Steve Totaro wrote: > Just for sh1t$ and giggles, try sip to sip and drop the IAX piece. > IAX2 is not all it is cracked up to be. > > Also, do a ping to see latency, 200ms is pretty much my standard. >
Coming from outside the network, setting up for a couple rounds of NATting isn't going to work well. They are not seeing it between phones. Others, using the polycom phones have reported echo between two SIP on a 4ms ping trip. _______________________________________________ -- 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
