problema freepbx

problema freepbx

Posted Agosto 30th, 2010 by jennereduardo

hola mi problema es q cuando entro a http://localhost/asterisk/ luego a freepbx me sale el siguiuente mensaje

Warning:

Cannot connect to Asterisk Manager with asterisk
Asterisk may not be running

aca les dejo una foto del error: http://img210.imageshack.us/f/error1m.jpg/

y en el terminal me sale q si se esta ejecutando el asterisk, les copio lo q tengo en manager.conf", "modules.conf" y "amportal.conf

---------------manager.conf:

;
; AMI - Asterisk Manager interface
;
; FreePBX needs this to be enabled. Note that if you enable it on a different IP, you need
; to assure that this can't be reached from un-authorized hosts with the ACL settings (permit/deny).
; Also, remember to configure non-default port or IP-addresses in amportal.conf.
;
; The AMI connection is used both by the portal and the operator's panel in FreePBX.
;
; FreePBX assumes an AMI connection to localhost:5038 by default.
;
[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0
displayconnects=no ;only effects 1.6+

[asterisk]
secret = asterisk
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,config,command,dtmf,reporting,cdr,dialplan,originate
write = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate

#include manager_additional.conf
#include manager_custom.conf

;debes agregar un usuario para tu panel.
[ELusuarioDELpanel]
secret = USpanelpass
deny=0.0.0.0/0.0.0.0
permit=127.0.0.1/255.255.255.0

;permit=192.168.0.0/255.255.255.0

----------------------amportal.conf

# This file is part of FreePBX.
#
# FreePBX is free software: you can redistribute it and/or modify
# it under the terms of the GNU General Public License as published by
# the Free Software Foundation, either version 2 of the License, or
# (at your option) any later version.
#
# FreePBX is distributed in the hope that it will be useful,
# but WITHOUT ANY WARRANTY; without even the implied warranty of
# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
# GNU General Public License for more details.
#
# You should have received a copy of the GNU General Public License
# along with FreePBX. If not, see .
#
# This file contains settings for components of the Asterisk Management Portal
# Spaces are not allowed!
# Run /usr/src/AMP/apply_conf.sh after making changes to this file

# FreePBX Database configuration
# AMPDBHOST: Hostname where the FreePBX database resides
# AMPDBENGINE: Engine hosting the FreePBX database (e.g. mysql)
# AMPDBNAME: Name of the FreePBX database (e.g. asterisk)
# AMPDBUSER: Username used to connect to the FreePBX database
# AMPDBPASS: Password for AMPDBUSER (above)
# AMPENGINE: Telephony backend engine (e.g. asterisk)
# AMPMGRUSER: Username to access the Asterisk Manager Interface
# AMPMGRPASS: Password for AMPMGRUSER
#
AMPDBHOST=localhost
AMPDBENGINE=mysql
# AMPDBNAME=asterisk
# AMPDBUSER=asteriskuser
# AMPDBPASS=amp109
AMPENGINE=asterisk
AMPMGRUSER=asteriskuser
AMPMGRPASS=asterisk

# AMPBIN: Location of the FreePBX command line scripts
# AMPSBIN: Location of (root) command line scripts
#
AMPBIN=/var/lib/asterisk/bin
AMPSBIN=/usr/local/sbin

# AMPWEBROOT: Path to Apache's webroot (leave off trailing slash)
# AMPCGIBIN: Path to Apache's cgi-bin dir (leave off trailing slash)
# AMPWEBADDRESS: The IP address or host name used to access the AMP web admin
#
AMPWEBROOT=/var/www/html/asterisk
AMPCGIBIN=/var/www/cgi-bin
# AMPWEBADDRESS=x.x.x.x|hostname

# FOPWEBROOT: Path to the Flash Operator Panel webroot (leave off trailing slash)
# FOPPASSWORD: Password for performing transfers and hangups in the Flash Operator Panel
# FOPRUN: Set to true if you want FOP started by freepbx_engine (amportal_start), false otherwise
# FOPDISABLE: Set to true to disable FOP in interface and retrieve_conf. Useful for sqlite3
# or if you don't want FOP.
#
FOPRUN=true
FOPWEBROOT=/var/www/html/asterisk/panel
FOPPASSWORD=asterisk

