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Подскажите как реализовать такое на *

Сообщений: 13

Re: Подскажите как реализовать такое на *

есть разница и она существенна ? не расскажете ?
2006-09-07 17:42

Сообщений: 6521

Re: Подскажите как реализовать такое на *

Я, я специалист по @home, FreePBX & TrixBox!
Всё управление - через вэб интерфейс, никаких копаний в конф файлах сначала. Потом, с опытом, можно осторожно ваять в дополнительных custom файлах.
2006-09-07 19:03

Откуда: Санкт-Петербург
Сообщений: 541

Re: Подскажите как реализовать такое на *

А теперь ищи где там установлен apache и по конфигам -
выяснишь URL веб-морды. Про забытые пароли - должно быть написано в @home handbook на voip-info.org
2006-09-07 19:04

Сообщений: 6521

Re: Подскажите как реализовать такое на *

..командой help-aah из шела увидишь всё, что можно изменить (или не изменить).
2006-09-07 19:26

Сообщений: 13

Re: Подскажите как реализовать такое на *

ded
доступ к веб-морде есть, там:
Welcome to the Asterisk Management Portal 1.10.010

я уже себя добавил в Extensions и могу звонить и принимать звонки с помощью софтфона SJPhone.

Только насколько я понял с просмотра меню AMP:

Incoming Calls
Extensions
Ring Groups
Queues
Digital Receptionist
Trunks
Inbound Routing
Outbound Routing
On Hold Music
System Recordings
Backup & Restore
General Settings

моя задачка с его помощью не решится.. поэтому спрашивал про конф.файлы и собственно сам подход... К сожалению так ничего и не нашел по команде Dial
в астерисковском шеле, который вызываю через asterisk -r, тоже такой команды нету..

кстати .. скачал VMWare образ Trixbox-1.1.. думаю на нем потренироваться.. связать их вместе чтоли... если получится конечно..
2006-09-07 21:01

Сообщений: 6521

Re: Подскажите как реализовать такое на *

Trixbox - это тот же АтХоме, только более продвинутая версия. Не надо их связывать. Тебе надо смотреть как сооружен транк на вашего оператора. Далее: изучить команду Dial со всеми опциями, и соорудить потом custom trunk руководствуясь получеными знаниями..
2006-09-07 22:05

Сообщений: 6521

Re: Подскажите как реализовать такое на *

После чего перевести весь Outbound Routing на твой новый транк.
2006-09-07 22:16

Сообщений: 6521

Re: Подскажите как реализовать такое на *

asterisk1*CLI> show application Dial
asterisk1*CLI>
-= Info about application 'Dial' =-

[Synopsis]
Place a call and connect to the current channel

[Description]
Dial(Technology/resource[&Tech2/resource2...][|timeout][|options][|URL]):
This applicaiton will place calls to one or more specified channels. As soon
as one of the requested channels answers, the originating channel will be
answered, if it has not already been answered. These two channels will then
be active in a bridged call. All other channels that were requested will then
be hung up.
Unless there is a timeout specified, the Dial application will wait
indefinitely until one of the called channels answers, the user hangs up, or
if all of the called channels are busy or unavailable. Dialplan executing will
continue if no requested channels can be called, or if the timeout expires.

This application sets the following channel variables upon completion:
DIALEDTIME - This is the time from dialing a channel until when it
is disconnected.
ANSWEREDTIME - This is the amount of time for actual call.
DIALSTATUS - This is the status of the call:
CHANUNAVAIL | CONGESTION | NOANSWER | BUSY | ANSWER | CANCEL
DONTCALL | TORTURE
For the Privacy and Screening Modes, the DIALSTATUS variable will be set to
DONTCALL if the called party chooses to send the calling party to the 'Go Away'
script. The DIALSTATUS variable will be set to TORTURE if the called party
wants to send the caller to the 'torture' script.
This application will report normal termination if the originating channel
hangs up, or if the call is bridged and either of the parties in the bridge
ends the call.
The optional URL will be sent to the called party if the channel supports it.
If the OUTBOUND_GROUP variable is set, all peer channels created by this
application will be put into that group (as in Set(GROUP()=...).

