<html><head><meta http-equiv="Content-Type" content="text/html charset=windows-1252"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;"><div>I seem to be going round in circles… I am trying to use dlg_bridge() from the dialog module to establish a call between two SIP endpoints. I have tested with Snom phones and linphone soft phone with the same result.</div><div><br></div><div>I get an outbound call to the first (from) end point, I answer the phone… and then nothing obvious happens.</div><div><br></div><div>I can get this same result when using kamcmd or from within the kamailio.cfg. For example using kamcmd:</div><div><br></div><div><span style="background-color: rgb(32, 36, 45); color: rgb(245, 245, 245); font-family: 'Espresso Mono'; font-size: 11px;">kamcmd dlg.bridge_dlg <a href="sip:105@mykamailioip">sip:105@mykamailioip</a> <a href="sip:106@mykamailioip">sip:106@mykamailioip</a> <a href="sip:mykamailioip:5060">sip:mykamailioip:5060</a></span></div><br><div apple-content-edited="true">
<div>I expected this to send an invite from "controller" via my KamailioIP to my registered local subscriber 105 (from), and then send a re-invite to 105 so that 105 creates a call leg to 106 (to).</div><div><br></div><div>The kamailio.cfg is pretty much the default. I am running 4.2 and rtpengine to proxy the RTP streams. </div><div><br></div><div>Not sure if it is necessary or useful but I call setflag() at the start of the request_route() to set the Dialog flag. What else should I have included in the kamailio.cfg to make this work?</div>
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<div><br></div>Also is there any way to control the SDP in the initial invite from dlg_bridge. By default I see RTP/AVP, with alaw and ulaw…. I’d like to offer RTP/SAVP swell or instead of RTP/AVP.<div><br></div><div>Thanks </div><div><br><div><br></div></div></body></html>