# FOPSORT=extension|lastname
# DEFAULT VALUE: extension
# FOP should sort extensions by Last Name [lastname] or by Extension [extension]

# This is the default admin name used to allow an administrator to login to ARI bypassing all security.
# Change this to whatever you want, don't forget to change the ARI_ADMIN_PASSWORD as well
ARI_ADMIN_USERNAME=admin

# This is the default admin password to allow an administrator to login to ARI bypassing all security.
# Change this to a secure password.
ARI_ADMIN_PASSWORD=ari_password

# AUTHTYPE=database|none
# Authentication type to use for web admininstration. If type set to 'database', the primary
# AMP admin credentials will be the AMPDBUSER/AMPDBPASS above.
AUTHTYPE=none

# AMPADMINLOGO=filename
# Defines the logo that is to be displayed at the TOP RIGHT of the admin screen. This enables
# you to customize the look of the administration screen.
# NOTE: images need to be saved in the ..../admin/images directory of your AMP install
# This image should be 55px in height
AMPADMINLOGO=logo.png

# USECATEGORIES=true|false
# DEFAULT VALUE: true
# Controls if the menu items in the admin interface are sorted by category (true), or sorted
# alphabetically with no categories shown (false).

# AMPEXTENSIONS=extensions|deviceanduser
# Sets the extension behavior in FreePBX. If set to 'extensions', Devices and Users are
# administered together as a unified Extension, and appear on a single page.
# If set to 'deviceanduser', Devices and Users will be administered seperately. Devices (e.g.
# each individual line on a SIP phone) and Users (e.g. '101') will be configured
# independent of each other, allowing association of one User to many Devices, or allowing
# Users to login and logout of Devices.
AMPEXTENSIONS=extensions

# ENABLECW=true|false
# DEFAULT VALUE: true
# Enable call waiting by default when an extension is created. Set to 'no' to if you don't want
# phones to be commissioned with call waiting already enabled. The user would then be required
# to dial the CW feature code (*70 default) to enable their phone. Most installations should leave
# this alone. It allows multi-line phones to receive multiple calls on their line appearances.

# CWINUSEBUSY=true|false
# DEFAULT VALUE: true
# For extensions that have CW enabled, report unanswered CW calls as 'busy' (resulting in busy
# voicemail greeting). If set to no, unanswered CW calls simply report as 'no-answer'.

# AMPBADNUMBER=true|false
# DEFAULT VALUE: true
# Generate the bad-number context which traps any bogus number or feature code and plays a
# message to the effect. If you use the Early Dial feature on some Grandstream phones, you
# will want to set this to false.

# AMPBACKUPSUDO=true|false
# DEFAULT VALUE: false
# This option allows you to use sudo when backing up files. Useful ONLY when using AMPPROVROOT
# Allows backup and restore of files specified in AMPPROVROOT, based on permissions in /etc/sudoers
# for example, adding the following to sudoers would allow the user asterisk to run tar on ANY file
# on the system:
# asterisk localhost=(root)NOPASSWD: /bin/tar
# Defaults:asterisk !requiretty
# PLEASE KEEP IN MIND THE SECURITY RISKS INVOLVED IN ALLOWING THE ASTERISK USER TO TAR/UNTAR ANY FILE

# CUSTOMASERROR=true|false
# DEFAULT VALUE: true
# If false, then the Destination Registry will not report unknown destinations as errors. This should be
# left to the default true and custom destinations should be moved into the new custom apps registry.

# DYNAMICHINTS=true|false
# DEFAULT VALUE: false
# If true, Core will not statically generate hints, but instead make a call to the AMPBIN php script,
# and generate_hints.php through an Asterisk's #exec call. This requires Asterisk.conf to be configured
# with "execincludes=yes" set in the [options] section.

