<div>Hi users ,</div>
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<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,
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<div> Actually he bills to his 6 phones and I will bill him ."o.k overall Scenario is well and good".</div>
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<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
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<div> SO How I can solve my problem :-( </div>
<div> any suggestions will be appreciated .</div>
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<div>below is the message i am getting from SER when I hung the phone on one side</div>
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<div>-------------------<this message is came from callshop Nat address>---------<"it sends bye to my SER "------------------------------------------------------</div>
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<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 sip:99106883@81.21.33.35:5060 SIP/2.0.<br>To: "99106883"<
sip:99106883@81.21.33.35:5060>;tag=78F9ECC4-166C.<br>From: "12345"<sip:12345@81.21.33.35:5060>;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: <
sip:99106883@81.21.33.35:5060;lr;nat=yes;ftag=837e5b2ff0b4cf4a>.<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"<sip:99106883@81.21.33.35:5060>;tag=78F9ECC4-166C.<br>From: "12345"<sip:12345@81.21.33.35:5060>;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"<
sip:99106883@81.21.33.35:5060>;tag=78F9ECC4-166C.<br>From: "12345"<sip:12345@81.21.33.35:5060>;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=
sip:99106883@81.21.33.35:5060 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>