vicidial.org

Hello, I have been working on integrating vicidial and trixbox. Everything so far has been relatively simple. The problem that I have been having is dialing from a vicidial campaign. In the asterisk output Vicidial is unable to locate a trunk to dial out on.
> Channel SIP/102-c2f1 was answered.
-- Executing MeetMe("SIP/102-c2f1", "8600051") in new stack
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing Macro("Local/917274514974@from-internal-45b6,2", "dialout-trunk|1|17274514974|") in new stack
-- Executing GotoIf("Local/917274514974@from-internal-45b6,2", "1?3:2") in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("Local/917274514974@from-internal-45b6,2", "user-callerid") in new stack
-- Executing GotoIf("Local/917274514974@from-internal-45b6,2", "1?report") in new stack
-- Goto (macro-user-callerid,s,9)
-- Executing NoOp("Local/917274514974@from-internal-45b6,2", "Using CallerID "V0822132458000000001" <0000000000>") in new stack
-- Executing Macro("Local/917274514974@from-internal-45b6,2", "record-enable|0000000000|OUT") in new stack
-- Executing GotoIf("Local/917274514974@from-internal-45b6,2", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("Local/917274514974@from-internal-45b6,2", "recordingcheck|20060822-132459|1156267499.9") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060822-132459|1156267499.9: No AMPUSER db entry for 0000000000. Not recording
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("Local/917274514974@from-internal-45b6,2", "No recording needed") in new stack
-- Executing Macro("Local/917274514974@from-internal-45b6,2", "outbound-callerid|1") in new stack
-- Executing GotoIf("Local/917274514974@from-internal-45b6,2", "0?start") in new stack
-- Executing Set("Local/917274514974@from-internal-45b6,2", "REALCALLERIDNUM=0000000000") in new stack
-- Executing NoOp("Local/917274514974@from-internal-45b6,2", "REALCALLERIDNUM is 0000000000") in new stack
-- Executing Set("Local/917274514974@from-internal-45b6,2", "USEROUTCID=") in new stack
-- Executing Set("Local/917274514974@from-internal-45b6,2", "EMERGENCYCID=") in new stack
-- Executing Set("Local/917274514974@from-internal-45b6,2", "TRUNKOUTCID=") in new stack
-- Executing GotoIf("Local/917274514974@from-internal-45b6,2", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,11)
-- Executing GotoIf("Local/917274514974@from-internal-45b6,2", "1?usercid") in new stack
-- Goto (macro-outbound-callerid,s,13)
-- Executing GotoIf("Local/917274514974@from-internal-45b6,2", "1?report") in new stack
-- Goto (macro-outbound-callerid,s,15)
-- Executing NoOp("Local/917274514974@from-internal-45b6,2", "CallerID set to "V0822132458000000001" <0000000000>") in new stack
-- Executing Set("Local/917274514974@from-internal-45b6,2", "GROUP()=OUT_1") in new stack
-- Executing GotoIf("Local/917274514974@from-internal-45b6,2", "0?108") in new stack
-- Executing Set("Local/917274514974@from-internal-45b6,2", "DIAL_NUMBER=17274514974") in new stack
-- Executing Set("Local/917274514974@from-internal-45b6,2", "DIAL_TRUNK=1") in new stack
-- Executing AGI("Local/917274514974@from-internal-45b6,2", "fixlocalprefix") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Set("Local/917274514974@from-internal-45b6,2", "OUTNUM=17274514974") in new stack
-- Executing Set("Local/917274514974@from-internal-45b6,2", "custom=") in new stack
-- Executing GotoIf("Local/917274514974@from-internal-45b6,2", "0?16") in new stack
-- Executing Dial("Local/917274514974@from-internal-45b6,2", "/17274514974|120|r") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Dial("Local/917274514974@from-internal-45b6,2", "/17274514974|55|to") in new stack
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Set("Local/917274514974@from-internal-45b6,2", "pre_num=") in new stack
-- Executing Set("Local/917274514974@from-internal-45b6,2", "the_num=") in new stack
-- Executing Set("Local/917274514974@from-internal-45b6,2", "post_num=") in new stack
-- Executing GotoIf("Local/917274514974@from-internal-45b6,2", "0?20:21") in new stack
-- Goto (macro-dialout-trunk,s,21)
-- Executing Dial("Local/917274514974@from-internal-45b6,2", "|120|r") in new stack

Any help with the dialplan would be greatly appriciated.

how did you get to the point you are at now with the integration? would be nice to pick you brain...

as soon as i catch up to you, i can help troubleshoot. i am looking to do the same thing.

have you checked things like dialing a 9 and including outbound routes in the vicidial contexts?

