[asterisk-bugs] [Asterisk 0014953]: Last digit missing when dialing out to pstn and echotraining=yes or echotraining=xx

[Asterisk Bug Tracker]

The following issue has been SUBMITTED. 
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http://bugs.digium.com/view.php?id=14953 
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Reported By:                rafuchoucv
Assigned To:                
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Project:                    Asterisk
Issue ID:                   14953
Category:                   Channels/chan_dahdi
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.0.7 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-04-22 19:25 CDT
Last Modified:              2009-04-22 19:25 CDT
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Summary:                    Last digit missing when dialing out to pstn and
echotraining=yes or echotraining=xx
Description: 
In asterisk 1.6.0.9 and dahdi 2.1.0.4 after I activated echotraining=yes,
calls to pstn through a Digium TDM400P failed because the last digit of the
number I dialed was not send, I can't even hear the busy tone produced by
the wrong number dialed. echocancel parameter does not make any diference
Commenting out echotraining=yes solved the problem, but i need
echotraining.

The log shows:

[Apr 22 19:30:58] DEBUG[5110] chan_dahdi.c: Dialing '04141060473'
[Apr 22 19:30:58] DEBUG[5110] chan_dahdi.c: Deferring dialing...
[Apr 22 19:30:58] DEBUG[5110] devicestate.c: Notification of state change
to be queued on device/channel DAHDI/1
[Apr 22 19:30:58] VERBOSE[5110] logger.c:     -- Called 1/04141060473
[Apr 22 19:30:58] DEBUG[5062] devicestate.c: Changing state for DAHDI/1 -
state 2 (In use)
[Apr 22 19:30:58] DEBUG[5073] app_queue.c: Device 'DAHDI/1' changed to
state '2' (In use) but we don't care because they're not a member of any
queue.
[Apr 22 19:30:58] DEBUG[5110] channel.c: Set channel DAHDI/1-1 to read
format gsm
[Apr 22 19:30:58] DEBUG[5110] channel.c: Set channel SIP/dfreepbx-01a081a0
to read format slin
[Apr 22 19:30:58] DEBUG[5110] channel.c: Set channel DAHDI/1-1 to write
format slin
[Apr 22 19:30:59] DEBUG[5110] chan_dahdi.c: Exception on 15, channel 1
[Apr 22 19:30:59] DEBUG[5110] chan_dahdi.c: Got event Hook Transition
Complete(12) on channel 1 (index 0)
[Apr 22 19:30:59] DEBUG[5110] chan_dahdi.c: Sent deferred digit string:
T0414106047

at the end is the number with the last digit stripped.

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Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-04-22 19:25 rafuchoucv     New Issue                                    
2009-04-22 19:25 rafuchoucv     Asterisk Version          => 1.6.0.7         
2009-04-22 19:25 rafuchoucv     Regression                => No              
2009-04-22 19:25 rafuchoucv     SVN Branch (only for SVN checkouts, not tarball
releases) => N/A             
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