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

[Asterisk Bug Tracker]

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:                     assigned
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.1 
SVN Revision (number only!): 167727 
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             2009-01-12 12:02 CST
Last Modified:              2009-01-14 11:11 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
====================================================================== 

---------------------------------------------------------------------- 
 (0097738) riksta (reporter) - 2009-01-14 11:11
 http://bugs.digium.com/view.php?id=14220#c97738 
---------------------------------------------------------------------- 
Still failing on r168500-168600 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-01-14 11:11 riksta         Note Added: 0097738                          
======================================================================