On Sat, Nov 21, 2009 at 11:37 AM, Gabe Casey <gca...@franklinamerican.com> wrote: > Guys I have tried and tried to find a reasonable solution to a PRI media > gateway in the OS realm Asterisk Freeswitch ect but still can not find a > solution to forwards to mobile devices via a users forwarding options as > well as unpredictable assisted transfers > > Note there is no nat in this Design and most features work as expected > except for Assisted Transfers and Call forwarding > SipXecs 4.0.4 > Using all polycom 450 and 650 with latest firmware
What is the level of your firmware? Please see note on Polycom here: http://sipx-wiki.calivia.com/index.php/SIP_Trunking_with_sipXecs:_Overview_and_Configuration#How_to_configure_sipXbridge Please see the troubleshooting section on the same page. I need more information to provide you a better diagnosis - for example, a sipx-snapshot would be of help. Regards, Ranga > Tested with Asterisk 1.6 / freeswitch 1.0.4 > > Case 1: Unmanaged Gateway toward asterisk or freeswitch in which the user > has set a simultaneous ring on his/her exten as well as a mobile device > Polycom rings one time and call proceeds to Mobile with no audio issues > however this is not the expected outcome > Assisted Transfers proceed until the called party attempts to pickup and the > call is dropped > > Case 2: Sip Trunk / SipXBridge > Polycom and Mobile ring at same time however pickup on mobile, yields a > cancel from sipxecs which disconnects the Call. > In a followme scenario in which the call is passed to mobile after Exten > fails to pick up the call has one way audio in which the mobile device can > not TX > > Both asterisk and freeswitch are both respecting the signaling sent via > sipxecs here so i don't how far i am going to get trying to get a response > from those groups. I have posted this issue before and was hopeful for the > changes in 4.04 however testing last night gives the same results. This > issue is keeping me from large deployment. I am using Sangoma A108D 's and > am not thrilled about the suggestions that I need to purchase a 16K audio > codes gateway to fix this. > > Can anyone advise. Does Nortel have support at this level ? > I can provide traces for these calls if need be however i have posted them > before and many people already seem to know exactly what im talking about. > > I love this product and am very excited about DimDim integration and Zimbra > Mash Ups but this is a basic scenario in my mind that needs some > documentation (happy to write it when i find a working config) > > Warm Regards > > Gabrial Casey > Telecommunications > Franklin American Mortgage Company > Direct:615-468-2945 > Cell: 615-693-2833 > Email:gca...@franklinamerican.com > > > _______________________________________________ > sipx-users mailing list sipx-users@list.sipfoundry.org > 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/ > -- M. Ranganathan _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org 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/
