[asterisk-bugs] [Asterisk 0016928]: [patch] The Dial c option returns answered elsewhere if the dial timeout occurs (only tested using SIP)

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16928 
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Reported By:                DennisD
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16928
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           1.6.2.3-rc2 
JIRA:                       SWP-998 
Regression:                 Yes 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-02-28 13:32 CST
Last Modified:              2010-05-26 19:58 CDT
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Summary:                    [patch] The Dial c option returns answered elsewhere
if the dial timeout occurs (only tested using SIP)
Description: 
[2010-02-28 14:04:53.293] DEBUG[10201] channel.c: Hanging up channel
'SIP/2003-0000002a'
[2010-02-28 14:04:53.293] DEBUG[10201] chan_sip.c: This call was answered
elsewhere[2010-02-28 14:04:53.293] DEBUG[10201] chan_sip.c: Hangup call
SIP/2003-0000002a, SIP callid 3f786f6a497f3cf57fe1e56e0db536ad at example.com
[2010-02-28 14:04:53.293] DEBUG[10201] chan_sip.c: Updating call counter
for outgoing call

This call was not answered anywhere.
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 (0122537) DennisD (reporter) - 2010-05-26 19:58
 https://issues.asterisk.org/view.php?id=16928#c122537 
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This is on 1.6.2.8-rc1.  The patch doesn't change the behavior.

Does not show there was a missed call on the GXP2000:


Dial(SIP/2024,30,c).  The call was not answered elsewhere:

CANCEL sip:2024 at 192.168.224.87:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.224.1:5060;branch=z9hG4bK5cd24dae;rport
Max-Forwards: 70
From: "Dennis' Office" <sip:2009 at dennisd.com>;tag=as7b4da91a
To: <sip:2024 at 192.168.224.87:5060;transport=udp>
Call-ID: 2097af5044935e6057cbfc38294ac5e4 at dennisd.com
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Reason: SIP;cause=200;text="Call completed elsewhere"
Content-Length: 0



Shows there was a missed call on the GXP2000:

Dial(SIP/2024,30).  The call was not answered elsewhere:

CANCEL sip:2024 at 192.168.224.87:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.224.1:5060;branch=z9hG4bK5997d6a5;rport
Max-Forwards: 70
From: "Dennis' Office" <sip:2009 at dennisd.com>;tag=as3e5ccd5d
To: <sip:2024 at 192.168.224.87:5060;transport=udp>
Call-ID: 2ea2eaaa4b97e5560702fe297d7e2dd8 at dennisd.com
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0



I dialed from 2009 to 2004 and let it ring a couple of times and then hung
up.

Dennis 

Issue History 
Date Modified    Username       Field                    Change               
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2010-05-26 19:58 DennisD        Note Added: 0122537                          
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