Hello
I am not able to dial to pstn with kamailio, the call is routed to my pstn-gw(asterisk), but the final phone rings 4 or 5 seconds and then it is hanged up. My outbound route is: route[PSTN] { if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } route(TOASTERISK); exit; return; }
route[TOASTERISK] { sl_send_reply("100","Trying"); uac_replace_from("$fn","sip:$fn@$fd"); route(NATMANAGE); ds_select_dst("1","4"); t_on_failure("1"); t_relay(); }
failure_route[1] { ds_mark_dst("i"); if (!ds_next_dst()) { t_reply("503", "Service unavailable: no more dst"); exit; } route(TOASTERISK); }
With a traffic capture i can see the traffic returning to my kamailio server.
Any suggestion will be appreciated.
Hello,
On 6/7/12 6:30 PM, Kr0m wrote:
Hello
I am not able to dial to pstn with kamailio, the call is routed to my pstn-gw(asterisk), but the final phone rings 4 or 5 seconds and then it is hanged up. My outbound route is: route[PSTN] { if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; } if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } route(TOASTERISK); exit; return; }
route[TOASTERISK] { sl_send_reply("100","Trying"); uac_replace_from("$fn","sip:$fn@$fd"); route(NATMANAGE); ds_select_dst("1","4"); t_on_failure("1"); t_relay(); }
failure_route[1] { ds_mark_dst("i"); if (!ds_next_dst()) { t_reply("503", "Service unavailable: no more dst"); exit; } route(TOASTERISK); }
With a traffic capture i can see the traffic returning to my kamailio server.
Any suggestion will be appreciated.
what side is ending (canceling) the call? Maybe we can give better hints if you send the ngrep trace with the SIP messages of such call.
Cheers, Daniel