[asterisk-users] DTMF forwarding and Page [SOLVED] [PATCH 1/1]

Matteo Fortini matteo.fortini at gmail.com
Mon Feb 13 09:37:26 CST 2012


Nevermind,
I checked the code, and A* is not using the "F" option in MeetMe for 
Page(), so it's not working by default.
Attached is a patch which fixes the problem for me, if anyone needs it.

Matteo

Il 11/02/2012 13:53, Matteo Fortini ha scritto:
> Noone knows that? Where/whom could I ask?
>
> Thanks
>
> Il 10/02/2012 12:30, Matteo Fortini ha scritto:
>> Hi,
>> I'd like to implement some way of controlling remote SIP clients 
>> while in a call, to execute remote commands.
>>
>> The call topology (think of a PA system) is this:
>> * the caller is in a MeetMe() conference room
>> * the callees are Page()d, then the dynamic conference room is 
>> connected to the previous one
>>
>> I'm wondering if Asterisk is relaying DTMF (SIP info or RTP) from the 
>> caller to the callees. I found option 'F' for MeetMe, but I have no 
>> control on Page().
>>
>> TIA,
>> Matteo
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