[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 Aug 14 09:06:14 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: 08-14-2007 09:06 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
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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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atca_pres - 08-14-07 09:06
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Maybe I live in a simple world, but isn't the server suppose to know at all
time when :
A transaction is unfinished
A call is over or not
With this, it would be simple : If a transaction is not finished, don't
send new ones, cancel. If the call is over (after a bye), clean whatever is
left.
I'm no programmer :)
Just wanted to thank you One47 for your efforts.
Issue History
Date Modified Username Field Change
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08-14-07 09:06 atca_pres Note Added: 0068822
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