screen -r

[bloureiro]

Hi all,

I'm using Slackware 10.2
astguiclient 2.0.1b2

My vicidial is only in test... but I have some problems, like when an agent is paused, he appeares as "0 agents logged in on all servers" it woul be as paused... I can only hangup a call clicking in LIVE CALLS IN YOUR SESSION: and HANGUP at the bottom of the vicidial page.

I don't know why I have repeats entires in screen -r. I think this is my problem.

There are several suitable screens on:
4839.ASTsend (Detached)
4766.ASTVDauto (Dead ???)
4920.ASTlisten (Dead ???)
4928.ASTlisten (Dead ???)
5015.ASTVDremote (Dead ???)
5021.ASTupdate (Dead ???)
4882.ASTlisten (Dead ???)
4888.ASTsend (Dead ???)
4893.ASTVDauto (Dead ???)
4898.ASTVDremote (Dead ???)
4779.asterisk (Detached)
4963.ASTlisten (Dead ???)
4969.ASTlisten (Dead ???)
4977.ASTsend (Dead ???)
5049.ASTVDremote (Dead ???)
5055.ASTupdate (Dead ???)
4861.ASTsend (Dead ???)
4860.ASTsend (Dead ???)
4865.ASTVDauto (Dead ???)
4870.ASTupdate (Dead ???)
4870.ASTVDremote (Dead ???)
4873.ASTVDremote (Dead ???)
4906.ASTsend (Dead ???)
4883.ASTupdate (Dead ???)
4886.ASTVDremote (Dead ???)
4828.ASTlisten (Detached)
4883.ASTlisten (Dead ???)
4890.ASTlisten (Dead ???)
4963.ASTupdate (Dead ???)
4969.ASTVDremote (Dead ???)
4866.ASTVDauto (Dead ???)
4871.ASTVDremote (Dead ???)
4875.ASTupdate (Dead ???)
4847.ASTVDadapt (Detached)
4849.ASTVDauto (Detached)
4856.ASTupdate (Detached)
4860.ASTVDremote (Detached)


my /etc/rc.d/rc.local

### start up the MySQL server
/usr/local/mysql/bin/mysqld_safe --user=mysql --skip-name-resolve --skip-host-cache &

### start up the apache web server
/usr/local/apache2/bin/apachectl start

### load digium tormenta 4xT1 drivers into system
/sbin/ztcfg -vvvvvvvvvvvv

### sybsys local login
touch /var/lock/subsys/local

### sleep for 20 seconds before launching Asterisk
sleep 20

### start up asterisk
#asterisk
/usr/share/astguiclient/start_asterisk_boot.pl



Best Regards,

Loureiro.

[mflorell]

What Asterisk version?

Can you post Asterisk CLI output of when you try to hangup a call properly in vicidial?

What is the loadavg of your server?

[bloureiro]

asterisk 1.2.9.1
===============================================
root@servidor:~# cat /proc/loadavg
0.06 0.09 0.07 2/112 6734

===============================================
in CLI sometimes print this, but when I click in Hangup Customer nothing is output in CLI, Only when the agent goes to the bottom of the page LIVE CaLLS in your Session.
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1



================================================

The paused agents doesnt working

0 agents logged in 0 agents in calls 0 agents waiting 0 paused agents
===============================================
root@servidor:~# screen -r
There are several suitable screens on:
4839.ASTsend (Dead ???)
4766.ASTVDauto (Dead ???)
4920.ASTlisten (Dead ???)
4928.ASTlisten (Dead ???)
5015.ASTVDremote (Dead ???)
5021.ASTupdate (Dead ???)
4882.ASTlisten (Dead ???)
4888.ASTsend (Dead ???)
4893.ASTVDauto (Dead ???)
4898.ASTVDremote (Dead ???)
4762.asterisk (Detached)
4963.ASTlisten (Dead ???)
4969.ASTlisten (Dead ???)
4977.ASTsend (Dead ???)
5049.ASTVDremote (Dead ???)
5055.ASTupdate (Dead ???)
4861.ASTsend (Dead ???)
4860.ASTsend (Dead ???)
4865.ASTVDauto (Dead ???)
4870.ASTupdate (Dead ???)
4870.ASTVDremote (Dead ???)
4873.ASTVDremote (Dead ???)
4906.ASTsend (Dead ???)
4883.ASTupdate (Dead ???)
4886.ASTVDremote (Dead ???)
4941.ASTlisten (Detached)
4883.ASTlisten (Dead ???)
4890.ASTlisten (Dead ???)
4963.ASTupdate (Dead ???)
4969.ASTVDremote (Dead ???)
4866.ASTVDauto (Dead ???)
4871.ASTVDremote (Dead ???)
4875.ASTupdate (Dead ???)
4950.ASTsend (Detached)
4957.ASTVDauto (Detached)
4961.ASTVDadapt (Detached)
4969.ASTupdate (Detached)
4972.ASTVDremote (Detached)
6477.pts-1.servidor (Attached)
Remove dead screens with 'screen -wipe'.
Type "screen [-d] -r [pid.]tty.host" to resume one of them.

