What does the Contact header in the INVITE look like? The BYE is actually a separate SIP dialog, and the SIP server has to send it to the address in the Contact header not the address the INVITE came from. NAT or a misbehaving SIP client could potentially screw that up by sending a bad Contact header, which might mean you're not seeing it in Wireshark because it's not being sent where you expect. -Steve On 7 September 2011 14:50, Simon Leck <simon0922@gmail.com> wrote:
Hi Peder, ****
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Thanks again for your swift reply. FS is running on a Public IP address. The UA is also running on a Public IP address. Yes, I did used wireshark to do a sip trace on FS and see that the bye was not present. ****
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One reason I could think of could be the on-hook and off-hook time interval or frequency for off-net calling is different that’s why it could not hang up? Or what else could be affecting.****
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Thanks in advanced for helping me out and I look forward to reply. ****
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Btw, are you on IRC and if you are can I chat with you?****
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Many thanks ****
Simon Leck****
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*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Peder *Sent:* Wednesday, September 07, 2011 8:42 PM
*To:* 'FreeSWITCH Users Help' *Subject:* Re: [Freeswitch-users] - freeswitch bye issue****
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Public IP on what? The phone, or FS? Did you do a packet capture at the FS box, or at the phone? If you did it at the phone, it still doesn’t prove that FS didn’t send a bye.****
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*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Simon Leck *Sent:* Tuesday, September 06, 2011 9:47 PM *To:* 'FreeSWITCH Users Help' *Subject:* Re: [Freeswitch-users] - freeswitch bye issue****
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Hi Pedar, ****
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Thanks for your reply. At first I thought it was NAT issue so to isolate the problem I used a Public IP but I still get the same result? I have used wireshark to do a trace but I don’t see FS sending a bye to A?****
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Kindly advice me on how else could I solved this issue. On the NAT Part, I have enable RTP keep alive as well. Under RTCP, I have the keep alive turned on as well?****
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Which other place are there setting I can configure with?****
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Thanks in advanced Pedar for your kindest help****
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Many thanks****
Simon Leck****
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*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Peder *Sent:* Wednesday, September 07, 2011 1:57 AM *To:* 'FreeSWITCH Users Help' *Subject:* Re: [Freeswitch-users] - freeswitch bye issue****
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It should happen automatically. Is A behind NAT? Sounds like a firewall/NAT may be dropping the bye. Also, have you done a debug on FS to see if it actually sends a bye or not to A?****
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*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Simon Leck *Sent:* Tuesday, September 06, 2011 11:42 AM *To:* 'FreeSWITCH Users Help' *Subject:* Re: [Freeswitch-users] - freeswitch bye issue****
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Hi Everyone, kindly help if you can ****
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At the moment I did encounter a issue, A leg user called a b leg user and when b leg user hanged up (Bye is not send by freeswitch to the A leg user) the only way I could hanged up this call to hanged up the A leg user manually? Would be great if someone could enlighten me on how this can be achieved?****
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Thanks in advanced to anyone for helping me out****
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Thanks again****
Simon Leck****
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