Call Recording on User Extension

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Call Recording on User Extension

Postby pdecker414 » Tue Aug 27, 2013 1:18 pm

I have been trying to get call recording working. I have enabled it in individual extensions to record everything. Nothing shows up on the Call monitor for that extension except for the call. I do see a reference to the recording in the CLI.

-- Executing [s@macro-one-touch-record:1] System("SIP/240-000000da", "/var/lib/asterisk/bin/one_touch_record.php SIP/240-000000da") in new stack

But nothing in the monitor folder. All lines are set to allow recording. The calls go through ring groups to the individual extensions.

Is this a bug?
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Re: Call Recording on User Extension

Postby navaismo » Tue Aug 27, 2013 1:25 pm

Seems like you are using FreePBX, are you sure that the extension has the "On Demand Recording" as enabled?
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Re: Call Recording on User Extension

Postby pdecker414 » Tue Aug 27, 2013 2:52 pm

Yes. I set Inbound External Calls, Outbound External Calls, Inbound Internal Calls, Outbound Internal Calls to always and tried both On Demand Recording enabled and disabled. Each line in Call Recording is set to Allow. The ring groups Record Calls have been set to On Demand and Always to no avail.
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Re: Call Recording on User Extension

Postby pdecker414 » Tue Aug 27, 2013 3:20 pm

Here is the CLI output when I press *1

-- Feature Found: apprecord exten: apprecord
-- Executing [s@macro-one-touch-record:1] System("SIP/240-000000fd", "/var/lib/asterisk/bin/one_touch_record.php SIP/240-000000fd") in new stack
-- Executing [s@macro-one-touch-record:2] MacroExit("SIP/240-000000fd", "") in new stack
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Re: Call Recording on User Extension

Postby navaismo » Tue Aug 27, 2013 3:23 pm

Check the apache log for errors since freepbx use an external PHP script to generate the recording.
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Re: Call Recording on User Extension

Postby pdecker414 » Wed Aug 28, 2013 3:14 pm

I looked through the log file which I assume is httpd but cannot find anything that might help find out what is going on. Any other ideas?
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Re: Call Recording on User Extension

Postby pdecker414 » Wed Aug 28, 2013 3:18 pm

If I choose to record all calls in the ring group, I do get .wav files in the monitor folder in the following structure /var/spool/asterisk/monitor/2013/08/28
/rg-601-unknown-20130828-085000-1377705000.446.wav
If I set it to on demand and press *1 on the extension I see events saying one touch record in the CLI asterisk -vvvrrrrrr
But no recordings are left in the monitor folder nor any other folder.
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Re: Call Recording on User Extension

Postby pdecker414 » Fri Aug 30, 2013 9:39 am

I am using AsteriskNow 3.0

Ok. After a lot of digging through PHP and logs and conf files, I found a few of my answers.

If I set an incoming call to go to my extension, on demand recording does not work but record always does. I see one touch record triggered on the CLI when I press *1 but no recording happens. If I set the extension rules to always record I get a recording for inbound and outbound calls. Just not for on demand. But......if I am using ring groups and set ring groups to on demand and I call out or call in, nothing happens. If I set ring groups to always record I see recordings for inbound calls but it is tied to the ring group so the extension cannot see their recorded call but an admin can if set in the main.conf.php file. Also if calls arrive through ring groups and the call record is set to on demand and the extension is set to always, no recording happens.

So it seams to me that there are two bugs present. One bug is on demand recordings do not happen even though I have enabled on demand on the extension and allowed call recording on the inbound and outbound routes but it is witnessed in the CLI. Maybe there is a hidden setting somewhere that needs to be set.

The second bug is in Ring Groups. Apparently if set to on demand and the extension is set to always record, ring groups is blocking or not passing the downstream setting. The second part of the bug related to ring groups is when the recording happens such as rg-601-unknown-20130830-081217-1377875537.349.wav the extension that needs to be able to playback the file is not allowed to see the file since it is attached to the ring group only (rg-601). Only the admin can see them.

