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[jmillican at sentinelc...]

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


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[brian at freeswitch.org]

Its looking for extension 2002 in context default on FreeSWITCH and
one doesn't exist so you get the NO ROUTE message.

Add a route to map 2002 so that it points at the Asterisk box.

/b

On Oct 10, 2008, at 1:00 PM, John Millican wrote:

[jmillican at sentinelc...]

Brian West wrote:

[brian at freeswitch.org]

I'm going to guess you added it at the very bottom of the default.xml?

It needs to be above this line:


<X-PRE-PROCESS cmd="include" data="default/*.xml"/>




/b




On Oct 10, 2008, at 1:22 PM, John Millican wrote:

[jmillican at sentinelc...]

Yep that was it! Now I just need to add a matching gateway, as I am
getting the error no matching gateway found, which I "think" I can
figure out.
Thank you for such quick and accurate help.
JohnM

Brian West wrote:

[brian at freeswitch.org]

The new default configs in SVN trunk have a HUGE warning at the very
bottom along with more documentation. I highly recommend you check it
out.

http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/dialplan/default.xml?r=9935

/b

On Oct 10, 2008, at 1:46 PM, John Millican wrote:

[jmillican at sentinelc...]

Thanks, will do that right now.
JohnM

Brian West wrote: