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    Ichi, como uso Asterisk Puro, n&atilde;o vou saber te ajudar.&nbsp; Mas essa
    solu&ccedil;&atilde;o que encontrei foi essa&nbsp; e resolveu meu problema.&nbsp; Ter&aacute;s que
    "futucar" no DialPlan do Asterisk.<br>
    Abs..<br>
    Em 31/10/2012 14:41, Ivan Maldonado Orosco escreveu:
    <blockquote cite="mid:COL105-DS21C061B869D505BF5A3F7FD2610@phx.gbl"
      type="cite">
      <div dir="ltr">
        <div style="FONT-FAMILY: 'Calibri'; COLOR: #000000; FONT-SIZE:
          12pt">
          <div>&Eacute; meio extenso, segue:</div>
          <div>&nbsp;</div>
          <div>[Oct 30 07:41:20] VERBOSE[18040] chan_sip.c: Audio is at
            189.2.20.134 port 16664<br>
            [Oct 30 07:41:20] VERBOSE[18040] chan_sip.c: Adding codec
            0x4 (ulaw) to SDP<br>
            [Oct 30 07:41:20] VERBOSE[18040] chan_sip.c: Adding codec
            0x8 (alaw) to SDP<br>
            [Oct 30 07:41:20] VERBOSE[18040] chan_sip.c: Adding
            non-codec 0x1 (telephone-event) to SDP<br>
            [Oct 30 07:41:20] VERBOSE[18040] chan_sip.c: Reliably
            Transmitting (no NAT) to 189.47.46.151:5060:<br>
            INVITE <a class="moz-txt-link-freetext" href="sip:97095313@189.47.46.151:5060">sip:97095313@189.47.46.151:5060</a> SIP/2.0<br>
            Via: SIP/2.0/UDP
            189.2.20.134:5060;branch=z9hG4bK4c8b3981;rport<br>
            Max-Forwards: 70<br>
            From: "100" <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134">&lt;sip:100@189.2.20.134&gt;</a>;tag=as57399d63<br>
            To: <a class="moz-txt-link-rfc2396E" href="sip:97095313@189.47.46.151:5060">&lt;sip:97095313@189.47.46.151:5060&gt;</a><br>
            Contact: <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134">&lt;sip:100@189.2.20.134&gt;</a><br>
            Call-ID: <a moz-do-not-send="true"
              href="mailto:3abbce8a451fb58b6726e36623e98a89@189.2.20.134">3abbce8a451fb58b6726e36623e98a89@189.2.20.134</a><br>
            CSeq: 102 INVITE<br>
            User-Agent: FPBX-2.8.1(1.6.2.17)<br>
            Date: Tue, 30 Oct 2012 09:41:20 GMT<br>
            Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
            NOTIFY, INFO<br>
            Supported: replaces, timer<br>
            Content-Type: application/sdp<br>
            Content-Length: 261</div>
          <div>&nbsp;</div>
          <div>v=0<br>
            o=root 1238957108 1238957108 IN IP4 189.2.20.134<br>
            s=Asterisk PBX 1.6.2.17<br>
            c=IN IP4 189.2.20.134<br>
            t=0 0<br>
            m=audio 16664 RTP/AVP 0 8 101<br>
            a=rtpmap:0 PCMU/8000<br>
            a=rtpmap:8 PCMA/8000<br>
            a=rtpmap:101 telephone-event/8000<br>
            a=fmtp:101 0-16<br>
            a=ptime:20<br>
            a=sendrecv</div>
          <div>&nbsp;</div>
          <div>---<br>
            [Oct 30 07:41:20] VERBOSE[8895] chan_sip.c: <br>
            &lt;--- SIP read from UDP:189.47.46.151:5060 ---&gt;<br>
            SIP/2.0 100 Trying<br>
            Via:SIP/2.0/UDP
            189.2.20.134:5060;rport;branch=z9hG4bK4c8b3981<br>
            From: "100" <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134">&lt;sip:100@189.2.20.134&gt;</a>;tag=as57399d63<br>
            To: <a class="moz-txt-link-rfc2396E" href="sip:97095313@189.47.46.151:5060">&lt;sip:97095313@189.47.46.151:5060&gt;</a><br>
            <a class="moz-txt-link-abbreviated" href="mailto:Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134">Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134</a><br>
            CSeq:102 INVITE<br>
            Content-Type:application/sdp<br>
            Content-Length:0</div>
          <div>&nbsp;</div>
          <div><br>
            &lt;-------------&gt;<br>
            [Oct 30 07:41:20] VERBOSE[8895] chan_sip.c: --- (8 headers 0
            lines) ---<br>
            [Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: <br>
            &lt;--- SIP read from UDP:189.47.46.151:5060 ---&gt;<br>
            SIP/2.0 183 Session in progress<br>
            Via:SIP/2.0/UDP
            189.2.20.134:5060;rport;branch=z9hG4bK4c8b3981<br>
            From: "100" <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134">&lt;sip:100@189.2.20.134&gt;</a>;tag=as57399d63<br>
            To:
            <a class="moz-txt-link-rfc2396E" href="sip:97095313@189.47.46.151:5060">&lt;sip:97095313@189.47.46.151:5060&gt;</a>;tag=8ed4511d-742398<br>
            <a class="moz-txt-link-abbreviated" href="mailto:Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134">Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134</a><br>
            CSeq:102 INVITE<br>
            Contact:<a class="moz-txt-link-rfc2396E" href="sip:6720@189.47.46.151:5060">&lt;sip:6720@189.47.46.151:5060&gt;</a><br>
            User-Agent:dlink 12-37-61926642-0.9.5.1.735<br>
            Content-Type:application/sdp<br>
            Content-Length:209</div>
          <div>&nbsp;</div>
          <div>v=0<br>
            o=6720 1849593660 1849593660 IN IP4 189.47.46.151<br>
