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<p class="MsoNormal"><span style="color:#1F497D">Btw, afaik this just means it didn’t get RTP setup properly. You’d need a full PCAP to diagnose. (Though Vitelity should be able to grab any SIP PCAP on demand.)<o:p></o:p></span></p>
<p class="MsoNormal"><a name="_MailEndCompose"><span style="color:#1F497D"><o:p> </o:p></span></a></p>
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<p class="MsoNormal"><b>From:</b> freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org]
<b>On Behalf Of </b>Sean Devoy<br>
<b>Sent:</b> Friday, 4 December, 2015 15:17<br>
<b>To:</b> FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org><br>
<b>Subject:</b> [Freeswitch-users] Vitelity Calls failing on my server - Ideas<o:p></o:p></p>
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<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Hi!<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">My VOIP provider is Vitelity. A few hours ago SOME of my customers reported phone failures – Rings no answer for the caller.<o:p></o:p></p>
<p class="MsoNormal">I understand this is NOT a FreeSwitch problem, but I thought someone here may be able to identify the problem quickly. I would greatly appreciate it if you could read the rest and comment if you know something that might help get me back
online.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">I have debugged what I know. I have identical calls wioth logs to 2 different numbers that are handled identically on my freeswitch server (going to an IVR).<o:p></o:p></p>
<p class="MsoNormal">The critical difference I see in the logs is here:<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">EXECUTE <a href="mailto:sofia/external_noauth/4108084699@66.241.96.237">
sofia/external_noauth/4108084699@66.241.96.237</a> ivr(estabrook_attendant_1)<o:p></o:p></p>
<p class="MsoNormal">2015-12-04 15:16:10.628821 [DEBUG] switch_ivr_menu.c:721 switch_ivr_menu_stack_xml_add binding 'menu-exit'<o:p></o:p></p>
<p class="MsoNormal">… ivr continues<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">2015-12-04 15:16:10.628821 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16@8000hz 1 channels 20ms<o:p></o:p></p>
<p class="MsoNormal"><b>2015-12-04 15:16:10.698805 [DEBUG] switch_rtp.c:3757 Correct ip/port confirmed.<o:p></o:p></b></p>
<p class="MsoNormal">2015-12-04 15:16:19.448558 [DEBUG] switch_core_session.c:1016 Send signal
<a href="mailto:sofia/external_noauth/4108084699@66.241.96.237">sofia/external_noauth/4108084699@66.241.96.237</a> [BREAK]<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">On the calls that are failing the debug line “<b>switch_rtp.c:3757 Correct ip/port confirmed”</b> does not happen. Instead I eventually get “[DEBUG] switch_ivr_menu.c:643 IVR menu 'bfis_attendant_1' no input detected”, all the while the
line is ringing on caller end.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">If someone understands what that means and I could tell them, it might get my customers back online quicker.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Thanks for your time,<o:p></o:p></p>
<p class="MsoNormal">Sean<o:p></o:p></p>
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