000] Connection Details: My address: 10.0.0.2:52044 Participants: (1 in total) Address 1: HostAddress:17364 /-------------------- END OF MEDIA SESSION --------------------/ /------------------------- BEGINNING --------------------------/ -------------------------------- Request: Test 2--&gt;Me: INVITE#102 -------------------------------- INVITE sip:430@Host SIP/2.0 Via: SIP/2.0/UDP HostAddress:5060;branch=z9hG4bK1fd06834;rport=5060;received=HostAddress From: "Test 2" &lt;sip:410@HostAddress&gt;;tag=as2b22eddf To: &lt;sip:430@Host&gt; Contact: &lt;sip:410@HostAddress&gt; Call-ID: 35e0e8655b20ad886f137a0c0e563809@HostAddress CSeq: 102 INVITE User-Agent: Freeswitch 1.2.3 Max-Forwards: 70 Date: Sun, 11 Jul 2010 02:44:43 GMT Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO Supported: replaces Content-Type: application/sdp Content-Length: 264 v=0 o=root 27669 27669 IN IP4 HostAddress s=session c=IN IP4 HostAddress t=0 0 m=audio 17364 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=silenceSupp:off - - - - a=ptime:20 a=sendrecv ------------------------ Response: Me ==&gt; Test 2: INVITE#102: 180 Ringing ------------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP HostAddress:5060;branch=z9hG4bK1fd06834;rport=5060;received=HostAddress From: "Test 2" &lt;sip:410@HostAddress&gt;;tag=as2b22eddf To: &lt;sip:430@Host&gt;;tag=e125be76 Call-ID: 35e0e8655b20ad886f137a0c0e563809@HostAddress CSeq: 102 INVITE Content-Length: 0 ------------------------ Response: Me ==&gt; Test 2: INVITE#102: 603 Decline ------------------------ SIP/2.0 603 Decline Via: SIP/2.0/UDP HostAddress:5060;branch=z9hG4bK1fd06834;rport=5060;received=HostAddress From: "Test 2" &lt;sip:410@HostAddress&gt;;tag=as2b22eddf To: &lt;sip:430@Host&gt;;tag=e125be76 Call-ID: 35e0e8655b20ad886f137a0c0e563809@HostAddress CSeq: 102 INVITE Content-Length: 0 /---------------------------- END -----------------------------/ </code></pre> <p>I must decline at the callee end, because if I don't respond to the request, the callee account get stuck in a loop and then the client returns busy forever, and requests does not reach that client, or at least until the account is deleted.</p> <p>And there is another thing, the B2BUA does not send anything back to the decline response, shouldn't I get an ACK from the server? </p> <p>And here is the <strong>Caller</strong> side:</p> <pre><code>/----------------------- MEDIA SESSION ------------------------/ --- Multimedia-Session: Audio --- 1. Media Session: "Audio" enabled=true States: [Disconnected] Capturers: (1 in total) Stream 1: audio device - DirectSoundCapture Formats: [PCMU/8000] [GSM/8000] [G723/8000] [DVI4/8000] [MPA/-1] [DVI4/11025] [DVI4/22050] Connection Details: My address: Participants: (0 in total) /-------------------- END OF MEDIA SESSION --------------------/ /------------------------- BEGINNING --------------------------/ -------------------------- Request: Client 410--&gt;Client 430: INVITE#81 -------------------------- INVITE sip:430@host SIP/2.0 Subject: Session Name: Nu-Art Software Via: SIP/2.0/UDP host:5060;branch=z9hG4bK4dd6bdf707a85fb5a73faec9ff648f703236 Contact: "Client 410" &lt;sip:410@host&gt; From: "Client 410" &lt;sip:410@host&gt;;tag=8f7b94cb To: "Client 430" &lt;sip:430@host&gt; Organization: Future Earth Max-Forwards: 32 CSeq: 81 INVITE Call-ID: 97ee019923d6a6d11a9476d71880e289@10.0.0.1 Allow: INVITE,ACK,CANCEL,BYE,OPTIONS Expires: 60 Content-Type: application/sdp Content-Length: 324 v=0 o=Client 410 699719 699719 IN IP4 MyAddress s=Audio i=Made by: Nu-Art Software 07-2010 c=IN IP4 MyAddress t=0 0 m=audio 2871 RTP/AVP 0 3 4 5 14 16 17 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:14 MPA/-1 a=rtpmap:16 DVI4/11025 a=rtpmap:17 DVI4/22050 ------- Response: Client 430 ==&gt; Client 410: INVITE#81: 407 Proxy Authentication Required ------- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP host:5060;branch=z9hG4bK4dd6bdf707a85fb5a73faec9ff648f703236;received=MyAddress From: "Client 410" &lt;sip:410@host&gt;;tag=8f7b94cb To: "Client 430" &lt;sip:430@host&gt;;tag=as78e28f4d Call-ID: 97ee019923d6a6d11a9476d71880e289@10.0.0.1 CSeq: 81 INVITE User-Agent: Freeswitch 1.2.3 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5,realm="asterisk",nonce="574b3d49" Content-Length: 0 ---------------------------- Request: Client 410--&gt;Client 430: ACK#81 ---------------------------- ACK sip:430@host SIP/2.0 Via: SIP/2.0/UDP host:5060;branch=z9hG4bK4dd6bdf707a85fb5a73faec9ff648f703236 From: "Client 