Doesn't know the call is connected

[djs]

Hi

I've been able to dig through the forum and find the answer to a lot of my initial problems, and I think I'm getting close to making everything work. I need a bit of help on what I hope is a few last things....

I'm very new to asterisk, and even newer to vicidial. Some of the terminology is lost on me, on what various terms are, what they do, etc. I'm working on that, but please pardon me if I use a term incorrectly. Asterisk with a vicidial dialect is not my native language.......

What do I have:
Asterisk 1.4.4, on CentOS, with vicidial 2.0.95. I have a Teliax SIP trunk configured for now.

What works:
Looking just at Asterisk:
- Meetme works. I can call into a meeting from several phones, no problem.
- Music on hold plays.
- Outgoing calls work
- Voicemail, etc, etc. As far as I've tried, the core asterisk installation is working pretty good.

What am I planning on using Vicidial for?
- Manual dial of leads lists. It's primarily going to be used to dial through call lists, nothing to fancy, 3-4 agents at most. I do not anticipate the need for closers, or incoming calls for vicidial.

Vicidial - what works:
- Loading lists
- Lists dropping into the hopper *
- Logging in to the agent side, getting a call on my phone for the conference.
- Outbound dialing, almost. I have it call my cell phone, my phone rings, I can answer and converse. **** This is where the problems start.

* Lists load into the hopper when I manually run the cron job that should be loading them automatically. Looking at the logs, cron is executing, but not all of the jobs. I'm trying to sort this out, but for testing, I just ran the hopper cron job manually.

PROBLEM>>>>
As an agent, I click on the Dial Next Number button. It dials, and rings the phone. After 60 seconds, whether or not the phone is answered, I get a 'dial timed out, contact your system administrator' message.

I'm trying to figure out what causes vicidial to know that the call was answered (or not), so I can focus my efforts in the right place. I'm a realist, and assume it is one of the cron jobs that isn't executing correctly, but I've tried to run them all while the phone was ringing, and it didn't work.

Any help is appreciated.....

Dan

[mflorell]

Please post results of "screen -r"

Is your crontab set up according to the SCRATCH_INSTALL?

I would recommend using Asterisk 1.2.18 over 1.4.4 for stability reasons.

[djs]

Thanks for looking at this.

[root@71-6-209-171 ~]# screen -r
There are several suitable screens on:
7827.ASTVDauto (Detached)
7235.pts-1.71-6-209-171 (Attached)
7818.ASTupdate (Detached)
7824.ASTlisten (Detached)
7821.ASTsend (Detached)
7642.asterisk (Detached)
7830.ASTVDremote (Detached)
Type "screen [-d] -r [pid.]tty.host" to resume one of them.


And my crontab is set as follows:
[root@71-6-209-171 ~]# crontab -l
### recording mixing for Asterisk run every 5 minutes
1,6,11,16,21,26,31,36,41,46,51,56 * * * 1,2,3,4,5,6 /usr/share/astguiclient/AST_
CRON_mix_recordings_BASIC.pl

### keepalive script for astguiclient processes
* * * * * /usr/share/astguiclient/ADMIN_keepalive_ALL.pl

### kill Hangup script for Asterisk updaters
* * * * * /usr/share/astguiclient/AST_manager_kill_hung_congested.pl

### updater for voicemail
* * * * * /usr/share/astguiclient/AST_vm_update.pl

### updater for conference validator
* * * * * /usr/share/astguiclient/AST_conf_update.pl

### flush queue DB table every hour for entries older than 1 hour
11 * * * * /usr/share/astguiclient/AST_flush_DBqueue.pl -q

### fix the vicidial_agent_log once every hour
33 * * * * /usr/share/astguiclient/AST_cleanup_agent_log.pl

### updater for VICIDIAL hopper
* * * * * /usr/share/astguiclient/AST_VDhopper.pl -q

### adjust the GMT offset for the leads in the vicidial_list table
1 1 * * * /usr/share/astguiclient/ADMIN_adjust_GMTnow_on_leads.pl --debug --post
al-code-gmt

