search for: slienc

Sorry for the fragmented messages from me - one last thing I forgot to ask in my last post. When incoming calls come to us, our PSTN line is picked up almost immediately - and then asterisk will proceed to dial the SIP extensions. During this time the caller hears dead slience - obviously not very good as some would think the line just went dead and hang up. I have toyed with the idea of playing a 'welcome... your call will be answered shortly' etc... message, but can't get it to work how want it. The caller will hear a recorded message, followed by music....

...ief period of silence. * The silence lasts anywhere from 3 to 10 seconds. * It is not due to silence suppression, because the silences generally occur in the middle of sentences. * Silences occur at various times of the day, for calls of various lengths, and at different times during calls. * Sliences have occured for incoming and outgoing calls, and have affected both people on the VoIP side and the PSTN side. * Silences have occured for devices from multiple manufacturers (Currently we are using Polycom IP 300, IP 500, and Grandstream 101). * Affected users are connecting to our asterisk...

I am interested to learn if I need to have ztdummy installed if I do not have any zaptel hardware in my machine? I have found a lot of references with RTP problems which were related to RTP timing (or lack of it). My problem is that voice coming from SIP hardware is OK, but voice going from asterisk to SIP hardware is choppy, full of noise or completely cut-off. Am I going to solve my problem