            s=Session SDP<br>
            c=IN IP4 189.47.46.151<br>
            t=0 0<br>
            m=audio 10000 RTP/AVP 0 101<br>
            a=rtpmap:0 PCMU/8000/1<br>
            a=rtpmap:101 telephone-event/8000/1<br>
            a=fmtp:101 0-16</div>
          <div>&nbsp;</div>
          <div>&lt;-------------&gt;<br>
            [Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: --- (10 headers
            9 lines) ---<br>
            [Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: Found RTP audio
            format 0<br>
            [Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: Found RTP audio
            format 101<br>
            [Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: Found audio
            description format PCMU for ID 0<br>
            [Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: Found audio
            description format telephone-event for ID 101<br>
            [Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: Capabilities: us
            - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0
            (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)<br>
            [Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: Non-codec
            capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
            (telephone-event), combined - 0x1 (telephone-event)<br>
            [Oct 30 07:41:23] VERBOSE[8895] chan_sip.c: Peer audio RTP
            is at port 189.47.46.151:10000<br>
            [Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: <br>
            &lt;--- SIP read from UDP:189.47.46.151:5060 ---&gt;<br>
            SIP/2.0 200 OK<br>
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE<br>
            Via:SIP/2.0/UDP
            189.2.20.134:5060;rport;branch=z9hG4bK4c8b3981<br>
            From: "100" <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134">&lt;sip:100@189.2.20.134&gt;</a>;tag=as57399d63<br>
            To:
            <a class="moz-txt-link-rfc2396E" href="sip:97095313@189.47.46.151:5060">&lt;sip:97095313@189.47.46.151:5060&gt;</a>;tag=8ed4511d-742398<br>
            <a class="moz-txt-link-abbreviated" href="mailto:Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134">Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134</a><br>
            CSeq:102 INVITE<br>
            Contact:<a class="moz-txt-link-rfc2396E" href="sip:6720@189.47.46.151:5060">&lt;sip:6720@189.47.46.151:5060&gt;</a><br>
            User-Agent:dlink 12-37-61926642-0.9.5.1.735<br>
            Content-Type:application/sdp<br>
            Content-Length:209</div>
          <div>&nbsp;</div>
          <div>v=0<br>
            o=6720 1849602720 1849602720 IN IP4 189.47.46.151<br>
            s=Session SDP<br>
            c=IN IP4 189.47.46.151<br>
            t=0 0<br>
            m=audio 10000 RTP/AVP 0 101<br>
            a=rtpmap:0 PCMU/8000/1<br>
            a=rtpmap:101 telephone-event/8000/1<br>
            a=fmtp:101 0-16</div>
          <div>&nbsp;</div>
          <div>&lt;-------------&gt;<br>
            [Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: --- (11 headers
            9 lines) ---<br>
            [Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: Found RTP audio
            format 0<br>
            [Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: Found RTP audio
            format 101<br>
            [Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: Found audio
            description format PCMU for ID 0<br>
            [Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: Found audio
            description format telephone-event for ID 101<br>
            [Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: Capabilities: us
            - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0
            (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)<br>
            [Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: Non-codec
            capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
            (telephone-event), combined - 0x1 (telephone-event)<br>
            [Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: Peer audio RTP
            is at port 189.47.46.151:10000<br>
            [Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: list_route: hop:
            <a class="moz-txt-link-rfc2396E" href="sip:6720@189.47.46.151:5060">&lt;sip:6720@189.47.46.151:5060&gt;</a><br>
            [Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: set_destination:
            Parsing <a class="moz-txt-link-rfc2396E" href="sip:6720@189.47.46.151:5060">&lt;sip:6720@189.47.46.151:5060&gt;</a> for address/port
            to send to<br>
            [Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: set_destination:
            set destination to 189.47.46.151, port 5060<br>
            [Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: Transmitting (no
            NAT) to 189.47.46.151:5060:<br>
            ACK <a class="moz-txt-link-freetext" href="sip:6720@189.47.46.151:5060">sip:6720@189.47.46.151:5060</a> SIP/2.0<br>
