5 Jan
2006
5 Jan
'06
5:48 p.m.
Hello,
"SIP/2.0 481 Call Leg/Transaction Does Not Exist" is for PRACK, because you do
not change the From header for it. You have to do the same translation for all requests
within the dialog. See the documentation of uac module: .
http://openser.org/docs/modules/1.1.x/uac.html
Cheers,
Daniel
On 01/04/06 12:21, unplug wrote:
Actually, I am replacing the username of the uri with
the alias that
stored in the database. In my configuration file, it is using
mediaproxy for NAT function (features-callfwd.5.0.cfg from getting
started). I also add the following codes in the very first of the
route routine for alias replacing purpose.
route {
...
if (!has_totag() && method=="INVITE") {
if (avp_db_load("$from/uri","s:alias")) {
xlog("L_INFO","sip408: have alias -
[$avp(s:alias)]\n");
uac_replace_from("anonymous","sip:$avp(s:alias)@$si");
} else {
xlog("L_INFO","sip411: no alias\n");
};
};
...
}
When I make a call from a phone to the PSTN phone, the caller drops
the ring when the callee rings. Callee hangs up and callee rings
again few seconds after. You can find an error message "SIP/2.0 481
Call Leg/Transaction Does Not Exist" in line 180. I wonder if there
is any thing wrong with the above code or my concept is wrong. Please
help. Below is the sip message.
[...]