[asterisk-bugs] [Asterisk 0011368]: chan_mobile does not recognize dtmf together with Authenticate or DISA
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Mar 30 15:52:59 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: 2009-03-30 15:52 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
----------------------------------------------------------------------
has duplicate 0012768 Multipile issues with chan_mobile
related to 0012567 Big latency (up to 3 sec) when call wai...
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(0102432) alexz (reporter) - 2009-03-30 15:52
http://bugs.digium.com/view.php?id=11368#c102432
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I did try the patch with Asterisk 2.6.1-rc03 and Addons 2.6.1-rc03 and
seems to me that issue is still here - I can see DTMF digits coming during
announcement play, but Authenticate is still deaf with regard to incoming
DTMF. Did I miss something somewhere? Can somebody confirm that patch fixed
the issue? I am using Ubuntu 9.04 beta with Bluez 4.32 - sound quality is
perfect, no delays, no noise, just perfect conversation. Spent yesterday
almost half an hour on a outbound call to another cell and not a sign of
any issue. Small observation though - depending on which party hangs first,
local cell phone may disconnect/reconnect. Anybody else observes the same?
Which part of the patch suppose to fix inbound DTMF problem? I am willing
to dig the code and see where extra changes should be done.
Issue History
Date Modified Username Field Change
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2009-03-30 15:52 alexz Note Added: 0102432
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