Hi,
I am running the following config:
Centos 5.2
Asterisk 1.4.22
Vicidial 2.0.4.1rc5
During the installation I loaded the file performance_test_leads.txt which contains all bogus numbers. I started using this list for testing and realized it does not have real number. I created my own list and included in the campaign. I deleted the old list. But the hopper is still getting the number from the old deleted list. The main question is how does the campaign figure out the the list it is currently using has been deleted and it should move on to the next list active in the campaign ????
The questions I have is:
1) How to purge the hopper and restart it with a new list ?
2) How stop and restart the campaign with new list ?
3) How to restart VICIDIAL as a last resort?
I have used the scratch_install steps and installed the vicidial.
And downloaded and followed the administrator free manual so far.
If this thing flies and goes to production, I assure you folks that I will buy the full manual.
Right now any help is greatly appreciated.
Regards,
Irfan
Stop and Restart VICIDIAL or CAMPAIGN or HOPPER
Moderators: gerski, enjay, williamconley, Op3r, Staydog, gardo, mflorell, MJCoate, mcargile, Kumba, Michael_N
27 posts
• Page 1 of 1
Stop and Restart VICIDIAL or CAMPAIGN or HOPPER
by irfanlateef » Wed Jan 14, 2009 4:20 pm
- irfanlateef
- Posts: 26
- Joined: Wed Jan 14, 2009 3:59 pm
- Location: New Jersey,USA
by irfanlateef » Thu Jan 15, 2009 11:16 am
Hi,
I think I did try that before but to be safe I tried again.
It does not have any effect.
I did set force reset hopper to Y and then SUBMIT
I went back and logged in an as agent and it still continues from the old
deleted list.
Is there something else I have to do to purge the old list?
BTW how do I stop a campaign.
My application is very simple. When it snows, I want to record a message
"The school is closed today" and broadcast out to 100 phone numbers.
So I dont want any campaings running. I want to start a campaign send the messages and stop the campaign until next time.
Appreciate if you can suggest what is the best way to do this. Which model should I use.
Thanks,
Irfan
I think I did try that before but to be safe I tried again.
It does not have any effect.
I did set force reset hopper to Y and then SUBMIT
I went back and logged in an as agent and it still continues from the old
deleted list.
Is there something else I have to do to purge the old list?
BTW how do I stop a campaign.
My application is very simple. When it snows, I want to record a message
"The school is closed today" and broadcast out to 100 phone numbers.
So I dont want any campaings running. I want to start a campaign send the messages and stop the campaign until next time.
Appreciate if you can suggest what is the best way to do this. Which model should I use.
Thanks,
Irfan
- irfanlateef
- Posts: 26
- Joined: Wed Jan 14, 2009 3:59 pm
- Location: New Jersey,USA
by mflorell » Thu Jan 15, 2009 1:53 pm
To make sure it is off you should set the campaign to active=N and your remote agent to INACTIVE and the system will not dial.
As for your lists, I don't know why your leads would still be loading. You could always just "delete from vicidial_list;" in mysql to wipe everything out and then just reload your desired leads.
As for your lists, I don't know why your leads would still be loading. You could always just "delete from vicidial_list;" in mysql to wipe everything out and then just reload your desired leads.
- mflorell
- Site Admin
- Posts: 18406
- Joined: Wed Jun 07, 2006 2:45 pm
- Location: Florida
by mflorell » Thu Jan 15, 2009 1:58 pm
Just got a PM from the poster:
Since I am just a two day old baby, the list does not allow me post more than one message per day I guess. It is giving that long winded error
and read the list and your general response is, " Post the message to me
i will post it" . So here it does please post it.
I wanted to add a few details to the previous post. Rather a few answers
to my own questions for the benefits of others and few more new questions.
1. I could successfully purge the list from the campaing by doing the folowing is the order mentioned below.
a. First to stop a campaign , just set the ACTIVE button in the campaign to N and SUBMIT , that will stop the campaign in the hopper.
b. Go to the list and set the list ACTIVE to N and submit, then go to the bottom and delete the list itself.
c. Wait for some time and it will clear the hopper and their campaign are
removed from the hopper and the lists are removed the campaign.
d. Then I created a small list of four phone numbers and loaded a new list.
added it to the campaign and started the campaign by setting it to
active. I am using the manual dial method to test until things start rolling.
When I clicked the dial next number button with preview selected
I see the correct number but I dont see anything happening in the
Asterisk console window. The hopper screen is just scrolling with the
following messages.
2009-01-15 13:15:31 CAMPAIGNS TO PROCESSES ADAPT FOR: 1|0 IT: 695174
TESTCAMP hopper READY count: 3 CAMPAIGN DIFFERENTIAL: 1 0 (0 - 0)
2009-01-15 13:15:32 CAMPAIGNS TO PROCESSES ADAPT FOR: 1|0 IT: 695175
TESTCAMP hopper READY count: 3 CAMPAIGN DIFFERENTIAL: 1 0 (0 - 0)
2009-01-15 13:15:33 CAMPAIGNS TO PROCESSES ADAPT FOR: 1|0 IT: 695176
TESTCAMP hopper READY count: 3 CAMPAIGN DIFFERENTIAL: 1 0 (0 - 0)
Appreciate your help !!
Since I am just a two day old baby, the list does not allow me post more than one message per day I guess. It is giving that long winded error
and read the list and your general response is, " Post the message to me
i will post it" . So here it does please post it.
I wanted to add a few details to the previous post. Rather a few answers
to my own questions for the benefits of others and few more new questions.
1. I could successfully purge the list from the campaing by doing the folowing is the order mentioned below.
a. First to stop a campaign , just set the ACTIVE button in the campaign to N and SUBMIT , that will stop the campaign in the hopper.
b. Go to the list and set the list ACTIVE to N and submit, then go to the bottom and delete the list itself.
c. Wait for some time and it will clear the hopper and their campaign are
removed from the hopper and the lists are removed the campaign.
d. Then I created a small list of four phone numbers and loaded a new list.
added it to the campaign and started the campaign by setting it to
active. I am using the manual dial method to test until things start rolling.
When I clicked the dial next number button with preview selected
I see the correct number but I dont see anything happening in the
Asterisk console window. The hopper screen is just scrolling with the
following messages.
2009-01-15 13:15:31 CAMPAIGNS TO PROCESSES ADAPT FOR: 1|0 IT: 695174
TESTCAMP hopper READY count: 3 CAMPAIGN DIFFERENTIAL: 1 0 (0 - 0)
2009-01-15 13:15:32 CAMPAIGNS TO PROCESSES ADAPT FOR: 1|0 IT: 695175
TESTCAMP hopper READY count: 3 CAMPAIGN DIFFERENTIAL: 1 0 (0 - 0)
2009-01-15 13:15:33 CAMPAIGNS TO PROCESSES ADAPT FOR: 1|0 IT: 695176
TESTCAMP hopper READY count: 3 CAMPAIGN DIFFERENTIAL: 1 0 (0 - 0)
Appreciate your help !!
- mflorell
- Site Admin
- Posts: 18406
- Joined: Wed Jun 07, 2006 2:45 pm
- Location: Florida
by irfanlateef » Thu Jan 15, 2009 2:14 pm
Here is the output of the screen:
[root@cbusivr extras]# screen -ls
There are screens on:
24914.ASTVDadapt (Attached)
6427.ASTVDauto (Detached)
2 Sockets in /var/run/screen/S-root.
[root@cbusivr extras]#
[root@cbusivr extras]# screen -ls
There are screens on:
24914.ASTVDadapt (Attached)
6427.ASTVDauto (Detached)
2 Sockets in /var/run/screen/S-root.
[root@cbusivr extras]#
- irfanlateef
- Posts: 26
- Joined: Wed Jan 14, 2009 3:59 pm
- Location: New Jersey,USA
by irfanlateef » Thu Jan 15, 2009 2:26 pm
Adding few more details...
When I try to attach to the second screen it give the following error and goes away
[root@cbusivr astguiclient_2.0.4.1rc5]# screen -ls
There are screens on:
24914.ASTVDadapt (Attached)
7481.ASTVDauto (Detached)
2 Sockets in /var/run/screen/S-root.
[root@cbusivr astguiclient_2.0.4.1rc5]# screen -r 7481.ASTVDauto
There is no screen to be resumed matching 7481.ASTVDauto.
And it comes back after sometime with a new ID.
I read in the Wiki FAQ that,
>>> Make sure you have leads in the vicidial_list table,
>>> and that you have run the gmt_adjust script.
>>> Also, make sure that AST_DVhopper.pl is running and inserting
>>> records into the hopper. Make sure that AST_VDautodial is running.
I made sure the ntpd is running but could not find the gmt_adjust script in the package I downloaded. And also the AST_VDautodial is not running.
But I guess that could be because I chose manual dial method ???
Thanks,
Irfan
When I try to attach to the second screen it give the following error and goes away
[root@cbusivr astguiclient_2.0.4.1rc5]# screen -ls
There are screens on:
24914.ASTVDadapt (Attached)
7481.ASTVDauto (Detached)
2 Sockets in /var/run/screen/S-root.
[root@cbusivr astguiclient_2.0.4.1rc5]# screen -r 7481.ASTVDauto
There is no screen to be resumed matching 7481.ASTVDauto.
