Am new to asterisk, so please bear with me if my question sounds naive. I have an SIP extension on asterisk that plays a music file when dialed to (Playback).
The problem is that, callers who fail to transmit RTP packet to asterisk would not hear any audio (users without mic). Asterisk wont sent RTP packets to caller unless it receives RTP packets from the caller. But in my case (Playback), the user does not need to send RTP packets to asterisk. Is there a workaround to this problem?
I did some basic research on this and I found that using internal timing can help solve the issue. I am using Asterisk 1.8.7 and the only timing module I have is res_timing_pthread.so. I tried setting internal_timing=yes for asterisk, but still no difference. How can I load other modules like res_timing_dahdi.so and res_timing_timerfd.so to asterisk.
Any help would be greatly appreciated. Thanks