5 Sep
2006
5 Sep
'06
11:36 a.m.
Hi users ,
I am proceeding in this way i have a SER-0.9.6 running fine :-)
and I had given a username and password to a call-shop and this callshop
owner with his username and password he connects to another 6 phones,
Actually he bills to his 6
phones and I will bill him ."o.k overall Scenario is well and good".
I made a little changes in ser.cfg and when the call made from the call shop
up to call connecting is O.K but when we hung the phone the SER is not
generating "BYE" messages to other party , so the call is on.. and i am not
getting "Acct Stop" packet also
SO How I can solve my problem :-(
any suggestions will be appreciated .
below is the message i am getting from SER when I hung the phone on one side
-------------------<this message is came from callshop Nat
address>---------<"it sends bye to my SER
"------------------------------------------------------
U 82.102.69.105:32768 -> 81.21.33.35:5060
BYE sip:99106883@81.21.33.35:5060 SIP/2.0.
To: "99106883"<sip:99106883@81.21.33.35:5060>;tag=78F9ECC4-166C.
From: "12345"<sip:12345@81.21.33.35:5060>;tag=837e5b2ff0b4cf4a.
Via: SIP/2.0/UDP 192.168.1.100:5060
;branch=z9hG4bK-d87543-5983e94f152d226c82a4e76799fe58e5-1--d87543-;rport.
Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK9bf920205fe3bef9.
Call-ID: f1b0fe2b6cdf1456(a)192.168.1.102.
CSeq: 9533 BYE.
Route: <sip:99106883@81.21.33.35:5060;lr;nat=yes;ftag=837e5b2ff0b4cf4a>.
Route: <sip:192.168.1.100:5060>.
Record-Route: <sip:192.168.1.100:5060>.
Contact: <sip:192.168.1.100:5060>.
Max-Forwards: 69.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE.
Proxy-Authorization: Digest username="12345", realm="81.21.33.35",
algorithm=MD5, uri="sip:192.168.1.100:5060",
nonce="44fd415a6754afa39f2668ce2ad11da6bfc65cde",
response="4001812db780010a557b40683efd2f9e".
Supported: replaces.
User-Agent: Grandstream BT110 1.0.8.23.
Content-Length: 0.
.
----------------<here it is as soon as SER recieve Bye message it has to
send Bye to other party >------<But it is sending to the hung up phone
itself>---------
#
U 81.21.33.35:5060 -> 192.168.1.100:5060
BYE sip:192.168.1.100:5060 SIP/2.0.
Record-Route: <sip:81.21.33.35;ftag=837e5b2ff0b4cf4a;lr=on>.
To: "99106883"<sip:99106883@81.21.33.35:5060>;tag=78F9ECC4-166C.
From: "12345"<sip:12345@81.21.33.35:5060>;tag=837e5b2ff0b4cf4a.
Via: SIP/2.0/UDP 81.21.33.35;branch=z9hG4bK936f.7ff7572.0.
Via: SIP/2.0/UDP 192.168.1.100:5060;received=82.102.69.105
;branch=z9hG4bK-d87543-5983e94f152d226c82a4e76799fe58e5-1--d87543-;rport=32768.
Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK9bf920205fe3bef9.
Call-ID: f1b0fe2b6cdf1456(a)192.168.1.102.
CSeq: 9533 BYE.
Record-Route: <sip:192.168.1.100:5060>.
Contact: <sip:192.168.1.100:5060>.
Max-Forwards: 16.
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO, SUBSCRIBE.
Proxy-Authorization: Digest username="12345", realm="81.21.33.35",
algorithm=MD5, uri="sip:192.168.1.100:5060",
nonce="44fd415a6754afa39f2668ce2ad11da6bfc65cde",
response="4001812db780010a557b40683efd2f9e".
Supported: replaces.
User-Agent: Grandstream BT110 1.0.8.23.
Content-Length: 0.
.
-----------------<And here I go iam getting this message and the call is not
being stopped >----------------------------------
#
U 81.21.33.35:5060 -> 82.102.69.105:32768
SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL).
To: "99106883"<sip:99106883@81.21.33.35:5060>;tag=78F9ECC4-166C.
From: "12345"<sip:12345@81.21.33.35:5060>;tag=837e5b2ff0b4cf4a.
Via: SIP/2.0/UDP 192.168.1.100:5060
;branch=z9hG4bK-d87543-5983e94f152d226c82a4e76799fe58e5-1--d87543-;rport=32768;received=
82.102.69.105.
Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK9bf920205fe3bef9.
Call-ID: f1b0fe2b6cdf1456(a)192.168.1.102.
CSeq: 9533 BYE.
Content-Length: 0.
Warning: 392 81.21.33.35:5060 "Noisy feedback tells: pid=27773 req_src_ip=
82.102.69.105 req_src_port=32768 in_uri=sip:99106883@81.21.33.35:5060
out_uri=sip:192.168.1.100:5060 via_cnt==2".
.
any suggestions will be appreciated:
Thank You.