[asterisk-bugs] [Asterisk 0014220]: SIP INVITE packets are incorrectly truncated with 1.6.1 svn after approx 1020 characters

Asterisk Bug Tracker noreply at bugs.digium.com
Fri Jan 16 15:20:39 CST 2009


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=14220 
====================================================================== 
Reported By:                riksta
Assigned To:                putnopvut
====================================================================== 
Project:                    Asterisk
Issue ID:                   14220
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.1 
SVN Revision (number only!): 167727 
Request Review:              
====================================================================== 
Date Submitted:             2009-01-12 12:02 CST
Last Modified:              2009-01-16 15:20 CST
====================================================================== 
Summary:                    SIP INVITE packets are incorrectly truncated with
1.6.1 svn after approx 1020 characters
Description: 
Since updating from 1.6.1b4 to SVN-branch-1.6.1-r167727 i can no longer
make calls to my SIP trunk provider if my call has a long CALLERID(name)
string.

The sip trunk provider's techie says that my SIP INVITE packets are being
truncated... eg if i set the callerid name as a short string and the INVITE
packet was 1017 characters, the call goes through perfectly.

If i set the callerid name as a longer string and the INVITE packet was
1020 chars the call fails to be sent.

In #asterisk-dev Corydon76 explains that it is probably to do with the
fact that SIP INVITES are no longer are built in a static buffer (That was
part of the ast_str work) 


I will attach a sip debug log file of a successful and unsuccessful call
====================================================================== 

---------------------------------------------------------------------- 
 (0098051) putnopvut (administrator) - 2009-01-16 15:20
 http://bugs.digium.com/view.php?id=14220#c98051 
---------------------------------------------------------------------- 
All right. At this point, the problem is fully identified, and is a bit
more complex to solve than I had originally thought. What I am going to do
is post a workaround patch here for you to use for now. Later, when I have
a full fix for the issue, I'll post it here. Thanks for your patience on
this one. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-01-16 15:20 putnopvut      Note Added: 0098051                          
======================================================================




More information about the asterisk-bugs mailing list