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
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 -
http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 -
http://asipto.com/u/kpw
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