<div>Thanks !!</div>
<div>it works <br><br></div>
<div class="gmail_quote">2009/11/10 Michael Wyres <span dir="ltr"><<a href="mailto:mwyres@cdm.com.au">mwyres@cdm.com.au</a>></span><br>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<div lang="EN-AU" vlink="purple" link="blue">
<div>
<p class="MsoNormal"><span style="FONT-SIZE: 11pt; COLOR: #1f497d">Try:</span></p>
<p class="MsoNormal"><span style="FONT-SIZE: 11pt; COLOR: #1f497d"> </span></p>
<p class="MsoNormal"><em>[tutorial]</em><i><br><em>exten => 1234,1,Dial(SIP/gianca,10,t)</em></i></p>
<p class="MsoNormal"><em>exten => 12345,1,Dial(SIP/giusy,10,t</em>)</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">You want a “/” between SIP and the name of the phone, not an “,”.</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">The “10” refers to the number of seconds you want the phone to ring. The “t” allows the channel to be transferred after pickup – not strictly needed, but I tend to put it in in most instances as generally you’ll want it.</p>
<p class="MsoNormal"> </p>
<p class="MsoNormal">For more information on the Dial application, see <a href="http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial" target="_blank">http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial</a></p>
<p class="MsoNormal"> </p>
<p class="MsoNormal"> </p>
<p class="MsoNormal"><span style="FONT-SIZE: 11pt; COLOR: #1f497d"> </span></p>
<p class="MsoNormal"><span style="FONT-SIZE: 11pt; COLOR: #1f497d"> </span></p>
<div style="BORDER-RIGHT: medium none; PADDING-RIGHT: 0cm; BORDER-TOP: #b5c4df 1pt solid; PADDING-LEFT: 0cm; PADDING-BOTTOM: 0cm; BORDER-LEFT: medium none; PADDING-TOP: 3pt; BORDER-BOTTOM: medium none">
<p class="MsoNormal"><b><span lang="EN-US" style="FONT-SIZE: 10pt">From:</span></b><span lang="EN-US" style="FONT-SIZE: 10pt"> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b>On Behalf Of </b>giancarlo lombardo<br>
<b>Sent:</b> Tuesday, 10 November 2009 09:03
<div class="im"><br><b>To:</b> <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br></div><b>Subject:</b> [asterisk-users] Call declined</span>
<p></p></p></div>
<div>
<div></div>
<div class="h5">
<p class="MsoNormal"> </p>
<div>
<p class="MsoNormal">Dear all,</p></div>
<div>
<p class="MsoNormal">I'm in basic setup of my network:</p></div>
<div>
<p class="MsoNormal"> </p></div>
<div>
<p class="MsoNormal">I try to do a call from a softphone to an other one but I got the error 603 Declined.</p></div>
<div>
<p class="MsoNormal"> </p></div>
<div>
<p class="MsoNormal">Below the</p></div>
<div>
<p class="MsoNormal">sip.conf:</p></div>
<div>
<p class="MsoNormal"><em>[gianca]</em><i><br><em>type=friend</em><br><em>username=gianca</em><br><em>secret=pwd_gianca</em><br><em>host=dynamic</em><br><em>context=tutorial</em></i></p></div>
<div>
<p class="MsoNormal"><em>[giusy]</em><i><br><em>type=friend</em><br><em>username=giusy</em><br><em>secret=pwd_giusy</em><br><em>host=dynamic</em><br><em>context=tutorial</em></i></p></div>
<div>
<p class="MsoNormal"> </p></div>
<div>
<p class="MsoNormal"> extension.conf:</p></div>
<div>
<p class="MsoNormal"><em>[tutorial]</em><i><br><em>exten => 1234,1,Dial(SIP,gianca)</em></i></p></div>
<div>
<p class="MsoNormal"><em>exten => 12345,1,Dial(SIP,giusy</em>)</p></div>
<div>
<p class="MsoNormal"> </p></div>
<div>
<p class="MsoNormal">Below the output of SIP debug of IP caller (192.168.1.116) in asterisk</p></div>
<div>
