Aloysius Thevarajah Lloyd wrote: > Hi, > > I am using pure voip call termination. Not using musiconhold or meetme. > Asterisk 1.2.9.1 <http://1.2.9.1> > > DID (SIP) <-> Asterisk <->Destination
SIP in Asterisk uses a remote timing source (which has problems of it's own but works for the most part), so you won't need any timing sources. -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://e164.org - Because e164.arpa is a tax on VoIP "In the long run the pessimist may be proved right, but the optimist has a better time on the trip."
