<div>Dear all,</div>
<div>I'm in basic setup of my network:</div>
<div> </div>
<div>I try to do a call from a softphone to an other one but I got the error 603 Declined.</div>
<div> </div>
<div>Below the</div>
<div>sip.conf:</div>
<div><em>[gianca]<br>type=friend<br>username=gianca<br>secret=pwd_gianca<br>host=dynamic<br>context=tutorial</em></div>
<div><em>[giusy]<br>type=friend<br>username=giusy<br>secret=pwd_giusy<br>host=dynamic<br>context=tutorial</em><br></div>
<div> </div>
<div> extension.conf:</div>
<div><em>[tutorial]<br>exten => 1234,1,Dial(SIP,gianca)</em></div>
<div><em>exten => 12345,1,Dial(SIP,giusy</em>)<br></div>
<div> </div>
<div>Below the output of SIP debug of IP caller (192.168.1.116) in asterisk</div>
<div> </div>
<div> </div>
<div><em>dhcppc0*CLI><br><--- SIP read from <a href="http://192.168.1.116:14862">192.168.1.116:14862</a> ---><br>INVITE <a href="mailto:sip%3A12345@192.168.1.100">sip:12345@192.168.1.100</a> SIP/2.0<br>Via: SIP/2.0/UDP 192.168.1.116:14862;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;rport<br>
Max-Forwards: 70<br>Contact: <<a href="http://sip:gianca@192.168.1.116:14862">sip:gianca@192.168.1.116:14862</a>><br>To: "12345"<<a href="mailto:sip%3A12345@192.168.1.100">sip:12345@192.168.1.100</a>><br>
From: "gianca"<<a href="mailto:sip%3Agianca@192.168.1.100">sip:gianca@192.168.1.100</a>>;tag=db428348<br>Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.<br>CSeq: 1 <strong>INVITE</strong><br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO<br>
Content-Type: application/sdp<br>User-Agent: X-Lite release 1103k stamp 53621<br>Content-Length: 265</em></div>
<div><em>v=0<br>o=- 6 2 IN IP4 192.168.1.116<br>s=CounterPath X-Lite 3.0<br>c=IN IP4 192.168.1.116<br>t=0 0<br>m=audio 5960 RTP/AVP 107 0 8 101<br>a=alt:1 1 : 6O20T816 ayXoiBh+ 192.168.1.116 5960<br>a=fmtp:101 0-15<br>a=rtpmap:107 BV32/16000<br>
a=rtpmap:101 telephone-event/8000<br>a=sendrecv</em></div>
<div><em><-------------><br>--- (12 headers 11 lines) ---<br>Sending to 192.168.1.116 : 14862 (NAT)<br>Using INVITE request as basis request - NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.</em></div>
<div><em><--- Reliably Transmitting (no NAT) to <a href="http://192.168.1.116:14862">192.168.1.116:14862</a> ---><br>SIP/2.0 407 <strong>Proxy Authentication Required<br></strong>Via: SIP/2.0/UDP 192.168.1.116:14862;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;received=192.168.1.116;rport=14862<br>
From: "gianca"<<a href="mailto:sip%3Agianca@192.168.1.100">sip:gianca@192.168.1.100</a>>;tag=db428348<br>To: "12345"<<a href="mailto:sip%3A12345@192.168.1.100">sip:12345@192.168.1.100</a>>;tag=as29d2b71c<br>
Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.<br>CSeq: 1 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>upported: replaces<br>Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="42ebb35e"<br>
Content-Length: 0</em></div>
<div><br><em><------------><br>Scheduling destruction of SIP dialog 'NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.' in 32000 ms (Method: INVITE)<br>Found user 'gianca'<br>dhcppc0*CLI><br><--- SIP read from <a href="http://192.168.1.116:14862">192.168.1.116:14862</a> ---><br>
ACK <a href="mailto:sip%3A12345@192.168.1.100">sip:12345@192.168.1.100</a> SIP/2.0<br>Via: SIP/2.0/UDP 192.168.1.116:14862;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;rport<br>To: "12345"<<a href="mailto:sip%3A12345@192.168.1.100">sip:12345@192.168.1.100</a>>;tag=as29d2b71c<br>
From: "gianca"<<a href="mailto:sip%3Agianca@192.168.1.100">sip:gianca@192.168.1.100</a>>;tag=db428348<br>Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.<br>CSeq: 1 ACK<br>Content-Length: 0</em></div>
<div><br><em><-------------><br>--- (7 headers 0 lines) ---<br>dhcppc0*CLI><br><--- SIP read from <a href="http://192.168.1.116:14862">192.168.1.116:14862</a> ---><br>INVITE <a href="mailto:sip%3A12345@192.168.1.100">sip:12345@192.168.1.100</a> SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.1.116:14862;branch=z9hG4bK-d8754z-dd20ff3e0753755b-1---d8754z-;rport<br>Max-Forwards: 70<br>Contact: <<a href="http://sip:gianca@192.168.1.116:14862">sip:gianca@192.168.1.116:14862</a>><br>To: "12345"<<a href="mailto:sip%3A12345@192.168.1.100">sip:12345@192.168.1.100</a>><br>