# XTNCONFLICTABORT=true|false
# BADDESTABORT=true|false
# DEFAULT VALUE: false
# Setting either of these to true will result in retrieve_conf aborting during a reload if an extension
# conflict is detected or a destination is detected. It is usually better to allow the reload to go
# through and then correct the problem but these can be set if a more strict behavior is desired.

# SERVERINTITLE=true|false
# DEFAULT VALUE: false
# Precede browser title with the server name.

# USEDEVSTATE = true|false
# DEFAULT VALUE: false
# If this is set, it assumes that you are running Asterisk 1.4 or higher and want to take advantage of the
# func_devstate.c backport available from Asterisk 1.6. This allows custom hints to be created to support
# BLF for server side feature codes such as daynight, followme, etc.

# MODULEADMINWGET=true|false
# DEFAULT VALUE: false
# Module Admin normally tries to get its online information through direct file open type calls to URLs that
# go back to the freepbx.org server. If it fails, typically because of content filters in firewalls that
# don't like the way PHP formats the requests, the code will fall back and try a wget to pull the information.
# This will often solve the problem. However, in such environment there can be a significant timeout before
# the failed file open calls to the URLs return and there are often 2-3 of these that occur. Setting this
# value will force FreePBX to avoid the attempt to open the URL and go straight to the wget calls.

# AMPDISABLELOG=true|false
# DEFAULT VALUE: true
# Whether or not to invoke the FreePBX log facility

# AMPSYSLOGLEVEL=LOG_EMERG|LOG_ALERT|LOG_CRIT|LOG_ERR|LOG_WARNING|LOG_NOTICE|LOG_INFO|LOG_DEBUG|LOG_SQL|SQL
# DEFAULT VALUE: LOG_ERR
# Where to log if enabled, SQL, LOG_SQL logs to old MySQL table, others are passed to syslog system to
# determine where to log

# AMPENABLEDEVELDEBUG=true|false
# DEFAULT VALUE: false
# Whether or not to include log messages marked as 'devel-debug' in the log system

# AMPMPG123=true|false
# DEFAULT VALUE: true
# When set to false, the old MoH behavior is adopted where MP3 files can be loaded and WAV files converted
# to MP3. The new default behavior assumes you have mpg123 loaded as well as sox and will convert MP3 files
# to WAV. This is highly recommended as MP3 files heavily tax the system and can cause instability on a busy
# phone system.

# CDR DB Settings: Only used if you don't use the default values provided by FreePBX.
# CDRDBHOST: hostname of db server if not the same as AMPDBHOST
# CDRDBPORT: Port number for db host
# CDRDBUSER: username to connect to db with if it's not the same as AMPDBUSER
# CDRDBPASS: password for connecting to db if it's not the same as AMPDBPASS
# CDRDBNAME: name of database used for cdr records
# CDRDBTYPE: mysql or postgres mysql is default
# CDRDBTABLENAME: Name of the table in the db where the cdr is stored cdr is default

# AMPVMUMASK=mask
# DEFAULT VALUE: 077
# Defaults to 077 allowing only the asterisk user to have any permission on VM files. If set to something
# like 007, it would allow the group to have permissions. This can be used if setting apache to a different
# user then asterisk, so that the apache user (and thus ARI) can have access to read/write/delete the
# voicemail files. If changed, some of the voicemail directory structures may have to be manually changed.

# DASHBOARD_STATS_UPDATE_TIME=integer_seconds
# DEFAULT VALUE: 6
# DASHBOARD_INFO_UPDATE_TIME=integer_seconds
# DEFAULT VALUE: 20
# These can be used to change the refresh rate of the System Status Panel. Most of
# the stats are updated based on the STATS interval but a few items are checked
# less frequently (such as Asterisk Uptime) based on the INFO value

# ZAP2DAHDICOMPAT=true|false
# DEFAULT VALUE: false
# If set to true, FreePBX will check if you have chan_dadhi installed. If so, it will
# automatically use all your ZAP configuration settings (devices and trunks) and
# silently convert them, under the covers, to DAHDI so no changes are needed. The
# GUI will continue to refer to these as ZAP but it will use the proper DAHDI channels.
# This will also keep Zap Channel DIDs working.

