<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=us-ascii" http-equiv="Content-Type">
</head>
<body bgcolor="#ffff99" text="#000000">
After much checking and puzzling, I cannot get my Polycom 601 to toggle
call recording with my Asterisk 1.4.21.1.<br>
<br>
Via FreePBX, I can set a user to always record, and the recording will
show up in /var/spool/asterisk/monitor.<br>
<br>
But if I try to start recording by toggling in-call, no luck.<br>
<br>
I can see this in the feature*.conf file set:<br>
<br>
automon=*1<br>
<br>
and I can see a 'Ww' in the logged/traced call to dial().<br>
<br>
and I can see the RFC2833 RTP packets going through Asterisk, both with
rtp debug and with wireshark.<br>
<br>
So my questions are:<br>
<br>
1) How do I verify that asterisk actually saw the feature code spec
upon restart/reload? I can't find any clues.<br>
<br>
2) Are there any other parameters that have a bearing on this?<br>
<br>
3) Is there anything I haven't thought of?<br>
<br>
Finally, it might be worth noting that the packet traces show three
RFC2833 end events for each DTMF code pressed. This might be perfectly
normal, and I even tried fudging the automon string to ***111 just to
compensate as an experiment, but it had no effect.<br>
<br>
If I've done everything necessary to configure enabling the toggle
function, then where should I see the failure/refusal to comply in any
logs. I'm getting nothing in logs/traces.<br>
<br>
A side question: freepbx is generating include statements with a
leading #, a la C includes - or a la Perl/Shell/et al comments! This
is OK? I've floundering with the suspicion that I'm overlooking
something really dumb...<br>
<br>
I would be grateful for some explicit diagnostic suggestions.<br>
<br>
<br>
</body>
</html>