<br><br><div><span class="gmail_quote">On 10/19/05, <b class="gmail_sendername">Darren Wiebe</b> <<a href="mailto:darren@aleph-com.net">darren@aleph-com.net</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Thanks for your feedback.<br><br>maka wrote:<br><br>> I'm using asterisk-1.0.6. The channel to dial is either SIP or IAX,<br>> I've had one missed call in both cases.<br>><br>> I commented out the $agi->verbose stuff in many places in the script,
<br>> and I limited my own print STDERR statements. I haven't seen the isue<br>> reappear since then, but I'm not sure whether excessive $agi->verbose<br>> output is what caused it.<br><br>Ok, just wondered.<br>
<br>><br>> I have also changed the way calls are billled in the calccost function<br>> to use includedseconds, and the billing increment period after that. I<br>> don't think this has anything to do with the problem anyway..
<br><br>Wasn't this fixed a while ago? I had a patch that I thought had been<br>accepted......</blockquote><div><br>
Absolutely, i just noticed it after i changed it myself..<br>
</div><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Darren Wiebe<br><a href="mailto:darren@aleph-com.net">darren@aleph-com.net</a><br>
<br>><br>><br>><br>> On 10/19/05, *Darren Wiebe* <<a href="mailto:darren@aleph-com.net">darren@aleph-com.net</a><br>> <mailto:<a href="mailto:darren@aleph-com.net">darren@aleph-com.net</a>>> wrote:
<br>><br>> What channel are you using to place the calls from ASTCC and what<br>> version of asterisk are you using? The get_variable and set_variable<br>> perl commands are not working in -HEAD due to stuff being deprecated.
<br>><br>> Darren Wiebe<br>> <a href="mailto:darren@aleph-com.net">darren@aleph-com.net</a> <mailto:<a href="mailto:darren@aleph-com.net">darren@aleph-com.net</a>><br>><br>> maka wrote:<br>
><br>> > Hello list,<br>> ><br>> > I just came into a strange problem wth astcc. the trouble is<br>> astcc.agi<br>> > does not bill some calls. The calls are logged in the
<br>> > cdr-csv/Master.csv file, but with a duration of 0, billsec of 0, an<br>> > empty dstchannel, and with a lastapp field of "hangup". I<br>> suppose that<br>> > astcc.agi
was not able to get the answeredime variable from the SIP<br>> > channel...<br>> ><br>> > I have added a few functions to the astcc default script, in<br>> order to<br>> > support different categories of users (functions to get the user
<br>> type,<br>> > get the routes and trunks tables for the user category before<br>> > trytrunk), as well as some 'print SDTERR' statements, in order to<br>> > trace any problems during execution. Could this be the problem, I
<br>> > noticed that there were reports on the list that get_variable has<br>> > issues with extensive $agi->verbose callings. I had a problem with<br>> > get_variable not catching answeredtime once before, and solved these
<br>> > by adding an additional agi->get_variable statement just underneath<br>> > the first one.<br>> ><br>> > Here's how the calls is logged in the csv file:<br>> > "","38607612","0016318674103","from-sip","""38607612""
<br>> > <38607612>","SIP/sip.mytel.net-0816afc8","","Hangup","","2005-10-17<br>> > 18:00:16","2005-10-17 18:00:16","2005-10-17
<br>> > 18:00:16",0,0,"ANSWERED","DOCUMENTATION"<br>> ><br>> ><br>> > The strangest thing is that this appears to happen at random<br>> times, so<br>
> > I can't just sit down and watch it through. I would appreciate any<br>> > ideas, cheers...<br>> ><br>> > maka<br>> ><br>> ><br>> ><br>> > --
<br>> > I'm sick and tired of being sick and tired...<br>> ><br>> >------------------------------------------------------------------------<br>> ><br>> >_______________________________________________
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</a><br></blockquote></div><br><br clear="all"><br>-- <br>I'm sick and tired of being sick and tired...