One final note is when using the user portal and call recordings happen, if you are already logged in and you press call monitor you will see the new call but no recording. If you refresh the page you will still not see the recording. If you log out and back in again you will then see the recording.

I hope my findings help others including the great programmers who work on these projects and updates/fixes are provided for the end users like myself.
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Re: Call Recording on User Extension

Postby navaismo » Fri Aug 30, 2013 3:55 pm

You can try to debug deeper by yourself or open a ticket in the freepbx tracker issue: http://issues.freepbx.org/secure/Dashboard.jspa

If you want to go deeper you need to generate a call, then connect to the asterisk cli and obtain the sip channel running the command: core show channels verbose

You need to copy the complete channel like sip/500-0001, then go to the location of the php file: /var/lib/asterisk/bin/one_touch_record.php

And run the script from linux shell with the channel name as argument, you can see the output of that script and see if there is an error.

When you perform the *1 feature you will see(always) in the asterisk cli that the application run, but that doesn't guarantee the recording, that only lunch the php script, then the script validate if the exten can record or not.
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Re: Call Recording on User Extension

Postby pdecker414 » Fri Aug 30, 2013 5:33 pm

-- SIP/240-00000141 connected line has changed. Saving it until answer for DAHDI/4-1
-- SIP/240-00000141 answered DAHDI/4-1
-- Executing [s@macro-auto-blkvm:1] Set("SIP/240-00000141", "__MACRO_RESULT=") in new stack
-- Executing [s@macro-auto-blkvm:2] Macro("SIP/240-00000141", "blkvm-clr,") in new stack
-- Executing [s@macro-blkvm-clr:1] Set("SIP/240-00000141", "SHARED(BLKVM,DAHDI/4-1)=") in new stack
-- Executing [s@macro-blkvm-clr:2] Set("SIP/240-00000141", "GOSUB_RETVAL=") in new stack
-- Executing [s@macro-blkvm-clr:3] MacroExit("SIP/240-00000141", "") in new stack
-- Executing [s@macro-auto-blkvm:3] ExecIf("SIP/240-00000141", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(num))=240)") in new stack
-- Executing [s@macro-auto-blkvm:4] ExecIf("SIP/240-00000141", "0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=Philip (Kalypso))") in new stack
-- Feature Found: apprecord exten: apprecord
-- Executing [s@macro-one-touch-record:1] System("SIP/240-00000141", "/var/lib/asterisk/bin/one_touch_record.php SIP/240-00000141") in new stack
-- Executing [s@macro-one-touch-record:2] MacroExit("SIP/240-00000141", "") in new stack


asterisk*CLI> core show channels verbose
Channel Context Extension Prio State Application Data CallerID Duration Accountcode PeerAccount BridgedTo
SIP/240-00000141 macro-dial s 1 Up AppDial (Outgoing Line) 240 00:00:07 DAHDI/4-1
DAHDI/4-1 macro-dial s 7 Up Dial SIP/210&SIP/220&SIP/230&S 00:00:07 SIP/240-00000141
2 active channels
1 active call
52 calls processed

OK here is what it said when I executed the php script.

[root@asterisk 30]# cd /var/lib/asterisk/bin/
[root@asterisk bin]# php one_touch_record.php SIP/240-00000141
Starting...
Channel: SIP/240-00000141
Gathering variables
Failed to get var PICKUP_EXTEN exiting


Me being from the windows world it takes a bit of getting used to linux but I am working on it. Does this tell you anything besides it being unable to determine the pickup extension?
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Re: Call Recording on User Extension

Postby navaismo » Fri Aug 30, 2013 7:24 pm

Well the error in the console is clear, the script can't obtain the value of that VAR, based on the php script in this part of the code:

Code: Select all
function getVariable($channel, $varName) {
   global $astman;

   $results = $astman->GetVar($channel, $varName, rand());

   if($results["Response"] != "Success"){
      ot_debug("Failed to get var {$varName} exiting");
      exit(1);
   }

   return $results["Value"];
}


It is connecting via manager to Asterisk and trying to get the value of the variable(in this case PICKUP_EXTEN) and its failing, so again looking at the code, the script is not receiving a SUCCESS response from asterisk. The documentation about getvar says this-->http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+GetVar

There are two responses: success or error, so or you are getting error because the channel doesn't exist or the script can't connect at all. You can add an: echo $results["Response"]; to see exactly the value of the response.
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Re: Call Recording on User Extension

Postby pdecker414 » Fri Aug 30, 2013 7:58 pm

I did what you suggested. Here is the response. By the way Echo did the same but I stayed with what was in the code. How do I drill down any further to find out what Error means?

