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

[noreply at bugs.digium.com]

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:                dbowerman
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Project:                    Asterisk
Issue ID:                   11368
Category:                   Addons/chan_mobile
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     assigned
Asterisk Version:            SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 89454 
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             11-25-2007 08:42 CST
Last Modified:              04-07-2008 17:53 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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---------------------------------------------------------------------- 
 aversa - 04-07-08 17:53  
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I used the latest trunk for asterisk 1.6 and a Nokia 2630. I call the Nokia
2630
and asterisk anwser, DTMF tones are accepted while asterisk plays an
announcement.
Nothing is accepted by a Read() command.

I noticed that in 
main/channel.c, in the function: 
static struct ast_channel *ast_waitfor_nandfds_classic
waits until timeout (10 seconds in my case) in this line:

res = poll(pfds, max, rms); 

and nothing is detected.
  
But if an announcement is played, the same line correctly reports the DTMF
tones. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
04-07-08 17:53  aversa         Note Added: 0085110                          
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