Hi All,
I was trying to play with the 3.1 feature specifically media handling
capabilities.
I want opensips act as a playing server by answering WebRTC based calls.
Here is a scenario I was trying to do.
-- Opensips will receive WSS call
-- Process the call
-- Play file with 200 OK
-- Sending to Voicemail (recording of file using rtpengine recording module)
-- Hangup call by caller or hangup after some time.
Here is sample routing I plan to develop. I tried it is not working as 200
OK is not generated with SDP.
route[VOICEMAIL]{
xlog("Receiving voicemail");
$var(rtpengine_flags) = "trust-address replace-origin
replace-session-connection rtcp-mux-offer ICE=force transcode-PCMU
transcode-G722 SDES-off UDP/TLS/RTP/SAVP";
rtpengine_offer("$var(rtpengine_flags)");
rtpengine_start_recording();
append_to_reply("Contact: <sip:voicemail@XXX_XXX_XXX_XXX>\r\n");
rtpengine_answer("$var(rtpengine_flags)");
t_reply(200, "Ok");
rtpengine_play_media("file=/etc/opensips/sounds/vm-isunavail.wav");
xlog("waiting for voicemail to be recorded");
sleep(30);
exit;
}
I want to know how this will be possible? Don't consider Asterisk and
Free-switch as a media server.
Any help suggestion would be appreciated.
--
Best Regards,
*Dhaval Indrodiya*
*skype: dki123sabse*
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