[asterisk-bugs] [Asterisk 0010890]: When parking lot ring back times out, error is generated, line is hung up and timeout extension isn't reached.

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=10890 
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Reported By:                kenw
Assigned To:                
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Project:                    Asterisk
Issue ID:                   10890
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.4.11  
SVN Branch (only for SVN checkouts, not tarball releases): N/A  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             10-04-2007 18:28 CDT
Last Modified:              03-12-2008 16:31 CDT
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Summary:                    When parking lot ring back times out, error is
generated, line is hung up and timeout extension isn't reached.
Description: 
The situation is repeated as follows:

1. Call is placed into parking lot
2. Call timesout from parking lot and rings back extension that put it
there to begin with
3. After http://bugs.digium.com/view.php?id=8#50 seconds of ringing back to
extension that placed the call in
the parking lot, the [park-dial] 't' extension should be invoked, instead
the following errors show up in the Asterisk-CLI:

[Oct  4 17:22:47] WARNING[7986]: chan_sip.c:12037 handle_response_invite:
Re-invite to non-existing call leg on other UA. SIP dialog
'224fe567089888351edffad76cdf16d9 at 10.200.26.202'. Giving up.
[Oct  4 17:38:18] WARNING[7986]: chan_sip.c:12536 handle_response: Remote
host can't match request CANCEL to call
'4c93b03d7f87f05950d21bd971edd97b at 10.200.26.202'. Giving up.

I'm not sure if this is a parking lot or SIP issue, but beings the errors
are from chan_sip.c I chose SIP Transfers.

I've attached CLI with debug & verbose set to 4 as well as sip debug
enabled.  I've also attached the parts of features.conf, extensions.conf &
sip.conf that apply.  Problem was noticed in 1.4.10.1, upgrade to 1.4.12
made no difference.

(1.4.12 not an option on Asterisk version btw).
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---------------------------------------------------------------------- 
 davidw - 03-12-08 16:31  
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The trace indicates that the session id (SIP tag?) *is* getting stripped
from channel name, although I'm not sure where. 

Issue History 
Date Modified   Username       Field                    Change               
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03-12-08 16:31  davidw         Note Added: 0083862                          
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