Ops, guess I should have read that a little closer. Not enough coffee yet. :(
Rich ------------------------ > I think what James is referring to is the delay once the call already been dialed. > It's not specific to Ciscos, as I'm experiencing the same problem on my polycom phones. Must be SIP related. > > The problem is that once a call is dialed, when the remote party picks up the phone, > the first half second is cutoff. The remote party won't hear the first half second of the call. I had this happend several times in the last few days. I've also had a few complaints from users recently. Here's what it looks like. > > SIP phone dials 555-1234 (outside line via PRI) > 555-1234 rings > 555-1234 answers and says "Hello" > SIP phone hears "o" or nothing at all. > > If 555-1234 is slow to say something, then everything is heard fine. > > Caveats. echotraining and echocancel are enabled on the PRI, however, similiar Zap > calls are not affected. > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
