[asterisk-users] Hangup issue

[Cyril SCETBON]

Nobody can help ?

I can provide the debug messages if needed.

Thanks

Cyril SCETBON wrote:
> 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