            Via: SIP/2.0/UDP
            189.2.20.134:5060;branch=z9hG4bK1c6b8c10;rport<br>
            Max-Forwards: 70<br>
            From: "100" <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134">&lt;sip:100@189.2.20.134&gt;</a>;tag=as57399d63<br>
            To:
            <a class="moz-txt-link-rfc2396E" href="sip:97095313@189.47.46.151:5060">&lt;sip:97095313@189.47.46.151:5060&gt;</a>;tag=8ed4511d-742398<br>
            Contact: <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134">&lt;sip:100@189.2.20.134&gt;</a><br>
            Call-ID: <a moz-do-not-send="true"
              href="mailto:3abbce8a451fb58b6726e36623e98a89@189.2.20.134">3abbce8a451fb58b6726e36623e98a89@189.2.20.134</a><br>
            CSeq: 102 ACK<br>
            User-Agent: FPBX-2.8.1(1.6.2.17)<br>
            Content-Length: 0</div>
          <div>&nbsp;</div>
          <div><br>
            ---<br>
            [Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: <br>
            &lt;--- SIP read from UDP:189.47.46.151:5060 ---&gt;<br>
            SIP/2.0 200 OK<br>
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE<br>
            Via:SIP/2.0/UDP
            189.2.20.134:5060;rport;branch=z9hG4bK4c8b3981<br>
            From: "100" <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134">&lt;sip:100@189.2.20.134&gt;</a>;tag=as57399d63<br>
            To:
            <a class="moz-txt-link-rfc2396E" href="sip:97095313@189.47.46.151:5060">&lt;sip:97095313@189.47.46.151:5060&gt;</a>;tag=8ed4511d-742398<br>
            <a class="moz-txt-link-abbreviated" href="mailto:Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134">Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134</a><br>
            CSeq:102 INVITE<br>
            Contact:<a class="moz-txt-link-rfc2396E" href="sip:6720@189.47.46.151:5060">&lt;sip:6720@189.47.46.151:5060&gt;</a><br>
            User-Agent:dlink 12-37-61926642-0.9.5.1.735<br>
            Content-Type:application/sdp<br>
            Content-Length:209</div>
          <div>&nbsp;</div>
          <div>v=0<br>
            o=6720 1849602720 1849602720 IN IP4 189.47.46.151<br>
            s=Session SDP<br>
            c=IN IP4 189.47.46.151<br>
            t=0 0<br>
            m=audio 10000 RTP/AVP 0 101<br>
            a=rtpmap:0 PCMU/8000/1<br>
            a=rtpmap:101 telephone-event/8000/1<br>
            a=fmtp:101 0-16</div>
          <div>&nbsp;</div>
          <div>&lt;-------------&gt;<br>
            [Oct 30 07:41:32] VERBOSE[8895] chan_sip.c: --- (11 headers
            9 lines) ---<br>
            [Oct 30 07:41:34] VERBOSE[8895] chan_sip.c: <br>
            &lt;--- SIP read from UDP:189.47.46.151:5060 ---&gt;<br>
            SIP/2.0 200 OK<br>
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE<br>
            Via:SIP/2.0/UDP
            189.2.20.134:5060;rport;branch=z9hG4bK4c8b3981<br>
            From: "100" <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134">&lt;sip:100@189.2.20.134&gt;</a>;tag=as57399d63<br>
            To:
            <a class="moz-txt-link-rfc2396E" href="sip:97095313@189.47.46.151:5060">&lt;sip:97095313@189.47.46.151:5060&gt;</a>;tag=8ed4511d-742398<br>
            <a class="moz-txt-link-abbreviated" href="mailto:Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134">Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134</a><br>
            CSeq:102 INVITE<br>
            Contact:<a class="moz-txt-link-rfc2396E" href="sip:6720@189.47.46.151:5060">&lt;sip:6720@189.47.46.151:5060&gt;</a><br>
            User-Agent:dlink 12-37-61926642-0.9.5.1.735<br>
            Content-Type:application/sdp<br>
            Content-Length:209</div>
          <div>&nbsp;</div>
          <div>v=0<br>
            o=6720 1849602720 1849602720 IN IP4 189.47.46.151<br>
            s=Session SDP<br>
            c=IN IP4 189.47.46.151<br>
            t=0 0<br>
            m=audio 10000 RTP/AVP 0 101<br>
            a=rtpmap:0 PCMU/8000/1<br>
            a=rtpmap:101 telephone-event/8000/1<br>
            a=fmtp:101 0-16</div>
          <div>&nbsp;</div>
          <div>&lt;-------------&gt;<br>
            [Oct 30 07:41:34] VERBOSE[8895] chan_sip.c: --- (11 headers
            9 lines) ---<br>
            [Oct 30 07:41:36] VERBOSE[8895] chan_sip.c: <br>
            &lt;--- SIP read from UDP:189.47.46.151:5060 ---&gt;<br>
            SIP/2.0 200 OK<br>
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE<br>
            Via:SIP/2.0/UDP
            189.2.20.134:5060;rport;branch=z9hG4bK4c8b3981<br>
            From: "100" <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134">&lt;sip:100@189.2.20.134&gt;</a>;tag=as57399d63<br>
            To:
            <a class="moz-txt-link-rfc2396E" href="sip:97095313@189.47.46.151:5060">&lt;sip:97095313@189.47.46.151:5060&gt;</a>;tag=8ed4511d-742398<br>
            <a class="moz-txt-link-abbreviated" href="mailto:Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134">Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134</a><br>
