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

noreply at bugs.digium.com noreply at bugs.digium.com
Sun Jun 1 10:04:44 CDT 2008


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:                     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:             05-28-2008 17:44 CDT
Last Modified:              06-01-2008 10:04 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. 
====================================================================== 

---------------------------------------------------------------------- 
 rjain - 06-01-08 10:04  
---------------------------------------------------------------------- 
I don't see debug entries in the log. Pls. post unstripped log. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
06-01-08 10:04  rjain          Note Added: 0087611                          
======================================================================




More information about the asterisk-bugs mailing list