Hi all. I have a (new) customer who is describing symptoms that I've not seen before.
They have 12 Polycom 430's behind a NAT, which is working OK. When phone A is on a call and phone B attempts to transfer another call to phone C, the conversation on phone A is interrupted for 15-20 seconds... The server is hardly loaded, and we have plenty of bandwidth to support our call level. I have these lines in the sip.cfg file: ================================================== nat = yes canreinvite = no ================================================== Has anyone seen these symptoms before? Any clues as to how to fix it? TIA, -- Take care and have fun, Mike Diehl. -- _____________________________________________________________________ -- 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