Hello, espencer, I am currently working to document the steps I have done so far. The only real trick to the trixbox and vicidial integration appears to be in the TB dialplan. It is possable to work around the TB dialplan but I am trying to integrate TB and VICIDIAL. Any help would be greatly appriciated.

Kevin Kelley

i noticed that i had to get into CPAN and get net::mysql as well as a few other things that did not get installed with trixbox. i also chose to use a separate database for the vicidial stuff which meant editing all the dbconnect.php files as well as a few others.

the asterisk dialplan is where i feel most at home so if you have both parts working in seperate contexts, i can't imagine it would take much more to get the two working together.

still need to catch up to where you are before i encounter that step.

also not sure how vicidial is making calls -- is it using the manager interface?

try this one this works on me under [macro-dialout-trunk]

exten => s,13,GotoIf($["${custom}" = "AMP"]?16)
exten => s,14,AGI(call_log.agi,${EXTEN})
exten => s,15,Dial(${OUT_${ARG1}}/${OUTNUM},120,${TRUNK_OPTIONS}) ; Regular Trunk Dial
exten => s,16,Goto(s-${DIALSTATUS},1)

there is a new line there so i think the goto in the first line needs to be 17 and then renumbering should take place below this section as well as a change to line s,20 goto.

exten => s,13,GotoIf($["${custom}" = "AMP"]?17)
exten => s,14,AGI(call_log.agi,${EXTEN})
exten => s,15,Dial(${OUT_${ARG1}}/${OUTNUM},120,${TRUNK_OPTIONS}) ; Regular Trunk Dial
exten => s,16,Goto(s-${DIALSTATUS},1)

did the original poster try this and get success?

thanks, Marvin!

i posted my files and instructions for getting this working on trixbox here:

http://iptn.org/vicidial/index.html

some things still don't quite work (no live status of agents from the vicidial admin page but the reports are there and accurate) but i am pretty excited at what can be done with this setup. from my reading of his posts I think Matt Florell will disagree with this but having done the scratch install once, i think it saves a fair bit of work to do it on top of trixbox as the OS, Asterisk, and many of the programs are installed automatically. not as lean and mean as Matt's scratch install but if you are doing small scale operations it shouldn't matter as much.

thanks to Marvin for his pointers on this -- they are in several parts of the forum and helped me get past tricky spots.

thanks especially to Matt Florell and his vast amount of accomplishment on his elegant and powerful software!

please let me know what needs fixing/doesn't work. I am a bit of a newbie so have probably made many mistakes and overlooked stuff etc.

Thank you very much for posting your instructions!

Hopefully if we can get a set that everyone agrees on we will be able to include a README.trixbox doc with the 2.0 release.

As for the SCRATCH_INSTALL, yes it is a tad large at 50 printed pages and I am planning on creating a smaller ASTERISK_INSTALL doc that will start from compiling Asterisk with patches and reference example conf files instead of going over their contents in detail in the file. It should be significantly smaller and easier for intermediate to expert Asterisk users to follow.

great job you got there espencer

the scratch install document is large but it is a huge gift. whatever amount of work it takes the newbie to muddle through it, it was so much more work to write the scratch install. nobody has any right to complain about it!

sorry if you thought i might have been complaining about the scratch install. People like me, who use and enjoy your code, should be writing the install guides.

if there is anything i can do to help with this project, let me know. i am a true non-expert/layperson but i can probably write/organize documentation, test things or do something low-skilled. the instructions mentioned in the above post have copies for download of all the dialplan additions just as they are written in the scratch install. maybe the next step would be for me to write a script that does a bunch of (or all of) that stuff i did to get vicidial running on trixbox?

Thanks for the compliments!

The Scratch_Install was in need of an overhaul which is why I created the BASE_INSTALL doc in the 2.0.1 beta release. There are many things like MySQL manual queries that I had separated in 3 different sections and people often had problems where they did not read all 2700 lines of the SCRATCH_INSTALL and they left one out. BASE _INSTALL groups all of that stuff together and references new sample conf files that also put many disconnected steps from the SCRATCH_INSTALL doc together into a more easy to understand format.