# CHECKREFERER=true|false
# DEFAULT VALUE: true
# When set to the default value of true, all requests into FreePBX that might possibly add/edit/delete
# settings will be validated to assure the request is coming from the server. This will protect the system
# from CSRF (cross site request forgery) attacks. It will have the effect of preventing legitimately entering
# URLs that could modify settings which can be allowed by changing this field to false.

# USEQUEUESTATE=true|false
# DEFAULT VALUE: false
# Setting this flag will generate the required dialplan to integrate with the following Asterisk patch:
# https://issues.asterisk.org/view.php?id=15168
# This feature is planned for a future 1.6 release but given the existence of the patch can be used prior. Once
# the release version is known, code will be added to automatically enable this format in versions of Asterisk
# that support it.

# USEGOOGLEDNSFORENUM=true|false
# DEFAULT VALUE: false
# Setting this flag will generate the required global variable so that enumlookup.agi will use Google DNS
# 8.8.8.8 when performing an ENUM lookup. Not all DNS deals with NAPTR record, but Google does. There is a
# drawback to this as Google tracks every lookup. If you are not comfortable with this, do not enable this
# setting. Please read Google FAQ about this: http://code.google.com/speed/public-dns/faq.html#privacy

# MOHDIR=subdirectory_name
# This is the subdirectory for the MoH files/directories which is located in ASTVARLIBDIR
# if not specified it will default to mohmp3 for backward compatibility.

# RELOADCONFIRM=true|false
# DEFAULT VALUE: true
# When set to false, will bypass the confirm on Reload Box

# FCBEEPONLY=true|false
# DEFAULT VALUE: false
# When set to true, a beep is played instead of confirmation message when activating/de-activating:
# CallForward, CallWaiting, DayNight, DoNotDisturb and FindMeFollow

# DISABLECUSTOMCONTEXTS=true|false
# DEFAULT VALUE: false
# Normally FreePBX auto-generates a custom context that may be usable for adding custom dialplan to modify the
# normal behavior of FreePBX. It takes a good understanding of how Asterisk processes these includes to use
# this and in many of the cases, there is no useful application. All includes will result in a WARNING in the
# Asterisk log if there is no context found to include though it results in no errors. If you know that you
# want the includes, you can set this to true. If you comment it out FreePBX will revert to legacy behavior
# and include the contexts.

AMPDBUSER=asteriskuser
AMPDBPASS=asterisk
AMPWEBADDRESS=10.0.2.15
AMPDBNAME=asterisk
ASTETCDIR=/etc/asterisk
ASTMODDIR=/usr/lib/asterisk/modules
ASTVARLIBDIR=/var/lib/asterisk
ASTAGIDIR=/var/lib/asterisk/agi-bin
ASTSPOOLDIR=/var/spool/asterisk
ASTRUNDIR=/var/run/asterisk
ASTLOGDIR=/var/log/asterisk

---------------------modules.conf

;
; Asterisk Module Loader configuration file
;
;

[modules]
autoload=yes
;
; Any modules that need to be loaded before the Asterisk core has been
; initialized (just after the logger has been initialized) can be loaded
; using 'preload'. This will frequently be needed if you wish to map all
; module configuration files into Realtime storage, since the Realtime
; driver will need to be loaded before the modules using those configuration
; files are initialized.
;
; An example of loading ODBC support would be:
;preload => res_odbc.so
;preload => res_config_odbc.so
;
; As FreePBX is using Local as the channel for queue members we need to make sure
; that pbx_config.so and chan_local.so are preloaded. If not, queue members
; will be marked as invalid until app_queue is reloaded.
preload => pbx_config.so
preload => chan_local.so
;
; Uncomment the following if you wish to use the Speech Recognition API
;preload => res_speech.so
;
; If you want, load the GTK console right away.
; KDE console is obsolete and was removed from Asterisk 2008-01-10
;
noload => pbx_gtkconsole.so
;load => pbx_gtkconsole.so
noload => pbx_kdeconsole.so
;
; Intercom application is obsoleted by
; chan_oss. Don't load it.
;
noload => app_intercom.so
;
; DON'T load the chan_modem.so, as they are obsolete in * 1.2

noload => chan_modem.so
noload => chan_modem_aopen.so
noload => chan_modem_bestdata.so
noload => chan_modem_i4l.so

