<div>Well, it appears that the all to sip_rtp_read dies when I call ast_read(p->owner) for some reason..</div>
<div> </div>
<div>See what I don't get is that when I call ast_set_read_format, I give it a channel and a format.</div>
<div> </div>
<div>when I call ast_dsp_call_progress I give it a DSP and a ast_frame</div>
<div> </div>
<div>what makes the ast_frame be in the right format?</div>
<div> </div>
<div>needless to say, I get:</div>
<div>dsp.c:1154 ast_dsp_call_progress: Can only check call progress in signed-linear frames<br> </div>
<div>any ideas?</div>
<div><br><br> </div>
<div><span class="gmail_quote">On 1/17/07, <b class="gmail_sendername">Leo Ann Boon</b> <<a href="mailto:leo@datvoiz.com">leo@datvoiz.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Brett List wrote:<br>> My SIP Provider (A large major carrier) passes me calls this way. And<br>> no, they won't disable early media and provide an appropriate
<br>> supervision code.<br>><br>> In fact, all carriers we've tested have transmitted the call this way.<br>> Even my cell phone gives me the tri-tone without "connecting"<br>In my part of the world, the telcos will drop the call immediately and
<br>return the cause code.<br><br>><br>> Yes, chan_sip would need to be modifed. That is the work I am requesting.<br>The problem with goertzel detection in G.729 is not trivial. Too much<br>spectral energy was lost in the transcoding. I'm speaking from
<br>experience with DTMF detection in G.729.<br>><br>> For what it's worth, I've attempted to modify chan_sip in the<br>> sip_rtp_read func to include adding a dsp with ast_dsp_new, converting<br>> to SLINEAR with ast_set_read_format and then passing it to
<br>> ast_dsp_call_progress. But it doesn't seem to be doing anything at<br>> all.. I've been performing my test on G.711.<br>Your logic is sound. Did you dump out the return values?<br><br>Leo<br><br>_______________________________________________
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