As for help on the project, right now we need people to test the 2.0.1 beta release for bugs so that it can be released in the next couple weeks officially.
http://www.eflo.net/VICIDIALforum/viewtopic.php?t=289

i have done some more testing and have found that manual dialing works great but auto-dialing does not so i suppose my newbieness is now shining through and my instructions posted earlier for getting vicidial working on trixbox are not quite up to snuff. i must say, however, that even if i don't get autodialing to work, this is really useful and functional with the manual dialing.

in search of a solution, i have read through the forums (man, Matt is helpful and patient!) and besides noticing how many times i see somebody say "i followed the scratch install" but then post output from their CLI that is clearly from freepbx i have found mention of the "o" flag, creating a symbolic link to the real location of mysql.sock, running AST_update.pl and AST_VDauto_dial.pl in debug, and installing the CLI delimiter patch.

i am using the beta version (2.0) now (that was a lot of work to Matt did) and really like the astguiclient.conf and interactive install.pl. very nice. i made sure i installed DBD::mysql -- already had DBI installed from previous version. followed other instructions in upgrade guide.

i was pretty sure these issues were related to my dialplan so i put vicidial in its very own (default) context with the scratch install dialplan lines and sectioned it off from the rest of trixbox but everything works the same as before with manual dialing going well, auto not so well -- calls the endpoint but never connects -- can see it get to vdad_transfer.agi but then it hangs up -- never gets to the agent.

i get the same symptoms on another computer with the 1.12 version (also a trixbox/vicidial hybrid attempt).

the only thing i have not done is applied the CLI delimiter patch. i see what it does and i am using an IAX trunk to dial out. could that cause this last little issue?

i also noticed that in table vicidial_auto_calls that channel and uniqueid are both null for the entry that is there when i start a campaign.

the output for AST_VDauto_dial:

TIME DEBUG: |1|1| GMT: 13:02
2006-08-31 14:02:47|LOGGED INTO MYSQL SERVER ON 1 CONNECTION||
TIME DEBUG: -5|-4|1| GMT: 18:02
2006-08-31 14:02:47|LIVE AGENTS LOGGED IN: 0|
2006-08-31 14:02:47| : agents: 1 dial_level: 0|
2006-08-31 14:02:47| : Calls to place: 0 (0 - 0) |
2006-08-31 14:02:47|CAMPAIGN DIFFERENTIAL: 0 0 (0 - 0)|
2006-08-31 14:02:47| : CALLING|
TIME DEBUG: -5|-4|1| GMT: 18:02
2006-08-31 14:02:47|| lagged call vla agent PAUSED 0E0|20060831140217|20060831140237|20060831140247||
2006-08-31 14:02:47|| lagged call vac agent DELETED 0E0|2006-08-31 14:00:47||

loop counter: |5759999|
TIME DEBUG: -5|-4|1| GMT: 18:02
2006-08-31 14:02:50|LIVE AGENTS LOGGED IN: 0|
2006-08-31 14:02:50| : agents: 1 dial_level: 0|
2006-08-31 14:02:50| : Calls to place: 0 (0 - 0) |
2006-08-31 14:02:50|CAMPAIGN DIFFERENTIAL: 0 0 (0 - 0)|
2006-08-31 14:02:50| : CALLING|
TIME DEBUG: -5|-4|1| GMT: 18:02
2006-08-31 14:02:50|| lagged call vla agent PAUSED 0E0|20060831140220|20060831140240|20060831140250||
2006-08-31 14:02:50|| lagged call vac agent DELETED 0E0|2006-08-31 14:00:50||

and AST_update:

LOOKING FOR Zap clients assigned to this server:
|SELECT extension FROM phones where protocol = 'Zap' and server_ip='192.168.0.112'|
LOOKING FOR IAX2 clients assigned to this server:
|SELECT extension FROM phones where protocol = 'IAX2' and server_ip='192.168.0.112'|
LOOKING FOR SIP clients assigned to this server:
|SELECT extension FROM phones where protocol = 'SIP' and server_ip='192.168.0.112'|
6604
Zap Clients: |
IAX2 Clients: |
SIP Clients: |6604|6604@6604|

|SELECT channel,extension FROM live_channels where server_ip = '192.168.0.112'|

|UPDATE server_updater set last_update='2006-08-31 14:05:24' where server_ip='192.168.0.112'|
concise: -1

|SELECT channel,extension FROM live_channels where server_ip = '192.168.0.112'|

|UPDATE server_updater set last_update='2006-08-31 14:05:25' where server_ip='192.168.0.112'|
concise: -1

nmap -sT -p 1-5570 localhost:

