vicidial login issue

[palani.sivagurunathan]

Team,

I have installed vicidial in ubuntu 10.04LTS server . Before a week Its working fine . i have upgraded apache-2.2.14 to 2.2.22 version and latest linux supporting file in this server. Asterisk and vicidial working but i can't login to vicidial agent screen . Asterisk meetme(conference) call is not bridged to softphone(Xlite).

Below the my version and configuration details :

asterisk-1.4.39.1
asterisk-addons-1.4.13
dahdi-linux-complete-2.6.0+2.6.0
libpri-1.4.12
Vicidial Version : 2.4-357a
PHP 5.3.10-1ubuntu3
Mysql 5.5.22-0ubuntu1

Manager.conf

[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0

[cron]
secret = 1234
read = system,call,log,verbose,command,agent,user
write = system,call,log,verbose,command,agent,user

[updatecron]
secret = 1234
read = command
write = command

[listencron]
secret = 1234
read = system,call,log,verbose,command,agent,user
write = command

[sendcron]
secret = 1234
read = command
write = system,call,log,verbose,command,agent,user

[check_asterisk_user]
secret = nagiosadmin
deny=0.0.0.0/0.0.0.0
permit=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,command




meetme.conf

[rooms]
conf => 8600
conf => 8601,1234
#include meetme-vicidial.conf

meetme-vicidial.conf

; WARNING- THIS FILE IS AUTO-GENERATED BY VICIDIAL, ANY EDITS YOU MAKE WILL BE LOST
; ViciDial Conferences:
conf => 8600051
conf => 8600052
conf => 8600053
conf => 8600054
conf => 8600055
conf => 8600056
conf => 8600057
conf => 8600058
conf => 8600059
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conf => 8600100
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conf => 8600297
conf => 8600298
conf => 8600299

; Conferences:
conf => 8600001
conf => 8600002
conf => 8600003
conf => 8600004
conf => 8600005
conf => 8600006
conf => 8600007
conf => 8600008
conf => 8600009
conf => 8600010
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conf => 8600051
conf => 8600052


; END OF FILE





astguiclient.conf

# /etc/astguiclient.conf - configuration elements for the astguiclient/vicidial package
# this is the astguiclient configuration file
# all comments will be lost if you run install.pl again

# Paths used by astGUIclient
PATHhome => /var/www
PATHlogs => /var/log/astguiclient
PATHagi => /var/lib/asterisk/agi-bin
PATHweb => /var/www
PATHsounds => /var/lib/asterisk/sounds
PATHmonitor => /var/spool/asterisk/monitor
PATHDONEmonitor => /var/spool/asterisk/monitorDONE

# The IP address of this machine
VARserver_ip => xx.xx.xxx.xx

# Database connection information
VARDB_server => localhost
VARDB_database => asterisk
VARDB_user => xxxxxxxx
VARDB_pass => xxxxxxxxxxxx
VARDB_custom_user => xxxxxxxxxxx
VARDB_custom_pass => xxxxxxxxx
VARDB_port => 3306

# Alpha-Numeric list of the astGUIclient processes to be kept running
# (value should be listing of characters with no spaces: 1234568)
# 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)
# 9 - Timeclock auto logout
VARactive_keepalives => 1234568

# Asterisk version VICIDIAL is installed for
VARasterisk_version => 1.4

# FTP recording archive connection information
VARasterisk_version => 1.4

# FTP recording archive connection information
VARFTP_host => 10.0.0.4
VARFTP_user => cron
VARFTP_pass => test
VARFTP_port => 21
VARFTP_dir => RECORDINGS
VARHTTP_path => http://10.0.0.4

# REPORT server connection information
VARREPORT_host => 10.0.0.4
VARREPORT_user => cron
VARREPORT_pass => test
VARREPORT_port => 21
VARREPORT_dir => REPORTS

# Settings for FastAGI logging server
VARfastagi_log_min_servers => 3
VARfastagi_log_max_servers => 16
VARfastagi_log_min_spare_servers => 2
VARfastagi_log_max_spare_servers => 8
VARfastagi_log_max_requests => 1000
VARfastagi_log_checkfordead => 30
VARfastagi_log_checkforwait => 60