And it comes back after sometime with a new ID.
I read in the Wiki FAQ that,
>>> Make sure you have leads in the vicidial_list table,
>>> and that you have run the gmt_adjust script.
>>> Also, make sure that AST_DVhopper.pl is running and inserting
>>> records into the hopper. Make sure that AST_VDautodial is running.
I made sure the ntpd is running but could not find the gmt_adjust script in the package I downloaded. And also the AST_VDautodial is not running.
But I guess that could be because I chose manual dial method ???
Thanks,
Irfan
- irfanlateef
- Posts: 26
- Joined: Wed Jan 14, 2009 3:59 pm
- Location: New Jersey,USA
by irfanlateef » Thu Jan 15, 2009 5:35 pm
Hi,
Please see the conf files as below:
# astguiclient.conf - configuration elements for the astguiclient package
# this is the astguiclient configuration file
# all comments will be lost if you run install.pl again
# Paths used by astGUIclient
PATHhome => /usr/share/astguiclient
PATHlogs => /var/log/astguiclient
PATHagi => /var/lib/asterisk/agi-bin
PATHweb => /usr/local/apache2/htdocs
PATHsounds => /var/lib/asterisk/sounds
PATHmonitor => /var/spool/asterisk/monitor
PATHDONEmonitor => /var/spool/asterisk/monitorDONE
# The IP address of this machine
VARserver_ip => X.Y.9.42
# Database connection information
VARDB_server => localhost
VARDB_database => asterisk
VARDB_user => cron
VARDB_pass => 1234
VARDB_port => 3306
# Alpha-Numeric list of the astGUIclient processes to be kept running
# (value should be listing of characters with no spaces: 123456)
# X - NO KEEPALIVE PROCESSES (use only if you want none to be keepalive)
# 1 - AST_update
# 2 - AST_send_listen
# 3 - AST_VDauto_dial
# 4 - AST_VDremote_agents
# 5 - AST_VDadapt (If multi-server system, this must only be on one server)
# 6 - FastAGI_log
# 7 - AST_VDauto_dial_FILL (only for multi-server, this must only be on one server)
# 8 - ip_relay (used for blind agent monitoring)
VARactive_keepalives => 1234568
I am using FreePBX to configure the Asterisk so the manager.conf
has some general settings and the FreePBX related config has been
put in manager_additional.conf which is included in manager.conf
I am including both files:
MANAGER.CONF
;
; Asterisk Call Management support
;
[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0
[admin]
secret = amp111
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
#include manager_additional.conf
#include manager_custom.conf
MANAGER_ADDITIONAL.conf
[cbusapp]
secret = cbus123
deny=
permit=0.0.0.0/0.0.0.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
[cron]
secret = 1234
deny=
permit=
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
[listencron]
secret = 1234
deny=
permit=
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
[phoneglue]
secret = phoneglue
deny=
permit=0.0.0.0/0.0.0.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
[sendcron]
secret = 1234
deny=
permit=
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
[updatecron]
secret = 1234
deny=
permit=
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
I was trying more basic stuff. I read the agent manual and it says
as soon as the agent logs in the phone should ring.
I logged in by SIP softphone also did not ring.
I saw the people are posting debug logs from the screen logs.
Is there any way to increase the debug level to see what is going on
when I log into the system as an agent ??
Thanks for your time.
Irfan
Please see the conf files as below:
# astguiclient.conf - configuration elements for the astguiclient package
# this is the astguiclient configuration file
# all comments will be lost if you run install.pl again
# Paths used by astGUIclient
PATHhome => /usr/share/astguiclient
PATHlogs => /var/log/astguiclient
PATHagi => /var/lib/asterisk/agi-bin
PATHweb => /usr/local/apache2/htdocs
PATHsounds => /var/lib/asterisk/sounds
PATHmonitor => /var/spool/asterisk/monitor
PATHDONEmonitor => /var/spool/asterisk/monitorDONE
# The IP address of this machine
VARserver_ip => X.Y.9.42
# Database connection information
VARDB_server => localhost
VARDB_database => asterisk
VARDB_user => cron
VARDB_pass => 1234
VARDB_port => 3306
# Alpha-Numeric list of the astGUIclient processes to be kept running
# (value should be listing of characters with no spaces: 123456)
# X - NO KEEPALIVE PROCESSES (use only if you want none to be keepalive)
# 1 - AST_update
# 2 - AST_send_listen
# 3 - AST_VDauto_dial
# 4 - AST_VDremote_agents
# 5 - AST_VDadapt (If multi-server system, this must only be on one server)
# 6 - FastAGI_log
# 7 - AST_VDauto_dial_FILL (only for multi-server, this must only be on one server)
# 8 - ip_relay (used for blind agent monitoring)
VARactive_keepalives => 1234568
I am using FreePBX to configure the Asterisk so the manager.conf
has some general settings and the FreePBX related config has been
put in manager_additional.conf which is included in manager.conf
I am including both files:
MANAGER.CONF
;
; Asterisk Call Management support
;
[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0
[admin]
secret = amp111
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
#include manager_additional.conf
#include manager_custom.conf
MANAGER_ADDITIONAL.conf
[cbusapp]
secret = cbus123
deny=
permit=0.0.0.0/0.0.0.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
[cron]
secret = 1234
deny=
permit=
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
[listencron]
secret = 1234
deny=
permit=
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
[phoneglue]
secret = phoneglue
deny=
permit=0.0.0.0/0.0.0.0
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
[sendcron]
secret = 1234
deny=
permit=
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
[updatecron]
secret = 1234
deny=
permit=
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user
I was trying more basic stuff. I read the agent manual and it says
as soon as the agent logs in the phone should ring.
I logged in by SIP softphone also did not ring.
I saw the people are posting debug logs from the screen logs.
Is there any way to increase the debug level to see what is going on
when I log into the system as an agent ??
Thanks for your time.
Irfan
- irfanlateef
- Posts: 26
- Joined: Wed Jan 14, 2009 3:59 pm
- Location: New Jersey,USA
by irfanlateef » Fri Jan 16, 2009 9:37 am
Hi,
See output below, looks Net::Telnet is not installed I am installing it now
and I will send another output if it works correctly.
[root@cbusivr asterisk]# /usr/share/astguiclient/AST_update.pl --debug
Can't locate Net/Telnet.pm in @INC (@INC contains: /usr/lib/perl5/site_perl/5.8.8/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.7/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.6/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.5/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.8 /usr/lib/perl5/site_perl/5.8.7 /usr/lib/perl5/site_perl/5.8.6 /usr/lib/perl5/site_perl/5.8.5 /usr/lib/perl5/site_perl /usr/lib/perl5/vendor_perl/5.8.8/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.7/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.6/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.5/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.8 /usr/lib/perl5/vendor_perl/5.8.7 /usr/lib/perl5/vendor_perl/5.8.6 /usr/lib/perl5/vendor_perl/5.8.5 /usr/lib/perl5/vendor_perl /usr/lib/perl5/5.8.8/i386-linux-thread-multi /usr/lib/perl5/5.8.8 .) at /usr/share/astguiclient/AST_update.pl line 217.
BEGIN failed--compilation aborted at /usr/share/astguiclient/AST_update.pl line 217.
Thanks and Regards,
Irfan
See output below, looks Net::Telnet is not installed I am installing it now
and I will send another output if it works correctly.
[root@cbusivr asterisk]# /usr/share/astguiclient/AST_update.pl --debug
Can't locate Net/Telnet.pm in @INC (@INC contains: /usr/lib/perl5/site_perl/5.8.8/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.7/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.6/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.5/i386-linux-thread-multi /usr/lib/perl5/site_perl/5.8.8 /usr/lib/perl5/site_perl/5.8.7 /usr/lib/perl5/site_perl/5.8.6 /usr/lib/perl5/site_perl/5.8.5 /usr/lib/perl5/site_perl /usr/lib/perl5/vendor_perl/5.8.8/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.7/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.6/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.5/i386-linux-thread-multi /usr/lib/perl5/vendor_perl/5.8.8 /usr/lib/perl5/vendor_perl/5.8.7 /usr/lib/perl5/vendor_perl/5.8.6 /usr/lib/perl5/vendor_perl/5.8.5 /usr/lib/perl5/vendor_perl /usr/lib/perl5/5.8.8/i386-linux-thread-multi /usr/lib/perl5/5.8.8 .) at /usr/share/astguiclient/AST_update.pl line 217.
BEGIN failed--compilation aborted at /usr/share/astguiclient/AST_update.pl line 217.
