Hi List, I have a question regarding jitterbuffer in Asterisk 1.4.24. I see that jitterbuffer is only effective on the receiving channels. My asterisk has only SIP accounts + 2 SIP trunk accounts to our branch office. Questions: 1. To enable jitter buffer on SIP channels it seems I have to enable and force it, right? 2. If I enable and force jitter buffer, Asterisk would always have to stay in media path to make it function, right? If I am right, this effectively disables native RTP bridging. 3. Is it possible to only enable jitter buffer on calls where the SIP trunk is involved? It is no use for me to enable the jitter buffer between SIP phones on the same LAN.
Many thanks for all answers, I have tried hard to google out them, but no success so far. Ondrej _______________________________________________ -- 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