Starting Nmap 4.11 ( http://www.insecure.org/nmap/ ) at 2006-08-31 14:11 MDT
Interesting ports
Not shown: 5554 closed ports
PORT STATE SERVICE
21/tcp open ftp
22/tcp open ssh
25/tcp open smtp
80/tcp open http
111/tcp open rpcbind
199/tcp open smux
631/tcp open ipp
800/tcp open mdbs_daemon
940/tcp open unknown
2000/tcp open callbook
3306/tcp open mysql
4445/tcp open unknown
4949/tcp open unknown
5038/tcp open unknown
5060/tcp open sip
5061/tcp open unknown

Nmap finished: 1 IP address (1 host up) scanned in 0.282 seconds
[root@ucb ~]#

Most likely a callerID issue. Trixbox changes callerID on all calls by default and you have to get it to not do that for VICIDIAL calls. VICIDIAL tracks it's calls by CallerIDName so they must remain unaltered. This is a very tricky one to go around and still preserve trixbox-functionality. I have just wiped out many trixbox functions by deleting the SetCallerID lines everywhere in all dialplans and that worked just fine for VICIDIAL, but not the most elegant solution.

The "o" Dial flag is to preserve callerID on outbound calls and must be present in the Dial string that is placing the outbound calls for VICIDIAL.

espencer wrote:i posted my files and instructions for getting this working on trixbox here:

http://iptn.org/vicidial/index.html

some things still don't quite work (no live status of agents from the vicidial admin page but the reports are there and accurate) but i am pretty excited at what can be done with this setup. from my reading of his posts I think Matt Florell will disagree with this but having done the scratch install once, i think it saves a fair bit of work to do it on top of trixbox as the OS, Asterisk, and many of the programs are installed automatically. not as lean and mean as Matt's scratch install but if you are doing small scale operations it shouldn't matter as much.

thanks to Marvin for his pointers on this -- they are in several parts of the forum and helped me get past tricky spots.

thanks especially to Matt Florell and his vast amount of accomplishment on his elegant and powerful software!

please let me know what needs fixing/doesn't work. I am a bit of a newbie so have probably made many mistakes and overlooked stuff etc.

Hi

I have fallowed ur procedure as it is. but when i dial from vicidial my softphone is not ringing and there is information in my asterisk -r that vicidial is trying to reach my extention. pls help me what could be the problem.

Thanks & Regards,
Shakeel.

Hi

Please Please Please Help me to resolve this problem.

i am writing a auto installation script for trix and vicidial and a very clear document with screen shots. Please help me to do this.
i am seriously working on this project i want to do some thing line qmailrocks. i have dedicated 3 servers only for this testing.
i need your help for this project which will be helpful for all.
i know we can do this.

My Campain is not in Auto mode. So when i press on Dial Next Number. my softphone is not responding. and there is no updates in my asterisk log.
if i press again for "Call Agent Again" it again tries to connect but no responce. after some time a popup says "Call time out Contact ur Admin"

if i dail any number directly from my softphone i can able to dial.


Steps what i have Done:

i have downloaded and installed iso image of trixbox latest version 2.0 beta.
1. yum -y update
2. installed perl modules.
perl -MCPAN -e shell
- install MD5
- install Digest::MD5
- install Digest::SHA1
- install readline
- install Bundle::CPAN
- reload cpan
- install DBI
- force install DBD::mysql
- install Net::Telnet
- install Time::HiRes
- install OLE::Storage_Lite
- install Spreadsheet::ParseExcel

- cd /usr/src/
- wget http://asterisk.gnuinter.net/files/aste ... .09.tar.gz
- gunzip asterisk-perl-0.09.tar.gz
- tar xvf asterisk-perl-0.09.tar
- cd asterisk-perl-0.09
- perl Makefile.PL
- make all
- make install

3. copied the 3 files from web to /etc/asterisk/ folder with the below names
vicidial_extensions.conf
vicidial_meetme.conf
vicidial_manager.conf

4. edited extensions_custom.conf
my extensions_custom.conf starting 5 lines are below

;;;;;;;;;;;extensions_custom.conf;;;;;;;;;;;;;;;;;;;;;
#include vicidial_extensions.conf
[custom-vicidial]
Include => default
#include extensions_trixbox.conf
#include extensions_hud.conf
;;;;;;;;;;;extensions_custom.conf;;;;;;;;;;;;;;;;;;;;;

5. modifed extensions.conf macro-dialout-trunk
my extension.conf macro-dailout-trunk
;;;;;;;; start line 257 ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
exten => s,13,GotoIf($["${custom}" = "AMP"]?17)
exten => s,14,AGI(call_log.agi,${EXTEN})
exten => s,15,Dial(${OUT_${ARG1}}/${OUTNUM},120,${TRUNK_OPTIONS}) ; Regular Trunk Dial
exten => s,16,Goto(s-${DIALSTATUS},1)

