6 Jul
2004
6 Jul
'04
12:39 p.m.
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();
};
}