<html>
<head>
<meta content="text/html; charset=ISO-8859-1"
http-equiv="Content-Type">
</head>
<body text="#000000" bgcolor="#FFFFFF">
On 13/09/13 12:31, Henrik Westerberg wrote:
<blockquote cite="mid:CE58C440.4161E%25henrik.westerberg@ain.se"
type="cite">
<meta http-equiv="Content-Type" content="text/html;
charset=ISO-8859-1">
<div>Hi,</div>
<div><br>
</div>
<div>I am running Asterisk 11.3 with both SIP and ISDN. When
dialing out (always over SIP) I want to keep track of who
answered and of the length of the call.</div>
<div><br>
</div>
<div>
<div>[outgoing-dev2]</div>
<div>exten =>
h,1,Agi(agi://localhost/ajpbxtest.agi?status=finished)</div>
<div><br>
</div>
<div>exten => _X.,1,NoOp(Will send call to ${CC_DIALSTRING})</div>
<div>exten => _X.,n,Dial(${CC_DIALSTRING}, 60,
M(uploadpeer-dev2^${CC_CALLID})em)</div>
<div>exten =>
_X.,n,Agi(agi://localhost/ajpbxtest.agi?status=failed&dialstatus=${DIALSTATUS})</div>
</div>
<div>
<div style="font-family: Consolas; font-size: medium; "><br>
</div>
<div style="font-family: Consolas; font-size: medium; ">The h
extension is called correctly when the call comes in over IP
and when I record the call. But when the call has come in over
SIP the h extension is called directly after the call is
answered so all the call gets length 0 in my own database.</div>
<div style="font-family: Consolas; font-size: medium; "><br>
</div>
<div style="font-family: Consolas; font-size: medium; ">I guess
that I could record the calls and throw away the recordings
afterwards. In this way the RTP would stay on the server. But
is there not a cleaner way to get Asterisk to execute the h
extension (or another possibility to fix a callback somewhere)
when the the Disconnect comes in over SIP?</div>
</div>
</blockquote>
<br>
I have no idea why you are seeing the h extension being run before
the call ends. Its not something I have ever seen happen.<br>
Whether or not Asterisk stays in the RTP media path makes no
difference as it will always stay in the SIP signalling path and its
that which controls the call establishment and termination.<br>
<br>
</body>
</html>