<div>I got it !!</div>
<div> </div>
<div> </div>
<div>host=192.168.0.151<br>port=5060<br>type=friend<br>nat=yes<br>qualify=yes<br>fromdomain=192.168.0.151<br>insecure=invite,port<br>dtmfmode=auto<br>disallow=all<br>allow=alaw&amp;g729 -----&lt;&lt;&lt;&lt;&lt;-----here! make a tention at the order! G729 is not allowed !</div>

<div> </div>
<div>i reorder it get work!!</div>
<div> </div>
<div> </div>
<div>thks a lot,all !<br></div>
<div><br><br> </div>
<div class="gmail_quote">On 26 March 2010 13:44, Alyed <span dir="ltr">&lt;<a href="mailto:alyed@vivoxie.com" target="_blank">alyed@vivoxie.com</a>&gt;</span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">it doesn&#39;t seems to be a problem of communication between A and B 
<div><br><br>&gt;    -- Executing [s@macro-dialout-trunk:19] Dial(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;ZAP/g0/15921256331|300|M(setmusic^none)Tt&quot;) in new stack<br>
<div>&gt; == Everyone is busy/congested at this time (1:0/0/1)<br></div><br></div>That&#39;s says it&#39;s more a problem with your Zap channels than your SIP connection.<br><br>First try playing a sound in B when receiving the call, that way you can be sure the connection is ok. If that one works then move to PSTN.<br>
<br>Alyed<br><br><br>
<div class="gmail_quote">2010/3/25 Aaron chen <span dir="ltr">&lt;<a href="mailto:evane1890@gmail.com" target="_blank">evane1890@gmail.com</a>&gt;</span><br>
<blockquote style="BORDER-LEFT: rgb(204,204,204) 1px solid; MARGIN: 0pt 0pt 0pt 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div>
<div></div>
<div>
<div>i have a prablom here,</div>
<div> </div>
<div>i want to send a call from A to B use sip trunk ,</div>
<div> </div>
<div>the call can sended B,but not work to PSTN.</div>
<div> </div>
<div>the message from B server. help pls,what&#39;s rong? </div>
<div> </div>
<div> </div>
<div>
<blockquote style="BORDER-LEFT: rgb(204,204,204) 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div> </div>
<div>&lt;--- SIP read from <a href="http://192.168.0.176:5060/" target="_blank">192.168.0.176:5060</a> ---&gt;<br>INVITE <a href="mailto:sip%3A15921256331@192.168.0.151" target="_blank">sip:15921256331@192.168.0.151</a> SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport<br>From: &quot;50005&quot; &lt;<a href="mailto:sip%3A50005@192.168.0.151" target="_blank">sip:50005@192.168.0.151</a>&gt;;tag=as72a55960<br>To: &lt;<a href="mailto:sip%3A15921256331@192.168.0.151" target="_blank">sip:15921256331@192.168.0.151</a>&gt;<br>
Contact: &lt;<a href="mailto:sip%3A50005@192.168.0.176" target="_blank">sip:50005@192.168.0.176</a>&gt;<br>Call-ID: <a href="mailto:28272ebb12ee6e4c1f06fca651456469@192.168.0.151" target="_blank">28272ebb12ee6e4c1f06fca651456469@192.168.0.151</a><br>
CSeq: 102 INVITE<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>Date: Fri, 26 Mar 2010 02:12:07 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>Supported: replaces<br>Content-Type: application/sdp<br>
Content-Length: 380</div>
<div>v=0<br>o=root 15081 15081 IN IP4 192.168.0.176<br>s=session<br>c=IN IP4 192.168.0.176<br>t=0 0<br>m=audio 12726 RTP/AVP 0 18 8 3 4 101<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:18 G729/8000<br>a=fmtp:18 annexb=no<br>a=rtpmap:8 PCMA/8000<br>
a=rtpmap:3 GSM/8000<br>a=rtpmap:4 G723/8000<br>a=fmtp:4 annexa=no<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv</div>
<div>&lt;-------------&gt;<br>--- (14 headers 18 lines) ---<br>Sending to 192.168.0.176 : 5060 (NAT)<br>Using INVITE request as basis request - <a href="mailto:28272ebb12ee6e4c1f06fca651456469@192.168.0.151" target="_blank">28272ebb12ee6e4c1f06fca651456469@192.168.0.151</a><br>
Found peer &#39;s1&#39;<br>Found RTP audio format 0<br>Found RTP audio format 18<br>Found RTP audio format 8<br>Found RTP audio format 3<br>Found RTP audio format 4<br>Found RTP audio format 101<br>Peer audio RTP is at port <a href="http://192.168.0.176:12726/" target="_blank">192.168.0.176:12726</a><br>
Found audio description format PCMU for ID 0<br>Found audio description format G729 for ID 18<br>Found audio description format PCMA for ID 8<br>Found audio description format GSM for ID 3<br>Found audio description format G723 for ID 4<br>
Found audio description format telephone-event for ID 101<br>Capabilities: us - 0x10f (g723|gsm|ulaw|alaw|g729), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10f (g723|gsm|ulaw|alaw|g729)<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://192.168.0.176:12726/" target="_blank">192.168.0.176:12726</a><br>
