Thanks for your reply. According to your reply, I add the following
code for the PRACK.
if (method=="PRACK") {
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");
};
}
The result is the same as before. It seems there is a conflict in the
from header and it causes the 481 error. To be more precise, below is
the protocol flow.
UA1 server1 server2
fromtag:taga -----from username:9000 INVITE----->
fromtag:taga <----from username:9000 100 trying---
after uac_replace_from("1234","sip:[EMAIL PROTECTED]")
fromtag:taga ------from username:1234 INVITE
---->
(callee rings but caller drops)
fromtag:taga <-----from username:1234 100
trying----
(callee still rings until answers the call)
fromtag:taga <----from username:1234 session
progress 183---
fromtag:taga <-----from username:9000 session prgress 183----
fromtag:taga -----from username:9000 PRACK----------->
fromtag:taga ----------from username:9000
PRACK--->
fromtag:taga <----from username:9000 481
call leg -----
fromtag:taga <----from username:9000 481 call leg -----
On 1/5/06, Daniel-Constantin Mierla <[EMAIL PROTECTED]> wrote:
> 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.
> > [...]
> >
>
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