Merry Christams (in July). This was a known issue back in August 2008 and was fixed. Something must have changed. I will post this relevant FS thread in the JIRA you opened.
http://jira.freeswitch.org/browse/SFSIP-86 On Fri, Jul 16, 2010 at 8:49 PM, Josh Patten <[email protected]>wrote: > The reason why this is an issue is because for end users migrating from > another system, as in my case, They are used to the "Transfer->dial > number->Transfer" method of transferring calls. The > "Transfer->Blind->Dial Number" concept is foreign to them so out of > force of habit they use the former method for transferring all calls. > > On 07/16/2010 06:53 PM, Paul Herron wrote: > > I concur with Matt's last two posts. I just tested attended transfer as > > described and had the same results -- fine to VM, failed to AA. Blind > > transfers are fine. Running 4.0.4 w/Polycom 650 3.1.3c split, BootROM > > 4.2.2 > > > > Of course, one might wonder why we need an attended transfer to AA > > (other than the user screwing-up and pressing the wrong button -- o.k., > > I guess that's a pretty good reason). > > > > I also just tested attended transfer of an external call (via SipBridge) > > and had the same results -- fine to VM, failed to AA. Blind is fine. > > > > -----Original Message----- > > From: Matthew Kitchin (public/usenet) [mailto:[email protected]] > > > > Sent: Friday, July 16, 2010 6:44 PM > > To: [email protected] > > Subject: Re: [sipx-users] Help with Patton gateway > > > > Attended for everything in what I described below. > > Blind is fine. > > > > On 7/16/2010 5:41 PM, Josh Patten wrote: > > > >> 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 > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > > ------------------------------------------------------------------------ > > > >>>>>>> > >>>>>>> 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/ > >>>>> > >>>>> > >>>>> > >>>> > >>>> > >>> _______________________________________________ > >>> 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/ > >> > >> > > > > > > _______________________________________________ > > 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/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.984.8431 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ Why do mathematicians always confuse Halloween and Christmas? Because 31 Oct = 25 Dec.
_______________________________________________ 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/
