Sorry i forget to give my extentions config.
[clientsG]
exten => _x.,1,Set(UserN=${CALLERID(all)})
exten => _x.,2,Set(CalledNum=${EXTEN})
exten => _x.,3,Set(Stime=${DATETIME})
exten => _x.,4,Set(CID=${CALLERID})
exten => _x.,5,Set(HCA=${HANGUPCAUSE})
exten => _x.,6,Set(Cun=${UNIQUEID})
exten => _x.,7,AGI(routing.pl)
exten => h,1,DeadAGI(stop.pl)
exten => h,1,Hangup
routing.pl
$AGI->exec('Set',"TIMEOUT(absolute)=$cstatus");
my $dialstr = "$gwtype/$gwip/$dialednum";
$AGI->exec('Dial', $dialstr); #//Dial the number
Abdul <[EMAIL PROTECTED]> wrote: We are not using any GSM Gateway for call
carriers we have
Asterisk ----> TELES(iSWITCH) -------> MCI
As Teles is world class telecoms product it should not make poor protocol stack.
In my AGI script already i am using
TIMEOUT(absolute)to limit the call according to registrar balance.
I am thinking my be exten => foo,n,Congestion(3) function can solve
the issue but how i can call this i should call it after dial or before?
is (3) is max Congestion time?
Thank You
Abdul <[EMAIL PROTECTED]> wrote:
Abdul <[EMAIL PROTECTED]> wrote: Good Day All,
I am facing a serious problem since I started to use asterisk. I dont know if
it is a bug or some one already solved this.
Currently I am running 120 VoIP SIP channels on my asterisk server but each day
2, 3 calls got hanged in asterisk, and on asterisk CLI show channels
showing us as call UP but in real there is no call.
When asterisk restarted the hanged calls removed from CLI with very high
duration which damaged our billing system and customers accounts goes in high
negative.
First I tried to get call info from asterisk mysql CDR using billsec field but
the same result then I create PERL AGI to get duration from ANSWEREDTIME and
same result.
Please have a look to the following URL which I put the result of show channel
<channelname> you can see the DIALSTATUS=CONGESTION but Elapsed Time:
20h54m16s which really strange and out of my mind to control such as call.
http://www.emafone.net/bugs.html
Please advice us if it is Bug and solved in some ver or its need some
configuration to avoid this issue.
This is in both ver of asterisk 1.2 and 1.4
--------
Regard,
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Regard,
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