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
>>>>
>>>>
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