<p class="MsoNormal"> </p></div>
<div>
<p class="MsoNormal"> </p></div>
<div>
<p class="MsoNormal"><em>dhcppc0*CLI></em><i><br><em><--- SIP read from <a href="http://192.168.1.116:14862/" target="_blank">192.168.1.116:14862</a> ---></em><br><em>INVITE <a href="mailto:sip%3A12345@192.168.1.100" target="_blank">sip:12345@192.168.1.100</a> SIP/2.0</em><br>
<em>Via: SIP/2.0/UDP 192.168.1.116:14862;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;rport</em><br><em>Max-Forwards: 70</em><br><em>Contact: <<a href="http://sip:gianca@192.168.1.116:14862" target="_blank">sip:gianca@192.168.1.116:14862</a>></em><br>
<em>To: "12345"<<a href="mailto:sip%3A12345@192.168.1.100" target="_blank">sip:12345@192.168.1.100</a>></em><br><em>From: "gianca"<<a href="mailto:sip%3Agianca@192.168.1.100" target="_blank">sip:gianca@192.168.1.100</a>>;tag=db428348</em><br>
<em>Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.</em><br><em>CSeq: 1 </em><strong>INVITE</strong><br><em>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO</em><br><em>Content-Type: application/sdp</em><br>
<em>User-Agent: X-Lite release 1103k stamp 53621</em><br><em>Content-Length: 265</em></i></p></div>
<div>
<p class="MsoNormal"><em>v=0</em><i><br><em>o=- 6 2 IN IP4 192.168.1.116</em><br><em>s=CounterPath X-Lite 3.0</em><br><em>c=IN IP4 192.168.1.116</em><br><em>t=0 0</em><br><em>m=audio 5960 RTP/AVP 107 0 8 101</em><br><em>a=alt:1 1 : 6O20T816 ayXoiBh+ 192.168.1.116 5960</em><br>
<em>a=fmtp:101 0-15</em><br><em>a=rtpmap:107 BV32/16000</em><br><em>a=rtpmap:101 telephone-event/8000</em><br><em>a=sendrecv</em></i></p></div>
<div>
<p class="MsoNormal"><em><-------------></em><i><br><em>--- (12 headers 11 lines) ---</em><br><em>Sending to 192.168.1.116 : 14862 (NAT)</em><br><em>Using INVITE request as basis request - NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.</em></i></p>
</div>
<div>
<p class="MsoNormal"><em><--- Reliably Transmitting (no NAT) to <a href="http://192.168.1.116:14862/" target="_blank">192.168.1.116:14862</a> ---></em><i><br><em>SIP/2.0 407 </em><strong>Proxy Authentication Required</strong><b><br>
</b><em>Via: SIP/2.0/UDP 192.168.1.116:14862;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;received=192.168.1.116;rport=14862</em><br><em>From: "gianca"<<a href="mailto:sip%3Agianca@192.168.1.100" target="_blank">sip:gianca@192.168.1.100</a>>;tag=db428348</em><br>
<em>To: "12345"<<a href="mailto:sip%3A12345@192.168.1.100" target="_blank">sip:12345@192.168.1.100</a>>;tag=as29d2b71c</em><br><em>Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.</em><br><em>CSeq: 1 INVITE</em><br>
<em>User-Agent: Asterisk PBX</em><br><em>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</em><br><em>upported: replaces</em><br><em>Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="42ebb35e"</em><br>
<em>Content-Length: 0</em></i></p></div>
<div>
<p class="MsoNormal"><br><em><------------></em><i><br><em>Scheduling destruction of SIP dialog 'NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.' in 32000 ms (Method: INVITE)</em><br><em>Found user 'gianca'</em><br>
<em>dhcppc0*CLI></em><br><em><--- SIP read from <a href="http://192.168.1.116:14862/" target="_blank">192.168.1.116:14862</a> ---></em><br><em>ACK <a href="mailto:sip%3A12345@192.168.1.100" target="_blank">sip:12345@192.168.1.100</a> SIP/2.0</em><br>
