[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. 
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http://bugs.digium.com/view.php?id=14220 
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Reported By:                riksta
Assigned To:                putnopvut
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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:              
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Date Submitted:             2009-01-12 12:02 CST
Last Modified:              2009-01-16 09:05 CST
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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
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 (0098012) putnopvut (administrator) - 2009-01-16 09:05
 http://bugs.digium.com/view.php?id=14220#c98012 
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No, another capture won't be necessary. Thanks for following up. If
something in the code doesn't jump out at me today while I'm looking, I may
be forced to place a patch up here which prints out a bunch of debug
information as SIP packets are being formed so I can track exactly what's
happening. Hopefully, I won't have to do that though. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-01-16 09:05 putnopvut      Note Added: 0098012                          
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