GOautodial CE 3.3 Unable to autorecord phone dials

[Merchant007]

Sir we have some users who dial directly via Phone (3cx phone) i changed context to defaultlog but still it is not recording

all users who are dialing via campain manual dial are getting recorded but not the users who dont log in to web


System
Fresh install of goautodial 3.3 - uptodate


Kernel Version 2.6.18-371.11.1.el5 (SMP)
Distro Name GoAutoDial CE 3.3
Asterisk Version:1.8.23.0

when login to
https://192.168.1.104/vicidial/admin.php

VERSION: 2.9-441a
BUILD: 140612-1628
© 2014 ViciDial Group

[williamconley]

How do you know it's not recording? Where did you look for the recordings?

http://goautodial.org/boards/1/topics/1082

viewtopic.php?t=23007

[Merchant007]

Yes thank you i saw both threads , its after i updated all phones from admin to defaultlog but still there is no logs shown in

Admine -phone --bottom - phone recordings

2) how do i know if i am in latest SVN 2.2 ?

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Kernel Version 2.6.18-371.11.1.el5 (SMP)
Distro Name GoAutoDial CE 3.3
Asterisk Version:1.8.23.0

when login to
192.168.1.104/vicidial/admin.php

VERSION: 2.9-441a
BUILD: 140612-1628
© 2014 ViciDial Group

it says Version 2.9 so i am in latest rite and goautodial3.3 is the latest in website

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[root@go ~]# find / -name  agi-NVA_recording.agi
/usr/src/vicidial-2.9.441a-140612.1628.2.go/agi/agi-NVA_recording.agi
/var/lib/asterisk/agi-bin/agi-NVA_recording.agi
[root@go ~]#


i find same files in two location

How to modify the file so it will autorecord all outbound calls dialled from SIP phone ?
my outbound call will start with 91-xxxxxxxxxx



is this correct ?

# ;custom dialplan entry example:
exten => _91XXXXXXXXXX.,1,AGI(agi-NVA_recording.agi,BOTH------Y---N---Y---N)
exten => _91XXXXXXXXXX.,n,Goto(default,${EXTEN},1)
exten => _91XXXXXXXXXX.,n,Hangup


where 91 is the country code
and No of X is the total number of digits in phone number

[williamconley]

Perhaps posting output from an asterisk CLI example call (not 3000 lines of unrelated code, please ... LOL ... just from beginning to end of a single example call).

[Merchant007]

Phone DIAL _LOG

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[Oct 22 23:39:31]   == Using SIP RTP CoS mark 5
[Oct 22 23:39:31]     -- Executing [918792460213@default:1] AGI("SIP/8077-000000                                                                                        08", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 22 23:39:31]     -- <SIP/8077-00000008>AGI Script agi://127.0.0.1:4577/call                                                                                        _log completed, returning 0
[Oct 22 23:39:31]     -- Executing [918792460213@default:2] Dial("SIP/8077-00000                                                                                        008", "sip/gsmtrunk/8792460213,,tTo") in new stack
[Oct 22 23:39:31]   == Using SIP RTP CoS mark 5
[Oct 22 23:39:31]     -- Called sip/gsmtrunk/8792460213
[Oct 22 23:39:32]     -- SIP/gsmtrunk-00000009 is making progress passing it to                                                                                         SIP/8077-00000008
[Oct 22 23:39:32]     -- SIP/gsmtrunk-00000009 is making progress passing it to                                                                                         SIP/8077-00000008
[Oct 22 23:39:32]     -- SIP/gsmtrunk-00000009 is making progress passing it to                                                                                         SIP/8077-00000008
[Oct 22 23:39:32]     -- SIP/gsmtrunk-00000009 is making progress passing it to                                                                                         SIP/8077-00000008
[Oct 22 23:39:41]     -- SIP/gsmtrunk-00000009 answered SIP/8077-00000008
[Oct 22 23:39:48]     -- Executing [h@default:1] AGI("SIP/8077-00000008", "agi://127.0.0.1:4577/call_log--HVcauses--PRI----                                             -NODEBUG-----16-----ANSWER-----17-----7") in new stack
[Oct 22 23:39:48]     -- <SIP/8077-00000008>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----                                             ANSWER-----17-----7 completed, returning 0
[Oct 22 23:39:48]   == Spawn extension (default, 918792460213, 2) exited non-zero on 'SIP/8077-00000008'


In carrier Dial plan entry

exten => _91X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91X.,2,Dial(sip/gsmtrunk/${EXTEN:2},,tTo)
exten => _91X.,3,Hangup


Account entry

[gsmtrunk]
disallow=all
allow=gsm
allow=ulaw
type=friend
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=invite,port
nat=yes
host=192.168.1.x