; Trunkisavail is a broken module supplied by Trixbox
noload => app_trunkisavail.so

; Ensure that format_* modules are loaded before res_musiconhold
;load => format_ogg_vorbis.so
load => format_wav.so

------------------------ he leido q debo poner en el amportal y el manager lo siguiente:

AMPDBUSER=admin
AMPDBPASS=secret123password

--------------------------en la linea 30 del amportal.conf estoy reemplazando el user y el pass

AMPDBHOST=localhost
AMPDBENGINE=mysql
# AMPDBNAME=asterisk
# AMPDBUSER=admin
# AMPDBPASS=secret123password
AMPENGINE=asterisk
AMPMGRUSER=asteriskuser
AMPMGRPASS=asterisk

--------------------------- y en la ultima parte tambien estoy poniendo el user y el pass

AMPDBUSER=admin
AMPDBPASS=secret123password
AMPWEBADDRESS=10.0.2.15
AMPDBNAME=asterisk
ASTETCDIR=/etc/asterisk
ASTMODDIR=/usr/lib/asterisk/modules
ASTVARLIBDIR=/var/lib/asterisk
ASTAGIDIR=/var/lib/asterisk/agi-bin
ASTSPOOLDIR=/var/spool/asterisk
ASTRUNDIR=/var/run/asterisk
ASTLOGDIR=/var/log/asterisk

-----------------------------------aca dejo el nuevo amportal.conf con el user y el pass reemplazados

-------------------amportal.config:

# This file is part of FreePBX.
#
# FreePBX is free software: you can redistribute it and/or modify
# it under the terms of the GNU General Public License as published by
# the Free Software Foundation, either version 2 of the License, or
# (at your option) any later version.
#
# FreePBX is distributed in the hope that it will be useful,
# but WITHOUT ANY WARRANTY; without even the implied warranty of
# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
# GNU General Public License for more details.
#
# You should have received a copy of the GNU General Public License
# along with FreePBX. If not, see .
#
# This file contains settings for components of the Asterisk Management Portal
# Spaces are not allowed!
# Run /usr/src/AMP/apply_conf.sh after making changes to this file

# FreePBX Database configuration
# AMPDBHOST: Hostname where the FreePBX database resides
# AMPDBENGINE: Engine hosting the FreePBX database (e.g. mysql)
# AMPDBNAME: Name of the FreePBX database (e.g. asterisk)
# AMPDBUSER: Username used to connect to the FreePBX database
# AMPDBPASS: Password for AMPDBUSER (above)
# AMPENGINE: Telephony backend engine (e.g. asterisk)
# AMPMGRUSER: Username to access the Asterisk Manager Interface
# AMPMGRPASS: Password for AMPMGRUSER
#
AMPDBHOST=localhost
AMPDBENGINE=mysql
# AMPDBNAME=asterisk
# AMPDBUSER=asteriskuser
# AMPDBPASS=amp109
AMPENGINE=asterisk
AMPMGRUSER=asteriskuser
AMPMGRPASS=asterisk

# AMPBIN: Location of the FreePBX command line scripts
# AMPSBIN: Location of (root) command line scripts
#
AMPBIN=/var/lib/asterisk/bin
AMPSBIN=/usr/local/sbin

# AMPWEBROOT: Path to Apache's webroot (leave off trailing slash)
# AMPCGIBIN: Path to Apache's cgi-bin dir (leave off trailing slash)
# AMPWEBADDRESS: The IP address or host name used to access the AMP web admin
#
AMPWEBROOT=/var/www/html/asterisk
AMPCGIBIN=/var/www/cgi-bin
# AMPWEBADDRESS=x.x.x.x|hostname

