[asterisk-bugs] [Asterisk 0011916]: Asterisk don't get the BYE packet from callee

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=11916 
====================================================================== 
Reported By:                mnnojd
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   11916
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 102238 
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             02-04-2008 04:47 CST
Last Modified:              02-07-2008 02:40 CST
====================================================================== 
Summary:                    Asterisk don't get the BYE packet from callee
Description: 
Hi,

There is problem when callee hangup the call which is connected between a
caller, Asterisk and callee. Asterisk don't get the bye packet from callee.
But if the caller hangup the call everything works fine.




======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
has duplicate       0011939 The "contact" section of the ...
====================================================================== 

---------------------------------------------------------------------- 
 mnnojd - 02-07-08 02:40  
---------------------------------------------------------------------- 
Hi,

Here I send you some logs.

There is two sip-clients and they are registered as;

A = sipcon1 at 192.168.0.1, this is on ip-address 192.168.0.31
B = allanec at 192.168.0.1, this is on ip-address 192.168.0.33
Asterisk at ip-address 192.168.0.32

and the sip-server is at ip-address 192.168.0.1. This environment is local
but I have tried on public ip-addresses with same results.

In this case A calls to Asterisk with address 11111 at 192.168.0.32 and in
Asterisk the extension 11111 at 192.168.0.32 switches to allanec at 192.168.0.1
which is B.

I collected some logs from Asterisk(file asterisk.log and sipsettings.log)
which you asked for, but if there is some missing information, just tell
me.

I also collected some logs with Wireshark(file Wireshark_at_asterisk) in
same computer as Asterisk. In the logs with Wireshark is it interesting to
se the sip invite from Asterisk to B, there you find in header "Contact"
there is port number 0... 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
02-07-08 02:40  mnnojd         Note Added: 0081836                          
======================================================================