Search jitter in sample sip.conf. Everything is well documented there.
Regards, Qasim On Tue, Apr 23, 2013 at 3:03 PM, Muhammad Yousuf <muyous...@gmail.com>wrote: > I am using asterisk as SIP/GSM gateway. I have 2 gsm cards installed in > server. I am having some issue in audio quality. I want to enable jitter > buffer on asterisk but don't know, how to do. Any one can help me. > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
