[asterisk-dev] SIP channel Indicate AST_CONTROL_FLASH support

[Raj Jain]

On Dec 28, 2007 6:22 AM, Kamanashis Roy Shuva <kamanashisroy at gmail.com>
wrote:

> Hi,
>
>   I am expecting someone to discuss this who have at least a little
> idea about sip actually.



You're already discussing this w/ Olle J., who is lead developer and a SIP
expert in the Asterisk community :-)

I guess the source of the confusion is that sometimes RFC 2833 is used
synonymously w/ DTMF, because typically many end-points support only the
DTMF portion (0-11) of the RFC. Asterisk supports 0-16 and negotiates that
range in the SDP if these events are expected to be conveyed through RTP.

With SIP INFO there is no negotiation of the range on what the participating
end-points support so all bets are off (that's why using SIP INFO for
telephony events is a bad idea).

--
Raj



> On Dec 28, 2007 5:20 PM, Kamanashis Roy Shuva <kamanashisroy at gmail.com>
> wrote:
> > Hi,
> >
> > Let me copy some code from chan_sip.c
> >
> > I am reading a function body
> >
> >  /*! \brief  Receive SIP INFO Message
> > \note    Doesn't read the duration of the DTMF signal */
> > static void handle_request_info(struct sip_pvt *p, struct sip_request
> *req)
> >
> > ....
> > ....
> >
> >         if (event == 16) {
> >             /* send a FLASH event */
> >             struct ast_frame f = { AST_FRAME_CONTROL, AST_CONTROL_FLASH,
> };
> >             ast_queue_frame(p->owner, &f);
> >             if (sipdebug)
> >                 ast_verbose("* DTMF-relay event received: FLASH\n");
> >         } else {
> >
> > .....
> > .....
> >
> > The above code surely works with info message where the message
> > body consists of signal value. And that value is matched to 16. This
> > is enough evidence that asterisk supports flash when it receives
> > info message.
> >
> >
> > On Dec 28, 2007 4:41 PM, Johansson Olle E <oej at edvina.net> wrote:
> > >
> > > 27 dec 2007 kl. 19.54 skrev Kamanashis Roy Shuva:
> > >
> > > > Hi,
> > > >
> > > > The patch is not successful adding this feature. I think there is
> more
> > > > to be added. Or the patch should follow a good design. I mean I am
> not
> > > > satisfied with this patch anyway.
> > > >
> > > > Here I have found the flash support for sip.
> > > >
> > > > http://www.voip-info.org/wiki/view/SIP+DTMF+Signalling
> > > >
> > > >
> > > This page is describing RFC 2833 DTMF.
> >
> > I think you should have a better look. Please recheck ..
> >
> > >
> > > > Again, asterisk can accept INFO dtmf flash .. I mean it is in the
> code
> > > > that when there is
> > > > a info message with "signal=16" it indicates flash.
> > > >
> > > > I hope we will be discussing this upto a solution.
> > >
> > > Cisco docs indicate that they only support hook flash in the RTP
> > > stream. I have not seen any support
> > > of hook flash in SIP INFO.
> > >
> > >
> > > /O
> > >
> > > _______________________________________________
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> > >
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> > >
> >
> >
> >
> > --
> > -- Thanks
> >
> > Kamanashis Roy
> >
>
>
>
> --
> -- Thanks
>
> Kamanashis Roy
>
> _______________________________________________
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