Use debian squeeze with Asterisk 1.8, I put intalei fop2.26 record conversations he writes in the /var/spool /asterisk/monitor but nothing appears in the interface, the rest is working just fine ..
Anyone know what might be missing?
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12 years ago
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If you want them to appear on fop2 recordings interface, you will have to edit fop2.cfg and set monitor_exec to run recordings_fop2.pl (it is not enabled by default).
Best regards,
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had already done that but it didn't work ..
but something?
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inspect the asterisk full log when enabling debug in the asterisk cli, and see if the script is being executed when a recording is finished. FOP2 does not record the call and it does not launch the script either, it just signals asterisk to do it. So, you have to check asterisk logs to see why it does not work as you expect to work.
Best regards,
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how do I see the log of asterisk?
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asterisk -rx "core set debug 1" asterisk -rx "core set verbose 10" tail -f /var/log/asterisk/full
(The above asumes you have the asterisk full log enabled in /etc/asterisk/logger.conf)
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[Oct 17 12:38:59] VERBOSE[2202] app_dial.c: -- SIP/403-00000003 answered SIP/400-00000002
[Oct 17 12:38:59] DEBUG[2202] chan_sip.c: SIP answering channel: SIP/400-00000002
[Oct 17 12:38:59] DEBUG[2202] chan_sip.c: Setting framing from config on incoming call
[Oct 17 12:38:59] DEBUG[2202] chan_sip.c: ** Our capability: 0x8 (alaw) Video flag: True Text flag: True
[Oct 17 12:38:59] DEBUG[2202] chan_sip.c: ** Our prefcodec: 0x0 (nothing)
[Oct 17 12:38:59] DEBUG[2202] features.c: Removing dialed interfaces datastore on SIP/403-00000003 since we're bridging
[Oct 17 12:38:59] DEBUG[2166] chan_sip.c: Stopping retransmission on <!-- e --><a href="mailto:'[email protected]">'[email protected]</a><!-- e -->' of Response 19637: Match Found
[Oct 17 12:38:59] DEBUG[2202] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw
[Oct 17 12:38:59] DEBUG[2202] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160
[Oct 17 12:38:59] DEBUG[2202] res_rtp_asterisk.c: Starting RTCP transmission on RTP instance '0x8e623b8'
[Oct 17 12:38:59] DEBUG[2202] res_rtp_asterisk.c: Got RTCP report of 84 bytes
[Oct 17 12:38:59] DEBUG[2202] res_rtp_asterisk.c: Ooh, format changed from unknown to alaw
[Oct 17 12:38:59] DEBUG[2202] res_rtp_asterisk.c: Created smoother: format: alaw ms: 20 len: 160
[Oct 17 12:38:59] DEBUG[2202] res_rtp_asterisk.c: Got RTCP report of 118 bytes
[Oct 17 12:39:04] DEBUG[2202] res_rtp_asterisk.c: Got RTCP report of 84 bytes
[Oct 17 12:39:07] DEBUG[2194] manager.c: Running action 'Setvar'
[Oct 17 12:39:07] DEBUG[2194] manager.c: Running action 'Monitor'
[Oct 17 12:39:09] DEBUG[2202] res_rtp_asterisk.c: Got RTCP report of 84 bytes
[Oct 17 12:39:13] DEBUG[2202] res_rtp_asterisk.c: Got RTCP report of 118 bytes
[Oct 17 12:39:13] DEBUG[2166] chan_sip.c: Allocating new SIP dialog for <!-- e --><a href="mailto:[email protected]:5060">[email protected]:5060</a><!-- e --> - OPTIONS (No RTP)