; This is a custom trunk. Substitute $OUTNUM$ with the actual number and rebuild the dialstring
; example trunks: "AMP:CAPI/XXXXXXXX:b$OUTNUM$,30,r", "AMP:OH323/$OUTNUM$@XX.XX.XX.XX:XXXX"
exten => s,17,Set(pre_num=${CUT(OUT_${ARG1},$,1)})
exten => s,18,Set(the_num=${CUT(OUT_${ARG1},$,2)}) ; this is where we expect to find string OUTNUM
exten => s,19,Set(post_num=${CUT(OUT_${ARG1},$,3)})
exten => s,20,GotoIf($["${the_num}" = "OUTNUM"]?21:22) ; if we didn't find "OUTNUM", then skip to Dial
exten => s,21,Set(the_num=${OUTNUM}) ; replace "OUTNUM" with the actual number to dial
exten => s,22,Dial(${pre_num:4}${the_num}${post_num},120,${TRUNK_OPTIONS})
exten => s,23,Goto(s-${DIALSTATUS},1)
;;;;;;;; end line 270 ;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

6. edited extension.conf
my extensions.conf end 5 lines are below

;;;;;;;;;;;extensions.conf;;;;;;;;;;;;;;;;;;;;;
;this is where parked calls go if they time-out. Should probably re-ring
;[default]
;include => ext-local
;exten => s,1,Playback(vm-goodbye)
;exten => s,2,Macro(hangupcall)
;;;;;;;;;;;extensions.conf;;;;;;;;;;;;;;;;;;;;;

7. i logged in to freepbx for the first time and i enabled the modules from tools ---> Module Admin

8. I enabled the "o" option in Freepbx ---> Setup ----> General Settings
Asterisk Dial command options: rTto
Asterisk Outbound Dial command options: rTto
saved the above settings.

9. i have copied the vicidail.txt file in to my root folder. and I changed all the IPs to 127.0.0.1

10. I created vicidial database
# mysqladmin -u root -p

mysql> create database vicidial;
mysql> use vicidial;
mysql> \. /root/vicidail.txt
mysql> GRANT SELECT,INSERT,UPDATE,DELETE on vicidial.* TO cron@'%' IDENTIFIED BY '1234';
mysql> GRANT SELECT,INSERT,UPDATE,DELETE on vicidial.* TO cron@localhost IDENTIFIED BY '1234';
mysql> GRANT SELECT on vicidial.phones TO idcheck@'%' IDENTIFIED BY '1234';
mysql> GRANT SELECT on vicidial.phones TO idcheck@localhost IDENTIFIED BY '1234';

i have executed vicidail.txt without any errors and all 47 tables got created.

11. i installed the below packages for astguiclient
# yum -y install screen
for webmin i have downloaded from webmin.com and installed the RPM.
# mkdir /usr/src/astguiclient
# cd /usr/src/astguiclient
# wget http://internap.dl.sourceforge.net/sour ... _2.0.1.zip
# unzip astguiclient_2.0.1.zip
# perl install.pl
i have falowed the installation procedure and i have given database name as "vicidial" and httpRoot as "/var/www/html"

12. i have updated the crontab with the below command and inserted the below values in to it.
# crontab -e

################Cron entry########################
### 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 root /usr/share/astguiclient/AST_CRON_mix_recordings_BASIC.pl
### keepalive script for Asterisk updaters
* * * * * root /usr/share/astguiclient/ADMIN_keepalive_AST_update.pl
### keepalive script for Asterisk manager queue system
* * * * * root /usr/share/astguiclient/ADMIN_keepalive_AST_send_listen.pl
### keepalive script for Asterisk VICIDIAL autodial system
* * * * * root /usr/share/astguiclient/ADMIN_keepalive_AST_VDautodial.pl
### keepalive script for VICIDIAL remote agents
* * * * * root /usr/share/astguiclient/ADMIN_keepalive_AST_VDremote_agents.pl
### kill Hangup script for Asterisk updaters
* * * * * root /usr/share/astguiclient/AST_manager_kill_hung_congested.pl
### updater for voicemail
* * * * * root /usr/share/astguiclient/AST_vm_update.pl
### updater for conference validator
* * * * * root /usr/share/astguiclient/AST_conf_update.pl
### flush queue DB table every hour for entries older than 1 hour
11 * * * * root /usr/share/astguiclient/AST_flush_DBqueue.pl -q
### fix the vicidial_agent_log once every hour
33 * * * * root /usr/share/astguiclient/AST_cleanup_agent_log.pl
### updater for VICIDIAL hopper
* * * * * root /usr/share/astguiclient/AST_VDhopper.pl -q
### adjust the GMT offset for the leads in the vicidial_list table
1 1 * * * root /usr/share/astguiclient/ADMIN_adjust_GMTnow_on_leads.pl --debug
### reset several temporary-info tables in the database
2 1 * * * root /usr/share/astguiclient/AST_reset_mysql_vars.pl
### optimize the database tables within the asterisk database
3 1 * * * root /usr/share/astguiclient/AST_DB_optimize.pl
## adjust time on the server with ntp
30 * * * * root /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 root /usr/share/astguiclient/AST_agent_week.pl
### keepalive script for VICIDIAL predictive script
* * * * * root /usr/share/astguiclient/ADMIN_keepalive_AST_VDadapt.pl
################Cron entry########################

