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<div class=Section1>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Try:<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><em>[tutorial]</em><i><br>
<em>exten => 1234,1,Dial(SIP/gianca,10,t)</em></i><o:p></o:p></p>
<p class=MsoNormal><em>exten => 12345,1,Dial(SIP/giusy,10,t</em>)<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>You want a “/” between SIP and the name of the phone, not an
“,”.<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></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.<o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>For more information on the Dial application, see <a
href="http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial">http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial</a><o:p></o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'>
<p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt;font-family:
"Tahoma","sans-serif"'>From:</span></b><span lang=EN-US style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>giancarlo
lombardo<br>
<b>Sent:</b> Tuesday, 10 November 2009 09:03<br>
<b>To:</b> asterisk-users@lists.digium.com<br>
<b>Subject:</b> [asterisk-users] Call declined<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<div>
<p class=MsoNormal>Dear all,<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal>I'm in basic setup of my network:<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></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.<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal>Below the<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal>sip.conf:<o:p></o:p></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><o:p></o:p></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><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal> extension.conf:<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><em>[tutorial]</em><i><br>
<em>exten => 1234,1,Dial(SIP,gianca)</em></i><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><em>exten => 12345,1,Dial(SIP,giusy</em>)<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal>Below the output of SIP debug of IP caller (192.168.1.116)
in asterisk<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><em>dhcppc0*CLI></em><i><br>
<em><--- SIP read from <a href="http://192.168.1.116:14862">192.168.1.116:14862</a>
---></em><br>
<em>INVITE <a href="mailto:sip%3A12345@192.168.1.100">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">sip:gianca@192.168.1.116:14862</a>></em><br>
<em>To: "12345"<<a href="mailto:sip%3A12345@192.168.1.100">sip:12345@192.168.1.100</a>></em><br>
<em>From: "gianca"<<a href="mailto:sip%3Agianca@192.168.1.100">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><o:p></o:p></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><o:p></o:p></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><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><em><--- Reliably Transmitting (no NAT) to <a
href="http://192.168.1.116:14862">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">sip:gianca@192.168.1.100</a>>;tag=db428348</em><br>
<em>To: "12345"<<a href="mailto:sip%3A12345@192.168.1.100">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><o:p></o:p></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">192.168.1.116:14862</a>
---></em><br>
<em>ACK <a href="mailto:sip%3A12345@192.168.1.100">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">sip:12345@192.168.1.100</a>>;tag=as29d2b71c</em><br>
<em>From: "gianca"<<a href="mailto:sip%3Agianca@192.168.1.100">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><o:p></o:p></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">192.168.1.116:14862</a>
---></em><br>
<em>INVITE <a href="mailto:sip%3A12345@192.168.1.100">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">sip:gianca@192.168.1.116:14862</a>></em><br>
<em>To: "12345"<<a href="mailto:sip%3A12345@192.168.1.100">sip:12345@192.168.1.100</a>></em><br>
<em>From: "gianca"<<a href="mailto:sip%3Agianca@192.168.1.100">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">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><o:p></o:p></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><o:p></o:p></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">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">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">sip:gianca@192.168.1.116:14862</a>></em></i><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><em><--- Transmitting (no NAT) to <a
href="http://192.168.1.116:14862">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">sip:gianca@192.168.1.100</a>>;tag=db428348</em><br>
<em>To: "12345"<<a href="mailto:sip%3A12345@192.168.1.100">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">sip:12345@192.168.1.100</a>></em><br>
<em>Content-Length: 0</em></i><o:p></o:p></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><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><em><--- Reliably Transmitting (no NAT) to <a
href="http://192.168.1.116:14862">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">sip:gianca@192.168.1.100</a>>;tag=db428348</em><br>
<em>To: "12345"<<a href="mailto:sip%3A12345@192.168.1.100">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><o:p></o:p></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">192.168.1.116:14862</a>
---></em><br>
<em>ACK <a href="mailto:sip%3A12345@192.168.1.100">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">sip:12345@192.168.1.100</a>>;tag=as12cbf532</em><br>
<em>From: "gianca"<<a href="mailto:sip%3Agianca@192.168.1.100">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><o:p></o:p></p>
</div>
<div>
<p class=MsoNormal> <o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><br clear=all>
<br>
-- <br>
Giancarlo Lombardo<o:p></o:p></p>
</div>
</div>
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