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The NAT or Grandstream may be making fun with you ... ;-)<br>
If the first message really is before SER receives it, look at this:<br>
Route: <
<a class="moz-txt-link-abbreviated" href="mailto:sip:99106883@81.21.33.35:5060;lr;nat=yes;ftag=837e5b2ff0b4cf4a">sip:99106883@81.21.33.35:5060;lr;nat=yes;ftag=837e5b2ff0b4cf4a</a>>.<br>
Route: <sip:<a href="http://192.168.1.100:5060">192.168.1.100:5060</a>>.<br>
Record-Route: <sip:<a href="http://192.168.1.100:5060">192.168.1.100:5060
</a>>.<br>
<br>
The last Route header and the Record-Route messes things up. Loose
route will the second Route as destination.<br>
<br>
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).<br>
g-)<br>
<br>
<br>
ravi reddy wrote:
<blockquote
cite="midcfbb389b0609050236i326bd465h2cca9b2e8987b4b1@mail.gmail.com"
type="cite">
<div>Hi users ,</div>
<div> </div>
<div> I am proceeding in this way i have a SER-0.9.6 running
fine :-)</div>
<div>
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, </div>
<div> </div>
<div> Actually he bills
to his 6 phones and I will bill him ."o.k overall Scenario is well and
good".</div>
<div> </div>
<div>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 </div>
<div> </div>
<div> </div>
<div> SO How I can solve my problem :-( </div>
<div> any suggestions will be appreciated .</div>
<div> </div>
<div>below is the message i am getting from SER when I hung the phone
on one side</div>
<div> </div>
<div> </div>
<div>-------------------<this message is came from callshop Nat
address>---------<"it sends bye to my SER
"------------------------------------------------------</div>
<div>
<p>U <a href="http://82.102.69.105:32768">82.102.69.105:32768</a>
-> <a href="http://81.21.33.35:5060">81.21.33.35:5060</a> <br>
BYE <a class="moz-txt-link-abbreviated" href="mailto:sip:99106883@81.21.33.35:5060">sip:99106883@81.21.33.35:5060</a> SIP/2.0.<br>
To: "99106883"<
<a class="moz-txt-link-abbreviated" href="mailto:sip:99106883@81.21.33.35:5060">sip:99106883@81.21.33.35:5060</a>>;tag=78F9ECC4-166C.<br>
From: "12345"<a class="moz-txt-link-rfc2396E" href="mailto:sip:12345@81.21.33.35:5060"><sip:12345@81.21.33.35:5060></a>;tag=837e5b2ff0b4cf4a.<br>
Via: SIP/2.0/UDP <a href="http://192.168.1.100:5060">192.168.1.100:5060</a>;branch=z9hG4bK-d87543-5983e94f152d226c82a4e76799fe58e5-1--d87543-;rport.
<br>
Via: SIP/2.0/UDP <a href="http://192.168.1.102">192.168.1.102</a>;branch=z9hG4bK9bf920205fe3bef9.<br>
Call-ID: <a href="mailto:f1b0fe2b6cdf1456@192.168.1.102">f1b0fe2b6cdf1456@192.168.1.102</a>.<br>
CSeq: 9533 BYE.<br>
Route: <
<a class="moz-txt-link-abbreviated" href="mailto:sip:99106883@81.21.33.35:5060;lr;nat=yes;ftag=837e5b2ff0b4cf4a">sip:99106883@81.21.33.35:5060;lr;nat=yes;ftag=837e5b2ff0b4cf4a</a>>.<br>
Route: <sip:<a href="http://192.168.1.100:5060">192.168.1.100:5060</a>>.<br>
Record-Route: <sip:<a href="http://192.168.1.100:5060">192.168.1.100:5060
</a>>.<br>
Contact: <sip:<a href="http://192.168.1.100:5060">192.168.1.100:5060</a>>.<br>
Max-Forwards: 69.<br>
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO,
SUBSCRIBE.<br>
Proxy-Authorization: Digest username="12345", realm="
<a href="http://81.21.33.35">81.21.33.35</a>", algorithm=MD5,
uri="sip:<a href="http://192.168.1.100:5060">192.168.1.100:5060</a>",
nonce="44fd415a6754afa39f2668ce2ad11da6bfc65cde",
response="4001812db780010a557b40683efd2f9e".
