[AsteriskBrasil] FOP
Duany Faustino
duany em grupolacerda.com.br
Quarta Abril 4 10:12:02 BRT 2007
Caro Bernardo eu instalei o FOP sosinho no meu asterisk que roda em um
Debiam
no meu manager.conf esta com o usuario e senha correto
manager.conf
[general]
enabled = yes
webenabled = yes
httptimeout = 600
displaysystemname = yes
port = 5038
bindaddr = 0.0.0.0
displayconnects = yes
[duany]
secret = SENHA
;deny=0.0.0.0/0.0.0.0
permit=0.0.0.0/0.0.0.0
op_server.cfg
[general]
; host or ip address of asterisk
manager_host=192.168.200.182
manager_port=5038
; user and secret for connecting to * manager
manager_user=duany
manager_secret=df08072005
; The optional event_mask for filtering manager events.
; Asterisk will send only the events you request
; with this parameter. Possible values are:
; on, off, system, call, log, verbose
event_mask=call
;
; You can specify many asterisk servers to
; monitor. Just repeat the manager_host, manager_user
; and manager_secret parameters in order. The first
; one will be server number 1, and so on.
;
; manager_host=1.2.3.4
; manager_user=john
; manager_secret=doe
; Enable MD5 auth to Asterisk manager
auth_md5=1
; you can use astmanproxy, if you enable it, all of the above
; connections and settings will be overriden. You have to define
; the host and port
; astmanproxy_host = 127.0.0.1
; astmanproxy_port = 1234
; You will also have to define the servers that are monitored trough
; astmanproxy, you have to enumerate them using the astmanproxy_server.
; astmanproxy_server = 192.168.10.1
; astmanproxy_server = 192.168.10.2
; astmanproxy_server = 192.168.10.3
;
; ip address to listen for inbound connections, default all
listen_addr=127.0.0.1
; port to listen for inbound flash connections, default 4445
listen_port=4445
; hostname or ip address used to connect to the webserver where
; the flash movie resides (just the hostname, without directories)
; This value might be omited. In that case the flash movie will
; try to connect to the same host as the web page.
web_hostname=voip
; location of the .swf file in your disk (must reside somewhere
; inside your web root)
flash_dir=/var/www/panel
; secret code for performing hangups and transfers
security_code=matarokacha
;dkd4393kld
; Frequency in second to poll for sip and iax status
poll_interval=120
; Poll for voicemail status (only necesary when you access the
; voicemail directories outside asterisk itself - eg. web access)
poll_voicemail=0
; 1 Enable automatic hangup of zombies
; 0 Disable
kill_zombies=0
; Debug level to stdout (bitmap)
; 1 Manager Events Received
; 2 Manager Commands Sent
; 4 Show Flash events Received
; 8 Show events sent to Flash Clients
; 16 Server 1st Debug Level
; 32 Server 2nd Debug Level
; 64 Server 3rd Debug Level
;
; Eg: to display manager events and
; commands sent set it to 3 (1+2)
;
; Maximum debug level 255
debug=0
; Default language to use (op_lang_XX.cfg file)
language=en
; Context in your diaplan where you have the conferences for barge in
; Example:
;
; meetme.conf
; [rooms]
; conf => 900
; conf => 901
; conf => 902
;
; extensions.conf
; [conferences]
; exten => 900,1,MeetMe(900)
; exten => 901,1,MeetMe(901)
; exten => 902,1,MeetMe(902)
conference_context=conferences
; Meetme room numbers to use for barge in. The room number must match
; the extension number (see above).