410" &lt;sip:410@host&gt;;tag=8f7b94cb To: "Client 430" &lt;sip:430@host&gt; Max-Forwards: 32 CSeq: 81 ACK Call-ID: 97ee019923d6a6d11a9476d71880e289@10.0.0.1 Content-Length: 0 -------------------------- Request: Client 410--&gt;Client 430: INVITE#82 -------------------------- INVITE sip:430@host SIP/2.0 Subject: Session Name: Nu-Art Software Via: SIP/2.0/UDP host:5060;branch=z9hG4bK34c52041066f24c6ac4499af25a948b63236 Contact: "Client 410" &lt;sip:410@host&gt; From: "Client 410" &lt;sip:410@host&gt;;tag=8f7b94cb To: "Client 430" &lt;sip:430@host&gt;;tag=as78e28f4d Organization: Future Earth Max-Forwards: 32 CSeq: 82 INVITE Call-ID: 97ee019923d6a6d11a9476d71880e289@10.0.0.1 Allow: INVITE,ACK,CANCEL,BYE,OPTIONS Expires: 60 Content-Type: application/sdp Proxy-Authorization: Digest username="410",nonce="574b3d49",realm="asterisk",uri="sip:410@host",algorithm=MD5,response="e674e15de7b6dd05c7fe6da6c155befd" Content-Length: 324 v=0 o=Client 410 699719 699719 IN IP4 MyAddress s=Audio i=Made by: Nu-Art Software 07-2010 c=IN IP4 MyAddress t=0 0 m=audio 2871 RTP/AVP 0 3 4 5 14 16 17 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=rtpmap:5 DVI4/8000 a=rtpmap:14 MPA/-1 a=rtpmap:16 DVI4/11025 a=rtpmap:17 DVI4/22050 ------------------- Response: Client 430 ==&gt; Client 410: INVITE#82: 100 Trying ------------------- SIP/2.0 100 Trying Via: SIP/2.0/UDP host:5060;branch=z9hG4bK34c52041066f24c6ac4499af25a948b63236;received=MyAddress From: "Client 410" &lt;sip:410@host&gt;;tag=8f7b94cb To: "Client 430" &lt;sip:430@host&gt;;tag=as78e28f4d Call-ID: 97ee019923d6a6d11a9476d71880e289@10.0.0.1 CSeq: 82 INVITE User-Agent: Freeswitch 1.2.3 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO Supported: replaces Contact: &lt;sip:430@HostAddress&gt; Content-Length: 0 ------------------ Response: Client 430 ==&gt; Client 410: INVITE#82: 180 Ringing ------------------ SIP/2.0 180 Ringing Via: SIP/2.0/UDP host:5060;branch=z9hG4bK34c52041066f24c6ac4499af25a948b63236;received=MyAddress From: "Client 410" &lt;sip:410@host&gt;;tag=8f7b94cb To: "Client 430" &lt;sip:430@host&gt;;tag=as78e28f4d Call-ID: 97ee019923d6a6d11a9476d71880e289@10.0.0.1 CSeq: 82 INVITE User-Agent: Freeswitch 1.2.3 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO Supported: replaces Contact: &lt;sip:430@HostAddress&gt; Content-Length: 0 -------------------------- Request: Client 410--&gt;Client 430: CANCEL#82 -------------------------- CANCEL sip:430@host SIP/2.0 Call-ID: 97ee019923d6a6d11a9476d71880e289@10.0.0.1 To: "Client 430" &lt;sip:430@host&gt;;tag=as78e28f4d CSeq: 82 CANCEL From: "Client 410" &lt;sip:410@host&gt;;tag=8f7b94cb Via: SIP/2.0/UDP host:5060;branch=z9hG4bK34c52041066f24c6ac4499af25a948b63236 Max-Forwards: 32 Content-Length: 0 --------------------- Response: Client 430 ==&gt; Client 410: CANCEL#82: 200 OK --------------------- SIP/2.0 200 OK Via: SIP/2.0/UDP host:5060;branch=z9hG4bK34c52041066f24c6ac4499af25a948b63236;received=MyAddress From: "Client 410" &lt;sip:410@host&gt;;tag=8f7b94cb To: "Client 430" &lt;sip:430@host&gt;;tag=as78e28f4d Call-ID: 97ee019923d6a6d11a9476d71880e289@10.0.0.1 CSeq: 82 CANCEL User-Agent: Freeswitch 1.2.3 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO Supported: replaces Content-Length: 0 ------------- Response: Client 430 ==&gt; Client 410: INVITE#82: 487 Request Terminated ------------- SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP host:5060;branch=z9hG4bK34c52041066f24c6ac4499af25a948b63236;received=MyAddress From: "Client 410" &lt;sip:410@host&gt;;tag=8f7b94cb To: "Client 430" &lt;sip:430@host&gt;;tag=as78e28f4d Call-ID: 97ee019923d6a6d11a9476d71880e289@10.0.0.1 CSeq: 82 INVITE User-Agent: Freeswitch 1.2.3 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO Supported: replaces Content-Length: 0 ---------------------------- Request: Client 410--&gt;Client 430: ACK#82 ---------------------------- ACK sip:430@host SIP/2.0 Via: SIP/2.0/UDP host:5060;branch=z9hG4bK34c52041066f24c6ac4499af25a948b63236 From: "Client 410" &lt;sip:410@host&gt;;tag=8f7b94cb To: "Client 430" &lt;sip:430@host&gt; Max-Forwards: 32 CSeq: 82 ACK Call-ID: 97ee019923d6a6d11a9476d71880e289@10.0.0.1 Content-Length: 0 /---------------------------- END -----------------------------/ </code></pre> <p>Frank, I tried to pay attention to your details, perhaps I missed something, since the other side still does not receives notification on an early hang up.</p> <p>Any idea why?</p> <p>Thanks in advance,</p> <p>Adam.</p>
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