I can test anything for about the next 10 min. I have a personal 
interest in it too...
I'm tentatively planning an upgrade from 4.0.4 to 4.2.1 this weekend.
I"ll go through with it if I know it doesn't at least make things worse.

On 7/16/2010 5:17 PM, Tony Graziano wrote:
> I can test with 4.2.1 and sipxbridge tomorrow.
> ============================
> Tony Graziano, Manager
> Telephone: 434.984.8430
> Fax: 434.984.8431
>
> Email: [email protected]
>
> LAN/Telephony/Security and Control Systems Helpdesk:
> Telephone: 434.984.8426
> Fax: 434.984.8427
>
> Helpdesk Contract Customers:
> http://www.myitdepartment.net/gethelp/
>
> ----- Original Message -----
> From: [email protected]
> <[email protected]>
> To: [email protected]<[email protected]>
> Sent: Fri Jul 16 18:15:32 2010
> Subject: Re: [sipx-users] Help with Patton gateway
>
> I just tested. I think I'm seeing the same thing.
> On 4.2.1, if I do attended transfer to the auto attendant, the call
> tranfers, but appears to stay on hold on the transferring phone.
> On 4.0.4, it appears to be even more broken. The call stays on hold and
> it doesn't actually transfer the call either.
>
> I"m using Sipxbridge and polycoms with 3.1.3
>
> On 7/16/2010 5:06 PM, Josh Patten wrote:
>    
>> Can I get other people on this list to test this scenario as well? It'll
>> only take a couple of minutes. I know all of you have at least two
>> phones on your desk :-P
>>
>> Josh Patten
>> Assistant Network Administrator
>> Brazos County IT Dept.
>> (979) 361-4676
>>
>>
>> On 7/16/2010 4:55 PM, Tony Graziano wrote:
>>
>>      
>>> I hope its fixable with a config change and not a deep down inside issue.
>>> ============================
>>> Tony Graziano, Manager
>>> Telephone: 434.984.8430
>>> Fax: 434.984.8431
>>>
>>> Email: [email protected]
>>>
>>> LAN/Telephony/Security and Control Systems Helpdesk:
>>> Telephone: 434.984.8426
>>> Fax: 434.984.8427
>>>
>>> Helpdesk Contract Customers:
>>> http://www.myitdepartment.net/gethelp/
>>>
>>> ----- Original Message -----
>>> From: [email protected]
>>> <[email protected]>
>>> To: [email protected]<[email protected]>
>>> Sent: Fri Jul 16 17:49:46 2010
>>> Subject: Re: [sipx-users] Help with Patton gateway
>>>
>>> I'm only talking about attended transfers to FreeSWITCH media services
>>> (AKA conference, voicemail, and auto attendant) not phone-to-phone
>>> transfers. Phone-to-phone transfers are working fine.
>>>
>>> Josh Patten
>>> Assistant Network Administrator
>>> Brazos County IT Dept.
>>> (979) 361-4676
>>>
>>>
>>> On 7/16/2010 4:48 PM, Michael Scheidell wrote:
>>>
>>>
>>>        
>>>> On 7/16/10 5:45 PM, Josh Patten wrote:
>>>>
>>>>
>>>>          
>>>>> I'm running 4.2.1
>>>>>
>>>>> I have just confirmed this has issues with Aastra phones as well.
>>>>>
>>>>> I've been saying for a while that FreeSWITCH has issues with the way
>>>>> attended transfers are handled.
>>>>> http://wiki.sipfoundry.org/display/xecsuserV4r2/Custom+FreeSWITCH+programming
>>>>> is a prime example.
>>>>>
>>>>> Fix the FreeSWITCH SIP stack and these issues will probably go away.
>>>>>
>>>>>
>>>>>
>>>>>            
>>>> attended transfers:
>>>> cisco to cisco is ok (sipx 4.2.0) (but funky.. you see the call drop,
>>>> the screen go blank, and then the calls start to go again)
>>>> cisco to polycom, doesn't work at all.
>>>>
>>>> (* just do you don't complain about me using cisco's.. looks like its
>>>> the FreeSWITCH SIP stack)
>>>>
>>>> --
>>>> Michael Scheidell, CTO
>>>> Phone: 561-999-5000, x 1259
>>>>
>>>>
>>>>          
>>>>> *| *SECNAP Network Security Corporation
>>>>>
>>>>>
>>>>>            
>>>>        * Certified SNORT Integrator
>>>>        * 2008-9 Hot Company Award Winner, World Executive Alliance
>>>>        * Five-Star Partner Program 2009, VARBusiness
>>>>        * Best in Email Security,2010: Network Products Guide
>>>>        * King of Spam Filters, SC Magazine 2008
>>>>
>>>>
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>>>>
>>>>          
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>>      
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