[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:              01-03-2008 10:44 CST
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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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---------------------------------------------------------------------- 
 kenw - 01-03-08 10:44  
---------------------------------------------------------------------- 
I've gone through the logs and code, and it appears my problem has to do
with the 408 response from the phone in chan_sip.c.

As a recap, a call comes in, is put into the parking lot.  X seconds later
it rings back to the phone that originally answered.  If this phone doesn't
pick the call up in 60 seconds, Asterisk drops the call.  Below is the
code.

        case 408: /* Request timeout */
        case 481: /* Call leg does not exist */
                /* Could be REFER caused INVITE with replaces */
                ast_log(LOG_WARNING, "Re-invite to non-existing call leg
on other UA. SIP dialog '%s'. Giving up.\n", p->callid);

                xmitres = transmit_request(p, SIP_ACK, seqno,
XMIT_UNRELIABLE, FALSE);
                if (p->owner)
                        ast_queue_control(p->owner,
AST_CONTROL_CONGESTION);
                sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
                break;


I've tried everything in extensions as far as exten,s,1 - exten,t,1 -
exten,h,1 - exten,.s, etc....literally dozens of 'capture a bad call'
extensions in all different contexts, regardless it drops the line.

So is this the way a 408 response to a parking lot ring back is expected
to be handled? 

Issue History 
Date Modified   Username       Field                    Change               
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01-03-08 10:44  kenw           Note Added: 0076254                          
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