<em>Via: SIP/2.0/UDP 192.168.1.116:14862;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;rport</em><br><em>To: "12345"<<a href="mailto:sip%3A12345@192.168.1.100" target="_blank">sip:12345@192.168.1.100</a>>;tag=as29d2b71c</em><br>
<em>From: "gianca"<<a href="mailto:sip%3Agianca@192.168.1.100" target="_blank">sip:gianca@192.168.1.100</a>>;tag=db428348</em><br><em>Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.</em><br><em>CSeq: 1 ACK</em><br>
<em>Content-Length: 0</em></i></p></div>
<div>
<p class="MsoNormal"><br><em><-------------></em><i><br><em>--- (7 headers 0 lines) ---</em><br><em>dhcppc0*CLI></em><br><em><--- SIP read from <a href="http://192.168.1.116:14862/" target="_blank">192.168.1.116:14862</a> ---></em><br>
<em>INVITE <a href="mailto:sip%3A12345@192.168.1.100" target="_blank">sip:12345@192.168.1.100</a> SIP/2.0</em><br><em>Via: SIP/2.0/UDP 192.168.1.116:14862;branch=z9hG4bK-d8754z-dd20ff3e0753755b-1---d8754z-;rport</em><br><em>Max-Forwards: 70</em><br>
<em>Contact: <<a href="http://sip:gianca@192.168.1.116:14862" target="_blank">sip:gianca@192.168.1.116:14862</a>></em><br><em>To: "12345"<<a href="mailto:sip%3A12345@192.168.1.100" target="_blank">sip:12345@192.168.1.100</a>></em><br>
<em>From: "gianca"<<a href="mailto:sip%3Agianca@192.168.1.100" target="_blank">sip:gianca@192.168.1.100</a>>;tag=db428348</em><br><em>Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.</em><br><em>CSeq: 2 INVITE</em><br>
<em>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO</em><br><em>Content-Type: application/sdp</em><br><em>Proxy-Authorization: Digest username="gianca",realm="asterisk",nonce="42ebb35e",uri="<a href="mailto:sip%3A12345@192.168.1.100" target="_blank">sip:12345@192.168.1.100</a>",response="8d00b3e1b28ed2e40681a3a9ee410046",algorithm=MD5</em><br>
<em>User-Agent: X-Lite release 1103k stamp 53621</em><br><em>Content-Length: 265</em></i></p></div>
<div>
<p class="MsoNormal"><em>v=0</em><i><br><em>o=- 6 2 IN IP4 192.168.1.116</em><br><em>s=CounterPath X-Lite 3.0</em><br><em>c=IN IP4 192.168.1.116</em><br><em>t=0 0</em><br><em>m=audio 5960 RTP/AVP 107 0 8 101</em><br><em>a=alt:1 1 : 6O20T816 ayXoiBh+ 192.168.1.116 5960</em><br>
<em>a=fmtp:101 0-15</em><br><em>a=rtpmap:107 BV32/16000</em><br><em>a=rtpmap:101 telephone-event/8000</em><br><em>a=sendrecv</em></i></p></div>
<div>
<p class="MsoNormal"><em><-------------></em><i><br><em>--- (13 headers 11 lines) ---</em><br><em>Sending to 192.168.1.116 : 14862 (NAT)</em><br><em>Using INVITE request as basis request - NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.</em><br>
<em>Found user 'gianca'</em><br><em>Found RTP audio format 107</em><br><em>Found RTP audio format 0</em><br><em>Found RTP audio format 8</em><br><em>Found RTP audio format 101</em><br><em>Peer audio RTP is at port <a href="http://192.168.1.116:5960/" target="_blank">192.168.1.116:5960</a></em><br>
<em>Found unknown media description format BV32 for ID 107</em><br><em>Found audio description format telephone-event for ID 101</em><br><em>Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)</em><br>
<em>Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)</em><br><em>Peer audio RTP is at port <a href="http://192.168.1.116:5960/" target="_blank">192.168.1.116:5960</a></em><br>
<em>Looking for 12345 in tutorial (domain 192.168.1.100)</em><br><em>list_route: hop: <<a href="http://sip:gianca@192.168.1.116:14862" target="_blank">sip:gianca@192.168.1.116:14862</a>></em></i></p></div>
<div>
