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:
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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
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