<div dir="ltr">Hi Pavel,<div>Thanks for the reply,</div><div>I tried with Hangup(16) but same result</div><div><br></div><div>what I get in console is </div><div><br></div><div><div> -- Executing [01XXXXXXX@public:1] Ringing("SIP/10.2.2.75-00000004", "") in new stack</div><div> -- Executing [01XXXXXXX@public:2] Wait("SIP/10.2.2.75-00000004", "1") in new stack</div><div> -- Executing [01XXXXXXX@public:3] Set("SIP/10.2.2.75-00000004", "vxmlurl=http%3A%2F%2F127.0.0.1:8080%2Ftest%2Findex.jsp?ani=07XXXXXX") in new stack</div><div> -- Executing [01XXXXXXX@public:4] AGI("SIP/10.2.2.75-00000004", "agi://<a href="http://127.0.0.1/url=http%3A%2F%2F127.0.0.1:8080%2Ftest%2Findex.jsp?ani=07XXXXXX">127.0.0.1/url=http%3A%2F%2F127.0.0.1:8080%2Ftest%2Findex.jsp?ani=07XXXXXX</a>") in new stack</div><div> -- <SIP/10.2.2.75-00000004>AGI Script agi://<a href="http://127.0.0.1/url=http%3A%2F%2F127.0.0.1:8080%2Ftest%2Findex.jsp?ani=07XXXXXX">127.0.0.1/url=http%3A%2F%2F127.0.0.1:8080%2Ftest%2Findex.jsp?ani=07XXXXXX</a> completed, returning 4</div><div>[Feb 13 10:25:24] ERROR[10111][C-00000004]: utils.c:1321 ast_carefulwrite: write() returned error: Broken pipe</div></div><div><br></div><div><br></div><div><div><--- Reliably Transmitting (no NAT) to <a href="http://11.200.1.53:9131">11.200.1.53:9131</a> ---></div><div>SIP/2.0 603 Declined</div><div>Via: SIP/2.0/UDP 11.200.1.53:9131;branch=z9hG4bKkcw3bikyw336f07x6xt6wbbc6;received=10.200.1.53</div><div>From: <<a href="mailto:sip%3A07XXXXXX@11.2.2.75">sip:07XXXXXX@11.2.2.75</a>;user=phone>;tag=sbc0403xbxkbx6w-CC-22</div><div>To: <<a href="mailto:sip%3A01XXXXXXX@11.2.2.75">sip:01XXXXXXX@11.2.2.75</a>;user=phone>;tag=as02deb6c4</div><div>Call-ID: isbcyc60z60kyw17fk6e1y36i7ey7tz33y3i@SoftX3000</div><div>CSeq: 1 INVITE</div><div>Server: Asterisk PBX 11.8.0</div><div>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH</div><div>Supported: replaces, timer</div><div>Content-Length: 0</div><div><br></div><div><br></div><div><------------></div><div><br></div><div><--- SIP read from UDP:<a href="http://11.200.1.53:9131">11.200.1.53:9131</a> ---></div><div>ACK sip:01XXXXXXX@11.2.2.75:5060;user=phone SIP/2.0</div><div>Via: SIP/2.0/UDP 11.200.1.53:9131;branch=z9hG4bKkcw3bikyw336f07x6xt6wbbc6;received=11.200.1.53</div><div>Call-ID: isbcyc60z60kyw17fk6e1y36i7ey7tz33y3i@SoftX3000</div><div>From: <<a href="mailto:sip%3A07XXXXXX@11.2.2.75">sip:07XXXXXX@11.2.2.75</a>;user=phone>;tag=sbc0403xbxkbx6w-CC-22</div><div>To: <<a href="mailto:sip%3A01XXXXXXX@11.2.2.75">sip:01XXXXXXX@11.2.2.75</a>;user=phone>;tag=as02deb6c4</div><div>CSeq: 1 ACK</div><div>Max-Forwards: 70</div><div>Content-Length: 0</div></div><div><br></div><div>I am running Asterisk 11.8.0 with voiceglue 0.14</div><div><br></div><div>When I try the .jsp page with wget, it works fine without any error.</div><div><br></div><div>Regards,</div><div>Roy.</div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Feb 12, 2015 at 4:58 PM, Pavel Troller <span dir="ltr"><<a href="mailto:patrol@sinus.cz" target="_blank">patrol@sinus.cz</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi Roy,<br>
<div><div class="h5"><br>
> Hi Friends,<br>
><br>
> I am trying to implement a simple dial plan with asterisk.<br>
> 1. Ring the inbound call<br>
> 2. wait for 2 seconds<br>
> 3. call agi script with cli<br>
> 4. hangup<br>
><br>
> But when it gets hangup I see the Declined is passed from the asterisk.<br>
> But my PSTN provider keep waiting in dialing state with no noise until 60<br>
> seconds.<br>
><br>
> any idea what I have done wrong.<br>
><br>
> here is my dial plan<br>
><br>
> exten => 01XXXX,1,Ringing()<br>
> exten => 01XXXX,2,Wait(2)<br>
> exten =><br>
> 01XXXX,3,Set(vxmlurl=http%3A%2F%2F127.0.0.1:8080%2Ftest%2Findex.jsp?ani=${CALLERID(num)})<br>
> exten => 01XXXX,4,Agi(agi://<a href="http://127.0.0.1/url=${vxmlurl}" target="_blank">127.0.0.1/url=${vxmlurl}</a>)<br>
> exten => 01XXXX,5,Hangup()<br>
<br>
</div></div>Did you try to put a specific Clear Cause code to the Hangup command ?<br>
For example, Hangup(17) means User Busy, i.e. 486 Busy Here on SIP, or<br>
Hangup(1) means Unallocated Number or 404 Not Found etc. If you want to<br>
find more codes, search for ISDN Cause Codes. Your PSTN provider will<br>
probably react better to other causes than the default one, which is<br>
probably 16 Normal Call Clearing.<br>
<br>
With regards,<br>
Pavel<br>
<div class="HOEnZb"><div class="h5"><br>
><br>
> Please advice,<br>
><br>
> Best Regards,<br>
> Roy.<br>
<br>
<br>
</div></div><span class="HOEnZb"><font color="#888888">--<br>
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