Hello,

I use ser and b2bua the free one from vovida and do not experience your mentioned problems

The rest of your points should be traced with something like ngrep. Especially looking if you receive ACKs.

Tjapko.

On Tue, 2004-07-06 at 12:39, Derek Toner wrote:

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();
    };

}

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