            CSeq:102 INVITE<br>
            Contact:<a class="moz-txt-link-rfc2396E" href="sip:6720@189.47.46.151:5060">&lt;sip:6720@189.47.46.151:5060&gt;</a><br>
            User-Agent:dlink 12-37-61926642-0.9.5.1.735<br>
            Content-Type:application/sdp<br>
            Content-Length:209</div>
          <div>&nbsp;</div>
          <div>v=0<br>
            o=6720 1849602720 1849602720 IN IP4 189.47.46.151<br>
            s=Session SDP<br>
            c=IN IP4 189.47.46.151<br>
            t=0 0<br>
            m=audio 10000 RTP/AVP 0 101<br>
            a=rtpmap:0 PCMU/8000/1<br>
            a=rtpmap:101 telephone-event/8000/1<br>
            a=fmtp:101 0-16</div>
          <div>&nbsp;</div>
          <div>&lt;-------------&gt;<br>
            [Oct 30 07:41:36] VERBOSE[8895] chan_sip.c: --- (11 headers
            9 lines) ---<br>
            [Oct 30 07:41:40] VERBOSE[8895] chan_sip.c: <br>
            &lt;--- SIP read from UDP:189.47.46.151:5060 ---&gt;<br>
            SIP/2.0 200 OK<br>
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE<br>
            Via:SIP/2.0/UDP
            189.2.20.134:5060;rport;branch=z9hG4bK4c8b3981<br>
            From: "100" <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134">&lt;sip:100@189.2.20.134&gt;</a>;tag=as57399d63<br>
            To:
            <a class="moz-txt-link-rfc2396E" href="sip:97095313@189.47.46.151:5060">&lt;sip:97095313@189.47.46.151:5060&gt;</a>;tag=8ed4511d-742398<br>
            <a class="moz-txt-link-abbreviated" href="mailto:Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134">Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134</a><br>
            CSeq:102 INVITE<br>
            Contact:<a class="moz-txt-link-rfc2396E" href="sip:6720@189.47.46.151:5060">&lt;sip:6720@189.47.46.151:5060&gt;</a><br>
            User-Agent:dlink 12-37-61926642-0.9.5.1.735<br>
            Content-Type:application/sdp<br>
            Content-Length:209</div>
          <div>&nbsp;</div>
          <div>v=0<br>
            o=6720 1849602720 1849602720 IN IP4 189.47.46.151<br>
            s=Session SDP<br>
            c=IN IP4 189.47.46.151<br>
            t=0 0<br>
            m=audio 10000 RTP/AVP 0 101<br>
            a=rtpmap:0 PCMU/8000/1<br>
            a=rtpmap:101 telephone-event/8000/1<br>
            a=fmtp:101 0-16</div>
          <div>&nbsp;</div>
          <div><br>
            &lt;-------------&gt;<br>
            [Oct 30 07:41:44] VERBOSE[8895] chan_sip.c: --- (10 headers
            0 lines) ---<br>
            [Oct 30 07:41:44] VERBOSE[8895] chan_sip.c: Really
            destroying SIP dialog <a moz-do-not-send="true"
              href="mailto:%27644015366b7e382f266705505a200615@189.2.20.134%27">'644015366b7e382f266705505a200615@189.2.20.134'</a>
            Method: OPTIONS<br>
            [Oct 30 07:41:48] VERBOSE[8895] chan_sip.c: <br>
            &lt;--- SIP read from UDP:189.47.46.151:5060 ---&gt;<br>
            SIP/2.0 200 OK<br>
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE<br>
            Via:SIP/2.0/UDP
            189.2.20.134:5060;rport;branch=z9hG4bK4c8b3981<br>
            From: "100" <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134">&lt;sip:100@189.2.20.134&gt;</a>;tag=as57399d63<br>
            To:
            <a class="moz-txt-link-rfc2396E" href="sip:97095313@189.47.46.151:5060">&lt;sip:97095313@189.47.46.151:5060&gt;</a>;tag=8ed4511d-742398<br>
            <a class="moz-txt-link-abbreviated" href="mailto:Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134">Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134</a><br>
            CSeq:102 INVITE<br>
            Contact:<a class="moz-txt-link-rfc2396E" href="sip:6720@189.47.46.151:5060">&lt;sip:6720@189.47.46.151:5060&gt;</a><br>
            User-Agent:dlink 12-37-61926642-0.9.5.1.735<br>
            Content-Type:application/sdp<br>
            Content-Length:209</div>
          <div>&nbsp;</div>
          <div>v=0<br>
            o=6720 1849602720 1849602720 IN IP4 189.47.46.151<br>
            s=Session SDP<br>
            c=IN IP4 189.47.46.151<br>
            t=0 0<br>
            m=audio 10000 RTP/AVP 0 101<br>
            a=rtpmap:0 PCMU/8000/1<br>
            a=rtpmap:101 telephone-event/8000/1<br>
            a=fmtp:101 0-16</div>
          <div>&nbsp;</div>
          <div>&lt;-------------&gt;<br>
            [Oct 30 07:41:48] VERBOSE[8895] chan_sip.c: --- (11 headers
            9 lines) ---<br>
            [Oct 30 07:41:53] WARNING[18042] func_db.c: DB_DELETE
            requires an argument, DB_DELETE(&lt;family&gt;/&lt;key&gt;)<br>
            [Oct 30 07:41:53] VERBOSE[8895] chan_sip.c: <br>
            &lt;--- SIP read from UDP:189.47.46.151:5060 ---&gt;<br>
            BYE <a class="moz-txt-link-freetext" href="sip:100@189.2.20.134">sip:100@189.2.20.134</a> SIP/2.0<br>
            Via:SIP/2.0/UDP
            189.47.46.151:5060;branch=z9hG4bK567b6611820f8f0c<br>
            From:
            <a class="moz-txt-link-rfc2396E" href="sip:97095313@189.47.46.151:5060">&lt;sip:97095313@189.47.46.151:5060&gt;</a>;tag=8ed4511d-742398<br>
