[Serusers] SIP INFO method

[Nils Ohlmeier]

Hi,

On Friday 01 July 2005 13:34, Iqbal wrote:
> Not really sure what I want to do with it as yet :-), but was looking
> into the grandstream 102 phones, and they can send DTMF via the INFO
> method, which I think helps with call transfers, and then that got me

how  does SIP INFO helps with call transfer? If I would program a UA (which I 
actually do), I would never submit any tones, no matter if in-band audio, 
RFC2833 RTP or SIP INFO, to the remote side while the user enters a transfer 
target.

  Nils

> thinking that if it could do this, then surely I could use the DTMF to
> make ser carry out a few more tasks, that initially I was going to pass
> into asterisk and let it do. Not sure what the downside is, except what
> you mentioned.
>
> Iqbal
>
> Zeus Ng wrote:
> >Iqbal,
> >
> >SIP is supposed to be extensible. So long as the end point understand the
> >method, it is allowed. A proxy should relay method it does not understand.
> >The INFO method is defined in RFC and so it's allowed.
> >
> >Seems like you are trying to extend SER beyond a proxy server. As long as
> >you have the right module, the thing you are asking is doable. However, I
> >strongly against this. How would you feel if the DTMF was supposed to be
> >received by the bank application and somehow SER intercept it and use it
> > for call routing?
> >
> >
> >Zeus
> >
> >Iqbal wrote:
> >>is this a "allowed" method if so has anyone used it to do
> >>clever things with dtmf digit handling, things like executing
> >>a script to amend call forwarding rules etc.
> >
> >.
>
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