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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Since there are no DAHDI lines involved,
polarity probably won’t help. Call-limit might or might not help with
this. Does “core show channels” show anything after the callee
hangs up?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>hugolivude<br>
<b><span style='font-weight:bold'>Sent:</span></b> Thursday, January 21, 2010
7:47 AM<br>
<b><span style='font-weight:bold'>To:</span></b> <st1:PersonName w:st="on">Asterisk
Users Mailing List - Non-Commercial Discussion</st1:PersonName><br>
<b><span style='font-weight:bold'>Subject:</span></b> [asterisk-users] Caller
hang up not detected</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>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></span></font></p>
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