<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&g729 -----<<<<<-----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"><<a href="mailto:alyed@vivoxie.com" target="_blank">alyed@vivoxie.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">it doesn't seems to be a problem of communication between A and B
<div><br><br>> -- Executing [s@macro-dialout-trunk:19] Dial("SIP/192.168.0.151-088e7938", "ZAP/g0/15921256331|300|M(setmusic^none)Tt") in new stack<br>
<div>> == Everyone is busy/congested at this time (1:0/0/1)<br></div><br></div>That's says it'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"><<a href="mailto:evane1890@gmail.com" target="_blank">evane1890@gmail.com</a>></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'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><--- SIP read from <a href="http://192.168.0.176:5060/" target="_blank">192.168.0.176:5060</a> ---><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: "50005" <<a href="mailto:sip%3A50005@192.168.0.151" target="_blank">sip:50005@192.168.0.151</a>>;tag=as72a55960<br>To: <<a href="mailto:sip%3A15921256331@192.168.0.151" target="_blank">sip:15921256331@192.168.0.151</a>><br>
Contact: <<a href="mailto:sip%3A50005@192.168.0.176" target="_blank">sip:50005@192.168.0.176</a>><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><-------------><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 's1'<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: <<a href="mailto:sip%3A50005@192.168.0.176" target="_blank">sip:50005@192.168.0.176</a>><br>gd-branch*CLI> <br><--- Transmitting (NAT) to <a href="http://192.168.0.176:5060/" target="_blank">192.168.0.176:5060</a> ---><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: "50005" <<a href="mailto:sip%3A50005@192.168.0.151" target="_blank">sip:50005@192.168.0.151</a>>;tag=as72a55960<br>
To: <<a href="mailto:sip%3A15921256331@192.168.0.151" target="_blank">sip:15921256331@192.168.0.151</a>><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: <<a href="mailto:sip%3A15921256331@192.168.0.151" target="_blank">sip:15921256331@192.168.0.151</a>><br>
Content-Length: 0</div>
<div><br><------------><br> -- Executing [15921256331@from-internal:1] Set("SIP/192.168.0.151-088e7938", "MOHCLASS=none") in new stack<br> -- Executing [15921256331@from-internal:2] Macro("SIP/192.168.0.151-088e7938", "user-callerid|SKIPTTL|") in new stack<br>
-- Executing [s@macro-user-callerid:1] Set("SIP/192.168.0.151-088e7938", "AMPUSER=50005") in new stack<br> -- Executing [s@macro-user-callerid:2] GotoIf("SIP/192.168.0.151-088e7938", "0?report") in new stack<br>
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/192.168.0.151-088e7938", "1|Set|REALCALLERIDNUM=50005") in new stack<br> -- Executing [s@macro-user-callerid:4] Set("SIP/192.168.0.151-088e7938", "AMPUSER=") in new stack<br>
-- Executing [s@macro-user-callerid:5] Set("SIP/192.168.0.151-088e7938", "AMPUSERCIDNAME=") in new stack<br> -- Executing [s@macro-user-callerid:6] GotoIf("SIP/192.168.0.151-088e7938", "1?report") in new stack<br>
-- Goto (macro-user-callerid,s,10)<br> -- Executing [s@macro-user-callerid:10] GotoIf("SIP/192.168.0.151-088e7938", "1?continue") in new stack<br> -- Goto (macro-user-callerid,s,19)<br> -- Executing [s@macro-user-callerid:19] NoOp("SIP/192.168.0.151-088e7938", "Using CallerID "50005" <50005>") in new stack<br>
-- Executing [15921256331@from-internal:3] Set("SIP/192.168.0.151-088e7938", "_NODEST=") in new stack<br> -- Executing [15921256331@from-internal:4] Macro("SIP/192.168.0.151-088e7938", "record-enable||OUT|") in new stack<br>
-- Executing [s@macro-record-enable:1] GotoIf("SIP/192.168.0.151-088e7938", "1?check") in new stack<br> -- Goto (macro-record-enable,s,4)<br> -- Executing [s@macro-record-enable:4] AGI("SIP/192.168.0.151-088e7938", "recordingcheck|20100326-101436|1269569676.20") 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("SIP/192.168.0.151-088e7938", "") in new stack<br> -- Executing [15921256331@from-internal:5] Macro("SIP/192.168.0.151-088e7938", "dialout-trunk|1|15921256331||") in new stack<br>
-- Executing [s@macro-dialout-trunk:1] Set("SIP/192.168.0.151-088e7938", "DIAL_TRUNK=1") in new stack<br> -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/192.168.0.151-088e7938", "0?sub-pincheck|s|1") in new stack<br>
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/192.168.0.151-088e7938", "0?disabletrunk|1") in new stack<br> -- Executing [s@macro-dialout-trunk:4] Set("SIP/192.168.0.151-088e7938", "DIAL_NUMBER=15921256331") in new stack<br>
-- Executing [s@macro-dialout-trunk:5] Set("SIP/192.168.0.151-088e7938", "DIAL_TRUNK_OPTIONS=Ttr") in new stack<br> -- Executing [s@macro-dialout-trunk:6] Set("SIP/192.168.0.151-088e7938", "OUTBOUND_GROUP=OUT_1") in new stack<br>
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/192.168.0.151-088e7938", "1?nomax") in new stack<br> -- Goto (macro-dialout-trunk,s,9)<br> -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/192.168.0.151-088e7938", "0?skipoutcid") in new stack<br>
-- Executing [s@macro-dialout-trunk:10] Set("SIP/192.168.0.151-088e7938", "DIAL_TRUNK_OPTIONS=Tt") in new stack<br> -- Executing [s@macro-dialout-trunk:11] Macro("SIP/192.168.0.151-088e7938", "outbound-callerid|1") in new stack<br>
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/192.168.0.151-088e7938", "0|SetCallerPres|") in new stack<br> -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/192.168.0.151-088e7938", "0|Set|REALCALLERIDNUM=50005") in new stack<br>