### reset several temporary-info tables in the database
2 1 * * * /usr/share/astguiclient/AST_reset_mysql_vars.pl

### optimize the database tables within the asterisk database
3 1 * * * /usr/share/astguiclient/AST_DB_optimize.pl

## adjust time on the server with ntp
30 * * * * /usr/local/bin/ntpdate -u 18.145.0.30 2>/dev/null 1>&2

### VICIDIAL agent time log weekly summary report generation
2 0 * * 0 /usr/share/astguiclient/AST_agent_week.pl


When I shut asterisk down for a few minutes, I receive emails about connection refused on port 5038 when AST_vm_update.pl is executed, so I know the cron jobs are running. Also, looking at /var/log/cron, there are entries such as:
May 19 07:38:01 71-6-209-171 crond[30787]: (root) CMD (/usr/share/astguiclient/AST_manager_kill_hung_congested.pl)
May 19 07:39:01 71-6-209-171 crond[31695]: (root) CMD (/usr/share/astguiclient/AST_vm_update.pl)
May 19 07:39:01 71-6-209-171 crond[31696]: (root) CMD (/usr/share/astguiclient/AST_manager_kill_hung_congested.pl)
May 19 07:40:01 71-6-209-171 crond[32331]: (root) CMD (/usr/share/astguiclient/AST_vm_update.pl)
May 19 07:40:01 71-6-209-171 crond[32332]: (root) CMD (/usr/share/astguiclient/ADMIN_keepalive_ALL.pl)
May 19 07:41:01 71-6-209-171 crond[1137]: (root) CMD (/usr/share/astguiclient/AST_CRON_mix_recordings_BASIC.pl)
May 19 07:42:01 71-6-209-171 crond[2027]: (root) CMD (/usr/share/astguiclient/ADMIN_keepalive_ALL.pl)

Dan

[mflorell]

Can you post thes results of this command in the asterisk CLI?

core show channels concise


And at the same time please post the results of these MySQL queries:

SELECT * from live_channels;
SELECT * from live_sip_channels;

[capmicro]

Hi

I have the same problem.

I use asterisk 1.2.18 and vicidial 2.0.129.


i have the same results for screen -r and crontab -l

show channels concise returns nothing

mysql> SELECT * from live_channels;
Empty set (0.00 sec)

mysql> SELECT * from live_sip_channels;
Empty set (0.00 sec)

Thanks

[mflorell]

You need to post those results when you have live calls going on with live agents logged in.

[capmicro]

When i have a live calls :

show channels concise
SIP/2000-007d4e00!default!!1!Up!Bridged Call!Local/8600052@default-c3f9,1!12000!!3!!Local/8600052@default-c3f9,1
Local/8600052@default-c3f9,2!default!8600052!1!Up!MeetMe!8600052!0000000000!!3!7!(None)
Local/8600052@default-c3f9,1!default!12000!2!Up!Dial!sip/2000|20|rt!0000000000!!3!7!SIP/2000-007d4e00
Zap/pseudo-1930048706!default!s!1!Rsrvd!(None)!!!!3!!(None)
SIP/3000-007c2a10!default!8600052!1!Up!MeetMe!8600052!!!3!29!(None)


mysql> SELECT * from live_channels; +-----------------------+---------------+---------------+-----------+--------------+
| channel | server_ip | channel_group | extension | channel_data |
+-----------------------+---------------+---------------+-----------+--------------+
| Zap/pseudo-1930048706 | 192.168.1.218 | NULL | ring | SIP/ring |
+-----------------------+---------------+---------------+-----------+--------------+




mysql> SELECT * from live_sip_channels; +------------------------------+---------------+---------------+-----------------------+------------------------------+
| channel | server_ip | channel_group | extension | channel_data |
+------------------------------+---------------+---------------+-----------------------+------------------------------+
| SIP/3000-007c2a10 | 192.168.1.218 | NULL | 8600052 | 8600052 |
| Local/8600052@default-238b,2 | 192.168.1.218 | NULL | 8600052 | 8600052 |
| Local/8600052@default-238b,1 | 192.168.1.218 | NULL | 2000 | sip/2000|20|rt |
| SIP/2000-007d4e00 | 192.168.1.218 | NULL | Local/8600052@default | Local/8600052@default-238b,1 |
+------------------------------+---------------+---------------+-----------------------+------------------------------+


I have two phones : 2000 and 3000
3000 is logged in vicidial and i want to call 2000.
2000 ring, the call is ok but vicidial doesn't know that all is ok (status is wait for ringing)

Any ideas ?

Thanks

[mflorell]

Are you using the sip-silence versions of the 8365 extensions?

[capmicro]

Are you using the sip-silence versions of the 8365 extensions?

if you talk about this following extention as describe in the stratch install:

Code: Select all

#### VDAD SIP UNREGISTERED TRANSFER ENTRIES ####
#### Use these entries IN PLACE OF the entries above if you are using SIP trunks
#### and are not registering your provider in sip.conf
; VICIDIAL_auto_dialer transfer script:
exten => 8365,1,Playback(sip-silence)
exten => 8365,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8365,3,AGI(agi-VDADtransfer.agi,${EXTEN})
exten => 8365,4,AGI(agi-VDADtransfer.agi,${EXTEN})
exten => 8365,5,Hangup


No i don't use it.

is it important????

[capmicro]

After some research in the code, it looks that the probleme is the callerid.
In "vicidial.php" there is a call to "vdc_db_query.php". In that page there is a request like "SELECT ..... from vicidial_manager WHERE status='UPDATED' AND caller_id=$MDnextCID ...." .

This request returns "NO" (no rows).
In the table i have a row with status="UPDATED" but another caller_id.
And a row with the good caller_id but "SEND" in status.

[capmicro]

I think i find the probleme. The caller_id is <unknown>.
So, when the request "UPDATE vicidial_manager set status='UPDATED' ... WHERE caller_id='' ...... " is called in AST_manager_listen.pl, nothing is updated

Any idea about this pb?

[mflorell]

CallerIDname is what is important, you cannot alter the callerIDname in your dialplan for calls being placed from VICIDIAL.

[mflorell]

Are you using the sip-silence versions of the 8365 extensions?

if you talk about this following extention as describe in the stratch install:

Code: Select all

#### VDAD SIP UNREGISTERED TRANSFER ENTRIES ####
#### Use these entries IN PLACE OF the entries above if you are using SIP trunks
#### and are not registering your provider in sip.conf
; VICIDIAL_auto_dialer transfer script:
exten => 8365,1,Playback(sip-silence)
exten => 8365,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8365,3,AGI(agi-VDADtransfer.agi,${EXTEN})
exten => 8365,4,AGI(agi-VDADtransfer.agi,${EXTEN})
exten => 8365,5,Hangup


No i don't use it.

is it important????

It can be for some SIP and IAX connections depending on the provider.

Try it and let us know.

[capmicro]

CallerIDname is what is important, you cannot alter the callerIDname in your dialplan for calls being placed from VICIDIAL.

i don't alter the callerIDName in my dialplan.

[capmicro]

if I run durring a call in vicidial:
/usr/share/astguiclient/AST_manager_listen.pl
i have this result :

Code: Select all

....
|UPDATE vicidial_manager set status='UPDATED', channel='SIP/4000-007e6b50', uniqueid = '1179997570.335' where server_ip = '192.168.1.218' and callerid = '<unknown>'|
|0E0 RINGINGs updated|
input buffer: 326     lines: 1     partial: 0
|Event: Newstate
Privilege: call,all
Channel: SIP/4000-007e6b50
State: Up
CallerID: 14000
CallerIDName: <unknown>
Uniqueid: 1179997570.335
......


the probleme is the CallerIDName is not correct and the script can't update the BD

[mflorell]

Are you using the 'o' Dial flag when you dial out?

[capmicro]

no i don't

[capmicro]

i have tried withe the 'o' dial flag and now it's ok. When i call, vicidial show the call and the image "live call" appears.

Thanks a lot for this.

But when the called phone hangs up, vicidial doesn't know that the call is finished.

[capmicro]

for the futur users of vicidial can you put more explicitly in the scratch install that the 'o' dial flag is needed.

if we use an existant dialplan we d'ont care about this. And it can be the cause of big headhaches!!

[mflorell]

It's in the examples in the SCRATCH INSTALL doc and it is mentioned explicitly in the REQUIREMENTS doc and web page.

I have added more notes to the SCRATCH_INSTALL doc and extensions.conf.sample file in SVN.