=============================================
Is my "screen -r" normal?

[mflorell]

can you try running "AST_manager_listen.pl --debug" directly in the command line and see if it dies and what the die message is?

[bloureiro]

root@servidor:/usr/share/astguiclient# ./AST_manager_listen.pl --debug
input buffer: 79 lines: 0 partial: 0
|Event: PeerStatus
Privilege: system,all
Peer: SIP/7203
PeerStatus: Registered

|
Killedounter: |863999|1|

[bloureiro]

I run again with call into the meetme. Agent in live call


root@servidor:/usr/share/astguiclient# ./AST_manager_listen.pl --debug

input buffer: 79 lines: 0 partial: 0
|Event: PeerStatus
Privilege: system,all
Peer: SIP/7203
PeerStatus: Registered

|
loop counter: |863999|1|



loop counter: |863998|2|




input buffer: 79 lines: 0 partial: 0
|Event: PeerStatus
Privilege: system,all
Peer: SIP/7203
PeerStatus: Registered

|
input buffer: 79 lines: 0 partial: 0
|Event: PeerStatus
Privilege: system,all
Peer: SIP/7203
PeerStatus: Registered

|
input buffer: 79 lines: 0 partial: 0
|Event: PeerStatus
Privilege: system,all
Peer: SIP/7203
PeerStatus: Registered

|
input buffer: 79 lines: 0 partial: 0
|Event: PeerStatus
Privilege: system,all
Peer: SIP/7203
PeerStatus: Registered

|
input buffer: 79 lines: 0 partial: 0
|Event: PeerStatus
Privilege: system,all
Peer: SIP/7203
PeerStatus: Registered

|
input buffer: 119 lines: 0 partial: 0
|Event: Registry
Privilege: system,all
Channel: IAX2
Username: 1112223333
Status: Rejected
Cause: Registration Refused

|
input buffer: 79 lines: 0 partial: 0
|Event: PeerStatus
Privilege: system,all
Peer: SIP/7203
PeerStatus: Registered

|
input buffer: 79 lines: 0 partial: 0
|Event: PeerStatus
Privilege: system,all
Peer: SIP/7203
PeerStatus: Registered

|
Killedounter: |863978|22|

[mflorell]

You will need to let it run for a while. from the look of your screen -r output it should crash after a few hours. Either that or you are doing hard reboots very often.

[bloureiro]

I started the process whithout agents logged. After some time I logged one agent and did auto-dial and put the agent in pause and resume several times.

Now, The status "paused" has ALWAYS been working for the first 5:00 min after it the ticket below disappear.

i.e If a user has been as ready for a call for less than 5.00 min and the user click in the "pause" , so the status paused below appear. But, if he stay as ready more the 5.00 when he click in pause the table below is empty.

Or if a user is paused more the 5.00 min its desappear too.
+------------|--------+-----------+--------+-----------------+-----------------+---------+------------+
| STATION | USER | SESSIONID | STATUS | SERVER IP | CALL SERVER IP | MM:SS | CAMPAIGN |
+------------|--------+-----------+--------+-----------------+-----------------+---------+------------+
| SIP/7205 | 7205 | 8600051 | PAUSED | 10.10.10.15 | | 1:22 | VENDAS2 |
+------------|--------+-----------+--------+-----------------+-----------------+---------+------------+
1 agents logged in on all servers
System Load Average: 0.04

This problem is only when put some agent as paused. When he is waiting for a call, is ready , in call or in call over 5 min NO PROBLEMs.

The other problem still the same. "hangup customer". I can only hangup someone by clicking at the bottom of the page in LIVE CALLS IN YOUR SESSION:

Sometimes if I press " hangup customer" very quick after call enter in meetme room its works.

I receice a lot of messages like this in terminal
=================================================
From: root@servidor.voipexpress
Subject: cron: /usr/share/astguiclient/ADMIN_keepalive_AST_send_listen.pl
usage: kill [ -s signal | -p ] [ -a ] pid ...
kill -l [ signal ]
==================================================

There are several suitable screens on:
4870.ASTsend (Detached)
4769.ASTVDauto (Detached)
4772.ASTVDadapt (Detached)
4826.asterisk (Detached)
5012.ASTupdate (Detached)
5016.ASTVDremote (Detached)
4855.ASTlisten (Detached)
Type "screen [-d] -r [pid.]tty.host" to resume one of them.
You have mail in /var/mail/root
===============================================

the ./AST_manager_listen.pl --debug is still running in loop in the terminal.
===============================================

I'm running the astguiclient beta 2

[mflorell]

The timeonVDADall.php script is set to not display PAUSED agent after 5 minutes of being paused.

As for the Hangup problem, what kind of trunks are you using?

What shows up in the channel field in vicidial.php after a call connects?

Does this change after the call has gone on for a while?

[bloureiro]

ok, so I have no problem with the paused agents. thanks!

I'm using SIP/Trunks.

