Re: [Serusers] SER & b2bua

Hi all, I have been experimenting with Ser and b2bua, and a Grandstream Budge Tone 100 SIP phone, although the same problem occurs for Cisco ATAs & softphones. When I try to make a call, the call will succeed, but there are a number of issues at the moment * it takes a long amount of time to connect, usually over 10 seconds. In some cases, it will connect even when the caller has disconnected * with b2bua in between the Ser and the SIP gateway, the return code is 484 - Address Incomplete * without b2bua, the return code is 408 - Request Timeout * the Ser gets into a loop when receiving an ACK. I have a feeling that these are all related, and could be down to the ser.cfg file. Below is the ser.cfg file. Any and all help would be appreciated. Regards, Derek # ------------------------- request routing logic ------------------- # main routing logic route{ # Standard max_forward_header checks... # A REGISTER is when a client attempts to register with a SIP Server if(method=="REGISTER") { if (!performRegistration("4ecalls.com")) { sl_send_reply("404", "Not Found"); break; }; save("location"); sl_send_reply("200", "OK"); }; # It is an attmept to make a call if(method=="INVITE") { sl_send_reply("100", "Trying.."); log("Attempting to re-route call ...\n"); # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # If we get to here, we intend to re-route throgh a PSTN network log("Lookup failed ...\n"); sl_send_reply("100", "Trying.."); if (performInvite("4ecalls.com")) { log("Am checking to see if it is in our domain...\n"); if(uri=~"^sip:0[0-9]*") { log("Am really attempting to re-route call \n"); rewritehostport("<SIP GATEWAY IP>:5060"); forward(<SIP GATEWAY IP>, 5060); # This should forward to the B2BUA server # rewritehostport("<b2bua IP>:5065"); # forward(<b2bua IP>, 5065); # log("Am done with that call ...\n"); # We may want to send a reply back here to prevent the client from # sending too many INVITE requests sl_send_reply("181", "Forwarding call"); }; } else { log("The call was not logged ...\n"); }; log("The call was redirected - hopefully ...\n"); } # Set the accounting flag - XXX do we actually need this? # setflag(1); }; # An ACK is when a call is acknowlowdged to have occured if(method=="ACK") { setStartTimeActiveCall(""); sl_send_reply("200", "OK"); }; # A CANCEL is when a call is cancelled in the middle of an attempt if(method=="CANCEL") { # Remove the active details... removeActiveCall(""); }; # A BYE is for when a call finishes, so what we would want to do here is to track these # messages, as these contain the info for the billing if(method=="BYE") { log("Found BYE message ...\n"); # This call here should slow down the dispatching of messages t_reply("100", "Trying to bill..."); # This will process the message and do all necessary work on it... # In this case, we just want to perform the BYE actions... log("Checking to see if the destination is a SIP phone or not ...\n"); # This lookup is here so we can determine if the call was to a SIP softphone or ATA # If it was a SIP Softphone or an ATA, then we won't need to bil, as this is over IP if (!lookup("location")) { log("Destination is a PSTN number ...\n"); # This call will bill both the provider and customer performBillingOperation("4ecalls.com"); } }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if(method=="INVITE") { record_route(); }; # loose-route processing if (loose_route()) { t_relay(); break; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } ______________________________________________________________________ _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers