<br><br><div><span class="gmail_quote">On 7/31/07, <b class="gmail_sendername">Richard Brady</b> <<a href="mailto:rnbrady@gmail.com">rnbrady@gmail.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
[Resent due to non-descriptive subject line.]<br><br>Hi folks<br><br>When connecting two SIP users, is there any way to stop Asterisk from<br>sending SIP 183 Session Progress messages, either globally or<br>per-peer?</blockquote>
<div><br>Yes, the option is progressinband in sip.conf, see: <a href="http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+progressinband">http://www.voip-info.org/wiki/index.php?page=Asterisk+sip+progressinband</a><br>
</div><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Scenario as follows:<br> Call from UA1 to Asterisk (UA2) to UA3.<br> UA3 sends RTP before SIP OK to Asterisk (UA2).
<br> Asterisk (UA2) detects early audio from UA3 and sends 183 Session<br>Progress with SDP to UA1.<br><br>Instead I would like it to just send on the early audio, is this possible?</blockquote><div><br> Why do you care about the 183 or not? Because ignoring anything you said about SIP 183 what you want is to send SIP 183 which would indicate there is inband indications. See:
<a href="http://www3.ietf.org/proceedings/99jul/slides/mmusic-sip183-99jul/index.htm">http://www3.ietf.org/proceedings/99jul/slides/mmusic-sip183-99jul/index.htm</a><br><br>Also it wouldnt hurt to read the SIP RFC's to have a better understanding of what is going on:
<br><br><a href="ftp://ftp.rfc-editor.org/in-notes/rfc3261.txt">ftp://ftp.rfc-editor.org/in-notes/rfc3261.txt</a><br><a href="ftp://ftp.rfc-editor.org/in-notes/rfc2543.txt">ftp://ftp.rfc-editor.org/in-notes/rfc2543.txt</a>
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