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

Tue Apr 21 19:48:59 CDT 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:              2009-04-21 19:48 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.
======================================================================
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...
====================================================================== 

---------------------------------------------------------------------- 
 (0103572) alexz (reporter) - 2009-04-21 19:48
 http://bugs.digium.com/view.php?id=11368#c103572 
---------------------------------------------------------------------- 
Ok. My current setup is:
Debian 5 Lenny
Bluez 4.36 (some dancing required in order not to build ipctest, which
fails to compile)
OpenVox TDM card with FXS/FXO
Asterisk trunk 189770
Add-ons trunk 878

Problem 1: On both inbound/outbound calls asterisk CLI craps out
immediately
Problem 2: On outbound call there is one way audio - called party doesn't
hear calling party (everything is OK with stock ast and addons with the
same setup above)
Problem 3: On outbound call if calling party hangs then call is not
hanging
Problem 4: On outbound call /var/log/asterisk/full is full of the
following messages: 
[Apr 21 20:22:15] VERBOSE[2922] pbx.c:     -- Executing
[s at macro-dialout-trunk:26] ^[[1;36
mDial^[[0m("^[[1;35mDAHDI/1-1^[[0m",
"^[[1;35mMOBILE/g1/6137978600,300,^[[0m") in new stac
k
[Apr 21 20:22:15] VERBOSE[2922] app_dial.c:     -- Called g1/6137978600
[Apr 21 20:22:24] ERROR[2922] res_timing_dahdi.c: Failed to configure
DAHDI timing fd for
0 sample timer ticks
[Apr 21 20:22:24] ERROR[2922] res_timing_dahdi.c: Failed to configure
DAHDI timing fd for
0 sample timer ticks
[Apr 21 20:22:24] ERROR[2922] res_timing_dahdi.c: Failed to configure
DAHDI timing fd for
0 sample timer ticks
..... and so on
AND THERE ARE A LOT OF THEM (17 megs actually)

all problems above are not observed with stock ast/addons

Problem 5: On inbound call sometimes it picks up digits in authenticate
(verbal confirmation from ast that password was accepted - can't confirm
the same from CLI since it dies), but most of the cases log is FULL (again
mega and megs) of 

[Apr 21 20:38:57] ERROR[2957] chan_mobile.c: read error 9
[Apr 21 20:38:57] ERROR[2957] chan_mobile.c: read error 9
[Apr 21 20:38:57] ERROR[2957] chan_mobile.c: read error 9
.........

I am completely lost here.

1) Stock ast/addons produce stable outbound call, but no dtmf digits even
in background
2) latest rc of ast/addons often produce garbled CLI output, but at least
I can see dtmf digits in background
3) Trunk sometimes authenticates, but the rest of functionality is broken

Where it leaves us?

mnicholson - can you please provide us with EXACT ast and addons trunk
version that you tested with? We would like to repeat your exact testing
(we already abandoned Ubuntu in order to have the same setup as you do) 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-04-21 19:48 alexz          Note Added: 0103572                          
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