Hello, I am a FreeSwitch newb but have been using asterisk for a while now. I have a project for which I think FreeSwitch will be the best answer, so I need to learn. Have been reading the docs and followed the example at: http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk when I call from a Polycom on the asterisk box to a polycom on the freeswitch box all is good. When id do the reverse I.E. call the ast polycom from the freeswitch polycom I get only the following in the freswitch CLI: 2008-10-10 13:33:24 [NOTICE] switch_channel.c:538 switch_channel_set_name() New Channel sofia/internal/1002@192.168.100.16 [521c96a2-5205-bf46-9f9f-31124757b0ef] 2008-10-10 13:33:24 [INFO] mod_dialplan_xml.c:228 dialplan_hunt() Processing John Millican->2002 in context default 2008-10-10 13:33:24 [NOTICE] switch_ivr.c:1098 switch_ivr_session_transfer() Transfer sofia/internal/1002@192.168.100.16 to enum[2002@default] 2008-10-10 13:33:24 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting 2008-10-10 13:33:24 [NOTICE] switch_core_state_machine.c:115 switch_core_standard_on_routing() Hangup sofia/internal/1002@192.168.100.16 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2008-10-10 13:33:24 [NOTICE] switch_core_session.c:812 switch_core_session_thread() Session 12 (sofia/internal/1002@192.168.100.16) Ended 2008-10-10 13:33:24 [NOTICE] switch_core_session.c:814 switch_core_session_thread() Close Channel sofia/internal/1002@192.168.100.16 [CS_HANGUP] It would seem that the line: 2008-10-10 13:33:24 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting is telling me my problem but I do not yet know why freeswitch does not have a route. I am certain that I have not correctly set the dial plan but haven't a clue what to look at. Both machines are on the 192.168.100.0 net, firewall is off on both the freeswitch box which is running on a VMware installation of WinXP SP3 and the asterisk box. I am using the default configs with the additions per the above page. I did have to change the following from the defaults in vars.xml to get 2 way audio when I call from asterisk to freeswitch: <X-PRE-PROCESS cmd="set" data="bind_server_ip=192.168.100.16"/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=192.168.100.16"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=192.168.100.16"/> Any ideas? Is there something else I need to post to help decipher what I have done wrong or have not yet done? Thanks, JohnM