[asterisk-bugs] [Asterisk 0017120]: (Regression) Pickup from Grandstream BLF button ignores the context specified in Pickup command

[Asterisk Bug Tracker]

The following issue requires your FEEDBACK. 
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https://issues.asterisk.org/view.php?id=17120 
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Reported By:                Airman
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17120
Category:                   Applications/app_directed_pickup
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.2.6 
JIRA:                        
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:             2010-03-30 13:58 CDT
Last Modified:              2010-03-31 10:07 CDT
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Summary:                    (Regression) Pickup from Grandstream BLF button
ignores the context specified in Pickup command
Description: 
We have Grandstream GXP2000 phones with buttons programmed as BLFs.  In
sip.conf, we specify:

context=default
subscribecontext=blf

In the default context, we have: 

exten => _**5XX,1,Pickup(${EXTEN:2}@extensions)

When you press the BLF button and the extension is busy, the phone dials
**5XX instead of 5XX.  Starting with Asterisk 1.6.2, this stopped working. 
Instead of picking up 5XX at extensions (as specified by the Pickup command),
it tries picking up 5XX at blf.  Dialing **5XX by hand still works correctly. 
Everything worked fine in Asterisk 1.6.1.x.

Here is the output from asterisk -r -v... when we try pressing the BLF
button:

[Mar 30 09:48:26] NOTICE[30223]: chan_sip.c:19860 handle_request_invite:
Trying to pick up 520 at blf
[Mar 30 09:48:27] NOTICE[30223]: chan_sip.c:19860 handle_request_invite:
Trying to pick up 520 at blf
[Mar 30 09:48:27] NOTICE[30223]: app_directed_pickup.c:257 pickup_exec: No
target channel found for 520.

Let me know if you need any more information.  Thanks.
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---------------------------------------------------------------------- 
 (0120019) lmadsen (administrator) - 2010-03-31 10:07
 https://issues.asterisk.org/view.php?id=17120#c120019 
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Required:

* complete console output from start to finish of call
* SIP trace from the console
* related configuration of dialplan and sip.conf in order to reproduce
this issue

The information from both 1.6.1 and 1.6.2 would be ideal so we can
determine what is different between the versions. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-03-31 10:07 lmadsen        Note Added: 0120019                          
2010-03-31 10:07 lmadsen        Status                   new => feedback     
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