[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:              2009-01-15 10:54 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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 (0097925) putnopvut (administrator) - 2009-01-15 10:54
 http://bugs.digium.com/view.php?id=14220#c97925 
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oej, after talking with Kevin, he said that the adjustment he made was to
allow for incoming INVITEs to have more than 64 lines in the SDP. This
issue seems to be dealing with outgoing INVITEs being truncated at 1024
characters. My immediate thought on this is that the ast_str object which
contains the request contents is not expanding as it is supposed to.

I see that riksta has attached pcap files here, so I will examine those in
comparison to what the SIP debug from Asterisk says. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-01-15 10:54 putnopvut      Note Added: 0097925                          
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