[SR-Users] Dispatcher PSTN

[Daniel-Constantin Mierla]

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
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