Error
Failed to get var PICKUP_EXTEN exiting


ot_debug($channel);
ot_debug($varName);
ot_debug(rand());
ot_debug($results["Response"]);
if($results["Response"] != "Success"){
ot_debug("Failed to get var {$varName} exiting");
exit(1);
}
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Re: Call Recording on User Extension

Postby navaismo » Fri Aug 30, 2013 8:59 pm

Add an echo with the response, with echo $results["Message"];
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Re: Call Recording on User Extension

Postby pdecker414 » Sat Aug 31, 2013 2:47 pm

Here is what I put in the PHP

$results = $astman->GetVar($channel, $varName, rand());
ot_debug($channel);
ot_debug($varName);
ot_debug(rand());
echo "Result: ";
echo $results["Response"];
echo "\n";
if($results["Response"] != "Success"){
ot_debug("Failed to get var {$varName} exiting");
exit(1);
}

return $results["Value"];
}


Here is the resultant output

Starting...
Channel: SIP/240-00000141
Gathering variables
SIP/240-00000141
PICKUP_EXTEN
1572509617
Result: Error
Failed to get var PICKUP_EXTEN exiting
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Re: Call Recording on User Extension

Postby navaismo » Sat Aug 31, 2013 8:34 pm

Like I said before you need to add the echo to see the explicit message of failure and cause that must be added inside the IF.
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Re: Call Recording on User Extension

Postby pdecker414 » Sun Sep 01, 2013 9:29 am

OK. I didn't see that in an earlier post you had put echo $results["Response"]; which I had been doing but then later put echo $results["Message"];. I had to read all the posts several times to notice the change. Sorry I didn't catch it earlier.

Anyway now I do get an error message which makes no sense to me.

No such channel

SIP/240 is valid as an extension

php one_touch_record.php SIP/240-00000141

Can you shed some light on this?

Thank you for your help thus far.
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Re: Call Recording on User Extension

Postby navaismo » Sun Sep 01, 2013 9:54 am

Paste the output of core show channels verbose, then paste command to run the PHP script & the result.
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Re: Call Recording on User Extension

Postby pdecker414 » Mon Sep 02, 2013 11:37 am

OK. I am getting somewhere. I was unaware that I must run the php while the call is in session. Learning. :?
When the call is in session I press one touch record and see the results:

-- Feature Found: apprecord exten: apprecord
-- Executing [s@macro-one-touch-record:1] System("SIP/240-00000012", "/var/lib/asterisk/bin/one_touch_record.php SIP/240-00000012") in new stack
-- Executing [s@macro-one-touch-record:2] MacroExit("SIP/240-00000012", "") in new stack


But no recording happens. But if I do the same thing again and run the PHP manually it records.

php one_touch_record.php SIP/240-00000012

So apparently even though it says it is executing the one_touch_record.php script it is not because manually running it works.
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Re: Call Recording on User Extension

Postby navaismo » Mon Sep 02, 2013 1:24 pm

I was talking with one of the FreePBX dev members and he mention that is something about saving the recording without wav format. Is this your case? Are you saving the recordings in gsm?

If you think the php is not running from asterisk, check the PHP error log when you trigger the *1 to find php or permission errors. Also check that the php script its owned by asterisk and the recordings paths too.

