Husnain, The same type of question is already answered by Kristian F. Høgh on the opensips mailing list. Try to search "*[OpenSIPS-Users] Playing caller a file before dialing callee*"
Hope it will solve your problem. Faheem On Thu, Jan 7, 2016 at 10:37 AM, Hamid Hashmi <[email protected]> wrote: > Try the following example. Change connection IP and codec order > accordingly. > > 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)"); > } > > > > *Hamid R. Hashmi* > Software Engineer - VoIP > Vopium A/S > > > ------------------------------ > Date: Wed, 6 Jan 2016 20:33:29 +0300 > From: [email protected] > To: [email protected] > Subject: [OpenSIPS-Users] Generating 183 reply and Playing Early Media > using rtpproxy_stream2uac() > > > Dear Users, > 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. > > 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. > > 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. > > Regards, > Husnain Taseer > VoIP Developer > > _______________________________________________ Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >
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