Thanks and Regards,
Irfan
- irfanlateef
- Posts: 26
- Joined: Wed Jan 14, 2009 3:59 pm
- Location: New Jersey,USA
by irfanlateef » Fri Jan 16, 2009 9:56 am
Hi,
Please find the complete output after installing Net::Telnet() module
[root@cbusivr ~]# /usr/share/astguiclient/AST_update.pl --debug
SHOW CHANNELS format: 2
LOOKING FOR Zap clients assigned to this server:
|SELECT extension FROM phones where protocol = 'Zap' and server_ip='X.Y1.9.42'|
LOOKING FOR IAX2 clients assigned to this server:
|SELECT extension FROM phones where protocol = 'IAX2' and server_ip='X.Y.9.42'|
cc300
cc301
cc302
cc303
cc304
cc305
cc306
cc307
cc308
cc309
cc310
cc311
cc312
cc313
cc314
cc315
cc316
cc317
cc318
cc319
cc320
cc321
cc322
cc323
cc324
cc325
cc326
cc327
cc328
cc329
cc330
cc331
cc332
cc333
cc334
cc335
cc336
cc337
cc338
cc339
cc340
cc341
cc342
cc343
cc344
cc345
cc346
cc347
cc348
cc349
cc350
LOOKING FOR SIP clients assigned to this server:
|SELECT extension FROM phones where protocol = 'SIP' and server_ip='X.Y.9.42'|
cc100
cc101
cc102
cc103
cc104
cc105
cc106
cc107
cc108
cc109
cc110
cc111
cc112
cc113
cc114
cc115
cc116
cc117
cc118
cc119
cc120
cc121
cc122
cc123
cc124
cc125
cc126
cc127
cc128
cc129
cc130
cc131
cc132
cc133
cc134
cc135
cc136
cc137
cc138
cc139
cc140
cc141
cc142
cc143
cc144
cc145
cc146
cc147
cc148
cc149
cc150
gs102
2005
gs102
Zap Clients: |
IAX2 Clients: |cc300|cc300@cc300|cc301|cc301@cc301|cc302|cc302@cc302|cc303|cc303@cc303|cc304|cc304@cc304|cc305|cc305@cc305|cc306|cc306@cc306|cc307|cc307@cc307|cc308|cc308@cc308|cc309|cc309@cc309|cc310|cc310@cc310|cc311|cc311@cc311|cc312|cc312@cc312|cc313|cc313@cc313|cc314|cc314@cc314|cc315|cc315@cc315|cc316|cc316@cc316|cc317|cc317@cc317|cc318|cc318@cc318|cc319|cc319@cc319|cc320|cc320@cc320|cc321|cc321@cc321|cc322|cc322@cc322|cc323|cc323@cc323|cc324|cc324@cc324|cc325|cc325@cc325|cc326|cc326@cc326|cc327|cc327@cc327|cc328|cc328@cc328|cc329|cc329@cc329|cc330|cc330@cc330|cc331|cc331@cc331|cc332|cc332@cc332|cc333|cc333@cc333|cc334|cc334@cc334|cc335|cc335@cc335|cc336|cc336@cc336|cc337|cc337@cc337|cc338|cc338@cc338|cc339|cc339@cc339|cc340|cc340@cc340|cc341|cc341@cc341|cc342|cc342@cc342|cc343|cc343@cc343|cc344|cc344@cc344|cc345|cc345@cc345|cc346|cc346@cc346|cc347|cc347@cc347|cc348|cc348@cc348|cc349|cc349@cc349|cc350|cc350@cc350|
SIP Clients: |cc100|cc100@cc100|cc101|cc101@cc101|cc102|cc102@cc102|cc103|cc103@cc103|cc104|cc104@cc104|cc105|cc105@cc105|cc106|cc106@cc106|cc107|cc107@cc107|cc108|cc108@cc108|cc109|cc109@cc109|cc110|cc110@cc110|cc111|cc111@cc111|cc112|cc112@cc112|cc113|cc113@cc113|cc114|cc114@cc114|cc115|cc115@cc115|cc116|cc116@cc116|cc117|cc117@cc117|cc118|cc118@cc118|cc119|cc119@cc119|cc120|cc120@cc120|cc121|cc121@cc121|cc122|cc122@cc122|cc123|cc123@cc123|cc124|cc124@cc124|cc125|cc125@cc125|cc126|cc126@cc126|cc127|cc127@cc127|cc128|cc128@cc128|cc129|cc129@cc129|cc130|cc130@cc130|cc131|cc131@cc131|cc132|cc132@cc132|cc133|cc133@cc133|cc134|cc134@cc134|cc135|cc135@cc135|cc136|cc136@cc136|cc137|cc137@cc137|cc138|cc138@cc138|cc139|cc139@cc139|cc140|cc140@cc140|cc141|cc141@cc141|cc142|cc142@cc142|cc143|cc143@cc143|cc144|cc144@cc144|cc145|cc145@cc145|cc146|cc146@cc146|cc147|cc147@cc147|cc148|cc148@cc148|cc149|cc149@cc149|cc150|cc150@cc150|gs102|gs102@gs102|2005|2005@2005|gs102|gs102@gs102|
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
|UPDATE server_updater set last_update='2009-01-16 9:40:26' where server_ip='X.Y.9.42'|
concise: -1
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
|UPDATE server_updater set last_update='2009-01-16 9:40:26' where server_ip='X.Y1.9.42'|
concise: -1
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
|UPDATE server_updater set last_update='2009-01-16 9:40:26' where server_ip='X.Y.9.42'|
concise: -1
|DELETE FROM live_sip_channels where server_ip='X.Y.9.42'|
|DELETE FROM live_channels where server_ip='X.Y.9.42'|
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
|UPDATE server_updater set last_update='2009-01-16 9:40:27' where server_ip='X.Y.9.42'|
concise: -1
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
|UPDATE server_updater set last_update='2009-01-16 9:40:27' where server_ip='X.Y.9.42'|
concise: -1
===== Keeps running indefinitely ================
Regards,
Irfan
Please find the complete output after installing Net::Telnet() module
[root@cbusivr ~]# /usr/share/astguiclient/AST_update.pl --debug
SHOW CHANNELS format: 2
LOOKING FOR Zap clients assigned to this server:
|SELECT extension FROM phones where protocol = 'Zap' and server_ip='X.Y1.9.42'|
LOOKING FOR IAX2 clients assigned to this server:
|SELECT extension FROM phones where protocol = 'IAX2' and server_ip='X.Y.9.42'|
cc300
cc301
cc302
cc303
cc304
cc305
cc306
cc307
cc308
cc309
cc310
cc311
cc312
cc313
cc314
cc315
cc316
cc317
cc318
cc319
cc320
cc321
cc322
cc323
cc324
cc325
cc326
cc327
cc328
cc329
cc330
cc331
cc332
cc333
cc334
cc335
cc336
cc337
cc338
cc339
cc340
cc341
cc342
cc343
cc344
cc345
cc346
cc347
cc348
cc349
cc350
LOOKING FOR SIP clients assigned to this server:
|SELECT extension FROM phones where protocol = 'SIP' and server_ip='X.Y.9.42'|
cc100
cc101
cc102
cc103
cc104
cc105
cc106
cc107
cc108
cc109
cc110
cc111
cc112
cc113
cc114
cc115
cc116
cc117
cc118
cc119
cc120
cc121
cc122
cc123
cc124
cc125
cc126
cc127
cc128
cc129
cc130
cc131
cc132
cc133
cc134
cc135
cc136
cc137
cc138
cc139
cc140
cc141
cc142
cc143
cc144
cc145
cc146
cc147
cc148
cc149
cc150
gs102
2005
gs102
Zap Clients: |
IAX2 Clients: |cc300|cc300@cc300|cc301|cc301@cc301|cc302|cc302@cc302|cc303|cc303@cc303|cc304|cc304@cc304|cc305|cc305@cc305|cc306|cc306@cc306|cc307|cc307@cc307|cc308|cc308@cc308|cc309|cc309@cc309|cc310|cc310@cc310|cc311|cc311@cc311|cc312|cc312@cc312|cc313|cc313@cc313|cc314|cc314@cc314|cc315|cc315@cc315|cc316|cc316@cc316|cc317|cc317@cc317|cc318|cc318@cc318|cc319|cc319@cc319|cc320|cc320@cc320|cc321|cc321@cc321|cc322|cc322@cc322|cc323|cc323@cc323|cc324|cc324@cc324|cc325|cc325@cc325|cc326|cc326@cc326|cc327|cc327@cc327|cc328|cc328@cc328|cc329|cc329@cc329|cc330|cc330@cc330|cc331|cc331@cc331|cc332|cc332@cc332|cc333|cc333@cc333|cc334|cc334@cc334|cc335|cc335@cc335|cc336|cc336@cc336|cc337|cc337@cc337|cc338|cc338@cc338|cc339|cc339@cc339|cc340|cc340@cc340|cc341|cc341@cc341|cc342|cc342@cc342|cc343|cc343@cc343|cc344|cc344@cc344|cc345|cc345@cc345|cc346|cc346@cc346|cc347|cc347@cc347|cc348|cc348@cc348|cc349|cc349@cc349|cc350|cc350@cc350|
SIP Clients: |cc100|cc100@cc100|cc101|cc101@cc101|cc102|cc102@cc102|cc103|cc103@cc103|cc104|cc104@cc104|cc105|cc105@cc105|cc106|cc106@cc106|cc107|cc107@cc107|cc108|cc108@cc108|cc109|cc109@cc109|cc110|cc110@cc110|cc111|cc111@cc111|cc112|cc112@cc112|cc113|cc113@cc113|cc114|cc114@cc114|cc115|cc115@cc115|cc116|cc116@cc116|cc117|cc117@cc117|cc118|cc118@cc118|cc119|cc119@cc119|cc120|cc120@cc120|cc121|cc121@cc121|cc122|cc122@cc122|cc123|cc123@cc123|cc124|cc124@cc124|cc125|cc125@cc125|cc126|cc126@cc126|cc127|cc127@cc127|cc128|cc128@cc128|cc129|cc129@cc129|cc130|cc130@cc130|cc131|cc131@cc131|cc132|cc132@cc132|cc133|cc133@cc133|cc134|cc134@cc134|cc135|cc135@cc135|cc136|cc136@cc136|cc137|cc137@cc137|cc138|cc138@cc138|cc139|cc139@cc139|cc140|cc140@cc140|cc141|cc141@cc141|cc142|cc142@cc142|cc143|cc143@cc143|cc144|cc144@cc144|cc145|cc145@cc145|cc146|cc146@cc146|cc147|cc147@cc147|cc148|cc148@cc148|cc149|cc149@cc149|cc150|cc150@cc150|gs102|gs102@gs102|2005|2005@2005|gs102|gs102@gs102|
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
|UPDATE server_updater set last_update='2009-01-16 9:40:26' where server_ip='X.Y.9.42'|
concise: -1
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
|UPDATE server_updater set last_update='2009-01-16 9:40:26' where server_ip='X.Y1.9.42'|
concise: -1
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
|UPDATE server_updater set last_update='2009-01-16 9:40:26' where server_ip='X.Y.9.42'|
concise: -1
|DELETE FROM live_sip_channels where server_ip='X.Y.9.42'|
|DELETE FROM live_channels where server_ip='X.Y.9.42'|
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
|UPDATE server_updater set last_update='2009-01-16 9:40:27' where server_ip='X.Y.9.42'|
concise: -1
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
|UPDATE server_updater set last_update='2009-01-16 9:40:27' where server_ip='X.Y.9.42'|
concise: -1
===== Keeps running indefinitely ================
Regards,
Irfan
- irfanlateef
- Posts: 26
- Joined: Wed Jan 14, 2009 3:59 pm
- Location: New Jersey,USA
by mflorell » Fri Jan 16, 2009 11:25 am
I was actually trying to see what was missing since it wasn't running, I don't need to see the outout now that it's running properly.
