[asterisk-bugs] [Asterisk 0015893]: SIP channels never close when using tcp or tls

[Asterisk Bug Tracker]

The following issue has been UPDATED. 
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https://issues.asterisk.org/view.php?id=15893 
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Reported By:                dvossel
Assigned To:                dvossel
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Project:                    Asterisk
Issue ID:                   15893
Category:                   Channels/chan_sip/TCP-TLS
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     closed
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 218330 
Request Review:              
Resolution:                 fixed
Fixed in Version:           
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Date Submitted:             2009-09-14 14:09 CDT
Last Modified:              2009-09-17 09:51 CDT
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Summary:                    SIP channels never close when using tcp or tls
Description: 
When making a call via tcp or tls from one Asterisk box to another, the
channel on the receiving box never goes away.  If I change the peers to use
udp this does not happen. 

The channel starts out looking something like this.            
10.24.18.230     asterisk    7019c6102fbff27  0x4 (ulaw)       No      
Rx: INVITE     

Then after the call is over, the format changes to (nothing) and it never
goes away.  It is as if the channel never gets hung up.           
10.24.18.230     asterisk    7019c6102fbff27  0x0 (nothing)    No      
Rx: INVITE


---dialplan logic for the extension being called---
exten => 5200,1, Answer()
exten => 5200,n, PlayBack(tt-weasels)
exten => 5200,n, hangup()
  
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Issue History 
Date Modified    Username       Field                    Change               
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2009-09-17 09:51 dvossel        Resolution               open => fixed       
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