# FOPWEBROOT: Path to the Flash Operator Panel webroot (leave off trailing slash)
# FOPPASSWORD: Password for performing transfers and hangups in the Flash Operator Panel
# FOPRUN: Set to true if you want FOP started by freepbx_engine (amportal_start), false otherwise
# FOPDISABLE: Set to true to disable FOP in interface and retrieve_conf. Useful for sqlite3
# or if you don't want FOP.
#
FOPRUN=true
FOPWEBROOT=/var/www/html/asterisk/panel
FOPPASSWORD=asterisk

# FOPSORT=extension|lastname
# DEFAULT VALUE: extension
# FOP should sort extensions by Last Name [lastname] or by Extension [extension]

# This is the default admin name used to allow an administrator to login to ARI bypassing all security.
# Change this to whatever you want, don't forget to change the ARI_ADMIN_PASSWORD as well
ARI_ADMIN_USERNAME=admin

# This is the default admin password to allow an administrator to login to ARI bypassing all security.
# Change this to a secure password.
ARI_ADMIN_PASSWORD=ari_password

# AUTHTYPE=database|none
# Authentication type to use for web admininstration. If type set to 'database', the primary
# AMP admin credentials will be the AMPDBUSER/AMPDBPASS above.
AUTHTYPE=none

# AMPADMINLOGO=filename
# Defines the logo that is to be displayed at the TOP RIGHT of the admin screen. This enables
# you to customize the look of the administration screen.
# NOTE: images need to be saved in the ..../admin/images directory of your AMP install
# This image should be 55px in height
AMPADMINLOGO=logo.png

# USECATEGORIES=true|false
# DEFAULT VALUE: true
# Controls if the menu items in the admin interface are sorted by category (true), or sorted
# alphabetically with no categories shown (false).

# AMPEXTENSIONS=extensions|deviceanduser
# Sets the extension behavior in FreePBX. If set to 'extensions', Devices and Users are
# administered together as a unified Extension, and appear on a single page.
# If set to 'deviceanduser', Devices and Users will be administered seperately. Devices (e.g.
# each individual line on a SIP phone) and Users (e.g. '101') will be configured
# independent of each other, allowing association of one User to many Devices, or allowing
# Users to login and logout of Devices.
AMPEXTENSIONS=extensions

# ENABLECW=true|false
# DEFAULT VALUE: true
# Enable call waiting by default when an extension is created. Set to 'no' to if you don't want
# phones to be commissioned with call waiting already enabled. The user would then be required
# to dial the CW feature code (*70 default) to enable their phone. Most installations should leave
# this alone. It allows multi-line phones to receive multiple calls on their line appearances.

# CWINUSEBUSY=true|false
# DEFAULT VALUE: true
# For extensions that have CW enabled, report unanswered CW calls as 'busy' (resulting in busy
# voicemail greeting). If set to no, unanswered CW calls simply report as 'no-answer'.

# AMPBADNUMBER=true|false
# DEFAULT VALUE: true
# Generate the bad-number context which traps any bogus number or feature code and plays a
# message to the effect. If you use the Early Dial feature on some Grandstream phones, you
# will want to set this to false.

# AMPBACKUPSUDO=true|false
# DEFAULT VALUE: false
# This option allows you to use sudo when backing up files. Useful ONLY when using AMPPROVROOT
# Allows backup and restore of files specified in AMPPROVROOT, based on permissions in /etc/sudoers
# for example, adding the following to sudoers would allow the user asterisk to run tar on ANY file
# on the system:
# asterisk localhost=(root)NOPASSWD: /bin/tar
# Defaults:asterisk !requiretty
# PLEASE KEEP IN MIND THE SECURITY RISKS INVOLVED IN ALLOWING THE ASTERISK USER TO TAR/UNTAR ANY FILE

# CUSTOMASERROR=true|false
# DEFAULT VALUE: true
# If false, then the Destination Registry will not report unknown destinations as errors. This should be
# left to the default true and custom destinations should be moved into the new custom apps registry.

# DYNAMICHINTS=true|false
# DEFAULT VALUE: false
# If true, Core will not statically generate hints, but instead make a call to the AMPBIN php script,
# and generate_hints.php through an Asterisk's #exec call. This requires Asterisk.conf to be configured
# with "execincludes=yes" set in the [options] section.