What is the output of screen -ls now?
Are things running better now?
What is the output of screen -ls now?
Are things running better now?
- mflorell
- Site Admin
- Posts: 18406
- Joined: Wed Jun 07, 2006 2:45 pm
- Location: Florida
by mflorell » Fri Jan 16, 2009 11:27 am
More PM'd from the poster:
Hi,
After install the Net::Telnet module the things look much better
and I think I am close to getting it working.
(1) I first check the screen -ls and found lot more processes there.
[root@cbusivr ~]# screen -ls
There are screens on:
8016.ASTupdate (Detached)
8019.ASTsend (Detached)
8028.ASTVDremote (Detached)
8025.ASTVDauto (Detached)
7857.ASTVDadapt (Detached)
8022.ASTlisten (Detached)
6 Sockets in /var/run/screen/S-root.
(2) Encouraged by this I logged into the Agent login again. And this time
my softphone at extension 2005 rang but when I answered the phone the
system hung up on me with a "Goodbye" message. I am not sure but the
problem could be with my dial plan see Asterisk and VICIDIAL logs below.
The extension 2005 is in [from-internal] context and 8600051 is a meetme room
in the meetme context but it is looking in the default context.
So it looks like it is not able to find it.
Please see the logs below and provide some guidance.
Thanks and Regards,
Irfan
================================================
ASTERISK LOGS
================================================
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
> Channel SIP/2005-09f79590 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
== Starting SIP/2005-09f79590 at default,8600051,1 failed so falling back to exten 's'
-- Executing [s@default:1] Playback("SIP/2005-09f79590", "vm-goodbye") in new stack
-- <SIP/2005-09f79590> Playing 'vm-goodbye' (language 'en')
-- Executing [s@default:2] Macro("SIP/2005-09f79590", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/2005-09f79590", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/2005-09f79590", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/2005-09f79590", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/2005-09f79590", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/2005-09f79590", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/2005-09f79590", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/2005-09f79590'
Really destroying SIP dialog '7c9c164a2ec6126762910643186467e3@mt.att.com' Method: INVITE
Really destroying SIP dialog '4bf42ae858f9daf3194bd79a3bd8197f@mt.att.com' Method: OPTIONS
Really destroying SIP dialog '20e71eef148af4075e3a048d2c6b042f@mt.att.com' Method: NOTIFY
=================================================
VICIDIAL LOGS
=================================================
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
|UPDATE server_updater set last_update='2009-01-16 10:59:47' where server_ip='X.Y.9.42'|
concise: 11
lines: 2
DBchn: 0
DBsip: 0
-|0|Response: Follows|
+|0|Response: Follows SIP/ring|
-|1|Privilege: Command|
+|1|Privilege: Command SIP/ring|
-|2|SIP/2005-09f79590!from-internal!!1!Ringing!(None)!!!!3!0!(None)|
+|2|SIP/2005-09f79590 SIP/ring|
channel: |SIP/2005-09f79590|
extension: |ring|
QRYchannel:|SIP/2005-09f79590__ring|
current sips: 0
|INSERT INTO live_sip_channels (channel,server_ip,extension,channel_data) values('SIP/2005-09f79590','X.Y.9.42','ring','SIP/ring')|
DB 0|| |COUNT: 3|2|5769989
loop counter: |5769988|
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
SIP/2005-09f79590|ring
|UPDATE server_updater set last_update='2009-01-16 10:59:47' where server_ip='X.Y.9.42'|
concise: 11
lines: 2
DBchn: 0
DBsip: 0
-|0|Response: Follows|
+|0|Response: Follows SIP/ring|
-|1|Privilege: Command|
+|1|Privilege: Command SIP/ring|
-|2|SIP/2005-09f79590!from-internal!!1!Ringing!(None)!!!!3!0!(None)|
+|2|SIP/2005-09f79590 SIP/ring|
channel: |SIP/2005-09f79590|
extension: |ring|
QRYchannel:|SIP/2005-09f79590__ring|
current sips: 0
DB 0|| |COUNT: 3|2|5769988
loop counter: |5769987|
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
SIP/2005-09f79590|ring
|UPDATE server_updater set last_update='2009-01-16 10:59:48' where server_ip='X.Y.9.42'|
concise: 11
lines: 2
DBchn: 0
DBsip: 0
-|0|Response: Follows|
+|0|Response: Follows SIP/ring|
-|1|Privilege: Command|
+|1|Privilege: Command SIP/ring|
-|2|SIP/2005-09f79590!from-internal!!1!Ringing!(None)!!!!3!1!(None)|
+|2|SIP/2005-09f79590 SIP/ring|
channel: |SIP/2005-09f79590|
extension: |ring|
QRYchannel:|SIP/2005-09f79590__ring|
current sips: 0
DB 0|| |COUNT: 3|2|5769987
loop counter: |5769986|
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
SIP/2005-09f79590|ring
|UPDATE server_updater set last_update='2009-01-16 10:59:48' where server_ip='X.Y.9.42'|
concise: 11
lines: 2
DBchn: 0
DBsip: 0
-|0|Response: Follows|
+|0|Response: Follows SIP/ring|
-|1|Privilege: Command|
+|1|Privilege: Command SIP/ring|
-|2|SIP/2005-09f79590!from-internal!!1!Ringing!(None)!!!!3!1!(None)|
+|2|SIP/2005-09f79590 SIP/ring|
channel: |SIP/2005-09f79590|
extension: |ring|
QRYchannel:|SIP/2005-09f79590__ring|
current sips: 0
DB 0|| |COUNT: 3|2|5769986
loop counter: |5769985|
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
SIP/2005-09f79590|ring
|UPDATE server_updater set last_update='2009-01-16 10:59:49' where server_ip='X.Y.9.42'|
concise: 1
bad grab, trying again
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
SIP/2005-09f79590|ring
|UPDATE server_updater set last_update='2009-01-16 10:59:49' where server_ip='X.Y.9.42'|
concise: 11
lines: 2
DBchn: 0
DBsip: 0
-|0|Response: Follows|
+|0|Response: Follows SIP/ring|
-|1|Privilege: Command|
+|1|Privilege: Command SIP/ring|
-|2|SIP/2005-09f79590!default!s!1!Up!Playback!vm-goodbye!!!3!2!(None)|
+|2|SIP/2005-09f79590 vm-goodbye|
channel: |SIP/2005-09f79590|
extension: |vm|
QRYchannel:|SIP/2005-09f79590__vm|
current sips: 0
|INSERT INTO live_sip_channels (channel,server_ip,extension,channel_data) values('SIP/2005-09f79590','X.Y.9.42','vm','vm-goodbye')|
DB 0|SIP/2005-09f79590__ring| |COUNT: 3|2|5769985
|DELETE FROM live_sip_channels where server_ip='X.Y.9.42' and channel='SIP/2005-09f79590' and extension='ring' limit 1|
loop counter: |5769984|
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
SIP/2005-09f79590|vm
|UPDATE server_updater set last_update='2009-01-16 10:59:49' where server_ip='X.Y.9.42'|
concise: 11
lines: 2
DBchn: 0
DBsip: 0
-|0|Response: Follows|
+|0|Response: Follows SIP/ring|
-|1|Privilege: Command|
+|1|Privilege: Command SIP/ring|
-|2|SIP/2005-09f79590!default!s!1!Up!Playback!vm-goodbye!!!3!3!(None)|
+|2|SIP/2005-09f79590 vm-goodbye|
channel: |SIP/2005-09f79590|
extension: |vm|
QRYchannel:|SIP/2005-09f79590__vm|
current sips: 0
DB 0|| |COUNT: 3|2|5769984
loop counter: |5769983|
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
SIP/2005-09f79590|vm
|UPDATE server_updater set last_update='2009-01-16 10:59:50' where server_ip='X.Y.9.42'|
concise: 1
bad grab, trying again
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
SIP/2005-09f79590|vm
|UPDATE server_updater set last_update='2009-01-16 10:59:50' where server_ip='X.Y.9.42'|
concise: -1
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
SIP/2005-09f79590|vm
|UPDATE server_updater set last_update='2009-01-16 10:59:50' where server_ip='X.Y.9.42'|
concise: -1
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
SIP/2005-09f79590|vm
|UPDATE server_updater set last_update='2009-01-16 10:59:51' where server_ip='X.Y.9.42'|
concise: -1
|DELETE FROM live_sip_channels where server_ip='X.Y.9.42'|
|DELETE FROM live_channels where server_ip='X.Y.9.42'|
Hi,
After install the Net::Telnet module the things look much better
and I think I am close to getting it working.
