The NAT or Grandstream may be making fun with you ... ;-)
If the first message really is before SER receives it, look at this:
Route: < sip:99106883@81.21.33.35:5060;lr;nat=yes;ftag=837e5b2ff0b4cf4a>.
Route: <sip:192.168.1.100:5060 <http://192.168.1.100:5060>>.
Record-Route: <sip:192.168.1.100:5060 <http://192.168.1.100:5060>>.
The last Route header and the Record-Route messes things up. Loose route
will the second Route as destination.
You need to find out how these came there (look at the dialog creation
with INVITE and OK because the route set used in BYE was created there).
g-)
ravi reddy wrote:
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 <http://82.102.69.105:32768> -> 81.21.33.35:5060
<http://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
<http://192.168.1.100:5060>;branch=z9hG4bK-d87543-5983e94f152d226c82a4e76799fe58e5-1--d87543-;rport.
Via: SIP/2.0/UDP 192.168.1.102
<http://192.168.1.102>;branch=z9hG4bK9bf920205fe3bef9.
Call-ID: f1b0fe2b6cdf1456(a)192.168.1.102
<mailto:f1b0fe2b6cdf1456@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 <http://192.168.1.100:5060>>.
Record-Route: <sip:192.168.1.100:5060 <http://192.168.1.100:5060>>.
Contact: <sip:192.168.1.100:5060 <http://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
<http://81.21.33.35>", algorithm=MD5, uri="sip:192.168.1.100:5060
<http://192.168.1.100:5060>",
nonce="44fd415a6754afa39f2668ce2ad11da6bfc65cde",
response="4001812db780010a557b40683efd2f9e".
Supported: replaces.
User-Agent: Grandstream BT110 1.0.8.23 <http://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 <http://81.21.33.35:5060> -> 192.168.1.100:5060
<http://192.168.1.100:5060>
BYE sip:192.168.1.100:5060 <http://192.168.1.100:5060> SIP/2.0.
Record-Route: <sip: 81.21.33.35
<http://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
<http://81.21.33.35>;branch=z9hG4bK936f.7ff7572.0.
Via: SIP/2.0/UDP 192.168.1.100:5060
<http://192.168.1.100:5060>;received=82.102.69.105
<http://82.102.69.105>;branch=z9hG4bK-d87543-5983e94f152d226c82a4e76799fe58e5-1--d87543-;rport=32768.
Via: SIP/2.0/UDP 192.168.1.102
<http://192.168.1.102>;branch=z9hG4bK9bf920205fe3bef9.
Call-ID: f1b0fe2b6cdf1456(a)192.168.1.102
<mailto:f1b0fe2b6cdf1456@192.168.1.102>.
CSeq: 9533 BYE.
Record-Route: <sip:192.168.1.100:5060 <http://192.168.1.100:5060>>.
Contact: <sip:192.168.1.100:5060 <http://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
<http://81.21.33.35>", algorithm=MD5, uri="sip: 192.168.1.100:5060
<http://192.168.1.100:5060>",
nonce="44fd415a6754afa39f2668ce2ad11da6bfc65cde",
response="4001812db780010a557b40683efd2f9e".
Supported: replaces.
User-Agent: Grandstream BT110 1.0.8.23 <http://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 <http://81.21.33.35:5060> -> 82.102.69.105:32768
<http://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
<http://192.168.1.100:5060>;branch=z9hG4bK-d87543-5983e94f152d226c82a4e76799fe58e5-1--d87543-;rport=32768;received=
82.102.69.105 <http://82.102.69.105>.
Via: SIP/2.0/UDP 192.168.1.102
<http://192.168.1.102>;branch=z9hG4bK9bf920205fe3bef9.
Call-ID: f1b0fe2b6cdf1456(a)192.168.1.102
<mailto:f1b0fe2b6cdf1456@192.168.1.102>.
CSeq: 9533 BYE.
Content-Length: 0.
Warning: 392 81.21.33.35:5060 <http://81.21.33.35:5060> "Noisy
feedback tells: pid=27773 req_src_ip=82.102.69.105
<http://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
<http://192.168.1.100:5060> via_cnt==2".
.
any suggestions will be appreciated:
Thank You.
------------------------------------------------------------------------
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