Looking for 15921256331 in from-internal (domain 192.168.0.151)<br>list_route: hop: &lt;<a href="mailto:sip%3A50005@192.168.0.176" target="_blank">sip:50005@192.168.0.176</a>&gt;<br>gd-branch*CLI&gt; <br>&lt;--- Transmitting (NAT) to <a href="http://192.168.0.176:5060/" target="_blank">192.168.0.176:5060</a> ---&gt;<br>
SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;received=192.168.0.176;rport=5060<br>From: &quot;50005&quot; &lt;<a href="mailto:sip%3A50005@192.168.0.151" target="_blank">sip:50005@192.168.0.151</a>&gt;;tag=as72a55960<br>
To: &lt;<a href="mailto:sip%3A15921256331@192.168.0.151" target="_blank">sip:15921256331@192.168.0.151</a>&gt;<br>Call-ID: <a href="mailto:28272ebb12ee6e4c1f06fca651456469@192.168.0.151" target="_blank">28272ebb12ee6e4c1f06fca651456469@192.168.0.151</a><br>
CSeq: 102 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Contact: &lt;<a href="mailto:sip%3A15921256331@192.168.0.151" target="_blank">sip:15921256331@192.168.0.151</a>&gt;<br>
Content-Length: 0</div>
<div><br>&lt;------------&gt;<br>    -- Executing [15921256331@from-internal:1] Set(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;MOHCLASS=none&quot;) in new stack<br>    -- Executing [15921256331@from-internal:2] Macro(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;user-callerid|SKIPTTL|&quot;) in new stack<br>
    -- Executing [s@macro-user-callerid:1] Set(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;AMPUSER=50005&quot;) in new stack<br>    -- Executing [s@macro-user-callerid:2] GotoIf(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;0?report&quot;) in new stack<br>
    -- Executing [s@macro-user-callerid:3] ExecIf(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;1|Set|REALCALLERIDNUM=50005&quot;) in new stack<br>    -- Executing [s@macro-user-callerid:4] Set(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;AMPUSER=&quot;) in new stack<br>
    -- Executing [s@macro-user-callerid:5] Set(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;AMPUSERCIDNAME=&quot;) in new stack<br>    -- Executing [s@macro-user-callerid:6] GotoIf(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;1?report&quot;) in new stack<br>
    -- Goto (macro-user-callerid,s,10)<br>    -- Executing [s@macro-user-callerid:10] GotoIf(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;1?continue&quot;) in new stack<br>    -- Goto (macro-user-callerid,s,19)<br>    -- Executing [s@macro-user-callerid:19] NoOp(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;Using CallerID &quot;50005&quot; &lt;50005&gt;&quot;) in new stack<br>
    -- Executing [15921256331@from-internal:3] Set(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;_NODEST=&quot;) in new stack<br>    -- Executing [15921256331@from-internal:4] Macro(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;record-enable||OUT|&quot;) in new stack<br>
    -- Executing [s@macro-record-enable:1] GotoIf(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;1?check&quot;) in new stack<br>    -- Goto (macro-record-enable,s,4)<br>    -- Executing [s@macro-record-enable:4] AGI(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;recordingcheck|20100326-101436|1269569676.20&quot;) in new stack<br>
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck<br>  recordingcheck|20100326-101436|1269569676.20: No AMPUSER db entry for . Not recording<br>    -- AGI Script recordingcheck completed, returning 0<br>
    -- Executing [s@macro-record-enable:5] MacroExit(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;&quot;) in new stack<br>    -- Executing [15921256331@from-internal:5] Macro(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;dialout-trunk|1|15921256331||&quot;) in new stack<br>
    -- Executing [s@macro-dialout-trunk:1] Set(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;DIAL_TRUNK=1&quot;) in new stack<br>    -- Executing [s@macro-dialout-trunk:2] GosubIf(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;0?sub-pincheck|s|1&quot;) in new stack<br>
    -- Executing [s@macro-dialout-trunk:3] GotoIf(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;0?disabletrunk|1&quot;) in new stack<br>    -- Executing [s@macro-dialout-trunk:4] Set(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;DIAL_NUMBER=15921256331&quot;) in new stack<br>
