<p>B.H.</p>
<p>But if the final response is 480 doesn't it mean that the call was placed but there was no reply?<br>
</p>
<div class="gmail_quote">On Aug 13, 2013 10:30 PM, "Shishir Pokharel" <<a href="mailto:Shishir.Pokharel@on24.com">Shishir.Pokharel@on24.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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<p class="MsoNormal">
<a name="14079286130689b2_section-21.1.5"></a><a href="http://tools.ietf.org/html/rfc3261#section-21.1.5" target="_blank"><b><span style="font-family:"Courier New"">21.1.5</span></b></a><b><span style="font-family:"Courier New""> 183 Session Progress<u></u><u></u></span></b></p>
<p class="MsoNormal"><span style="font-family:"Courier New""><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-family:"Courier New""><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-family:"Courier New""> The 183 (Session Progress) response is used to convey information<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-family:"Courier New""> about the progress of the call that is not otherwise classified. The<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-family:"Courier New""> Reason-Phrase, header fields, or message body MAY be used to convey<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-family:"Courier New""> more details about the call progress.<u></u><u></u></span></p>
<p class="MsoNormal"><b><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></b></p>
<h4><a name="14079286130689b2_section-21.1.2"></a><a href="http://tools.ietf.org/html/rfc3261#section-21.1.2" target="_blank"><span style="font-family:"Courier New"">21.1.2</span></a><span style="font-family:"Courier New"">
180 Ringing<u></u><u></u></span></h4>
<pre><span style="font-size:12.0pt"><u></u> <u></u></span></pre>
<pre><span style="font-size:12.0pt"><u></u> <u></u></span></pre>
<pre><span style="font-size:12.0pt"> The UA receiving the INVITE is trying to alert the user. This<u></u><u></u></span></pre>
<pre><span style="font-size:12.0pt"> response MAY be used to initiate local ringback.<u></u><u></u></span></pre>
<p class="MsoNormal"><b><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></b></p>
<p class="MsoNormal"><a href="http://tools.ietf.org/html/rfc3261#section-21.1.2" target="_blank">http://tools.ietf.org/html/rfc3261#section-21.1.2</a><b><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u><u></u></span></b></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>]
<b>On Behalf Of </b>Mordechay Kaganer<br>
<b>Sent:</b> Tuesday, August 13, 2013 10:55 AM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> Re: [asterisk-users] SIP trunk and congestion handling<u></u><u></u></span></p>
<p class="MsoNormal"><u></u> <u></u></p>
<p>B.H.<u></u><u></u></p>
<p>Asterisk 1.8.22<u></u><u></u></p>
<p>Thanks<u></u><u></u></p>
<div>
<p class="MsoNormal">On Aug 12, 2013 8:05 PM, "Shishir Pokharel" <<a href="mailto:Shishir.Pokharel@on24.com" target="_blank">Shishir.Pokharel@on24.com</a>> wrote:<u></u><u></u></p>
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<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Which version of asterisk are you using ?
</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"> </span><u></u><u></u></p>
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">
<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>]
<b>On Behalf Of </b>Mordechay Kaganer<br>
<b>Sent:</b> Sunday, August 11, 2013 8:59 AM<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> [asterisk-users] SIP trunk and congestion handling</span><u></u><u></u></p>
<p class="MsoNormal"> <u></u><u></u></p>
<div>
<p class="MsoNormal">B.H.<u></u><u></u></p>
<div>
<p class="MsoNormal"> <u></u><u></u></p>
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<p class="MsoNormal">Hello, all. We have a dialer software that runs outgoing telephony campaigns. We have been using it successfully with PRI cards, now we're evaluating it's use also with a SIP trunk.
Most of the things run perfectly good without a need to change anything except for dial string, but there's some strange problem with asterisk interpreting SIP result codes. <u></u><u></u></p>
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<p class="MsoNormal"> <u></u><u></u></p>
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<p class="MsoNormal">Our software is written in Java using asterisk-java library. It is using Asterisk's reason code from OriginateResponseEvent to determine if it should redial the number. Our consideration
is that if Asterisk returns reason code 8 (Congestion) this means that the call has never actually reached the destination number, and it's OK to try to redial again.<u></u><u></u></p>
</div>
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<p class="MsoNormal"> <u></u><u></u></p>
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<p class="MsoNormal">But with SIP trunk, many times i can see a really strange sequence of events:<u></u><u></u></p>
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<p class="MsoNormal"> <u></u><u></u></p>
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<p class="MsoNormal">After INVITE i get the following responses (example from a real conversation)<u></u><u></u></p>
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<div>
<p class="MsoNormal">[17:01:40] SIP/2.0 100 Trying<u></u><u></u></p>
</div>
<div>
<div>
<p class="MsoNormal">[17:01:40] SIP/2.0 183 Session Progress<u></u><u></u></p>
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<div>
<p class="MsoNormal">[17:01:51] SIP/2.0 480 Temporarily not available<u></u><u></u></p>
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<p class="MsoNormal"> <u></u><u></u></p>
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<p class="MsoNormal">As far as i understand, this means that the remote phone was ringing for 10 seconds and then the call failed due to a timeout. As far as i understand, i'm supposed to get reason
code 3, but actually the java application gets OriginateResponseEvent with failure reason code 8.<u></u><u></u></p>
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<p class="MsoNormal"> <u></u><u></u></p>
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<div>
<p class="MsoNormal">This behavior is hard to reproduce. I was trying with my own phone number and then i get the expected reason code 3, but i constantly get this situation running our customer's campaigns.<u></u><u></u></p>
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<p class="MsoNormal"> <u></u><u></u></p>
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<p class="MsoNormal"> <u></u><u></u></p>
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<p class="MsoNormal">--
<u></u><u></u></p>
<div>
<div>
<p class="MsoNormal" dir="RTL" style="text-align:right;direction:rtl">
<span lang="HE" style="color:#666666">משיח </span><span dir="LTR" style="color:#666666">NOW</span><span dir="RTL"></span><span dir="RTL"></span><span lang="HE" style="color:#666666"><span dir="RTL"></span><span dir="RTL"></span>!</span><span dir="LTR"><u></u><u></u></span></p>
</div>
<p class="MsoNormal"><span style="color:#666666">Moshiach is coming very soon, prepare yourself!</span><span lang="AR-SA" dir="RTL"><u></u><u></u></span></p>
<div>
<p class="MsoNormal" dir="RTL" style="text-align:right;direction:rtl">
<span lang="HE" style="color:#666666">יחי אדוננו מורינו ורבינו מלך המשיח לעולם ועד!</span><span dir="LTR"><u></u><u></u></span></p>
</div>
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<p class="MsoNormal"><br>
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