(1) I first check the screen -ls and found lot more processes there.
[root@cbusivr ~]# screen -ls
There are screens on:
8016.ASTupdate (Detached)
8019.ASTsend (Detached)
8028.ASTVDremote (Detached)
8025.ASTVDauto (Detached)
7857.ASTVDadapt (Detached)
8022.ASTlisten (Detached)
6 Sockets in /var/run/screen/S-root.
(2) Encouraged by this I logged into the Agent login again. And this time
my softphone at extension 2005 rang but when I answered the phone the
system hung up on me with a "Goodbye" message. I am not sure but the
problem could be with my dial plan see Asterisk and VICIDIAL logs below.
The extension 2005 is in [from-internal] context and 8600051 is a meetme room
in the meetme context but it is looking in the default context.
So it looks like it is not able to find it.
Please see the logs below and provide some guidance.
Thanks and Regards,
Irfan
================================================
ASTERISK LOGS
================================================
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
> Channel SIP/2005-09f79590 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
== Starting SIP/2005-09f79590 at default,8600051,1 failed so falling back to exten 's'
-- Executing [s@default:1] Playback("SIP/2005-09f79590", "vm-goodbye") in new stack
-- <SIP/2005-09f79590> Playing 'vm-goodbye' (language 'en')
-- Executing [s@default:2] Macro("SIP/2005-09f79590", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/2005-09f79590", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/2005-09f79590", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/2005-09f79590", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/2005-09f79590", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/2005-09f79590", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/2005-09f79590", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/2005-09f79590'
Really destroying SIP dialog '7c9c164a2ec6126762910643186467e3@mt.att.com' Method: INVITE
Really destroying SIP dialog '4bf42ae858f9daf3194bd79a3bd8197f@mt.att.com' Method: OPTIONS
Really destroying SIP dialog '20e71eef148af4075e3a048d2c6b042f@mt.att.com' Method: NOTIFY
=================================================
VICIDIAL LOGS
=================================================
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
|UPDATE server_updater set last_update='2009-01-16 10:59:47' where server_ip='X.Y.9.42'|
concise: 11
lines: 2
DBchn: 0
DBsip: 0
-|0|Response: Follows|
+|0|Response: Follows SIP/ring|
-|1|Privilege: Command|
+|1|Privilege: Command SIP/ring|
-|2|SIP/2005-09f79590!from-internal!!1!Ringing!(None)!!!!3!0!(None)|
+|2|SIP/2005-09f79590 SIP/ring|
channel: |SIP/2005-09f79590|
extension: |ring|
QRYchannel:|SIP/2005-09f79590__ring|
current sips: 0
|INSERT INTO live_sip_channels (channel,server_ip,extension,channel_data) values('SIP/2005-09f79590','X.Y.9.42','ring','SIP/ring')|
DB 0|| |COUNT: 3|2|5769989
loop counter: |5769988|
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
SIP/2005-09f79590|ring
|UPDATE server_updater set last_update='2009-01-16 10:59:47' where server_ip='X.Y.9.42'|
concise: 11
lines: 2
DBchn: 0
DBsip: 0
-|0|Response: Follows|
+|0|Response: Follows SIP/ring|
-|1|Privilege: Command|
+|1|Privilege: Command SIP/ring|
-|2|SIP/2005-09f79590!from-internal!!1!Ringing!(None)!!!!3!0!(None)|
+|2|SIP/2005-09f79590 SIP/ring|
channel: |SIP/2005-09f79590|
extension: |ring|
QRYchannel:|SIP/2005-09f79590__ring|
current sips: 0
DB 0|| |COUNT: 3|2|5769988
loop counter: |5769987|
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
SIP/2005-09f79590|ring
|UPDATE server_updater set last_update='2009-01-16 10:59:48' where server_ip='X.Y.9.42'|
concise: 11
lines: 2
DBchn: 0
DBsip: 0
-|0|Response: Follows|
+|0|Response: Follows SIP/ring|
-|1|Privilege: Command|
+|1|Privilege: Command SIP/ring|
-|2|SIP/2005-09f79590!from-internal!!1!Ringing!(None)!!!!3!1!(None)|
+|2|SIP/2005-09f79590 SIP/ring|
channel: |SIP/2005-09f79590|
extension: |ring|
QRYchannel:|SIP/2005-09f79590__ring|
current sips: 0
DB 0|| |COUNT: 3|2|5769987
loop counter: |5769986|
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
SIP/2005-09f79590|ring
|UPDATE server_updater set last_update='2009-01-16 10:59:48' where server_ip='X.Y.9.42'|
concise: 11
lines: 2
DBchn: 0
DBsip: 0
-|0|Response: Follows|
+|0|Response: Follows SIP/ring|
-|1|Privilege: Command|
+|1|Privilege: Command SIP/ring|
-|2|SIP/2005-09f79590!from-internal!!1!Ringing!(None)!!!!3!1!(None)|
+|2|SIP/2005-09f79590 SIP/ring|
channel: |SIP/2005-09f79590|
extension: |ring|
QRYchannel:|SIP/2005-09f79590__ring|
current sips: 0
DB 0|| |COUNT: 3|2|5769986
loop counter: |5769985|
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
SIP/2005-09f79590|ring
|UPDATE server_updater set last_update='2009-01-16 10:59:49' where server_ip='X.Y.9.42'|
concise: 1
bad grab, trying again
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
SIP/2005-09f79590|ring
|UPDATE server_updater set last_update='2009-01-16 10:59:49' where server_ip='X.Y.9.42'|
concise: 11
lines: 2
DBchn: 0
DBsip: 0
-|0|Response: Follows|
+|0|Response: Follows SIP/ring|
-|1|Privilege: Command|
+|1|Privilege: Command SIP/ring|
-|2|SIP/2005-09f79590!default!s!1!Up!Playback!vm-goodbye!!!3!2!(None)|
+|2|SIP/2005-09f79590 vm-goodbye|
channel: |SIP/2005-09f79590|
extension: |vm|
QRYchannel:|SIP/2005-09f79590__vm|
current sips: 0
|INSERT INTO live_sip_channels (channel,server_ip,extension,channel_data) values('SIP/2005-09f79590','X.Y.9.42','vm','vm-goodbye')|
DB 0|SIP/2005-09f79590__ring| |COUNT: 3|2|5769985
|DELETE FROM live_sip_channels where server_ip='X.Y.9.42' and channel='SIP/2005-09f79590' and extension='ring' limit 1|
loop counter: |5769984|
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
SIP/2005-09f79590|vm
|UPDATE server_updater set last_update='2009-01-16 10:59:49' where server_ip='X.Y.9.42'|
concise: 11
lines: 2
DBchn: 0
DBsip: 0
-|0|Response: Follows|
+|0|Response: Follows SIP/ring|
-|1|Privilege: Command|
+|1|Privilege: Command SIP/ring|
-|2|SIP/2005-09f79590!default!s!1!Up!Playback!vm-goodbye!!!3!3!(None)|
+|2|SIP/2005-09f79590 vm-goodbye|
channel: |SIP/2005-09f79590|
extension: |vm|
QRYchannel:|SIP/2005-09f79590__vm|
current sips: 0
DB 0|| |COUNT: 3|2|5769984
loop counter: |5769983|
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
SIP/2005-09f79590|vm
|UPDATE server_updater set last_update='2009-01-16 10:59:50' where server_ip='X.Y.9.42'|
concise: 1
bad grab, trying again
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
SIP/2005-09f79590|vm
|UPDATE server_updater set last_update='2009-01-16 10:59:50' where server_ip='X.Y.9.42'|
concise: -1
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
SIP/2005-09f79590|vm
|UPDATE server_updater set last_update='2009-01-16 10:59:50' where server_ip='X.Y.9.42'|
concise: -1
|SELECT channel,extension FROM live_channels where server_ip = 'X.Y.9.42'|
SIP/2005-09f79590|vm
|UPDATE server_updater set last_update='2009-01-16 10:59:51' where server_ip='X.Y.9.42'|
concise: -1
|DELETE FROM live_sip_channels where server_ip='X.Y.9.42'|
|DELETE FROM live_channels where server_ip='X.Y.9.42'|
- mflorell
- Site Admin
- Posts: 18406
- Joined: Wed Jun 07, 2006 2:45 pm
- Location: Florida
by irfanlateef » Fri Jan 16, 2009 11:30 am
The only other thing is VoiceGlue 0.8 for VXML support.
