[Asterisk-Users] Weird Can-Reinvite problem

[Brett N]

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 <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