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/192.168.0.151-088e7938", "1?normcid") in new stack<br> -- Goto (macro-outbound-callerid,s,6)<br> -- Executing [s@macro-outbound-callerid:6] Set("SIP/192.168.0.151-088e7938", "USEROUTCID=") in new stack<br>
-- Executing [s@macro-outbound-callerid:7] Set("SIP/192.168.0.151-088e7938", "EMERGENCYCID=") in new stack<br> -- Executing [s@macro-outbound-callerid:8] Set("SIP/192.168.0.151-088e7938", "TRUNKOUTCID=64858162") in new stack<br>
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/192.168.0.151-088e7938", "1?trunkcid") in new stack<br> -- Goto (macro-outbound-callerid,s,12)<br> -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/192.168.0.151-088e7938", "1|Set|CALLERID(all)=64858162") in new stack<br>
-- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/192.168.0.151-088e7938", "0|Set|CALLERID(all)=") in new stack<br> -- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/192.168.0.151-088e7938", "0|SetCallerPres|prohib_passed_screen") in new stack<br>
-- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/192.168.0.151-088e7938", "0|AGI|fixlocalprefix") in new stack<br> -- Executing [s@macro-dialout-trunk:13] Set("SIP/192.168.0.151-088e7938", "OUTNUM=15921256331") in new stack<br>
-- Executing [s@macro-dialout-trunk:14] Set("SIP/192.168.0.151-088e7938", "custom=ZAP/g0") in new stack<br> -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/192.168.0.151-088e7938", "1|Set|DIAL_TRUNK_OPTIONS=M(setmusic^none)Tt") in new stack<br>
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/192.168.0.151-088e7938", "dialout-trunk-predial-hook|") in new stack<br> -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/192.168.0.151-088e7938", "") in new stack<br>
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/192.168.0.151-088e7938", "0?bypass|1") in new stack<br> -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/192.168.0.151-088e7938", "0?customtrunk") in new stack<br>
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/192.168.0.151-088e7938", "ZAP/g0/15921256331|300|M(setmusic^none)Tt") in new stack<br> == Everyone is busy/congested at this time (1:0/0/1)<br> -- Executing [s@macro-dialout-trunk:20] Goto("SIP/192.168.0.151-088e7938", "s-CHANUNAVAIL|1") in new stack<br>
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)<br> -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/192.168.0.151-088e7938", "1?noreport") in new stack<br> -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)<br>
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/192.168.0.151-088e7938", "TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 58) - failing through to other trunks") in new stack<br> -- Executing [15921256331@from-internal:6] Macro("SIP/192.168.0.151-088e7938", "outisbusy|") in new stack<br>
-- Executing [s@macro-outisbusy:1] Playback("SIP/192.168.0.151-088e7938", "all-circuits-busy-now|noanswer") in new stack<br> -- Executing [s@macro-outisbusy:2] Playback("SIP/192.168.0.151-088e7938", "pls-try-call-later|noanswer") in new stack<br>
-- Executing [s@macro-outisbusy:3] Macro("SIP/192.168.0.151-088e7938", "hangupcall") in new stack<br> -- Executing [s@macro-hangupcall:1] GotoIf("SIP/192.168.0.151-088e7938", "1?skiprg") in new stack<br>
-- Goto (macro-hangupcall,s,4)<br> -- Executing [s@macro-hangupcall:4] GotoIf("SIP/192.168.0.151-088e7938", "1?skipblkvm") in new stack<br> -- Goto (macro-hangupcall,s,7)<br> -- Executing [s@macro-hangupcall:7] GotoIf("SIP/192.168.0.151-088e7938", "1?theend") in new stack<br>
-- Goto (macro-hangupcall,s,9)<br> -- Executing [s@macro-hangupcall:9] Hangup("SIP/192.168.0.151-088e7938", "") in new stack<br> == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/192.168.0.151-088e7938' in macro 'hangupcall'<br>
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/192.168.0.151-088e7938' in macro 'outisbusy'<br> == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/192.168.0.151-088e7938'<br>
Scheduling destruction of SIP dialog <a href="mailto:%2728272ebb12ee6e4c1f06fca651456469@192.168.0.151%27" target="_blank">'28272ebb12ee6e4c1f06fca651456469@192.168.0.151'</a> in 6400 ms (Method: INVITE)<br>gd-branch*CLI> <br>
<--- Reliably Transmitting (NAT) to <a href="http://192.168.0.176:5060/" target="_blank">192.168.0.176:5060</a> ---><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: "50005" <<a href="mailto:sip%3A50005@192.168.0.151" target="_blank">sip:50005@192.168.0.151</a>>;tag=as72a55960<br>To: <<a href="mailto:sip%3A15921256331@192.168.0.151" target="_blank">sip:15921256331@192.168.0.151</a>>;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><------------><br>gd-branch*CLI> <br><--- SIP read from <a href="http://192.168.0.176:5060/" target="_blank">192.168.0.176:5060</a> ---><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: "50005" <<a href="mailto:sip%3A50005@192.168.0.151" target="_blank">sip:50005@192.168.0.151</a>>;tag=as72a55960<br>To: <<a href="mailto:sip%3A15921256331@192.168.0.151" target="_blank">sip:15921256331@192.168.0.151</a>>;tag=as12db2697<br>
Contact: <<a href="mailto:sip%3A50005@192.168.0.176" target="_blank">sip:50005@192.168.0.176</a>><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><-------------><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>