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
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