You can open and issue on the freepbx tracker, there you get help from freepbx developers i'm just telling you stuff based on your results.
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Re: Call Recording on User Extension

Postby pdecker414 » Mon Sep 02, 2013 6:42 pm

I am saving in wav format.
I checked the ownership of the php script and the ownership of the path and both are asterisk.
I am having trouble finding any php error log. Where might I find this? I tried to change the default in php.ini but that still does not create a log file.
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Re: Call Recording on User Extension

Postby navaismo » Mon Sep 02, 2013 8:25 pm

Normally the logs are in /var/log/ in the case of PHP in /var/log/httpd/error_log if you are using HTTPS see the ssl_error_log.
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Re: Call Recording on User Extension

Postby pdecker414 » Tue Sep 03, 2013 8:24 pm

Unfortunately there are no errors reported. It simply does not run but works if I manually run it from a command prompt.

-- Feature Found: apprecord exten: apprecord

One touch command is run by pressing *1

-- Executing [s@macro-one-touch-record:1] System("SIP/240-0000018a", "/var/lib/asterisk/bin/one_touch_record.php SIP/240-0000018a") in new stack
-- Executing [s@macro-one-touch-record:2] MacroExit("SIP/240-0000018a", "") in new stack
-- Executing [h@macro-dial:1] Macro("DAHDI/4-1", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("DAHDI/4-1", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("DAHDI/4-1",
"0?Set(CDR(recordingfile)=)") in new stack

Running the command /var/lib/asterisk/bin/php one_touch_record.php SIP/240-0000018a
causes the following:


== Begin MixMonitor Recording SIP/240-0000018a
-- Executing [h@macro-dial:1] Macro("DAHDI/4-1", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("DAHDI/4-1", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("DAHDI/4-1", "1?Set(CDR(recordingfile)=
rg-601-unknown-20130903-191145-1378260705.525.wav)") in new stack

As you can see asterisk is the owner of the php file so there should b no reason for it not to run.

-rwxrwxr--. 1 asterisk asterisk 10262 Aug 15 10:00 archive_recordings
lrwxrwxrwx 1 asterisk asterisk 49 Aug 15 10:24 backup.php -> /var/www/html/admin/modules/backup/bin/backup.php
-rwxrwxr--. 1 asterisk asterisk 1920 Aug 15 10:00 freepbx-cron-scheduler.php
-rwxrwxr-x. 1 asterisk asterisk 17269 Aug 15 10:00 freepbx_engine
lrwxrwxrwx. 1 asterisk asterisk 75 Aug 15 10:00 freepbx_engine_hook_dahdiconfig -> /var/www/html/admin/modules/dahdiconfig/bin/freepbx_engine_hook_dahdiconfig
-rwxrwxr--. 1 asterisk asterisk 17177 Aug 15 10:00 freepbx_engine.orig
-rwxrwxr-x. 1 asterisk asterisk 3206 Aug 15 10:00 freepbx_setting
-rwxrwxr-x. 1 asterisk asterisk 776 Aug 15 10:00 gen_amp_conf.php
-rwxrwxr--. 1 asterisk asterisk 2739 Aug 15 10:00 generate_hints.php
lrwxrwxrwx. 1 asterisk asterisk 63 Aug 15 10:00 generate_queue_hints.php -> /var/www/html/admin/modules/queues/bin/generate_queue_hints.php
-rwxrwxr--. 1 asterisk asterisk 9495 Aug 15 10:00 libfreepbx.confgen.php
-rwxrwxr--. 1 asterisk asterisk 23315 Aug 15 10:00 module_admin

lrwxrwxrwx 1 asterisk asterisk 66 Aug 15 10:29 one_touch_record.php -> /var/www/html/admin/modules/callrecording/bin/one_touch_record.php
lrwxrwxrwx. 1 asterisk asterisk 60 Aug 15 10:00 queue_reset_stats.php -> /var/www/html/admin/modules/queues/bin/queue_reset_stats.php
lrwxrwxrwx 1 asterisk asterisk 50 Aug 15 10:24 restore.php -> /var/www/html/admin/modules/backup/bin/restore.php
-rwxrwxr--. 1 asterisk asterisk 35060 Aug 15 10:00 retrieve_conf
-rwxrwxr--. 1 asterisk asterisk 621 Aug 15 10:00 retrieve_parse_amportal_conf.pl
lrwxrwxrwx 1 asterisk asterisk 58 Aug 15 10:24 schedtc.php -> /var/www/html/admin/modules/timeconditions/bin/schedtc.php


Any other ideas?
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Re: Call Recording on User Extension

Postby navaismo » Tue Sep 03, 2013 11:10 pm

Just one thing the script is running in both ways, but it isn't generating the recording when you trigger from the sip peer. Try to write the output of the script to a log file, then compare the log file when you trigger the script from the sip peer against the Linux shell.

