Jono wrote:
> Paul Hayes wrote on 17/07/2007 :
>> Jono wrote:
>>> Gordon Henderson explained on 15/07/2007 :
>>>
>>>>
>>>> You need both t and T in the Dial() command in the dialplan.
>>>>
>>>> I've no idea how you implement this in Trixbox though.
>>>>
>>>> Gordon
>>>
>>> Yes, thanks. I had considered that, however, any IVRs that require a
>>> # to be dialled result in the transfer process commencing.
>>>
>>> I wish the Siemens' R key worked.
>>>
>>>
>>
>> The Siemens "R" key does work, Asterisk doesn't know how to handle the
>> flash-hook signal that the R key sends.
>
> Righty-Oh. Perhaps I could've worded what I said slight differently: "I
> wish the Siemens' R key worked in a way that Asterisk understood"
>
>> The actual signal depends on your selected DTMF method, rfc2833 sends
>> an RTP DTMF Event message with a value of 16. SIP Info sends a SIP
>> INFO message with event value 16 but you can change this in the
>> Siemens config because there is currently no standardised way of
>> sending a flash hook in SIP Info.
>
> Paul, ....if I read you correctly, the R key /will/ work with Asterisk,
> if the Siemens config is set correctly.......I'm fairly sure I tried all
> three choices, individually & combined, however, I don't think I used
> all possible combinations....
>
>

I don't think so at the moment. I've been searching around for a while
for a way to get Asterisk to act on receiving either a SIP INFO Hook
Flash message or (probably more likely) an RFC2833 RTP Hook Flash
message. Sadly there doesn't seem to be anything, all I can find is
articles relating to generating/reading hook flashes through analogue
channels, not SIP. I guess Asterisk developers just expect the SIP
phone to deal with transfers. If I find anything else out I'll post it
here.

cheers,
Paul.

--
Working Email:

paul-at-polog40-dot-org-dot-uk