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 tohis 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@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@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@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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