[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:                     assigned
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:              2009-01-14 15:52 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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 (0097829) putnopvut (administrator) - 2009-01-14 15:52
 http://bugs.digium.com/view.php?id=14220#c97829 
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I've been looking at the two files you attached, specifically looking at
the outgoing INVITEs from Asterisk. The INVITEs do not appear to be
truncated. Now this could just be that Asterisk is not actually sending the
same thing that is being printed when executing a sip debug on the CLI.

So here are some things that may be helpful:

1) Compare a capture using something like wireshark and what Asterisk
shows when executing a sip debug. If they are different, then upload a
.pcap file showing the actual INVITE that Asterisk is sending.

2) Is openSER running locally or remotely? I keep seeing that openSER is
what is responding to our INVITEs, and I just was wondering if the problem
might lie there instead of in Asterisk. The fact that the problem started
occurring when you made an upgrade to Asterisk makes me doubt this is the
case, but it doesn't hurt to ask.

I'll also take a look in the code, but your help would be very much
appreciated here. 

Issue History 
Date Modified    Username       Field                    Change               
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2009-01-14 15:52 putnopvut      Note Added: 0097829                          
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