Hi all,<br><br>My architecture is:<br><br>PSTN-----E1----OldPBX----E1-----Asterisk<br><br>I've a similar problem, SIP user agents using X-Lite:<br><br><br>Sip User Agent "A" calls to PSTN user "B"<br>"B" user hangs the call
<br>"A" user starts listening busy indications on the phone, and if he doesn't hangup correctly on Xlite <br>The calls seems to be alive.... Only solved it with soft hangup, and that is not an acceptable solution.
<br><br>I have on user that seems to have turned off the pc ( at least he reports me that) and the call (at least on Asterisk CDR) remained alive....didn't disconnect....<br><br>It is working fine only if SIP user agents dials to an extension in the Old PBX, that case if the called party Hangs, the Old Pbx immediately sends a DISCONNECT message to Asterisk and the call hangs...
<br><br>I hope someone could help US.<br><br>Best regards,<br>Marco Mouta<br><br><br><br><div><span class="gmail_quote">On 4/12/06, <b class="gmail_sendername">Abhimanyu Rapria</b> <<a href="mailto:Abhimanyu@synotek.com">
Abhimanyu@synotek.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div style="direction: ltr;">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>vicidial2*CLI> show channels <br></p>
<p>Channel<span>
</span>Location<span> </span>State<span> </span>Application(Data)</p>
<p>Local/78600051@defau 78600051@default:1<span> </span>Up<span>
</span>MeetMe(8600051|q)</p>
<p>Local/78600051@defau 8309@default:3<span> </span>Up<span>
</span>Wait(3600)</p>
<p>SIP/primus-8f43<span>
</span>(None)<span> </span>Ringing
AppDial((Outgoing Line))</p>
<p>Local/761394353177@d 761394353177@default Ring<span> </span>Dial(SIP/61394353177@primus||t</p>
<p>Local/761394353177@d s@default:1<span> </span>Down<span> </span>(None)</p>
<p>Local/78600053@defau 78600053@default:1<span> </span>Up<span>
</span>MeetMe(8600053|q)</p>
<p>Local/78600053@defau 8309@default:3<span> </span>Up<span>
</span>Wait(3600)</p>
<p>SIP/primus-00fe<span>
</span>(None)<span> </span><span> </span>Ringing AppDial((Outgoing Line))</p>
<p>Local/761394357078@d 761394357078@default Ring<span> </span>Dial(SIP/61394357078@primus||t</p>
<p>Local/761394357078@d s@default:1<span> </span>Down<span> </span>(None)</p>
<p>Local/78600054@defau 78600054@default:1<span> </span>Up<span>
</span>MeetMe(8600054|q)</p>
<p>Local/78600054@defau 8309@default:3<span> </span>Up<span>
</span>Wait(3600)</p>
<p>SIP/primus-95db<span>
</span>8600051@default:1<span> </span>Up<span> </span>MeetMe(8600051)</p>
<p>Zap/pseudo-122590356 s@default:1<span> </span>Rsrvd<span> </span>(None)</p>
<p>SIP/agent7-44fa<span>
</span>8600055@default:1<span> </span>Up<span> </span>MeetMe(8600055)</p>
<p>SIP/primus-0a7c<span>
</span>8600053@default:1<span> </span>Up<span> </span>MeetMe(8600053)</p>
<p>SIP/primus-7c73<span>
</span>8600054@default:1<span> </span>Up<span> </span>MeetMe(8600054)</p>
<p>Local/78600052@defau 78600052@default:1<span> </span>Up<span>
</span>MeetMe(8600052|q)</p>
<p>Local/78600052@defau 8309@default:3<span> </span>Up<span>
</span>Wait(3600)</p>
<p>SIP/primus-2ed8<span>
</span>8600052@default:1<span> </span>Up<span> </span>MeetMe(8600052)</p>
<p>Zap/pseudo-104079549 s@default:1<span> </span>Rsrvd<span> </span>(None)</p>
<p>SIP/agent1-32b5<span>
</span>8600054@default:1<span> </span>Up<span> </span>MeetMe(8600054)</p>
<p>Zap/pseudo-204709889 s@default:1<span> </span>Rsrvd<span> </span>(None)</p>
<p>SIP/agent8-d3ab<span>
</span>8600056@default:1<span> </span>Up<span> </span>MeetMe(8600056)</p>
<p>SIP/agent5-ec77<span>
</span>8600051@default:1<span> </span>Up<span> </span>MeetMe(8600051)</p>
<p>Zap/pseudo-926666046 s@default:1<span> </span>Rsrvd<span> </span>(None)</p>
<p>SIP/agent3-2df5<span>
</span>8600053@default:1<span> </span>Up<span> </span>MeetMe(8600053)</p>
<p>Zap/pseudo-204290210 s@default:1<span> </span>Rsrvd<span> </span>(None)</p>
<p>SIP/agent2-4ff6<span>
</span>8600052@default:1<span> </span>Up<span> </span>MeetMe(8600052)</p>
<p>SIP/primus-fc90<span>
</span>8600051@default:1<span> </span>Up<span> </span>MeetMe(8600051)</p>
<p>Zap/pseudo-170346238 s@default:1<span> </span>Rsrvd<span> </span>(None)</p>
<p>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" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">
203.63.248.197</a> 122001 20a58a2e251 00651/00000 unkn No<br><a href="http://203.196.128.56" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">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><span> </span>* SIP Call</p>
<p><span> </span>Direction:<span> </span>Outgoing</p>
<p><span> </span>Call-ID:<span>
</span><a href="mailto:5420f80176e3e56679ce4e537ffdbd3f@203.196.128.56" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">5420f80176e3e56679ce4e537ffdbd3f@203.196.128.56</a></p>
<p><span> </span>Our Codec
Capability:<span> </span>256</p>
<p><span> </span>Non-Codec
Capability:<span> </span>1</p>
<p><span> </span>Their Codec
Capability:<span> </span>256</p>
<p><span> </span>Joint Codec
Capability:<span> </span>256</p>
<p><span> </span>Format<span> </span>g729</p>
<p><span> </span>Theoretical
Address:<span> </span><a href="http://203.196.128.56:5060" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">203.196.128.56:5060</a></p>
<p><span> </span>Received
Address:<span> </span><a href="http://203.196.128.56:5060" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">203.196.128.56:5060</a></p>
<p><span> </span>NAT Support:<span> </span>RFC3581</p>
<p><span> </span>Audio IP:<span> </span><a href="http://220.227.174.4" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">220.227.174.4</a> (local)</p>
<p><span> </span>Our Tag:<span> </span><span> </span>as7a55ac7a</p>
<p><span> </span>Their Tag:<span> </span>29258</p>
<p><span> </span>SIP User agent:</p>
<p><span> </span>Username:<span> </span>61356251162</p>
<p><span> </span>Peername:<span> </span>90340</p>
<p><span> </span>Original uri:<span> </span>sip:61356251162@216.181.122.44:5060</p>
<p><span> </span>Need Destroy:<span> </span>0</p>
<p><span> </span>Last Message:<span> </span>Tx: ACK</p>
<p><span> </span>Promiscuous
Redir:<span> </span>No</p>
<p><span> </span>Route:<span>
</span><a href="mailto:sip:61356251162@203.196.128.56" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">sip:61356251162@203.196.128.56</a>;ftag=as7a55ac7a;lr=on</p>
<p><span> </span>DTMF Mode:<span> </span>rfc2833</p>
<p><span> </span>SIP Options:<span> </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><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></div><div style="direction: ltr;"><span class="sg"><br>Abhimanyu<br>
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