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

Asterisk Bug Tracker noreply at bugs.digium.com
Fri May 1 20:53:48 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
====================================================================== 
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:              
====================================================================== 
Date Submitted:             2007-11-25 08:42 CST
Last Modified:              2009-05-01 20:53 CDT
====================================================================== 
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...
====================================================================== 

---------------------------------------------------------------------- 
 (0104053) ughnz (reporter) - 2009-05-01 20:53
 http://bugs.digium.com/view.php?id=11368#c104053 
---------------------------------------------------------------------- 
Tested with latest trunk of addons. Same setup as before.

1. Needed to re-pair the phone before chan_mobile would connect.
2. Outgoing calls would not work. chan_mobile would dial the number then
the phone would switch to normal profile and activate handsfree on the
handset. Asterisk would report all channels are busy.
3. Incoming calls on the mobile are ignored. The phone would just ring. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-05-01 20:53 ughnz          Note Added: 0104053                          
======================================================================




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