[asterisk-users] Hangup issue

[Cyril SCETBON]

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 [483062608 at accueil: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 
> '4D1E12-22A811DD-8829ADFE-87DC36C1 at 192.168.19.1' in 384 ms (Method: ACK)
> set_destination: Parsing <sip:489989614 at 192.168.19.1:5060> 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:489989614 at 192.168.19.1:5060 SIP/2.0
> 
> SIP/2.0 200 OK
> 
> <------------->
> --- (9 headers 0 lines) ---
> SIP Response message for INCOMING dialog BYE arrived
> Really destroying SIP dialog 
> '4D1E12-22A811DD-8829ADFE-87DC36C1 at 192.168.19.1' Method: ACK
> 
> SIP/2.0 200 OK
> 
> Any idea about what's happening and how to resolve it ?
> 
> Regards

-- 
Cyril SCETBON