[Serusers] SER + Asterisk PSTN calls don't hung up
Ricardo Carvalho
rcarvalho at iric.up.pt
Mon Aug 7 16:09:03 CEST 2006
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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