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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