[asterisk-users] menu issue

[salaheddine elharit]

Hi khalid

thank you for your response i have verify the chapter that talks about
contexts and also menu in asterisk book
and now i can create my menu without issue,if i have any question i will
send an email to this perfect list

thanks evrey one for the help and support

Kind Regards



2011/6/21 khalid touati <khalidtouati at gmail.com>

> Hi I wanted to help out with dial plan, but it's not obvious what you want
> to achieve, also I do recommend to read the chapter that talks about
> contexts and dialplan from future of asterisk book. but if you're in rush
> just try to make clear how you want your system to behave and i'll be glad
> to help.
>
>
> On Tue, Jun 21, 2011 at 9:22 AM, salaheddine elharit <
> salah.elharit200 at gmail.com> wrote:
>
>>  Hello
>>
>>
>>
>> I have created the menu below, with this menu when I call 520460XXX I can
>> hear the welcome message [home] context and when I press the # I can go to
>> the [menu] context and hear the menu message
>>
>>
>>
>> When I press 1 in order to go to the [call] I can hear the call message
>>
>>
>> Now I have 2 client sip 222 and 223 with x-lite, how can I do in order to
>> receive the call in 223 with [call] and receive a call in 222 with [support]
>>
>> thanks and best regards
>>
>>
>> exten => 520460XXX,1,Ringing()
>> exten => 520460XXX,2,Wait(4)
>> exten => 520460XXX,3,Goto(home,s,1)
>>
>> [home]
>> exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
>> exten => s,2,Background(${sounds_path}welcome)
>> exten => s,3,WaitExten(10)
>>
>> exten => #,1,Goto(menu,s,1)
>> exten => i,1,Playback(${sounds_path}error-key)
>> exten => t,1,Goto(home,s,1)
>>
>> [menu]
>> exten => s,1,Background(${sounds_path}menu)
>>
>> exten => 0,1,Goto(menu,s,1)
>> exten => 1,1,Goto(call,s,1)
>> exten => 2,1,Goto(support,s,1)
>>
>> exten => i,1,Playback(${sounds_path}error-key)
>> exten => i,2,Goto(call,s,1)
>> exten => t,1,Goto(call,s,1)
>> [call]
>> exten => s,1,Background(${sounds_path}call)
>>
>> exten => 0,1,Goto(menu,s,1)
>> exten => 223,1,Dial(SIP/${EXTEN},20,tr)
>>
>> exten => i,1,Playback(${sounds_path}error-key)
>> exten => t,1,Goto(appel,s,1)
>>
>>
>> [support]
>> exten => s,1,GoToIfTime(09:00-17:00|mon-fri|*|*?s,4)
>> exten => s,2,Playback(${sounds_path}no-relation-support)
>> exten => s,3,Goto(menu,s,1)
>> exten => s,4,Playback(${sounds_path}relation-support)
>> exten => s,5,Queue(default)
>> exten => t,1,Hangup()
>>
>>
>>  2011/6/20 Warren Selby <wcselby at selbytech.com>
>>
>>>   On Mon, Jun 20, 2011 at 12:17 PM, salaheddine elharit <
>>> salah.elharit200 at gmail.com> wrote:
>>>
>>>> [home]
>>>> exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
>>>> exten => s,2,Background(${sounds_path}welcome)
>>>> exten => #,1,Goto(menu,s,1)
>>>> exten => i,1,Playback(${sounds_path}error-key)
>>>> exten => t,1,Goto(home,s,1)
>>>>
>>>>
>>> You need to add the following to the [home] context:
>>>
>>> exten => s,3,WaitExten(10)
>>>
>>> which will cause the call to wait 10 seconds for input, otherwise it will
>>> timeout and go to the 't' extension.  The way you currently have it, the
>>> call will end after the Background() app finishes playing because it has no
>>> additional steps and nothing that will tell it to go to the 't' extension.
>>>
>>> Also, consider switching your dialplan priorities away from "1,2,3..."
>>> and go to "1,n,n,n..." as this reduces headaches in the longrun.
>>>
>>> --
>>> Thanks,
>>> --Warren Selby, dCAP
>>> http://www.SelbyTech.com <http://www.selbytech.com/>
>>>
>>>
>>> --
>>> _____________________________________________________________________
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>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>               http://www.asterisk.org/hello
>>
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
> Abdullah
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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