Hi All,
Need help with a nagging issue: UA->Opensips 1->Opensips 2->PSTN UA sends an invite on Opensips 1, and is routed via do_routing() to Opensips 2, Opensips 2 uses do_routing to get to the PSTN, call starts ringing. UA cancels call before answer, but now t_check_trans fails and the CANCEL is not passed onto the PSTN, with the result that the call rings forever and can only be terminated by the remote answering and dropping the call or through a timeout. The scripts on Opensips 1 & Opensips 2 is virtuall identical: How do I get the CANCEL to the PSTN ? route{ ..... if (is_method("CANCEL") ) { route(5); # drop media proxy if (t_check_trans()){ # this always fails after a do_routing() xlog("L_INFO","CANCEL Transaction[$fd/$fu/$rd/$ru/$si/]\n"); t_relay(); exit; }; exit; } } route[4] { xlog("L_INFO","Route4 [$fd/$fu/$rd/$ru/$si/]\n"); $avp(i:102)=1; # Default dr-group route(10); # Do custom stuff t_on_failure("4"); if (do_routing("$avp(i:102)")){ xlog("L_INFO","Route4 Route to Dyna Group: $avp(i:102)[$fd/$fu/$rd/$ru/$si/]\n"); t_newtran(); route(1); exit; }; xlog("L_INFO","Route4 No Route to Host[$fd/$fu/$rd/$ru/$si/]\n"); sl_reply_error(); exit; } Regards Juri Nysschen <http://www.greydotelecom.net/bcard/jnysschen.htm>
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