[asterisk-bugs] [Asterisk 0011368]: chan_mobile does not recognize dtmf together with Authenticate or DISA
Asterisk Bug Tracker
noreply at bugs.digium.com
Tue Apr 28 11:43:24 CDT 2009
A NOTE has been added to this issue.
======================================================================
http://bugs.digium.com/view.php?id=11368
======================================================================
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: 2009-04-28 11:43 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 "No audio" on incoming blueto...
related to 0012567 Big latency (up to 3 sec) when call wai...
======================================================================
----------------------------------------------------------------------
(0103882) mnicholson (administrator) - 2009-04-28 11:43
http://bugs.digium.com/view.php?id=11368#c103882
----------------------------------------------------------------------
@franciscoce
Can you test with the patch from issue http://bugs.digium.com/view.php?id=14878
to see if it fixes the
first call issues.
Issue History
Date Modified Username Field Change
======================================================================
2009-04-28 11:43 mnicholson Note Added: 0103882
======================================================================
More information about the asterisk-bugs
mailing list