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[cyril.scetbon at free.fr]

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

[cyril.scetbon at free.fr]

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:

[cyril.scetbon at free.fr]

Nobody can help ?

I can provide the debug messages if needed.

Thanks

Cyril SCETBON wrote: