Inbund calls

[SuPrSluG]

I can't seem to get it to work.
extensions.conf
exten => 8665551212,1,Ringing
exten => 8665551212,2,Wait(1)
exten => 8665551212,3,Answer
exten=>8662619713,4,AGI(agi-VDADcloser_inboundCID.agi,SalesLine-----8662619713-----Closer---------------999-----1)
exten => 8662619713,5,Hangup

when I call my number (real, not above) it immediately hangs up. the agent closer is logged in.

I followed the free Manager Manual example. Is the meetme or transfer not working. I am using an IAX2 channel.

-- Accepting AUTHENTICATED call from 66.225.202.80:
> requested format = ulaw,
> requested prefs = (gsm|speex|ilbc|adpcm|g726|g723|g729|ulaw|alaw),
> actual format = ulaw,
> host prefs = (ulaw|gsm),
> priority = mine
-- Executing Ringing("IAX2/66.225.202.80:4569-2", "") in new stack
-- Executing Wait("IAX2/66.225.202.80:4569-2", "1") in new stack
-- Executing Answer("IAX2/66.225.202.80:4569-2", "") in new stack
-- Executing AGI("IAX2/66.225.202.80:4569-2", "agi-VDADcloser_inboundCID.agi|SalesLine-----8665551212-----Closer----------999-----1") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDADcloser_inboundCID.agi
-- AGI Script agi-VDADcloser_inboundCID.agi completed, returning 0
-- Executing Hangup("IAX2/66.225.202.80:4569-2", "") in new stack
== Spawn extension (default, 8662619713, 5) exited non-zero on 'IAX2/66.225.202.80:4569-2'
-- Executing DeadAGI("IAX2/66.225.202.80:4569-2", "call_log.agi|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing DeadAGI("IAX2/66.225.202.80:4569-2", "VD_hangup.agi|PRI-----NODEBUG-----16---------------") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script VD_hangup.agi completed, returning 0
-- Hungup 'IAX2/66.225.202.80:4569-2'

[mflorell]

Asterisk version?
astGUIclient version?
Real Asterisk CLI output? (not asterisk -r output)

[SuPrSluG]

Asterisk version?
Asterisk 1.2.11

astGUIclient version?
2.01.b1

Real Asterisk CLI output? (not asterisk -r output)
screen -r
There are several suitable screens on:
4116.ASTupdate (Detached)
4110.ASTVDremote (Detached)
4103.ASTVDauto (Detached)
4098.ASTVDadapt (Detached)
4091.ASTsend (Detached)
4081.ASTlisten (Detached)
Type "screen [-d] -r [pid.]tty.host" to resume one of them.

I have the boot_asterisk script in rc.local
and under ps aux
root 3970 0.0 0.2 2864 1104 ? Ss 12:18 0:00 /usr/bin/SCREEN -d -m -S asterisk /usr/sbin/asterisk -vvvvvvvvvvvvvvvvvvvvvgc
root 3971 0.4 1.5 16824 7804 pts/0 Ssl+ 12:18 0:09 /usr/sbin/asterisk -vvvvvvvvvvvvvvvvvvvvvgc

why doesn't it show under screen -r ?

[mflorell]

I have no idea why it wouldn't be showing there. What Linux distro are you running?

You can also get the AGI output by going to the astguiclient/admin.php servers section and setting the AGI output to FILE and then just viewing the agi output log file in your logs directory.

[SuPrSluG]

