[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:                     feedback
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:              11-01-2007 10:42 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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---------------------------------------------------------------------- 
 jharragi - 11-01-07 10:42  
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...an additional observation...

You can find your afflicted sip peer by entering 'sip show channel (then
enter the channel listed in the WARNING - or press tab to locate it)' at
the cli. 

  SIP User agent:         Cisco-CP7960G/8.0
  Username:               6702
  Peername:               6702
  Original uri:           sip:6702 at 10.0.10.248:5060                       
       

...restarting that peer clears the error (on this, and most occurances
I've seen, there has been no noticable audio problem on the peer). 

Issue History 
Date Modified   Username       Field                    Change               
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11-01-07 10:42  jharragi       Note Added: 0072894                          
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