<p class="MsoNormal"><em><--- Transmitting (no NAT) to <a href="http://192.168.1.116:14862/" target="_blank">192.168.1.116:14862</a> ---></em><i><br><em>SIP/2.0 100 Trying</em><br><em>Via: SIP/2.0/UDP 192.168.1.116:14862;branch=z9hG4bK-d8754z-dd20ff3e0753755b-1---d8754z-;received=192.168.1.116;rport=14862</em><br>
<em>From: "gianca"<<a href="mailto:sip%3Agianca@192.168.1.100" target="_blank">sip:gianca@192.168.1.100</a>>;tag=db428348</em><br><em>To: "12345"<<a href="mailto:sip%3A12345@192.168.1.100" target="_blank">sip:12345@192.168.1.100</a>></em><br>
<em>Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.</em><br><em>CSeq: 2 INVITE</em><br><em>User-Agent: Asterisk PBX</em><br><em>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</em><br><em>Supported: replaces</em><br>
<em>Contact: <<a href="mailto:sip%3A12345@192.168.1.100" target="_blank">sip:12345@192.168.1.100</a>></em><br><em>Content-Length: 0</em></i></p></div>
<div>
<p class="MsoNormal"><br><em><------------></em><i><br><em> -- Executing [12345@tutorial:1] Dial("SIP/gianca-088b96e0", "SIP|giusy") in new stack</em><br><em> == Spawn extension (tutorial, 12345, 1) exited non-zero on 'SIP/gianca-088b96e0'</em><br>
<em>Scheduling destruction of SIP dialog 'NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.' in 32000 ms (Method: INVITE)</em></i></p></div>
<div>
<p class="MsoNormal"><em><--- Reliably Transmitting (no NAT) to <a href="http://192.168.1.116:14862/" target="_blank">192.168.1.116:14862</a> ---></em><i><br><em>SIP/2.0 603 Declined</em><br><em>Via: SIP/2.0/UDP 192.168.1.116:14862;branch=z9hG4bK-d8754z-dd20ff3e0753755b-1---d8754z-;received=192.168.1.116;rport=14862</em><br>
<em>From: "gianca"<<a href="mailto:sip%3Agianca@192.168.1.100" target="_blank">sip:gianca@192.168.1.100</a>>;tag=db428348</em><br><em>To: "12345"<<a href="mailto:sip%3A12345@192.168.1.100" target="_blank">sip:12345@192.168.1.100</a>>;tag=as12cbf532</em><br>
<em>Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.</em><br><em>CSeq: 2 INVITE</em><br><em>User-Agent: Asterisk PBX</em><br><em>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY</em><br><em>Supported: replaces</em><br>
<em>Content-Length: 0</em></i></p></div>
<div>
<p class="MsoNormal"><br><em><------------></em><i><br><em>dhcppc0*CLI></em><br><em><--- SIP read from <a href="http://192.168.1.116:14862/" target="_blank">192.168.1.116:14862</a> ---></em><br><em>ACK <a href="mailto:sip%3A12345@192.168.1.100" target="_blank">sip:12345@192.168.1.100</a> SIP/2.0</em><br>
<em>Via: SIP/2.0/UDP 192.168.1.116:14862;branch=z9hG4bK-d8754z-dd20ff3e0753755b-1---d8754z-;rport</em><br><em>To: "12345"<<a href="mailto:sip%3A12345@192.168.1.100" target="_blank">sip:12345@192.168.1.100</a>>;tag=as12cbf532</em><br>
<em>From: "gianca"<<a href="mailto:sip%3Agianca@192.168.1.100" target="_blank">sip:gianca@192.168.1.100</a>>;tag=db428348</em><br><em>Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.</em><br><em>CSeq: 2 ACK</em><br>
<em>Content-Length: 0</em></i></p></div>
<div>
<p class="MsoNormal"> </p></div>
<div>
<p class="MsoNormal"><br clear="all"><br>-- <br>Giancarlo Lombardo</p></div></div></div></div><pre>IMPORTANT NOTICE TO RECIPIENT
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</pre></div><br>_______________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a> --<br><br>asterisk-users mailing list<br>
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<br>-- <br>Giancarlo Lombardo<br>