[SR-Users] failure route correct..?

Daniel-Constantin Mierla miconda at gmail.com
Fri Jul 13 10:05:13 CEST 2012


Hello,

one audio problem is due to nat issues -- be sure you engage rtpproxy 
for this call in the failure route, if you didn't do it at all.

A ngrep trace of the call may help spotting if that is the case or not.

Cheers,
Daniel

On 7/11/12 8:41 PM, Gertjan Wolzak wrote:
> Hello All,
>
> I am trying to get a failure route to work, I have got it working 
> partially.
>
> When a call comes in, first I check the db_alias, if that is positive 
> I do a lookup location and relay if the location is valid.
>
> But sometimes the sip client is still registered in the location 
> table, but not connected anymore, mainly with wifi connected clients.
>
> So, I have got the time out on 3 seconds (fr_timer). When that hits I 
> have configured the following failure route:
>
>
> failure_route[NOTONLINE]
> {
>
>         xlog("SCRIPT: Notonline failure route\n");
>
>         t_on_failure("STOP");
>
>         if (t_is_canceled())
>         {
>                 exit;
>         }
>
>         if (t_check_status("408"))
>         {
>                 xlog("SCRIPT: Status is time out");
>                 $rU = $avp(orig_called); /( called number and alias id 
> not equal, so have to revert the rU back to the called number)/
>                 prefix("9993"); /( needed to get the right 
> manipulation done within asterisk)/
>                 xlog("SCRIPT: uri is $ru");
>                 $ru = "sip:" + $rU + "@w.x.y.x:5060";
>                 xlog("SCRIPT: uri is $ru"); (w.x.y.z ip address of the 
> asterisk box)
>                 append_branch();
>                 t_relay_to_udp("w.x.y.x","5060");
>                 break;
>         }
> }
>
>
> I am not sure if the above is correct. I have based this on an old 
> "voicemail" failure route I could find.
>
> It is working correct, the call is forwarded to an Asterisk box, where 
> some manipulation is done and then send to an pstn gateway.
>
> The only problem I have is one way audio. RTP from the called number 
> reaches the callee but not vice versa.
>
> Now I am wondering, can that be caused by the failure route, or should 
> I be looking in another direction?
>
> Hope someone can give me a pointer.
>
> Thanks.
>
> Gertjan Wolzak
>
>
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-- 
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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