Johansson Olle E wrote: > In my series of articles about Asterisk 1.4, I've now arrived to the > new jitter buffer > that enhances voice quality for those of you using Asterisk as a PSTN > gateway. > > Please read > http://www.voip-forum.com/category/asterisk/asterisk14/
I wrote a patch that lets you use the jitterbuffer in Asterisk 1.4 for more than just PSTN gateway functionality. Originally, there was no way to use it when connecting to Asterisk applications that did not create outbound channels and bridge calls (basically only Dial and Queue). This is actually still the case, but what I did was add support for using the jitterbuffer when you are bridged to a Local channel. That way, you can use it when connected to Voicemail, Meetme, or whatever else you want. See this post for more information: http://www.russellbryant.net/blog/index.php/2007/10/09/asterisk-jitterbuffer-support-for-applications/ -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. _______________________________________________ -- 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