From: "gianca"<<a href="mailto:sip%3Agianca@192.168.1.100">sip:gianca@192.168.1.100</a>>;tag=db428348<br>Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.<br>CSeq: 2 INVITE<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO<br>
Content-Type: application/sdp<br>Proxy-Authorization: Digest username="gianca",realm="asterisk",nonce="42ebb35e",uri="<a href="mailto:sip%3A12345@192.168.1.100">sip:12345@192.168.1.100</a>",response="8d00b3e1b28ed2e40681a3a9ee410046",algorithm=MD5<br>
User-Agent: X-Lite release 1103k stamp 53621<br>Content-Length: 265</em></div>
<div><em>v=0<br>o=- 6 2 IN IP4 192.168.1.116<br>s=CounterPath X-Lite 3.0<br>c=IN IP4 192.168.1.116<br>t=0 0<br>m=audio 5960 RTP/AVP 107 0 8 101<br>a=alt:1 1 : 6O20T816 ayXoiBh+ 192.168.1.116 5960<br>a=fmtp:101 0-15<br>a=rtpmap:107 BV32/16000<br>
a=rtpmap:101 telephone-event/8000<br>a=sendrecv</em></div>
<div><em><-------------><br>--- (13 headers 11 lines) ---<br>Sending to 192.168.1.116 : 14862 (NAT)<br>Using INVITE request as basis request - NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.<br>Found user 'gianca'<br>
Found RTP audio format 107<br>Found RTP audio format 0<br>Found RTP audio format 8<br>Found RTP audio format 101<br>Peer audio RTP is at port <a href="http://192.168.1.116:5960">192.168.1.116:5960</a><br>Found unknown media description format BV32 for ID 107<br>
Found audio description format telephone-event for ID 101<br>Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)<br>Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)<br>
Peer audio RTP is at port <a href="http://192.168.1.116:5960">192.168.1.116:5960</a><br>Looking for 12345 in tutorial (domain 192.168.1.100)<br>list_route: hop: <<a href="http://sip:gianca@192.168.1.116:14862">sip:gianca@192.168.1.116:14862</a>></em></div>
<div><em><--- Transmitting (no NAT) to <a href="http://192.168.1.116:14862">192.168.1.116:14862</a> ---><br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP 192.168.1.116:14862;branch=z9hG4bK-d8754z-dd20ff3e0753755b-1---d8754z-;received=192.168.1.116;rport=14862<br>
From: "gianca"<<a href="mailto:sip%3Agianca@192.168.1.100">sip:gianca@192.168.1.100</a>>;tag=db428348<br>To: "12345"<<a href="mailto:sip%3A12345@192.168.1.100">sip:12345@192.168.1.100</a>><br>
Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.<br>CSeq: 2 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Contact: <<a href="mailto:sip%3A12345@192.168.1.100">sip:12345@192.168.1.100</a>><br>
Content-Length: 0</em></div>
<div><br><em><------------><br> -- Executing [12345@tutorial:1] Dial("SIP/gianca-088b96e0", "SIP|giusy") in new stack<br> == Spawn extension (tutorial, 12345, 1) exited non-zero on 'SIP/gianca-088b96e0'<br>
Scheduling destruction of SIP dialog 'NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.' in 32000 ms (Method: INVITE)</em></div>
<div><em><--- Reliably Transmitting (no NAT) to <a href="http://192.168.1.116:14862">192.168.1.116:14862</a> ---><br>SIP/2.0 603 Declined<br>Via: SIP/2.0/UDP 192.168.1.116:14862;branch=z9hG4bK-d8754z-dd20ff3e0753755b-1---d8754z-;received=192.168.1.116;rport=14862<br>
From: "gianca"<<a href="mailto:sip%3Agianca@192.168.1.100">sip:gianca@192.168.1.100</a>>;tag=db428348<br>To: "12345"<<a href="mailto:sip%3A12345@192.168.1.100">sip:12345@192.168.1.100</a>>;tag=as12cbf532<br>
Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.<br>CSeq: 2 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Content-Length: 0</em></div>
<div><br><em><------------><br>dhcppc0*CLI><br><--- SIP read from <a href="http://192.168.1.116:14862">192.168.1.116:14862</a> ---><br>ACK <a href="mailto:sip%3A12345@192.168.1.100">sip:12345@192.168.1.100</a> SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.1.116:14862;branch=z9hG4bK-d8754z-dd20ff3e0753755b-1---d8754z-;rport<br>To: "12345"<<a href="mailto:sip%3A12345@192.168.1.100">sip:12345@192.168.1.100</a>>;tag=as12cbf532<br>From: "gianca"<<a href="mailto:sip%3Agianca@192.168.1.100">sip:gianca@192.168.1.100</a>>;tag=db428348<br>
Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.<br>CSeq: 2 ACK<br>Content-Length: 0<br></em></div>
<div><em></em> </div>
<div><br clear="all"><br>-- <br>Giancarlo Lombardo<br></div>