    -- Executing [s@macro-dialout-trunk:5] Set(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;DIAL_TRUNK_OPTIONS=Ttr&quot;) in new stack<br>    -- Executing [s@macro-dialout-trunk:6] Set(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;OUTBOUND_GROUP=OUT_1&quot;) in new stack<br>
    -- Executing [s@macro-dialout-trunk:7] GotoIf(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;1?nomax&quot;) in new stack<br>    -- Goto (macro-dialout-trunk,s,9)<br>    -- Executing [s@macro-dialout-trunk:9] GotoIf(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;0?skipoutcid&quot;) in new stack<br>
    -- Executing [s@macro-dialout-trunk:10] Set(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;DIAL_TRUNK_OPTIONS=Tt&quot;) in new stack<br>    -- Executing [s@macro-dialout-trunk:11] Macro(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;outbound-callerid|1&quot;) in new stack<br>
    -- Executing [s@macro-outbound-callerid:1] ExecIf(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;0|SetCallerPres|&quot;) in new stack<br>    -- Executing [s@macro-outbound-callerid:2] ExecIf(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;0|Set|REALCALLERIDNUM=50005&quot;) in new stack<br>
    -- Executing [s@macro-outbound-callerid:3] GotoIf(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;1?normcid&quot;) in new stack<br>    -- Goto (macro-outbound-callerid,s,6)<br>    -- Executing [s@macro-outbound-callerid:6] Set(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;USEROUTCID=&quot;) in new stack<br>
    -- Executing [s@macro-outbound-callerid:7] Set(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;EMERGENCYCID=&quot;) in new stack<br>    -- Executing [s@macro-outbound-callerid:8] Set(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;TRUNKOUTCID=64858162&quot;) in new stack<br>
    -- Executing [s@macro-outbound-callerid:9] GotoIf(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;1?trunkcid&quot;) in new stack<br>    -- Goto (macro-outbound-callerid,s,12)<br>    -- Executing [s@macro-outbound-callerid:12] ExecIf(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;1|Set|CALLERID(all)=64858162&quot;) in new stack<br>
    -- Executing [s@macro-outbound-callerid:13] ExecIf(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;0|Set|CALLERID(all)=&quot;) in new stack<br>    -- Executing [s@macro-outbound-callerid:14] ExecIf(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;0|SetCallerPres|prohib_passed_screen&quot;) in new stack<br>
    -- Executing [s@macro-dialout-trunk:12] ExecIf(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;0|AGI|fixlocalprefix&quot;) in new stack<br>    -- Executing [s@macro-dialout-trunk:13] Set(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;OUTNUM=15921256331&quot;) in new stack<br>
    -- Executing [s@macro-dialout-trunk:14] Set(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;custom=ZAP/g0&quot;) in new stack<br>    -- Executing [s@macro-dialout-trunk:15] ExecIf(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;1|Set|DIAL_TRUNK_OPTIONS=M(setmusic^none)Tt&quot;) in new stack<br>
    -- Executing [s@macro-dialout-trunk:16] Macro(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;dialout-trunk-predial-hook|&quot;) in new stack<br>    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;&quot;) in new stack<br>
    -- Executing [s@macro-dialout-trunk:17] GotoIf(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;0?bypass|1&quot;) in new stack<br>    -- Executing [s@macro-dialout-trunk:18] GotoIf(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;0?customtrunk&quot;) in new stack<br>
    -- Executing [s@macro-dialout-trunk:19] Dial(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;ZAP/g0/15921256331|300|M(setmusic^none)Tt&quot;) in new stack<br>  == Everyone is busy/congested at this time (1:0/0/1)<br>    -- Executing [s@macro-dialout-trunk:20] Goto(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;s-CHANUNAVAIL|1&quot;) in new stack<br>
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)<br>    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;1?noreport&quot;) in new stack<br>    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)<br>