            To: "100" <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134">&lt;sip:100@189.2.20.134&gt;</a>;tag=as57399d63<br>
            <a class="moz-txt-link-abbreviated" href="mailto:Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134">Call-ID:3abbce8a451fb58b6726e36623e98a89@189.2.20.134</a><br>
            CSeq:8 BYE<br>
            Contact:<a class="moz-txt-link-rfc2396E" href="sip:6720@189.47.46.151:5060">&lt;sip:6720@189.47.46.151:5060&gt;</a><br>
            Max-Forwards:70<br>
            User-Agent:dlink 12-37-61926642-0.9.5.1.735<br>
            Content-Length:0</div>
          <div>&nbsp;</div>
          <div><br>
            &lt;-------------&gt;<br>
            [Oct 30 07:41:53] VERBOSE[8895] chan_sip.c: --- (10 headers
            0 lines) ---<br>
            [Oct 30 07:41:53] VERBOSE[8895] chan_sip.c: Sending to
            189.47.46.151 : 5060 (no NAT)<br>
            [Oct 30 07:41:53] VERBOSE[8895] chan_sip.c: <br>
            &lt;--- Transmitting (no NAT) to 189.47.46.151:5060 ---&gt;<br>
            SIP/2.0 200 OK<br>
            Via: SIP/2.0/UDP
            189.47.46.151:5060;branch=z9hG4bK567b6611820f8f0c;received=189.47.46.151<br>
            From:
            <a class="moz-txt-link-rfc2396E" href="sip:97095313@189.47.46.151:5060">&lt;sip:97095313@189.47.46.151:5060&gt;</a>;tag=8ed4511d-742398<br>
            To: "100" <a class="moz-txt-link-rfc2396E" href="sip:100@189.2.20.134">&lt;sip:100@189.2.20.134&gt;</a>;tag=as57399d63<br>
            Call-ID: <a moz-do-not-send="true"
              href="mailto:3abbce8a451fb58b6726e36623e98a89@189.2.20.134">3abbce8a451fb58b6726e36623e98a89@189.2.20.134</a><br>
            CSeq: 8 BYE<br>
            Server: FPBX-2.8.1(1.6.2.17)<br>
            Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
            NOTIFY, INFO<br>
            Supported: replaces, timer<br>
            Content-Length: 0</div>
          <div>&nbsp;</div>
          <div><br>
            &lt;------------&gt;<br>
            [Oct 30 07:41:54] VERBOSE[8895] chan_sip.c: Really
            destroying SIP dialog <a moz-do-not-send="true"
              href="mailto:%273abbce8a451fb58b6726e36623e98a89@189.2.20.134%27">'3abbce8a451fb58b6726e36623e98a89@189.2.20.134'</a>
            Method: BYE<br>
            [Oct 30 07:42:15] VERBOSE[8895] chan_sip.c: Really
            destroying SIP dialog <a moz-do-not-send="true"
              href="mailto:%27CE40-3FA8-4674241856E1641BE1B1-193@SipHost%27">'CE40-3FA8-4674241856E1641BE1B1-193@SipHost'</a>
            Method: REGISTER<br>
            [Oct 30 07:42:16] VERBOSE[8895] chan_sip.c: Really
            destroying SIP dialog <a moz-do-not-send="true"
              href="mailto:%27CE40-3FA8-4674242047F2EB3729E8-194@SipHost%27">'CE40-3FA8-4674242047F2EB3729E8-194@SipHost'</a>
            Method: REGISTER<br>
          </div>
          <div>[]&#8217;s</div>
          <div style="FONT-STYLE: normal; DISPLAY: inline; FONT-FAMILY:
            'Calibri'; COLOR: #000000; FONT-SIZE: small; FONT-WEIGHT:
            normal; TEXT-DECORATION: none">
            <div style="FONT: 10pt tahoma">
              <div>&nbsp;</div>
              <div style="BACKGROUND: #f5f5f5">
                <div style="font-color: black"><b>From:</b> <a
                    moz-do-not-send="true" title="ivan.paes@gmail.com"
                    href="mailto:ivan.paes@gmail.com">Ivan Paes Jos&eacute;</a>
                </div>
                <div><b>Sent:</b> Tuesday, October 30, 2012 12:20 AM</div>
                <div><b>To:</b> <a moz-do-not-send="true"
                    title="asteriskbrasil@listas.asteriskbrasil.org"
                    href="mailto:asteriskbrasil@listas.asteriskbrasil.org">asteriskbrasil@listas.asteriskbrasil.org</a>
                </div>
                <div><b>Subject:</b> Re: [AsteriskBrasil] DVG 6004S
                  St_VoipAnswering Timeout</div>
              </div>
            </div>
            <div>&nbsp;</div>
          </div>
          <div style="FONT-STYLE: normal; DISPLAY: inline; FONT-FAMILY:
            'Calibri'; COLOR: #000000; FONT-SIZE: small; FONT-WEIGHT:
            normal; TEXT-DECORATION: none">
            <div>Ol&aacute;!</div>
            <div>&nbsp;</div>
            <div>Consegues o debug do sip do CLI do asterisk?</div>
            <br clear="all">
            <font size="1"><span style="LINE-HEIGHT: normal;
                TEXT-TRANSFORM: none; FONT-VARIANT: normal; FONT-STYLE:
                normal; TEXT-INDENT: 0px; LETTER-SPACING: normal;
                BORDER-COLLAPSE: separate; FONT-FAMILY: 'Times New
                Roman'; WHITE-SPACE: normal; COLOR: rgb(0,0,0);
                FONT-WEIGHT: normal; WORD-SPACING: 0px"><span
                  style="FONT-FAMILY: arial">Atenciosamente,<br>
                  <br>
                  Ivan P</span></span><span style="LINE-HEIGHT: normal;
                TEXT-TRANSFORM: none; FONT-VARIANT: normal; FONT-STYLE:
                normal; TEXT-INDENT: 0px; LETTER-SPACING: normal;
