[Users] NAT+uac_replace_from

Daniel-Constantin Mierla daniel at voice-system.ro
Thu Jan 5 16:48:04 CET 2006


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.
> [...]
>   




More information about the sr-users mailing list