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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Hi, <o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Couple of questions...<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Are you allowing reinvites, and what happens if you change the
dialplan to this?<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal>exten => 1,n,Dial(SIP/14168724765@6135551212-sw1|120|gtT)<br>
exten => 1,n,Playback(vm-goodbye)<br>
<span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>exten
=> 1,n,Hangup()<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>help this helps </span><span style='font-size:11.0pt;font-family:
Wingdings;color:#1F497D'>J</span><span style='font-size:11.0pt;font-family:
"Calibri","sans-serif";color:#1F497D'><o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Steven Davison<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Net Technial Solutions<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0cm 0cm 0cm'>
<p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt;font-family:
"Tahoma","sans-serif"'>From:</span></b><span lang=EN-US style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>hugolivude<br>
<b>Sent:</b> 21 January 2010 13:47<br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> [asterisk-users] Caller hang up not detected<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Hi, <br>
<br>
I'm having trouble getting Dial to exit when the caller hangs up in Asterisk
1.4.21.2.<br>
<br>
I use a POTS line to call into the DiD given to me by VOIP service
provider. When the call comes in, I have the VOIP provider send it to
another POTS line. All this works fine however when the caller (me) hangs
up, the Dial command does not exit. The callee stays connected (and my
billing continues!). Dial doesn't exit until the callee hangs up. Here's a snip
from extensions.conf:<br>
<br>
exten => 1,n,Dial(SIP/14168724765@6135551212-sw1|120|gtT)<br>
exten => 1,n,Playback(vm-goodbye)<br>
<br>
Here's the CLI output (verbosity = 4):<br>
<br>
-- Executing [1@Trunk-0001:1] NoOp("SIP/77.57.127.163-09023590",
"") in new stack<br>
-- Executing [1@Trunk-0001:2] Dial("SIP/77.57.127.163-09023590",
"SIP/14168724765@6135551212-sw1|120|gtT") in new stack<br>
-- Called 14168724765@6135551212-sw1<br>
-- SIP/6135551212-sw1-090275d0 is making progress passing it to
SIP/77.57.127.163-09023590<br>
-- SIP/6135551212-sw1-090275d0 answered SIP/77.57.127.163-09023590<br>
*** I hang up here, but the call continues. A while later the callee
hangs up:<br>
-- Executing [1@Trunk-0001:3] Playback("SIP/77.57.127.163-09023590",
"vm-goodbye") in new stack<br>
*** obviously I don't here this, just see it in the CLI<br>
<br>
I'd be grateful for any troubleshooting tips that will help me get asterisk to
quit the Dial command when the originator hangs up.<br>
<br>
Thanks,<br>
H<o:p></o:p></p>
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