[Freeswitch-users] Newb question on dialplan config

John Millican jmillican at sentinelcommunications.com
Fri Oct 10 11:00:15 PDT 2008

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:


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 at
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 at to enum[2002 at 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 at [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 at Ended
2008-10-10 13:33:24 [NOTICE] switch_core_session.c:814
switch_core_session_thread() Close Channel
sofia/internal/1002 at [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 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="/>

  <X-PRE-PROCESS cmd="set" data="external_rtp_ip="/>

  <X-PRE-PROCESS cmd="set" data="external_sip_ip="/>

Any ideas? Is there something else I need to post to help decipher what
I have done wrong or have not yet done?


More information about the FreeSWITCH-users mailing list