ok. here's the output. Something about no VDAC
record found

2006-09-06 13:27:33|call_log.agi|AGI Environment Dump:
2006-09-06 13:27:33|call_log.agi| -- accountcode =
2006-09-06 13:27:33|call_log.agi| -- callerid = 7164080354
2006-09-06 13:27:33|call_log.agi| -- calleridname = 7164080354
2006-09-06 13:27:33|call_log.agi| -- callingani2 = 0
2006-09-06 13:27:33|call_log.agi| -- callingpres = 0
2006-09-06 13:27:33|call_log.agi| -- callingtns = 0
2006-09-06 13:27:33|call_log.agi| -- callington = 0
2006-09-06 13:27:33|call_log.agi| -- channel = IAX2/66.225.202.80:4569-3
2006-09-06 13:27:33|call_log.agi| -- context = default
2006-09-06 13:27:33|call_log.agi| -- dnid = unknown
2006-09-06 13:27:33|call_log.agi| -- enhanced = 0.0
2006-09-06 13:27:33|call_log.agi| -- extension = h
2006-09-06 13:27:33|call_log.agi| -- language = en
2006-09-06 13:27:33|call_log.agi| -- priority = 1
2006-09-06 13:27:33|call_log.agi| -- rdnis = unknown
2006-09-06 13:27:33|call_log.agi| -- request = call_log.agi
2006-09-06 13:27:33|call_log.agi| -- type = IAX2
2006-09-06 13:27:33|call_log.agi| -- uniqueid = 1157563651.2
2006-09-06 13:27:33|call_log.agi|AGI Variables: |1157563651.2|IAX2/66.225.202.80:4569-3|h|IAX2|7164080354|
2006-09-06 13:27:33|call_log.agi||CALL HUNG UP|
2006-09-06 13:27:33|call_log.agi||DELETE from live_inbound where uniqueid='1157563651.2' and server_ip='192.168.2.133'|
2006-09-06 13:27:33|call_log.agi|+++++ CALL LOG HUNGUP: |1157563651.2|IAX2/66.225.202.80:4569-3|h|2006-09-06 13:27:33|min: |
2006-09-06 13:27:33|VD_hangup.agi|Perl Environment Dump:
2006-09-06 13:27:33|VD_hangup.agi|0|PRI-----NODEBUG-----16---------------
2006-09-06 13:27:33|VD_hangup.agi|AGI Environment Dump:
2006-09-06 13:27:33|VD_hangup.agi| -- accountcode =
2006-09-06 13:27:33|VD_hangup.agi| -- callerid = 7164080354
2006-09-06 13:27:33|VD_hangup.agi| -- calleridname = 7164080354
2006-09-06 13:27:33|VD_hangup.agi| -- callingani2 = 0
2006-09-06 13:27:33|VD_hangup.agi| -- callingpres = 0
2006-09-06 13:27:33|VD_hangup.agi| -- callingtns = 0
2006-09-06 13:27:33|VD_hangup.agi| -- callington = 0
2006-09-06 13:27:33|VD_hangup.agi| -- channel = IAX2/66.225.202.80:4569-3
2006-09-06 13:27:33|VD_hangup.agi| -- context = default
2006-09-06 13:27:33|VD_hangup.agi| -- dnid = unknown
2006-09-06 13:27:33|VD_hangup.agi| -- enhanced = 0.0
2006-09-06 13:27:33|VD_hangup.agi| -- extension = h
2006-09-06 13:27:33|VD_hangup.agi| -- language = en
2006-09-06 13:27:33|VD_hangup.agi| -- priority = 2
2006-09-06 13:27:33|VD_hangup.agi| -- rdnis = unknown
2006-09-06 13:27:33|VD_hangup.agi| -- request = VD_hangup.agi
2006-09-06 13:27:33|VD_hangup.agi| -- type = IAX2
2006-09-06 13:27:33|VD_hangup.agi| -- uniqueid = 1157563651.2
2006-09-06 13:27:33|VD_hangup.agi|AGI Variables: |1157563651.2|IAX2/66.225.202.80:4569-3|h|IAX2|7164080354|
2006-09-06 13:27:33|VD_hangup.agi|DEBUG: NODEBUG
2006-09-06 13:27:33|VD_hangup.agi|VD_hangup : 7164080354 IAX2/66.225.202.80:4569-3 2 0
2006-09-06 13:27:33|VD_hangup.agi||SELECT lead_id,callerid FROM vicidial_auto_calls where uniqueid = '1157563651.2' limit 1;|
2006-09-06 13:27:33|VD_hangup.agi|VD hangup: no VDAC record found: 1157563651.2 7164080354

[mflorell]

You need to post the agi-VDADcloser_inboundCID.agi output from that file.

[SuPrSluG]

There is none. Although it runs and exits w/ 0 from
a asterisk -r output.

[SuPrSluG]

Is this my problem? How/where would I insert IAX2 into live channels?

mysql> select * FROM `live_channels`;
+----------------------+---------------+---------------+-----------+--------------+
| channel | server_ip | channel_group | extension | channel_data |
+----------------------+---------------+---------------+-----------+--------------+
| Zap/pseudo-964923550 | 192.168.2.133 | NULL | ring | SIP/ring |
+----------------------+---------------+---------------+-----------+--------------+

[mflorell]

There was a bug in this script that was fixed in SVN 2-X. Please use this version. Otherwise I will be releasing another beta later this week with these changes in it.

