I've tried using a SIP client and when asterisk issue the Hangup
function the SIP client indicate that the call is terminated.
Maybe a SIP parameter with the pstn gateway ?
Cyril SCETBON wrote:
> Hi guys,
>
> My asterisk server is connected to a pstn gateway using SIP. When I
> receive a call and use the Hangup command the pstn seems to not
> correctly see the request and the caller gets a 'number unknown" message.
>
> Below are the debug message printed on the CLI :
>
>
> -- Executing [EMAIL PROTECTED]:3]
> Hangup("SIP/192.168.19.1-0818f100", "") in new stack
> == Spawn extension (accueil, 483062608, 3) exited non-zero on
> 'SIP/192.168.19.1-0818f100'
> Scheduling destruction of SIP dialog
> '[EMAIL PROTECTED]' in 384 ms (Method: ACK)
> set_destination: Parsing <sip:[EMAIL PROTECTED]:5060> for
> address/port to send to
> set_destination: set destination to 192.168.19.1, port 5060
> Reliably Transmitting (NAT) to 192.168.19.1:53728:
> BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0
>
> SIP/2.0 200 OK
>
> <------------->
> --- (9 headers 0 lines) ---
> SIP Response message for INCOMING dialog BYE arrived
> Really destroying SIP dialog
> '[EMAIL PROTECTED]' Method: ACK
>
> SIP/2.0 200 OK
>
> Any idea about what's happening and how to resolve it ?
>
> Regards
--
Cyril SCETBON
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