<div dir="ltr">Husnain, <div>The same type of question is already answered by <span style="font-size:12.8px;font-weight:bold;white-space:nowrap">Kristian F. Høgh</span> on the opensips mailing list. Try to search "<b>[<span class="" style="background-color:rgb(255,255,255)">OpenSIPS</span>-Users] Playing caller a file before dialing callee</b>"</div><div><br></div><div>Hope it will solve your problem.</div><div><br></div><div>Faheem</div><div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Jan 7, 2016 at 10:37 AM, Hamid Hashmi <span dir="ltr"><<a href="mailto:hamid2kviii@hotmail.com" target="_blank">hamid2kviii@hotmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><div dir="ltr"><font color="#2672EC">Try the following example. Change connection IP and codec order accordingly.</font><div><font color="#2672EC"><br></font></div><div><pre style="overflow:auto;font-family:Consolas,'Liberation Mono',Menlo,Courier,monospace;font-size:11.9px;font-stretch:normal;line-height:1.45;padding:16px;border-radius:3px;word-wrap:normal;color:rgb(51,51,51);margin-top:0px!important;margin-bottom:0px!important;background-color:rgb(247,247,247)"><code style="font-family:Consolas,'Liberation Mono',Menlo,Courier,monospace;font-size:11.9px;padding:0px;margin:0px;border-radius:3px;word-break:normal;border:0px;display:inline;max-width:initial;overflow:initial;line-height:inherit;word-wrap:normal;background:transparent">if (is_method("INVITE") && has_body("application/sdp")) {
$var(Session_owner) = $rb[1];
append_to_reply("Content-Type: application/sdp\r\nv=0\r\n$var(Session_owner)\r\ns=call\r\nc=IN IP4 10.130.130.114\r\nt=0 0\r\nm=audio 61896 RTP 0 8 3 101\r\na=rtpmap:0 pcmu/8000\r\na=rtpmap:8 pcma/8000\r\na=rtpmap:3 gsm/8000\r\na=rtpmap:101 telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n")';
t_reply_with_body("183", "Session Progress", "$var(body)");
}</code></pre><br><br><b><font color="#5133ab" face="Times New Roman" size="3"><i>Hamid R. Hashmi</i></font></b><div><font size="2">Software Engineer - VoIP</font></div><div><font color="#008a17" size="2" style="font-size:10pt">Vopium A/S</font></div><br><br><div><hr>Date: Wed, 6 Jan 2016 20:33:29 +0300<br>From: <a href="mailto:husnain.taseer@gmail.com" target="_blank">husnain.taseer@gmail.com</a><br>To: <a href="mailto:users@lists.opensips.org" target="_blank">users@lists.opensips.org</a><br>Subject: [OpenSIPS-Users] Generating 183 reply and Playing Early Media using rtpproxy_stream2uac()<div><div class="h5"><br><br><div dir="ltr">Dear Users,<div>I have a scenario where I want to Play an announcement as early media to the UAC before answering the call but I don't want to use any media server like asterisk/Freeswitch. </div><div><br></div><div>When user agent sends an INVITE I am calling rtpproxy_offer() and sending INVITE to B party. On 100 Trying from B party I am calling rtpproxy_stream2uac() and streaming the file I can see that RTPs are going towards the UAC (caller) but softphone is not accepting those RTPs because 183 was not sent to the softphone so he don't know the media details of the rtpproxy. but as 200 Ok reaches to the softphone last part of the audio can be heard immediately after Answer. </div><div><br></div><div>So I think on 100 Trying from B Part if I send 183 Session Progress to the softphone and then starting the RTP stream will work. So can you please tell me is there a way to generate 183 Session Progress with media details of RTPPROXY in opensips ? so that my scenario starts work.<br><br>Regards,</div><div>Husnain Taseer</div><div>VoIP Developer</div></div>
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