# XTNCONFLICTABORT=true|false
# BADDESTABORT=true|false
# DEFAULT VALUE: false
# Setting either of these to true will result in retrieve_conf aborting during a reload if an extension
# conflict is detected or a destination is detected. It is usually better to allow the reload to go
# through and then correct the problem but these can be set if a more strict behavior is desired.

# SERVERINTITLE=true|false
# DEFAULT VALUE: false
# Precede browser title with the server name.

# USEDEVSTATE = true|false
# DEFAULT VALUE: false
# If this is set, it assumes that you are running Asterisk 1.4 or higher and want to take advantage of the
# func_devstate.c backport available from Asterisk 1.6. This allows custom hints to be created to support
# BLF for server side feature codes such as daynight, followme, etc.

# MODULEADMINWGET=true|false
# DEFAULT VALUE: false
# Module Admin normally tries to get its online information through direct file open type calls to URLs that
# go back to the freepbx.org server. If it fails, typically because of content filters in firewalls that
# don't like the way PHP formats the requests, the code will fall back and try a wget to pull the information.
# This will often solve the problem. However, in such environment there can be a significant timeout before
# the failed file open calls to the URLs return and there are often 2-3 of these that occur. Setting this
# value will force FreePBX to avoid the attempt to open the URL and go straight to the wget calls.

# AMPDISABLELOG=true|false
# DEFAULT VALUE: true
# Whether or not to invoke the FreePBX log facility

# AMPSYSLOGLEVEL=LOG_EMERG|LOG_ALERT|LOG_CRIT|LOG_ERR|LOG_WARNING|LOG_NOTICE|LOG_INFO|LOG_DEBUG|LOG_SQL|SQL
# DEFAULT VALUE: LOG_ERR
# Where to log if enabled, SQL, LOG_SQL logs to old MySQL table, others are passed to syslog system to
# determine where to log

# AMPENABLEDEVELDEBUG=true|false
# DEFAULT VALUE: false
# Whether or not to include log messages marked as 'devel-debug' in the log system

# AMPMPG123=true|false
# DEFAULT VALUE: true
# When set to false, the old MoH behavior is adopted where MP3 files can be loaded and WAV files converted
# to MP3. The new default behavior assumes you have mpg123 loaded as well as sox and will convert MP3 files
# to WAV. This is highly recommended as MP3 files heavily tax the system and can cause instability on a busy
# phone system.

# CDR DB Settings: Only used if you don't use the default values provided by FreePBX.
# CDRDBHOST: hostname of db server if not the same as AMPDBHOST
# CDRDBPORT: Port number for db host
# CDRDBUSER: username to connect to db with if it's not the same as AMPDBUSER
# CDRDBPASS: password for connecting to db if it's not the same as AMPDBPASS
# CDRDBNAME: name of database used for cdr records
# CDRDBTYPE: mysql or postgres mysql is default
# CDRDBTABLENAME: Name of the table in the db where the cdr is stored cdr is default

# AMPVMUMASK=mask
# DEFAULT VALUE: 077
# Defaults to 077 allowing only the asterisk user to have any permission on VM files. If set to something
# like 007, it would allow the group to have permissions. This can be used if setting apache to a different
# user then asterisk, so that the apache user (and thus ARI) can have access to read/write/delete the
# voicemail files. If changed, some of the voicemail directory structures may have to be manually changed.

# DASHBOARD_STATS_UPDATE_TIME=integer_seconds
# DEFAULT VALUE: 6
# DASHBOARD_INFO_UPDATE_TIME=integer_seconds
# DEFAULT VALUE: 20
# These can be used to change the refresh rate of the System Status Panel. Most of
# the stats are updated based on the STATS interval but a few items are checked
# less frequently (such as Asterisk Uptime) based on the INFO value

# ZAP2DAHDICOMPAT=true|false
# DEFAULT VALUE: false
# If set to true, FreePBX will check if you have chan_dadhi installed. If so, it will
# automatically use all your ZAP configuration settings (devices and trunks) and
# silently convert them, under the covers, to DAHDI so no changes are needed. The
# GUI will continue to refer to these as ZAP but it will use the proper DAHDI channels.
# This will also keep Zap Channel DIDs working.

