Hola tele ;) I have tested an example similar to yours, and it works ok. It's de following: UA1 ---[1]--> WeSIP --[2]--486 Busy Here---> UA2 | [3] ---OK----_> UA3 [1] UA1 generates an INVITE to UA2 ant it 's forwarded to UA2 with a B2BUA behaviour. In do INVITE. I have implemented the following: protected void doInvite(SipServletRequest invite){ SipServletRequest otherInvite = sf.createRequest(invite, false); SipURI sipUri = sf.createSipURI("UA2","proxy"); otherInvite.setRequestURI(sipUri); otherInvite.getSession().setAttribute("REQUEST", otherInvite); otherInvite.getSession().setAttribute("PEER_SESSION",invite.getSession()); invite.getSession().setAttribute("PEER_SESSION",otherInvite.getSession()); invite.getSession().setAttribute("REQUEST", invite); otherInvite.send(); } [2] UA2 declines the call and a 486 response is received by WeSIP and process it in doErrorResponse: protected void doErrorResponse(SipServletResponse errorResponse) { switch(errorResponse.getStatus()){ case 486: SipServletRequest request = (SipServletRequest) errorResponse.getSession().getAttribute("REQUEST"); SipSession otherSession = (SipSession) errorResponse.getSession().getAttribute("PEER_SESSION"); SipURI sipUri = sf.createSipURI("UA3", "proxy"); request.setRequestURI(sipUri); SipServletRequest newRequest = sf.createRequest(request,false); newRequest.getSession().setAttribute("REQUEST", newRequest); newRequest.getSession().setAttribute("PEER_SESSION", otherSession); otherSession.setAttribute("PEER_SESSION", newRequest.getSession()); newRequest.setHeader("X-SSVTPBX", "yes"); newRequest.send(); break; } } [3] UA3 Recevies the second call and it takes down. I do not understand the cause of error. I annex you the code of simple sip application example that replies to your problem. You can start it up and checking if it works. Else if work for you, I would attempt watching the particular case when a 5XX responses are received. Sorry... Best regards. Antonio. -----Mensaje original----- De: tele [mailto:tele@plexia.com] Enviado el: lunes, 19 de marzo de 2007 14:33 Para: Antonio Abajo Álvarez Asunto: RE: [Users] WeSIP session question Hola Antonio, The scenario is more complex, i'll try to explain it: PSTN | | UA---->GW ------> WeSIP(B2BUA) | | mediaserver GW: 82.215.163.5 WeSIP: 82.215.163.67 MS: 82.215.133.50 in the mediaserver there is an vxml script that i play in early media and in case of particular event return to wesip a 503 temporaly unavailable or e 410 Gone. So my B2BUA application have control of this and can do stuff with the 503 and 410. in particular, in case of 503 temporaly unavailable i get the upstream session and create a new invite to the media server for play another announcement associated. in case of 410 gone i generate a new invite to the GW with the original URI request for the correct termination. Yes when the session is removed i'm able to send another call. attached here the logs and the servlet. don't care about the hardcoded IP and the repeated code :-) i'm doing testing.. regards, :tele On Mon, 2007-03-19 at 13:29 +0100, Antonio Abajo Álvarez wrote: _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users