Something tells me that's not the case. It seems all the major open source VoIP platforms (Asterisk, FreeSWITCH, YaTE, etc.) all have some kind of problem dealing with REFER in some way or another. This is why I had to buy an Audiocodes gateway instead of continuing to use my Adtran because REFER caused problems with it and I couldn't get either Asterisk, FreeSWITCH, or YaTE to compensate for REFER because they too had broken implementations.
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/ >> >> ------------------------------------------------------------------------ >> >> >> _______________________________________________ >> sipx-users mailing list [email protected] >> List Archive: http://list.sipfoundry.org/archive/sipx-users >> Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users >> sipXecs IP PBX -- http://www.sipfoundry.org/ >> _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