13. i have created a SIP trunk in freepbx which will connect to my VOIP Service provider.
freepbs ---> Setup ----> Trunks ---> Add SIP Trunk

14. i have created a Outbound Route and selected the above created trunk in Trunk Sequence
freepbs ---> Setup ----> Outbound Routes
my Dial pattern is:
1NXXNXXXXXX
NXXNXXXXXX
NXXXXXX

15. now i creared a extension using freepbx ---> Setup ---> Extensions ---> SIP
SIP Extension: 786
context : default

16. i have not configured inbound calling.

17. by default there is 2 CAMPAIGN in the Database of vicidial.
CAMPAIGN LISTINGS:
21 testing Y NEW MODIFY
22 test3961 Y N DROP MODIFY

18. i have created a list with 101 ID, test01 as name and assigned to CAMPAIGN ID 21.
LIST LISTINGS:
101 test01 21 Y MODIFY

19. i logged in to the VICIDIAL and i created a user 786.

20. i have logged in to astguiclient/admin.php and i created a phone 786

Note: at this point i have a extension 786 and user 786 and phone 786

21. i have added the leads of scratch_install Document using mysql prompt and changed the phone numbers to my own numbers.
# mysql -u cron -p
mysql> use vicidial;
mysql>
insert into vicidial_list values('','2004-01-06','','NEW','','','TEST01','101','TESTCAMP','N','1','7275551212','Mr','Matt','X','lead01','1234 Fake St.','','','Clearwater','FL','','33760','USA','M','1970-01-01','','test@test.com','suprise','comments go here','0');
insert into vicidial_list values('','2004-01-06','','NEW','','','TEST01','101','TESTCAMP','N','1','7275551212','Mr','Matt','X','lead02','1234 Fake St.','','','Clearwater','FL','','33760','USA','M','1970-01-01','','test@test.com','suprise','comments go here','0');
insert into vicidial_list values('','2004-01-07','','NEW','','','TEST01','101','TESTCAMP','N','1','7275551212','Mr','Matt','X','lead03','1234 Fake St.','','','Clearwater','FL','','33760','USA','M','1970-01-01','','test@test.com','suprise','comments go here','0');
insert into vicidial_list values('','2004-01-07','','NEW','','','TEST01','101','TESTCAMP','N','1','7275551212','Mr','Matt','X','lead04','1234 Fake St.','','','Clearwater','FL','','33760','USA','M','1970-01-01','','test@test.com','suprise','comments go here','0');
insert into vicidial_list values('','2004-01-07','','NEW','','','TEST01','101','TESTCAMP','N','1','7275551212','Mr','Matt','X','lead05','1234 Fake St.','','','Clearwater','FL','','33760','USA','M','1970-01-01','','test@test.com','suprise','comments go here','0');
insert into vicidial_list values('','2004-01-07','','NEW','','','TEST01','101','TESTCAMP','N','1','7275551212','Mr','Matt','X','lead06','1234 Fake St.','','','Clearwater','FL','','33760','USA','M','1970-01-01','','test@test.com','suprise','comments go here','0');
insert into vicidial_list values('','2004-01-07','','NEW','','','TEST01','101','TESTCAMP','N','1','7275551212','Mr','Matt','X','lead07','1234 Fake St.','','','Clearwater','FL','','33760','USA','M','1970-01-01','','test@test.com','suprise','comments go here','0');

22. I edited /usr/sbin/amportal script and per the suggestion.
my /usr/sbin/amportal script from line 165 to 179
#################amportal script##################
# run_asterisk
/usr/share/astguiclient/start_asterisk_boot.pl
run_fop
;;
stop)
stop_asterisk
stop_fop
;;
restart)
stop_asterisk
stop_fop
sleep 1
chown_asterisk
# run_asterisk
/usr/share/astguiclient/start_asterisk_boot.pl
#################amportal script##################

23. i dont have any zaptel device. so i dint changed rc.local file.

