<div dir="ltr">B.H.<div><br></div><div>While dialing out i get a lot of AMI responses like this:</div><div><br></div><div><div>Event: Hangup</div><div>Privilege: call,all</div><div>Channel: SIP/TRK012-000336b0</div><div>Uniqueid: S5-1376567634.218719</div>
<div>CallerIDNum: XXXXXXXXX</div><div>CallerIDName: YYYYYYYYYY</div><div>ConnectedLineNum: XXXXXXXXX</div><div>ConnectedLineName: YYYYYYYYYY</div><div><i><b>Cause: 19</b></i></div><div><i><b>Cause-txt: User alerting, no answer</b></i></div>
<div><br></div><div>Event: OriginateResponse</div><div>Privilege: call,all</div><div>ActionID: 249867518_255525#YD_UFOzWQx30Wm6PM3USxGE</div><div>Response: Failure</div><div>Channel: SIP/TRK012/YYYYYYYYYY</div><div>Context: YemotDialer_Bridge</div>
<div>Exten: s</div><div><b><i>Reason: 8</i></b></div><div>Uniqueid: <null></div><div>CallerIDNum: XXXXXXXXX</div><div>CallerIDName: YYYYYYYYYY</div></div><div><br></div><div>As mentioned in the previous mails, SIP response code is 480. I would expect to get reason 3 not 8. Reason 8 is confusing my dialer software so it wants to redial the number.</div>
<div><br></div><div>I use Asterisk 1.8.22. Is this a bug in asterisk or is a problem with my SIP trunk provider?</div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Aug 14, 2013 at 9:00 AM, Mordechay Kaganer <span dir="ltr"><<a href="mailto:mkaganer@gmail.com" target="_blank">mkaganer@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 .8ex;border-left:1px #ccc solid;border-right:1px #ccc solid;padding-left:1ex;padding-right:1ex"><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="HOEnZb"><div class="h5">
<div class="gmail_quote">On Aug 13, 2013 10:30 PM, "Shishir Pokharel" <<a href="mailto:Shishir.Pokharel@on24.com" target="_blank">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">
<div lang="EN-US" link="blue" vlink="purple">
<div>
<p class="MsoNormal">
<a name="1407b69185a068e2_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="1407b69185a068e2_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>
<div>
<div>
<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>
</div>
<div>
<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>
</div>
<div>
<p class="MsoNormal"> <u></u><u></u></p>
</div>
<div>
<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>
<div>
<p class="MsoNormal"> <u></u><u></u></p>
</div>
<div>
<p class="MsoNormal">But with SIP trunk, many times i can see a really strange sequence of events:<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"> <u></u><u></u></p>
</div>
<div>
<p class="MsoNormal">After INVITE i get the following responses (example from a real conversation)<u></u><u></u></p>
</div>
<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>
</div>
</div>
<div>
<p class="MsoNormal">[17:01:51] SIP/2.0 480 Temporarily not available<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"> <u></u><u></u></p>
</div>
<div>
<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>
</div>
<div>
<p class="MsoNormal"> <u></u><u></u></p>
</div>
<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>
</div>
<div>
<p class="MsoNormal"> <u></u><u></u></p>
</div>
<div>
<div>
<p class="MsoNormal"> <u></u><u></u></p>
</div>
<p class="MsoNormal">--
<u></u><u></u></p>
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<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>
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<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>
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<span lang="HE" style="color:#666666">יחי אדוננו מורינו ורבינו מלך המשיח לעולם ועד!</span><span dir="LTR"><u></u><u></u></span></p>
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</div></div></blockquote></div><br><br clear="all"><div><br></div>-- <br><div dir="ltr"><div style="direction:rtl"><font color="#666666">משיח NOW!</font></div><font color="#666666">Moshiach is coming very soon, prepare yourself!</font><div dir="rtl">
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