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

[Asterisk Bug Tracker]

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:              2009-04-27 17:23 CDT
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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.
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Relationships       ID      Summary
----------------------------------------------------------------------
related to          0011801 mobile to asterisk audio stability stro...
has duplicate       0012768 Multipile issues with chan_mobile
has duplicate       0011556 &quot;No audio&quot; on incoming blueto...
related to          0012567 Big latency (up to 3 sec) when call wai...
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---------------------------------------------------------------------- 
 (0103859) alexz (reporter) - 2009-04-27 17:23
 http://bugs.digium.com/view.php?id=11368#c103859 
---------------------------------------------------------------------- 
I tested virtually with all the possible permutations of different
trunk/release/rc versions of ast and addons, but the only combination that
works for us is release, where DTMF is not working (as we know). All of
rc/trunk do produce log full of different errors (regardless to the
patch/no patch applied in case of rc of addons) including timing. We
probably better wait till official release where fix will be submitted, and
then will test it again. Too much time spent chasing the ghosts. We do go
through the same setup routine (Ubuntu/Debian + LAMP + OpenSSH + Bluez
(4.32-4.37) + Release DAHDI + Ast_X + Addons_X + FreePBX) and the only case
when we can reliably plece/receive the call is release versions of
ast/addons. We tried today exact the same ast r185846 + addons r878 with
absolutely the same result as before. Another possible issue is that whole
setup procedure that we are using is not valid anymore for trunk/rc
versions of Asterisk.

So, our understanding is that fix will be included in the next release of
addons 1.6.0 and 1.6.1 - is it correct? WHEN APPROXIMATELY IT WILL HAPPEN? 

Issue History 
Date Modified    Username       Field                    Change               
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2009-04-27 17:23 alexz          Note Added: 0103859                          
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