Hi, I have started using the call recording facilities in Asterisk 1.2 recently, and having worked out some of the foibles regarding call forwarding etc etc, I think I have a mostly working system.
I do still seem to have a problem with recording volume though. It seems that all SIP call legs are recorded at "normal" volume, but all my Zap (ISDN) and IAX (via Provider -> ISDN) calls are recorded at a massively reduced volume. -> It does not matter whether the call originates inside or outside the box -> It does not matter which channel is Monitored (Zap, IAX or SIP) -> The caller/callee can hear each other fine regardless of the call source and destination. -> I also tried both Monitor and MixMonitor with the same results. -> The recording of the ISDN or IAX leg is so quiet that it is often impossible to hear. -> SIP to SIP records 100% okay -> Recording using different codecs makes no difference -> Voicemail recording volume is fine, regardless of call source. I considered using MixMonitor's volume settings, but cannot always identify which channel needs a volume boost (Local channels can obscure the call source or destination) I can use 'sox' to modify the levels to a usable point, but this amplifies background noise to a ridiculous degree so is not particularly satisfactory. Given that the call proceeds "normally" where is all of the volume being lost? We generally use aLaw end-to-end (which is the codec used on UK ISDN lines) so there should be almost no modification of the voice packets required at-all. Why does the recording differ from the audio being heard? I looked at the source and could see no obvious reason! Thanks for any pointers. I am happy to try experiments on our development system if it helps... Regards, Steve _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
