hi everybody:
I use Asterisk and SER(with nathelp moudle) in on box, SER as sip
registrar and sip proxy, Asterisk as media gw and pstn connector. Here
is my configuration: SER use 192.168.2.10:5060,Asterisk use
192.168.2.10:5065,my pstn gw is 192.168.2.20:5060
in ser.cfg
if (method=="INVITE") {
if (!(uri=~"^sip:[^9][0-9]{3}@.*")) {
rewritehostport("192.168.2.10:5065");
forward(uri:host, uri:port);
break;
};
};
in extensions.conf
[Out]
exten => _9XXX,1,Dial(SIP/${EXTEN}@192.168.2.20,,rT)
exten => _9XXX,n,hangup
I use xlite dial 9001 and the 9001 phone ring,then i pick up the 9001 phone,
all seem good.but when i drop the 9001 phone,xlite doesn't disconnect.xlite
still show connected status till i manual huangup it.it seem the BYE message
cann't be sended to the xlite,what can i do?
Regards.
here is some asterisk sip debug info:
set_destination: Parsing <sip:192.168.2.10;ftag=4b08853d;lr=on> for
address/port to send to
set_destination: set destination to 192.168.2.10, port 5060
Reliably Transmitting (NAT) to 192.168.2.10:5060:
BYE sip:2002@192.168.2.160:9753 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.10:5065;branch=z9hG4bK37f9a0d6;rport
Route: <sip:192.168.2.10;ftag=4b08853d;lr=on>
From: <sip:9001@192.168.2.10>;tag=as1f67b017
To: 2002<sip:2002@192.168.2.10>;tag=4b08853d
Contact: <sip:9001@192.168.2.10:5065>
Call-ID: 3d4b5923d31edb66
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Retransmitting #1 (NAT) to 192.168.2.10:5060:
BYE sip:2002@192.168.2.160:9753 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.10:5065;branch=z9hG4bK37f9a0d6;rport
Route: <sip:192.168.2.10;ftag=4b08853d;lr=on>
From: <sip:9001@192.168.2.10>;tag=as1f67b017
To: 2002<sip:2002@192.168.2.10>;tag=4b08853d
Contact: <sip:9001@192.168.2.10:5065>
Call-ID: 3d4b5923d31edb66
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0