Hey Jim, No I haven't. What does ICMP redirect have to do with this? Have you had this problem? Did this fix it for you? -Brett
On Tue, June 6, 2006 1:18 pm, Jim Freeze wrote: > Have you tried turning off icmp redirect on your router? > > > On 6/6/06, Brett N <[EMAIL PROTECTED]> wrote: >> >> Hi All, >> I'm having a really weird can reinvite issue. I've been banging my head >> around on this for days now.. >> >> >> I have two asterisk servers. One at 172.20.0.11 One at 172.20.2.5 >> >> 172.20.0.11 is a hosted box and serves multiple offices >> 172.20.2.5 is a box on site at a customer's office. >> >> >> A phone at 172.20.128.10 makes a call using server 172.20.0.11 to a >> phone >> at 172.20.2.80 via server 172.20.2.5: >> >> Phone A-->asterisk A----->SER----->asterisk B--->PhoneB >> >> All devices all have ip connectivity (No Firewalls! No Natting) to each >> other. so phone a can ping phone b and server b, etc, etc, etc.. >> >> >> Can reinvite is enabled on both the ser connection (on both sides) and >> for >> both phones.. >> >> Making a call from phone A to phone B works great.. Except you can hear >> a >> pop when the reinvite happens. After the call is connected Phone B can >> transfer the phone just fine.. However if phone A (the originator) tries >> to transfer FIRST (either to the pstn via SER or to another local >> extension on asterisk A) the call will have 0 way audio. If the call is >> transfered back, there will be one way audio. >> >> It seems this is Always how it is, over and over.. The Originator Cannot >> transfer the call first. I THINK if the destination transfers first, >> THEN >> the originator can transfer.. >> >> I've checked netmasks, ips, gateways, etc, etc.. The SDP on the >> reinvites >> looks ok.. >> >> No Nat, no funny business here.. just IP routing.. >> >> Any ideas? >> -Brett _______________________________________________ --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
