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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