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Summary:ASTERISK-22123: Mixmonitor does not create the file and call is muted
Reporter:Roberto (tel.medola)Labels:
Date Opened:2013-07-19 09:45:31Date Closed:2013-07-19 11:32:47
Priority:MajorRegression?
Status:Closed/CompleteComponents:Applications/app_mixmonitor
Versions:1.8.23.0 Frequency of
Occurrence		Constant
Related
Issues:
Environment:Linux asterisk02 2.6.32-358.14.1.el6.x86_64 #1 SMP Tue Jul 16 23:51:20 UTC 2013 x86_64 x86_64 x86_64 GNU/Linux Attachments:( 0) debug
( 1) debug.0
( 2) full
( 3) full
( 4) full.0
( 5) messages
( 6) messages
( 7) messages.0
Description:When I call with MixMontitor activated, the audio is not heard on the target and not the file is created in /var/spool/asterisk/monitor

exten => 8035,1,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
same => n, MixMonitor(${UNIQUEID}.wav,amb)
same => n,Dial(SIP/IP_ADDRESS/8035)
same => n,Hangup
Comments:By: Roberto (tel.medola) 2013-07-19 10:03:56.432-0500

How apply the patch? I can not do...

https://issues.asterisk.org/jira/browse/ASTERISK-19426?focusedCommentId=189761&page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel#comment-189761
By: Roberto (tel.medola) 2013-07-19 10:19:54.612-0500

I apply the patch:

wget -q -O - https://issues.asterisk.org/jira/secure/attachment/43471/32-bit-time-overflow-2012-04-27.diff | patch -p0

./configure
make
make install

the problem persists.

One question: My Centos is 6.4 64bits

By: Matt Jordan (mjordan) 2013-07-19 11:32:40.334-0500

# Your log files don't contain debug information, nor sufficient information to diagnose any problems here. Please read the instructions on the Asterisk wiki for reporting problems.
# The patch you linked to has been in Asterisk for quite some time, and should already be in Asterisk 1.8. Even so, the problem addressed in that patch only affected 32-bit systems.


All of that aside, there's some obvious WARNING messages in your log file that something is amiss:

{noformat}
[Jul 19 11:43:03] WARNING[2324] chan_sip.c: Got Siren7 offer at 24000 bps, but only 32000 bps supported; ignoring.
[Jul 19 11:43:03] VERBOSE[2452] pbx.c:     -- Executing [8035@callcenter:1] Goto("SIP/9007-00000002", "padrao,8035,1") in new stack
[Jul 19 11:43:03] VERBOSE[2452] pbx.c:     -- Goto (padrao,8035,1)
[Jul 19 11:43:03] VERBOSE[2452] pbx_realtime.c:     -- Executing [8035@padrao:1] Set("SIP/9007-00000002", "AUDIOHOOK_INHERIT(MixMonitor)=yes")
[Jul 19 11:43:03] VERBOSE[2452] pbx_realtime.c:     -- Executing [8035@padrao:2] MixMonitor("SIP/9007-00000002", "1374244983.2.wav,amb")
[Jul 19 11:43:03] VERBOSE[2453] app_mixmonitor.c:   == Begin MixMonitor Recording SIP/9007-00000002
[Jul 19 11:43:03] VERBOSE[2452] pbx_realtime.c:     -- Executing [8035@padrao:3] Dial("SIP/9007-00000002", "SIP/172.18.100.20/8035")
[Jul 19 11:43:03] VERBOSE[2452] netsock2.c:   == Using SIP RTP CoS mark 5
[Jul 19 11:43:03] VERBOSE[2452] app_dial.c:     -- Called SIP/172.18.100.20/8035
[Jul 19 11:43:03] VERBOSE[2452] app_dial.c:     -- SIP/172.18.100.20-00000003 is ringing
[Jul 19 11:43:04] VERBOSE[2452] app_dial.c:     -- SIP/172.18.100.20-00000003 answered SIP/9007-00000002
[Jul 19 11:43:08] VERBOSE[2452] cdr_pgsql.c:        > [INSERT INTO cdr ("clid","src","dst","dcontext","channel","dstchannel","lastapp","lastdata","duration","billsec","disposition","amaflags","uniqueid","useragent","answer","end","calldate") VALUES ('"Ramal 9007" <9007>','9007','8035','padrao','SIP/9007-00000002','SIP/172.18.100.20-00000003','Dial','SIP/172.18.100.20/8035',5,4,'ANSWERED',3,'1374244983.2','','2013-07-19 14:43:04','2013-07-19 14:43:08','2013-07-19 14:43:03')]
[Jul 19 11:43:08] VERBOSE[2452] pbx.c:   == Spawn extension (padrao, 8035, 3) exited non-zero on 'SIP/9007-00000002'
[Jul 19 11:43:08] VERBOSE[2453] app_mixmonitor.c:   == End MixMonitor Recording SIP/9007-00000002
{noformat}

In particular, the WARNING message regarding Siren7 indicates that something is not completely supported in the audio codecs being negotiated. You probably don't have audio - which is why you have no call recording and why the call is "muted".

As it is, this sounds like a configuration issue. As such, I'm closing out this issue as "Not a Bug". If you need assistance with your configuration, please contact the asterisk-users mailing list or the #asterisk IRC channel.
By: Roberto (tel.medola) 2013-07-19 11:55:51.345-0500

news logs
By: Roberto (tel.medola) 2013-07-19 11:55:56.694-0500

My problem is not the this WARNING( regardin Siren 7). My problem is when i call with MixMonitor on, the call is muted and the file is not create.

See new logs(with debug information) please.
By: Roberto (tel.medola) 2013-07-19 12:17:22.967-0500

Hi,
analyzing the new log, i found the "bug".

My iptables was blocking the new port created by MixMonitor.

Sorry for issue wrong.

Thanks for the help.