WORKING OF VICIDIALER??

All installation and configuration problems and questions

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Postby jlodvo » Mon Aug 14, 2006 6:05 am

got the same problem

Syntax error on line 2 of /usr/local/apache2/conf/httpd.conf:
Cannot load /usr/local/apache2/libexec/libphp4.so into server: /usr/local/apache2/libexec/libphp4.so: cannot open shared object file: No such file or directory




root@phone:~# locate libphp4.so
/usr/local/php-4.4.2/.libs/libphp4.so
/usr/local/php-4.4.2/libs/libphp4.so
/usr/local/apache2/modules/libphp4.so
/usr/libexec/apache/libphp4.so
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Postby mflorell » Mon Aug 14, 2006 6:13 am

go back into your httpd.conf file and change
/usr/local/apache2/libexec/libphp4.so
to
/usr/local/apache2/modules/libphp4.so
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Postby jlodvo » Mon Aug 14, 2006 6:57 am

its working now great thanks
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Postby prabhakar_asterisk » Tue Aug 15, 2006 10:01 pm

NO.THE CURSOR IS JUST BLINKING.I AM NOT GETTING PROMPT EVEN AFTER 1 HOUR
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Postby prabhakar_asterisk » Thu Aug 17, 2006 3:41 am

thanks its working fine now.
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Postby prabhakar_asterisk » Thu Aug 17, 2006 11:46 pm

while configuring mysql for vicidialer

# /home/cron/ADMIN_area_code_populate.pl
Access denied for user: 'cron@Asterisk' (Using password: YES) at /home/cron/ADMIN_area_code_populate.pl line 23

i never set any password for mysql and i dunno how to solve this issue

let me know

thanks
prabhakar
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Postby mflorell » Fri Aug 18, 2006 6:21 am

look for the "GRANT" statements in the SCRATCH_INSTALL document.
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configuration of vicidialer using SIP account

Postby prabhakar_asterisk » Sun Aug 20, 2006 11:27 pm

hi
thanks for your reply and working fine..i was trying to coinfigure vicidialer for my organisation..i completed till phase 6.1 and i am getting the admin page of astguiclient
i am testing it with 1 sip phone in asterisk server and 1 hard phone in the network and asterisk is working perefctly.
i am going to get a SIP account from teh service provider then want to make outbound calls .
can you tell me how to configure vicidialer in this scenario..i dont have database right now..i want to test vicidilaer first then i want to go live with my database
thanks
prabhakar
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Postby mflorell » Mon Aug 21, 2006 1:37 am

The SCRATCH_INSTALL has an example of SIP trunks that you can use in most cases.

You cannot use VICIDIAL without a database.
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Postby prabhakar_asterisk » Mon Aug 21, 2006 3:02 am

thanks
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how to see viciidialer webadmin

Postby prabhakar_asterisk » Mon Aug 21, 2006 5:22 am

hi matt
i know this question seem very silly
how to find the viicidialers webadmin page
thanks
prabhakar
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Postby prabhakar_asterisk » Mon Aug 21, 2006 5:39 am

i found the answer..
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problem connecting asterisk to sip account

Postby prabhakar_asterisk » Tue Aug 22, 2006 11:35 pm

hi
i couldnt able to connect asterisk with sip account (net2phone)
when i try to call i am getting unable to create channel of type 'SIP' (CAUSE 3-NO ROUTE TO DESTINATION)
when i type sip peers
status is unreacheable ..i am behind nat..do you think its a natting problem.
thanks
prabhakar
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Postby mflorell » Wed Aug 23, 2006 12:14 am

This is more of an Asterisk-users general question than a VICIDIAL one. you probably are having NAT issues if you are using SIP behind a firewall. Search google or voip-info.org for more information on your router and how you would have to set it up to get it to work.
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Postby Trilochan Panda » Fri Sep 01, 2006 7:51 pm

hello,

I have installed all the package which is given n scarch install. I am useing X-Lite. Now the login screen is comeing and through X-Lite i am able to call, but the problem is iafter login into the agc/vicidial.php screen i am not getting beep tone . agent manual is telling after login into the agent screen that should be ring that i don't found. Please help me sir. Its very urgent.

