Thanks responding guys. It appears that it's the canreinvite that's causing the problem. Interesting results tho:<br><br>With canreinvite=yes, leaving out the transfer options leads to a Dial command that _never_ exits:<br>
exten => 1,n,Dial(SIP/14168724765@6135551212-sw1|120|g)<br><br>I have 2 channels seemingly forever - kinda scary!<br><br>With canreinvite=yes, but with transfer options, I get the behaviour I reported - it hangs up but only after the callee hangs up. <br>
exten => 1,n,Dial(SIP/14168724765@6135551212-sw1|120|gtT)<br><br>adding this had no effect:<br>exten => 1,n,Hangup<br><br>except of course that it hangs up after the callee hangs up (but not when the caller hangs up)<br>
<br>With canreinvite=no, Dial exits when the caller (me) hangs up in both cases, which is of course the desired behaviour. <br><br>Will I still be able to transfer calls w/ canreinvite=no? I'd like to test that but my second problem is that the feature codes don't seem to be working for me! I posted that problem in a separate thread, but it didn't get posted on the list; that seems to happen to me frequently.<br>
<br>Thanks again guys.<br><br>H<br>
<br><br><br><div class="gmail_quote">On Thu, Jan 21, 2010 at 10:39 AM, Steven Davison <span dir="ltr"><<a href="mailto:steven.davison@ntsols.com">steven.davison@ntsols.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div link="blue" vlink="purple" lang="EN-GB">
<div>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">Hi, </span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">Couple of questions...</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">Are you allowing reinvites, and what happens if you change the
dialplan to this?</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p class="MsoNormal"></p><div class="im">exten => 1,n,Dial(SIP/14168724765@6135551212-sw1|120|gtT)<br>
exten => 1,n,Playback(vm-goodbye)<br>
</div><span style="font-size: 11pt; color: rgb(31, 73, 125);">exten
=> 1,n,Hangup()</span>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">help this helps </span><span style="font-size: 11pt; font-family: Wingdings; color: rgb(31, 73, 125);">J</span><span style="font-size: 11pt; color: rgb(31, 73, 125);"></span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">Steven Davison</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);">Net Technial Solutions</span></p>
<p class="MsoNormal"><span style="font-size: 11pt; color: rgb(31, 73, 125);"> </span></p>
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<p class="MsoNormal"><b><span style="font-size: 10pt;" lang="EN-US">From:</span></b><span style="font-size: 10pt;" lang="EN-US"> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b>On Behalf Of </b>hugolivude<br>
<b>Sent:</b> 21 January 2010 13:47<div class="im"><br>
<b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br>
<b>Subject:</b> [asterisk-users] Caller hang up not detected</div></span></p>
</div>
<p class="MsoNormal"> </p>
<p class="MsoNormal">Hi, <br></p><div><div></div><div class="h5">
<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</div></div>
</div>
</div>
</blockquote></div><br>