Today it's only in test, it will be up until Friday with 3E1s in auto dial

The channel field is SIP/7203

If I press "hangup customer" quick during the first seconds of the call sometimes works. Another way no.

I will conncect the 3E1s in the next days. Maybe with zap in group, no problems...

I still running the AST_manager_listen.pl

I will have to do it always?

[bloureiro]

Does this change after the call has gone on for a while?

<> after a while I cant hangup only at the bottom.


Some times i can hangup at the begin, never after 2.00 min or more ...

[mflorell]

Is this hangup issue only happening with autodial, manualdial or both?

After 10-20 seconds, compare the channel in the channel field to the manual hangup field at the bottom and post both please.

[bloureiro]

In AUTO-Dial:

in channel field SIP/7203-609b
at the bottom of the page SIP/7203-609b

Hangup Customer has Failed several times (call duration >5s)
It only worked in one at 3:39 min.
----------------------------------------------------------------------------------------------------------------------------------------------
SIP/7203-3510
SIP/7203-3510

2.00 min FAILED
----------------------------------------------------------------------------------------------------------------------------------------------
SIP/7203-ce21
SIP/7203-ce21
is always the same channel in the field or at the manual link at the bottom
----------------------------------------------------------------------------------------------------------------------------------------

I can hangup anyone customer <5 s ;
anything >5s only one hanged up in several calls

Clicking at the bottom link ---> no problem.
=================================================
In MAnual Dial

The agent is in the meetme room
the Screen Detail of the Campaign
NO LIVE CALLS WAITING NO AGENTS ON
xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx

when the agent dial the number goes
Waiting for Ring... from 60 to 1 seconds

at the bottom of the page the channel is Local/8600051@default-3657,2 (but it's the SIP/7203 like the auto-dial example)

the "hangup customer" button is red

Channel field, Seconds, Cust Time are all of them empties
pop-up a window - "Dial timed out, contact your system adminitrator"


But they are both, agent and outbound call, in the meetme room they can talk... and they appear at the bottom links

Maybe something is wrong in my extensions.conf? do I need answer or ring before dial?

Thanks

[bloureiro]

<> You will need to let it run for a while. from the look of your screen -r output it should crash after a few hours. Either that or you are doing hard reboots very often.

I didn't understand it very well...

do I need to run manualy ./AST_manager_listen.pl every time that I boot the server? Do I need to check it during the day to see if it crashed? What does it do?

[mflorell]

You do not need to run AST_manager_listen.pl manually when you reboot. That was just to test whether it was dying.

As for your hangup issue, I cannot duplicate it using SIP trunks on the latest 2.0.1b2 release.

Are you able to hangup properly in manual dial mode?

[bloureiro]

In manual dial I cant hangup customer too.
I have a lot of problems in manual dial.
In auto dial I can only hangup customer in the firsts 5s of call.

below a RESUME :
------------------------------------------------------------------------------------------------------------------------------------
In MAnual Dial

The agent is in the meetme room
the Screen Detail of the Campaign
NO LIVE CALLS WAITING NO AGENTS ON
xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx

when the agent dial the number goes
Waiting for Ring... from 60 to 1 seconds

at the bottom of the page the channel is Local/8600051@default-3657,2 (but it's the SIP/7203 like the auto-dial example)

the "hangup customer" button is red

Channel field, Seconds, Cust Time are all of them empties
pop-up a window - "Dial timed out, contact your system adminitrator"


But they are both, agent and outbound call, in the meetme room they can talk... and they appear at the bottom links

Maybe something is wrong in my extensions.conf? do I need answer or ring before dial?

Thanks Wink

[mflorell]

can you post the results of these queries while agent and customer are both on:

SELECT * from live_channels;
SELECT * from live_sip_channels;

[bloureiro]

in auto dial:

I have

the sip channel field:
SIP/7203-ea94

LIVE CALLS IN YOUR SESSION:
# REMOTE CHANNEL HANGUP
1 SIP/7203-ea94 HANGUP
2 SIP/7205-b8a2 HANGUP


but the mysql live_channels table is the same with or without calls:

mysql> select * from live_channels;
+-----------------------+-------------+---------------+-----------+--------------+
| channel | server_ip | channel_group | extension | channel_data |
+-----------------------+-------------+---------------+-----------+--------------+
| Zap/pseudo-1092587195 | 10.10.10.15 | NULL | ring | SIP/ring |
+-----------------------+-------------+---------------+-----------+--------------+
Can I drop this?


the other is rigth:



mysql> select * from live_sip_channels;
+---------------+-------------+---------------+-----------+--------------+
| channel | server_ip | channel_group | extension | channel_data |
+---------------+-------------+---------------+-----------+--------------+
| SIP/7203-ea94 | 10.10.10.15 | NULL | 8600051 | 8600051 |
| SIP/7205-b8a2 | 10.10.10.15 | NULL | 8600051 | 8600051 |
+---------------+-------------+---------------+-----------+--------------+

[mflorell]

You cannot do test outbound dialing with phones on your local system. Have you tried dialing test calls out over outbound telco trunks?

[bloureiro]

I will do it tonight.

Thanks