[asterisk-bugs] [Asterisk 0012792]: Dropped calls with: Maximum retries exceeded on transmission

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=12792 
====================================================================== 
Reported By:                digmaster
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   12792
Category:                   Channels/chan_sip/General
Reproducibility:            sometimes
Severity:                   major
Priority:                   normal
Status:                     new
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:             06-05-2008 01:23 CDT
Last Modified:              06-05-2008 04:27 CDT
====================================================================== 
Summary:                    Dropped calls with: Maximum retries exceeded on
transmission
Description: 
We get lots of dropped calls with the following warning in the logs:

chan_sip.c: Maximum retries exceeded on transmission
2cc62c79934cc54c at 192.168.1.102 for seqno 35083 (Critical Response).

We use asterisk 1.4.19 with trixbox, a beronet 2-port ISDN PCI card and
some Grandstream phones. This happens on 2 out of 10 calls, but when it
happens it drops the call while the two people are actively talking.

After further tests, we thought that the UDP port was timing out, but that
wasn't the case since communication seemed ok.


====================================================================== 

---------------------------------------------------------------------- 
 davidw - 06-05-08 04:27  
---------------------------------------------------------------------- 
Which ends of the failing call are using SIP?  Have you done anything
special with the call.  (See http://bugs.digium.com/view.php?id=12548 for an
example of one "something
special" that provokes this symptom.)

You may well need to follow the full procedures for reporting SIP problems
<http://www.asterisk.org/developers/bug-guidelines>, but, in this case,
using sip history just before the call fails (type "sip history", on the
CLI, before the call, and "sip show history <SIP channel ID>" just before
the final failure) will tell you which SIP operation is failng.  Generally
this error presents about 20 seconds after things started to go wrong and
there are multiple attempts to send the failing request. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
06-05-08 04:27  davidw         Note Added: 0087835                          
======================================================================