[asterisk-bugs] [Asterisk 0011368]: chan_mobile does not recognize dtmf together with Authenticate or DISA

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Feb 26 16:35:33 CST 2009


A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=11368 
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Reported By:                bt047265
Assigned To:                mnicholson
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Project:                    Asterisk
Issue ID:                   11368
Category:                   Addons/chan_mobile
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     ready for testing
Asterisk Version:            SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 89454 
Request Review:              
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Date Submitted:             2007-11-25 08:42 CST
Last Modified:              2009-02-26 16:35 CST
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Summary:                    chan_mobile does not recognize dtmf together with
Authenticate or DISA
Description: 
Hello,

chan_mobile is configured according to the documentation. Incoming and
outgoing calls are working via the new channel "Mobile". 

Mobile.conf:

[adapter]
id=stick1
address=00:08:F4:16:3A:E2
[SGH-F200]
;address=00:1D:25:73:0E:76
address=00:1B:59:14:77:38
port=4
context=incoming_mobile
adapter=stick1
dtmfskip=50

This dialplan was added to the extensions.conf:

[incoming_mobile]
exten => _!,1,Answer()
exten => _!,n,Wait(1)
exten => _!,n,Verbose(${EXTEN})
exten => _!,n,Verbose(${CALLERID})
exten => _!,n,Authenticate(1234)
exten => _!,n,Background(vm-enter-num-to-call)
exten => _!,n,DISA(no-password,phones,"sipgate" <7001>)

No DTMF tones are regocnized by the Authenticate function.  If the same
context is assigned to the SIP channel Authenticate and DISA is working.

Attached the output of /var/log/asterisk/full for:

- incoming mobile authenticate
- icoming mobile to SIP extension
- incoming SIP authenticate

If the incoming call from the mobile is directly routed to an SIP
extension, DTMF is sended to the SIP extension.
======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
has duplicate       0012768 Multipile issues with chan_mobile
related to          0012567 Big latency (up to 3 sec) when call wai...
====================================================================== 

---------------------------------------------------------------------- 
 (0100854) mnicholson (administrator) - 2009-02-26 16:35
 http://bugs.digium.com/view.php?id=11368#c100854 
---------------------------------------------------------------------- 
I reproduced your experiment here, and I initially got the same results. 
When I stripped out dtmf processing, the recording of the tones sounded
perfect.  This means that chan_mobile is accurately passing the tones into
the dtmf processing engine, and the artifacts in your recording are
actually the dtmf processing engine's attempts to strip out the dtmf tone
from the audio stream.  I will look into this further, but basically this
means the problems are in the dtmf detection code rather then in chan
mobile.

You may see similar issues with tones if you test with a sip phone using
inband dtmf. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-02-26 16:35 mnicholson     Note Added: 0100854                          
======================================================================




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