                BORDER-COLLAPSE: separate; FONT-FAMILY: 'Times New
                Roman'; WHITE-SPACE: normal; COLOR: rgb(0,0,0);
                FONT-WEIGHT: normal; WORD-SPACING: 0px"><span
                  style="FONT-FAMILY: arial"></span></span><span
                style="LINE-HEIGHT: normal; TEXT-TRANSFORM: none;
                FONT-VARIANT: normal; FONT-STYLE: normal; TEXT-INDENT:
                0px; LETTER-SPACING: normal; BORDER-COLLAPSE: separate;
                FONT-FAMILY: 'Times New Roman'; WHITE-SPACE: normal;
                COLOR: rgb(0,0,0); FONT-WEIGHT: normal; WORD-SPACING:
                0px"><span style="FONT-FAMILY: arial">aes Jos&eacute;<br>
                </span></span><span style="LINE-HEIGHT: normal;
                TEXT-TRANSFORM: none; FONT-VARIANT: normal; FONT-STYLE:
                normal; TEXT-INDENT: 0px; LETTER-SPACING: normal;
                BORDER-COLLAPSE: separate; FONT-FAMILY: 'Times New
                Roman'; WHITE-SPACE: normal; COLOR: rgb(0,0,0);
                FONT-WEIGHT: normal; WORD-SPACING: 0px"><span
                  style="FONT-FAMILY: arial"></span></span><br>
              <span style="LINE-HEIGHT: normal; TEXT-TRANSFORM: none;
                FONT-VARIANT: normal; FONT-STYLE: normal; TEXT-INDENT:
                0px; LETTER-SPACING: normal; BORDER-COLLAPSE: separate;
                FONT-FAMILY: 'Times New Roman'; WHITE-SPACE: normal;
                COLOR: rgb(0,0,0); FONT-WEIGHT: normal; WORD-SPACING:
                0px"><span style="FONT-FAMILY: arial">Acad&ecirc;mico de
                  Biblioteconomia - UFSC<br>
                </span></span><span style="LINE-HEIGHT: normal;
                TEXT-TRANSFORM: none; FONT-VARIANT: normal; FONT-STYLE:
                normal; TEXT-INDENT: 0px; LETTER-SPACING: normal;
                BORDER-COLLAPSE: separate; FONT-FAMILY: 'Times New
                Roman'; WHITE-SPACE: normal; COLOR: rgb(0,0,0);
                FONT-WEIGHT: normal; WORD-SPACING: 0px"><span
                  style="FONT-FAMILY: arial">T&eacute;cnico em Telecomunica&ccedil;&otilde;es
                  - IFSC<br>
                  <br>
                  E-mail/MSN/GTalk:<span>&nbsp;</span><a
                    moz-do-not-send="true"
                    href="mailto:ivan.paes@gmail.com" target="_blank">ivan.paes@gmail.com</a><br>
                  Oi: +55 48 84291055<br>
                </span></span><span style="LINE-HEIGHT: normal;
                TEXT-TRANSFORM: none; FONT-VARIANT: normal; FONT-STYLE:
                normal; TEXT-INDENT: 0px; LETTER-SPACING: normal;
                BORDER-COLLAPSE: separate; FONT-FAMILY: 'Times New
                Roman'; WHITE-SPACE: normal; COLOR: rgb(0,0,0);
                FONT-WEIGHT: normal; WORD-SPACING: 0px"><span
                  style="FONT-FAMILY: arial">Skype: ivanpaesjose<br>
                </span></span><span style="LINE-HEIGHT: normal;
                TEXT-TRANSFORM: none; FONT-VARIANT: normal; FONT-STYLE:
                normal; TEXT-INDENT: 0px; LETTER-SPACING: normal;
                BORDER-COLLAPSE: separate; FONT-FAMILY: 'Times New
                Roman'; WHITE-SPACE: normal; COLOR: rgb(0,0,0);
                FONT-WEIGHT: normal; WORD-SPACING: 0px"><span
                  style="FONT-FAMILY: arial">Palho&ccedil;a - Santa Catarina -
                  Brasil<br>
                  <br>
                  *** Muito Importante *** NETiqueta<br>
                  Se repassar esta mensagem, por gentileza:<br>
                  * Apague todos os endere&ccedil;os que aparecem nele.<br>
                  * E, por op&ccedil;&atilde;o de seguran&ccedil;a endere&ccedil;&aacute;-lo no Cco ou Bcc.<br>
                  Desta forma voc&ecirc; estar&aacute; protegendo a mim, seus amigos
                  e a voc&ecirc; mesmo.<br>
                  Eu, juntamente com a campanha contra a propaga&ccedil;&atilde;o de
                  v&iacute;rus agradecemos sinceramente.</span></span></font><br>
            <br>
            <br>
            <div class="gmail_quote">Em 29 de outubro de 2012 17:28,
              Ivan Maldonado Orosco <span dir="ltr">&lt;<a
                  moz-do-not-send="true"
                  href="mailto:ivanorosco@hotmail.com" target="_blank">ivanorosco@hotmail.com</a>&gt;</span>
              escreveu:<br>
              <blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN:
                0px 0px 0px 0.8ex; PADDING-LEFT: 1ex"
                class="gmail_quote">
                <div dir="ltr">
                  <div dir="ltr">
                    <div style="FONT-FAMILY: 'Calibri'; FONT-SIZE: 12pt">
                      <div>Boa tarde,</div>
                      <div>&nbsp;</div>
                      <div>Adquirimos um ata Dlink DVG 6004S e
                        configuramos o mesmo para utilizar nossas linhas
                        anal&oacute;gicas, no entanto, deparamos com um
                        problema aparentemente simples para se resolver,
                        no entanto, n&atilde;o conseguimos encontrar a solu&ccedil;&atilde;o.
