search for: transmit_sil

Hello, I've got a problem at the moment, that setting "transmit_silence = yes" seems to have no effect on Asterisk 1.8-Certified. Although it's enabled and "core show settings" confirms, that it is really enabled, there are no RTP packets sent by Asterisk when waiting for DMTF input or when "Wait()" is called. Also, there seems to be a...

hello, since while recoding asterisk is not sending an audio stream, the remote party times-out rtp and is hanging up on us. is it possible to send a blank audio stream while recording app is running? thanks, jrun

Hi, We are using MixMonitor to record the call. When the call is bridged, the latency is significant. We tried to increase the internet speed and the server RAM and processor speed and still we are having that issue. We use VoiceTrading and Gafachi's Termination minutes to make calls. As we are in US and VoiceTrading in Europe, somebody suggested to move the termination minute provider

Hello, I have my provider dropping the calls after 41 seconds of not receiving any RTP from my asterisk. Obviously there is no RTP back when the caller is leaving a message in the voicemail. Is it possible to have asterisk generate some RTP packet back? Leandro -------------- next part -------------- An HTML attachment was scrubbed... URL: