asterisk users - Mar 2005 - FW: (still problems) Dialing phone number and extension together to avoid listening to voice menu (incoming call)

Thanks a lot for all the suggestions!

Unfortunately, it still gives problems.

Most common error message is "ast_realaudio_callback Failed to write
frame" after "paying the beep". Then it says "User
disconnected".

Also, it doesn't react to any extension entered and doesn't do any
forwarding (as it should in "exten =>
_XXX.,6,Macro(fhostaff,${mynumber},SIP/${mynumber})"


Here's my complete context:

[fhostaffmenu]
;include => fhostaff

exten => s,1,Ringing			; Ring
exten => s,2,Wait(2)			; Give them 2 seconds of ringing
exten => s,3,Answer			; Answer the line
exten => s,4,DigitTimeout(3)		; Set Digit Timeout to 3 seconds
exten => s,5,ResponseTimeout(10)		; Set Response Timeout
to 10 seconds
exten => s,n,Read(mynumber|beep|3)
exten => _XXX.,6,Macro(fhostaff,${mynumber},SIP/${mynumber})
exten
=>i,1,playback(some_file_to_say_you_entered_an_invalid_extension_try_aga
in)
exten => i,8,Goto(s,4)
exten
=>t,1,playback(some_file_to_say_you_did_not_enter_an_extension_try_again
)
exten => t,8,Goto(s,4)


I've tried different dtmf modes, turning of "silence" on the
client
(SJPhone), etc, but nothing seems to help.

Any ideas?
Please help me to figure it out.

I'm using an around 2 weeks old CVS version.



Isn't there an easy way to redirect a call coming from a provider
(sipgate.de) to one of the extensions based on the last 3 numbers a
caller entered?

By the way, I'm using RealTime (mysql) for sip (just users) and
extensions. Can it pose a problem?


Thank you very much,
Roman Zhovtulya




-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Adam
Goryachev
Sent: Donnerstag, 3. M?rz 2005 15:05
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] (another try) Dialing phone number
andextension together to avoid listening to voice menu (incoming call)

> ***************
> ; defining the voice menu for incoming calls:
> 
> [fhostaffmenu]
> exten => s,1,Ringing			; Make them comfortable with
> some seconds of ringback
> exten => s,2,Answer			; Answer the line
You haven't actually given them any ringing, you need to add this: exten
=> s,3,wait(2) ; Give them 2 seconds of ringing
> exten => s,4,DigitTimeout(1)		; Set Digit Timeout to 5 seconds
> exten => s,5,ResponseTimeout(3)	; Set Response Timeout to 10
seconds

Rather than doing the below, if you simply stop all processing at this
point, and don't have any more extensions, then asterisk will wait 3
seconds for the user to press a number, then 1 second for each extra
number. When they don't press a number for more than the 1 second, or
asterisk matches an extension, then it will try to dial the entered
number.
> exten => s,5,Read(mynumber,beep,3)  ; Read DTMF input and save it into
> "mynumber" variable exten =>
> s,6,Macro(fhostaff,${mynumber},SIP/${mynumber}) ; dial the extension 
> that is saved in "mynumber"
> ***************
OK, hard to get asterisk to do this, but something like:
exten => _XXX.,Macro(fhostaff,${mynumber},SIP/${mynumber})

So, the user can dial 3 or more digits, and then it will go to your
macro.

You can also add:
exten =>
i,1,playback(some_file_to_say_you_entered_an_invalid_extension_try_again
)
exten => i,2,Goto(s,4)

and also:
exten =>
t,1,playback(some_file_to_say_you_did_not_enter_an_extension_try_again)
exten => t,2,Goto(s,4)

I hope that helps you...

Regards,
Adam

-- 
 -- 
Adam Goryachev
Website Managers
Ph:  +61 2 9345 4395                        adam@websitemanagers.com.au
Fax: +61 2 9345 4396                        www.websitemanagers.com.au

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