http://track.sipfoundry.org/browse/XX-8652 Once again Dale solves the 
mystery. Should this be reported to FreeSWITCH? From what I've seen they 
have a pretty good track record for fixing bugs.

On 07/16/2010 07:49 PM, Josh Patten wrote:
> The reason why this is an issue is because for end users migrating from
> another system, as in my case, They are used to the "Transfer->dial
> number->Transfer" method of transferring calls. The
> "Transfer->Blind->Dial Number" concept is foreign to them so out of
> force of habit they use the former method for transferring all calls.
>
> On 07/16/2010 06:53 PM, Paul Herron wrote:
>    
>> I concur with Matt's last two posts.  I just tested attended transfer as
>> described and had the same results -- fine to VM, failed to AA.  Blind
>> transfers are fine. Running 4.0.4 w/Polycom 650 3.1.3c split, BootROM
>> 4.2.2
>>
>> Of course, one might wonder why we need an attended transfer to AA
>> (other than the user screwing-up and pressing the wrong button -- o.k.,
>> I guess that's a pretty good reason).
>>
>> I also just tested attended transfer of an external call (via SipBridge)
>> and had the same results -- fine to VM, failed to AA.  Blind is fine.
>>
>> -----Original Message-----
>> From: Matthew Kitchin (public/usenet) [mailto:[email protected]]
>>
>> Sent: Friday, July 16, 2010 6:44 PM
>> To: [email protected]
>> Subject: Re: [sipx-users] Help with Patton gateway
>>
>> Attended for everything in what I described below.
>> Blind is fine.
>>
>> On 7/16/2010 5:41 PM, Josh Patten wrote:
>>
>>      
>>> Attended transfer to VM or blind?
>>>
>>> I haven't seen any issues with blind transfer.
>>>
>>> On 07/16/2010 05:32 PM, Matthew Kitchin (public/usenet) wrote:
>>>
>>>
>>>        
>>>> A little more info, 4.0.4 seems fine when transferring to someone's
>>>>
>>>>          
>> VM,
>>
>>      
>>>> but fails when transferring to Auto attendant. IT totally fails going
>>>>
>>>>          
>> to
>>
>>      
>>>> the AA.
>>>> 4.2.1 seems to exhibit the same behavior (described below) when
>>>> transferring to either VM or AA.
>>>>
>>>> On 7/16/2010 5:15 PM, Matthew Kitchin (public/usenet) wrote:
>>>>
>>>>
>>>>
>>>>          
>>>>> 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
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>                  
>> ------------------------------------------------------------------------
>>
>>      
>>>>>>>> This email has been scanned and certified safe by SpammerTrap.
>>>>>>>> For Information please see
>>>>>>>>
>>>>>>>>                  
>> http://www.secnap.com/products/spammertrap/
>>
>>      
>>>>>>>>
>>>>>>>>                  
>> ------------------------------------------------------------------------
>>
>>      
>>>>>>>>
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>>
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>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
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>>>>>
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