Did you open the case with the freepbx team?
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Re: Call Recording on User Extension

Postby pdecker414 » Wed Sep 04, 2013 8:34 pm

When I run the script manually lots of information is displayed on the Telnet window. I can copy this information and send it if you would like. As far as a log showing this detail. I do see it in the freepbx_dbug log but only when I manually trigger the php script. Otherwise...nothing at all.

I have not officially started a case with Freepbx team as of yet. I wanted to make sure I wasn't being stupid first.
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Re: Call Recording on User Extension

Postby pdecker414 » Wed Sep 04, 2013 9:08 pm

Got new information. If I do a core show channel SIP/240-00000xxx
of course the xxx is a changing number but anyway....
I get this:


-- General --
Name: SIP/240-0000028e
Type: SIP
UniqueID: 1378349677.834
LinkedID: 1378349677.830
Caller ID: 240
Caller ID Name: Philip (Kalypso)
Connected Line ID: (N/A)
Connected Line ID Name: AfterHours
Eff. Connected Line ID: (N/A)
Eff. Connected Line ID Name: AfterHours
DNID Digits: (N/A)
Language: en
State: Up (6)
Rings: 0
NativeFormats: (ulaw)
WriteFormat: ulaw
ReadFormat: ulaw
WriteTranscode: No
ReadTranscode: No
1st File Descriptor: 48
Frames in: 3256
Frames out: 3244
Time to Hangup: 0
Elapsed Time: 0h1m7s
Direct Bridge: DAHDI/4-1
Indirect Bridge: DAHDI/4-1
-- PBX --
Context: macro-dial
Extension: s
Priority: 1
Call Group: 0
Pickup Group: 0
Application: AppDial
Data: (Outgoing Line)
Blocking in: ast_waitfor_nandfds
Call Identifer: [C-000000d2]
Variables:
BRIDGEPEER=DAHDI/4-1
MACRO_DEPTH=0
SYSTEMSTATUS=APPERROR
DYNAMIC_FEATURENAME=apprecord
DYNAMIC_PEERNAME=DAHDI/4-1
DB_RESULT=Philip (Kalypso)
GOSUB_RETVAL=
MACRO_RESULT=
DIALEDPEERNUMBER=240
SIPCALLID=2734e6356899dc2650de489172c9c1d8@192.168.1.20:5060
SIPADDHEADER01=Alert-Info:Classic-1
KEEPCID=TRUE
CALLFILENAME=rg-601-unknown-20130904-195437-1378349677.830
FROMEXTEN=unknown
TIMESTR=20130904-195437
YEAR=2013
MONTH=09
DAY=04
REC_STATUS=INITIALIZED
REC_POLICY_MODE=dontcare
MON_FMT=wav
ALERT_INFO=Classic-1
RGPREFIX=AfterHours
NODEST=601
BLKVM_CHANNEL=DAHDI/4-1
TTL=64
CALLINGPRES_SV=allowed_not_screened
FROM_DID=8188414279

CDR Variables:
level 1: dnid=
level 1: clid="Philip (Kalypso)" <240>
level 1: src=240
level 1: dst=s
level 1: dcontext=from-internal
level 1: channel=SIP/240-0000028e
level 1: lastapp=MacroExit
level 1: start=2013-09-04 19:54:37
level 1: answer=2013-09-04 19:54:39
level 1: duration=66
level 1: billsec=64
level 1: disposition=ANSWERED
level 1: amaflags=DOCUMENTATION
level 1: uniqueid=1378349677.834
level 1: linkedid=1378349677.830
level 1: sequence=982

