[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 04:32:17 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: 2009-05-01 04:32 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
----------------------------------------------------------------------
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...
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----------------------------------------------------------------------
(0104015) ughnz (reporter) - 2009-05-01 04:32
http://bugs.digium.com/view.php?id=11368#c104015
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Tested the audio patch with:
asterisk 1.6.1.0
addons 1.6.1.0
bluez 3.36
kernel 2.6.18-128.1.6.el5
Audio delay has gone. Still get a few hci_scodata_packet: hci0 SCO packet
for unknown connection handle messages in the kernel log.
DISA will still not work. It will either crash asterisk, hang with no
dialtone or give dialtone then hang the channel, only a asterisk kill will
release the channel.
Issue History
Date Modified Username Field Change
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2009-05-01 04:32 ughnz Note Added: 0104015
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