Campain Manual Dial LOG

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[Oct 22 23:43:36]     -- Executing [8600052@default:1] MeetMe("Local/8600052@default-00000001;2", "8600052,F") in new stack
[Oct 22 23:43:36]        > Channel Local/8600052@default-00000001;1 was answered.
[Oct 22 23:43:36]     -- Executing [91918792460213@default:1] AGI("Local/8600052@default-00000001;1", "agi://127.0.0.1:4577/call_log") in new stack
[Oct 22 23:43:36]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct 22 23:43:36]     -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=BACKEND))
[Oct 22 23:43:36]     -- <Local/8600052@default-00000001;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Oct 22 23:43:36]     -- Executing [91918792460213@default:2] Dial("Local/8600052@default-00000001;1", "sip/gsmtrunk/918792460213,,tTo") in new stack
[Oct 22 23:43:36]   == Using SIP RTP CoS mark 5
[Oct 22 23:43:36]     -- Called sip/gsmtrunk/918792460213
[Oct 22 23:43:37]   == Manager 'sendcron' logged on from 127.0.0.1
[Oct 22 23:43:37]     -- Executing [58600052@default:1] MeetMe("Local/58600052@default-00000002;2", "8600052,Fmq") in new stack
[Oct 22 23:43:37]        > Channel Local/58600052@default-00000002;1 was answered.
[Oct 22 23:43:37]     -- Executing [8309@default:1] Answer("Local/58600052@default-00000002;1", "") in new stack
[Oct 22 23:43:37]     -- Executing [8309@default:2] Monitor("Local/58600052@default-00000002;1", "wav,20151023-091335_8792460213_BACKEND_admin") in new stack
[Oct 22 23:43:37]   == Manager 'sendcron' logged off from 127.0.0.1
[Oct 22 23:43:37]     -- Executing [8309@default:3] Wait("Local/58600052@default-00000002;1", "3600") in new stack
[Oct 22 23:43:38]     -- SIP/gsmtrunk-0000000a is making progress passing it to Local/8600052@default-00000001;1
[Oct 22 23:43:38]     -- SIP/gsmtrunk-0000000a is making progress passing it to Local/8600052@default-00000001;1
[Oct 22 23:43:38]     -- SIP/gsmtrunk-0000000a is making progress passing it to Local/8600052@default-00000001;1
[Oct 22 23:43:38]     -- SIP/gsmtrunk-0000000a is making progress passing it to Local/8600052@default-00000001;1
[Oct 22 23:43:51]     -- SIP/gsmtrunk-0000000a answered Local/8600052@default-00000001;1
[Oct 22 23:43:56]     -- Executing [h@default:1] AGI("Local/8600052@default-00000001;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----20-----5") in new stack
[Oct 22 23:43:56]     -- <Local/8600052@default-00000001;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----20-----5 completed, returning 0
[Oct 22 23:43:56]   == Spawn extension (default, 91918792460213, 2) exited non-zero on 'Local/8600052@default-00000001;1'
[Oct 22 23:43:56]   == Spawn extension (default, 8600052, 1) exited non-zero on 'Local/8600052@default-00000001;2'
[Oct 22 23:43:56]     -- Executing [h@default:1] AGI("Local/8600052@default-00000001;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Oct 22 23:43:56]     -- <Local/8600052@default-00000001;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0


[williamconley]

I'm not sure which of those calls was which, but both executed in "default", neither went through "defaultlog" first. So ... that's why there's no recording.

What precisely did you change to "defaultlog from admin"?

[Merchant007]

I'm not sure which of those calls was which, but both executed in "default", neither went through "defaultlog" first. So ... that's why there's no recording.

What precisely did you change to "defaultlog from admin"?

Exten Context: default
Phone Context: defaultlog

[Phone Details]

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hone Extension/Login:    8001
Phone Password:   
goautodial
Dial Plan Number:   
8001
Voicemail ID:   
8001
Outbound CallerID:   
0000000000
Server IP:   
Agent Screen Login:   
8001
SIP Registration Password:   
goautodial
Status:   
Active Account:   
Full Name:   
8001
Email:   
Delete Voicemail After Email:   
Company:   
Picture:   
New Messages:    0
Old Messages:    0
Client Protocol:   
Local GMT:      (Do NOT adjust for DST)
Phone Ring Timeout:   
60
On-Hook Agent:   
Default User:   
Default Pass:   
Default Campaign:   
Park Exten:   
8301
Conf Exten:   
8302
Monitor Prefix:   
8612
Recording Exten:   
8309
Voicemail Exten:   
8501
Voicemail Dump Exten:   
85026666666666
Exten Context:   
default
Phone Context:   
defaultlog
Call Logging:   
User Switching:   
Conferencing:   
Admin Hang Up:   
Admin Hijack:   
Admin Monitor:   
Call Park:   
Updater Check:   
AF Logging:   
Queue Enabled:   
CallerID Popup:   
Voicemail Button:   
Fast Refresh:   
Fast Refresh Rate:   
1000
Persistant MySQL:   
Auto Dial Next Number:   
Stop Recording After Each Call:   
Enable SIPSAK Messages:   