24. i have rebooted the system and executed " asterisk -r "

25. according to "STEP 21" i have logged in to my SIP phone (X-light) with 786 extension. and asterisk log updateed me that 786 user is registered.

26. now logged in to http://<myip>/agc/vicidial.php

Up to this point everything is fine.

27. My Campain is not in Auto mode. So when i press on Dial Next Number. my softphone is not responding. and there is no updates in my asterisk log.
if i press again for "Call Agent Again" it again tries to connect but no responce. after some time a popup says "Call time out Contact ur Admin"

28. if i dail any number directly from my softphone i can able to dial.


Please Please Please Help me to resolve this problem.

i am writing a auto installation script for trix and vicidial and a very clear document with screen shots. Please help me to do this.
i am seriously working on this project i want to do some thing line qmailrocks. i have dedicated 3 servers only for this testing.
i need your help for this project which will be helpful for all.
i know we can do this.

******* Teach to Learn *********

Thanks & Regards,
SHAAN.
shaantheadmin@gmail.com

After we spent many weeks trying to write some consistent instructions for instaling VICIDIAL on Trixbox, I can tell you that unless you are willing to spend the time to retest and rewrite your instructions weekly that any instructions you write now will not work one month from now if you do a yum update on your trixbox installation.

The problem is that Trixbox is not maintained like a lot of other projects. There are random updates that are not documented that drastically change the default dialplans and dozens of macros and AGI script with no notice of changes and no effort to remain consistent.

Then there is the other issue of system capacity. Our testing showed that using VICIDIAL on a trixbox server cuts your dialing capacity in half unless you turn off many of the dialing paths that are default in Trixbox.

The bottom line is that once you get VICIDIAL working on Trixbox, you can never update it unless you want to go through the fun of trying to figure out what has changed. You also have to be very careful when editing the dialplan so that you don't remove entries that VICIDIAL needs to run.

Trixbox should not be used for a production VICIDIAL system and actually takes longer to setup on Trixbox than just installing Linux and VICIDIAL from scratch.

I never really understood why me and some people need to integrate a@h with VICIDIAL before. I managed to actually do it (with only 4 weeks of learning asterisk!) then yes it works and now what?

Trixbox and VICIDIAL isnt really compatible with each other. and yes you will encounter a lot of headaches maintaining it. I am not saying that let's all stop integrating VICIDIAL with trixbox but have you ever asked yourself whats the point of doing this?

Let's make an assumption now.

With trixbox in the server , yes some parts can be easily manageable. you can add extensions and add routing and trunks by point and click and after you hit red line on top. oh wait of course youre still going to use the cli and edit confs to make it work on vicidial.

without trixbox, open up your putty change your confs , go to asterisk cli type reload and its done.

simple stuff really can make an administrator's life easier.

Hi,

I really appreciate your suggestions. Yes ur right i agree with you, trix and vicidial is not really a good combination. please suggest me the best architecture for predictive dialer for a production environment. i have seen users struggling for a predictive dialer and standard VOIP Server configuration. Please suggest me the best combination for a predictive dialer.
i am free for more 2 weeks i want to do some thing for open source. i want to pour a small drop in the open source sea.



*******Teach to Lean******

Thanks & Regards,
SHAAN.
shaantheadmin@gmail.com

Hi,

i want to understand how extentions.conf sip.conf meetme.conf manager.conf works exactly. some times i am confused with the syntax.

could you please suggest me any manual or book for reference to understand the file syntax of asterisk.

Thanks & Regards
SHAAN.
shaantheadmin@gmail.com

try to read the Asterisk - the future of telephony and can be downloaded at the asterisk website.

also I suggest that you go to voip-info to learn some few tips and tricks

and the best site that I went to is asteriskguru.com - got some stuff there that taught me almost everything I need to know to modify and create some cool dial plans.

Shaan,

sorry I did not see your post earlier. I wrote the instructions you were following. I will read your log more carefully later today but in the meantime, I have found a few things behave differently when using different versions of Asterisk/Trixbox. I have not tried the beta version of trixbox but the latest stable version works the best of the bunch. If the agent is not getting called, here are a few things to test:

dial a meetme directly from an extension

check to see what type of extension you built in the astguiclient/admin.php part of Matt's cool software (external, sip ...) -- if it is external, the manager originated call will need to have a dialplan line in the vicidial context.

