[asterisk-bugs] [Asterisk 0011764]: MixMonitor doesn't work right with SIP and FLASH on FXS channels

[noreply at bugs.digium.com]

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 08:50 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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---------------------------------------------------------------------- 
 viniciusfontes - 01-14-08 08:50  
---------------------------------------------------------------------- 
OK, but could I continue to use SIP and FLASH transfers using chan_local?
If so, a simple example would be very appreciated. 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
01-14-08 08:50  viniciusfontes Note Added: 0076889                          
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