[SuPrSluG]

That was it!! I'm now recieving calls. You da MAN!!!
Seriously I can't thank you enough for all your help. And I thought Asterisk had a learning curve !?!
I'm not a programer but if I can help in any way please let me know. Help writing docs etc..

[mflorell]

I'm glad it's working for you now.

The best help we can have is more testing of the 2.0.1 beta code and the SVN 2-X code. We are getting closer to a release and I should have beta2 out on Friday with several bug fixes and a few little feature additions with some documentation fixes as well.

[SuPrSluG]

I'm using the svn trunk. When I try to set autodial level it is not taken.
from admin_changes_log.txt after trying to change the level.


MODIFY CAMPAIGN INFO|6666|192.168.2.137|UPDATE vicidial_campaigns set campaign_name='Closer and inbound campaign',active='Y',dial_status_a='NEW',dial_status_b='',dial_status_c='',dial_status_d='',dial_status_e='',lead_order='DOWN',allow_closers='Y',hopper_level='5', auto_dial_level='0', next_agent_call='oldest_call_finish', local_call_time='24hours', voicemail_ext='', dial_timeout='60', dial_prefix='9', campaign_cid='8662619713', campaign_vdad_exten='8365',

[mflorell]

Can you please post all of this line?

What is the Dial Method?

[SuPrSluG]

Ok I can set it manually in mysql and then can reset it back to 0 through the web interface, but when I try to set it back to 1 in the web interface it fails.

[mflorell]

Can you please post all of this line?

What is the Dial Method?

[SuPrSluG]

Dial method is manual.

Here's what in after I change auto dial level
admin_changes_log.txt

Thu, 07 Sep 2006 10:50:33 -0400|MODIFY CAMPAIGN INFO|6666|192.168.2.137|UPDATE vicidial_campaigns set campaign_name='Closer and inbound campaign',active='Y',dial_status_a='NEW',dial_status_b='',dial_status_c='',dial_status_d='',dial_status_e='',lead_order='DOWN',allow_closers='Y',hopper_level='5', auto_dial_level='0', next_agent_call='oldest_call_finish', local_call_time='24hours', voicemail_ext='', dial_timeout='60', dial_prefix='9', campaign_cid='8662619713', campaign_vdad_exten='8365', web_form_address='', park_ext='', park_file_name='', campaign_rec_exten='8309', campaign_recording='ONDEMAND', campaign_rec_filename='FULLDATE_CUSTPHONE', campaign_script='NEWTEST', get_call_launch='NONE', am_message_exten='', amd_send_to_vmx='N', xferconf_a_dtmf='',xferconf_a_number='', xferconf_b_dtmf='',xferconf_b_number='',lead_filter_id='NONE',alt_number_dialing='N',scheduled_callbacks='N',safe_harbor_message='N',drop_call_seconds='5',safe_harbor_exten='8307',wrapup_seconds='0',wrapup_message='Wrapup Call',closer_campaigns=' SalesLine -',use_internal_dnc='N',allcalls_delay='0',omit_phone_code='N',dial_method='MANUAL',available_only_ratio_tally='N',adaptive_dropped_percentage='3',adaptive_maximum_level='3.0',adaptive_latest_server_time='7',adaptive_intensity='0',adaptive_dl_diff_target='7' where campaign_id='CLOSER';|N|

[mflorell]

If you have read the help(little question mark next to the field):

Dial Method - This field is the way to define how dialing is to take place. If MANUAL then the auto_dial_level will be locked at 0 unless Dial Method is changed...

[SuPrSluG]

Sorry I didn't see that. I followed the Free Manager Manual and it never mentioned that. I set it to Ratio
I hope that is correct. It is now correctly setting the values.

PS. I also noticed that in /etc/astguiclient.conf
I had to change (not paying attention during install.pl) Then the recording mixer worked properly. Maybe this will save someone some frustration.
PATHmonitor => /var/spool/asterisk/
PATHmonitor => /var/spool/asterisk/monitor

[mflorell]

The free manual is still on 1.1.12-3 so it wouldn't have any of the predictive stuff or any changes that are in 2.0 in it.

Thanks for reminding me about the PATHmonitor variable. I keep meaning to change that. I just changed and committed it to SVN 2-X.