[asterisk-bugs] [Asterisk 0017613]: [patch] After a blind transfer by the calling party the transferees peer cannot be dialed again within the same call

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17613 
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Reported By:                ramonpeek
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17613
Category:                   Channels/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           1.4.33 
JIRA:                       SWP-1833 
Regression:                 No 
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:             2010-07-09 05:38 CDT
Last Modified:              2010-12-15 19:26 CST
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Summary:                    [patch] After a blind transfer by the calling party
the transferees peer cannot be dialed again within the same call
Description: 
After a blind transfer by the calling party the transferees peer cannot be
dialed again within the same call. This ONLY occurs when dialing through a
Local channel.

Asterisk will show this warning on the CLI>

[Jul  9 12:18:37] WARNING[27865]: app_dial.c:1296 dial_exec_full: Skipping
dialing interface 'SIP/401' again since it has already been dialed


NOTE: See steps to reproduce (in advanced view)
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 (0129639) cmaj (reporter) - 2010-12-15 19:26
 https://issues.asterisk.org/view.php?id=17613#c129639 
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I can sort-of reproduce this in 1.8.1 on Polycoms as follows:

1. Extension 701 is watching buddy 702.
2. Buddy 702 starts ringing and extension 701 hits the "Pickup" soft key.
3. Extension 701 hits "Transfer" hard key then "Blind" soft key then
enters "702".
4. Dial() immediately fails with DIALSTATUS=INVALIDARGS.  The CLI WARNING
is the same as OP, "Skipping dialing interface 'SIP/401' again since it has
already been dialed."

So I worked around it by adding the 'g' option to Dial() then checking the
DIALSTATUS variable set internally by app_dial.c:

Dial(SIP/1234,30,g);
if("${DIALSTATUS}"="INVALIDARGS") {
Voicemail(1234);
}


I know this is not a solution addressing the original bug directly, but it
might help somebody coming at this "blind transfer failure to a
twice-dialed extension problem" from a slightly different error scenario. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-12-15 19:26 cmaj           Note Added: 0129639                          
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