I have made some changes on my dialer here at home for high performance (then trixbox is only providing a GUI for managing extensions and trunks) i have changed my outbound dialing in vicidial to directly dial the trunks rather than use the Trixbox dialplan which can be very costly for running a real campaign. this is more in line with the way Matt has detailed things in his scratch install guide. You will want an entry in globals_custom.conf defining the trunk variable.

lastly, depending on the version of asterisk/trixbox, you will probably want to start asterisk in something other than the safemode it is started in by default by the amportal script. It looks like you did this. Also, you will need zaptel regardless of whether you have a zaptel device or not or meetme will not function. To see if you have this part nailed simply dial a meetme room directly (e.g. 8600100). it won't work if you don't have ztdummy up and running.

Hi espencer,

Thanks for kind reply,

Could you please suggest me the version match for this setup. and if u can can attach ur /etc/asterisk dir to me it will be really helpful to understand it.

i stucked up in basic installation only...i dont know when i will write Auto instalation for Vicidial.

Thanks & Regards,
SHAAN,
shaantheadmin@gmail.com

shaan

have you find a way to auto install those required perl packages?

Op3r wrote:shaan

have you find a way to auto install those required perl packages?

I am Working on it. not a big deal we can do it. download all the packages and install locally.. or create a virtual CPN source in local system and give that URL: that way also we can do.

Any how i will get back to you on this with a solution ASAP.

thanks & Regards,
SHAAN.

Hi,

I am after configuring everything. i can able to login from from vicidial and x-light get a ring. but when i answer the call it will hangup.
my asterisk -r says the below. 1001 is my extension number. what could be the problem.


#####################error:######################
> Channel SIP/1001-09074f10 was answered.

== Starting SIP/1001-09334cc0 at default,,1 failed so falling back to exten 's'
== Starting SIP/1001-09334cc0 at default,s,1 still failed so falling back to context 'default'
[Nov 26 10:35:52] WARNING[27093]: pbx.c:2405 __ast_pbx_run: Channel 'SIP/1001-09334cc0' sent into invalid extension 's' in context 'default', but no invalid handler
-- Executing [h@default:1] DeadAGI("SIP/1001-09334cc0", "call_log.agi|h") in new stack

#####################error:######################

Thanks & Regards,
SHAAN.

Shaan,

are you using this wget http://asterisk.gnuinter.net/files/aste ... .09.tar.gz ?

dont, get the 0.08 version

hope you learn the lesson!

Hi,

yes i am using the latest asterisk perl "asterisk-perl-0.09.tar.gz"
is there any issues with it.

Thanks & Regards,
SHAAN.

get the 0.08 version

I think matt said that the 0.09 is not compatible with vicidial

Correct, asterisk-perl 0.09 does not work with VICIDIAL, they changed it too much so that it is not backward compatible with a lot of their old features, they also added a lot of new code that is unnecessary for VICIDIAL.

Hi,

I got it. its working gr8. iam preparing clear document for vici.

i need some inputs regarding g729

Open Source G.729 and G.723.1 codec i installed in my my system. my first question is if i install this codec with intel IPP evolution (Integrated Performance Primitives) for 30 days trial. how many concurrent calls it will support. i mean is there any license limit for g729 calls.

after installing the codec do i need to buy the licence from digium.com for g729 to make calls.

intel IPP is 110$ and one g729 codec license is 10$. please suggest me do we need both of this in production to make calls using g729. or any one is fine.

if any one of the above is fine then with intel IPP how many concurrent calls i can make. i mean is there license limit after buying IPP.

I want to Use g729 codec. suggest me whats the procedure do i need to fallow in production environment.

i need the above inputs for my documentation. i am explaining about the codecs. and how to use them in production.

Thanks & Regards,
SHAAN.

Hi shaan,

Can you please post the document that you were talking about, I am also facing the same problems that you have faced long time ago with trixbox and vicidial. How did u fix that problem. I followed the instructions in http://iptn.org/vicidial/index.html and scratch install just like u did, when i login as agent iam not getting the ring on my soft phone.

Can you please tell me where should i look.

Iam using trixbox 1.2.3 version.

Thanks & Regards,
Sundar.

Please post full document on the forum.

We all are waiting for you.

Could you please also post on forum the 'scratch_installation' document , bcz I tried to use it so many times but not successful.
Any help is welcome

Thanks to all in forums.

best regards
Kaz