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Summary:ASTERISK-06758: Originate records Disposition 'answered' in cdr when 'originating device' answers
Reporter:Christian Lauinger (clauinger)Labels:
Date Opened:2006-04-28 18:57:18Date Closed:2011-06-07 14:07:47
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Core/ManagerInterface
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:Originate a call in AMI, (with FOP, or something else),
pick up the ringing device, but the called party doesn't pick up.

In cdr there is always disposition answered.
Comments:By: Andrey S Pankov (casper) 2006-04-28 21:17:00

Maybe some cdr entries, your Originate command and extentions.conf?
By: Christian Lauinger (clauinger) 2006-04-29 07:20:37

2006-04-29 14:16:43   SIP/10-...   10   "Christian" <10>   Dial   SIP/11|20|tTr   11   ANSWERED   00:02

Thats what in cdr.

Originate command:
<PRE>
<?php
$socket = fsockopen("127.0.0.1","5058", $errno, $errstr, $timeout);
fputs($socket, "Action: Login\r\n");
fputs($socket, "UserName: voiper\r\n");
fputs($socket, "Secret: voiper\r\n\r\n");
fputs($socket, "Action: Originate\r\n");
fputs($socket, "Channel: SIP/10\r\n");
fputs($socket, "Context: default\r\n");
fputs($socket, "Exten: 11\r\n");
fputs($socket, "Priority: 1\r\n");
fputs($socket, "Callerid: 11\r\n\r\n");
fputs($socket, "Action: Logoff\r\n\r\n");
while (!feof($socket)) {
$wrets[] = fread($socket, 8192);
}
fclose($socket);
var_dump($wrets);
?>
</pre>
By: Andrey S Pankov (casper) 2006-04-30 10:19:13

What's in extensions.conf context default? Please upload, do not paste... thanks.
By: Christian Lauinger (clauinger) 2006-05-01 13:14:31

Only this:

exten => 11,1,Dial(SIP/11,20,tTr)
By: Andrey S Pankov (casper) 2006-05-01 14:44:27

Can you upgrade to 1.2.7.1 and test it again please. That's because it may be already fixed:

------------------------------------------------------------------------
r14234 | file | 2006-03-22 23:38:32 +0200 (&ASTERISK-1051;&ASTERISK-1082;&ASTERISK-1070;, 22 &ASTERISK-1046;&ASTERISK-1066;&ASTERISK-1082; 2006) | 2 lines

Issue ASTERISK-5763 - Disposition showing FAILED even though call is answered successfully (Reported by tracinet)

------------------------------------------------------------------------
r13161 | russell | 2006-03-16 00:16:15 +0200 (&ASTERISK-1057;&ASTERISK-1084;&ASTERISK-1068;, 16 &ASTERISK-1046;&ASTERISK-1066;&ASTERISK-1082; 2006) | 3 lines

- remove some calculations that will always result in 0
- if a CDR was never started, don't try to calculate a duration and consider it failed
By: Christian Lauinger (clauinger) 2006-05-09 12:39:45

1.2.7.1.dfsg-2 is now my version, out of debian unstable.

but still same effect
By: Serge Vecher (serge-v) 2006-05-09 12:50:56

is anything of interest reported on the console at high verbosity setting?
By: Andrey S Pankov (casper) 2006-05-09 13:45:12

Hey, guy! But you do pick up SIP/10, right? That's why you have ANSWERED for SIP/10 and duration is 0:02...
By: Christian Lauinger (clauinger) 2006-05-09 15:00:10

-- Executing Dial("SIP/10-3a80", "SIP/11|20|tTr") in new stack
   -- Called 11
   -- SIP/11-9541 is ringing

that is in the console

###########################

yes i picked up SIP/10, but this is the phone that originates the call.
the call is anwered if SIP/11 picks up, not the 10 !
By: Andrey S Pankov (casper) 2006-05-09 15:06:16

It seems like this is the way it works. Please consider using chan_local to get correct CDRs.

Is there any difference if you originate a call using pbx_spool's call file?
By: Christian Lauinger (clauinger) 2006-05-11 07:57:43

tryed call file, no difference.

chan_local ?
By: Andrey S Pankov (casper) 2006-05-11 11:53:03

> chan_local ?

Yes... http://voip-info.org/ for more... :)
By: Andrey S Pankov (casper) 2006-05-15 13:06:14

Not a bug, configuration issue... Let's close it.
By: Serge Vecher (serge-v) 2006-05-15 13:40:48

I second casper on this one: Asterisk is correct in creating a CDR record for "originating device" call-leg (in reality, it is AMI that is the originator of call, not "originating device" itself). If you need a change from existing behaviour, please discuss such change either on the asterisk-dev list or #asterisk-dev channel on IRC. If an outcome of such discussion is a patch, please feel free to reopen the issue with the patch attached.

Thanks.