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
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Jim Freeze
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