[asterisk-bugs] [Asterisk 0017692]: [patch] subchannel remains half-open after call transfer

[Asterisk Bug Tracker]

The following issue has been UPDATED. 
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https://issues.asterisk.org/view.php?id=17692 
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Reported By:                jmhunter
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17692
Category:                   Channels/chan_skinny
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 278578 
Request Review:              
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Date Submitted:             2010-07-23 18:02 CDT
Last Modified:              2010-07-27 12:32 CDT
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Summary:                    [patch] subchannel remains half-open after call
transfer
Description: 
Call transfers don't seem to work properly using chan_skinny.

To reproduce:

 [ jmhtest1 is a Skinny phone (Cisco 7905G)   ]
 [ 5022 is a DAHDI-connected analogue handset ]

1. jmhtest1 picks up handset and calls 6001 (music on hold)
2. jmhtest1 presses 'Transfer'
3. jmhtest1 dials 5022
4. 5022 picks up
5. jmhtest1 presses 'Transfer'
6. jmhtest1 replaces handset in cradle

At this point, jmhtest1 still shows "Connected" on its display, and now
won't give out a dialtone when handset is picked up. Pressing the 'EndCall'
softkey leads to messages such as:
[Jul 23 23:26:21] WARNING[19173]: chan_skinny.c:1673
find_subchannel_by_instance_reference: Could not find subchannel with
reference '1' on 'jmhtest1'
 Received Softkey Event: End Call(1/1)
[Jul 23 23:26:22] WARNING[19173]: chan_skinny.c:1673
find_subchannel_by_instance_reference: Could not find subchannel with
reference '1' on 'jmhtest1'
 Received Softkey Event: End Call(1/1)

Hanging up 5022 (the recipient of the call transfer) does not make any
difference, jmhtest1 still does not work and I have to reboot the phone.

Trying a blind transfer (missing out step 5 above) seems to be a bit hit
and miss - this either works fine, or places the call on hold. It's done
both for me at various times, I think it's to do with whether Asterisk has
crashed recently (see https://issues.asterisk.org/view.php?id=17680).

In addition, at step 3 above, I can hear the audio from the original call
to 6001, as well as the audio from 5022 (where I am transferring to). That
surely shouldn't happen - jmhtest1 should talk to 5022 and only 5022 at
that point, right?
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Issue History 
Date Modified    Username       Field                    Change               
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2010-07-27 12:32 lmadsen        Description Updated                          
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