Through vicidial agent scrren i want to diall with the help of X-Lite softphone. Please help me so that i will be grateful to you

Thanks for advance
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Postby mflorell » Fri Sep 01, 2006 8:30 pm

Please post Asterisk CLI output when you try to login.
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Postby Trilochan Panda » Sat Sep 02, 2006 3:38 pm

Thanks for reply

I resolved that previous proble. The problem was in crontab -e. I have checked all the script which was in crontab -e. Now another problem is where and how to add the 'o' flag in extension.conf please give me the suggesation. Thanks
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dial option

Postby espencer » Sat Sep 02, 2006 4:14 pm

the dial option should be in the dial string in extensions.conf like so:

; exten => _91NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN})
; exten => _91NXXNXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},,tTo)
; exten => _91NXXNXXXXXX,3,Hangup
; dial a long distance outbound number through BINFONE
exten => _91NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN})
exten => _91NXXNXXXXXX,2,Dial(${TRUNKIAX}/${EXTEN},55,o)
exten => _91NXXNXXXXXX,3,Hangup


(example taken from the sample extensions.conf entries in the scratch install)
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espencer

Postby Trilochan Panda » Sat Sep 02, 2006 4:38 pm

Hello Sir,

the following is the default endty of my outbound plan. I get a service
unavailable message in my soft phone.
; dial a long distance outbound number
exten => _91NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN})
exten => _91NXXNXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},,tTo)
exten => _91NXXNXXXXXX,3,Hangup

Neither I am able to do out bound calls through auto or manual mode.


But If I change it to the following I am able to dial manually without the 91
and I am able to connect to any number in the US.
but Iam unable to dial though VICIDIAL from the leads list. But I can make a
manual call from the manual dial option in the vicidial interface. In both the
cases I am able to get the link in the soft sip phone (Xlite)
I have one TDM400p card with one fxs and one fxo port. My out US Line is
connected on ZAP 4.

Also please guide me how to add the 'o' flag and where.


exten => _XXXXXXXXXX,1,Dial(zap/4/${EXTEN})
exten => _XXXXXXXX,1,Dial(zap/4/${EXTEN})
exten => _XXX,1,Dial(SIP/${EXTEN})

Also

Any help in this regard will be highly appreciated.

Regards,
Trilochan Panda
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mflorell

Postby Trilochan Panda » Sat Sep 02, 2006 5:13 pm

Hello Sir,

the following is the default endty of my outbound plan. I get a service
unavailable message in my soft phone.
; dial a long distance outbound number
exten => _91NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN})
exten => _91NXXNXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},,tTo)
exten => _91NXXNXXXXXX,3,Hangup

Neither I am able to do out bound calls through auto or manual mode.


But If I change it to the following I am able to dial manually without the 91
and I am able to connect to any number in the US.
but Iam unable to dial though VICIDIAL from the leads list. But I can make a
manual call from the manual dial option in the vicidial interface. In both the
cases I am able to get the link in the soft sip phone (Xlite)
I have one TDM400p card with one fxs and one fxo port. My out US Line is
connected on ZAP 4.

Also please guide me how to add the 'o' flag and where.


exten => _XXXXXXXXXX,1,Dial(zap/4/${EXTEN})
exten => _XXXXXXXX,1,Dial(zap/4/${EXTEN})
exten => _XXX,1,Dial(SIP/${EXTEN})

Also

Any help in this regard will be highly appreciated.
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Postby espencer » Sat Sep 02, 2006 5:33 pm

so it sounds like you may need to define TRUNKX=Zap/4 in your globals section of extensions.conf for the original dialplan lines to work properly.

also, your default campaign prefixes numbers with a "9" so you will want to change that part of your campaign.

the "o" flag only changes the way the CDR gets written so it should not affect whether or not you get connected. if you want to use your lines below you will need do this:

exten => _XXXXXXXXXX,1,Dial(zap/4/${EXTEN},,tTo)
exten => _XXXXXXXX,1,Dial(zap/4/${EXTEN},,tTo)
exten => _XXX,1,Dial(SIP/${EXTEN},,tTo)

however, from what you have written there, you are sending only 10 digits out for an LD call which is weird (and 8 digits for a local call -- even odder). i would expect the following to work better in the US:

exten => _1NXXNXXXXXX,1,Dial(zap/4/${EXTEN},,tTo)
exten => _NXXXXXX,1,Dial(zap/4/${EXTEN},,tTo)
exten => _XXX,1,Dial(SIP/${EXTEN},,tTo)
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espencer

