[asterisk-users] Hangup issue

[Cyril SCETBON]

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