                        O que acontece &eacute; que o ata recebe liga&ccedil;&otilde;es e
                        transfere para os ramais SIP corretamente
                        (entrada 100% funcional), mas na execu&ccedil;&atilde;o de
                        discagem, o asterisk comanda a discagem, o ata
                        faz a discagem, o destino atende a liga&ccedil;&atilde;o e os
                        dois pontos se falam, mas ap&oacute;s 20 segundos da
                        discagem a liga&ccedil;&atilde;o &eacute; derrubada pelo ata.</div>
                      <div>&nbsp;</div>
                      <div>Depurando as a&ccedil;&otilde;es dos comandos do ata pelo
                        SLmon (programinha da Dlink), vimos que o mesmo
                        efetua todo o processo de discagem e envia por
                        &uacute;ltimo o comando &#8220;==13: VoipAnswering&#8221; (com a
                        liga&ccedil;&atilde;o j&aacute; atendida) e ap&oacute;s 20 segundos ele
                        responde novamente com St_VoipAnswering Timeout
                        e derruba a liga&ccedil;&atilde;o.</div>
                      <div>&nbsp;</div>
                      <div>Vejam o log gerado:</div>
                      <div>&nbsp;</div>
                      <div>16:58:00 [010667] 5: 6720=OFFERING<br>
                        16:58:00 [010667] 5: Get CallerId=100<br>
                        16:58:00 [010668] 5: Check Trunk FixLine<br>
                        16:58:00 [010668] 5: Hunting Trunk Line<br>
                        16:58:00 [010668] 0: Peer PTime=20 #2<br>
                        16:58:00 [010668] 0: Peer=<a
                          moz-do-not-send="true"
                          href="http://189.2.20.138:19292"
                          target="_blank">189.2.20.138:19292</a>, PT=0,
                        RecvOnly=0<br>
                        16:58:00 [010668] 0: TrunkPrefix=,
                        Dest=39064886, Dialno=39064886<br>
                        16:58:00 [010668] 0: FxoHookOff<br>
                        16:58:00 [010668] 0: SetInputGain(-2)<br>
                        16:58:00 [010668] 0: ==18:TrunkDialOut<br>
                        16:58:01 [010679] 0: DialOut(39064886)=0<br>
                        16:58:02 [010692] 0: RtpApiTalk[1,1],Peer=<a
                          moz-do-not-send="true"
                          href="http://189.2.20.138:19292"
                          target="_blank">189.2.20.138:19292</a>,PT=0,FX=2,NewOOB=1<br>
                        16:58:02 [010692] 0: Substatus=3<br>
                        16:58:11 [010782] 0: Fxo Still No RingTone, Talk<br>
                        16:58:11 [010782] 0: Fxo DialOut OK<br>
                        16:58:11 [010782] 5: 6720=ACCEPT<br>
                        16:58:11 [010783] 0: ==13:VoipAnswering</div>
                      <div><br>
                        16:58:33 [010996] 0: St_VoipAnswering Timeout<br>
                        16:58:33 [010996] 0: DSP_ch0_check=0<br>
                        16:58:33 [010996] 0: FxoHookOn<br>
                        16:58:33 [010996] 0: ==15:PlayBusyTone<br>
                        16:58:33 [010996] 0: DSP_ch0_check=0<br>
                        16:58:33 [010996] 0: FxoHookOn<br>
                        16:58:33 [010996] 0: ==3:Idle<br>
                        16:58:33 [010996] 0: SetInputGain(4)<br>
                        16:58:33 [010996] 0: SetFax(1)=0<br>
                        16:58:33 [010996] 0: 6720=DISCONNECT</div>
                      <div>&nbsp;</div>
                      <div>J&aacute; fiz todas as configura&ccedil;&otilde;es poss&iacute;veis no
                        ATA e n&atilde;o houve nenhum resultado diferente que
                        fa&ccedil;a ele dar uma outra resposta ap&oacute;s o
                        VoipAnswering que n&atilde;o seja Timeout.</div>
                      <div>&nbsp;</div>
                      <div>As configura&ccedil;&otilde;es de meu tronco SIP s&atilde;o:</div>
                      <div>host=dynamic<br>
                        username=6720<br>
                        secret=<br>
                        type=friend<br>
                        qualify=yes<br>
                        canreinvite=yes<br>
                        dtmfmode=rfc2833<br>
                        alow=all<br>
                      </div>
                      <div>O ATA est&aacute; ligado diretamente na internet sem
                        firewal na frente, assim como nosso asterisk
                        (para evitar qualquer problema relacionado a
                        libera&ccedil;&atilde;o de portas)</div>
                      <div>&nbsp;</div>
                      <div>Algu&eacute;m tem alguma id&eacute;ia ?</div>
                      <div>&nbsp;</div>
                      <div>[]&#8217;s</div>
                      <div>&nbsp;</div>
                      <div><br>
                        &nbsp;</div>
                      <div>&nbsp;</div>
                    </div>
                  </div>
                </div>
                <br>
                _______________________________________________<br>
                KHOMP Inova&ccedil;&atilde;o: External Board Series<br>
                M&oacute;dulos de 1/2 rack e 1U para todas as interfaces e
                solu&ccedil;&otilde;es Asterisk e FreeSWITCH.<br>
                Tenha a External Series Experience na sua aplica&ccedil;&atilde;o.
