[asterisk-bugs] [Asterisk 0012746]: reINVITE --> Ignoring this INVITE request --> Hanging up call - no reply to our critical packet.

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=12746 
====================================================================== 
Reported By:                johan
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   12746
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.19 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             05-28-2008 17:44 CDT
Last Modified:              07-03-2008 03:16 CDT
====================================================================== 
Summary:                    reINVITE --> Ignoring this INVITE request -->
Hanging up call - no reply to our critical packet.
Description: 
Under light load this does not occur. However, when Asterisk serves more
than a few calls my provider resends the initial INVITE. Then you see the
following in the log:
[May 22 18:46:05] VERBOSE[26472] logger.c: Ignoring this INVITE request

Then follows retries and about 20 seconds after the call is initiated it
is terminated by Asterisk:
[May 22 18:46:25] WARNING[26472] chan_sip.c: Maximum retries exceeded on
transmission N2EzNDEyOGFmNGNmZmY0NGM4NmJlMmZmZWIzMDg3Mzk. for seqno 1
(Critical Response)
[May 22 18:46:25] WARNING[26472] chan_sip.c: Hanging up call
N2EzNDEyOGFmNGNmZmY0NGM4NmJlMmZmZWIzMDg3Mzk. - no reply to our critical
packet.

Below the SIP HISTORY for the dialogue is attached. I've tried to sort
this out with no success. There seem to be some miscommunication between
Asterisk and my providers sip server. This can and have also occurred
during calls, exactly the same outcome. 
====================================================================== 

---------------------------------------------------------------------- 
 ranahimal - 07-03-08 03:16  
---------------------------------------------------------------------- 
Yes i resolved the issue by very differently!
But it is not the resolution for the problem.
I m not auto answering the calls but rather waiting for the upstream ACK
to come and fulfill the transaction then everything seems to work
perfectly.
The scenario is like this way we are dealing with the vpn gateway which is
giving us sip traffic so there will be delay in the transactions.
In this case there will be re-transmits going on.

-------------Problem is like this way--------------------
Problem arises when the i m auto answering the call so asterisk sends OK
to ongoing call then there comes the ACK from the upstream call and we are
sending that OK to downstream user(VPN one- with retransmits) too. 
So there is two OK transaction going on at same time from asterisk which
is very unusual and leads to destruction of call when retransmit of any two
transaction exceeds.
And u will be disconnected on connected call.. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
07-03-08 03:16  ranahimal      Note Added: 0089658                          
======================================================================