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. [...]