[asterisk-biz] SIT Tone Detection

[Brett List]

My SIP Provider (A large major carrier) passes me calls this way. And no,
they won't disable early media and provide an appropriate supervision code.

In fact, all carriers we've tested have transmitted the call this way. Even
my cell phone gives me the tri-tone without "connecting"

Yes, chan_sip would need to be modifed. That is the work I am requesting.

For what it's worth, I've attempted to modify chan_sip in the sip_rtp_read
func to include adding a dsp with ast_dsp_new, converting to SLINEAR with
ast_set_read_format and then passing it to ast_dsp_call_progress. But it
doesn't seem to be doing anything at all.. I've been performing my test on
G.711.


Thanks for your reply!
-Brett



On 1/17/07, Leo Ann Boon <leo at datvoiz.com> wrote:
>
> Brett List wrote:
> > Hi All,
> > I have a specific Need to detect SIT Tones is SIP RTP Early Media. I
> > specifically need to be able to tell the difference between a RNA and
> > a SIT.
> It should be a function of your gateway. chap_sip doesn't support analog
> call progress. And, since you're using G.729 it could be rather
> difficult to detect the tri-tone reliability all the time.
>
> Leo
>
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