asterisk users - Aug 2006 - SER + Asterisk PSTN calls don't hung up

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.

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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