On Mon, Mar 3, 2008 at 12:40 AM, NOC Ph <[EMAIL PROTECTED]> wrote: > > [NOCPH] I have to open the SIP port and web. Another question, the SIP port > is 5060 UDP, how about the conference? Does it use the same port also?
That's a good start, but you'll also need to open the RTP ports as well - these usually fall in the 10k-20k udp range. 5060/udp is used for call signalling only, the actual voice data can use a variety of ports, depending on how you're set up. You can specify what RTP ports you want to use in your rtp.conf. -erik _______________________________________________ -- 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