Now I will post another call with the manually triggered recording

-- General --
Name: SIP/240-00000296
Type: SIP
UniqueID: 1378349879.843
LinkedID: 1378349879.839
Caller ID: 240
Caller ID Name: Philip (Kalypso)
Connected Line ID: (N/A)
Connected Line ID Name: AfterHours
Eff. Connected Line ID: (N/A)
Eff. Connected Line ID Name: AfterHours
DNID Digits: (N/A)
Language: en
State: Up (6)
Rings: 0
NativeFormats: (ulaw)
WriteFormat: ulaw
ReadFormat: ulaw
WriteTranscode: No
ReadTranscode: No
1st File Descriptor: 48
Frames in: 1123
Frames out: 1117
Time to Hangup: 0
Elapsed Time: 0h0m25s
Direct Bridge: DAHDI/4-1
Indirect Bridge: DAHDI/4-1
-- PBX --
Context: macro-dial
Extension: s
Priority: 1
Call Group: 0
Pickup Group: 0
Application: AppDial
Data: (Outgoing Line)
Blocking in: ast_waitfor_nandfds
Call Identifer: [C-000000d3]
Variables:
ONETOUCH_RECFILE=rg-601-unknown-20130904-195759-1378349879.839.wav
MON_FMT=wav
MIXMONITOR_FILENAME=/var/spool/asterisk/monitor/2013/09/04/rg-601-unknown-20130904-195759-1378349879.839.wav
REC_STATUS=RECORDING
ONETOUCH_REC=RECORDING
MASTERONETOUCH=TRUE
THISEXTEN=601

BRIDGEPEER=DAHDI/4-1
MACRO_DEPTH=0
DB_RESULT=Philip (Kalypso)
GOSUB_RETVAL=
MACRO_RESULT=
DIALEDPEERNUMBER=240
SIPCALLID=3cd856c608e17efc19892f504d5042aa@192.168.1.20:5060
SIPADDHEADER01=Alert-Info:Classic-1
KEEPCID=TRUE
CALLFILENAME=rg-601-unknown-20130904-195759-1378349879.839
FROMEXTEN=unknown
TIMESTR=20130904-195759
YEAR=2013
MONTH=09
DAY=04
REC_POLICY_MODE=dontcare
ALERT_INFO=Classic-1
RGPREFIX=AfterHours
NODEST=601
BLKVM_CHANNEL=DAHDI/4-1
TTL=64
CALLINGPRES_SV=allowed_not_screened
FROM_DID=8188414279

CDR Variables:
level 1: recordingfile=rg-601-unknown-20130904-195759-1378349879.839.wav
level 1: dnid=
level 1: clid="Philip (Kalypso)" <240>
level 1: src=240
level 1: dst=s
level 1: dcontext=from-internal
level 1: channel=SIP/240-00000296
level 1: lastapp=ExecIf
level 1: lastdata=0?Set(MASTER_CHANNEL(CONNECTEDLINE(name))=Philip (Kalypso))
level 1: start=2013-09-04 19:57:59
level 1: answer=2013-09-04 19:58:01
level 1: duration=24
level 1: billsec=22
level 1: disposition=ANSWERED
level 1: amaflags=DOCUMENTATION
level 1: uniqueid=1378349879.843
level 1: linkedid=1378349879.839
level 1: sequence=992
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Re: Call Recording on User Extension

Postby navaismo » Wed Sep 04, 2013 9:18 pm

Well we don't know the real behavior behind the script so asking to the developers is the best way and the worst scenario is the dev team close your case with the response about the non issue.


To create the log file you can add an echo below every ot_debug string like exec("echo $thevar >> mylog.txt"); or using file manipulation routines(change thevar for the actual var inside ot_debug).
navaismo
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Re: Call Recording on User Extension

Postby navaismo » Wed Sep 04, 2013 10:33 pm

Open the ticket with that info.
navaismo
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Re: Call Recording on User Extension

Postby renatofb » Fri Jan 09, 2015 5:30 am

I am experiencing the exactly same problem as described, it seems that it has not been fixed yet.
Did you have any solution?
Do you have a ticket # I can relate to?

BR
Renato

PBX Firmware: 6.12.65-24
ASTERISK 11
FreePBX 12.0.25
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