Hi,<br><br>We are using Vicidial and sometime even when agent disconnects, outgoing call originated by dialer is still active. Since call was initiated by dialer and then bought into meetme conference of agent and we can't corelate this call to any agent channel.
<br><br><br>When agents are dialing, channels doesn't show calls<br>
<p class="MsoNormal">vicidial2*CLI> show channels <br></p>
<p class="MsoNormal">Channel<span style="">
</span>Location<span style=""> </span>State<span style=""> </span>Application(Data)</p>
<p class="MsoNormal">Local/78600051@defau 78600051@default:1<span style=""> </span>Up<span style="">
</span>MeetMe(8600051|q)</p>
<p class="MsoNormal">Local/78600051@defau 8309@default:3<span style=""> </span>Up<span style="">
</span>Wait(3600)</p>
<p class="MsoNormal">SIP/primus-8f43<span style="">
</span>(None)<span style=""> </span>Ringing
AppDial((Outgoing Line))</p>
<p class="MsoNormal">Local/761394353177@d 761394353177@default Ring<span style=""> </span>Dial(SIP/61394353177@primus||t</p>
<p class="MsoNormal">Local/761394353177@d s@default:1<span style=""> </span>Down<span style=""> </span>(None)</p>
<p class="MsoNormal">Local/78600053@defau 78600053@default:1<span style=""> </span>Up<span style="">
</span>MeetMe(8600053|q)</p>
<p class="MsoNormal">Local/78600053@defau 8309@default:3<span style=""> </span>Up<span style="">
</span>Wait(3600)</p>
<p class="MsoNormal">SIP/primus-00fe<span style="">
</span>(None)<span style=""> </span><span style=""> </span>Ringing AppDial((Outgoing Line))</p>
<p class="MsoNormal">Local/761394357078@d 761394357078@default Ring<span style=""> </span>Dial(SIP/61394357078@primus||t</p>
<p class="MsoNormal">Local/761394357078@d s@default:1<span style=""> </span>Down<span style=""> </span>(None)</p>
<p class="MsoNormal">Local/78600054@defau 78600054@default:1<span style=""> </span>Up<span style="">
</span>MeetMe(8600054|q)</p>
<p class="MsoNormal">Local/78600054@defau 8309@default:3<span style=""> </span>Up<span style="">
</span>Wait(3600)</p>
<p class="MsoNormal">SIP/primus-95db<span style="">
</span>8600051@default:1<span style=""> </span>Up<span style=""> </span>MeetMe(8600051)</p>
<p class="MsoNormal">Zap/pseudo-122590356 s@default:1<span style=""> </span>Rsrvd<span style=""> </span>(None)</p>
<p class="MsoNormal">SIP/agent7-44fa<span style="">
</span>8600055@default:1<span style=""> </span>Up<span style=""> </span>MeetMe(8600055)</p>
<p class="MsoNormal">SIP/primus-0a7c<span style="">
</span>8600053@default:1<span style=""> </span>Up<span style=""> </span>MeetMe(8600053)</p>
<p class="MsoNormal">SIP/primus-7c73<span style="">
</span>8600054@default:1<span style=""> </span>Up<span style=""> </span>MeetMe(8600054)</p>
<p class="MsoNormal">Local/78600052@defau 78600052@default:1<span style=""> </span>Up<span style="">
</span>MeetMe(8600052|q)</p>
<p class="MsoNormal">Local/78600052@defau 8309@default:3<span style=""> </span>Up<span style="">
</span>Wait(3600)</p>
<p class="MsoNormal">SIP/primus-2ed8<span style="">
</span>8600052@default:1<span style=""> </span>Up<span style=""> </span>MeetMe(8600052)</p>
<p class="MsoNormal">Zap/pseudo-104079549 s@default:1<span style=""> </span>Rsrvd<span style=""> </span>(None)</p>
<p class="MsoNormal">SIP/agent1-32b5<span style="">
</span>8600054@default:1<span style=""> </span>Up<span style=""> </span>MeetMe(8600054)</p>
<p class="MsoNormal">Zap/pseudo-204709889 s@default:1<span style=""> </span>Rsrvd<span style=""> </span>(None)</p>
<p class="MsoNormal">SIP/agent8-d3ab<span style="">
</span>8600056@default:1<span style=""> </span>Up<span style=""> </span>MeetMe(8600056)</p>
<p class="MsoNormal">SIP/agent5-ec77<span style="">
</span>8600051@default:1<span style=""> </span>Up<span style=""> </span>MeetMe(8600051)</p>
<p class="MsoNormal">Zap/pseudo-926666046 s@default:1<span style=""> </span>Rsrvd<span style=""> </span>(None)</p>
<p class="MsoNormal">SIP/agent3-2df5<span style="">
</span>8600053@default:1<span style=""> </span>Up<span style=""> </span>MeetMe(8600053)</p>
<p class="MsoNormal">Zap/pseudo-204290210 s@default:1<span style=""> </span>Rsrvd<span style=""> </span>(None)</p>
<p class="MsoNormal">SIP/agent2-4ff6<span style="">
</span>8600052@default:1<span style=""> </span>Up<span style=""> </span>MeetMe(8600052)</p>
<p class="MsoNormal">SIP/primus-fc90<span style="">
</span>8600051@default:1<span style=""> </span>Up<span style=""> </span>MeetMe(8600051)</p>
<p class="MsoNormal">Zap/pseudo-170346238 s@default:1<span style=""> </span>Rsrvd<span style=""> </span>(None)</p>
<p class="MsoNormal">31 active channels</p>
