Hi Bogdan I have redirecting the user if not available to to Voice mail of Asterisk that run on myserver-ipaddress:5090 i have integrated OpenSer+Asterisk so when the local user not availble it should go to Voice mail but when iam dialing from X-lite call going out. but when iam dialing from Sipura the call Going to voice mail .. why iam not sure here is my openser.cfg looks like ----------- modparam("uac","credential","99999:provider.com:99999") # ------------------------- request routing logic ------------------- route { #check for old messages: could mean a problem withthe DNS entries or some other loop-causer... if (!mf_process_maxfwd_header("10")) { xlog("L_WARN", "WARNING: Too many hops\n"); sl_send_reply("483", "Too many hops, forward count exceeded limit\n"); return; }; #check for extremely large messages; we don't need a sip dos attack if (msg:len >= 2048) { xlog("L_WARN", "WARNING: Message too large, >= 2048 bytes\n"); sl_send_reply("513", "Message too large, exceeded limit\n"); return; }; #record everything besides registers and acks if(method!="REGISTER" && method!="ACK") { setflag(1); }; #do not send to voicemail if BYE or CANCEL #is used to end call before user pickup or timeout if(method=="CANCEL" || method=="BYE") { setflag(10); }; #grant route if route headers already present if (loose_route()) { route(1); return; }; #Always require authentication, which could result in a PSTN, ie $$$ if (method=="REGISTER") { if(!www_authorize("mydomain.com", "subscriber")) { www_challenge("mydomain.com", "0"); return; } else { if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; #Save into user database, used below when checkingif user is available xlog("L_INFO", "REGISTER: User Authenticated Correctly\n"); save("location"); return; }; }; #} if (method=="INVITE") { if(uri=~"sip:\*98@.*") #if(uri=~"sip:\*86@.*") { #authorize if a call is going to PSTN if(!proxy_authorize("mydomain.com", "subscriber")) { proxy_challenge("mydomain.com", "0"); return; }; xlog("L_INFO", "CALL: Call to check voicemail\n"); rewritehostport("myserver-ipaddress:5090"); } else { if (does_uri_exist()) { #Call is to sip client, so do nothing but route xlog("L_INFO", "CALL: Sip client\n"); if (!lookup("location")) { # sl_send_reply("404", "Not Found"); # log(1, "ERROR: User Not Found\n"); rewritehostport("myserver-ipaddress:5090"); t_relay(); return; }; } else { #authorize if a call is going to PSTN if(!proxy_authorize("mydomain.com", "subscriber")) { proxy_challenge("mydomain.com", "0"); return; }; #Call destination is PSTN, so send it to the gateway (Net.com) xlog("L_INFO", "CALL: PSTN gateway1\n"); rewritehostport("provider-ip:5060"); }; }; #Make sure that all subsequent requests go through us; record_route(); } else { if (does_uri_exist()) { #Call is to sip client, so do nothing but route xlog("L_INFO", "CALL: Sip client\n"); if (!lookup("location")) { # sl_send_reply("404", "Not Found"); # log(1, "ERROR: User Not Found\n"); rewritehostport("myserver-ipaddress:5090"); t_relay(); return; }; } else { #Call destination is PSTN, so send it to the gateway (Net.com) xlog("L_INFO", "CALL: PSTN gateway2\n"); rewritehostport("provider-ip:5060"); }; record_route(); }; #ALL PROCESSING IS DONE, SO ROUTE route(1); } route[1] { #send the call outward if(method=="INVITE" && !isflagset(10)) { t_on_failure("2"); }; if (!t_relay()) { xlog("L_WARN", "ERROR: t_relay failed"); sl_reply_error(); }; } failure_route[2] { if(!t_was_cancelled()) { revert_uri(); rewritehostport("myserver-ipaddress:5090"); append_branch(); #PREVENT SOME CRAZY VOICEMAIL LOOP xlog("L_INFO", "INFO: CALL TO VOICEMAIL"); setflag(10); route(1); } } ram