Options:
A(x) - Play an announcement to the called party, using 'x' as the file.
C - Reset the CDR for this call.
d - Allow the calling user to dial a 1 digit extension while waiting for
a call to be answered. Exit to that extension if it exists in the
current context, or the context defined in the EXITCONTEXT variable,
if it exists.
D([called][:calling]) - Send the specified DTMF strings *after* the called
party has answered, but before the call gets bridged. The 'called'
DTMF string is sent to the called party, and the 'calling' DTMF
string is sent to the calling party. Both parameters can be used
alone.
f - Force the callerid of the *calling* channel to be set as the
extension associated with the channel using a dialplan 'hint'.
For example, some PSTNs do not allow CallerID to be set to anything
other than the number assigned to the caller.
g - Proceed with dialplan execution at the current extension if the
destination channel hangs up.
G(context^exten^pri) - If the call is answered, transfer the calling party to
the specified priority and the called party to the specified priority+1.
Optionally, an extension, or extension and context may be specified.
Otherwise, the current extension is used.
h - Allow the called party to hang up by sending the '*' DTMF digit.
H - Allow the calling party to hang up by hitting the '*' DTMF digit.
j - Jump to priority n+101 if all of the requested channels were busy.
L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are
left. Repeat the warning every 'z' ms. The following special
variables can be used with this option:
* LIMIT_PLAYAUDIO_CALLER yes|no (default yes)
Play sounds to the caller.
* LIMIT_PLAYAUDIO_CALLEE yes|no
Play sounds to the callee.
* LIMIT_TIMEOUT_FILE File to play when time is up.
* LIMIT_CONNECT_FILE File to play when call begins.
* LIMIT_WARNING_FILE File to play as warning if 'y' is defined.
The default is to say the time remaining.
m([class]) - Provide hold music to the calling party until a requested
channel answers. A specific MusicOnHold class can be
specified.
M(x[^arg]) - Execute the Macro for the *called* channel before connecting
to the calling channel. Arguments can be specified to the Macro
using '^' as a delimeter. The Macro can set the variable
MACRO_RESULT to specify the following actions after the Macro is
finished executing.
* ABORT Hangup both legs of the call.
* CONGESTION Behave as if line congestion was encountered.
* BUSY Behave as if a busy signal was encountered. This will also
have the application jump to priority n+101 if the
'j' option is set.
* CONTINUE Hangup the called party and allow the calling party
to continue dialplan execution at the next priority.
* GOTO:<context>^<exten>^<priority> - Transfer the call to the
specified priority. Optionally, an extension, or
extension and priority can be specified.
n - This option is a modifier for the screen/privacy mode. It specifies
that no introductions are to be saved in the priv-callerintros
directory.
N - This option is a modifier for the screen/privacy mode. It specifies
that if callerID is present, do not screen the call.
o - Specify that the CallerID that was present on the *calling* channel
be set as the CallerID on the *called* channel. This was the
behavior of Asterisk 1.0 and earlier.
p - This option enables screening mode. This is basically Privacy mode
without memory.
P([x]) - Enable privacy mode. Use 'x' as the family/key in the database if
it is provided. The current extension is used if a database
family/key is not specified.
r - Indicate ringing to the calling party. Pass no audio to the calling
party until the called channel has answered.
S(x) - Hang up the call after 'x' seconds *after* the called party has
answered the call.
t - Allow the called party to transfer the calling party by sending the
DTMF sequence defined in features.conf.
T - Allow the calling party to transfer the called party by sending the
DTMF sequence defined in features.conf.
w - Allow the called party to enable recording of the call by sending
the DTMF sequence defined for one-touch recording in features.conf.
W - Allow the calling party to enable recording of the call by sending
the DTMF sequence defined for one-touch recording in features.conf.

asterisk1*CLI>
2006-09-07 22:30

Сообщений: 13

Re: Подскажите как реализовать такое на *

ded
Спасибо. Читаю...

Кстати,
ded..командой help-aah из шела увидишь всё, что можно изменить (или не изменить).

в * шелле такой команды вроде нету..или может у меня не получается..
2006-09-08 11:44

Сообщений: 6521

Re: Подскажите как реализовать такое на *

ramz
в * шелле такой команды вроде нету..или может у меня не получается..
ты из-под рута?
когда логинишься через ssh дает такой motd -

Welcome to Asterisk@Home
-------------------------------------------------

For access to the Asterisk@Home web GUI use this URL
http://10.11.12.13

Может это и не @home?
[root@localhost]#uname -a ?


2006-09-08 14:30

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