Validating idenity linksys
If you are having issues with proper codec selection, make sure your XML IP phone configuration contains: For Cisco SPA 5xx/3xx phones, I had to do quite a bit of NAT setup. I also found on many routers, that NAT requires unique local SIP ports.Login to phone through web interface as ADMIN default password is 456. On each extension tab (Ext 1, Ext 2, ...) that you register with Free Switch, scroll down to the SIP Settings and set the SIP Port to a Unique Number (at least unique on this phone).YB3.bin) The Cisco UC520 is a fairly flexible PBX geared for small businesses; and the Cisco IOS SIP stack it uses seems to be reasonably compatible with Free SWITCH without too much messing around.Cisco Integrated Services Routers, such as the 1861, 2800, 3800 can also connect to Free SWITCH using Unified Call Manager Express as PBX or using Cisco Unified Border Element as a session border controller.
Firmware versions up through 8.5.3 can be found here.
Audio Codes are willing to make changes necessary for Freeswitch support (more details at the phone specific details).
The 310HD model is an entry level phone intended for the simple user and is ready for production use.
Enabling STUN support may also help on the newer firmwares.
A lot of users have complained about presence not working properly with Aastra phones.
Currently, changing the transport from UDP to TCP seems to resolve these issues.