----------------------------------------------i got mail from system ----------

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From root@go.goautodial.org  Fri Oct 23 01:45:13 2015
Return-Path: <root@go.goautodial.org>
Received: from go.goautodial.org (localhost.localdomain [127.0.0.1])
   by go.goautodial.org (8.13.8/8.13.8) with ESMTP id t9N5jDQ3000722
   for <root@go.goautodial.org>; Fri, 23 Oct 2015 01:45:13 -0400
Received: (from root@localhost)
   by go.goautodial.org (8.13.8/8.13.8/Submit) id t9N5j9ZC032625;
   Fri, 23 Oct 2015 01:45:09 -0400
Date: Fri, 23 Oct 2015 01:45:09 -0400
Message-Id: <201510230545.t9N5j9ZC032625@go.goautodial.org>
From: root@go.goautodial.org (Cron Daemon)
To: root@go.goautodial.org
Subject: Cron <root@go> /usr/share/astguiclient/AST_CRON_audio_1_move_mix.pl
Content-Type: text/plain; charset=UTF-8
Auto-Submitted: auto-generated
X-Cron-Env: <SHELL=/bin/sh>
X-Cron-Env: <HOME=/root>
X-Cron-Env: <PATH=/usr/bin:/bin>
X-Cron-Env: <LOGNAME=root>
X-Cron-Env: <USER=root>

mv: cannot stat `/var/spool/asterisk/monitor/20151023-111406_984532****_70661251_Agent1-in.wav': No such file or directory
mv: cannot stat `/var/spool/asterisk/monitor/20151023-111406_9845321***_70661251_Agent1-out.wav': No such file or directory
mv: cannot stat `/var/spool/asterisk/monitor/20151023-111337_9886414***_TEST1_Agent2-in.wav': No such file or directory
mv: cannot stat `/var/spool/asterisk/monitor/20151023-111337_9886414***_TEST1_Agent3-out.wav': No such file or directory




[root@go ~]# find / -name *20151023-111406_984532*
/var/spool/asterisk/monitorDONE/MP3/20151023-111406_98453****_70661251_Agent1-all.mp3
/var/spool/asterisk/monitorDONE/ORIG/20151023-111406_98453****_70661251_Agent1-all.wav
find: /proc/3282/task/14010: No such file or directory
find: /proc/3282/task/14011: No such file or directory
You have mail in /var/spool/mail/root



My Crontab recording

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### recording mixing/compressing/ftping scripts
0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * * /usr/share/astguiclient/AST_CRON_audio_1_move_mix.pl
0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * * /usr/share/astguiclient/AST_CRON_audio_1_move_mix.pl --MIX
0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * * /usr/share/astguiclient/AST_CRON_audio_1_move_VDonly.pl
1,4,7,10,13,16,19,22,25,28,31,34,37,40,43,46,49,52,55,58 * * * * /usr/share/astguiclient/AST_CRON_audio_2_compress.pl --MP3
#2,5,8,11,14,17,20,23,26,29,32,35,38,41,44,47,50,53,56,59 * * * * /usr/share/astguiclient/AST_CRON_audio_3_ftp.pl --GSM



[williamconley]

I did not see "defaultlog" in the asterisk CLI output you sent. Unless I'm getting blind in my old age, or you did not post all of the output, the calls executed in [default] instead of [defaultlog].

That being said, if there was a recording and it disappeared before the mixing script could find it, it's possible there was another mixing script running already and it was processed before the second script could access it. In which case, you should find the mixed script in the ORIG/MP3 or another audio folder. Note that "updatedb" does not work on some of these folders as they are excluded from indexing (spool folders are often excluded from drive indexers), so you may need to actually check the directories manually.

Otherwise, you may need to upgrade to fix a bug if the file was somehow deleted (or troubleshoot that deletion).

[Merchant007]

I did not see "defaultlog" in the asterisk CLI output you sent. Unless I'm getting blind in my old age, or you did not post all of the output, the calls executed in [default] instead of [defaultlog].

That being said, if there was a recording and it disappeared before the mixing script could find it, it's possible there was another mixing script running already and it was processed before the second script could access it. In which case, you should find the mixed script in the ORIG/MP3 or another audio folder. Note that "updatedb" does not work on some of these folders as they are excluded from indexing (spool folders are often excluded from drive indexers), so you may need to actually check the directories manually.

Otherwise, you may need to upgrade to fix a bug if the file was somehow deleted (or troubleshoot that deletion).

Sir but i have enabled in each phone



is there any other reason why it is not loading ?

[williamconley]

I just read what I saw in the log. Did you see the word "defaultlog" in the cli entry you posted?

As I said: Either it's there and I overlooked it, or you didn't post the entire CLI entry for the call, or defaultlog was not used. If it was not used, the next place to check is the sip context for that phone to see if the entry has been updated to defaultlog.

[Merchant007]

Found a solution , by default goautodial was adding


Conf override for all Phones (admin-Phones)
type=friend
host=dynamic
secret=goautodial
canreinvite=no
context=default
qualify=yes
disallow=all
allow=ulaw
allow=g729
allow=gsm
qualify=yes

once i removed these overrides and executed reload from asterisk cli its recording

so now i need to remove all these values from all SIP phones settings and issue will be fixed