<div>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. </div>
<div> </div>
<div>In fact, all carriers we've tested have transmitted the call this way. Even my cell phone gives me the tri-tone without "connecting"</div>
<div> </div>
<div>Yes, chan_sip would need to be modifed. That is the work I am requesting. </div>
<div> </div>
<div>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.</div>
<div> </div>
<div> </div>
<div>Thanks for your reply!</div>
<div>-Brett</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>> Hi All,<br>> I have a specific Need to detect SIT Tones is SIP RTP Early Media. I
<br>> specifically need to be able to tell the difference between a RNA and<br>> a SIT.<br>It should be a function of your gateway. chap_sip doesn't support analog<br>call progress. And, since you're using G.729
it could be rather<br>difficult to detect the tri-tone reliability all the time.<br><br>Leo<br><br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com
</a> --<br><br>asterisk-biz mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-biz">http://lists.digium.com/mailman/listinfo/asterisk-biz</a><br></blockquote>
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