[asterisk-bugs] [Asterisk 0011368]: chan_mobile does not recognize dtmf together with Authenticate or DISA
Asterisk Bug Tracker
noreply at bugs.digium.com
Fri Feb 20 07:24:25 CST 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: 2009-02-20 07:24 CST
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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
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has duplicate 0012768 Multipile issues with chan_mobile
related to 0012567 Big latency (up to 3 sec) when call wai...
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(0100441) nixon (reporter) - 2009-02-20 07:24
http://bugs.digium.com/view.php?id=11368#c100441
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Delay of voice is the best now! Thnxs!
But DTMF...
Voice is passed splendidly, but as soon as dtmf tones, each digit is torn
to fragments and heard as the interrupted parts - by ear. Therefore with
any 'dtmfskip'(20,40,50,70,100,150,200,400,500), the best from my attempts
is with dtmfskip=50:
-- Called caca_gate/3804447221992228
originally was 380445721928
In addition, if send many tones from asterisk to cellular there are
enormous interruptions between sending of digits and with every next digit
an interruption is multiplied.
Issue History
Date Modified Username Field Change
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2009-02-20 07:24 nixon Note Added: 0100441
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