Stop and ensure you are NOT using firmware later than 3.1.3Rev3. You say "latest", and 3.2 is known to have issues which polycom if working on.
IMO hardware gateways (I prefer patton over audiocodes), because they. Are much less dependent on moving parts, and are more reliable. I didn't know AC made a 16k dollar gateway. What you are trying to do works spendidly with a real PRI gateway. Tony ============================ Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net 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: sipx-users-boun...@list.sipfoundry.org <sipx-users-boun...@list.sipfoundry.org> To: sipx-users@list.sipfoundry.org <sipx-users@list.sipfoundry.org> Sent: Sat Nov 21 11:37:23 2009 Subject: [sipx-users] Hair Pinned Calls on Forwards to PSTN 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 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/