barge_rooms=900-902
; When doing barge ins, you can make the 3rd party to start
; the meetme muted, so the other parties wont notice they are
; now being monitored
barge_muted=1
; Formatting of the callerid field
; where 'x' is a number
clid_format=(xxx)xxx-xxxx
; If you want not to show the callerid on the buttons, set this
; to one
clid_privacy=0
; To display the ip address of sip or iax peer inside the button
; set this to 1
show_ip=0
; It will hide queue position buttons and show only the active ones
queue_hide=0
; Will change the button label on AgentLogin
rename_label_agentlogin=0
; Will change the button label on Agentcallbacklogin
rename_label_callbacklogin=0
; Will rename the label for a wildcard button
rename_label_wildcard=0
; Will rename to the name specified in agents.conf
; If disabled the renaming will be Agent/XXXX
rename_to_agent_name=1
; Will display IDLE time for agents, as well as
; update the queue status after an agent hangs up
; the call, so you don't need to reload to get
; queue statistics
agent_status=0
; Will rename labels for queuemembers
; If you use addqueuemember in your dialplan, you
; can fake an AgengLogin event by sending it with
; the UserEvent application. Eg:
;
; exten => 25,1,AddQueueMember(sales|SIP/${CALLERIDNUM}
; exten => 25,2,UserEvent(Agentlogin|Agent: ${CALLERIDNUM});
; exten => 25,3,Answer
; exten => 25,4,Playback(added-to-sales-queue)
; exten => 25,5,Hangup
;
; exten => 26,1,RemoveQueueMember(sales|SIP/${CALLERIDNUM})
; exten => 26,2,UserEvent(RefreshQueue);
; exten => 26,3,Answer
; exten => 26,4,Playback(removed-from-sales-queue)
; exten => 26,5,Hangup
rename_queue_member=0
; Will change the led color when the agent logs in
; The color is configurable in op_style.cfg
change_led_agent=1
; If set to 1, you will transfer the linked channel instead
; of the current channel when you drag the icon on a button
reverse_transfer=0
; If enabled, it will not ask forthe security code
; when performing a click to dial
; Will change the led color when the agent logs in
; The color is configurable in op_style.cfg
change_led_agent=1
; If set to 1, you will transfer the linked channel instead
; of the current channel when you drag the icon on a button
reverse_transfer=0
; If enabled, it will not ask forthe security code
; when performing a click to dial
clicktodial_insecure=1
; Enable select box with absolutetimeout for the call after
; a transfer is performed within the panel
transfer_timeout= "0,No timeout|300,5 minutes|600,10 minutes|1200,20
minutes|2400,40 minutes|3000,50 minutes"
; If set to 1, when hitting the reload button on the flash
; client it will instead restart the 1st asterisk box
; (For asterisk to restart you have to start it with
; safe_asterisk, if you dont do that, asterisk will just
; shut down)
enable_restart = 0
; If you set this parameter to your voicemailmain
; extension em context, it will originate a call to
; voicemailmain when double clicking on the MWI icon
; for any button.
voicemail_extension = 3000 em features
; You can have panel contexts with their own
; button layout and configuration. The following entry
; will create a context called sip with a different
; security code. In the online documentation you will
; find how to use contexts
;
;[sip]
;security_code=djdjdi43
;web_hostname=www.virtualwebserver.com
;flash_dir=/var/www/virtualwebserver/html/panel
;barge_rooms=800-802
;conference_context=otherconferences
;transfer_timeout="0,No timeout|60,1 minute"
;voicemail_extension=1000 em nine
;language=es
o iptables nao ta confiugrado e ele esta em um link direto
Grato pela atenção
----- Original Message -----
From: "Bernardo Vieira" <bernardo.vieira em terra.com.br>
To: <asteriskbrasil em listas.asteriskbrasil.org>
Sent: Wednesday, April 04, 2007 9:54 AM
Subject: Re: [AsteriskBrasil] FOP
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Compartilha um pouco mais de informação aí porque só o placar num vai
adiantar muito não.
- - Descrição do problema?
- - FOP + FreePbx ou FOP sozinho?
- - /etc/asterisk/manager.conf
- - /etc/amportal.conf (se for FreePBX)
- - /var/www/html/panel/op_server.cfg
- - ifconfig
- - iptables-save | grep INPUT (do seu servidor *)
- - iptables-save | grep FORWARD (do seu gateway, se o * estiver atrás
de nat)
acho que aí já dá.
Duany Faustino escreveu:
> Pessoal alguem tem uma luz pra me dar pois ja ta de 10000 X 0 pro FOP e
> eu não consegui fazer funcionar eu to com o * 1.2 e uso o Debian
>
>
> Grato
>
> Duany Faustino
> ----------------------------------------
> Estação VoIP 2006
> 5 e 6 Dezembro
> Curitiba PR
> http://www.estacaovoip.com.br
>
> _______________________________________________
> LIsta de discussões AsteriskBrasil.org
> AsteriskBrasil em listas.asteriskbrasil.org
> http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil
>
> _______________________________________________
> Acesse o wiki AsteriskBrasil.org:
> http://www.asteriskbrasil.org
>
>
- --
"What most profoundly divides two men is a different sense and degree of
cleanliness. What help is all honesty and mutual utility, what help is
all the good will for each other: in the end the fact remains-they can't
stand each other?s smell!"
- - Nietzsche
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----------------------------------------
Estação VoIP 2006
5 e 6 Dezembro
Curitiba PR
http://www.estacaovoip.com.br
_______________________________________________
LIsta de discussões AsteriskBrasil.org
AsteriskBrasil em listas.asteriskbrasil.org
http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil
_______________________________________________
Acesse o wiki AsteriskBrasil.org:
http://www.asteriskbrasil.org
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