Postby Trilochan Panda » Sat Sep 02, 2006 5:51 pm

Sir thank you very much for your guidence. Sir but how to include the agi script in extensions for call back. Another problem is Whenever i am dialing from leads NO LIVE CALLS option is not high lighting please give me the guide of this. Thank you again
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Postby espencer » Sun Sep 03, 2006 10:33 am

paste your asterisk CLI output here so I can see. sounds like the call is not getting transferred to your conference which can be from lack of an "o" in the dialplan (already doing this, i gather) or running on Trixbox. don't forget to make the asterisk CLI output verbose:

asterisk -vvvvvvvvvvvvvvvvvvvgrdc
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Postby prabhakar_asterisk » Thu Sep 07, 2006 5:52 am

hi guys
when i log on to clinets page of vicidialers
//ipaddress/agc/vicidial.php
i am getting connetion refused from asterisk server
cheers
prabhakar
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Postby prabhakar_asterisk » Fri Sep 15, 2006 6:41 am

hi matt
i configgiures vicidialer as given the scratch install..my asteriks is working perfectly..i can maek ooutbound calls and i can connect to audiocodes with it

i am trying to check the vicidial and doing the simple exercise you given the managers manual..the lead i loaded using the load leads is not seen in the campaigns and also when i log into the client i couldnt see any leads
"If there are no leads in the hopper something is either wrong with the leads you loaded or there is something wrong with your hopper loading script"

how to find where is the problem
thanks in advance
prabhakar
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Postby gerski » Fri Sep 15, 2006 11:37 am

when you load the leads do you actually set the list ID? it is important so the system know where will he put the leads... and then set it to yes and check on your campaign and see if it loads correctly.
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Postby prabhakar_asterisk » Mon Sep 18, 2006 12:23 am

i am still having the problem.i set the list id in the field provided but i still couldnt see anything in the campaign.i can see the active list in the campaign.when i load the new loads


its showing like this

Processing pipe-delimited file... (0|22)

LIST ID OVERRIDE FOR THIS FILE: 107
Done GOOD: 1 BAD: 0 TOTAL: 1

but when i click the appropriate campaign it shows

This campaign has 2 active lists and 0 inactive lists

This campaign has 0 leads to be dialed in those lists - HIDE

This campaign has 0 leads in the dial hopper


Thanks in advance
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Postby prabhakar_asterisk » Wed Oct 04, 2006 6:30 am

i can connect to sip server connected in india but i couldnt connect to the sip server in US
thanks in advance
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freepbx and VICIDIAL

Postby shaan » Sun Nov 12, 2006 5:45 pm

mflorell wrote:freepbx creates a mess of dialplan rules that cause all sorts of problems for VICIDIAL. The biggest one is the callerID issue where freepbx overrides the callerID set by VICIDIAL unless you setup several special rules in freepbx so that it will not mess with callerID on VICIDIAL calls. The issue you are having with it is unrelated to the callerID issue and I do not know how to fix that either within freepbx.

We are working on testing freepbx with VICIDIAL but we still haven't gotten it to work. When we do, we will post our results on this forum. Until then I suggest you remove freepbx and use basic conf files.


Hi,

i am working on auto installation of freepbx and VICIDIAL on Fedora core6.
i want to write a document on freepbx and VICIDIAL just like qmail rocks. my motive is install freepbx and VICIDIAL without any issues.
want to rock the asterisk with freepbx and VICIDIAL. iam sure we can make this so that it will be helpful for all.

Current Status:
i have prepared a detail installation Document for freepbx and VICIDIAL. and working on Auto installation scripts.

Chalenges:
freepbx work fine. i can do manual dialing from x-light. and from VICIDIAL<http://192.168.0.99/agc/vicidial.php> if i call it rings but when i pick up the call it says goodbye. i have 8600051 in extention.conf and in meetme.conf too.

error details:
####################localhost*CLI>#################
== Starting SIP/786-08b0e528 at default,8600051,1 failed so falling back to exten 's'
-- Executing [s@default:1] Playback("SIP/786-08b0e528", "vm-goodbye") in new stack
-- Playing 'vm-goodbye' (language 'en')
-- Executing [s@default:2] Macro("SIP/786-08b0e528", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/786-08b0e528", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/786-08b0e528", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/786-08b0e528", "1?theend") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] Wait("SIP/786-08b0e528", "5") in new stack
-- Executing [s@macro-hangupcall:7] Hangup("SIP/786-08b0e528", "") in new stack
== Spawn extension (macro-hangupcall, s, 7) exited non-zero on 'SIP/786-08b0e528' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 7) exited non-zero on 'SIP/786-08b0e528'
####################localhost*CLI>#################

*Teach to Learn*
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