<br>
Supported: replaces.<br>
User-Agent: Grandstream BT110 <a href="http://1.0.8.23">1.0.8.23</a>.<br>
Content-Length: 0.<br>
.</p>
<p>----------------<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>---------</p>
<p>#<br>
U <a href="http://81.21.33.35:5060">81.21.33.35:5060</a> -> <a
href="http://192.168.1.100:5060">192.168.1.100:5060</a><br>
BYE sip:<a href="http://192.168.1.100:5060">192.168.1.100:5060</a>
SIP/2.0.<br>
Record-Route: <sip:
<a href="http://81.21.33.35">81.21.33.35</a>;ftag=837e5b2ff0b4cf4a;lr=on>.<br>
To: "99106883"<a class="moz-txt-link-rfc2396E" href="mailto:sip:99106883@81.21.33.35:5060"><sip:99106883@81.21.33.35:5060></a>;tag=78F9ECC4-166C.<br>
From: "12345"<a class="moz-txt-link-rfc2396E" href="mailto:sip:12345@81.21.33.35:5060"><sip:12345@81.21.33.35:5060></a>;tag=837e5b2ff0b4cf4a.
<br>
Via: SIP/2.0/UDP <a href="http://81.21.33.35">81.21.33.35</a>;branch=z9hG4bK936f.7ff7572.0.<br>
Via: SIP/2.0/UDP <a href="http://192.168.1.100:5060">192.168.1.100:5060</a>;received=<a
href="http://82.102.69.105">82.102.69.105
</a>;branch=z9hG4bK-d87543-5983e94f152d226c82a4e76799fe58e5-1--d87543-;rport=32768.<br>
Via: SIP/2.0/UDP <a href="http://192.168.1.102">192.168.1.102</a>;branch=z9hG4bK9bf920205fe3bef9.<br>
Call-ID: <a href="mailto:f1b0fe2b6cdf1456@192.168.1.102">
f1b0fe2b6cdf1456@192.168.1.102</a>.<br>
CSeq: 9533 BYE.<br>
Record-Route: <sip:<a href="http://192.168.1.100:5060">192.168.1.100:5060</a>>.<br>
Contact: <sip:<a href="http://192.168.1.100:5060">192.168.1.100:5060</a>
>.<br>
Max-Forwards: 16.<br>
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, INFO,
SUBSCRIBE.<br>
Proxy-Authorization: Digest username="12345", realm="<a
href="http://81.21.33.35">81.21.33.35</a>", algorithm=MD5, uri="sip:
<a href="http://192.168.1.100:5060">192.168.1.100:5060</a>",
nonce="44fd415a6754afa39f2668ce2ad11da6bfc65cde",
response="4001812db780010a557b40683efd2f9e".<br>
Supported: replaces.<br>
User-Agent: Grandstream BT110 <a href="http://1.0.8.23">1.0.8.23</a>.<br>
Content-Length: 0.<br>
.</p>
<p>-----------------<And here I go iam getting this message and
the call is not being stopped >----------------------------------</p>
<p>#<br>
U <a href="http://81.21.33.35:5060">81.21.33.35:5060</a> -> <a
href="http://82.102.69.105:32768">82.102.69.105:32768</a><br>
SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL).<br>
To: "99106883"<
<a class="moz-txt-link-abbreviated" href="mailto:sip:99106883@81.21.33.35:5060">sip:99106883@81.21.33.35:5060</a>>;tag=78F9ECC4-166C.<br>
From: "12345"<a class="moz-txt-link-rfc2396E" href="mailto:sip:12345@81.21.33.35:5060"><sip:12345@81.21.33.35:5060></a>;tag=837e5b2ff0b4cf4a.<br>
Via: SIP/2.0/UDP <a href="http://192.168.1.100:5060">192.168.1.100:5060</a>;branch=z9hG4bK-d87543-5983e94f152d226c82a4e76799fe58e5-1--d87543-;rport=32768;received=
<a href="http://82.102.69.105">82.102.69.105</a>.<br>
Via: SIP/2.0/UDP <a href="http://192.168.1.102">192.168.1.102</a>;branch=z9hG4bK9bf920205fe3bef9.<br>
Call-ID: <a href="mailto:f1b0fe2b6cdf1456@192.168.1.102">f1b0fe2b6cdf1456@192.168.1.102
</a>.<br>
CSeq: 9533 BYE.<br>
Content-Length: 0.<br>
Warning: 392 <a href="http://81.21.33.35:5060">81.21.33.35:5060</a>
"Noisy feedback tells: pid=27773 req_src_ip=<a
href="http://82.102.69.105">82.102.69.105</a> req_src_port=32768
in_uri=
<a class="moz-txt-link-abbreviated" href="mailto:sip:99106883@81.21.33.35:5060">sip:99106883@81.21.33.35:5060</a> out_uri=sip:<a
href="http://192.168.1.100:5060">192.168.1.100:5060</a> via_cnt==2".<br>
.</p>
<p>any suggestions will be appreciated:</p>
<p>Thank You.<br>
</p>
</div>
<pre wrap="">
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</blockquote>
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