The only impact of that software on asterisk is it adds and
extension under its own phoneglue context.
-IL
The only impact of that software on asterisk is it adds and
extension under its own phoneglue context.
-IL
- irfanlateef
- Posts: 26
- Joined: Wed Jan 14, 2009 3:59 pm
- Location: New Jersey,USA
by irfanlateef » Fri Jan 16, 2009 11:31 am
Sorry if I did not mention before I do use FreePBX 2.5.1.0
to configure the Asterisk.
to configure the Asterisk.
- irfanlateef
- Posts: 26
- Joined: Wed Jan 14, 2009 3:59 pm
- Location: New Jersey,USA
by mflorell » Fri Jan 16, 2009 8:18 pm
FreePBX is very difficult to integrate with successfully, especially maintaining that integration as you make changes with the system. I would recommend having a separate server if possible or just not using FreePBX.
- mflorell
- Site Admin
- Posts: 18406
- Joined: Wed Jun 07, 2006 2:45 pm
- Location: Florida
by mflorell » Fri Jan 16, 2009 8:24 pm
More from the poster:
Matt,
Unable to post on the list. Can I ask you the favor to posting again.
In order to debug further, I made a call from the extension to the meetme room 8600 which works fine. See log below.
Then I logged into the VICIDIAL agent screen and it triggered the call to the same
8600 extension but it does the flow and calls are entirely different.
In the VICIDIAL it does not even seem to going to a meetme room.
Can you please compare the log and advise. Another question is,
Has anybody got it working with Asterisk/FreePBX/Vicidial working before ? I am getting especially concerned becuase after asking if
I was using FreePBX there is total silence on the list.
Appreciate your reponse.
Thanks,
Irfan
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.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
-- Executing [8600@from-internal:1] Macro("SIP/2005-09f79c40", "user-callerid|") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/2005-09f79c40", "AMPUSER=2005") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/2005-09f79c40", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/2005-09f79c40", "1|Set|REALCALLERIDNUM=2005") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/2005-09f79c40", "AMPUSER=2005") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/2005-09f79c40", "AMPUSERCIDNAME=Irfan-2005") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/2005-09f79c40", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/2005-09f79c40", "AMPUSERCID=2005") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/2005-09f79c40", "CALLERID(all)="Irfan-2005" <2005>") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/2005-09f79c40", "REALCALLERIDNUM=2005") in new stack
-- Executing [s@macro-user-callerid:10] ExecIf("SIP/2005-09f79c40", "0|Set|CHANNEL(language)=") in new stack
-- Executing [s@macro-user-callerid:11] GotoIf("SIP/2005-09f79c40", "0?continue") in new stack
-- Executing [s@macro-user-callerid:12] Set("SIP/2005-09f79c40", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:13] GotoIf("SIP/2005-09f79c40", "1?continue") in new stack
-- Goto (macro-user-callerid,s,20)
-- Executing [s@macro-user-callerid:20] NoOp("SIP/2005-09f79c40", "Using CallerID "Irfan-2005" <2005>") in new stack
-- Executing [8600@from-internal:2] Set("SIP/2005-09f79c40", "MEETME_ROOMNUM=8600") in new stack
-- Executing [8600@from-internal:3] GotoIf("SIP/2005-09f79c40", "0?USER") in new stack
-- Executing [8600@from-internal:4] Answer("SIP/2005-09f79c40", "") in new stack
-- Executing [8600@from-internal:5] Wait("SIP/2005-09f79c40", "1") in new stack
-- Executing [8600@from-internal:6] Set("SIP/2005-09f79c40", "MEETME_OPTS=") in new stack
-- Executing [8600@from-internal:7] Goto("SIP/2005-09f79c40", "STARTMEETME|1") in new stack
-- Goto (from-internal,STARTMEETME,1)
-- Executing [STARTMEETME@from-internal:1] MeetMe("SIP/2005-09f79c40", "8600||") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
== Parsing '/etc/asterisk/meetme_additional.conf': Found
-- Created MeetMe conference 1023 for conference '8600'
-- <SIP/2005-09f79c40> Playing 'conf-onlyperson' (language 'en')
-- Hungup 'DAHDI/pseudo-305149690'
== Spawn extension (from-internal, STARTMEETME, 1) exited non-zero on 'SIP/2005-09f79c40'
-- Executing [h@from-internal:1] Macro("SIP/2005-09f79c40", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/2005-09f79c40", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/2005-09f79c40", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/2005-09f79c40", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/2005-09f79c40", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/2005-09f79c40", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/2005-09f79c40", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/2005-09f79c40'
Really destroying SIP dialog '2A86D287B9514BB8995353B63440B1EA0x875bf69a' Method: BYE
Really destroying SIP dialog '14e3e9005016af7b7c395ca55855e6af@mt.att.com' Method: OPTIONS
Really destroying SIP dialog '70d713d6331fa7661953e0686dc3a951@mt.att.com' Method: OPTIONS
===============================
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
Really destroying SIP dialog '2914fa5f7d4c8104219ab724422249e9@mt.att.com' Method: OPTIONS
Really destroying SIP dialog '5c5f7b6a57cb371606569a284e35dda3@mt.att.com' Method: OPTIONS
> Channel SIP/2005-09f79c40 was answered.
== Starting SIP/2005-09f79c40 at default,8600,1 failed so falling back to exten 's'
-- Executing [s@default:1] Playback("SIP/2005-09f79c40", "vm-goodbye") in new stack
-- <SIP/2005-09f79c40> Playing 'vm-goodbye' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing [s@default:2] Macro("SIP/2005-09f79c40", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/2005-09f79c40", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/2005-09f79c40", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/2005-09f79c40", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/2005-09f79c40", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/2005-09f79c40", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/2005-09f79c40", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/2005-09f79c40'
Really destroying SIP dialog '41b18dc16e4121be2d00ecb601f5592b@mt.att.com' Method: INVITE
Matt,
Unable to post on the list. Can I ask you the favor to posting again.
In order to debug further, I made a call from the extension to the meetme room 8600 which works fine. See log below.
Then I logged into the VICIDIAL agent screen and it triggered the call to the same
8600 extension but it does the flow and calls are entirely different.
In the VICIDIAL it does not even seem to going to a meetme room.
Can you please compare the log and advise. Another question is,
Has anybody got it working with Asterisk/FreePBX/Vicidial working before ? I am getting especially concerned becuase after asking if
I was using FreePBX there is total silence on the list.
Appreciate your reponse.
Thanks,
Irfan
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.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
-- Executing [8600@from-internal:1] Macro("SIP/2005-09f79c40", "user-callerid|") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/2005-09f79c40", "AMPUSER=2005") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/2005-09f79c40", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/2005-09f79c40", "1|Set|REALCALLERIDNUM=2005") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/2005-09f79c40", "AMPUSER=2005") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/2005-09f79c40", "AMPUSERCIDNAME=Irfan-2005") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/2005-09f79c40", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/2005-09f79c40", "AMPUSERCID=2005") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/2005-09f79c40", "CALLERID(all)="Irfan-2005" <2005>") in new stack
-- Executing [s@macro-user-callerid:9] Set("SIP/2005-09f79c40", "REALCALLERIDNUM=2005") in new stack
-- Executing [s@macro-user-callerid:10] ExecIf("SIP/2005-09f79c40", "0|Set|CHANNEL(language)=") in new stack
-- Executing [s@macro-user-callerid:11] GotoIf("SIP/2005-09f79c40", "0?continue") in new stack
-- Executing [s@macro-user-callerid:12] Set("SIP/2005-09f79c40", "__TTL=64") in new stack
-- Executing [s@macro-user-callerid:13] GotoIf("SIP/2005-09f79c40", "1?continue") in new stack
-- Goto (macro-user-callerid,s,20)
-- Executing [s@macro-user-callerid:20] NoOp("SIP/2005-09f79c40", "Using CallerID "Irfan-2005" <2005>") in new stack
-- Executing [8600@from-internal:2] Set("SIP/2005-09f79c40", "MEETME_ROOMNUM=8600") in new stack
-- Executing [8600@from-internal:3] GotoIf("SIP/2005-09f79c40", "0?USER") in new stack
-- Executing [8600@from-internal:4] Answer("SIP/2005-09f79c40", "") in new stack
-- Executing [8600@from-internal:5] Wait("SIP/2005-09f79c40", "1") in new stack
-- Executing [8600@from-internal:6] Set("SIP/2005-09f79c40", "MEETME_OPTS=") in new stack
-- Executing [8600@from-internal:7] Goto("SIP/2005-09f79c40", "STARTMEETME|1") in new stack
-- Goto (from-internal,STARTMEETME,1)
-- Executing [STARTMEETME@from-internal:1] MeetMe("SIP/2005-09f79c40", "8600||") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
== Parsing '/etc/asterisk/meetme_additional.conf': Found
-- Created MeetMe conference 1023 for conference '8600'
-- <SIP/2005-09f79c40> Playing 'conf-onlyperson' (language 'en')
-- Hungup 'DAHDI/pseudo-305149690'
== Spawn extension (from-internal, STARTMEETME, 1) exited non-zero on 'SIP/2005-09f79c40'
-- Executing [h@from-internal:1] Macro("SIP/2005-09f79c40", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/2005-09f79c40", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/2005-09f79c40", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/2005-09f79c40", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/2005-09f79c40", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/2005-09f79c40", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/2005-09f79c40", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/2005-09f79c40'
Really destroying SIP dialog '2A86D287B9514BB8995353B63440B1EA0x875bf69a' Method: BYE
Really destroying SIP dialog '14e3e9005016af7b7c395ca55855e6af@mt.att.com' Method: OPTIONS
Really destroying SIP dialog '70d713d6331fa7661953e0686dc3a951@mt.att.com' Method: OPTIONS
===============================
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
Really destroying SIP dialog '2914fa5f7d4c8104219ab724422249e9@mt.att.com' Method: OPTIONS
Really destroying SIP dialog '5c5f7b6a57cb371606569a284e35dda3@mt.att.com' Method: OPTIONS
> Channel SIP/2005-09f79c40 was answered.
