On Mon, Nov 16, 2009 at 9:40 AM, Bharath B. Reddy Bynagari
<bynag...@mavensphere.com> wrote:
> We are using MixMonitor to record the call. When the call is bridged, the
> latency is significant.
> $ConversationFile =
> $ConversationPath."conv_"."$CallQID-$ConversationID.wav";
>
> $self->agi->answer();
>
> $self->agi->exec("MixMonitor", "$ConversationFile|ba");You're obviously using SIP. I don't like to admit it, but I've seen this problem before. Please try modifying the voice-activity-detection sections of your SIP settings and see if this fixes the problems. My hunch, which is not proven, is that when SIP silence detection thinks it should stop transmitting packets, the recording module thinks it shouldn't record the lack of voice transmission, and then the timing in the recording gets farther and farther from the truth the longer the call goes on. in asterisk.conf transmit_silence = yes transmit_silence_during_record = yes in dsp.conf silencethreshold=1000 in codecs.conf vad => false pp_vad => false _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
