[Asterisk-Users] Weird Can-Reinvite problem

[Jim Freeze]

Have you tried turning off icmp redirect on your router?


On 6/6/06, Brett N <brettlist at nemeroff.com> 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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