== Starting SIP/2005-09f79c40 at default,8600,1 failed so falling back to exten 's'
-- Executing [s@default:1] Playback("SIP/2005-09f79c40", "vm-goodbye") in new stack
-- <SIP/2005-09f79c40> Playing 'vm-goodbye' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing [s@default:2] Macro("SIP/2005-09f79c40", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/2005-09f79c40", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/2005-09f79c40", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/2005-09f79c40", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/2005-09f79c40", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/2005-09f79c40", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/2005-09f79c40", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/2005-09f79c40'
Really destroying SIP dialog '41b18dc16e4121be2d00ecb601f5592b@mt.att.com' Method: INVITE
- mflorell
- Site Admin
- Posts: 18406
- Joined: Wed Jun 07, 2006 2:45 pm
- Location: Florida
by mflorell » Fri Jan 16, 2009 8:25 pm
FreePBX creates a big mess of the dialplan which is why we don't recommend using it. Also, if you do get VICIDIAL to work with FreePBX you will have half of the concurrent call capacity that you have on the same hardware without FreePBX.
- mflorell
- Site Admin
- Posts: 18406
- Joined: Wed Jun 07, 2006 2:45 pm
- Location: Florida
by irfanlateef » Mon Jan 19, 2009 12:22 pm
Matt,
I dont have the choice of not using FreePBX because I am using
an already installed Asterisk+FreePBX box and we dont have big in-house
Asterisk expertise to live without FreePBX.
Before taking this route I did research on the net and found that people
have successfully installed and using VICIDIAL on A+F platform.
I understand your concern about capacity but we are not a commercial
call center. This is only for emergency notifications. And this is the beauty
of open source that, when it is not being used for profit, it can used for free with the help from the community for a
common good cause.
Now coming back to our problem which I think is dialplan related.
If you can look into the log where I made a call to the meetme room 8600
from my extension 2005 and suggest what and where we can tweak in
the VICIDIAL to get the same behavior than we may be able to make some progress and get this to work.
As soon as the agent extension answers the asterisk logs this line.
== Starting SIP/2005-09f79c40 at default,8600,1 failed so falling back to exten 's'
If you can explain the meaning of this line I can try to fix it.
Is it trying to find and extension and failing ??
Instead of the above line, if we can get the VICIDIAL to trigger the following
line it might work.
-- Executing [8600@from-internal:1] Macro("SIP/2005-09f79c40", "user-callerid|") in new stack
But I dont know VICIDIAL enough to believe this might work.
Appreciate you r comments...
Thanks and Regards,
Irfan Lateef
I dont have the choice of not using FreePBX because I am using
an already installed Asterisk+FreePBX box and we dont have big in-house
Asterisk expertise to live without FreePBX.
Before taking this route I did research on the net and found that people
have successfully installed and using VICIDIAL on A+F platform.
I understand your concern about capacity but we are not a commercial
call center. This is only for emergency notifications. And this is the beauty
of open source that, when it is not being used for profit, it can used for free with the help from the community for a
common good cause.
Now coming back to our problem which I think is dialplan related.
If you can look into the log where I made a call to the meetme room 8600
from my extension 2005 and suggest what and where we can tweak in
the VICIDIAL to get the same behavior than we may be able to make some progress and get this to work.
As soon as the agent extension answers the asterisk logs this line.
== Starting SIP/2005-09f79c40 at default,8600,1 failed so falling back to exten 's'
If you can explain the meaning of this line I can try to fix it.
Is it trying to find and extension and failing ??
Instead of the above line, if we can get the VICIDIAL to trigger the following
line it might work.
-- Executing [8600@from-internal:1] Macro("SIP/2005-09f79c40", "user-callerid|") in new stack
But I dont know VICIDIAL enough to believe this might work.
Appreciate you r comments...
Thanks and Regards,
Irfan Lateef
- irfanlateef
- Posts: 26
- Joined: Wed Jan 14, 2009 3:59 pm
- Location: New Jersey,USA
by williamconley » Mon Jan 19, 2009 9:16 pm
ifranlateef, you DO have the choice of not using FreePBX.
They may reside on the same box but they can ignore each other entirely while running. I install FreePBX on my systems for the sole purpose of allowing FreePBX to create SIP/IAX2 extensions and trunks (easily, even for an end-user), and then use those trunks and extensions in Vicidial without using the associated dialplan from FreePBX.
So NONE of the macros fire because that dialplan is not used.
Vicidial uses [default] context, which is almost completely ignored by FreePBX. All you need to do is keep them separate, and you can continue to use this box.
They may reside on the same box but they can ignore each other entirely while running. I install FreePBX on my systems for the sole purpose of allowing FreePBX to create SIP/IAX2 extensions and trunks (easily, even for an end-user), and then use those trunks and extensions in Vicidial without using the associated dialplan from FreePBX.
So NONE of the macros fire because that dialplan is not used.
Vicidial uses [default] context, which is almost completely ignored by FreePBX. All you need to do is keep them separate, and you can continue to use this box.
Vicidial Installation and Repair, plus Hosting and Colocation
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http://www.PoundTeam.com # 352-269-0000 # 888-883-8488 # +44(203) 769-2294
- williamconley
- Posts: 20427
- Joined: Wed Oct 31, 2007 4:17 pm
- Location: Bartow, FL (In the boondocks)
by irfanlateef » Tue Jan 20, 2009 10:45 am
willianconley,
Your post earlier in 2008 was my starting point where you declared that
it works and gave the sample scripts and the mechanism to change the
config files.
So I am following the model you suggested. May be to clarify we should
define the two models. The model you described should called
"Colocated Model".
Now you need to define what you understood and what it means
when you say "Integrated Model".
The only change I made on that was, instead of default I called
the context [vicidial] because I was not sure if asterisk or FreePBX
was already having anything tucked with default context.
Therefore my question earlier, requesting someone(read mflorell) to explain the meaning of default.
If VICIDIAL is sending a command to look in default then there should be
someplace where I can change that to vicidial.
If it is hardcoded then I will change it back to default and try to debug.
If that default not coming from VICIDIAL I need to further debug and
find what and where could be causing the problem.
So why are the macros be triggered in my implementation and how to
prevent it ???
Thanks for comments and appreciate your help.
Regards,
Irfan
Your post earlier in 2008 was my starting point where you declared that
it works and gave the sample scripts and the mechanism to change the
config files.
So I am following the model you suggested. May be to clarify we should
define the two models. The model you described should called
"Colocated Model".
Now you need to define what you understood and what it means
when you say "Integrated Model".
The only change I made on that was, instead of default I called
the context [vicidial] because I was not sure if asterisk or FreePBX
was already having anything tucked with default context.
Therefore my question earlier, requesting someone(read mflorell) to explain the meaning of default.
If VICIDIAL is sending a command to look in default then there should be
someplace where I can change that to vicidial.
If it is hardcoded then I will change it back to default and try to debug.
If that default not coming from VICIDIAL I need to further debug and
find what and where could be causing the problem.
So why are the macros be triggered in my implementation and how to
prevent it ???
Thanks for comments and appreciate your help.
Regards,
Irfan
- irfanlateef
- Posts: 26
- Joined: Wed Jan 14, 2009 3:59 pm
- Location: New Jersey,USA
by irfanlateef » Tue Jan 20, 2009 3:33 pm
There is little progress here and some confusion.
I am unable to use agent which I had configured. For some reason
the campaign does not show in the login, but I used the 6666
administrative login and I was able to make progress.
Instead of immediately saying good bye it says
"You are the only person in the conf" when I login and keeps the channel open, but when I dial an external number it hangs up and says
good bye. See log below of the Asterisk
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
> Channel SIP/2005-0a09d020 was answered.