[Oct 17 12:39:13] DEBUG[2166] chan_sip.c: Initializing initreq for method OPTIONS - callid <!-- e --><a href="mailto:[email protected]:5060">[email protected]:5060</a><!-- e -->
[Oct 17 12:39:14] DEBUG[2166] chan_sip.c: Stopping retransmission on <!-- e --><a href="mailto:'[email protected]:5060">'[email protected]:5060</a><!-- e -->' of Request 102: Match Found
[Oct 17 12:39:14] DEBUG[2202] res_rtp_asterisk.c: Got RTCP report of 84 bytes
[Oct 17 12:39:19] DEBUG[2202] res_rtp_asterisk.c: Got RTCP report of 84 bytes
[Oct 17 12:39:21] DEBUG[2166] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8e5a8b8'
[Oct 17 12:39:21] DEBUG[2202] channel.c: Didn't get a frame from channel: SIP/400-00000002
[Oct 17 12:39:21] DEBUG[2202] channel.c: Bridge stops bridging channels SIP/400-00000002 and SIP/403-00000003
[Oct 17 12:39:21] DEBUG[2202] channel.c: Hanging up channel 'SIP/403-00000003'
[Oct 17 12:39:21] DEBUG[2202] chan_sip.c: Hangup call SIP/403-00000003, SIP callid <!-- e --><a href="mailto:[email protected]:5060">[email protected]:5060</a><!-- e -->
[Oct 17 12:39:21] DEBUG[2202] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8e623b8'
[Oct 17 12:39:21] DEBUG[2202] app_dial.c: Exiting with DIALSTATUS=ANSWER.
[Oct 17 12:39:21] VERBOSE[2202] app_macro.c: == Spawn extension (macro-disca, s, 300) exited non-zero on 'SIP/400-00000002' in macro 'disca'
[Oct 17 12:39:21] DEBUG[2202] pbx.c: Spawn extension (telefonista,403,2) exited non-zero on 'SIP/400-00000002'
[Oct 17 12:39:21] VERBOSE[2202] pbx.c: == Spawn extension (telefonista, 403, 2) exited non-zero on 'SIP/400-00000002'
[Oct 17 12:39:21] DEBUG[2202] channel.c: Soft-Hanging up channel 'SIP/400-00000002'
[Oct 17 12:39:21] DEBUG[2202] channel.c: Hanging up channel 'SIP/400-00000002'
[Oct 17 12:39:21] DEBUG[2202] chan_sip.c: Hangup call SIP/400-00000002, SIP callid <!-- e --><a href="mailto:[email protected]">[email protected]</a><!-- e -->
[Oct 17 12:39:21] DEBUG[2202] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x8e5a8b8'
[Oct 17 12:39:21] DEBUG[2202] res_monitor.c: monitor executing /usr/local/fop2/recording_fop2.pl "/var/spool/asterisk/monitor/400_400_123907_1350488335.2-in.wav" "/var/spool/asterisk/monitor/400_400_123907_1350488335.2-out.wav" "/var/spool/asterisk/monitor/400_400_123907_1350488335.2.wav" &
[Oct 17 12:39:21] DEBUG[2166] chan_sip.c: Stopping retransmission on <!-- e --><a href="mailto:'[email protected]:5060">'[email protected]:5060</a><!-- e -->' of Request 103: Match Found
[Oct 17 12:39:21] DEBUG[2166] rtp_engine.c: Destroyed RTP instance '0x8e623b8' -
The script is being execute correctly:
[Oct 17 12:39:21] DEBUG[2202] res_monitor.c: monitor executing /usr/local/fop2/recording_fop2.pl "/var/spool/asterisk/monitor/400_400_123907_1350488335.2-in.wav" "/var/spool/asterisk/monitor/400_400_123907_1350488335.2-out.wav" "/var/spool/asterisk/monitor/400_400_123907_1350488335.2.wav" &
If you have the db and tables and permissions populated correctly, you not only will see the final /var/spool/asterisk/monitor/400_400_123907_1350488335.2.wav file but also the proper entries in the database. In FreePBX you do not have to do anything, but if you do not use FreePBX, please read the script as you will have to create databases or databases, set credentials, etc.. Open the file and read it, as it explains everything:
/usr/local/fop2/recording_fop2.pl
