[asterisk-bugs] [Asterisk 0011764]: MixMonitor doesn't work right with SIP and FLASH on FXS channels
noreply at bugs.digium.com
noreply at bugs.digium.com
Mon Jan 14 08:18:08 CST 2008
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=11764
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Reported By: viniciusfontes
Assigned To:
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Project: Asterisk
Issue ID: 11764
Category: Applications/app_mixmonitor
Reproducibility: always
Severity: minor
Priority: normal
Status: new
Asterisk Version: 1.4.17
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 01-14-2008 04:20 CST
Last Modified: 01-14-2008 08:18 CST
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Summary: MixMonitor doesn't work right with SIP and FLASH on
FXS channels
Description:
I want to record *every* call in my Asterisk box, so I use the MixMonitor()
application like this is my extensions.conf:
exten => _0X.,1,Answer()
exten =>
_0X.,n,MixMonitor(${CALLERID(num)}-${STRFTIME(${EPOCH},America/Sao_Paulo,%Y-%m-%d-%H-%M-%S)}-${EXTEN}.wav)
exten => _0X.,n,Dial(IAX2/pabx-canall/${EXTEN},60,tT)
exten => _2XX,1,Answer() exten =>
_2XX,n,MixMonitor(${CALLERID(num)}-${STRFTIME(${EPOCH},America/Sao_Paulo,%Y-%m-%d-%H-%M-%S)}-${EXTEN}.wav)
exten => _2XX,n,Dial(SIP/${EXTEN},60,tT)
The scenario is as following:
1) 201 asks operator for an external call, hangs up. The audio file is
stored correctly. From the CLI:
[Jan 8 16:20:19] -- Executing [200 at default:1] Answer("SIP/201-081d8740",
"") in new stack
[Jan 8 16:20:19] -- Executing [200 at default:2]
MixMonitor("SIP/201-081d8740", "201-2008-01-08-16-20-19-200.wav") in new
stack
[Jan 8 16:20:19] -- Executing [200 at default:3] Dial("SIP/201-081d8740",
"SIP/200|60|tT") in new stack
[Jan 8 16:20:19] == Begin MixMonitor Recording SIP/201-081d8740
[Jan 8 16:20:19] -- Called 200
[Jan 8 16:20:19] -- SIP/200-081fac90 is ringing
[Jan 8 16:20:23] -- SIP/200-081fac90 answered SIP/201-081d8740
[Jan 8 16:20:27] == Spawn extension (default, 200, 3) exited non-zero on
'SIP/201-081d8740'
[Jan 8 16:20:27] == End MixMonitor Recording SIP/201-081d8740
2) 200 dials to the PSTN. So far so good.
[Jan 8 16:20:35] -- Executing [021047020 at default:1]
Answer("SIP/200-081d8740", "") in new stack
[Jan 8 16:20:35] -- Executing [021047020 at default:2]
MixMonitor("SIP/200-081d8740", "200-2008-01-08-16-20-35-021047020.wav") in
new stack
[Jan 8 16:20:35] -- Executing [021047020 at default:3]
Dial("SIP/200-081d8740", "IAX2/pabx-canall/021047020|60|tT") in new stack
[Jan 8 16:20:35] == Begin MixMonitor Recording SIP/200-081d8740
[Jan 8 16:20:35] -- Called pabx-canall/021047020
[Jan 8 16:20:35] -- Call accepted by 200.248.136.140 (format alaw)
[Jan 8 16:20:35] -- Format for call is alaw [Jan 8 16:20:35] --
IAX2/pabx-canall-16384 answered SIP/200-081d8740
3) Extension 200 is a Polycom SoundPoint 301 IP phone. It presses the
Transfer button, putting 021047020 in hold and dialing to 201 who answers
the call:
[Jan 8 16:20:45] -- Started music on hold, class 'default', on
IAX2/pabx-canall-16384
[Jan 8 16:20:51] -- Executing [201 at default:1] Answer("SIP/200-081fac90",
"") in new stack
[Jan 8 16:20:51] -- Executing [201 at default:2]
MixMonitor("SIP/200-081fac90", "200-2008-01-08-16-20-51-201.wav") in new
stack
[Jan 8 16:20:51] -- Executing [201 at default:3] Dial("SIP/200-081fac90",
"SIP/201|60|tT") in new stack
[Jan 8 16:20:51] -- Called 201
[Jan 8 16:20:51] == Begin MixMonitor Recording SIP/200-081fac90
[Jan 8 16:20:51] -- SIP/201-081edf80 is ringing
[Jan 8 16:20:54] -- SIP/201-081edf80 answered SIP/200-081fac90
4) The operator says "here's your call" to 201 and presses Transfer on the
phone once more. The call is transferred correctly, but:
[Jan 8 16:20:57] -- Stopped music on hold on IAX2/pabx-canall-16384
[Jan 8 16:20:57] == Spawn extension (default, 021047020, 3) exited
non-zero on 'SIP/200-081d8740'
[Jan 8 16:20:57] == End MixMonitor Recording SIP/200-081d8740
[Jan 8 16:20:57] == End MixMonitor Recording SIP/200-081fac90
Notice that all the MixMonitor processes stopped!
5) 201 finally hangs up the phone:
[Jan 8 16:21:45] == Spawn extension (default, 201, 3) exited non-zero on
'IAX2/pabx-canall-16384'
[Jan 8 16:21:45] -- Hungup 'IAX2/pabx-canall-16384'
So, all the audio regarding the important part -- the call to the PSTN
itself -- is simply lost.
Although I'm using SIP in this example, the very same thing happens when I
use a TDM2400 with FXS channels and press the FLASH button on the phone to
transfer.
I noticed that if I use Asterisk's built-in transfer features (atxfer,
blindxfer) everything works fine.
Any ideas on workarounds will be welcome as well (IE mapping the FLASH on
the analog phone to the sequence configured on features.conf).
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file - 01-14-08 08:18
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It actually is doing the logical thing... let me explain.
In your instance when you start MixMonitor it is on the calling channel,
always a channel created by SIP/200 in your above scenario. This channel
ultimately disappears though once the transfer completes. You actually want
MixMonitor to start on the called channel instead. The one thing that comes
to mind is using chan_local in between with /n so you can start MixMonitor
on a channel that shouldn't go away.
Issue History
Date Modified Username Field Change
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01-14-08 08:18 file Note Added: 0076884
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