-- Executing [8600051@default:1] MeetMe("SIP/2005-0a09d020", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
== Parsing '/etc/asterisk/meetme_additional.conf': Found
== Parsing '/etc/asterisk/meetme_custom.conf': Found
== Manager 'sendcron' logged off from 127.0.0.1
-- Created MeetMe conference 1023 for conference '8600051'
-- <SIP/2005-0a09d020> Playing 'conf-onlyperson' (language 'en')
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [8600051@default:1] MeetMe("Local/8600051@default-5e4a,2", "860 0051|F") in new stack
> Channel Local/8600051@default-5e4a,1 was answered.
== Starting Local/8600051@default-5e4a,1 at default,917324201919,1 failed so f alling back to exten 's'
-- Executing [s@default:1] Playback("Local/8600051@default-5e4a,1", "vm-good bye") in new stack
-- <Local/8600051@default-5e4a,1> Playing 'vm-goodbye' (language 'en')
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [78600051@default:1] MeetMe("Local/78600051@default-a6d7,2", "8 600051|Fq") in new stack
> Channel Local/78600051@default-a6d7,1 was answered.
-- Executing [8309@default:1] Answer("Local/78600051@default-a6d7,1", "") in new stack
-- Executing [8309@default:2] Monitor("Local/78600051@default-a6d7,1", "wav| ") in new stack
-- Executing [8309@default:3] Wait("Local/78600051@default-a6d7,1", "3600") in new stack
-- Executing [s@default:2] Macro("Local/8600051@default-5e4a,1", "hangupcall ") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("Local/8600051@default-5e4a,1", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("Local/8600051@default-5e4a,1", "" ) in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("Local/8600051@default-5e4a,1", " 1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("Local/8600051@default-5e4a,1", " 1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("Local/8600051@default-5e4a,1", " 1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("Local/8600051@default-5e4a,1", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'Local/8600051 @default-5e4a,1' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'Local/8600051 @default-5e4a,1'
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@def ault-5e4a,2'
-- Executing [h@default:1] DeadAGI("Local/8600051@default-5e4a,2", "agi://12 7.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new s tack
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
cbusivr*CLI>
I am unable to use agent which I had configured. For some reason
the campaign does not show in the login, but I used the 6666
administrative login and I was able to make progress.
Instead of immediately saying good bye it says
"You are the only person in the conf" when I login and keeps the channel open, but when I dial an external number it hangs up and says
good bye. See log below of the Asterisk
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
> Channel SIP/2005-0a09d020 was answered.
-- Executing [8600051@default:1] MeetMe("SIP/2005-0a09d020", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
== Parsing '/etc/asterisk/meetme_additional.conf': Found
== Parsing '/etc/asterisk/meetme_custom.conf': Found
== Manager 'sendcron' logged off from 127.0.0.1
-- Created MeetMe conference 1023 for conference '8600051'
-- <SIP/2005-0a09d020> Playing 'conf-onlyperson' (language 'en')
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [8600051@default:1] MeetMe("Local/8600051@default-5e4a,2", "860 0051|F") in new stack
> Channel Local/8600051@default-5e4a,1 was answered.
== Starting Local/8600051@default-5e4a,1 at default,917324201919,1 failed so f alling back to exten 's'
-- Executing [s@default:1] Playback("Local/8600051@default-5e4a,1", "vm-good bye") in new stack
-- <Local/8600051@default-5e4a,1> Playing 'vm-goodbye' (language 'en')
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing [78600051@default:1] MeetMe("Local/78600051@default-a6d7,2", "8 600051|Fq") in new stack
> Channel Local/78600051@default-a6d7,1 was answered.
-- Executing [8309@default:1] Answer("Local/78600051@default-a6d7,1", "") in new stack
-- Executing [8309@default:2] Monitor("Local/78600051@default-a6d7,1", "wav| ") in new stack
-- Executing [8309@default:3] Wait("Local/78600051@default-a6d7,1", "3600") in new stack
-- Executing [s@default:2] Macro("Local/8600051@default-5e4a,1", "hangupcall ") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("Local/8600051@default-5e4a,1", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("Local/8600051@default-5e4a,1", "" ) in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("Local/8600051@default-5e4a,1", " 1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("Local/8600051@default-5e4a,1", " 1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("Local/8600051@default-5e4a,1", " 1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("Local/8600051@default-5e4a,1", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'Local/8600051 @default-5e4a,1' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'Local/8600051 @default-5e4a,1'
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@def ault-5e4a,2'
-- Executing [h@default:1] DeadAGI("Local/8600051@default-5e4a,2", "agi://12 7.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new s tack
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
cbusivr*CLI>
- irfanlateef
- Posts: 26
- Joined: Wed Jan 14, 2009 3:59 pm
- Location: New Jersey,USA
by irfanlateef » Tue Jan 20, 2009 4:44 pm
Gentlemen,
I would like to bring this thread to a close and thank everybody
for helping me make this maiden journey of initiation into VICIDIAL world.
The answer to my question, where do you change the context
from default to something is the conclusion of this thread.
Namely when you define the phone there is a field called the
Exten Context : which set to default.
Change this to the context in which the extension in Asterisk has been
defined. In my case FreePBX by default assign them to a context
called [from-internal] .
When I changed the context to this everything started working.
Namely it calls my extension 2005 when I login and in the manual
mode when I click DIAL LEAD, it successfull dials the external
number.
I have two suggestions for VICIDIAL folks.
(1) This step which I described above can be added to the SCRATCH_INSTALL.txt . If this step would have been mentioned
I would have saved atleast one week of struggle.
(2) The SCRATCH_INSTALL has steps to load the performance leads
and the list of lead with dummy number and , fake list of extensions.
These three list totally throws the first-time user/installer into a confusing
state. Instead you should add steps to do the folowing
a. add a single user , and extension and phone
b. add a list with two good valid numbers
c. create a campaign
d. Add the list and user to the campaign
e. Client login and verify extension rign
f. Manual dial lead and verify external dialling is fine.
Let a add a third suggestion here
(3) At the end where you declare as installation complete and suggest
client login, you should a installation verificatin step and say
screen -ls should look like this otherwise something is mising in installation.
Somewhere along the away, I had to struggle with Net::Telnet module
missing. You should add this to the list of requirements.
After this minimal installation verification sep(As I am calling it now)
You can put the performance lead testing scripts etc ...
Hope my suggestions will be helpful.
My special thanks to Matt for his patience with me.
Thanks and Regards,
Irfan
I would like to bring this thread to a close and thank everybody
for helping me make this maiden journey of initiation into VICIDIAL world.
The answer to my question, where do you change the context
from default to something is the conclusion of this thread.
Namely when you define the phone there is a field called the
Exten Context : which set to default.
Change this to the context in which the extension in Asterisk has been
defined. In my case FreePBX by default assign them to a context
called [from-internal] .
When I changed the context to this everything started working.
Namely it calls my extension 2005 when I login and in the manual
mode when I click DIAL LEAD, it successfull dials the external
number.
I have two suggestions for VICIDIAL folks.
(1) This step which I described above can be added to the SCRATCH_INSTALL.txt . If this step would have been mentioned
I would have saved atleast one week of struggle.
(2) The SCRATCH_INSTALL has steps to load the performance leads
and the list of lead with dummy number and , fake list of extensions.
These three list totally throws the first-time user/installer into a confusing
state. Instead you should add steps to do the folowing
a. add a single user , and extension and phone
b. add a list with two good valid numbers
c. create a campaign
d. Add the list and user to the campaign
e. Client login and verify extension rign
f. Manual dial lead and verify external dialling is fine.
Let a add a third suggestion here
(3) At the end where you declare as installation complete and suggest
client login, you should a installation verificatin step and say
screen -ls should look like this otherwise something is mising in installation.
Somewhere along the away, I had to struggle with Net::Telnet module
missing. You should add this to the list of requirements.
After this minimal installation verification sep(As I am calling it now)
You can put the performance lead testing scripts etc ...
Hope my suggestions will be helpful.
My special thanks to Matt for his patience with me.
Thanks and Regards,
Irfan
- irfanlateef
- Posts: 26
- Joined: Wed Jan 14, 2009 3:59 pm
- Location: New Jersey,USA
by mflorell » Tue Jan 20, 2009 7:03 pm
Thank you very much for the post-solution response, we really like ot get these.
In response,
- SCRATCH_INSTALL is written from the point of view of a new install not using any other utilities like FreePBX. Adding a separate document specifically about freePBX would seem like a better idea in this case.
- Net::Telnet is mentioned specifically in the REQUIREMENTS.txt document and several other places in the documentation.
Thanks again and let us know how your VICIDIAL system is running after you have a few campaigns under your belt.
In response,
- SCRATCH_INSTALL is written from the point of view of a new install not using any other utilities like FreePBX. Adding a separate document specifically about freePBX would seem like a better idea in this case.
- Net::Telnet is mentioned specifically in the REQUIREMENTS.txt document and several other places in the documentation.
Thanks again and let us know how your VICIDIAL system is running after you have a few campaigns under your belt.
- mflorell
- Site Admin
- Posts: 18406
- Joined: Wed Jun 07, 2006 2:45 pm
- Location: Florida
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