VoIP monitor Files

version 4.1
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Optimizations 
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- implement storing pcap and graph files to cache/directory and move it to spool directory right after the call ends. the option is -C, --cachedir <dir> (cachedir in voipmonitor.conf). Moving all files are guaranteed to be serialized which solves slow random write I/O on magnetic or other media. Typical cache directory is /dev/shm which is in RAM and grows automatically or /mnt/ssd/voipmonitor which is mounted to SSD disk or some very fast SAS/SATA disk where spool can be network storage or raid5 etc. wav files are not implemented yet

Features
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- implement --norecord-header option: if any of SIP message during the call contains header X-VoipMonitor-norecord call will be not converted to wav and pcap file will be deleted.
- implement --norecord-dtmf option which allows to turn of recording of wav/pcap file once DTMF SIP INFO sequence == "*0" (in other words if you press * and 0 on your phone and you send DTMF via SIP INFO, the call will be not recorded). This option can be given on command line or in voipmonitor.conf.

- Store domain part of uri in From and To header in caller_domain and called_domain which is new cdr columns. This is not configurable and you MUST upgrade database.  ALTER TABLE cdr ADD caller_domain VARCHAR(255) DEFAULT NULL AFTER caller,  ADD called_domain VARCHAR(255) DEFAULT NULL AFTER called, ADD KEY `caller_domain` (`caller_domain`), add KEY `called_domain` (`called_domain`); (upgrade large db can take hours so schedule it off peak)

- add experimental option --rtp-nosig allows analyze of RTP streams without signalization.  Handy if you want to extract RTP without SIP or unsupported H.323 signalization.


Bug fixes
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- fix answer duration for calls where there is 200 OK to another message than the first INVITE - for example PRACK.
- fix annoying noise in recordings which are put on hold and voipmonitor generates silence.
- fix high PDD in case there are more 18X responses during call.
- set progress_time also if RING 18X is not received and the call is connected or aborted so the PDD (Post Dial Delay) is accurate for those cases too 
- If one RTP direction changes SSRC after connect (200 OK) and one of the RTP direction is missing (not captured or whatever reason) cdr was saved bad - web gui showed presence of both directions - one direction for the first SSRC and the second direction secon SSRC so it seemed like there are both directions. Thanks POUILLY GUILLAUME for reporting and providing pcap file.
- Fix out of sync audio for cases where call was put on hold and the SSRC of RTP stream changed.
- Fix some G729 recording issues (fast playback) in rare cases.
- Fix WAV recording for PCMU/PCMA streams with CNG (comfort noise generators - payload 13 and payload 19)
- Fix ./configure on centos