[asterisk-users] SER + Asterisk PSTN calls don't hung up

[Ricardo Carvalho]

Problem solved.

It was needed to insert the following code in ser.cfg:

-------------------------------------------------------------
if (method=="CANCEL") {
          route(1);
          break;
}
-------------------------------------------------------------

and also:

-------------------------------------------------------------
exten => _0.,2,Busy
exten => _0.,3,Hangup
-------------------------------------------------------------

Ricardo.










Ricardo Carvalho wrote:
> Hi,
>
> I'm deploying a SER + Asterisk architecture, where SER is used to 
> manage acc, users database and sip routing, and Asterisk is used for 
> voicemail and PSTN gateway.
> The system is already able to make and receive calls from the PSTN, 
> although, only after the call has been established it can be hung up 
> with success; when it is still ringing, if any side hungs up the call, 
> it still keeps ringing on the other side. Observing with Ethereal, we 
> concluded that in this erroneous cases, the CANCEL SIP request isn't 
> transmitted from the SER to Asterisk (if cancelled from the VoIP side) 
> being transmitted a "404  User Not Found" message from SER to Sip 
> Phone. If hung from the PSTN side, the sip phone keeps calling after 
> that, and ends calling by time-out being observed a "486 Busy Here" 
> status message from Asterisk to SER and then from SER to sip phone.
>
> Any help, please?
>
> Regards,
>
> Ricardo.
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