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 58) - failing through to other trunks&quot;) in new stack<br>    -- Executing [15921256331@from-internal:6] Macro(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;outisbusy|&quot;) in new stack<br>
    -- Executing [s@macro-outisbusy:1] Playback(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;all-circuits-busy-now|noanswer&quot;) in new stack<br>    -- Executing [s@macro-outisbusy:2] Playback(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;pls-try-call-later|noanswer&quot;) in new stack<br>
    -- Executing [s@macro-outisbusy:3] Macro(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;hangupcall&quot;) in new stack<br>    -- Executing [s@macro-hangupcall:1] GotoIf(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;1?skiprg&quot;) in new stack<br>
    -- Goto (macro-hangupcall,s,4)<br>    -- Executing [s@macro-hangupcall:4] GotoIf(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;1?skipblkvm&quot;) in new stack<br>    -- Goto (macro-hangupcall,s,7)<br>    -- Executing [s@macro-hangupcall:7] GotoIf(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;1?theend&quot;) in new stack<br>
    -- Goto (macro-hangupcall,s,9)<br>    -- Executing [s@macro-hangupcall:9] Hangup(&quot;SIP/192.168.0.151-088e7938&quot;, &quot;&quot;) in new stack<br>  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on &#39;SIP/192.168.0.151-088e7938&#39; in macro &#39;hangupcall&#39;<br>
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on &#39;SIP/192.168.0.151-088e7938&#39; in macro &#39;outisbusy&#39;<br>  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on &#39;SIP/192.168.0.151-088e7938&#39;<br>
Scheduling destruction of SIP dialog <a href="mailto:%2728272ebb12ee6e4c1f06fca651456469@192.168.0.151%27" target="_blank">&#39;28272ebb12ee6e4c1f06fca651456469@192.168.0.151&#39;</a> in 6400 ms (Method: INVITE)<br>gd-branch*CLI&gt; <br>
&lt;--- Reliably Transmitting (NAT) to <a href="http://192.168.0.176:5060/" target="_blank">192.168.0.176:5060</a> ---&gt;<br>SIP/2.0 488 Not Acceptable Here<br>Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;received=192.168.0.176;rport=5060<br>
From: &quot;50005&quot; &lt;<a href="mailto:sip%3A50005@192.168.0.151" target="_blank">sip:50005@192.168.0.151</a>&gt;;tag=as72a55960<br>To: &lt;<a href="mailto:sip%3A15921256331@192.168.0.151" target="_blank">sip:15921256331@192.168.0.151</a>&gt;;tag=as12db2697<br>
Call-ID: <a href="mailto:28272ebb12ee6e4c1f06fca651456469@192.168.0.151" target="_blank">28272ebb12ee6e4c1f06fca651456469@192.168.0.151</a><br>CSeq: 102 INVITE<br>User-Agent: Asterisk PBX<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Supported: replaces<br>Content-Length: 0</div>
<div><br>&lt;------------&gt;<br>gd-branch*CLI&gt; <br>&lt;--- SIP read from <a href="http://192.168.0.176:5060/" target="_blank">192.168.0.176:5060</a> ---&gt;<br>ACK <a href="mailto:sip%3A15921256331@192.168.0.151" target="_blank">sip:15921256331@192.168.0.151</a> SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport<br>From: &quot;50005&quot; &lt;<a href="mailto:sip%3A50005@192.168.0.151" target="_blank">sip:50005@192.168.0.151</a>&gt;;tag=as72a55960<br>To: &lt;<a href="mailto:sip%3A15921256331@192.168.0.151" target="_blank">sip:15921256331@192.168.0.151</a>&gt;;tag=as12db2697<br>
Contact: &lt;<a href="mailto:sip%3A50005@192.168.0.176" target="_blank">sip:50005@192.168.0.176</a>&gt;<br>Call-ID: <a href="mailto:28272ebb12ee6e4c1f06fca651456469@192.168.0.151" target="_blank">28272ebb12ee6e4c1f06fca651456469@192.168.0.151</a><br>
CSeq: 102 ACK<br>User-Agent: Asterisk PBX<br>Max-Forwards: 70<br>Content-Length: 0</div>
<div><br>&lt;-------------&gt;<br>--- (10 headers 0 lines) ---<br>sip no debug<br>SIP Debugging Disabled</div></blockquote></div>
<div> </div>
<div> </div>
<div>Best regards!</div>
<div><br>Aaron Chen <br></div><br></div></div>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>              <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br>
  <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br></blockquote></div><br><br clear="all"><br>--<br>_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com/" target="_blank">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br>              <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br>  <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</blockquote></div><br><br clear="all"><br>-- <br>祝您愉快!!<br><br>Aaron Chen <br>陈江涛<br>