[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 CLOSED 
====================================================================== 
https://issues.asterisk.org/view.php?id=14953 
====================================================================== 
Reported By:                rafuchoucv
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   14953
Category:                   Channels/chan_dahdi
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     closed
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:              
Resolution:                 open
Fixed in Version:           
====================================================================== 
Date Submitted:             2009-04-22 19:25 CDT
Last Modified:              2009-09-18 08:54 CDT
====================================================================== 
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.

====================================================================== 

---------------------------------------------------------------------- 
 (0110955) lmadsen (administrator) - 2009-09-18 08:54
 https://issues.asterisk.org/view.php?id=14953#c110955 
---------------------------------------------------------------------- 
I'm closing as it looks like tzafrir asked a bunch of questions, of which
the reporter didn't respond back. If they are still interested in this
issue, then feel free to reopen the issue and respond. Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-09-18 08:54 lmadsen        Note Added: 0110955                          
2009-09-18 08:54 lmadsen        Status                   new => closed       
======================================================================