FOP2 Community Forum

[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:'1adbcf9ddf504386@10.1.1.2">'1adbcf9ddf504386@10.1.1.2</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:06988c2608e88f53051a08874bd2b6d4@127.0.1.1:5060">06988c2608e88f53051a08874bd2b6d4@127.0.1.1:5060</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:419821e607861e33261730cd55ae0937@192.168.254.10:5060">419821e607861e33261730cd55ae0937@192.168.254.10:5060</a><!-- e -->
[Oct 17 12:39:14] DEBUG[2166] chan_sip.c: Stopping retransmission on <!-- e --><a href="mailto:'419821e607861e33261730cd55ae0937@192.168.254.10:5060">'419821e607861e33261730cd55ae0937@192.168.254.10:5060</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:79d5ff2d74285e762c3795b204264298@192.168.254.10:5060">79d5ff2d74285e762c3795b204264298@192.168.254.10:5060</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:1adbcf9ddf504386@10.1.1.2">1adbcf9ddf504386@10.1.1.2</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:'79d5ff2d74285e762c3795b204264298@192.168.254.10:5060">'79d5ff2d74285e762c3795b204264298@192.168.254.10:5060</a><!-- e -->' of Request 103: Match Found
[Oct 17 12:39:21] DEBUG[2166] rtp_engine.c: Destroyed RTP instance '0x8e623b8'

how do I see the log of asterisk?

had already done that but it didn't work ..

but something?

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?