[asterisk-bugs] [JIRA] (ASTERISK-22123) Mixmonitor does not create the file and call is muted
Matt Jordan (JIRA)
noreply at issues.asterisk.org
Fri Jul 19 11:34:03 CDT 2013
[ https://issues.asterisk.org/jira/browse/ASTERISK-22123?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=208082#comment-208082 ]
Matt Jordan commented on ASTERISK-22123:
----------------------------------------
# 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 at 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 at padrao:1] Set("SIP/9007-00000002", "AUDIOHOOK_INHERIT(MixMonitor)=yes")
[Jul 19 11:43:03] VERBOSE[2452] pbx_realtime.c: -- Executing [8035 at 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 at 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.
> Mixmonitor does not create the file and call is muted
> -----------------------------------------------------
>
> Key: ASTERISK-22123
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-22123
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Applications/app_mixmonitor
> Affects Versions: 1.8.23.0
> 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
> Reporter: Roberto
> Attachments: full, messages
>
>
> 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
--
This message is automatically generated by JIRA.
If you think it was sent incorrectly, please contact your JIRA administrators: https://issues.asterisk.org/jira/secure/ContactAdministrators!default.jspa
For more information on JIRA, see: http://www.atlassian.com/software/jira
More information about the asterisk-bugs
mailing list