Sip.conf

[general]
context=trunkinbound ; Default context for incoming calls
allowguest=no ; Allow or reject guest calls (default is yes)
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
;realm=mydomain.tld ; Realm for digest authentication
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;domain=mydomain.tld ; Set default domain for this host
;pedantic=yes ; Enable checking of tags in headers,
;tos_sip=cs3 ; Sets TOS for SIP packets.
;tos_audio=ef ; Sets TOS for RTP audio packets.
;tos_video=af41 ; Sets TOS for RTP video packets.
;maxexpiry=3600 ; Maximum allowed time of incoming registrations
;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120 ; Default length of incoming/outgoing registration
;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10 ; Default time between mailbox checks for peers
;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
;allow=gsm
mohinterpret=default
mohsuggest=default
language=en ; Default language setting for all users/peers
relaxdtmf=yes ; Relax dtmf handling
trustrpid = no ; If Remote-Party-ID should be trusted
sendrpid = yes ; If Remote-Party-ID should be sent
progressinband=no ; If we should generate in-band ringing always
;useragent=Asterisk PBX ; Allows you to change the user agent string
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
;compactheaders = yes ; send compact sip headers.
videosupport=no ; Turn on support for SIP video. You need to turn this on
;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
callevents=yes ; generate manager events when sip ua
alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
;regcontext=sipregistrations
rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
;sipdebug = yes ; Turn on SIP debugging by default, from
;recordhistory=yes ; Record SIP history by default
;dumphistory=yes ; Dump SIP history at end of SIP dialogue
;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
limitonpeers = yes ; Apply call limits on peers only. This will improve
;t38pt_udptl = yes ; Default false
;register => 1234:password@mysipprovider.com
;registertimeout=20 ; retry registration calls every 20 seconds (default)
;registerattempts=10 ; Number of registration attempts before we give up
;externip = 192.168.1.1 ; Address that we're going to put in outbound SIP
;externhost=test.test.com ; Alternatively you can specify a domain
;externrefresh=10 ; How often to refresh externhost if
localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
nat=yes ; Global NAT settings (Affects all peers and users)
canreinvite=no ; Asterisk by default tries to redirect the
;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
;rtsavesysname=yes ; Save systemname in realtime database at registration
;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
;ignoreregexpire=yes ; Enabling this setting has two functions:
;domain=mydomain.tld,mydomain-incoming
;domain=1.2.3.4 ; Add IP address as local domain
;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
;autodomain=yes ; Turn this on to have Asterisk add local host
;fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
jbmaxsize = 100 ; Max length of the jitterbuffer in milliseconds.
jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
qualify=yes ; By default, qualify all peers at 2000ms
limitonpeer = yes ; enable call limit on a per peer basis, different from limitonpeers

/etc/apache2/sites-available/default

<VirtualHost *:80>
ServerAdmin webmaster@localhost
DocumentRoot /var/www
<Directory />
Options FollowSymLinks
AllowOverride None
</Directory>
<Directory /var/www/>
Options Indexes FollowSymLinks MultiViews
AllowOverride None
Order allow,deny
allow from all
</Directory>

ScriptAlias /cgi-bin/ /usr/lib/cgi-bin/
<Directory "/usr/lib/cgi-bin">
AllowOverride None
Options +ExecCGI -MultiViews +SymLinksIfOwnerMatch
Order allow,deny
Allow from all
</Directory>

ErrorLog ${APACHE_LOG_DIR}/error.log

# Possible values include: debug, info, notice, warn, error, crit,
# alert, emerg.
LogLevel warn

CustomLog ${APACHE_LOG_DIR}/access.log combined

Alias /doc/ "/usr/share/doc/"
<Directory "/usr/share/doc/">
Options Indexes MultiViews FollowSymLinks
AllowOverride None
Order deny,allow
Deny from all
Allow from 127.0.0.0/255.0.0.0 ::1/128
</Directory>

</VirtualHost>


what am i doing wrong with my configuration . please help me .

Thanks,
Palanivel S

[uperformkarthick]

If this is new install please go with vicibox install. It is a easiest and convenient way of installing vicidial.

Thanks
Karthik Raja

[williamconley]

there are many "scratch install instructions" on the net for Vicidial. Several are for Ubuntu. Which one did you use?

What errors are you seeing in your logs?

Why (Why!) are you not installing Vicibox (from Vicibox.com) or GoAutoDial (from GoAutoDial.com)?

Also, please refrain from posting massive files that have merely been copied from your system. We've seen them before and they are not an indication of a flaw or problem.

If you MUST install from scratch ... find an existing install instruction set, and follow it ... and ask for help when you hit each small bump. You'll get help that way. But if you ask the group to help you build your ferrari all at once, right now ... odds are not very good.