<br>After agents have logged out<br><br><br>vicidial2*CLI> show channels<br>Channel Location State Application(Data)<br>SIP/primus-fc90 8600051@default:1 Up MeetMe(8600051)<br>Zap/pseudo-170346238
s@default:1 Rsrvd (None)<br><br><br>Calls doesn't show channels<br><br>vicidial2*CLI> sip show channels<br>Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message<br><a href="http://203.63.248.197">
203.63.248.197</a> 122001 20a58a2e251 00651/00000 unkn No<br><a href="http://203.196.128.56">203.196.128.56</a> 6135625116 5420f80176e 00102/00000 g729 No Tx: ACK<br><br>calls doesn't show channel<br>
CLI>sip show channel 5420f80176e
<p class="MsoNormal"><span style=""> </span>* SIP Call</p>
<p class="MsoNormal"><span style=""> </span>Direction:<span style=""> </span>Outgoing</p>
<p class="MsoNormal"><span style=""> </span>Call-ID:<span style="">
</span><a href="mailto:5420f80176e3e56679ce4e537ffdbd3f@203.196.128.56">5420f80176e3e56679ce4e537ffdbd3f@203.196.128.56</a></p>
<p class="MsoNormal"><span style=""> </span>Our Codec
Capability:<span style=""> </span>256</p>
<p class="MsoNormal"><span style=""> </span>Non-Codec
Capability:<span style=""> </span>1</p>
<p class="MsoNormal"><span style=""> </span>Their Codec
Capability:<span style=""> </span>256</p>
<p class="MsoNormal"><span style=""> </span>Joint Codec
Capability:<span style=""> </span>256</p>
<p class="MsoNormal"><span style=""> </span>Format<span style=""> </span>g729</p>
<p class="MsoNormal"><span style=""> </span>Theoretical
Address:<span style=""> </span><a href="http://203.196.128.56:5060">203.196.128.56:5060</a></p>
<p class="MsoNormal"><span style=""> </span>Received
Address:<span style=""> </span><a href="http://203.196.128.56:5060">203.196.128.56:5060</a></p>
<p class="MsoNormal"><span style=""> </span>NAT Support:<span style=""> </span>RFC3581</p>
<p class="MsoNormal"><span style=""> </span>Audio IP:<span style=""> </span><a href="http://220.227.174.4">220.227.174.4</a> (local)</p>
<p class="MsoNormal"><span style=""> </span>Our Tag:<span style=""> </span><span style=""> </span>as7a55ac7a</p>
<p class="MsoNormal"><span style=""> </span>Their Tag:<span style=""> </span>29258</p>
<p class="MsoNormal"><span style=""> </span>SIP User agent:</p>
<p class="MsoNormal"><span style=""> </span>Username:<span style=""> </span>61356251162</p>
<p class="MsoNormal"><span style=""> </span>Peername:<span style=""> </span>90340</p>
<p class="MsoNormal"><span style=""> </span>Original uri:<span style=""> </span>sip:61356251162@216.181.122.44:5060</p>
<p class="MsoNormal"><span style=""> </span>Need Destroy:<span style=""> </span>0</p>
<p class="MsoNormal"><span style=""> </span>Last Message:<span style=""> </span>Tx: ACK</p>
<p class="MsoNormal"><span style=""> </span>Promiscuous
Redir:<span style=""> </span>No</p>
<p class="MsoNormal"><span style=""> </span>Route:<span style="">
</span><a href="mailto:sip:61356251162@203.196.128.56">sip:61356251162@203.196.128.56</a>;ftag=as7a55ac7a;lr=on</p>
<p class="MsoNormal"><span style=""> </span>DTMF Mode:<span style=""> </span>rfc2833</p>
<p class="MsoNormal"><span style=""> </span>SIP Options:<span style=""> </span>(none)</p>BUT ONE THING IS COMMON IS THAT OLDEST SIP CALL WILL COME IN THE BOTTOM OF THE LIST of COMMAND sip show channels (agents will be above it) so it is hung and needs to be destroyed manually. Also channel corresponding to this call will also come in the bottom of SHOW Channels command for same technology
i.e. it will be last SIP/XYZ entry so to destroy this call lets try destroy last SIP channel entry.<br><p class="MsoNormal"><br></p>vicidial2*CLI> soft hangup SIP/primus-fc90<br>Requested Hangup on channel 'SIP/primus-fc90'
<br> -- Hungup 'Zap/pseudo-1703462386'<br> == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/primus-fc90'<br> -- Executing DeadAGI("SIP/primus-fc90", "call_log.agi|h") in new stack
<br> -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi<br> -- AGI Script call_log.agi completed, returning 0<br> -- Executing DeadAGI("SIP/primus-fc90", "VD_hangup.agi|h") in new stack
<br> -- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi<br> -- AGI Script VD_hangup.agi completed, returning 0<br><br><br><br><br>vicidial2*CLI> sip show channels<br>Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message
<br>0 active SIP channels<br><br><br>IT WORKS!! A crude way but very important to save 100 of dollars of hung call while agent are dialing. You can always do stop now but then whole operations will stop.<br><br>Dont know why this happens in first place but atleast I have seen it coming twice and now keep a vigil that no call is below the agents in sip show channels, it there is any it means its a hung call costing you money
<br><br>Abhimanyu<br>