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
>>>>>
>>>>>
>>>>> ------------------------------------------------------------------------
>>>>>
>>>>>
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>>>>>
>>>>>
>>>>>
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>>      
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