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,

 

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.

 

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.

 

Thanks in advanced for helping me out and I look forward to reply.

 

Btw, are you on IRC and if you are can I chat with you?

 

Many thanks

Simon Leck

 

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

 

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.

 

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

 

Hi Pedar,

 

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?

 

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?

 

Which other place are there setting I can configure with?

 

Thanks in advanced Pedar for your kindest help

 

Many thanks

Simon Leck

 

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

 

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?

 

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

 

Hi Everyone, kindly help if you can

 

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?

 

Thanks in advanced to anyone for helping me out

 

Thanks again

Simon Leck

 

 

 

 

 

 

 



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