[Asterisk-Users] astcc missing to bill random calls?
Darren Wiebe
darren at aleph-com.net
Tue Oct 18 20:00:25 MST 2005
What channel are you using to place the calls from ASTCC and what
version of asterisk are you using? The get_variable and set_variable
perl commands are not working in -HEAD due to stuff being deprecated.
Darren Wiebe
darren at aleph-com.net
maka wrote:
> Hello list,
>
> I just came into a strange problem wth astcc. the trouble is astcc.agi
> does not bill some calls. The calls are logged in the
> cdr-csv/Master.csv file, but with a duration of 0, billsec of 0, an
> empty dstchannel, and with a lastapp field of "hangup". I suppose that
> astcc.agi was not able to get the answeredime variable from the SIP
> channel...
>
> I have added a few functions to the astcc default script, in order to
> support different categories of users (functions to get the user type,
> get the routes and trunks tables for the user category before
> trytrunk), as well as some 'print SDTERR' statements, in order to
> trace any problems during execution. Could this be the problem, I
> noticed that there were reports on the list that get_variable has
> issues with extensive $agi->verbose callings. I had a problem with
> get_variable not catching answeredtime once before, and solved these
> by adding an additional agi->get_variable statement just underneath
> the first one.
>
> Here's how the calls is logged in the csv file:
> "","38607612","0016318674103","from-sip","""38607612""
> <38607612>","SIP/sip.mytel.net-0816afc8","","Hangup","","2005-10-17
> 18:00:16","2005-10-17 18:00:16","2005-10-17
> 18:00:16",0,0,"ANSWERED","DOCUMENTATION"
>
>
> The strangest thing is that this appears to happen at random times, so
> I can't just sit down and watch it through. I would appreciate any
> ideas, cheers...
>
> maka
>
>
>
> --
> I'm sick and tired of being sick and tired...
>
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