[asterisk-bugs] [Asterisk 0009305]: [patch] REINVITE before 200ok causes a call to be ended

noreply at bugs.digium.com noreply at bugs.digium.com
Tue Sep 25 13:11:15 CDT 2007


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=9305 
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Reported By:                atca_pres
Assigned To:                oej
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Project:                    Asterisk
Issue ID:                   9305
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     confirmed
Asterisk Version:            SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  1.4 
SVN Revision (number only!): 58957 
Disclaimer on File?:        Yes 
Request Review:              
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Date Submitted:             03-16-2007 13:28 CDT
Last Modified:              09-25-2007 13:11 CDT
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Summary:                    [patch] REINVITE before 200ok causes a call to be
ended
Description: 
When Flash hooking:
Box 1 sends a INVITE with a contact 0.0.0.0 (Hold)
Asterisk sends a invite to box 2
Box 2 sends a trying
Asterisk sends a second invite to box 2 with a differetn CSeq, Branch and
Session version(SDP). This trigger a 500 msg.
The 500 is triggered because you cannot (according to RFC) send a second
invite when you have an unfinished dialog

Note : No RTP Portal
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Relationships       ID      Summary
----------------------------------------------------------------------
related to          0009142 Placing a call on hold sends two INVITE...
related to          0009209 race condition in sip hangup with reinv...
related to          0009649 BYE calls too fast when connected to a ...
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---------------------------------------------------------------------- 
 jcmoore - 09-25-07 13:11  
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I was having a very similar problem where Asterisk would send a BYE
immediately after receiving a "100 Trying" in response to a re-INVITE. 
sip_reinvite6.diff seems to have fixed this for me.  Can we have some
others look over this patch and try to get it in to 1.4/trunk ASAP as it's
a pretty serious bug in the SIP stack.

The patch as it is seems to be correct but I'd prefer we had more eyes on
it before committing it. 

Issue History 
Date Modified   Username       Field                    Change               
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09-25-07 13:11  jcmoore        Note Added: 0071079                          
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