# CHECKREFERER=true|false
# DEFAULT VALUE: true
# When set to the default value of true, all requests into FreePBX that might possibly add/edit/delete
# settings will be validated to assure the request is coming from the server. This will protect the system
# from CSRF (cross site request forgery) attacks. It will have the effect of preventing legitimately entering
# URLs that could modify settings which can be allowed by changing this field to false.

# USEQUEUESTATE=true|false
# DEFAULT VALUE: false
# Setting this flag will generate the required dialplan to integrate with the following Asterisk patch:
# https://issues.asterisk.org/view.php?id=15168
# This feature is planned for a future 1.6 release but given the existence of the patch can be used prior. Once
# the release version is known, code will be added to automatically enable this format in versions of Asterisk
# that support it.

# USEGOOGLEDNSFORENUM=true|false
# DEFAULT VALUE: false
# Setting this flag will generate the required global variable so that enumlookup.agi will use Google DNS
# 8.8.8.8 when performing an ENUM lookup. Not all DNS deals with NAPTR record, but Google does. There is a
# drawback to this as Google tracks every lookup. If you are not comfortable with this, do not enable this
# setting. Please read Google FAQ about this: http://code.google.com/speed/public-dns/faq.html#privacy

# MOHDIR=subdirectory_name
# This is the subdirectory for the MoH files/directories which is located in ASTVARLIBDIR
# if not specified it will default to mohmp3 for backward compatibility.

# RELOADCONFIRM=true|false
# DEFAULT VALUE: true
# When set to false, will bypass the confirm on Reload Box

# FCBEEPONLY=true|false
# DEFAULT VALUE: false
# When set to true, a beep is played instead of confirmation message when activating/de-activating:
# CallForward, CallWaiting, DayNight, DoNotDisturb and FindMeFollow

# DISABLECUSTOMCONTEXTS=true|false
# DEFAULT VALUE: false
# Normally FreePBX auto-generates a custom context that may be usable for adding custom dialplan to modify the
# normal behavior of FreePBX. It takes a good understanding of how Asterisk processes these includes to use
# this and in many of the cases, there is no useful application. All includes will result in a WARNING in the
# Asterisk log if there is no context found to include though it results in no errors. If you know that you
# want the includes, you can set this to true. If you comment it out FreePBX will revert to legacy behavior
# and include the contexts.

AMPDBUSER=admin
AMPDBPASS=secret123password
AMPWEBADDRESS=10.0.2.15
AMPDBNAME=asterisk
ASTETCDIR=/etc/asterisk
ASTMODDIR=/usr/lib/asterisk/modules
ASTVARLIBDIR=/var/lib/asterisk
ASTAGIDIR=/var/lib/asterisk/agi-bin
ASTSPOOLDIR=/var/spool/asterisk
ASTRUNDIR=/var/run/asterisk
ASTLOGDIR=/var/log/asterisk

---- ahora el manager.conf

;
; AMI - Asterisk Manager interface
;
; FreePBX needs this to be enabled. Note that if you enable it on a different IP, you need
; to assure that this can't be reached from un-authorized hosts with the ACL settings (permit/deny).
; Also, remember to configure non-default port or IP-addresses in amportal.conf.
;
; The AMI connection is used both by the portal and the operator's panel in FreePBX.
;
; FreePBX assumes an AMI connection to localhost:5038 by default.
;
[general]
enabled = yes
port = 5038
bindaddr = 0.0.0.0
displayconnects=no ;only effects 1.6+

[admin]
secret = admin
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,config,command,dtmf,reporting,cdr,dialplan,originate
write = system,call,log,verbose,command,agent,user,config,command,dtmf,reporting,cdr,dialplan,originate

#include manager_additional.conf
#include manager_custom.conf

-------en el modules.conf no mo difique nada

y ahora ya no me sale el errro anterios sino me sale este error:

http://img814.imageshack.us/f/error2qw.jpg/

---ojala me puedan decir como puedo resolverlo.