                Visite <a moz-do-not-send="true"
                  href="http://www.khomp.com" target="_blank">www.khomp.com</a><br>
                _______________________________________________<br>
                DIGIVOICE&nbsp; Fabricante de Placas de Voz e Channel Bank<br>
                20 anos de experi&ecirc;ncia com E1(R2/ISDN), FXS, FXO e GSM<br>
                Centro Treinamento - Curso de PABX IP -&nbsp; Asterisk&nbsp; -
                Site&nbsp; <a moz-do-not-send="true"
                  href="http://www.digivoice.com.br" target="_blank">www.digivoice.com.br</a><br>
                ________<br>
                YEALINK: Telefones IP e V&iacute;deoPhones IP com o melhor
                custo/benef&iacute;cio do mercado.<br>
                email: <a moz-do-not-send="true"
                  href="mailto:yealink@commlogik.com.br">yealink@commlogik.com.br</a>
                | <a moz-do-not-send="true"
                  href="http://www.commlogik.com.br" target="_blank">www.commlogik.com.br</a>
                | <a moz-do-not-send="true"
                  href="tel:%2811%29%205503-1011" value="+551155031011">(11)
                  5503-1011</a><br>
                ______________________________________________<br>
                Para remover seu email desta lista, basta enviar um
                email em branco para <a moz-do-not-send="true"
                  href="mailto:asteriskbrasil-unsubscribe@listas.asteriskbrasil.org">asteriskbrasil-unsubscribe@listas.asteriskbrasil.org</a><br>
              </blockquote>
            </div>
            <br>
            <p>
            </p>
            <hr>
            _______________________________________________<br>
            KHOMP Inova&ccedil;&atilde;o: External Board Series<br>
            M&oacute;dulos de 1/2 rack e 1U para todas as interfaces e solu&ccedil;&otilde;es
            Asterisk e FreeSWITCH.<br>
            Tenha a External Series Experience na sua aplica&ccedil;&atilde;o. Visite
            <a class="moz-txt-link-abbreviated" href="http://www.khomp.com">www.khomp.com</a><br>
            _______________________________________________<br>
            DIGIVOICE&nbsp; Fabricante de Placas de Voz e Channel Bank<br>
            20 anos de experi&ecirc;ncia com E1(R2/ISDN), FXS, FXO e GSM<br>
            Centro Treinamento - Curso de PABX IP -&nbsp; Asterisk&nbsp; - Site&nbsp;
            <a class="moz-txt-link-abbreviated" href="http://www.digivoice.com.br">www.digivoice.com.br</a><br>
            ________<br>
            YEALINK: Telefones IP e V&iacute;deoPhones IP com o melhor
            custo/benef&iacute;cio do mercado.<br>
            email: <a class="moz-txt-link-abbreviated" href="mailto:yealink@commlogik.com.br">yealink@commlogik.com.br</a> | <a class="moz-txt-link-abbreviated" href="http://www.commlogik.com.br">www.commlogik.com.br</a> |
            (11) 5503-1011<br>
            ______________________________________________<br>
            Para remover seu email desta lista, basta enviar um email em
            branco para
            <a class="moz-txt-link-abbreviated" href="mailto:asteriskbrasil-unsubscribe@listas.asteriskbrasil.org">asteriskbrasil-unsubscribe@listas.asteriskbrasil.org</a></div>
        </div>
      </div>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_______________________________________________
KHOMP Inova&ccedil;&atilde;o: External Board Series
M&oacute;dulos de 1/2 rack e 1U para todas as interfaces e solu&ccedil;&otilde;es Asterisk e FreeSWITCH.
Tenha a External Series Experience na sua aplica&ccedil;&atilde;o. Visite <a class="moz-txt-link-abbreviated" href="http://www.khomp.com">www.khomp.com</a>
_______________________________________________
DIGIVOICE  Fabricante de Placas de Voz e Channel Bank
20 anos de experi&ecirc;ncia com E1(R2/ISDN), FXS, FXO e GSM
Centro Treinamento - Curso de PABX IP -  Asterisk  - Site  <a class="moz-txt-link-abbreviated" href="http://www.digivoice.com.br">www.digivoice.com.br</a>
________
YEALINK: Telefones IP e V&iacute;deoPhones IP com o melhor custo/benef&iacute;cio do mercado.
email: <a class="moz-txt-link-abbreviated" href="mailto:yealink@commlogik.com.br">yealink@commlogik.com.br</a> | <a class="moz-txt-link-abbreviated" href="http://www.commlogik.com.br">www.commlogik.com.br</a> | (11) 5503-1011
______________________________________________
Para remover seu email desta lista, basta enviar um email em branco para <a class="moz-txt-link-abbreviated" href="mailto:asteriskbrasil-unsubscribe@listas.asteriskbrasil.org">asteriskbrasil-unsubscribe@listas.asteriskbrasil.org</a></pre>
    </blockquote>
    <br>
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