Yep, using SIP for users, IAX for trunks. Can't seem to figure out how to help out the RTP streams though. Once in a while, calls seem clear, but most of the time they're choppy as anything...
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Friday, July 28, 2006 2:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk VOIP / Mikrotik On Jul 28, 2006, at 10:55 AM, Curt Shaffer wrote: > And, someone correct me if I am wrong here, you want to make sure RTP > is getting quality as well. SIP is setting up, tearing down, and a few > other things but RTP is where the conversation is taking place. Yes, if he is using SIP. He didn't mention that. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
