[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
Tue Jan 15 14:02:36 CST 2008
The following issue has been RESOLVED.
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http://bugs.digium.com/view.php?id=11764
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Reported By: viniciusfontes
Assigned To: file
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Project: Asterisk
Issue ID: 11764
Category: Applications/app_mixmonitor
Reproducibility: always
Severity: minor
Priority: normal
Status: resolved
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:
Resolution: won't fix
Fixed in Version:
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Date Submitted: 01-14-2008 04:20 CST
Last Modified: 01-15-2008 14:02 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 - 01-15-08 14:02
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Closed as a suitable way to do it has been described.
Issue History
Date Modified Username Field Change
======================================================================
01-15-08 14:02 file Status feedback => resolved
01-15-08 14:02 file Resolution open => won't fix
01-15-08 14:02 file Assigned To => file
01-15-08 14:02 file Note Added: 0076994
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